From helmut.kuper at ewetel.de  Thu Oct  1 01:52:59 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Thu, 01 Oct 2009 10:52:59 +0200
Subject: [Freeswitch-users] Problem with subscription expire
Message-ID: <4AC46DEB.3090506@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

it seems exired subsciptions are never cleared in FS.

A look into sofia_presence.c confirms explains this

/* negative in exptime means keep bumping up sub time to avoid a snafu
where every device has it's own rules about subscriptions
                           that somehow barely resemble the RFC not that
I blame them because the RFC MAY be amibiguous and SHOULD be deleted.
                           So to avoid the problem we keep resetting the
expiration date of the subscription so it never expires.

                           Eybeam completely ignores this option and
most other subscription-state: directives from rfc3265 and still expires.
                           Polycom is happy to keep upping the
subscription expiry back to the original time on each new notify.
                           The rest ... who knows...?

                        */

For some reasons subscriptions created by Snom phones are filling up the
sip_subscriptions table over time. This leads to some kind of DOS by FS
against the subscribing phone ... The subscribtions are differentiate by
call-id. This can be explained by RFC 3842 chapter 3.6 where expired
subscriptions must be renewed with a NEW call-id. Because there is no
hint about unsubscribing the old subscription I guess the clean up
process has to be done by FS.

Any way to get FS to do this job? Since there is no creation date or
expire value which represents the expire as a timestamp I have no way to
clean up the table manually via sql and cronjob - except cleaning the
whole table ...


A further (but background) question is, why do the subscriptions expire
in snom phones at all ...



regards
helmut
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From nagalenoj at gmail.com  Thu Oct  1 06:50:00 2009
From: nagalenoj at gmail.com (Nagalenoj)
Date: Thu, 1 Oct 2009 06:50:00 -0700 (PDT)
Subject: [Freeswitch-users]  Listening to a connected call [barge in]
Message-ID: <25696889.post@talk.nabble.com>


In ES outbound, I need to do the following, 
* A calls 2000(FS ES outbound extension)
* In the script, It'll answer the call, play some files and get the reply
from A(as voice).
* Simultaneously(when doing the above), the script has to call B.
* When B attends the call, B has to listen to the live conversation between
2000 & A.

How should I do.?

I've tried this with async mode and by listening to the events. But I
couldn't do it. Help me to do this..

Regards,
Nagalenoj
-- 
View this message in context: http://www.nabble.com/Listening-to-a-connected-call--barge-in--tp25696889p25696889.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From vinuth.madinur at gmail.com  Thu Oct  1 07:03:19 2009
From: vinuth.madinur at gmail.com (Vinuth Madinur)
Date: Thu, 1 Oct 2009 19:33:19 +0530
Subject: [Freeswitch-users] Listening to a connected call [barge in]
In-Reply-To: <25696889.post@talk.nabble.com>
References: <25696889.post@talk.nabble.com>
Message-ID: <910309030910010703j151d71v3c9e5700a6355939@mail.gmail.com>

Hi,
Use the eavesdrop command.
Just supply it with the call UUID and the extension of B.
Wiki has more details.

Thanks,
Vinuth.


On Thu, Oct 1, 2009 at 7:20 PM, Nagalenoj <nagalenoj at gmail.com> wrote:

>
> In ES outbound, I need to do the following,
> * A calls 2000(FS ES outbound extension)
> * In the script, It'll answer the call, play some files and get the reply
> from A(as voice).
> * Simultaneously(when doing the above), the script has to call B.
> * When B attends the call, B has to listen to the live conversation between
> 2000 & A.
>
> How should I do.?
>
> I've tried this with async mode and by listening to the events. But I
> couldn't do it. Help me to do this..
>
> Regards,
> Nagalenoj
> --
> View this message in context:
> http://www.nabble.com/Listening-to-a-connected-call--barge-in--tp25696889p25696889.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sicfslist at gmail.com  Thu Oct  1 07:27:47 2009
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Thu, 1 Oct 2009 09:27:47 -0500
Subject: [Freeswitch-users] Dialplan Issue
Message-ID: <35b355e90910010727v2efcbff6p97a0a883857818e4@mail.gmail.com>

Hello:

I asked this on IRC yesterday and I think I confused everyone involved.  So
I apologize in advance here for reposting the question and if I wasted
anyone's time.

So here is the issue I'm having.  I'm trying to use FS as a redirect server
(specifically to serve up LNP queries via 302 redirects).  But I'm having an
issue where based on the string in the dialplan FS will respond with a 500
internal error message instead of a 300 redirect.

The call flow should be this:
   -- remote party sends an Invite to my FS instance
   -- FS should respond with a 302

The following works as expected (FS will send a 302 when it receives an
Invite):

<action application="redirect" data="sip:${destination_number}@
${network_addr};rn=${rn};npdi=yes"/>

However if I do this (which is the way the response should look) FS will
respond with a 500 internal server error:

<action application="redirect"
data="sip:${destination_number};rn=${rn};npdi=yes@${network_addr}"/>

So the issue is the placement of the user params .... if they are before the
@ FS will send a 500 internal server error ... if they are after the @ FS
will send a 302.  Unfortunately placing the user params after the @ doesn't
quite conform to the way other devices expect to receive the 302 for this
application.

Any help would be greatly appreciated.

Shelby

PS ... hats off to the author of mod_memcache ... that is extremely useful!
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From testa at voicetechnology.com.br  Thu Oct  1 07:35:24 2009
From: testa at voicetechnology.com.br (Fernando Testa)
Date: Thu, 1 Oct 2009 11:35:24 -0300
Subject: [Freeswitch-users] REGISTER fails with 407 after minutes of
	success register
In-Reply-To: <7B7B543E-ACC9-4D00-A246-6C8D2608B960@freeswitch.org>
References: <9cb0e15e0909291728y2d75cea0k96547ea727d3dff9@mail.gmail.com>
	<9F9D51D2-2EBE-441D-BABF-6CA1DE8A9372@freeswitch.org>
	<9cb0e15e0909300506g6306d86dg39ade3fafa3ae750@mail.gmail.com>
	<7B7B543E-ACC9-4D00-A246-6C8D2608B960@freeswitch.org>
Message-ID: <9cb0e15e0910010735q3364cca8m8333740b1fd1a31@mail.gmail.com>

In the link below you have the entire SIP trace from system startup until
start receiving this annoying 407 Proxy Auth Required, preventing FS to
register successfully on the Ericsson Pabx.You can notice multiple
registrations from named ericsson_1050 to ericsson_1064 that starts failing
after ~50 minutes after the boot. Issuing a 'sofia external profile restart'
solves the registration problems.
Brian, thanks for reply, but I really didn't get your point.
Thank you, I apreciate any help.

http://dl.getdropbox.com/u/410277/sip.log.gz

On Wed, Sep 30, 2009 at 10:33 AM, Brian West <brian at freeswitch.org> wrote:

> I don't see a challenge in your 407 so how can we answer properly
> against the far end if they don't challenge us?
>
> /b
>
> On Sep 30, 2009, at 7:06 AM, Fernando Testa wrote:
>
> > Brian,
> >
> > Thanks for the reply. The SIP trace is mixed with the log (+ sofia
> > loglevel all 9) on the pastebin I mention on the previous email
> > (http://pastebin.freeswitch.org/10517 ). That log is from FS 1.0.4.
> >
> >
> > On Tue, Sep 29, 2009 at 10:13 PM, Brian West <brian at freeswitch.org>
> > wrote:
> >> I need the sip trace.
> >>
> >> /b
> >>
> >> On Sep 29, 2009, at 7:28 PM, Fernando Testa wrote:
> >>
> >>> Hi all,
> >>>
> >>> I have a FS that registers on an Ericsson pabx as gateway under
> >>> sip_external.
> >>> This gateway start registering on the Ericsson ok, but after a
> >>> while,
> >>> around 50mins, it fails with the logs below.
> >>> If I hit *sofia profile external restart* on fs_cli then the gateway
> >>> returns to register with success (that means, we get 200 OK from
> >>> Ericsson).
> >>> This happens with FS 1.0.4 release tarball, and trunk r15011.
> >>> I found similar situations on these links, but not actually found a
> >>> solution.
> >>> Any help is very welcome.
> >>>
> >>> *OS*
> >>> CentOS 5.3 x86_64
> >>> 4Gb RAM
> >>>
> >>> log at http://pastebin.freeswitch.org/10517
> >>>
> >>> conf/sip_profiles/external/ericsson.xml:
> >>>  <gateway name="ericsson_1064">
> >>>    <param name="username" value="1064"/>
> >>>    <param name="realm" value="10.227.0.3"/>
> >>>    <param name="password" value="5080"/>
> >>>    <param name="register" value="true"/>
> >>>    <param name="register-proxy" value="10.227.0.3"/>
> >>>    <param name="expire-seconds" value="60"/>
> >>>    <param name="retry-seconds" value="10"/>
> >>>  </gateway>
> >>>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Fernando Gregianin Testa
Voice Technology Ltda
+55 11 35882166
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From brian at freeswitch.org  Thu Oct  1 07:42:23 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 1 Oct 2009 09:42:23 -0500
Subject: [Freeswitch-users] Dialplan Issue
In-Reply-To: <35b355e90910010727v2efcbff6p97a0a883857818e4@mail.gmail.com>
References: <35b355e90910010727v2efcbff6p97a0a883857818e4@mail.gmail.com>
Message-ID: <3DA34BA8-5E67-4A0C-A163-B72EF5F8638F@freeswitch.org>


On Oct 1, 2009, at 9:27 AM, Shelby Ramsey wrote:

> <action application="redirect" data="sip:${destination_number};rn=${rn};npdi=yes@${network_addr} 
> "/>

This will produce an INVALID sip uri... You can not feed this to sofia  
it'll get PISSED.

Its missing the host portion.

>
> So the issue is the placement of the user params .... if they are  
> before the @ FS will send a 500 internal server error ... if they  
> are after the @ FS will send a 302.  Unfortunately placing the user  
> params after the @ doesn't quite conform to the way other devices  
> expect to receive the 302 for this application.
>
> Any help would be greatly appreciated.
>
> Shelby
>
> PS ... hats off to the author of mod_memcache ... that is extremely  
> useful!

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From brian at freeswitch.org  Thu Oct  1 07:46:53 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 1 Oct 2009 09:46:53 -0500
Subject: [Freeswitch-users] REGISTER fails with 407 after minutes of
	success register
In-Reply-To: <9cb0e15e0910010735q3364cca8m8333740b1fd1a31@mail.gmail.com>
References: <9cb0e15e0909291728y2d75cea0k96547ea727d3dff9@mail.gmail.com>
	<9F9D51D2-2EBE-441D-BABF-6CA1DE8A9372@freeswitch.org>
	<9cb0e15e0909300506g6306d86dg39ade3fafa3ae750@mail.gmail.com>
	<7B7B543E-ACC9-4D00-A246-6C8D2608B960@freeswitch.org>
	<9cb0e15e0910010735q3364cca8m8333740b1fd1a31@mail.gmail.com>
Message-ID: <F5F435C3-9295-402D-9D65-A886232E7859@freeswitch.org>

Thanks for posting the logs... But I'm not going to spend the time to  
download it.. unzip it and look at it... I would rather just click a  
link with the logs in plain text and read them in my browser.

I'll do it now but next time lets not add steps to the process that  
are not needed.  This goes for Jira too don't upload zip files of text  
logs that just makes it harder for us to quickly help you.

This isn't going to help me much know why Sofia/FreeSWITCH isn't  
working.

sofia profile xxx siptrace on
press F8
sofia loglevel all 9

Then post that please.

/b

On Oct 1, 2009, at 9:35 AM, Fernando Testa wrote:

> In the link below you have the entire SIP trace from system startup  
> until start receiving this annoying 407 Proxy Auth Required,  
> preventing FS to register successfully on the Ericsson Pabx.
> You can notice multiple registrations from named ericsson_1050 to  
> ericsson_1064 that starts failing after ~50 minutes after the boot.  
> Issuing a 'sofia external profile restart' solves the registration  
> problems.
> Brian, thanks for reply, but I really didn't get your point.
> Thank you, I apreciate any help.
>
> http://dl.getdropbox.com/u/410277/sip.log.gz

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From anthony.minessale at gmail.com  Thu Oct  1 07:55:54 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 1 Oct 2009 09:55:54 -0500
Subject: [Freeswitch-users] Dialplan Issue
In-Reply-To: <35b355e90910010727v2efcbff6p97a0a883857818e4@mail.gmail.com>
References: <35b355e90910010727v2efcbff6p97a0a883857818e4@mail.gmail.com>
Message-ID: <191c3a030910010755l387faba8x1d8f9bbf51c8ffa1@mail.gmail.com>

you left too fast.

1) I told you if you put <> around the sip uri it will work.
2) I told you I added a patch in tree to add one for you if it's not
supplied so update to trunk.


On Thu, Oct 1, 2009 at 9:27 AM, Shelby Ramsey <sicfslist at gmail.com> wrote:

> Hello:
>
> I asked this on IRC yesterday and I think I confused everyone involved.  So
> I apologize in advance here for reposting the question and if I wasted
> anyone's time.
>
> So here is the issue I'm having.  I'm trying to use FS as a redirect server
> (specifically to serve up LNP queries via 302 redirects).  But I'm having an
> issue where based on the string in the dialplan FS will respond with a 500
> internal error message instead of a 300 redirect.
>
> The call flow should be this:
>    -- remote party sends an Invite to my FS instance
>    -- FS should respond with a 302
>
> The following works as expected (FS will send a 302 when it receives an
> Invite):
>
> <action application="redirect" data="sip:${destination_number}@
> ${network_addr};rn=${rn};npdi=yes"/>
>
> However if I do this (which is the way the response should look) FS will
> respond with a 500 internal server error:
>
> <action application="redirect"
> data="sip:${destination_number};rn=${rn};npdi=yes@${network_addr}"/>
>
> So the issue is the placement of the user params .... if they are before
> the @ FS will send a 500 internal server error ... if they are after the @
> FS will send a 302.  Unfortunately placing the user params after the @
> doesn't quite conform to the way other devices expect to receive the 302 for
> this application.
>
> Any help would be greatly appreciated.
>
> Shelby
>
> PS ... hats off to the author of mod_memcache ... that is extremely useful!
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From jerry.richards at teotech.com  Thu Oct  1 07:57:29 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Thu, 1 Oct 2009 07:57:29 -0700
Subject: [Freeswitch-users] FS Does Not Relay Presence PUBLISH To
	Subscribing Phones
Message-ID: <B4B4D2747EEC4852B41A7D8D2DA36006@greyhawk.tonecommander.com>


By the way, I see the following lines at the FS console, which might be a
clue as to why this is happening.  Could someone point me toward what might
cause this?  I set the "manage-presence" parameter to "true" in each XML
file where I saw it defined.

[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
   ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
   ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
   ...
[WARNING] sofia_presence.c:565  192.168.72.38 is an alias, skipping

 
Best Regards,
Jerry
 

-----Original Message-----
From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Wednesday, September 30, 2009 9:12 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones

I have two phones configured to subscribe to each other's presence status.
When I change the presence status in one phone, I see the SIP PUBLISH
message going to FS, but I don't see FS relaying that presence status to the
subscribing phone.  Does anyone know why?

Best Regards,
Jerry



From sicfslist at gmail.com  Thu Oct  1 08:59:53 2009
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Thu, 1 Oct 2009 10:59:53 -0500
Subject: [Freeswitch-users] Dialplan Issue
In-Reply-To: <3DA34BA8-5E67-4A0C-A163-B72EF5F8638F@freeswitch.org>
References: <35b355e90910010727v2efcbff6p97a0a883857818e4@mail.gmail.com>
	<3DA34BA8-5E67-4A0C-A163-B72EF5F8638F@freeswitch.org>
Message-ID: <35b355e90910010859t5c7cd401n7a33cfe7a13ec962@mail.gmail.com>

Brian,

Thanks for the info.  I guess I'll go read section 19.1 of RFC3261 again.  I
do think the above has a valid host portion (I don't think the port is
required).

I'm not so sure that putting params in the user portion of the uri is valid
(from the RFC it states sip:user:password at host:port;uri-parameters?headers).

The issue is that in the real world this is done all the time .... SIP is
fantastic :)

Shelby


On Thu, Oct 1, 2009 at 9:42 AM, Brian West <brian at freeswitch.org> wrote:

>
> On Oct 1, 2009, at 9:27 AM, Shelby Ramsey wrote:
>
> <action application="redirect" data="
> sip:${destination_number};rn=${rn};npdi=yes@${network_addr}"/>
>
>
> This will produce an INVALID sip uri... You can not feed this to sofia
> it'll get PISSED.
>
> Its missing the host portion.
>
>
> So the issue is the placement of the user params .... if they are before
> the @ FS will send a 500 internal server error ... if they are after the @
> FS will send a 302.  Unfortunately placing the user params after the @
> doesn't quite conform to the way other devices expect to receive the 302 for
> this application.
>
> Any help would be greatly appreciated.
>
> Shelby
>
> PS ... hats off to the author of mod_memcache ... that is extremely useful!
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sicfslist at gmail.com  Thu Oct  1 09:01:37 2009
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Thu, 1 Oct 2009 11:01:37 -0500
Subject: [Freeswitch-users] Dialplan Issue
In-Reply-To: <191c3a030910010755l387faba8x1d8f9bbf51c8ffa1@mail.gmail.com>
References: <35b355e90910010727v2efcbff6p97a0a883857818e4@mail.gmail.com>
	<191c3a030910010755l387faba8x1d8f9bbf51c8ffa1@mail.gmail.com>
Message-ID: <35b355e90910010901j63a36531m7550eba26564f01@mail.gmail.com>

Tony,

Once again ... you are the man!

I'll try this right now.

SDR
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From brian at freeswitch.org  Thu Oct  1 09:15:33 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 1 Oct 2009 11:15:33 -0500
Subject: [Freeswitch-users] Dialplan Issue
In-Reply-To: <35b355e90910010859t5c7cd401n7a33cfe7a13ec962@mail.gmail.com>
References: <35b355e90910010727v2efcbff6p97a0a883857818e4@mail.gmail.com>
	<3DA34BA8-5E67-4A0C-A163-B72EF5F8638F@freeswitch.org>
	<35b355e90910010859t5c7cd401n7a33cfe7a13ec962@mail.gmail.com>
Message-ID: <DB2DDCF0-3E7A-4EF9-B35A-E700676B3F0B@freeswitch.org>

I wouldn't go that far! :P  You might be able to get away with it on  
the patch tony wrote but not sure.

/b

On Oct 1, 2009, at 10:59 AM, Shelby Ramsey wrote:

> SIP is fantastic :)



From sicfslist at gmail.com  Thu Oct  1 09:18:01 2009
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Thu, 1 Oct 2009 11:18:01 -0500
Subject: [Freeswitch-users] Dialplan Issue
In-Reply-To: <35b355e90910010901j63a36531m7550eba26564f01@mail.gmail.com>
References: <35b355e90910010727v2efcbff6p97a0a883857818e4@mail.gmail.com>
	<191c3a030910010755l387faba8x1d8f9bbf51c8ffa1@mail.gmail.com>
	<35b355e90910010901j63a36531m7550eba26564f01@mail.gmail.com>
Message-ID: <35b355e90910010918w7a01c4d1l7bc8b7986d2a3434@mail.gmail.com>

Just to confirm ... works like a champ.

Thanks again!!!
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From anthony.minessale at gmail.com  Thu Oct  1 09:35:34 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 1 Oct 2009 11:35:34 -0500
Subject: [Freeswitch-users] FS Does Not Relay Presence PUBLISH To
	Subscribing Phones
In-Reply-To: <B4B4D2747EEC4852B41A7D8D2DA36006@greyhawk.tonecommander.com>
References: <B4B4D2747EEC4852B41A7D8D2DA36006@greyhawk.tonecommander.com>
Message-ID: <191c3a030910010935s69e65569v76f4d26503a94d96@mail.gmail.com>

which phone is it,
we tested it with eyebeam and it appears to work for us.


On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>
> By the way, I see the following lines at the FS console, which might be a
> clue as to why this is happening.  Could someone point me toward what might
> cause this?  I set the "manage-presence" parameter to "true" in each XML
> file where I saw it defined.
>
> [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal)
> [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
>   ...
> [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6)
> [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
>   ...
> [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external)
> [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
>   ...
> [WARNING] sofia_presence.c:565  192.168.72.38 is an alias, skipping
>
>
> Best Regards,
> Jerry
>
>
> -----Original Message-----
> From: Jerry Richards [mailto:jerry.richards at teotech.com]
> Sent: Wednesday, September 30, 2009 9:12 AM
> To: 'freeswitch-users at lists.freeswitch.org'
> Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones
>
> I have two phones configured to subscribe to each other's presence status.
> When I change the presence status in one phone, I see the SIP PUBLISH
> message going to FS, but I don't see FS relaying that presence status to
> the
> subscribing phone.  Does anyone know why?
>
> Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mike at van.lammeren.net  Thu Oct  1 09:45:23 2009
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Thu, 1 Oct 2009 12:45:23 -0400
Subject: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
In-Reply-To: <5d2828f0909290720y6bb4997uc4b9afba4b0fe74c@mail.gmail.com>
References: <5d2828f0909290720y6bb4997uc4b9afba4b0fe74c@mail.gmail.com>
Message-ID: <5d2828f0910010945q30ed76e2x50d6ab3b13bc8f88@mail.gmail.com>

Guess what? I have two FreeSWITCH servers working behind UltraMonkey, using
heartbeat and ldirectord for load-balancing, fail-over and high
availability! I'm probably not the first one to do it, but as near as Google
and I can tell, I'm the first one to write about it.
Here's how you can duplicate my setup:

1. Install Ubuntu Server 8 on four machines, either real or VM.
2. Compile and install FreeSWITCH v1.0.4 from source on two machines,
following these instructions:
http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start
<http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start>3. Configure both
FreeSWITCH boxes, and make sure they are both working.
4. Follow (most of) these instructions from Daniel Aliaman's blog. They were
written for Asterisk, but since a SIP connection is a SIP connection, most
of the document applies to FreeSWITCH:
http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf

<http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf>The one
problem I ran into was the IP address and port to which FreeSWITCH was
bound. The default is to use the primary address, which works great
out-of-the-box for everything else. When a client tried to register, all it
got back was an ICMP error -- Destination Unreachable, Port Unreachable.
That error is returned when no sockets are listening for UDP packets. To get
FreeSWITCH to listen for your Virtual IP, you need to set it in two places:

5. In /opt/freeswitch/conf/vars.xml, set "bind_server_ip".
6. In /opt/freeswitch/conf/sip_profiles/internal.xml, set "sip-ip".

That should do it! If you have any success, please report to this list.

Keep in mind that if you want to do something like conferencing between two
registered clients, then you have to deal with the fact that the clients may
or may not be on the same box.

Mike van Lammeren



On Tue, Sep 29, 2009 at 10:20 AM, Mike van Lammeren
<mike at van.lammeren.net>wrote:

>
> On Mon, Sep 28, 2009 at 9:05 PM, "Even Andr? Fiskvik" <grevenx at me.com
> > wrote:
>
>> From: "Even Andr? Fiskvik" <grevenx at me.com>
>> To: freeswitch-users at lists.freeswitch.org
>> Date: Mon, 28 Sep 2009 22:52:13 +0200
>> Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
>> I have been working with a similar setup myself, but for some reason I
>> ended up ditching theUltraMonkey setup because I just couldn't get it to
>> work right.
>>
>> It's been quite a while since my effort, so I don't remember what the
>> exact issue was.
>> I got registrations to work, but had some other sip-dialog issues.
>>
>> We have since then changed over to running OpenSIPs as a loadbalancer in
>> front of
>> multiple FreeSWITCH instances. This setup is still in testing, but
>> seemlingy works fine
>> (and if it doesn't, it's my own fault for writing a bad opensips config).
>>
>> After we have done some more testing I can create a wiki-page with config
>> details.
>>
>>
>> Best regards,
>> Even Andr?
>>
>>
> Thanks, Even, that would be great! I might have to give up on the
> ultramonkey solution, since I can't find anyone who has made it work. It's
> too bad, because it would fit well with the rest of our architecture.
>
> Mike van Lammeren
>
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From Russell.Mosemann at cune.org  Thu Oct  1 07:37:10 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Thu, 1 Oct 2009 14:37:10 -0000
Subject: [Freeswitch-users] Connecting FS to Hicom 300
Message-ID: <20091001143710.AE8063C4C4D@mail.cune.org>

We have connected FS to a Siemens Hicomm 300. As you might guess, it's
not working right. Here is what we are working with.

Dell 1750 (dual socket, dual core Xeon 2.8GHz)
Debian 5
FS (15029), OpenZAP (without libpri)
TE110P T1 card (Zaptel driver)
Handles 71xx extensions

Siemens Hicom 300
TMDN64P T1 card
Handles 74xx extensions

We are pretty much using the stock FS configuration, yet, because we're
trying to get this to work. I have configured OpenZAP and the associated
files like the examples on the wiki (see below) to work with a PRI T1.
There are 23 B channels and 1 D channel. The Zaptel side looks fine.
OpenZAP is able to open the channels when FS boots. So far, so good.

When a call is made from 74xx (Rolmphone 624) to 71xx (X-Lite 4.0 beta
from CounterPath on an office PC), X-Lite rings. The call can be
answered, and the conversation sounds fine. That means the routing,
registration and authorization are working on the network between X-Lite
and FS. It also means that FS is able to communicate with the Hicom over
the T1. Great.

When the caller presses the transfer button on the 74xx phone, the Hicom
sends a message over the D channel, and the call is disconnected
(watching with fs_cli). As best I can interpret the bytes in the message,
the Hicom sends a disconnect message when 74xx presses the transfer key.

In order to call 74xx, I created dialplan/default/02_hicom.xml. The
contents are

<include>
  <extension name="hicom">
    <condition field="destination_number" expression="^(74\d{2})$">
      <action application="bridge" data="openzap/1/a/$1"/>
    </condition>
  </extension>
</include>

If a call is made from 71xx to 74xx, the Hicom forwards the call to the
switchboard with "7100->7445 connection not possible" (or whatever
extensions) in the switchboard display.

1. Are these issues related to the way I have configured FS?

The Hicom is maintained by the local phone company. I do not have access
to view or configure the T1 card on the Hicom. According to the phone
guy, there isn't anything else that needs to be configured on the Hicom.
He believes that if 74xx can call 71xx, then 71xx should be able to call
74xx.

I suspect that something more needs to be done on the Hicom in order to
accept calls from FS and bridge/transfer them to a local extension on the
Hicom. It's as if the Hicom doesn't know how or is not permitted to route
incoming calls on the T1 to local extensions. I have no way to know,
though. I'm hoping someone else has connected FS to a Hicom 300 and can
provide configuration details. If I could tell the phone guy something
like, "You need to look at <this>," that would help him out.

2. Should I receive CID/ANI from the Hicom?

X-Lite displays "OpenZAP" as the call and "1" as Other when the call
comes in, which is the information for the endpoint. Is there something I
need to do in the FS configuration to capture CID/ANI information from
the Hicom and make it available (or is it not being provided by the Hicom)?

3. When dialing from the Rolmphone is there a way for FS to send the
called name back to the Hicom for it to appear in the display?

When dialing 74xx to 74xx, of course, it shows the called number and name
in the display. We also have a HiPath 4000 connected to the Hicom 300.
When dialing an extension on the HiPath from the Hicom, the HiPath ships
the called name back to the Hicom for display on the phone. It would be
nice to do that from FS.

Let me know if you need additional information. Thanks for any pointers
or insight as to how things work.

-- 
Russell Mosemann


openzap.conf
[span zt PRI_1]
name => OpenZAP
number => 1
trunk_type => t1
b-channel => 1-23
d-channel => 24

zt.conf
[defaults]
codec_ms => 20
wink_ms => 150
flash_ms => 750
echo_cancel_level => 64
rxgain => 0.0
txgain => 0.0

openzap.conf
<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    <param name="debug" value="0"/>
    <!--<param name="hold-music" value="$${moh_uri}"/>-->
    <!--<param name="enable-analog-option" value="call-swap"/>-->
    <!--<param name="enable-analog-option" value="3-way"/>-->
  </settings>
   <pri_spans>
     <span name="PRI_1">
       <param name="q921loglevel" value="alert"/>
       <param name="q931loglevel" value="alert"/>
       <param name="mode" value="user"/>
       <param name="dialect" value="national"/>
       <param name="dialplan" value="XML"/>
       <param name="context" value="public"/>
     </span>
   </pri_spans>
</configuration>

zaptel.conf
# Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" (MASTER)
span=1,1,0,esf,b8zs
# termtype: te
bchan=1-23
dchan=24

# Global data
loadzone        = us
defaultzone     = us


________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From dmitry.bely at gmail.com  Thu Oct  1 10:01:10 2009
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Thu, 1 Oct 2009 21:01:10 +0400
Subject: [Freeswitch-users] How to limit the number of incoming+outgoing
	calls via specific gateway?
Message-ID: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>

My SIP provider allows only one call (incoming or outgoing) via one
SIP account. For FreeSWITCH I have configured it as public DID
extension and outgoing gateway. Now I would like to transfer to
another gw (or generate "limit exceded") when one tries to place an
outgoing call while incoming call is in progress. How tho do that?
Limiting the number of outgoing calls is easy (mod_limit), but how to
take into account incoming one?

- Dmitry Bely


From jerry.richards at teotech.com  Thu Oct  1 10:28:40 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Thu, 1 Oct 2009 10:28:40 -0700
Subject: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
	ToSubscribing Phones
In-Reply-To: <191c3a030910010935s69e65569v76f4d26503a94d96@mail.gmail.com>
References: <B4B4D2747EEC4852B41A7D8D2DA36006@greyhawk.tonecommander.com>
	<191c3a030910010935s69e65569v76f4d26503a94d96@mail.gmail.com>
Message-ID: <C92528AAF40D4D41B0F6B69A4CE644F5@greyhawk.tonecommander.com>

I am using two Bria Professional Version 2.5.4 Build 54835 softphones.
 
Thanks,
Jerry


  _____  

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Thursday, October 01, 2009 9:36 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
ToSubscribing Phones


which phone is it,
we tested it with eyebeam and it appears to work for us.



On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards <jerry.richards at teotech.com>
wrote:



By the way, I see the following lines at the FS console, which might be a
clue as to why this is happening.  Could someone point me toward what might
cause this?  I set the "manage-presence" parameter to "true" in each XML
file where I saw it defined.

[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[WARNING] sofia_presence.c:565  192.168.72.38 is an alias, skipping


Best Regards,
Jerry



-----Original Message-----
From: Jerry Richards [mailto:jerry.richards at teotech.com]
Sent: Wednesday, September 30, 2009 9:12 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones

I have two phones configured to subscribe to each other's presence status.
When I change the presence status in one phone, I see the SIP PUBLISH
message going to FS, but I don't see FS relaying that presence status to the
subscribing phone.  Does anyone know why?

Best Regards,
Jerry


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:213-799-1400


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From raffaele.p.guidi at gmail.com  Thu Oct  1 11:02:25 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Thu, 1 Oct 2009 20:02:25 +0200
Subject: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
In-Reply-To: <5d2828f0910010945q30ed76e2x50d6ab3b13bc8f88@mail.gmail.com>
References: <5d2828f0909290720y6bb4997uc4b9afba4b0fe74c@mail.gmail.com>
	<5d2828f0910010945q30ed76e2x50d6ab3b13bc8f88@mail.gmail.com>
Message-ID: <b0a8792f0910011102x691ba8a9s6a06cfd42483c935@mail.gmail.com>

And, should someone succed replicating this setup, consider writing about it
on the wiki :)

On Thu, Oct 1, 2009 at 18:45, Mike van Lammeren <mike at van.lammeren.net>wrote:

> Guess what? I have two FreeSWITCH servers working behind UltraMonkey, using
> heartbeat and ldirectord for load-balancing, fail-over and high
> availability! I'm probably not the first one to do it, but as near as Google
> and I can tell, I'm the first one to write about it.
> Here's how you can duplicate my setup:
>
> 1. Install Ubuntu Server 8 on four machines, either real or VM.
> 2. Compile and install FreeSWITCH v1.0.4 from source on two machines,
> following these instructions:
> http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start
>  <http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start>3. Configure both
> FreeSWITCH boxes, and make sure they are both working.
> 4. Follow (most of) these instructions from Daniel Aliaman's blog. They
> were written for Asterisk, but since a SIP connection is a SIP connection,
> most of the document applies to FreeSWITCH:
> http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf
>
> <http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf>The one
> problem I ran into was the IP address and port to which FreeSWITCH was
> bound. The default is to use the primary address, which works great
> out-of-the-box for everything else. When a client tried to register, all it
> got back was an ICMP error -- Destination Unreachable, Port Unreachable.
> That error is returned when no sockets are listening for UDP packets. To get
> FreeSWITCH to listen for your Virtual IP, you need to set it in two places:
>
> 5. In /opt/freeswitch/conf/vars.xml, set "bind_server_ip".
> 6. In /opt/freeswitch/conf/sip_profiles/internal.xml, set "sip-ip".
>
> That should do it! If you have any success, please report to this list.
>
> Keep in mind that if you want to do something like conferencing between two
> registered clients, then you have to deal with the fact that the clients may
> or may not be on the same box.
>
> Mike van Lammeren
>
>
>
> On Tue, Sep 29, 2009 at 10:20 AM, Mike van Lammeren <mike at van.lammeren.net
> > wrote:
>
>>
>> On Mon, Sep 28, 2009 at 9:05 PM, "Even Andr? Fiskvik" <grevenx at me.com
>> > wrote:
>>
>>> From: "Even Andr? Fiskvik" <grevenx at me.com>
>>> To: freeswitch-users at lists.freeswitch.org
>>> Date: Mon, 28 Sep 2009 22:52:13 +0200
>>> Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
>>> I have been working with a similar setup myself, but for some reason I
>>> ended up ditching theUltraMonkey setup because I just couldn't get it to
>>> work right.
>>>
>>> It's been quite a while since my effort, so I don't remember what the
>>> exact issue was.
>>> I got registrations to work, but had some other sip-dialog issues.
>>>
>>> We have since then changed over to running OpenSIPs as a loadbalancer in
>>> front of
>>> multiple FreeSWITCH instances. This setup is still in testing, but
>>> seemlingy works fine
>>> (and if it doesn't, it's my own fault for writing a bad opensips config).
>>>
>>> After we have done some more testing I can create a wiki-page with config
>>> details.
>>>
>>>
>>> Best regards,
>>> Even Andr?
>>>
>>>
>> Thanks, Even, that would be great! I might have to give up on the
>> ultramonkey solution, since I can't find anyone who has made it work. It's
>> too bad, because it would fit well with the rest of our architecture.
>>
>> Mike van Lammeren
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From grevenx at me.com  Thu Oct  1 11:12:59 2009
From: grevenx at me.com (=?iso-8859-1?Q?Even_Andr=E9_Fiskvik?=)
Date: Thu, 01 Oct 2009 20:12:59 +0200
Subject: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
In-Reply-To: <5d2828f0910010945q30ed76e2x50d6ab3b13bc8f88@mail.gmail.com>
References: <5d2828f0909290720y6bb4997uc4b9afba4b0fe74c@mail.gmail.com>
	<5d2828f0910010945q30ed76e2x50d6ab3b13bc8f88@mail.gmail.com>
Message-ID: <32C47FA5-774E-4584-890B-0A066F5CFE14@me.com>

That's very cool Mike!

I'm going to try to configure four boxes with this as well (Btw, did  
you use physical hardware or virtualization?)
and see how it goes. I followed Daniel Aliaman's blog as well, but I  
can try it again with the tips
you provided on FreeSWITCH config to see if I can get it working  
properly this time.
We did the setup on CentOS, but I wouldn't think that would be any  
issue.

Perhaps you or we could write up a complete guide about this on the  
wiki since this is an scenario
commonly used? Also it would be great if we could outline possible  
issues (and even better solutions)
to this kind of setup with regards to stuff like conferencing,  
bridging between registered users and presence.


Best regards,
Even Andr?


On 1. okt. 2009, at 18.45, Mike van Lammeren wrote:

> Guess what? I have two FreeSWITCH servers working behind  
> UltraMonkey, using heartbeat and ldirectord for load-balancing, fail- 
> over and high availability! I'm probably not the first one to do it,  
> but as near as Google and I can tell, I'm the first one to write  
> about it.
>
> Here's how you can duplicate my setup:
>
> 1. Install Ubuntu Server 8 on four machines, either real or VM.
> 2. Compile and install FreeSWITCH v1.0.4 from source on two  
> machines, following these instructions: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start
> 3. Configure both FreeSWITCH boxes, and make sure they are both  
> working.
> 4. Follow (most of) these instructions from Daniel Aliaman's blog.  
> They were written for Asterisk, but since a SIP connection is a SIP  
> connection, most of the document applies to FreeSWITCH:
> http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf
>
> The one problem I ran into was the IP address and port to which  
> FreeSWITCH was bound. The default is to use the primary address,  
> which works great out-of-the-box for everything else. When a client  
> tried to register, all it got back was an ICMP error -- Destination  
> Unreachable, Port Unreachable. That error is returned when no  
> sockets are listening for UDP packets. To get FreeSWITCH to listen  
> for your Virtual IP, you need to set it in two places:
>
> 5. In /opt/freeswitch/conf/vars.xml, set "bind_server_ip".
> 6. In /opt/freeswitch/conf/sip_profiles/internal.xml, set "sip-ip".
>
> That should do it! If you have any success, please report to this  
> list.
>
> Keep in mind that if you want to do something like conferencing  
> between two registered clients, then you have to deal with the fact  
> that the clients may or may not be on the same box.
>
> Mike van Lammeren
>
>
>
> On Tue, Sep 29, 2009 at 10:20 AM, Mike van Lammeren <mike at van.lammeren.net 
> > wrote:
>
> On Mon, Sep 28, 2009 at 9:05 PM, "Even Andr? Fiskvik"  
> <grevenx at me.com> wrote:
> From: "Even Andr? Fiskvik" <grevenx at me.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Mon, 28 Sep 2009 22:52:13 +0200
> Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with  
> Ultramonkey
> I have been working with a similar setup myself, but for some reason  
> I ended up ditching the
> UltraMonkey setup because I just couldn't get it to work right.
>
> It's been quite a while since my effort, so I don't remember what  
> the exact issue was.
> I got registrations to work, but had some other sip-dialog issues.
>
> We have since then changed over to running OpenSIPs as a  
> loadbalancer in front of
> multiple FreeSWITCH instances. This setup is still in testing, but  
> seemlingy works fine
> (and if it doesn't, it's my own fault for writing a bad opensips  
> config).
>
> After we have done some more testing I can create a wiki-page with  
> config details.
>
>
> Best regards,
> Even Andr?
>
>
> Thanks, Even, that would be great! I might have to give up on the  
> ultramonkey solution, since I can't find anyone who has made it  
> work. It's too bad, because it would fit well with the rest of our  
> architecture.
>
> Mike van Lammeren
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From jerry.richards at teotech.com  Thu Oct  1 11:29:20 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Thu, 1 Oct 2009 11:29:20 -0700
Subject: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
	ToSubscribing Phones
References: <B4B4D2747EEC4852B41A7D8D2DA36006@greyhawk.tonecommander.com>
	<191c3a030910010935s69e65569v76f4d26503a94d96@mail.gmail.com> 
Message-ID: <9B5214D274044021A57C6857844C10BA@greyhawk.tonecommander.com>

If you have time to take a look, I could put a trace in the pastebin?
 
Jerry


  _____  

From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Thursday, October 01, 2009 10:29 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
ToSubscribing Phones


I am using two Bria Professional Version 2.5.4 Build 54835 softphones.
 
Thanks,
Jerry


  _____  

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Thursday, October 01, 2009 9:36 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
ToSubscribing Phones


which phone is it,
we tested it with eyebeam and it appears to work for us.



On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards <jerry.richards at teotech.com>
wrote:



By the way, I see the following lines at the FS console, which might be a
clue as to why this is happening.  Could someone point me toward what might
cause this?  I set the "manage-presence" parameter to "true" in each XML
file where I saw it defined.

[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[WARNING] sofia_presence.c:565  192.168.72.38 is an alias, skipping


Best Regards,
Jerry



-----Original Message-----
From: Jerry Richards [mailto:jerry.richards at teotech.com]
Sent: Wednesday, September 30, 2009 9:12 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones

I have two phones configured to subscribe to each other's presence status.
When I change the presence status in one phone, I see the SIP PUBLISH
message going to FS, but I don't see FS relaying that presence status to the
subscribing phone.  Does anyone know why?

Best Regards,
Jerry


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:213-799-1400


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From anthony.minessale at gmail.com  Thu Oct  1 12:14:14 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 1 Oct 2009 14:14:14 -0500
Subject: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
In-Reply-To: <32C47FA5-774E-4584-890B-0A066F5CFE14@me.com>
References: <5d2828f0909290720y6bb4997uc4b9afba4b0fe74c@mail.gmail.com>
	<5d2828f0910010945q30ed76e2x50d6ab3b13bc8f88@mail.gmail.com>
	<32C47FA5-774E-4584-890B-0A066F5CFE14@me.com>
Message-ID: <191c3a030910011214i3a9eaad1xf242ecf19dc80da3@mail.gmail.com>

can we do it without advertising to use ubuntu =D
We don't like encouraging our users to use bleeding edge OS for our own
sanity with debugging.
Not to say you are not allowed to I just don't want to encourage it =p



On Thu, Oct 1, 2009 at 1:12 PM, Even Andr? Fiskvik <grevenx at me.com> wrote:

> That's very cool Mike!
> I'm going to try to configure four boxes with this as well (Btw, did you
> use physical hardware or virtualization?)
> and see how it goes. I followed Daniel Aliaman's blog as well, but I can
> try it again with the tips
> you provided on FreeSWITCH config to see if I can get it working properly
> this time.
> We did the setup on CentOS, but I wouldn't think that would be any issue.
>
> Perhaps you or we could write up a complete guide about this on the wiki
> since this is an scenario
> commonly used? Also it would be great if we could outline possible issues
> (and even better solutions)
> to this kind of setup with regards to stuff like conferencing, bridging
> between registered users and presence.
>
>
> Best regards,
> Even Andr?
>
>
> On 1. okt. 2009, at 18.45, Mike van Lammeren wrote:
>
> Guess what? I have two FreeSWITCH servers working behind UltraMonkey, using
> heartbeat and ldirectord for load-balancing, fail-over and high
> availability! I'm probably not the first one to do it, but as near as Google
> and I can tell, I'm the first one to write about it.
> Here's how you can duplicate my setup:
>
> 1. Install Ubuntu Server 8 on four machines, either real or VM.
> 2. Compile and install FreeSWITCH v1.0.4 from source on two machines,
> following these instructions:
> http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start
>  <http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start>3. Configure both
> FreeSWITCH boxes, and make sure they are both working.
> 4. Follow (most of) these instructions from Daniel Aliaman's blog. They
> were written for Asterisk, but since a SIP connection is a SIP connection,
> most of the document applies to FreeSWITCH:
> http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf
>
> <http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf>The one
> problem I ran into was the IP address and port to which FreeSWITCH was
> bound. The default is to use the primary address, which works great
> out-of-the-box for everything else. When a client tried to register, all it
> got back was an ICMP error -- Destination Unreachable, Port Unreachable.
> That error is returned when no sockets are listening for UDP packets. To get
> FreeSWITCH to listen for your Virtual IP, you need to set it in two places:
>
> 5. In /opt/freeswitch/conf/vars.xml, set "bind_server_ip".
> 6. In /opt/freeswitch/conf/sip_profiles/internal.xml, set "sip-ip".
>
> That should do it! If you have any success, please report to this list.
>
> Keep in mind that if you want to do something like conferencing between two
> registered clients, then you have to deal with the fact that the clients may
> or may not be on the same box.
>
> Mike van Lammeren
>
>
>
> On Tue, Sep 29, 2009 at 10:20 AM, Mike van Lammeren <mike at van.lammeren.net
> > wrote:
>
>>
>> On Mon, Sep 28, 2009 at 9:05 PM, "Even Andr? Fiskvik" <grevenx at me.com
>> > wrote:
>>
>>> From: "Even Andr? Fiskvik" <grevenx at me.com>
>>> To: freeswitch-users at lists.freeswitch.org
>>> Date: Mon, 28 Sep 2009 22:52:13 +0200
>>> Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
>>> I have been working with a similar setup myself, but for some reason I
>>> ended up ditching theUltraMonkey setup because I just couldn't get it to
>>> work right.
>>>
>>> It's been quite a while since my effort, so I don't remember what the
>>> exact issue was.
>>> I got registrations to work, but had some other sip-dialog issues.
>>>
>>> We have since then changed over to running OpenSIPs as a loadbalancer in
>>> front of
>>> multiple FreeSWITCH instances. This setup is still in testing, but
>>> seemlingy works fine
>>> (and if it doesn't, it's my own fault for writing a bad opensips config).
>>>
>>> After we have done some more testing I can create a wiki-page with config
>>> details.
>>>
>>>
>>> Best regards,
>>> Even Andr?
>>>
>>>
>> Thanks, Even, that would be great! I might have to give up on the
>> ultramonkey solution, since I can't find anyone who has made it work. It's
>> too bad, because it would fit well with the rest of our architecture.
>>
>> Mike van Lammeren
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Thu Oct  1 12:17:11 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 1 Oct 2009 14:17:11 -0500
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <20091001143710.AE8063C4C4D@mail.cune.org>
References: <20091001143710.AE8063C4C4D@mail.cune.org>
Message-ID: <191c3a030910011217o6aed6b2ag1ded3d26a62430dc@mail.gmail.com>

You might want to try the ozmod_pri instead of ozmod_isdn until the new
revision of ozmod_isdn is published into the source tree.


On Thu, Oct 1, 2009 at 9:37 AM, <Russell.Mosemann at cune.org> wrote:

> We have connected FS to a Siemens Hicomm 300. As you might guess, it's
> not working right. Here is what we are working with.
>
> Dell 1750 (dual socket, dual core Xeon 2.8GHz)
> Debian 5
> FS (15029), OpenZAP (without libpri)
> TE110P T1 card (Zaptel driver)
> Handles 71xx extensions
>
> Siemens Hicom 300
> TMDN64P T1 card
> Handles 74xx extensions
>
> We are pretty much using the stock FS configuration, yet, because we're
> trying to get this to work. I have configured OpenZAP and the associated
> files like the examples on the wiki (see below) to work with a PRI T1.
> There are 23 B channels and 1 D channel. The Zaptel side looks fine.
> OpenZAP is able to open the channels when FS boots. So far, so good.
>
> When a call is made from 74xx (Rolmphone 624) to 71xx (X-Lite 4.0 beta
> from CounterPath on an office PC), X-Lite rings. The call can be
> answered, and the conversation sounds fine. That means the routing,
> registration and authorization are working on the network between X-Lite
> and FS. It also means that FS is able to communicate with the Hicom over
> the T1. Great.
>
> When the caller presses the transfer button on the 74xx phone, the Hicom
> sends a message over the D channel, and the call is disconnected
> (watching with fs_cli). As best I can interpret the bytes in the message,
> the Hicom sends a disconnect message when 74xx presses the transfer key.
>
> In order to call 74xx, I created dialplan/default/02_hicom.xml. The
> contents are
>
> <include>
>  <extension name="hicom">
>    <condition field="destination_number" expression="^(74\d{2})$">
>      <action application="bridge" data="openzap/1/a/$1"/>
>    </condition>
>  </extension>
> </include>
>
> If a call is made from 71xx to 74xx, the Hicom forwards the call to the
> switchboard with "7100->7445 connection not possible" (or whatever
> extensions) in the switchboard display.
>
> 1. Are these issues related to the way I have configured FS?
>
> The Hicom is maintained by the local phone company. I do not have access
> to view or configure the T1 card on the Hicom. According to the phone
> guy, there isn't anything else that needs to be configured on the Hicom.
> He believes that if 74xx can call 71xx, then 71xx should be able to call
> 74xx.
>
> I suspect that something more needs to be done on the Hicom in order to
> accept calls from FS and bridge/transfer them to a local extension on the
> Hicom. It's as if the Hicom doesn't know how or is not permitted to route
> incoming calls on the T1 to local extensions. I have no way to know,
> though. I'm hoping someone else has connected FS to a Hicom 300 and can
> provide configuration details. If I could tell the phone guy something
> like, "You need to look at <this>," that would help him out.
>
> 2. Should I receive CID/ANI from the Hicom?
>
> X-Lite displays "OpenZAP" as the call and "1" as Other when the call
> comes in, which is the information for the endpoint. Is there something I
> need to do in the FS configuration to capture CID/ANI information from
> the Hicom and make it available (or is it not being provided by the Hicom)?
>
> 3. When dialing from the Rolmphone is there a way for FS to send the
> called name back to the Hicom for it to appear in the display?
>
> When dialing 74xx to 74xx, of course, it shows the called number and name
> in the display. We also have a HiPath 4000 connected to the Hicom 300.
> When dialing an extension on the HiPath from the Hicom, the HiPath ships
> the called name back to the Hicom for display on the phone. It would be
> nice to do that from FS.
>
> Let me know if you need additional information. Thanks for any pointers
> or insight as to how things work.
>
> --
> Russell Mosemann
>
>
> openzap.conf
> [span zt PRI_1]
> name => OpenZAP
> number => 1
> trunk_type => t1
> b-channel => 1-23
> d-channel => 24
>
> zt.conf
> [defaults]
> codec_ms => 20
> wink_ms => 150
> flash_ms => 750
> echo_cancel_level => 64
> rxgain => 0.0
> txgain => 0.0
>
> openzap.conf
> <configuration name="openzap.conf" description="OpenZAP Configuration">
>  <settings>
>    <param name="debug" value="0"/>
>    <!--<param name="hold-music" value="$${moh_uri}"/>-->
>    <!--<param name="enable-analog-option" value="call-swap"/>-->
>    <!--<param name="enable-analog-option" value="3-way"/>-->
>  </settings>
>   <pri_spans>
>     <span name="PRI_1">
>       <param name="q921loglevel" value="alert"/>
>       <param name="q931loglevel" value="alert"/>
>       <param name="mode" value="user"/>
>       <param name="dialect" value="national"/>
>       <param name="dialplan" value="XML"/>
>       <param name="context" value="public"/>
>     </span>
>   </pri_spans>
> </configuration>
>
> zaptel.conf
> # Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" (MASTER)
> span=1,1,0,esf,b8zs
> # termtype: te
> bchan=1-23
> dchan=24
>
> # Global data
> loadzone        = us
> defaultzone     = us
>
>
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From nicolas at medularis.com  Thu Oct  1 12:29:09 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Thu, 1 Oct 2009 15:29:09 -0400
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <1b46b4e80909291542k71440965g75bb3bfcf38e8005@mail.gmail.com>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<191c3a030909291122t770af037j5b1068197b31de87@mail.gmail.com>
	<1b46b4e80909291542k71440965g75bb3bfcf38e8005@mail.gmail.com>
Message-ID: <1b46b4e80910011229k31f7233du10b0fc661730d3f3@mail.gmail.com>

Any ideas about this?

The SIP provider is offering H323, but I'm not quite sure about that, is
mod_opal working right?

Thanks!

Nicolas

On Tue, Sep 29, 2009 at 6:42 PM, Nicolas Brenner <nicolas at medularis.com>wrote:

> Anthony, thanks. Below are my config files for the two gateways from the
> sip trace. Both files are located in conf/directory/default.
>
> ---------------------
>
> redvoiss.xml (the one that works)
>
> <include>
>   <user id="gateway_redvoiss">
>     <gateways>
>       <gateway name="redvoiss-pp">
>         <param name="username" value="xxxxxxx"/>
>         <param name="password" value="xxxxxxx"/>
>         <param name="from-domain" value="pxextmy.redvoiss.net"/>
>         <param name="realm" value="pxextmy.redvoiss.net"/>
>         <param name="proxy" value="pxextmy.redvoiss.net"/>
>         <param name="from-user" value="xxxxxxx"/>
>         <param name="caller-id-in-from" value="false"/>
>         <param name="expire-seconds" value="600"/>
>         <param name="register" value="true"/>
>         <param name="retry_seconds" value="5"/>
>         <param name="extension" value="2010"/>
>         <param name="context" value="public"/>
>         <param name="codec-prefs" value="G729"/>
>         <param name="rfc2833-pt" value="101"/>
>       </gateway>
>     </gateways>
>     <params>
>       <param name="password" value="4321"/>
>     </params>
>   </user>
> </include>
>
> ---------------------
>
> orange.xml (the one that doesn't work)
>
> <include>
>   <user id="gateway_orange">
>     <gateways>
>       <gateway name="orange">
>         <param name="username" value="xxxxxxx"/>
>         <param name="password" value="xxxxxxx"/>
>         <param name="from-domain" value="216.72.10.39"/>
>         <param name="realm" value="216.72.10.39"/>
>         <param name="proxy" value="216.72.10.39"/>
>         <param name="from-user" value="xxxxxxx"/>
>         <param name="caller-id-in-from" value="false"/>
>         <param name="expire-seconds" value="600"/>
>         <param name="register" value="true"/>
>         <param name="retry_seconds" value="5"/>
>         <param name="extension" value="2011"/>
>         <param name="context" value="public"/>
>         <param name="codec-prefs" value="G729"/>
>         <param name="rfc2833-pt" value="101"/>
>       </gateway>
>     </gateways>
>     <params>
>       <param name="password" value="4321"/>
>     </params>
>   </user>
> </include>
>
> ---------------------
>
> If I remove the register=true param for the non-working gateway, I don't
> get the registration error on the cli, but then all call attempts get
> rejected with a 401 Unauthorized, and I get a hangup cause of
> NORMAL_UNSPECIFIED.
>
>
> Best,
>
> Nicolas
>
>
>
> On Tue, Sep 29, 2009 at 2:22 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> 900 level errors are sofia internal errors so probably something is wrong
>> with your gateway config xml.
>> if you want to send it with any critical info replaced with XXX maybe we
>> can see the issue for you.
>>
>>
>>
>> On Tue, Sep 29, 2009 at 1:05 PM, Nicolas Brenner <nicolas at medularis.com>wrote:
>>
>>> Hello everyone,
>>>
>>> I am trying to add a gateway, but after configuring it just like the
>>> others gateways I have, it is failing to register with a message like this:
>>>
>>> 2009-09-29 12:54:40.853440 [ERR] sofia_reg.c:1402 orange Registration
>>> Failed with status Operation has no matching challenge  [904]. failure #1
>>> 2009-09-29 12:54:40.906798 [WARNING] sofia_reg.c:364 orange Failed
>>> Registration, setting retry to 10 seconds.
>>>
>>> I captured the sip traffic and noticed that when trying to register with
>>> one gateway (the one that works), I get a "Trying" reply immediately
>>> followed by a "401 Unauthorized" which contains a "WWW-Authenticate: digest"
>>> with a "qop=auth" parameter. Then Freeswitch replies with a second REGISTER
>>> including a large "Authorization: digest" section with cnonce and
>>> nc=00000001 parameters.
>>>
>>> The gateway which doesn't register, doesn't send the "qop=auth" parameter
>>> together with the "401 Unauthorized", and then Freeswitch sends a
>>> "Authorization: digest" section on the second REGISTER with no cnonce or nc
>>> parameters.
>>>
>>> I know very little abouth SIP, so I'm wondering what this "qop=auth"
>>> parameter means and how does it affect the registration process. Is there
>>> any way to do without the qop=auth parameter?
>>>
>>> Also, I tried registering with X-Lite directly to the gateway, and it
>>> worked, so it appears to be a problem in the Freeswitch/gateway combination.
>>> (Note: X-Lite sends an "Authorization: digest" section on the _first_
>>> REGISTER, apparently this makes a difference)
>>>
>>> Attached is a sip trace for the registration traffic when doing "sofia
>>> profile external restart reloadxml" on the cli, captured with "tshark -i
>>> eth0 -o "rtp.heuristic_rtp: TRUE" -w /tmp/capture.pcap -b filesize:51200 -b
>>> files:100 -R 'sip or rtp or icmp or dns or rtcp or t38'"
>>>
>>> Thanks!
>>>
>>> Nicolas
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From hads at nice.net.nz  Thu Oct  1 12:41:24 2009
From: hads at nice.net.nz (Hadley Rich)
Date: Fri, 02 Oct 2009 08:41:24 +1300
Subject: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
In-Reply-To: <191c3a030910011214i3a9eaad1xf242ecf19dc80da3@mail.gmail.com>
References: <5d2828f0909290720y6bb4997uc4b9afba4b0fe74c@mail.gmail.com>
	<5d2828f0910010945q30ed76e2x50d6ab3b13bc8f88@mail.gmail.com>
	<32C47FA5-774E-4584-890B-0A066F5CFE14@me.com>
	<191c3a030910011214i3a9eaad1xf242ecf19dc80da3@mail.gmail.com>
Message-ID: <1254426084.3779.29.camel@sodium>

On Thu, 2009-10-01 at 14:14 -0500, Anthony Minessale wrote:
> can we do it without advertising to use ubuntu =D
> We don't like encouraging our users to use bleeding edge OS for our
> own sanity with debugging. 

I understand your stance, though if we're talking about Ubuntu 8.04 LTS
(Long Term Support - 5 years) it's not really bleeding edge anymore. 18
months ago when it was released it may have been a little, but the LTS
releases still aren't as bleeding edge as the standard support in
between releases.

hads
-- 
http://nicegear.co.nz
New Zealand's Open Source Hardware Supplier



From peder at networkoblivion.com  Thu Oct  1 12:44:08 2009
From: peder at networkoblivion.com (Peder)
Date: Thu, 1 Oct 2009 14:44:08 -0500
Subject: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
In-Reply-To: <191c3a030910011214i3a9eaad1xf242ecf19dc80da3@mail.gmail.com>
References: <5d2828f0909290720y6bb4997uc4b9afba4b0fe74c@mail.gmail.com>	<5d2828f0910010945q30ed76e2x50d6ab3b13bc8f88@mail.gmail.com>	<32C47FA5-774E-4584-890B-0A066F5CFE14@me.com>
	<191c3a030910011214i3a9eaad1xf242ecf19dc80da3@mail.gmail.com>
Message-ID: <0ca401ca42cf$8a709d20$9f51d760$@com>

Looking thru the example, it looks like each box has a real address of 21,
22 or 23 and they all have a loopback of .17, right?  So even though they
connections are being load balanced, each box really thinks it is .17 and
each client that connects thinks it is connecting to .17, right?  If that?s
the case, how does a client on one box call a client on the other box?
Since every box thinks it is .17 how would you bridge to another user on
another box that also thinks it is .17?  Or am I totally missing how it
works?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Thursday, October 01, 2009 2:14 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey

 

can we do it without advertising to use ubuntu =D
We don't like encouraging our users to use bleeding edge OS for our own
sanity with debugging.
Not to say you are not allowed to I just don't want to encourage it =p




On Thu, Oct 1, 2009 at 1:12 PM, Even Andr? Fiskvik <grevenx at me.com> wrote:

That's very cool Mike!

 

I'm going to try to configure four boxes with this as well (Btw, did you use
physical hardware or virtualization?)

and see how it goes. I followed Daniel Aliaman's blog as well, but I can try
it again with the tips

you provided on FreeSWITCH config to see if I can get it working properly
this time.

We did the setup on CentOS, but I wouldn't think that would be any issue.

 

Perhaps you or we could write up a complete guide about this on the wiki
since this is an scenario

commonly used? Also it would be great if we could outline possible issues
(and even better solutions)

to this kind of setup with regards to stuff like conferencing, bridging
between registered users and presence.

 

 

Best regards,

Even Andr?

 

 

On 1. okt. 2009, at 18.45, Mike van Lammeren wrote:

 

Guess what? I have two FreeSWITCH servers working behind UltraMonkey, using
heartbeat and ldirectord for load-balancing, fail-over and high
availability! I'm probably not the first one to do it, but as near as Google
and I can tell, I'm the first one to write about it.

 

Here's how you can duplicate my setup:

 

1. Install Ubuntu Server 8 on four machines, either real or VM.

2. Compile and install FreeSWITCH v1.0.4 from source on two machines,
following these instructions:
http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start

3. Configure both FreeSWITCH boxes, and make sure they are both working.

4. Follow (most of) these instructions from Daniel Aliaman's blog. They were
written for Asterisk, but since a SIP connection is a SIP connection, most
of the document applies to FreeSWITCH:

http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf

 

The one problem I ran into was the IP address and port to which FreeSWITCH
was bound. The default is to use the primary address, which works great
out-of-the-box for everything else. When a client tried to register, all it
got back was an ICMP error -- Destination Unreachable, Port Unreachable.
That error is returned when no sockets are listening for UDP packets. To get
FreeSWITCH to listen for your Virtual IP, you need to set it in two places:

 

5. In /opt/freeswitch/conf/vars.xml, set "bind_server_ip".

6. In /opt/freeswitch/conf/sip_profiles/internal.xml, set "sip-ip".

 

That should do it! If you have any success, please report to this list.

 

Keep in mind that if you want to do something like conferencing between two
registered clients, then you have to deal with the fact that the clients may
or may not be on the same box.

 

Mike van Lammeren

 

 

 

On Tue, Sep 29, 2009 at 10:20 AM, Mike van Lammeren <mike at van.lammeren.net>
wrote:

 

On Mon, Sep 28, 2009 at 9:05 PM, "Even Andr? Fiskvik" <grevenx at me.com>
wrote:

From: "Even Andr? Fiskvik" <grevenx at me.com>
To: freeswitch-users at lists.freeswitch.org
Date: Mon, 28 Sep 2009 22:52:13 +0200
Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey

I have been working with a similar setup myself, but for some reason I ended
up ditching the

UltraMonkey setup because I just couldn't get it to work right.

 

It's been quite a while since my effort, so I don't remember what the exact
issue was.

I got registrations to work, but had some other sip-dialog issues.

 

We have since then changed over to running OpenSIPs as a loadbalancer in
front of 

multiple FreeSWITCH instances. This setup is still in testing, but seemlingy
works fine

(and if it doesn't, it's my own fault for writing a bad opensips config).

 

After we have done some more testing I can create a wiki-page with config
details.

 

 

Best regards,

Even Andr?

 

 

Thanks, Even, that would be great! I might have to give up on the
ultramonkey solution, since I can't find anyone who has made it work. It's
too bad, because it would fit well with the rest of our architecture.

 

Mike van Lammeren

 

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:213-799-1400

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From siniypin at gmail.com  Thu Oct  1 14:03:38 2009
From: siniypin at gmail.com (RobertT)
Date: Fri, 2 Oct 2009 01:03:38 +0400
Subject: [Freeswitch-users] conference participant from behind NAT
In-Reply-To: <A3EEB6BE-19E2-49F3-8E41-FC74307A18E2@jerris.com>
References: <2160023e0909290132p2a5d0b62jbcb7d37625686866@mail.gmail.com>
	<681a20520909290618i2fbe7199rb18ff32933bba952@mail.gmail.com>
	<2160023e0909290634h3102d81bj9e3dd3d4d56f43e0@mail.gmail.com>
	<A3EEB6BE-19E2-49F3-8E41-FC74307A18E2@jerris.com>
Message-ID: <2160023e0910011403r5e9d0bb0ib5699f0523aaa606@mail.gmail.com>

I am still experiencing problem with lost media in conference on a client
behind NAT.
This is what I've done - disabled VAD on a NATed client and asked my friend
to produce lots of animal sounds in order to keep channel busy. But at the
end of minute sounds of wild nature disapeared again. We reproduced that
without security with tcp SIP transport and got the same result.
Then I started to dig into SIP trace and this is what I found.
This client (behind NAT) recieve subsequent INVITE message from FS which
seem to destroy dialog and causes client app to close media stream after a
session being established normally. I performed the same call from box with
public ip and saw no subsequent INVITE's from FS. How come FS sends an
INVITE message to already connected client? Is it OK? Should client handle
this normally?

Below is client's SIP trace:

INVITE sip:1.conference.dw at 74.208.167.44:5081;transport=TLS SIP/2.0
...
User-Agent: DoxWox SIP user agent
..

SIP/2.0 100 Trying
..
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M
..

SIP/2.0 407 Proxy Authentication Required
..
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M
..

ACK sip:1.conference.dw at 74.208.167.44:5081;transport=TLS SIP/2.0
..

SIP/2.0 100 Trying
..
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M
..

SIP/2.0 183 Session Progress
..
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M
..

SIP/2.0 200 OK
..
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M
..

ACK sip:1.conference.dw at 74.208.167.44:5081;transport=tls SIP/2.0
..

*Finally, this message cause media stream closing*
INVITE sip:1001 at 87.184.52.45:64183;transport=tls SIP/2.0
Via: SIP/2.0/TLS 74.208.167.44:5081;branch=z9hG4bK8269NDyXQNjyH
Max-Forwards: 70
From: <sip:1.conference.dw at 74.208.167.44<sip%3A1.conference.dw at 74.208.167.44>>;tag=vQH234QtN2U8Q

To: <sip:1001 at 74.208.167.44
<sip%3A1001 at 74.208.167.44>>;tag=3a231ba86c894ceca81d5021b68d3b6c

Call-ID: 37edc38329f64fe98c36cc0a6ddcbd9d
CSeq: 121093810 INVITE
Contact: <sip:1.conference.dw at 74.208.167.44:5081;transport=tls>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY
Supported: timer, precondition, path, replaces
Session-Expires: 120;refresher=uac
Min-SE: 120
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 340
v=0
o=FreeSWITCH 1254396503 1254396504 IN IP4 74.208.167.44
s=FreeSWITCH
c=IN IP4 74.208.167.44
t=0 0
m=audio 27726 RTP/SAVP 103 101
a=rtpmap:103 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:J6x2UgRVUY8GfbwjCuyyttrtnXnwwuWA9Pt+o3VW
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From frank at carmickle.com  Thu Oct  1 14:26:02 2009
From: frank at carmickle.com (Frank Carmickle)
Date: Thu, 1 Oct 2009 17:26:02 -0400
Subject: [Freeswitch-users] register problem
Message-ID: <20091001212602.GC17256@base.carmickle.com>

Can someone point out what is wrong here.  Thanks.

Siptrace at http://carmickle.com/fs.txt

<include>
      <gateway name="voiptalk.org">
        <param name="username" value="81100068"/>
        <param name="password" value="my_pass"/>
        <param name="from-user" value="81100068"/>
        <param name="from-domain" value="voiptalk.org"/>
    <param name="proxy" value="voiptalk.org"/>
        <param name="expire-seconds" value="600"/>
        <param name="register" value="true"/>
        <param name="retry-seconds" value="30"/>
        <param name="extension" value="1000"/>
        <param name="contact-params" value="domain_name=$${domain}"/>
        <param name="context" value="public"/>
      </gateway>
</include>
 
--FC


From lists at venturevoip.com  Thu Oct  1 14:36:32 2009
From: lists at venturevoip.com (Matt Riddell)
Date: Fri, 02 Oct 2009 10:36:32 +1300
Subject: [Freeswitch-users] bgapi jobid to uuid
Message-ID: <4AC520E0.4020507@venturevoip.com>

Hi,

I decided to go with a linked list for current channels and maintain 
that through state changes.

So, basically it works like this:

1. Originate a call using bgapi (we get a jobid in response)
2. Receive an event with the jobid and a uuid
3. Lookup the linked list for the jobid, set the uuid
4. Receive hangup etc (with uuid) remove from linked list

The problem is that sometimes I receive the hangup with the uuid before 
I'm told what the match between the jobid and the uuid are.

Any ideas?

-- 
Cheers,

Matt Riddell
Director
_______________________________________________

http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)


From Russell.Mosemann at cune.org  Thu Oct  1 14:19:14 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Thu, 1 Oct 2009 21:19:14 -0000
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <191c3a030910011217o6aed6b2ag1ded3d26a62430dc@mail.gmail.com>
Message-ID: <20091001211914.6533641A1FA@mail.cune.org>

Anthony Minessale <anthony.minessale at gmail.com> said:

> You might want to try the ozmod_pri instead of ozmod_isdn until the new
> revision of ozmod_isdn is published into the source tree.

libpri took care of the problem with the transfer. Now, someone can call
into FS from the Hicomm and then transfer the call to another extension
on the Hicomm.

A call from FS to the Hicomm still transfers to the switchboard. I'm not
seeing any CID/ANI on the X-Lite. It shows up as "OpenZAP" and "1". Is
there something like ngrep for the D channel of a PRI? It would be nice
to see what data is being sent between FS and the Hicom.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From siniypin at gmail.com  Thu Oct  1 14:44:22 2009
From: siniypin at gmail.com (RobertT)
Date: Fri, 2 Oct 2009 01:44:22 +0400
Subject: [Freeswitch-users] conference participant from behind NAT
In-Reply-To: <2160023e0910011403r5e9d0bb0ib5699f0523aaa606@mail.gmail.com>
References: <2160023e0909290132p2a5d0b62jbcb7d37625686866@mail.gmail.com>
	<681a20520909290618i2fbe7199rb18ff32933bba952@mail.gmail.com>
	<2160023e0909290634h3102d81bj9e3dd3d4d56f43e0@mail.gmail.com>
	<A3EEB6BE-19E2-49F3-8E41-FC74307A18E2@jerris.com>
	<2160023e0910011403r5e9d0bb0ib5699f0523aaa606@mail.gmail.com>
Message-ID: <2160023e0910011444y24448djede1bc24fc6d9c1@mail.gmail.com>

And here is a short piece of log from the server side:
...
nua(): refersh session after 62 seconds (in [55..65])...
send INVITE ...
rcv OK...
send ACK...
rcv BYE...

I see now that sdp for natted client has additional lines in OK response
compared to client with public ip.
Session-Expires: 120;refresher=uas
Min-SE: 120

How come that they differs? And how do I resolve this situation? Should
client handle these refresher messages normally?

Best regards, Robert.
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From anthony.minessale at gmail.com  Thu Oct  1 14:50:04 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 1 Oct 2009 16:50:04 -0500
Subject: [Freeswitch-users] bgapi jobid to uuid
In-Reply-To: <4AC520E0.4020507@venturevoip.com>
References: <4AC520E0.4020507@venturevoip.com>
Message-ID: <191c3a030910011450r1e0c1645r395c2024a25dcde9@mail.gmail.com>

if you pick your own job-uuid you can set it in the originate too

send

job-uuid: 1234

in your bgapi event

and

{job_uuid=1234}sofia/internal/foo at bar.com

in your dial string.

*shrug* i am not sure exactly what the goal is so maybe it's not a useful
suggestion...



On Thu, Oct 1, 2009 at 4:36 PM, Matt Riddell <lists at venturevoip.com> wrote:

> Hi,
>
> I decided to go with a linked list for current channels and maintain
> that through state changes.
>
> So, basically it works like this:
>
> 1. Originate a call using bgapi (we get a jobid in response)
> 2. Receive an event with the jobid and a uuid
> 3. Lookup the linked list for the jobid, set the uuid
> 4. Receive hangup etc (with uuid) remove from linked list
>
> The problem is that sometimes I receive the hangup with the uuid before
> I'm told what the match between the jobid and the uuid are.
>
> Any ideas?
>
> --
> Cheers,
>
> Matt Riddell
> Director
> _______________________________________________
>
> http://www.venturevoip.com/news.php (Daily Asterisk News)
> http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
> http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From anthony.minessale at gmail.com  Thu Oct  1 14:52:39 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 1 Oct 2009 16:52:39 -0500
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <20091001211914.6533641A1FA@mail.cune.org>
References: <191c3a030910011217o6aed6b2ag1ded3d26a62430dc@mail.gmail.com>
	<20091001211914.6533641A1FA@mail.cune.org>
Message-ID: <191c3a030910011452r2c163a83g7b5d0988d54facd7@mail.gmail.com>

there was a feature to generate a pcap from the debug logs but i forgot who
posted it.


On Thu, Oct 1, 2009 at 4:19 PM, <Russell.Mosemann at cune.org> wrote:

> Anthony Minessale <anthony.minessale at gmail.com> said:
>
> > You might want to try the ozmod_pri instead of ozmod_isdn until the new
> > revision of ozmod_isdn is published into the source tree.
>
> libpri took care of the problem with the transfer. Now, someone can call
> into FS from the Hicomm and then transfer the call to another extension
> on the Hicomm.
>
> A call from FS to the Hicomm still transfers to the switchboard. I'm not
> seeing any CID/ANI on the X-Lite. It shows up as "OpenZAP" and "1". Is
> there something like ngrep for the D channel of a PRI? It would be nice
> to see what data is being sent between FS and the Hicom.
>
> --
> Russell Mosemann
>
>
>
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From lists at venturevoip.com  Thu Oct  1 15:02:41 2009
From: lists at venturevoip.com (Matt Riddell)
Date: Fri, 02 Oct 2009 11:02:41 +1300
Subject: [Freeswitch-users] bgapi jobid to uuid
In-Reply-To: <191c3a030910011450r1e0c1645r395c2024a25dcde9@mail.gmail.com>
References: <4AC520E0.4020507@venturevoip.com>
	<191c3a030910011450r1e0c1645r395c2024a25dcde9@mail.gmail.com>
Message-ID: <4AC52701.4020504@venturevoip.com>

On 2/10/09 10:50 AM, Anthony Minessale wrote:
> if you pick your own job-uuid you can set it in the originate too
>
> send
>
> job-uuid: 1234
>
> in your bgapi event
>
> and
>
> {job_uuid=1234}sofia/internal/foo at bar.com <mailto:foo at bar.com>
>
> in your dial string.
>
> *shrug* i am not sure exactly what the goal is so maybe it's not a
> useful suggestion...

Does the job-uuid get used as the actual call uuid?  The problem is that 
the uuid for the call changes once it's answered (or hungup etc) and 
that the mapping doesn't come till after occasionally.

I'm getting the job uuid no problem, but it's the replacement of the 
job-uuid with a call-uuid that is the issue.

If this doesn't make too much sense I can post an example.

-- 
Cheers,

Matt Riddell
Director
_______________________________________________

http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)


From anthony.minessale at gmail.com  Thu Oct  1 15:36:14 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 1 Oct 2009 17:36:14 -0500
Subject: [Freeswitch-users] bgapi jobid to uuid
In-Reply-To: <4AC52701.4020504@venturevoip.com>
References: <4AC520E0.4020507@venturevoip.com>
	<191c3a030910011450r1e0c1645r395c2024a25dcde9@mail.gmail.com>
	<4AC52701.4020504@venturevoip.com>
Message-ID: <191c3a030910011536p403e4681u9375b53f594616a4@mail.gmail.com>

if you make your own uuids you could set them in the originate string with
{origination_uuuid=foo}

where foo is a real uuid.
if you have no other way to make them you can ask FS for one with the
create_uuid api call.


On Thu, Oct 1, 2009 at 5:02 PM, Matt Riddell <lists at venturevoip.com> wrote:

> On 2/10/09 10:50 AM, Anthony Minessale wrote:
> > if you pick your own job-uuid you can set it in the originate too
> >
> > send
> >
> > job-uuid: 1234
> >
> > in your bgapi event
> >
> > and
> >
> > {job_uuid=1234}sofia/internal/foo at bar.com <mailto:foo at bar.com>
> >
> > in your dial string.
> >
> > *shrug* i am not sure exactly what the goal is so maybe it's not a
> > useful suggestion...
>
> Does the job-uuid get used as the actual call uuid?  The problem is that
> the uuid for the call changes once it's answered (or hungup etc) and
> that the mapping doesn't come till after occasionally.
>
> I'm getting the job uuid no problem, but it's the replacement of the
> job-uuid with a call-uuid that is the issue.
>
> If this doesn't make too much sense I can post an example.
>
> --
> Cheers,
>
> Matt Riddell
> Director
> _______________________________________________
>
> http://www.venturevoip.com/news.php (Daily Asterisk News)
> http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
> http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Thu Oct  1 16:33:53 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 1 Oct 2009 16:33:53 -0700
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <20091001211914.6533641A1FA@mail.cune.org>
References: <191c3a030910011217o6aed6b2ag1ded3d26a62430dc@mail.gmail.com>
	<20091001211914.6533641A1FA@mail.cune.org>
Message-ID: <87f2f3b90910011633o1f561b76vf77e8e152d21e7a4@mail.gmail.com>

On Thu, Oct 1, 2009 at 2:19 PM, <Russell.Mosemann at cune.org> wrote:

> Anthony Minessale <anthony.minessale at gmail.com> said:
>
> > You might want to try the ozmod_pri instead of ozmod_isdn until the new
> > revision of ozmod_isdn is published into the source tree.
>
> libpri took care of the problem with the transfer. Now, someone can call
> into FS from the Hicomm and then transfer the call to another extension
> on the Hicomm.
>
> A call from FS to the Hicomm still transfers to the switchboard. I'm not
> seeing any CID/ANI on the X-Lite. It shows up as "OpenZAP" and "1". Is
> there something like ngrep for the D channel of a PRI? It would be nice
> to see what data is being sent between FS and the Hicom.
>
> I believe the "OpenZAP" and "1" are coming from your conf file:
openzap.conf
[span zt PRI_1]
name => OpenZAP
number => 1

As far as debugging with ozmod_libpri I believe the syntax is:
oz libpri debug 1 all

It will do a traditional libpri-style debug, just like "pri debug span 1" in
Asterisk.
-MC
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From msc at freeswitch.org  Thu Oct  1 16:41:02 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 1 Oct 2009 16:41:02 -0700
Subject: [Freeswitch-users] bgapi jobid to uuid
In-Reply-To: <4AC520E0.4020507@venturevoip.com>
References: <4AC520E0.4020507@venturevoip.com>
Message-ID: <87f2f3b90910011641l7f8f90c6td19ad8f70fbf914b@mail.gmail.com>

On Thu, Oct 1, 2009 at 2:36 PM, Matt Riddell <lists at venturevoip.com> wrote:

> Hi,
>
> I decided to go with a linked list for current channels and maintain
> that through state changes.
>
> So, basically it works like this:
>
> 1. Originate a call using bgapi (we get a jobid in response)
> 2. Receive an event with the jobid and a uuid
> 3. Lookup the linked list for the jobid, set the uuid
> 4. Receive hangup etc (with uuid) remove from linked list
>
> The problem is that sometimes I receive the hangup with the uuid before
> I'm told what the match between the jobid and the uuid are.
>
> Any ideas?
>

If I may ask, what's the application? Are you working on Vicidial-ish stuff
for FreeSWITCH? Also, you might want to call the FreeSWITCH conference
tomorrow. We have some Q&A time with the FS devs and this kind of thing
might work better in realtime rather than email threads.
-MC
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From lists at venturevoip.com  Thu Oct  1 16:46:01 2009
From: lists at venturevoip.com (Matt Riddell)
Date: Fri, 02 Oct 2009 12:46:01 +1300
Subject: [Freeswitch-users] bgapi jobid to uuid
In-Reply-To: <191c3a030910011536p403e4681u9375b53f594616a4@mail.gmail.com>
References: <4AC520E0.4020507@venturevoip.com>	<191c3a030910011450r1e0c1645r395c2024a25dcde9@mail.gmail.com>	<4AC52701.4020504@venturevoip.com>	<191c3a030910011536p403e4681u9375b53f594616a4@mail.gmail.com>
Message-ID: <4AC53F39.4030302@venturevoip.com>

On 2/10/09 11:36 AM, Anthony Minessale wrote:
> if you make your own uuids you could set them in the originate string with
> {origination_uuuid=foo}
>
> where foo is a real uuid.
> if you have no other way to make them you can ask FS for one with the
> create_uuid api call.

Awesome, thanks - will give it a whirl over the weekend.

-- 
Cheers,

Matt Riddell
Director
_______________________________________________

http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)


From jmesquita at freeswitch.org  Thu Oct  1 20:14:12 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 2 Oct 2009 00:14:12 -0300
Subject: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
	ToSubscribing Phones
In-Reply-To: <9B5214D274044021A57C6857844C10BA@greyhawk.tonecommander.com>
References: <B4B4D2747EEC4852B41A7D8D2DA36006@greyhawk.tonecommander.com>
	<191c3a030910010935s69e65569v76f4d26503a94d96@mail.gmail.com>
	<9B5214D274044021A57C6857844C10BA@greyhawk.tonecommander.com>
Message-ID: <ebbc2d550910012014ic655d84tbf000c084a98e51d@mail.gmail.com>

Piece of advice, don't ask, just do it. ;)

jmesquita

On Thu, Oct 1, 2009 at 3:29 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>  If you have time to take a look, I could put a trace in the pastebin?
>
> Jerry
>
>  ------------------------------
> *From:* Jerry Richards [mailto:jerry.richards at teotech.com]
> *Sent:* Thursday, October 01, 2009 10:29 AM
> *To:* 'freeswitch-users at lists.freeswitch.org'
> *Subject:* RE: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
> ToSubscribing Phones
>
>  I am using two Bria Professional Version 2.5.4 Build 54835 softphones.
>
> Thanks,
> Jerry
>
>  ------------------------------
> *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com]
> *Sent:* Thursday, October 01, 2009 9:36 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
> ToSubscribing Phones
>
> which phone is it,
> we tested it with eyebeam and it appears to work for us.
>
>
> On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards <jerry.richards at teotech.com
> > wrote:
>
>>
>> By the way, I see the following lines at the FS console, which might be a
>> clue as to why this is happening.  Could someone point me toward what
>> might
>> cause this?  I set the "manage-presence" parameter to "true" in each XML
>> file where I saw it defined.
>>
>> [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal)
>> [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
>>   ...
>> [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6)
>> [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
>>   ...
>> [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external)
>> [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
>>   ...
>> [WARNING] sofia_presence.c:565  192.168.72.38 is an alias, skipping
>>
>>
>> Best Regards,
>> Jerry
>>
>>
>> -----Original Message-----
>> From: Jerry Richards [mailto:jerry.richards at teotech.com]
>> Sent: Wednesday, September 30, 2009 9:12 AM
>> To: 'freeswitch-users at lists.freeswitch.org'
>> Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones
>>
>> I have two phones configured to subscribe to each other's presence status.
>> When I change the presence status in one phone, I see the SIP PUBLISH
>> message going to FS, but I don't see FS relaying that presence status to
>> the
>> subscribing phone.  Does anyone know why?
>>
>> Best Regards,
>> Jerry
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dule.maillist at gmail.com  Thu Oct  1 20:59:54 2009
From: dule.maillist at gmail.com (Dan Le)
Date: Thu, 1 Oct 2009 23:59:54 -0400
Subject: [Freeswitch-users] Minimum audio length for uuid_record
In-Reply-To: <20091001033138.GA18723@jdc.jasonjgw.net>
References: <914fc92a0909301944l116455b3tc25009fa9ea499d5@mail.gmail.com>
	<20091001033138.GA18723@jdc.jasonjgw.net>
Message-ID: <914fc92a0910012059v16bd018bpca397ec6c7b339d8@mail.gmail.com>

Thanks, I think I found the thread you were referring to
("[Freeswitch-users] session record does not for very short calls"), which
doesn't seem to be a solution for my situation. However, I did find that
using session:recordFile() didn't delete the file if it was really short.
And following that thread lead me to an interesting channel variable that
could be useful to us, record_ms, but having trouble getting it to reflect
the audio length. I can see the variable when printing data from the info
application, but it's always 0. My snippet of code is very simple:
if session:ready() then
   session:recordFile("C:/Temp/recording.wav", 30, 600, 6);

   local record_length = session:getVariable("record_ms");
   freeswitch.consoleLog("INFO", "Recorded a " .. record_length .. " ms
file.\n");
end

Another side question, the silence secs parameter (in this example, 6), is
that 6s silence hits during the entire recording session or 6s of
consecutive silence? From a few tests, it seems to be the former, but just
wanted to verify, something that would make a good addition to the wiki.

Dan


On Wed, Sep 30, 2009 at 11:31 PM, Jason White <jason at jasonjgw.net> wrote:

> Dan Le <dule.maillist at gmail.com> wrote:
> > We're running into a problem with the minimum file size when recording
> using
> > uuid_record. It seems if the audio is too short it deletes the audio
> file.
> > Is there a way to override that?
>
> Yes. It was discussed on the list recently. I suggest searching the list
> archives. Someone may have documented it on the wiki by now also.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From irmatov at gmail.com  Fri Oct  2 01:10:39 2009
From: irmatov at gmail.com (Timur Irmatov)
Date: Fri, 2 Oct 2009 13:10:39 +0500
Subject: [Freeswitch-users] internal & external ip addresses of freeswitch
Message-ID: <241d382f0910020110l5f3728e7k3e521e81353b5d6f@mail.gmail.com>

Hi.

We have a local network 192.168.1.0/24, where all the users are. Out
FreeSWITCH server is connected to this network, and has ip address
192.168.1.242. Through different network card it is connected to
external gateway, and has address 172.16.12.11 in this network.

I set up a test client with softphone. When incoming call is
deliviered to this client, call is set up normally, but client can't
hang it up. It sends BYE to external address - 172.16.12.11 - which is
not reachable from the client. It seems this address is coming from
Contact: field in INVITE that FreeSWITCH sends:

U 192.168.1.242:5060 -> 192.168.1.34:37169
INVITE sip:100 at 192.168.1.34:37169 SIP/2.0.
Via: SIP/2.0/UDP 172.16.12.11;rport;branch=z9hG4bKrvp6jm3myyaaF.
Max-Forwards: 70.
From: "FreeSWITCH" <sip:0000000000 at 172.16.12.11>;tag=v817pS9c6v6Fe.
To: <sip:100 at 192.168.1.34:37169>.
Call-ID: 797bd088-29cd-122d-9b93-0060979d54c5.
CSeq: 121117089 INVITE.
Contact: <sip:mod_sofia at 172.16.12.11:5060>.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14898.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 267.
Remote-Party-ID: "FreeSWITCH"
<sip:0000000000 at 172.16.12.11>;party=calling;screen=yes;privacy=off.

What should I tweak in freeswitch to change this behaviour?


-- 
Timur Irmatov, xmpp:irmatov at jabber.ru


From siniypin at gmail.com  Fri Oct  2 01:32:26 2009
From: siniypin at gmail.com (RobertT)
Date: Fri, 2 Oct 2009 12:32:26 +0400
Subject: [Freeswitch-users] conference participant from behind NAT
In-Reply-To: <2160023e0910011444y24448djede1bc24fc6d9c1@mail.gmail.com>
References: <2160023e0909290132p2a5d0b62jbcb7d37625686866@mail.gmail.com>
	<681a20520909290618i2fbe7199rb18ff32933bba952@mail.gmail.com>
	<2160023e0909290634h3102d81bj9e3dd3d4d56f43e0@mail.gmail.com>
	<A3EEB6BE-19E2-49F3-8E41-FC74307A18E2@jerris.com>
	<2160023e0910011403r5e9d0bb0ib5699f0523aaa606@mail.gmail.com>
	<2160023e0910011444y24448djede1bc24fc6d9c1@mail.gmail.com>
Message-ID: <2160023e0910020132i3be2d3a2o615cb18dba992878@mail.gmail.com>

Hi folks!

Suddenly I found this
http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-February/002015.htmltopic
and that explains a lot.
>From there I see that sofia sends refresher messages for NATed client in
order to check if it still alive.
It means I have problems in my client. Sorry for the mess.

Cheers, Robert.
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From mcampbellsmith at gmail.com  Fri Oct  2 01:58:58 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Fri, 2 Oct 2009 18:58:58 +1000
Subject: [Freeswitch-users] [ERR] mod_dingaling.c:980 Stun Failed!
In-Reply-To: <33c87fa30909290415y178b6506x120a6e5c39839ce0@mail.gmail.com>
References: <33c87fa30909290415y178b6506x120a6e5c39839ce0@mail.gmail.com>
Message-ID: <33c87fa30910020158q7589e18fj53bd4cc95ff3926f@mail.gmail.com>

Anyone have this issue?

On Tue, Sep 29, 2009 at 9:15 PM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> Hi!
>
> I have just started to use dingaling again, and noticed I constantly
> get a stun error.
>
> 2009-09-29 21:11:03.175002 [ERR] mod_dingaling.c:980 Stun Failed!
> stun.fwdnet.net:3478 [Remote Address Error!]
>
> I have tried with stun.freeswitch.org and stun.fwdnet.net stun servers
> and keep getting this error with dingaling. ?I have no problems with
> inbound sip calls, so I don't think ?its the actual stun server.
>
> Has anyone else seen this? ?I am using: FreeSWITCH Version 1.0.trunk (14952)
>
> Thanks!
>


From shaheryarkh at googlemail.com  Fri Oct  2 03:00:03 2009
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Fri, 2 Oct 2009 16:00:03 +0600
Subject: [Freeswitch-users] [ERR] mod_dingaling.c:980 Stun Failed!
In-Reply-To: <33c87fa30910020158q7589e18fj53bd4cc95ff3926f@mail.gmail.com>
References: <33c87fa30909290415y178b6506x120a6e5c39839ce0@mail.gmail.com>
	<33c87fa30910020158q7589e18fj53bd4cc95ff3926f@mail.gmail.com>
Message-ID: <b16156850910020300x6cd7dd9cm2348942b04c79030@mail.gmail.com>

Yes, i had same problem, then i changed stun server to one of our own
servers. You may try some of public stun servers listed on below link,

http://www.voip-info.org/wiki/view/STUN

Thank you.


On Fri, Oct 2, 2009 at 2:58 PM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> Anyone have this issue?
>
> On Tue, Sep 29, 2009 at 9:15 PM, Mark Campbell-Smith
> <mcampbellsmith at gmail.com> wrote:
> > Hi!
> >
> > I have just started to use dingaling again, and noticed I constantly
> > get a stun error.
> >
> > 2009-09-29 21:11:03.175002 [ERR] mod_dingaling.c:980 Stun Failed!
> > stun.fwdnet.net:3478 [Remote Address Error!]
> >
> > I have tried with stun.freeswitch.org and stun.fwdnet.net stun servers
> > and keep getting this error with dingaling.  I have no problems with
> > inbound sip calls, so I don't think  its the actual stun server.
> >
> > Has anyone else seen this?  I am using: FreeSWITCH Version 1.0.trunk
> (14952)
> >
> > Thanks!
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
________________________________________________________
|
                     |
| FATAL ERROR                                                           ---
O X |
|_______________________________________________________|
|                        You have moved the mouse.
 |
| Windows must be restarted for the changes to take effect.   |
|                                        <OK>
               |
####################################/


Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From aep.lists at it46.se  Fri Oct  2 04:11:23 2009
From: aep.lists at it46.se (Alberto Escudero)
Date: Fri, 2 Oct 2009 13:11:23 +0200
Subject: [Freeswitch-users] How to limit the number of incoming+outgoing
 calls via specific gateway?
In-Reply-To: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>
References: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>
Message-ID: <e403cddca1c90c83ce4fb6f85b026e63.squirrel@correo.nodo50.org>


You can use the api and check that the channel is occupied with "show
channels"?
You can write a small javascript that checks if the channel is occupied by
means of session.execute api.

/aep
-- 
Stopping junk mailers is good for the environment

> My SIP provider allows only one call (incoming or outgoing) via one
> SIP account. For FreeSWITCH I have configured it as public DID
> extension and outgoing gateway. Now I would like to transfer to
> another gw (or generate "limit exceded") when one tries to place an
> outgoing call while incoming call is in progress. How tho do that?
> Limiting the number of outgoing calls is easy (mod_limit), but how to
> take into account incoming one?
>
> - Dmitry Bely
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>




From Russell.Mosemann at cune.org  Fri Oct  2 04:48:35 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Fri, 2 Oct 2009 11:48:35 -0000
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <87f2f3b90910011633o1f561b76vf77e8e152d21e7a4@mail.gmail.com>
Message-ID: <20091002114835.4C0FA419B7A@mail.cune.org>

Michael Collins <msc at freeswitch.org> said:

> > I believe the "OpenZAP" and "1" are coming from your conf file:
> openzap.conf
> [span zt PRI_1]
> name => OpenZAP
> number => 1

That is correct. If that information is removed, then X-Lite displays

FreeSWITCH
[Other: 0000000000]

Are there any variables to set to get CID, or is OpenZap supposed to be
filling that in?

> As far as debugging with ozmod_libpri I believe the syntax is:
> oz libpri debug 1 all

That works. Now, I have to figure out what some of these abbreviations mean.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From tculjaga at gmail.com  Fri Oct  2 05:05:55 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 2 Oct 2009 14:05:55 +0200
Subject: [Freeswitch-users] configure FS: config.status: error: cannot find
	input file: Makefile.in
Message-ID: <65d96fc80910020505t48a6237ao19884c23fedf228d@mail.gmail.com>

hello,
i just got the last trunk and tried to compile it on one of my development
machines... Well configure fails on tiff-3.8.2 where it is unable to find
Makefile.in ... Can someone advice?



checking if g++ static flag -static works... yes
checking if g++ supports -c -o file.o... yes
checking if g++ supports -c -o file.o... (cached) yes
checking whether the g++ linker (/usr/bin/ld) supports shared libraries...
yes
checking dynamic linker characteristics... GNU/Linux ld.so
checking how to hardcode library paths into programs... immediate
checking for OpenGL Utility library... no
checking for GLUT library... no
configure: creating ./config.status
config.status: error: cannot find input file: Makefile.in



tculjaga at subZero:~/freeswitch-trunk/libs/tiff-3.8.2$ ls -l
total 2224
-rw-r--r--  1 tculjaga tculjaga  23741 2009-10-02 13:19 acinclude.m4
-rw-r--r--  1 tculjaga tculjaga 316978 2009-10-02 13:28 aclocal.m4
-rwxr-xr-x  1 tculjaga tculjaga    121 2009-10-02 13:19 autogen.sh
-rw-r--r--  1 tculjaga tculjaga 124047 2009-10-02 13:19 ChangeLog
drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:28 config
-rw-r--r--  1 tculjaga tculjaga 103926 2009-10-02 14:00 config.log
-rwxr-xr-x  1 tculjaga tculjaga  73065 2009-10-02 14:00 config.status
-rwxr-xr-x  1 tculjaga tculjaga 740145 2009-10-02 13:28 configure
-rw-r--r--  1 tculjaga tculjaga  20492 2009-10-02 13:19 configure.ac
-rwxr-xr-x  1 tculjaga tculjaga     56 2009-10-02 13:19 configure.gnu
-rwxr-xr-x  1 tculjaga tculjaga 737794 2009-10-02 13:57 configure.lineno
drwxr-xr-x 16 tculjaga tculjaga   4096 2009-10-02 13:19 contrib
-rw-r--r--  1 tculjaga tculjaga   1146 2009-10-02 13:19 COPYRIGHT
-rw-r--r--  1 tculjaga tculjaga   1570 2009-10-02 13:19 HOWTO-RELEASE
drwxr-xr-x  5 tculjaga tculjaga   4096 2009-10-02 13:19 html
drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:28 libtiff
drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 m4
-rw-r--r--  1 tculjaga tculjaga   1908 2009-10-02 13:19 Makefile.am
-rw-r--r--  1 tculjaga tculjaga   1724 2009-10-02 13:19 Makefile.vc
drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 man
-rw-r--r--  1 tculjaga tculjaga   6270 2009-10-02 13:19 nmake.opt
drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 port
-rw-r--r--  1 tculjaga tculjaga   2363 2009-10-02 13:19 README
-rw-r--r--  1 tculjaga tculjaga      9 2009-10-02 13:19 RELEASE-DATE
-rw-r--r--  1 tculjaga tculjaga   5893 2009-10-02 13:19 SConstruct
drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 test
-rw-r--r--  1 tculjaga tculjaga    433 2009-10-02 13:19 TODO
drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 tools
-rw-r--r--  1 tculjaga tculjaga      6 2009-10-02 13:19 VERSION
tculjaga at subZero:~/freeswitch-trunk/libs/tiff-3.8.2$
tculjaga at subZero:~/freeswitch-trunk/libs/tiff-3.8.2$
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From tculjaga at gmail.com  Fri Oct  2 05:32:31 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 2 Oct 2009 14:32:31 +0200
Subject: [Freeswitch-users] How to limit the number of incoming+outgoing
	calls via specific gateway?
In-Reply-To: <e403cddca1c90c83ce4fb6f85b026e63.squirrel@correo.nodo50.org>
References: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>
	<e403cddca1c90c83ce4fb6f85b026e63.squirrel@correo.nodo50.org>
Message-ID: <65d96fc80910020532i4ba33353g7ac141624a4bb89e@mail.gmail.com>

what if you are running some huge traffic e.g. 2000 calls with media?

a typical application for that is an IVR system handling several different
services. I'd like to "dedicate" some capacity for inbound on per service
basis.


e.g.

DID 10001 limit to 500 calls
DID 10002 limit to 400 calls
DID 10003 limit to 100 calls
DID 10005 limit to 1000 calls


This will be a total of 2000 calls.


don't you think js is simply too weak for that? It should cont
calls/channels, brake counts per service/DID and update the counters on
every call hit.




in the DP you would have something like this for every DID:


<include>
  <extension name="MY_DID_NUM">
    <condition field="destination_number" expression="^MY_DID_NUMBER$">
    <action application="set" data="SERVICE_LIMIT=500"/>

      <!--
             count number of active channels going towards MY_DID_NUMBER and
store it into COUNT_MY_DID_NUMBER
      -->

      <action application="transfer" data="do_MY_SERVICE XML public"/>
    </condition>
  </extension>
</include>



<include>
  <extension name="SERVICE1">
    <condition field="destination_number" expression="^do_MY_SERVICE$"/>
    <condition field="${COUNT_MY_DID_NUMBER}" expression="^SERVICE_LIMIT$">

       <!-- do your service here -->
      <action application="playback" data="I_Accept_Your_Call.wav"/>
      <action application="hangup" data="NORMAL_CLEARING"/>

       <!-- do your limitation here -->
      <anti-action application="respond" data="403 Forbidden"/> <= put your
response here!

    </condition>
  </extension>
</include>





but the question is ... how powerful a JavaScript can be? Will it be enough
to handle that load?



Tihomir.





On Fri, Oct 2, 2009 at 1:11 PM, Alberto Escudero <aep.lists at it46.se> wrote:

>
> You can use the api and check that the channel is occupied with "show
> channels"?
> You can write a small javascript that checks if the channel is occupied by
> means of session.execute api.
>
> /aep
> --
> Stopping junk mailers is good for the environment
>
> > My SIP provider allows only one call (incoming or outgoing) via one
> > SIP account. For FreeSWITCH I have configured it as public DID
> > extension and outgoing gateway. Now I would like to transfer to
> > another gw (or generate "limit exceded") when one tries to place an
> > outgoing call while incoming call is in progress. How tho do that?
> > Limiting the number of outgoing calls is easy (mod_limit), but how to
> > take into account incoming one?
> >
> > - Dmitry Bely
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tculjaga at gmail.com  Fri Oct  2 05:38:30 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 2 Oct 2009 14:38:30 +0200
Subject: [Freeswitch-users] Dialplan Issue
In-Reply-To: <35b355e90910010918w7a01c4d1l7bc8b7986d2a3434@mail.gmail.com>
References: <35b355e90910010727v2efcbff6p97a0a883857818e4@mail.gmail.com>
	<191c3a030910010755l387faba8x1d8f9bbf51c8ffa1@mail.gmail.com>
	<35b355e90910010901j63a36531m7550eba26564f01@mail.gmail.com>
	<35b355e90910010918w7a01c4d1l7bc8b7986d2a3434@mail.gmail.com>
Message-ID: <65d96fc80910020538s51c47425vf9142d4fe47e16bb@mail.gmail.com>

anyhow, this is how it works for me!




<include>
  <context name="public">

      <extension name="LNP">
                <condition field="destination_number"
expression="(^300000)(.*)">
                      <action application="lnp_getprefix" data="in $2, out
reroutingalias"/>
                        <action application="redirect" data="sip:${
reroutingalias}@10.4.13.11:5060"/>
                </condition>
        </extension>


        <extension name="LBS">
                <condition field="destination_number"
expression="(^300010)(.*)">
                        <action application="lbs_getpublicphone" data="in
${caller_id_number}, in $2, out reroutingalias"/>
                        <action application="redirect" data="sip:${
reroutingalias}@10.4.13.11:5060"/>
              </condition>
        </extension>

        <extension name="CPS">
                <condition field="destination_number"
expression="(^300020)(.*)">
                        <action application="cps_verifyphone" data="in
${caller_id_number}, in $2, out radiusacc"/>
                 </condition>
              <condition field="radiusacc" expression="1">
                            <action application="redirect" data="sip:${
caller_id_number}@10.4.13.11:5060"/>
                            <anti-action application="respond" data="403
Forbidden"/>
                </condition>
        </extension>



   <extension name="ServiceLookup">
      <condition field="destination_number" expression="(^300030)(.*)">
         <action application="lookup_service_destination" data="in
${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, in $
1, in ${network_addr}:5060, out red_contact, out authResult"/>
         <action application="log" data="INFO ########################
ServiceLookup ########################\n"/>
         <action application="log" data="INFO ########################
contact = '${red_contact}' ##############\n"/>
         <action application="log" data="INFO ########################
CallerNum = '${caller_id_number:6:16}' ##########\n"/>
         <action application="log" data="INFO ########################
RADIUS auth = '${authResult}' ##########\n"/>

         <action application="execute_extension" data="doRedirect XML
public"/>
        </condition>
   </extension>


   <extension name="doRedirect">
      <condition field="destination_number" expression="^doRedirect$"/>
      <condition field="${authResult}" expression="^0$|^60$">
         <action application="log" data="INFO ########################
RADIUS auth OK!!!' ##########\n"/>
         <action application="redirect" data="${red_contact}"/>
         <anti-action application="log" data="INFO ########################
RADIUS auth NOK!! ##########\n"/>
         <anti-action application="respond" data="403 Forbidden"/>
      </condition>

   </extension>


  </context>
</include>

On Thu, Oct 1, 2009 at 6:18 PM, Shelby Ramsey <sicfslist at gmail.com> wrote:

> Just to confirm ... works like a champ.
>
> Thanks again!!!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From orien at tx.rr.com  Thu Oct  1 18:11:40 2009
From: orien at tx.rr.com (Orien Love)
Date: Thu, 01 Oct 2009 20:11:40 -0500
Subject: [Freeswitch-users] New to freeswitch and have a few questions
Message-ID: <4AC5534C.6050202@tx.rr.com>

Hello Everybody,
    I am new to freeswitch, so forgive me if I ask stupid questions. I 
am planning a test setup consisting of:
    1 - Pfsense router with the freeswitch package installed.
    1 - Cisco WS-C3524-PWR-XL-EN 24 FE Switch, POE for the phones.
    1 - LINKSYS SPA3000 to connect to my existing land line and phones.
    2 - POLYCOM SOUNDPOINT IP601 SIP IP PHONEs

The first question I have, Are the IP601 phones supported? The wiki 
lists 320, 431, 501, 550, 650 but not the 601.

Second, is there a place that helps a person new to the IP phone world 
learn what is needed to set up a PBX using freeswitch at a small office?

Finally is my test setup a good one? is there something I am missing or 
that I need to get the learning process started, I have found in the 
past, with a little information and a test system, I can learn what I am 
doing by breaking and fixing the test bed.

Thanks for your time
Orien


From msc at freeswitch.org  Fri Oct  2 09:00:45 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 2 Oct 2009 09:00:45 -0700
Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Starting,
	Please Call In!
Message-ID: <87f2f3b90910020900k20f65141rcb2414402ce388a1@mail.gmail.com>

Hey folks, the weekly conference call is starting. Please see the agenda for
instructions on dialing:

http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_02

Looking forward to speaking with you all!

-Michael
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From msc at freeswitch.org  Fri Oct  2 09:08:03 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 2 Oct 2009 09:08:03 -0700
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <20091002114835.4C0FA419B7A@mail.cune.org>
References: <87f2f3b90910011633o1f561b76vf77e8e152d21e7a4@mail.gmail.com>
	<20091002114835.4C0FA419B7A@mail.cune.org>
Message-ID: <87f2f3b90910020908p1aa5f5d4o8b69894cd4a58bef@mail.gmail.com>

On Fri, Oct 2, 2009 at 4:48 AM, <Russell.Mosemann at cune.org> wrote:

> Michael Collins <msc at freeswitch.org> said:
>
> > > I believe the "OpenZAP" and "1" are coming from your conf file:
> > openzap.conf
> > [span zt PRI_1]
> > name => OpenZAP
> > number => 1
>
> That is correct. If that information is removed, then X-Lite displays
>
> FreeSWITCH
> [Other: 0000000000]
>

do something like:
name => XYZ Corp
number => 8005551212

>
> Are there any variables to set to get CID, or is OpenZap supposed to be
> filling that in?
>
> > As far as debugging with ozmod_libpri I believe the syntax is:
> > oz libpri debug 1 all
>
> That works. Now, I have to figure out what some of these abbreviations
> mean.
>
> Welcome to the wacky world of Q931. The wiki has info:
http://wiki.freeswitch.org/wiki/ISDN:_Integrated_Services_Digital_Network

-MC
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From mgende at gendesign.com  Fri Oct  2 09:10:30 2009
From: mgende at gendesign.com (Michael Gende)
Date: Fri, 2 Oct 2009 11:10:30 -0500
Subject: [Freeswitch-users] New to freeswitch and have a few questions
In-Reply-To: <4AC5534C.6050202@tx.rr.com>
References: <4AC5534C.6050202@tx.rr.com>
Message-ID: <a87d43fd0910020910l5db88c5fv980c8c8fe312f380@mail.gmail.com>

Hey Orien,

I'm not using exactly your set up, but am using pfsense/FreeBSD. Since
you're using that, I assume you're going "dual homed". I've got a starter
guide that might help you out. If nothing else, I'd be interested in a
candid assessment of its usefulness or lack thereof, especially to a guy
like you.

I've included it here. Its all just text at the moment so be advised. Also
be advised that there's a lot of great information on the freeswitch site
and in this group. The goal of my document was so that someone just starting
would have to hunt a little less.

Hope its good for something, let me know either way, especially if you find
errors.

Regards,

Mike G.

On Thu, Oct 1, 2009 at 8:11 PM, Orien Love <orien at tx.rr.com> wrote:

> Hello Everybody,
>    I am new to freeswitch, so forgive me if I ask stupid questions. I
> am planning a test setup consisting of:
>    1 - Pfsense router with the freeswitch package installed.
>    1 - Cisco WS-C3524-PWR-XL-EN 24 FE Switch, POE for the phones.
>    1 - LINKSYS SPA3000 to connect to my existing land line and phones.
>    2 - POLYCOM SOUNDPOINT IP601 SIP IP PHONEs
>
> The first question I have, Are the IP601 phones supported? The wiki
> lists 320, 431, 501, 550, 650 but not the 601.
>
> Second, is there a place that helps a person new to the IP phone world
> learn what is needed to set up a PBX using freeswitch at a small office?
>
> Finally is my test setup a good one? is there something I am missing or
> that I need to get the learning process started, I have found in the
> past, with a little information and a test system, I can learn what I am
> doing by breaking and fixing the test bed.
>
> Thanks for your time
> Orien
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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FREESWITCH FOR DUMMIES: DUAL-HOMED HOST EXAMPLE mgende at gendesign.com September, 2009. 

 - Stuff One Ought to Know Before Starting 
 - set FS as a dual-homed host (seperate LAN and WAN interfaces), 
 - use the pre-set 1000-1019 extension numbers to configure and register your local SIP phones, 
 - set up FS to register with your SIP/PSTN provider, 
 - tell FS how to use that provider gateway for out-bound calls, 
 - Group your LAN phone extensions in a useful way, 
 - tell FS to ring all LAN SIP phones when a call comes in. 
 - Add a main Voice Mail account
 - Ring a Group, if no answer, transfer to Main VM 
 - Adding Another DID

STUFF ONE OUGHT TO KNOW BEFORE STARTING

PURPOSE OF THIS DOCUMENT

When I was getting my own FS up and running, I did a lot of Wiki reading (and still do). There's a lot of information (including a really good "Getting Started" guide) on the FreeSwitch site, but I nonetheless found myself looking here and there. So, I tried to put into one place some things I had found about the basics. While I keep this pretty straightforward, there is the assumption that one is comfortable with computer systems, networking, VoIP in general, XML, these kinds of things. This is not developer documentation, just "getting it going and trying to understand what's what." You'll get some of my opinions, keep 'em or chuck 'em, up to you. But, one will also learn some important basics about how to do some hopfully useful things, where things are, and how to get FS to act like a phone system (in this case). From there, the sky's the limit. The idea here is to provide a foothold to step up higher.

Note that this document has much in common with the "Getting Started" section on the FS website. This is more tuned to dual-homed setups, thus its creation. If you find it helpful, great. Find errors, point them out to me, please, and I'll correct them. If you hate it, then delete your copy.  

Oh, and any external static IP addresses, DIDs, etc, in this document are all pure fiction, so please don't waste any time trying to hack them, if you were so inclined (which I'm sure you were not).


PRE-INSTALL CONSIDERATIONS

THE COMPUTER:

Before one starts anything, ask yourself a few questions about how you would like FS to be deployed in your situation. One way, and a simple one at that, is to add a computer (or use an existing one) to your LAN. Just a box with, say, a Gb of RAM and a dual-core CPU will do it (see the FS site for more precise details on system requirements). FS comes set up to work this way without much fooling around. However, using a dual-homed host has its advantages. The largest one being lessening the traffic introduced to your LAN because a WAN interface is available. I find this a good thing and worth the extra effort. You may not. 


THE OS:

FS can run on a variety of OS's (the top three being MacOS, Windows, and Linux I believe). If you can work with Linux, BSD and the like, you'll be fine. The FS site has a list of ways to install on different distros, take your pick. We initially went the CentOS, which is a good Red Hat Linux compatable distro. For a computer with only one Ethernet internface on your LAN, that'd be my personal choice. However, if you use a computer with seperate LAN and WAN interfaces, I'd opt for PfSense (pfsense.org). This is a very nice FreeBSD system that is already a powerful firewall (and other great stuff). If you're connecting direct to the Internet, you'll need that. Further, with PfSense the instalation of the OS and Firewall, via install disk image they provide, is pretty straightforward. Once FreeBSD/PfSense is installed, one adds FS as a package (see the directions at pfsense.org) with just a couple of mouse clicks. While FS on PfSense has some handy interfaces to provide aids in administration, I tend to run and configure it from the command line just the same. I do so because I want to know and understand the FS directory structure and XML files that make things work.  


SIDEBAR: Want to run FS on a dual homed host? 

That's what we do. One Ethernet port is set for the LAN while the other connects direct to the Internet. We opt for a static WAN IP address. I personally suggest using a fixed IP if you're going to go this route. It will make configuring FS, as well as well as potentially registering with your SIP providor, easier in my opinion. Also, one will have less new LAN traffic with the advent of you new VoIP system as one would with a single-homed host. This is what pursuaded me. However, please note the FS scripts are pretty intellegent and can sense an IP change on your WAN port, letting the system know about it. However, for clarity and ease, largely my own, I'll go on about a static IP on the WAN and a seperate IP address for the LAN port of your FS box. 

NOTE: Just call your cable provider or DSL provider for info on getting a static IP if you'd like one. We have both DSL and Cable at my office, it was pretty simple to set up. These Internet providors, they'll do anything for money. If you have a T1 or better, you already have a static IP. Also note: going this way (dual-homed, fixed IP) gets your hands a little dirty, and I do mean "a little", with FS. That's something I think you'll want to do to understand more about how things work. It sure helped me.


MOVING ON: A FEW IMPORTANT FS COMMANDS FOR STARTERS

At this point, I'm assuming that you'll have the computer of your choice up and running with an OS and FS installed and waiting for your commands (if you don't, pretend). When I was getting FS up and running, there were a few concepts and commands that helped me get on my feet. So, for your edification, here they are:

To Start FS: The executable is in /usr/local/freeswitch/bin/freeswitch. Invoking it long-hand (or going to the ~/bin directory and typing ./freeswitch) will start FS in the foreground, attach to your terminal, and provide output and a command line. 

To Check FS Status: If the screen is full of "junk", hit enter. You should see a prompt something like this: "freeswitch at FreeSwitch.local>".  From there, you can tell FS to do things and ask about status.  You'll use commands like "sofia status", "sofia status profile internal", and "sofia status profile external". One can accomplish quite a lot with just those few. More on these later in the document. Or, go try 'em out.  

To Stop FS: At the command line mentioned above, just enter the command "shutdown" (no quotes). Or, one could kill it if its running in the background (as it does by default in PfSense on FreeBSD). I've done so with a "kill -15" followed by the FS PID (ps uax | grep free) and things shut down fine. There's probably a better way to do that, but this works.  

To Reload XML Files into FS: You'll likely be making changes to the XML files that configure FS, especially initially. Once you edit an XML file, FS has to be told about this so your changes will take effect. That can be done with the "reloadxml" command. Alternatively, one can issue a "stop" command and then start FS from the command line. Take your choice. 

For More Info: one can type "help" at the FS command prompt. There's a lot there; don't let it intimidate you. 


THE FS FILE SYSTEM: WHERE STUFF IS FOR STARTERS:

Here's the basics on where things are. Master this and you've got a good, starting grasp on how to "get around" when configuring and maintaining your FS:
NOTE: Everythings starts with /usr/local/freeswitch, so I'm just going to assume that with the tilde (~) here to save me from typing it again and again. 

~/bin is where all the binaries, like the FS executable itself, resides. 
~/conf is a starting point for your exloration. Some important files here, like vars.xml, and more. 
~/conf/sip_profiles is very important. Here are all the SIP User Agent instructions. UAs listen for registrations of SIP Phones, etc. 
~/conf/dialplan is where one sets up instructions telling FS about events and what to do when they take place.
~/conf/directory is where information on physical SIP phone extensions (and their groupings) is stored. 

These directories have subdirectories (which themselves have subdirectories) that you'll want to get familiar with. I'll go into more detail later on. Also, I've left out where language stuff goes, and more. But, one has to start somewhere. Now, let's get to work. 



SETTING THE SIP PROFILES TO USE DIFFERENT ETHERNET PORTS

INTERNAL LAN

Back to "dual-homed host setup with fixed IP on the WAN port". To make this work, one has to tell FS about where to be listening for SIP registrations, and other SIP traffic. That's done in the /usr/local/freeswitch/conf/sip_profiles directory. There, one finds what are conveniently called "sip profiles". One profile is for internal traffic (in my case called "internal.xml", for the phones on your LAN, etc) the other is for outside traffic (in my case called "external.xml", outside phone registrations, etc).

In the internal.xml file, we'll make some changes to accommodate the LAN network. Its a big file, but just a small tweak needs to be made. I'd edit the file and search for rtp-ip, sip-ip, ext-rtp-ip, and ext-sip-ip, respectively. Initially, you'll likely see "value=$${local_ip_v4}" in these four lines. Change the "value=" as below, using your own LAN IP address, for those four lines. Note that in the actual file there's a lot more stuff.

 <param name="rtp-ip" value="192.168.0.199"/>

 <param name="sip-ip" value="192.168.0.199"/>

Since I'm writting about a dual-homed example, we don't need a STUN server (i.e, we don't really need external rtp or sip) you can either leave the following two enteries alone or set them as I have below:

 <param name="ext-rtp-ip" value="192.168.0.199"/>

 <param name="ext-sip-ip" value="192.168.0.199"/>

Another reminder to use the LAN IP of your computer running FS, not mine. Also remember this is a dual-homed host in this particular example. You don't need to mess with this - the the internal.xml or external.xml files - if you have a single Ethernet port connected to a LAN with a gateway to the Internet. The settings "from the factory" will do just fine "out of the box" with a single homed host (though you will likely need STUN, not covered here). But, what fun is that?


EXTERNAL WAN

So, now we've got to do the same sort of thing to the /usr/local/freeswitch/conf/sip_profiles/external.xml file too right? Actually, while you could, one does not need to. Strangely enough, and to one's great happiness, this profile will go out and get the address of the WAN side and make the proper adjustments. Again, it will also, if a dynamic IP is being used, sense the address and any changes to it. I prefer the static IP and that's what I'm writting about here. The long and short of it is you don't have to change this file at this time. 


GETTING YOUR LAN PHONES REGISTERED

Also, we make a similar change elsewhere. In the /usr/local/freeswitch/conf/directory area sits a file that is - I hope - called default.xml. This file also has something to do with SIP on your LAN. In this case, what end-points (phones, soft phones) FS should expect to see, and other info (such as grouping these end-points into, well, groups). So, edit that file but use your LAN IP instead of mine, just like in the profiles above:

<!--the domain or ip (the right hand side of the @ in the addr-->
  <domain name="192.168.0.199">

While we're in the /usr/local/freeswitch/conf/directory area, let's consider how the SIP phones on your LAN are going to register with FS. I suggest using the pre-set 1000-1017 extension numbers - already set up for you - to configure and register your local SIP phones. If you need more, you'll know where/how to enter them in. So, while still in the default.xml file here - remember there are other default.xml files in other directories doing different things - note everything between the <groups> and </groups> tags. There are singular <group> </group> tags nested inside, setting up logical groupings of extension numbers into Sales, Billing, and Support. You can change these groupings to suite yourself, our go with what's there. Either way, you can see where this configuration is set now.

Also in this file, near the top, you'll see a pre-processor directive to go a directory below and include any .XML files found there. Leave this file, and cd down to the ./default directory. Just to be clear, we in directory /usr/local/freeswitch/conf/directory/default. Here's 1015.xml, one of the several you'll find there:

<include>
  <user id="1015">
    <params>
      <param name="password" value="$${default_password}"/>
      <param name="vm-password" value="1015"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="1015"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="Extension 1015"/>
      <!-- The value Extension 1015 can be personalized to your name, for instance, if you'd like -->
      <variable name="effective_caller_id_number" value="1015"/>
      <!-- Again, one could type in your main DID phone number instead of just 1015 if desired -->
      <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
      <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>

That's the whole XML file. Now, these can get more complicated, but this is right there and ready to go. While the variables pretty much tell you what they do, the point here is that if you need to modify something about a LAN SIP phone's "personal info", you come to /usr/local/freeswitch/conf/directory/default and look at the *.xml files here. And/or go "up" one level and look in the default.xml file there. NOTE: On my system, there's another default.xml file here. Keep in mind that while the names are the same, where they are has a lot to do with their function, so associate the place with the name. It gets easier with time.  

CHANGING THE PASSWORD FOR SIP REGISTRATION ON YOUR LAN

Last thing: You'll want to change the default password that your LAN SIP phones (or external phones for that matter) will use when authenticating. Leaving the one set up "out of the box" is a security risk since anyone could know it. So, make up your own. Edit the /usr/local/freeswitch/conf/vars.xml file. Near the top you'll see:

 <X-PRE-PROCESS cmd="set" data="default_password=1234"/>
  <!-- Did you change it yet? -->

Change the default_password parameter to some other four numbers that appeal to you. Now, only SIP phones that know the secret can register. Having done all this, set up your LAN SIP phones. Assign each a number between 1000 and 1017 for starters because these are already configured. Point your SIP phones the the IP address we've hard coded in the profiles (192.168.0.199 in my example). Be sure to use the new password you just set up.  


APPLYING YOUR CHANGES AND CHECKING YOUR WORK

Once you've made all these changes, either reloadxml the FS or simply stop and start it (don't know how? See "Stuff I Ought to Know Before Starting" section). To check, after the reload or restart, issue this command at the FS command line:

freeswitch at FreeSwitch.local> sofia status profile internal

You'll get something like the following if all is well:

API CALL [sofia(status profile internal)] output:
=================================================================================================
Name             	internal
Domain Name      	N/A
DBName           	sofia_reg_internal
Pres Hosts       	
Dialplan         	XML
Context          	public
Challenge Realm  	auto_to
RTP-IP           	192.168.0.199
Ext-RTP-IP       	192.168.0.199
SIP-IP           	192.168.0.199
Ext-SIP-IP       	192.168.0.199
URL              	sip:mod_sofia at 192.168.0.199:5060
BIND-URL         	sip:mod_sofia at 192.168.0.199:5060
HOLD-MUSIC       	local_stream://moh
OUTBOUND-PROXY   	N/A
CODECS           	G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
TEL-EVENT        	101
DTMF-MODE        	rfc2833
CNG              	13
SESSION-TO       	0
MAX-DIALOG       	0
NOMEDIA          	false
LATE-NEG         	false
PROXY-MEDIA      	false
AGGRESSIVENAT    	false
STUN-ENABLED     	true
STUN-AUTO-DISABLE	false
CALLS-IN         	2
FAILED-CALLS-IN  	0
CALLS-OUT        	3
FAILED-CALLS-OUT 	0

Registrations:
=================================================================================================
Call-ID:    	e6c864e9c4a3d at 192.168.0.80
User:       	1013 at 192.168.0.199
Contact:    	"user" <sip:1013 at 192.168.0.80:5062;transport=udp;fs_nat=yes;fs_path=sip%013%40192.168.0.80%062%3Btransport%3Dudp>
Agent:      	Grandstream GXP2000 1.1.6.46
Status:     	Registered(UDP-NAT)(unknown) EXP(2009-09-23 16:37:10)
Host:       	FreeSwitch.local
IP:         	192.168.0.80
Port:       	5062
Auth-User:  	1013
Auth-Realm: 	192.168.0.199

Call-ID:    	88ee646b068da at 192.168.0.41
User:       	1012 at 192.168.0.199
Contact:    	"user" <sip:1012 at 192.168.0.41:5062;transport=udp;user=phone;fs_nat=yes;fs_path=sip%012%192.168.0.41%3A5062%3Btransport%udp%3Buser%3Dphone>
Agent:      	Grandstream GXP2000 1.1.5.15
Status:     	Registered(UDP-NAT)(unknown) EXP(2009-09-23 16:37:12)
Host:       	FreeSwitch.local
IP:         	192.168.0.41
Port:       	5062
Auth-User:  	1012
Auth-Realm: 	192.168.0.199

Call-ID:    	45ee6e1b083da at 192.168.0.57
User:       	1017 at 192.168.0.199
Contact:    	"user" <sip:1017 at 192.168.0.57:5062;transport=udp;fs_nat=yes;fs_path=sip%17%40192.168.0.57%62%3Btransport%3Dudp>
Agent:      	Grandstream GXP2000 1.1.5.15
Status:     	Registered(UDP-NAT)(unknown) EXP(2009-09-23 16:37:15)
Host:       	FreeSwitch.local
IP:         	192.168.0.57
Port:       	5062
Auth-User:  	1017
Auth-Realm: 	192.168.0.199

=================================================================================================

Note that everwhere there's an IP address (RTP_IP, SIP_IP, EXT_RTP_IP, EXT_SIP_IP), its the LAN Ethernet address on our FS box or the address of a LAN SIP phone (under Registrations). That's what we want. The internal.xml profile is listening ONLY on the LAN. 

Do the same thing for the external profile:

freeswitch at FreeSwitch.local> sofia status profile external
API CALL [sofia(status profile external)] output:
=================================================================================================
Name             	external
Domain Name      	N/A
DBName           	sofia_reg_external
Pres Hosts       	
Dialplan         	XML
Context          	public
Challenge Realm  	auto_to
RTP-IP           	yy.yy.yy.yy
Ext-RTP-IP       	yy.yy.yy.yy
SIP-IP           	yy.yy.yy.yy
Ext-SIP-IP       	yy.yy.yy.yy
URL              	sip:mod_sofia at yy.yy.yy.yy:5080
BIND-URL         	sip:mod_sofia at yy.yy.yy.yy:5080
HOLD-MUSIC       	local_stream://moh
OUTBOUND-PROXY   	N/A
CODECS           	PCMU,PCMA,GSM
TEL-EVENT        	101
DTMF-MODE        	rfc2833
CNG              	13
SESSION-TO       	0
MAX-DIALOG       	0
NOMEDIA          	false
LATE-NEG         	false
PROXY-MEDIA      	false
AGGRESSIVENAT    	false
STUN-ENABLED     	true
STUN-AUTO-DISABLE	false
CALLS-IN         	1
FAILED-CALLS-IN  	0
CALLS-OUT        	1
FAILED-CALLS-OUT 	0

Registrations:
=================================================================================================
=================================================================================================

Here, we have only the external Ethernet static-ip address showing up. Again, that's as it should be. This profile is listening only to the WAN and not interfearing with the LAN. Note that in my case, I don't have any registrations on this profile. That's OK, but I could set things up for phones to register from the WAN if that was desirable (which in our case, it is not as yet). As we'll see, these two will work together well for out-going and in-coming calls. Read on.



REGISTERING WITH A SIP PROVIDER

So that we can make and receive calls, we'll need to register with a SIP/PSTN provider. Getting one is outside the scope of this document, but check out the Internet, many are available. You'll have to have FS register with your "gateway provider" so things can work. Here's how we did it: In /usr/local/freeswitch/conf/sip_profiles/external, I created a file called urbancom.xml (happens to be the name of my provider). As long as this file is in that directory, FS will use it to register with my SIP/PSTN provider, making calling "in and out" possible. Here's what's inside:

<include>
  <gateway name="xxx.xxx.xxx.xxx">
     <!-- Remember, you'll need this name for your "out-bound calls" extension -->
     <param name="username" value="8158381212"/>
     <!-- Natch', your own username, ours is like a NPANXXTN -->
     <param name="password" value="somekindapassw0rd"/>
     <!-- Natch', your own password -->
     <param name="expire-seconds" value="60"/>
     <!-- Time out value, 60 works for us -->
     <param name="register" value="true"/>
     <!-- You want to register, so "true" -->
     <param name="register-transport" value="udp"/>
     <!-- You'll probably use udp, but ask your provider -->
     <param name="retry-seconds" value="30"/>
     <!-- Retry parameter, 30 seconds works for us -->
     <param name="caller-id-in-from" value="false"/>
     <!-- Use the callerid of an inbound call in the from field on outbound calls via this gateway. We don't, up to you. -->
     <param name="contact-params" value="tport=5060"/>
     <!-- Port 5060 is customary, but again, ask the provider -->
     <param name="ping" value="25"/>
     <!-- We like to send a ping just to keep some traffic traversing and keep us registered -->
  </gateway>
</include>

That's the whole file. The "name" has to be something DNS can translate or an actual IP address. Clearly, I used the latter. Username and password come from the provider, as do the transport value (could be tcp, ask the provider) and the tport value (5060 is typical, but ask the...you get the message). I like the ping value, kinda making sure the link is kept alive in the absence of other traffic.

Now, we want to set something to happen "call-wise". That is, when our LAN phones register with FS and dial a phone number, something good will transpire. Here's one way to get that done: We'll add an "extension" - not to be confused with a physical telephone extension - to what's called a "context". Contexts are just containers for extensions. These context extensions are really directions to FS, telling it how to react when certain things take place. Our "thing" in this case is a registered sip phone dialing a 10 digit number, trying to call someone on the outside.

To make this work, we add the following "extension" to the "context" in the /usr/local/freeswitch/conf/dialplan/default.xml file. Don't confuse this default.xml with the one in /usr/local/freeswitch/conf/directory, different files doing different things. Enter the following in as the first extension in the file, note the <extension> tags that delimit the text you're inputting. What's the point? Well, this paragraph lets FS know what to do when an outgoing call is dialed. Note the <action application> tag. This directs FS to use a specific gateway, the one we just set up, to "get the call out".

   <! -- Mike stole from elsewhere: Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here -->
   <extension name="UrbanOut">
     <! -- Call this what you want, above. Can be anything that makes sense to you. -->
     <condition field="destination_number" expression="^(1{0,1}\d{10})$">
     <!-- The above translates to: "a 10 digit number, with or without a preceeding 0 or 1" -->
       <action application="set" data="effective_caller_id_number=18158381212"/>
       <!-- Change the above effective_caller_id_number to your own DID. -->
       <!-- If your provider does not provide ringback (180 or 183) you may simulate ringback by uncommenting the following line. -->
       <!-- action application="ringback" /-->
       <action application="bridge" data="sofia/gateway/xxx.xxx.xxx.xxx/$1"/>
       <!-- Change the above to your own gateway name, between "/gateway/" and "/$1" -->
     </condition>
   </extension>

NOTE: One can know if the gateway is registered or not, plus other very useful information, by using the following command at the FS command line:

freeswitch at FreeSwitch.local> sofia status

Don't forget to hit "enter". That will get you something like the following:

API CALL [sofia(status)] output:

    Name                Type                       Data                        State
=================================================================================================
   internal            profile          sip:mod_sofia at 192.168.0.199:5060      RUNNING (0)
   external            profile            sip:mod_sofia at yy.yy.yy.yy:5080      RUNNING (0)
   xxx.xxx.xxx.xxx     gateway           sip:8158381212 at xxx.xxx.xxx.xxx       REGED
   default               alias                                  internal      ALIASED
   192.168.0.199         alias                                  internal      ALIASED
=================================================================================================
2 profiles 2 aliases

Note that my gateway (check the "Type" column) is in state REGED, meaning it is registered with my provider and all is well with the world. If it isn't, check with your provider. If they're nice, they'll look at your registration attempt and provide clues as to what's not right. Hey, its a start.

While were at it, also note that the internal profile is up and running. That's what we messed with in the first place to direct calls to the gateway. Lo, the beautiful symmetry of it all. Further, the internal profile is listening on the LAN address (192.168.0.199 happens to be ours) while the external listens to the outside world on a different Ethernet static IP (it isn't our real one so don't try to hack it) as well as a different UDP port (a logical port for UDP, 5080 for the outside, 5060 for the inside here). All make sense? The alias types are just that, different names for the internal and external profiles. Used for some FS shorthand we won't talk about at the moment.


LETS MAKE SOME CALLS (OUT-GOING)

At this point, your FS has SIP phones registered locally, a SIP/PSTN gateway that it is registered to, and SIP profiles listening for work to do. Your SIP phone - there are many types so I assume you know how to configure and use your device - should itself show as registered to the FS. So, go ahead. Make that call out. I called my cellphone the first time. If all is well, and if your output from above looks like it should then all IS well, your outside phone is ringing. Rejoice.   


IN-COMING CALLS, RINGING A GROUP

OK, that was fun. But, one would want people to be able to call FS from the outside too. It would also be nice if FS would ring my SIP phone so I could take the call. As a matter of fact, at least in my office, we like all the phones to ring and whoever is available takes the call. One way to set up incoming calls in this way (ring all extensions in a group) is the following:

First, one has to have a group to ring. Forget, for a second, that there are groups already setup in the /usr/local/freeswitch/conf/directory. There is a feature of FS that allows one, from the SIP phone itself, to join a group that FS can then use. Go to each SIP phone in turn and dial extension 8101 (or *8101 on my PfSense version). 81 means "I'm putting this phone into a group". The last two digits tell what group, 01 in this case. You'll hear a tone letting you know it worked. Hang up and go join the next phone to group 01 until you've done 'em all. 

NOTE: Want to delete a phone from the group (it will still be able to call out)? Dial extension 8001 (or *8001 in my case on PfSense) and that particular phone will be deleted from the group. Keep in mind that once you are in the group, that group will persist in FS. You can re-register, restart FS or even reboot the computer FS is running on. You have to take action to get an extension out of the group (which is a good thing). 

Also note that if you'd like to see how 8001, 8101, or any other "logic" works in FS on your LAN, edit the /usr/local/freeswitch/conf/dialplan/default.xml file. Search for 80, for instance. This file contains "contexts" which themselves containg "extensions". Here, and extention is logic for taking some action when FS sees a particular event take place (like someone dialing 8201). We are at the heart of FS in this file, don't forget about it or what its for. 

Now, we have our group, 01 in this case (could be anything between 00 and 99). Dial extension 8201 (or, again, *8201 depending) and all phones joined to group 01 will ring. Fun stuff. Now, let's get FS to use that group to ring the same group. To do so, go to the /usr/local/freeswitch/conf/dialplan/public directory. I created a file that would be read first by using the file name 00_inbound_did.xml. The name is significant, especially the leading 00s. In this case, its the only file here but one could have many. Anyway, enter the following in that file, correcting for your own private info: 

<include>
  <extension name="Urban1212">
  <!-- Call it what you'd like. -->
    <condition field="destination_number" expression="^(8158381212)$">
    <!-- When FS sees the incoming number above - please use your own DID - it will engage the actions below -->
      <action application="set" data="domain_name=192.168.0.199"/>
      <!-- again, use the domain for your LAN from earlier in this document -->
      <action application="transfer" data="*8201 XML default"/>
      <!-- Here's the group ring. I need the *, you may not. But, ringing 8201, just like from your phone, will ring the whole group -->
      <!-- The "*8201 XML default" string means: find extension *8201, its in the default.xml file in the dialplan directory. -->
    </condition>
  </extension>
</include>

OK, having done the above, reloadxml or stop and start your FS. Then, dial in from a cellphone or other outside line. All phone in your group should ring. The DID of your calling phone should show. Now, we've got something useful going.  

Now, this is pretty powerful stuff. You could create another file in /usr/local/freeswitch/conf/dialplan/public and call it 01_inbound_did.xml. There one could set up another group (or single phone) to be rung if one has another DID. Much is possible. But, we'll keep to getting things functional and operational. 



MOVING ON TO MORE ADVANCED TOPICS


VOICE MAIL (OR "HOW TO ACCESS SOME THINGS THAT FS IS PREPARED TO DO")

If one has a registered phone (at this point in our document, only possible on the internal profile) then you have a voicemail extension at your disposal too. To check if an extension is in fact registered with your FS, go to your console screen and type:

freeswitch at FreeSwitch.local> sofia status profile internal

As in the scection on APPLYING YOUR CHANGES AND CHECKING YOUR WORK, one sees output about the internal profile itself, followed by all registered extensions for the profile. 

Now, we're going to use the FS dialplan file to figure out how to access voicemail. You'll likely want to do things like create a message for people to hear when they get VM. Also, you will want to hear what messages are left there, delteing some, leaving others. How? We'll answer that question; in the process learning  more about FS dialplans, contexts, extensions, and how to understand and use them. 

NOTE: You may recall a line in the REGISTERING WITH A SIP PROVIDOR section stating in passing that a dial plan containted "contexts". Contexts themselves are containters for instruction sets called "extensions". Don't confuse the term  "extension" here with a SIP phone or other device (which has an extension number) and these instruction blocks within contexts.


A CLOSER LOOK AT YOUR DEFAULT DIAL PLAN

Let's edit the /usr/local/freeswitch/conf/dialplan/default.xml file. Near the top of the file, one notes a <context=default> tag. All the way at the bottom of the file, one sees the closing </context> tag. In between these tags, one can see many groupings, deliniated with the <extension="somename"> and </extension> tags. Each of these extensions is actually a reaction to an event. The event being serviced is seen in the <condition> tag set, inside each extension. Usually, that condition is a number that someone has dialed. That could be an outside caller coming into your FS via your gateway provider, or a registered phone on your LAN doing the dialing. When FS "sees" a number, coming or going, it looks at the dial plans, matching that number to each extension's <condition>. FS then performs the <actions> within the matching extension.

NOTE: If the pattern matching syntax within the  "expresion=" of the <condition> tag makes no sense, please have a look at "regular expressions" ("regex" for short) and how they work. There's a great regex primer on the freeswitch.org site that's well worth finding and reading it you need it. However, that's outside the scope of this document. Nonetheless, you'll need some familiarity with regex's to understand conditions.

So, now that you're in the file and have the gist of what's going on here, search for the string "voicemail". In mine, courtesy of PfSense (pfsense.org), I find five extensions dealing with voicemail. We'll talk about each a little bit. Once finished, you'll know a lot more about voice mail on my FS. Yours will likely be similar. Here's my five, right from ~/conf/dialplan/default.xml on my system:

    <!-- voicemail operator extension -->
    <extension name="operator">
      <condition field="destination_number" expression="^\*operator$|^0$">
        <action application="set" data="transfer_ringback=$${hold_music}"/>
        <action application="transfer" data="1000 XML features"/>
      </condition>
    </extension>

The "operator" extension above is invoked by the string *operator or a dialed 0. On PfSense, that author likes to use a "*" with special extensions. Others do not. You can set it how you like it. If you have an extension like my "operator", go to a registered phone and hit 0. Note that you'll need a "real" extension 1000 on your LAN to be able to use this. We'll provide a way to put this to work later.   

    <!-- voicemail main2 extension -->
    <extension name="vmain2">
      <condition field="destination_number" expression="^vmain2$|^\*97$|^\*4000$">
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="voicemail" data="check default ${domain_name}"/>
      </condition>
    </extension>

In extension "vmain2", one can type *97, *4000 or a literal vmain2 to access the <action> here. Looking at the last <action> tag, one notes voicemail being invoked to check a user's VM contents. You'll need your user ID and password to gain access. Try this one out, if you have it, and follow the prompts. 

    <!-- voicemail main extension -->
    <extension name="vmain">
      <condition field="destination_number" expression="^vmain$|^\*98$">
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="voicemail" data="check default ${domain_name} ${sip_from_user}"/>
      </condition>
    </extension>

Much like vmai2, you can type *98 (or set it how you'd like it) for this VM entry. Note the <action> here works thus: if you call from your own phone extension, you'll only need the password to gain entry. The invocation of voicemail here uses the number you are dialing from as the user. 

    <extension name="send_to_voicemail_5digits">
      <condition field="destination_number" expression="^\*99(\d{5})$">
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
        <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
      </condition>
    </extension>

Above, a handy extension to transfer a caller to a five-digit extension, if you have them. Mine are all four digit, 1000-1019, the defaults. You could change this to handle any extension lenth you like. 

    <extension name="send_to_voicemail_4digits">
      <condition field="destination_number" expression="^\*99(\d{4})$">
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
        <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
      </condition>
    </extension>

Finally, an extension so that anyone taking a call can transfer that party to any four digit extension they'd like (much like the 5 digit above this). Just hit "tranfer" on your phone, and then enter *991013 to go to 1013's voicemail without ringing his/her phone. Nice. Remember, if your FS isn't configured as above, you can do so with the extension as you see them here, or by modifiying them to suit yourself. 

Now that you can get around in this particular default.xml, have a look at other .xml files in the /usr/local/freeswitch/conf/dialplan directory. Note that we've been here before in the  IN-COMING CALLS, RINGING A GROUP section of this document. There, we were directing in-coming calls to the proper default.xml extension via the ~/conf/dialplan/public subdirectory. Hopefully, this is all beginning to make sense as a whole. 

So, back to Voicemail. Now, you can access VM from your registered phone, set up your own voice prompt, and administer VM in general. How did we figure that out? By reading the "instructions" in the default dial plan. There's more to know, and a lot more that can be done. The idea is to build on what you've learned up to this point. Now, let's tie that in with calls coming into the system to make our FS even more useful.  



INCOMING CALLS RING A GROUP AND GOES TO VM AFTER NO ANSWER

This time, I'm just going to "lay out" some files that are accessed, in order, when an incoming call hits FS. Actually, FS isn't looking at the literal file, it has read them into the running application (assuming you've told it to). If not, FS can't "see" them so be sure you've reloadxml'd or shutdown/started FS. So, have a look and you'll see the flow FS follows to find out what to do. Let's say someone on the "outside" dials us in Illinois at 8158381212.


First, the call "comes in" and FS wants to know where to send it. Since this is a public incoming call, it looks for a match in ~/conf/dialplan/public/ and finds the match - in my case - in 01_incoming_did.xml:

     <include>
       <extension name="Urban1212">
       <!-- Call it what you'd like. -->
         <condition field="destination_number" expression="^(8158381212)$">
         <!-- When FS sees the incoming number above  please use your own DID  it will engage the actions below -->
           <action application="set" data="domain_name=192.168.0.199"/>
           <!-- again, use the domain for your LAN from earlier in this document -->
           <action application="transfer" data="8201 XML default"/>
           <!-- Go to this extension, 8201, to find more instructions. -->
         </condition>
       </extension>
     </include>

Notice above that I changed the transfer extension from *8201 (IN-COMING CALLS, RINGING A GROUP) to a new extension, 8201 (no preceeding *), seen below. I did that so I could add the <actions> that I needed and also demonstrate another way to ring groups, as you will see:  


Next, FS looks in ~/conf/dialplan/default.xml, just as it was told to, above, for the 8201 extension. Here it is:

    <!-- Mike Added: Incoming Urban Calls -->
    <extension name="UrbanIn">
      <condition field="destination_number" expression="^(8201)$">
         <action application="set" data="call_timeout=20"/>
         <action application="set" data="domain_name=192.168.0.199"/>
         <action application="set" data="hangup_after_bridge=true"/>
         <action application="set" data="continue_on_fail=true"/>
         <action application="bridge" data="group/support@${domain_name}"/>
         <action application="answer"/>
         <action application="sleep" data="1000"/>
         <action application="voicemail" data="default 192.168.0.199 1013"/>
      </condition>
    </extension>

In this extension, I have the included actions do a few things. I set a timeout value to 20. That means if no one answers in that amount of time (from when the bridge is invoked later on), FS will skip down to the next action, invoking voicemail (via dialplan default.xml) for extension 1013. 

Also notice that instead of bridging to *8201, which I could have still done, I opted to use one of the groups already set up for me in ~/conf/directory/default.xml (There's a lot of default.xml files. But, being in different directories, they do completely different things. Actually, one could call them anything you'd like. For the moment, I'm sticking to the names as installed). 


Here's a snippet of ~/conf/directory/default.xml. I simply located the <group name> set equal to "support", and added the users I wanted to that group. Note that the same users can be in different groups if you'd like: 

    <group name="support">
      <users>
        <user id="1010" type="pointer"/>
        <user id="1012" type="pointer"/>
        <user id="1013" type="pointer"/>
        <user id="1017" type="pointer"/>
      </users>
    </group>

Assuming you've set up extension 1013 (the physical phone's voicemail session) with a message about everybody being busy and please leave us a call back number, you're good to go (remember to reloadxml or shutdown/start if you've changed XML files). My phone's (admittedly an older Grandstream GXP-2000) message light blinked red as soon as I called in and left one (I'm 1013). Using the appropriate numbers from A CLOSER LOOK AT YOUR DIAL PLAN, you can now get your VM and deal with it as you'd like.  



ADDING ANOTHER DID

Let's say that you would like to have more than one direct inward dial number for your home or business. We saw in the section on REGISTERING WITH A SIP PROVIDER who to have one DID registered. To have more than one, naturally, you will have to have another <gateway> set up as that one is. However, if you want to use the same providor, we'll have to change the syntax of the XML file in /usr/local/freeswitch/conf/sip_profiles/external. Have a look: 

my ~/conf/sip_profiles/external/urban.xml file for registering and using one DID:

<include>
  <gateway name="xxx.xxx.xxx.xxx">
     <!-- Remember, you'll need this name for your "out-bound calls" extension -->
     <param name="username" value="8158381212"/>
     <!-- Natch', your own username, ours is like a NPANXXTN -->
     <param name="password" value="somekindapassw0rd"/>
     <!-- Natch', your own password -->
     <param name="expire-seconds" value="60"/>
     <!-- Time out value, 60 works for us -->
     <param name="register" value="true"/>
     <!-- You want to register, so "true" -->
     <param name="register-transport" value="udp"/>
     <!-- You'll probably use udp, but ask your provider -->
     <param name="retry-seconds" value="30"/>
     <!-- Retry parameter, 30 seconds works for us -->
     <param name="caller-id-in-from" value="false"/>
     <!-- Use the callerid of an inbound call in the from field on outbound calls via this gateway. We don't, up to you. -->
     <param name="contact-params" value="tport=5060"/>
     <!-- Port 5060 is customary, but again, ask the provider -->
     <param name="ping" value="25"/>
     <!-- We like to send a ping just to keep some traffic traversing and keep us registered -->
  </gateway>
</include>


Here's the same file, registering with the same providor, but now we have two registrations, one for each DID. Check the comments in the XML code below:


<include>
  <gateway name="urban1212">
	<!-- Any name that makes sense -->
        <param name="username" value="8158381212"/>
        <param name="password" value="somekindapassw0rd"/>
        <param name="realm" value="xxx.xxx.xxx.xxx"/>
	<!-- The gateway name from the "One DID" example. -->
        <param name="expire-seconds" value="60"/>>
        <!--/// do not register if value="false"///-->
        <param name="register" value="true"/>
        <param name="register-transport" value="udp"/>
        <param name="retry-seconds" value="30"/>
        <!--Use the callerid of an inbound call in the from field on outbound calls via this gateway -->
        <param name="caller-id-in-from" value="false"/>
        <param name="contact-params" value="tport=5060"/>
        <!--send an options ping every x seconds, failure will unregister and/or mark it down-->
        <param name="ping" value="25"/>
  </gateway>
  <gateway name="urban1234">
	<!-- Any name that makes sense, but not the one from above. The gateway names must be unique. --> 
        <param name="username" value="8158381234"/>
        <param name="password" value="anotherpassword1"/>
        <param name="realm" value="xxx.xxx.xxx.xxx"/>
	<!-- The gateway name from the "One DID" example. This is the same as above, but its OK here. -->
        <param name="expire-seconds" value="60"/>>
        <!--/// do not register if value="false"///-->
        <param name="register" value="true"/>
        <param name="register-transport" value="udp"/>
        <param name="retry-seconds" value="30"/>
        <!--Use the callerid of an inbound call in the from field on outbound calls via this gateway -->
        <param name="caller-id-in-from" value="false"/>
        <param name="contact-params" value="tport=5060"/>
        <!--send an options ping every x seconds, failure will unregister and/or mark it down-->
        <param name="ping" value="25"/>
  </gateway>
</include>


Notice that I had to use a different gateway name in this example for each DID registration, even though the same providor is registering both DIDs. This was done using the <param name="realm" value=xxx.xxx.xxx.xxx"/> tag. Of course, use your own gatway IP address. If you just double the entries of the single DID example, only one will actually register. You could use the syntax of the two DID example for one DID. As you can see, there's more than one way to do things. 

IMPORTANT: If you only make the changes above - assuming you have a working system with the "one DID" setup - you'll find that you can not make outgoing calls anymore (but incoming calls still work). Why? Remember in REGISTERING WITH A SIP PROVIDOR, we told FS what to do with out-going calls. That is, to use your newly registered gatwway. In the "two DID" example, we've changed the name of that gateway. Had to, can't use the same one twice. So, pick one of your DID gateway names above and update the extension in ~/conf/dialplan/default.xml for out-going calls (I use the urban1212 gateway, see below).

    <!-- Mike Added: Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here -->
    <extension name="UrbanOut">
     <condition field="destination_number" expression="^(1{0,1}\d{10})$">
       <action application="set" data="effective_caller_id_number=18158384096"/>
       <!-- If your provider does not provide ringback (180 or 183) you may simulate ringback by uncommenting the following line. -->
       <!-- action application="ringback" /-->
       <action application="bridge" data="sofia/gateway/urban1212/$1"/>
     </condition>
    </extension>

From jerry.richards at teotech.com  Fri Oct  2 09:40:28 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 2 Oct 2009 09:40:28 -0700
Subject: [Freeswitch-users] Call Forward All/Busy/No-Answer
Message-ID: <ABB03FEA28D4411393F2389EF02C9EA0@greyhawk.tonecommander.com>

How would I configure FS to Call Forward All or Call Forward when Busy or
Call Forward when No-Answer?  Can this be done at the server, rather than at
the phone?

Best Regards,
Jerry



From mike at van.lammeren.net  Fri Oct  2 09:51:11 2009
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Fri, 2 Oct 2009 12:51:11 -0400
Subject: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
In-Reply-To: <1254426084.3779.29.camel@sodium>
References: <5d2828f0909290720y6bb4997uc4b9afba4b0fe74c@mail.gmail.com>
	<5d2828f0910010945q30ed76e2x50d6ab3b13bc8f88@mail.gmail.com>
	<32C47FA5-774E-4584-890B-0A066F5CFE14@me.com>
	<191c3a030910011214i3a9eaad1xf242ecf19dc80da3@mail.gmail.com>
	<1254426084.3779.29.camel@sodium>
Message-ID: <5d2828f0910020951n784c569ak5af4bf4f25a8ac00@mail.gmail.com>

I only mentioned the OS I used as a reference for people. If they want to do
the same thing on another OS, then they might not have apt-get, etc.
Mike van Lammeren

On Thu, Oct 1, 2009 at 3:41 PM, Hadley Rich <hads at nice.net.nz> wrote:

> On Thu, 2009-10-01 at 14:14 -0500, Anthony Minessale wrote:
> > can we do it without advertising to use ubuntu =D
> > We don't like encouraging our users to use bleeding edge OS for our
> > own sanity with debugging.
>
> I understand your stance, though if we're talking about Ubuntu 8.04 LTS
> (Long Term Support - 5 years) it's not really bleeding edge anymore. 18
> months ago when it was released it may have been a little, but the LTS
> releases still aren't as bleeding edge as the standard support in
> between releases.
>
> hads
> --
> http://nicegear.co.nz
> New Zealand's Open Source Hardware Supplier
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at van.lammeren.net  Fri Oct  2 09:56:22 2009
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Fri, 2 Oct 2009 12:56:22 -0400
Subject: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
In-Reply-To: <0ca401ca42cf$8a709d20$9f51d760$@com>
References: <5d2828f0909290720y6bb4997uc4b9afba4b0fe74c@mail.gmail.com>
	<5d2828f0910010945q30ed76e2x50d6ab3b13bc8f88@mail.gmail.com>
	<32C47FA5-774E-4584-890B-0A066F5CFE14@me.com>
	<191c3a030910011214i3a9eaad1xf242ecf19dc80da3@mail.gmail.com>
	<0ca401ca42cf$8a709d20$9f51d760$@com>
Message-ID: <5d2828f0910020956w5c3fd2dfi481bb2952b0beb85@mail.gmail.com>

The load balancer listens to the virtual IP address, and port-forwards to
one of the FreeSWITCH boxes. Each FreeSWITCH box listens for the same
virtual IP address for SIP registrations and connections, which is what
FreeSWITCH needs to bind to. All other traffic actually travels over their
real IP address, which is what the FreeSWITCH servers would use to talk to
each other.

On Thu, Oct 1, 2009 at 3:44 PM, Peder <peder at networkoblivion.com> wrote:

>  Looking thru the example, it looks like each box has a real address of
> 21, 22 or 23 and they all have a loopback of .17, right?  So even though
> they connections are being load balanced, each box really thinks it is .17
> and each client that connects thinks it is connecting to .17, right?  If
> that?s the case, how does a client on one box call a client on the other
> box?  Since every box thinks it is .17 how would you bridge to another user
> on another box that also thinks it is .17?  Or am I totally missing how it
> works?
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Thursday, October 01, 2009 2:14 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
>
>
>
> can we do it without advertising to use ubuntu =D
> We don't like encouraging our users to use bleeding edge OS for our own
> sanity with debugging.
> Not to say you are not allowed to I just don't want to encourage it =p
>
>
>  On Thu, Oct 1, 2009 at 1:12 PM, Even Andr? Fiskvik <grevenx at me.com>
> wrote:
>
> That's very cool Mike!
>
>
>
> I'm going to try to configure four boxes with this as well (Btw, did you
> use physical hardware or virtualization?)
>
> and see how it goes. I followed Daniel Aliaman's blog as well, but I can
> try it again with the tips
>
> you provided on FreeSWITCH config to see if I can get it working properly
> this time.
>
> We did the setup on CentOS, but I wouldn't think that would be any issue.
>
>
>
> Perhaps you or we could write up a complete guide about this on the wiki
> since this is an scenario
>
> commonly used? Also it would be great if we could outline possible issues
> (and even better solutions)
>
> to this kind of setup with regards to stuff like conferencing, bridging
> between registered users and presence.
>
>
>
>
>
> Best regards,
>
> Even Andr?
>
>
>
>
>
> On 1. okt. 2009, at 18.45, Mike van Lammeren wrote:
>
>
>
>  Guess what? I have two FreeSWITCH servers working behind UltraMonkey,
> using heartbeat and ldirectord for load-balancing, fail-over and high
> availability! I'm probably not the first one to do it, but as near as Google
> and I can tell, I'm the first one to write about it.
>
>
>
> Here's how you can duplicate my setup:
>
>
>
> 1. Install Ubuntu Server 8 on four machines, either real or VM.
>
> 2. Compile and install FreeSWITCH v1.0.4 from source on two machines,
> following these instructions:
> http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start
>
> 3. Configure both FreeSWITCH boxes, and make sure they are both working.
>
> 4. Follow (most of) these instructions from Daniel Aliaman's blog. They
> were written for Asterisk, but since a SIP connection is a SIP connection,
> most of the document applies to FreeSWITCH:
>
> http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf
>
>
>
> The one problem I ran into was the IP address and port to which FreeSWITCH
> was bound. The default is to use the primary address, which works great
> out-of-the-box for everything else. When a client tried to register, all it
> got back was an ICMP error -- Destination Unreachable, Port Unreachable.
> That error is returned when no sockets are listening for UDP packets. To get
> FreeSWITCH to listen for your Virtual IP, you need to set it in two places:
>
>
>
> 5. In /opt/freeswitch/conf/vars.xml, set "bind_server_ip".
>
> 6. In /opt/freeswitch/conf/sip_profiles/internal.xml, set "sip-ip".
>
>
>
> That should do it! If you have any success, please report to this list.
>
>
>
> Keep in mind that if you want to do something like conferencing between two
> registered clients, then you have to deal with the fact that the clients may
> or may not be on the same box.
>
>
>
> Mike van Lammeren
>
>
>
>
>
>
>
> On Tue, Sep 29, 2009 at 10:20 AM, Mike van Lammeren <mike at van.lammeren.net>
> wrote:
>
>
>
> On Mon, Sep 28, 2009 at 9:05 PM, "Even Andr? Fiskvik" <grevenx at me.com
> > wrote:
>
> From: "Even Andr? Fiskvik" <grevenx at me.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Mon, 28 Sep 2009 22:52:13 +0200
> Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
>
> I have been working with a similar setup myself, but for some reason I
> ended up ditching the
>
> UltraMonkey setup because I just couldn't get it to work right.
>
>
>
> It's been quite a while since my effort, so I don't remember what the exact
> issue was.
>
> I got registrations to work, but had some other sip-dialog issues.
>
>
>
> We have since then changed over to running OpenSIPs as a loadbalancer in
> front of
>
> multiple FreeSWITCH instances. This setup is still in testing, but
> seemlingy works fine
>
> (and if it doesn't, it's my own fault for writing a bad opensips config).
>
>
>
> After we have done some more testing I can create a wiki-page with config
> details.
>
>
>
>
>
> Best regards,
>
> Even Andr?
>
>
>
>
>
> Thanks, Even, that would be great! I might have to give up on the
> ultramonkey solution, since I can't find anyone who has made it work. It's
> too bad, because it would fit well with the rest of our architecture.
>
>
>
> Mike van Lammeren
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at van.lammeren.net  Fri Oct  2 09:58:16 2009
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Fri, 2 Oct 2009 12:58:16 -0400
Subject: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
In-Reply-To: <32C47FA5-774E-4584-890B-0A066F5CFE14@me.com>
References: <5d2828f0909290720y6bb4997uc4b9afba4b0fe74c@mail.gmail.com>
	<5d2828f0910010945q30ed76e2x50d6ab3b13bc8f88@mail.gmail.com>
	<32C47FA5-774E-4584-890B-0A066F5CFE14@me.com>
Message-ID: <5d2828f0910020958o1176ab71rc361d9b2e96d6e58@mail.gmail.com>

I am running the servers on the free version of VMware's ESX platform, but
only for development purposes. We will be setting up real machines sometime
in Spring 2010.

On Thu, Oct 1, 2009 at 2:12 PM, Even Andr? Fiskvik <grevenx at me.com> wrote:

> That's very cool Mike!
> I'm going to try to configure four boxes with this as well (Btw, did you
> use physical hardware or virtualization?)
> and see how it goes. I followed Daniel Aliaman's blog as well, but I can
> try it again with the tips
> you provided on FreeSWITCH config to see if I can get it working properly
> this time.
> We did the setup on CentOS, but I wouldn't think that would be any issue.
>
> Perhaps you or we could write up a complete guide about this on the wiki
> since this is an scenario
> commonly used? Also it would be great if we could outline possible issues
> (and even better solutions)
> to this kind of setup with regards to stuff like conferencing, bridging
> between registered users and presence.
>
>
> Best regards,
> Even Andr?
>
>
> On 1. okt. 2009, at 18.45, Mike van Lammeren wrote:
>
> Guess what? I have two FreeSWITCH servers working behind UltraMonkey, using
> heartbeat and ldirectord for load-balancing, fail-over and high
> availability! I'm probably not the first one to do it, but as near as Google
> and I can tell, I'm the first one to write about it.
> Here's how you can duplicate my setup:
>
> 1. Install Ubuntu Server 8 on four machines, either real or VM.
> 2. Compile and install FreeSWITCH v1.0.4 from source on two machines,
> following these instructions:
> http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start
>  <http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start>3. Configure both
> FreeSWITCH boxes, and make sure they are both working.
> 4. Follow (most of) these instructions from Daniel Aliaman's blog. They
> were written for Asterisk, but since a SIP connection is a SIP connection,
> most of the document applies to FreeSWITCH:
> http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf
>
> <http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf>The one
> problem I ran into was the IP address and port to which FreeSWITCH was
> bound. The default is to use the primary address, which works great
> out-of-the-box for everything else. When a client tried to register, all it
> got back was an ICMP error -- Destination Unreachable, Port Unreachable.
> That error is returned when no sockets are listening for UDP packets. To get
> FreeSWITCH to listen for your Virtual IP, you need to set it in two places:
>
> 5. In /opt/freeswitch/conf/vars.xml, set "bind_server_ip".
> 6. In /opt/freeswitch/conf/sip_profiles/internal.xml, set "sip-ip".
>
> That should do it! If you have any success, please report to this list.
>
> Keep in mind that if you want to do something like conferencing between two
> registered clients, then you have to deal with the fact that the clients may
> or may not be on the same box.
>
> Mike van Lammeren
>
>
>
> On Tue, Sep 29, 2009 at 10:20 AM, Mike van Lammeren <mike at van.lammeren.net
> > wrote:
>
>>
>> On Mon, Sep 28, 2009 at 9:05 PM, "Even Andr? Fiskvik" <grevenx at me.com
>> > wrote:
>>
>>> From: "Even Andr? Fiskvik" <grevenx at me.com>
>>> To: freeswitch-users at lists.freeswitch.org
>>> Date: Mon, 28 Sep 2009 22:52:13 +0200
>>> Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
>>> I have been working with a similar setup myself, but for some reason I
>>> ended up ditching theUltraMonkey setup because I just couldn't get it to
>>> work right.
>>>
>>> It's been quite a while since my effort, so I don't remember what the
>>> exact issue was.
>>> I got registrations to work, but had some other sip-dialog issues.
>>>
>>> We have since then changed over to running OpenSIPs as a loadbalancer in
>>> front of
>>> multiple FreeSWITCH instances. This setup is still in testing, but
>>> seemlingy works fine
>>> (and if it doesn't, it's my own fault for writing a bad opensips config).
>>>
>>> After we have done some more testing I can create a wiki-page with config
>>> details.
>>>
>>>
>>> Best regards,
>>> Even Andr?
>>>
>>>
>> Thanks, Even, that would be great! I might have to give up on the
>> ultramonkey solution, since I can't find anyone who has made it work. It's
>> too bad, because it would fit well with the rest of our architecture.
>>
>> Mike van Lammeren
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From davis.erwin at gmail.com  Fri Oct  2 08:50:58 2009
From: davis.erwin at gmail.com (Erwin Davis)
Date: Fri, 2 Oct 2009 11:50:58 -0400
Subject: [Freeswitch-users] looking for qualified and cheap TISP
Message-ID: <a38a91d50910020850v5e19b8e0r751823b9d1e4ba5d@mail.gmail.com>

Hi, I installed internal freeSWITCH in my LAN and want to see if I can dial
out / dial in. Could anyone suggest one Telephone Service provider which is
capable of connecting with FreeSWITCH and CHEAP/even FREE if possible? At
this moment, I want to prove it is working with the real outside world.
Thanks,

e
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From jerry.richards at teotech.com  Fri Oct  2 10:12:45 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 2 Oct 2009 10:12:45 -0700
Subject: [Freeswitch-users] FS Does Not Relay Presence
	PUBLISHToSubscribing Phones
In-Reply-To: <ebbc2d550910012014ic655d84tbf000c084a98e51d@mail.gmail.com>
References: <B4B4D2747EEC4852B41A7D8D2DA36006@greyhawk.tonecommander.com><191c3a030910010935s69e65569v76f4d26503a94d96@mail.gmail.com><9B5214D274044021A57C6857844C10BA@greyhawk.tonecommander.com>
	<ebbc2d550910012014ic655d84tbf000c084a98e51d@mail.gmail.com>
Message-ID: <E407DD09040D408F8A130C2BFAFFCD17@greyhawk.tonecommander.com>

Okay, I put a log up on the pastebin that shows the PUBLISH event coming
from a CounterPath Bria Professional phone.  For some reason, FS is getting
an error and not relaying the presence status to the subscriber.
 
Best Regards,
Jerry


  _____  

From: Jo?o Mesquita [mailto:jmesquita at freeswitch.org] 
Sent: Thursday, October 01, 2009 8:14 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Does Not Relay Presence
PUBLISHToSubscribing Phones


Piece of advice, don't ask, just do it. ;)

jmesquita


On Thu, Oct 1, 2009 at 3:29 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:


If you have time to take a look, I could put a trace in the pastebin?
 
Jerry


  _____  


From: Jerry Richards [mailto:jerry.richards at teotech.com] 

Sent: Thursday, October 01, 2009 10:29 AM 

To: 'freeswitch-users at lists.freeswitch.org'

Subject: RE: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
ToSubscribing Phones


I am using two Bria Professional Version 2.5.4 Build 54835 softphones.
 
Thanks,
Jerry


  _____  


From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 

Sent: Thursday, October 01, 2009 9:36 AM 

To: freeswitch-users at lists.freeswitch.org

Subject: Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
ToSubscribing Phones


which phone is it,
we tested it with eyebeam and it appears to work for us.



On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards <jerry.richards at teotech.com>
wrote:



By the way, I see the following lines at the FS console, which might be a
clue as to why this is happening.  Could someone point me toward what might
cause this?  I set the "manage-presence" parameter to "true" in each XML
file where I saw it defined.

[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[WARNING] sofia_presence.c:565  192.168.72.38 is an alias, skipping


Best Regards,
Jerry



-----Original Message-----
From: Jerry Richards [mailto:jerry.richards at teotech.com]
Sent: Wednesday, September 30, 2009 9:12 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones

I have two phones configured to subscribe to each other's presence status.
When I change the presence status in one phone, I see the SIP PUBLISH
message going to FS, but I don't see FS relaying that presence status to the
subscribing phone.  Does anyone know why?

Best Regards,
Jerry


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:213-799-1400



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




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From Russell.Mosemann at cune.org  Fri Oct  2 10:24:18 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Fri, 2 Oct 2009 17:24:18 -0000
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <87f2f3b90910020908p1aa5f5d4o8b69894cd4a58bef@mail.gmail.com>
Message-ID: <20091002172418.9E8AF2E19CE@mail.cune.org>

Michael Collins <msc at freeswitch.org> said:

> do something like:
> name => XYZ Corp
> number => 8005551212

I was expecting that information to be filled with the caller name and
number. It doesn't really help if someone calls from outside the
business, and it looks like my business is calling me. Doesn't OpenZAP
extract caller information from a PRI T1?

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From mgende at gendesign.com  Fri Oct  2 10:27:55 2009
From: mgende at gendesign.com (Michael Gende)
Date: Fri, 2 Oct 2009 12:27:55 -0500
Subject: [Freeswitch-users] looking for qualified and cheap TISP
In-Reply-To: <a38a91d50910020850v5e19b8e0r751823b9d1e4ba5d@mail.gmail.com>
References: <a38a91d50910020850v5e19b8e0r751823b9d1e4ba5d@mail.gmail.com>
Message-ID: <a87d43fd0910021027q4d4766c9n2930a92d1b7fce3f@mail.gmail.com>

Hey Erwin,

Can't give any personal recommendations, but on the FS site, there's several
examples. Some have "free" or "cheap" in the name. Might be a good place to
start, plus the means to connect is demonstrated to-boot.

http://wiki.freeswitch.org/wiki/SIP_Provider_Examples

Regards,

Mike G.

On Fri, Oct 2, 2009 at 10:50 AM, Erwin Davis <davis.erwin at gmail.com> wrote:

> Hi, I installed internal freeSWITCH in my LAN and want to see if I can dial
> out / dial in. Could anyone suggest one Telephone Service provider which is
> capable of connecting with FreeSWITCH and CHEAP/even FREE if possible? At
> this moment, I want to prove it is working with the real outside world.
> Thanks,
>
>
>
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From mike at van.lammeren.net  Fri Oct  2 10:30:45 2009
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Fri, 2 Oct 2009 13:30:45 -0400
Subject: [Freeswitch-users] looking for qualified and cheap TISP
In-Reply-To: <a38a91d50910020850v5e19b8e0r751823b9d1e4ba5d@mail.gmail.com>
References: <a38a91d50910020850v5e19b8e0r751823b9d1e4ba5d@mail.gmail.com>
Message-ID: <2F7DF2F3-4A48-426C-91F6-9A2F79E17B06@van.lammeren.net>

Hello!

For dialing in, there are a number of sites that provide free DIDs,  
such as http://freephonelines.ca/ .

For dialing out, you can get 1.5 cents per minute calling to N.  
America from http://les.net/ .

Mike

On 2009-10-02, at 11:50 AM, Erwin Davis <davis.erwin at gmail.com> wrote:

> Hi, I installed internal freeSWITCH in my LAN and want to see if I  
> can dial out / dial in. Could anyone suggest one Telephone Service  
> provider which is capable of connecting with FreeSWITCH and CHEAP/ 
> even FREE if possible? At this moment, I want to prove it is working  
> with the real outside world. Thanks,
>
> e
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From anthony.minessale at gmail.com  Fri Oct  2 10:52:17 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 2 Oct 2009 12:52:17 -0500
Subject: [Freeswitch-users] FS Does Not Relay Presence
	PUBLISHToSubscribing Phones
In-Reply-To: <E407DD09040D408F8A130C2BFAFFCD17@greyhawk.tonecommander.com>
References: <B4B4D2747EEC4852B41A7D8D2DA36006@greyhawk.tonecommander.com>
	<191c3a030910010935s69e65569v76f4d26503a94d96@mail.gmail.com>
	<9B5214D274044021A57C6857844C10BA@greyhawk.tonecommander.com>
	<ebbc2d550910012014ic655d84tbf000c084a98e51d@mail.gmail.com>
	<E407DD09040D408F8A130C2BFAFFCD17@greyhawk.tonecommander.com>
Message-ID: <191c3a030910021052y2e074b4r648024507206e25@mail.gmail.com>

connect to sqlite directly with sqlite3 app and try that sql stmt and see
why it doesn't match anything.

sqlite3 /usr/local/freeswitch/db/sofia_reg_internal.db

select
sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Away','away','
192.168.72.38',sip_presence.status,sip_presence.rpid from sip_subscriptions
left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user
and sip_subscriptions.sub_to_host=sip_presence.sip_host and
sip_subscriptions.profile_name=sip_presence.profile_name) where
(event='presence'
or event='presence') and sub_to_user='1001' and (sub_to_host='192.168.72.38'
or presence_hosts like '%192.168.72.38%') and (sip_subscriptions.profile_name
= 'external' or sip_subscriptions.presence_hosts !=
sip_subscriptions.sub_to_host)


On Fri, Oct 2, 2009 at 12:12 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>  Okay, I put a log up on the pastebin that shows the PUBLISH event coming
> from a CounterPath Bria Professional phone.  For some reason, FS is getting
> an error and not relaying the presence status to the subscriber.
>
> Best Regards,
> Jerry
>
>  ------------------------------
> *From:* Jo?o Mesquita [mailto:jmesquita at freeswitch.org]
> *Sent:* Thursday, October 01, 2009 8:14 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] FS Does Not Relay Presence
> PUBLISHToSubscribing Phones
>
> Piece of advice, don't ask, just do it. ;)
>
> jmesquita
>
> On Thu, Oct 1, 2009 at 3:29 PM, Jerry Richards <jerry.richards at teotech.com
> > wrote:
>
>>  If you have time to take a look, I could put a trace in the pastebin?
>>
>> Jerry
>>
>>  ------------------------------
>>  *From:* Jerry Richards [mailto:jerry.richards at teotech.com]
>> *Sent:* Thursday, October 01, 2009 10:29 AM
>> *To:* 'freeswitch-users at lists.freeswitch.org'
>> *Subject:* RE: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
>> ToSubscribing Phones
>>
>>  I am using two Bria Professional Version 2.5.4 Build 54835 softphones.
>>
>> Thanks,
>> Jerry
>>
>>  ------------------------------
>>  *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com]
>> *Sent:* Thursday, October 01, 2009 9:36 AM
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
>> ToSubscribing Phones
>>
>>   which phone is it,
>> we tested it with eyebeam and it appears to work for us.
>>
>>
>> On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards <
>> jerry.richards at teotech.com> wrote:
>>
>>>
>>> By the way, I see the following lines at the FS console, which might be a
>>> clue as to why this is happening.  Could someone point me toward what
>>> might
>>> cause this?  I set the "manage-presence" parameter to "true" in each XML
>>> file where I saw it defined.
>>>
>>> [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal)
>>> [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
>>>   ...
>>> [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6)
>>> [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
>>>   ...
>>> [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external)
>>> [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
>>>   ...
>>> [WARNING] sofia_presence.c:565  192.168.72.38 is an alias, skipping
>>>
>>>
>>> Best Regards,
>>> Jerry
>>>
>>>
>>> -----Original Message-----
>>> From: Jerry Richards [mailto:jerry.richards at teotech.com]
>>> Sent: Wednesday, September 30, 2009 9:12 AM
>>> To: 'freeswitch-users at lists.freeswitch.org'
>>> Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones
>>>
>>> I have two phones configured to subscribe to each other's presence
>>> status.
>>> When I change the presence status in one phone, I see the SIP PUBLISH
>>> message going to FS, but I don't see FS relaying that presence status to
>>> the
>>> subscribing phone.  Does anyone know why?
>>>
>>> Best Regards,
>>> Jerry
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From csa at nowthor.com  Fri Oct  2 11:12:01 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Fri, 02 Oct 2009 14:12:01 -0400
Subject: [Freeswitch-users] looking for qualified and cheap TISP
In-Reply-To: <a38a91d50910020850v5e19b8e0r751823b9d1e4ba5d@mail.gmail.com>
References: <a38a91d50910020850v5e19b8e0r751823b9d1e4ba5d@mail.gmail.com>
Message-ID: <4AC64271.2050707@nowthor.com>

Hi!

Callcentric 
<http://wiki.freeswitch.org/wiki/Provider_Configuration:_Callcentric> 
offers a package called IP Freedom 
<http://www.callcentric.com/rate_plans01.php>. It costs nothing and will 
allow you to test FS.

Carlos

Erwin Davis wrote:
> Hi, I installed internal freeSWITCH in my LAN and want to see if I can 
> dial out / dial in. Could anyone suggest one Telephone Service 
> provider which is capable of connecting with FreeSWITCH and CHEAP/even 
> FREE if possible? At this moment, I want to prove it is working with 
> the real outside world. Thanks,
>
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From msc at freeswitch.org  Fri Oct  2 11:16:58 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 2 Oct 2009 11:16:58 -0700
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <20091002172418.9E8AF2E19CE@mail.cune.org>
References: <87f2f3b90910020908p1aa5f5d4o8b69894cd4a58bef@mail.gmail.com>
	<20091002172418.9E8AF2E19CE@mail.cune.org>
Message-ID: <87f2f3b90910021116l209a44bcg124016e58735ca37@mail.gmail.com>

On Fri, Oct 2, 2009 at 10:24 AM, <Russell.Mosemann at cune.org> wrote:

> Michael Collins <msc at freeswitch.org> said:
>
> > do something like:
> > name => XYZ Corp
> > number => 8005551212
>
> I was expecting that information to be filled with the caller name and
> number. It doesn't really help if someone calls from outside the
> business, and it looks like my business is calling me. Doesn't OpenZAP
> extract caller information from a PRI T1?
>

Can you pastebin a dialplan snippet (or put it here) so I can see what
you're doing?
-MC
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From jerry.richards at teotech.com  Fri Oct  2 11:28:07 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 2 Oct 2009 11:28:07 -0700
Subject: [Freeswitch-users] FS Does Not Relay
	PresencePUBLISHToSubscribing Phones
In-Reply-To: <191c3a030910021052y2e074b4r648024507206e25@mail.gmail.com>
References: <B4B4D2747EEC4852B41A7D8D2DA36006@greyhawk.tonecommander.com><191c3a030910010935s69e65569v76f4d26503a94d96@mail.gmail.com><9B5214D274044021A57C6857844C10BA@greyhawk.tonecommander.com><ebbc2d550910012014ic655d84tbf000c084a98e51d@mail.gmail.com><E407DD09040D408F8A130C2BFAFFCD17@greyhawk.tonecommander.com>
	<191c3a030910021052y2e074b4r648024507206e25@mail.gmail.com>
Message-ID: <57502048DB624686B96B8AE76F697AA7@greyhawk.tonecommander.com>

I put the sqlite3 select query in the paste bin, and prior to that, I
entered the .dump command.  The select command came back with a "...>"
prompt which I don't understand.  I don't know enough about sqlite3 to know
what that means?
 
Best Regards,
Jerry


  _____  

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Friday, October 02, 2009 10:52 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Does Not Relay
PresencePUBLISHToSubscribing Phones


connect to sqlite directly with sqlite3 app and try that sql stmt and see
why it doesn't match anything.

sqlite3 /usr/local/freeswitch/db/sofia_reg_internal.db

select
sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_hos
t,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscripti
ons.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscripti
ons.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subsc
riptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,
'Away','away','192.168.72.38',sip_presence.status,sip_presence.rpid from
sip_subscriptions left join sip_presence on
(sip_subscriptions.sub_to_user=sip_presence.sip_user and
sip_subscriptions.sub_to_host=sip_presence.sip_host and
sip_subscriptions.profile_name=sip_presence.profile_name) where
(event='presence' or event='presence') and sub_to_user='1001' and
(sub_to_host='192.168.72.38' or presence_hosts like '%192.168.72.38%') and
(sip_subscriptions.profile_name = 'external' or
sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host)



On Fri, Oct 2, 2009 at 12:12 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:


Okay, I put a log up on the pastebin that shows the PUBLISH event coming
from a CounterPath Bria Professional phone.  For some reason, FS is getting
an error and not relaying the presence status to the subscriber.
 
Best Regards,
Jerry


  _____  

From: Jo?o Mesquita [mailto:jmesquita at freeswitch.org] 
Sent: Thursday, October 01, 2009 8:14 PM 

To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Does Not Relay Presence
PUBLISHToSubscribing Phones


Piece of advice, don't ask, just do it. ;)

jmesquita


On Thu, Oct 1, 2009 at 3:29 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:


If you have time to take a look, I could put a trace in the pastebin?
 
Jerry


  _____  


From: Jerry Richards [mailto:jerry.richards at teotech.com] 

Sent: Thursday, October 01, 2009 10:29 AM 

To: 'freeswitch-users at lists.freeswitch.org'

Subject: RE: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
ToSubscribing Phones


I am using two Bria Professional Version 2.5.4 Build 54835 softphones.
 
Thanks,
Jerry


  _____  


From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 

Sent: Thursday, October 01, 2009 9:36 AM 

To: freeswitch-users at lists.freeswitch.org

Subject: Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
ToSubscribing Phones


which phone is it,
we tested it with eyebeam and it appears to work for us.



On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards <jerry.richards at teotech.com>
wrote:



By the way, I see the following lines at the FS console, which might be a
clue as to why this is happening.  Could someone point me toward what might
cause this?  I set the "manage-presence" parameter to "true" in each XML
file where I saw it defined.

[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[WARNING] sofia_presence.c:565  192.168.72.38 is an alias, skipping


Best Regards,
Jerry



-----Original Message-----
From: Jerry Richards [mailto:jerry.richards at teotech.com]
Sent: Wednesday, September 30, 2009 9:12 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones

I have two phones configured to subscribe to each other's presence status.
When I change the presence status in one phone, I see the SIP PUBLISH
message going to FS, but I don't see FS relaying that presence status to the
subscribing phone.  Does anyone know why?

Best Regards,
Jerry


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:213-799-1400



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org






-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:213-799-1400


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From rupa at rupa.com  Fri Oct  2 11:42:29 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 2 Oct 2009 13:42:29 -0500
Subject: [Freeswitch-users] FS Does Not Relay
	PresencePUBLISHToSubscribing Phones
In-Reply-To: <57502048DB624686B96B8AE76F697AA7@greyhawk.tonecommander.com>
References: <B4B4D2747EEC4852B41A7D8D2DA36006@greyhawk.tonecommander.com>
	<191c3a030910010935s69e65569v76f4d26503a94d96@mail.gmail.com>
	<9B5214D274044021A57C6857844C10BA@greyhawk.tonecommander.com>
	<ebbc2d550910012014ic655d84tbf000c084a98e51d@mail.gmail.com>
	<E407DD09040D408F8A130C2BFAFFCD17@greyhawk.tonecommander.com>
	<191c3a030910021052y2e074b4r648024507206e25@mail.gmail.com>
	<57502048DB624686B96B8AE76F697AA7@greyhawk.tonecommander.com>
Message-ID: <b39273530910021142p546680felc6c6ec19d86c922f@mail.gmail.com>

You are missing the trailing ;

On Fri, Oct 2, 2009 at 1:28 PM, Jerry Richards
<jerry.richards at teotech.com> wrote:
> I put the sqlite3 select query in the paste bin, and prior to that, I
> entered the .dump command.? The select command came back with a "...>"
> prompt which I don't understand.? I don't know enough about sqlite3 to know
> what that means?
>
> Best Regards,
> Jerry


-- 
-Rupa


From Russell.Mosemann at cune.org  Fri Oct  2 11:53:59 2009
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Fri, 2 Oct 2009 13:53:59 -0500
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <87f2f3b90910021116l209a44bcg124016e58735ca37@mail.gmail.com>
References: <87f2f3b90910020908p1aa5f5d4o8b69894cd4a58bef@mail.gmail.com><20091002172418.9E8AF2E19CE@mail.cune.org>
	<87f2f3b90910021116l209a44bcg124016e58735ca37@mail.gmail.com>
Message-ID: <E2C130F6CC9B4F59B069CEF8D0B0AC42@cune.pri>

> Can you pastebin a dialplan snippet (or put it here) so I can see what
> you're doing?
> -MC

It is the stock FS configuration with a small change. We're still testing things, getting them to work. This is from public.xml. It detects calls to internal 71xx extensions and transfers them. The transfer works. Do some additional variables need to be set here?

<extension name="public_extensions">
  <condition field="destination_number" expression="^(10[01][0-9]|71\d{2})$">
    <action application="transfer" data="$1 XML default"/>
  </condition>
</extension>

--
Russell Mosemann



From ujjval at simplesignal.com  Fri Oct  2 12:10:40 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Fri, 2 Oct 2009 12:10:40 -0700
Subject: [Freeswitch-users] Asterisk vs Freeswitch
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701C61EE3@EXMBXCLUS01.citservers.local>

Is there benchmark test results on how many simultaneous calls Freeswtich can do (with RTP anchored through it) vs the Asterisk.

For any hardware/CPU/Mem  that anyone may have performed this performance testing.

Any numbers on average how much Freeswitch scores over the Asterisk in terms of capacity will help.


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From msc at freeswitch.org  Fri Oct  2 12:18:30 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 2 Oct 2009 12:18:30 -0700
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <E2C130F6CC9B4F59B069CEF8D0B0AC42@cune.pri>
References: <87f2f3b90910020908p1aa5f5d4o8b69894cd4a58bef@mail.gmail.com>
	<20091002172418.9E8AF2E19CE@mail.cune.org>
	<87f2f3b90910021116l209a44bcg124016e58735ca37@mail.gmail.com>
	<E2C130F6CC9B4F59B069CEF8D0B0AC42@cune.pri>
Message-ID: <87f2f3b90910021218i434f67cbjc33073f8550c8d1a@mail.gmail.com>

On Fri, Oct 2, 2009 at 11:53 AM, Russell Mosemann <Russell.Mosemann at cune.org
> wrote:

> > Can you pastebin a dialplan snippet (or put it here) so I can see what
> > you're doing?
> > -MC
>
> It is the stock FS configuration with a small change. We're still testing
> things, getting them to work. This is from public.xml. It detects calls to
> internal 71xx extensions and transfers them. The transfer works. Do some
> additional variables need to be set here?
>
> <extension name="public_extensions">
>  <condition field="destination_number"
> expression="^(10[01][0-9]|71\d{2})$">
>    <action application="transfer" data="$1 XML default"/>
>  </condition>
> </extension>
>
> cool. can you pastebin a debug log on an incoming call?
-MC
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From kadantsev.d at gmail.com  Fri Oct  2 13:00:42 2009
From: kadantsev.d at gmail.com (Dmitry Kadantsev)
Date: Fri, 2 Oct 2009 22:00:42 +0200
Subject: [Freeswitch-users] Asterisk vs Freeswitch
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71701C61EE3@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701C61EE3@EXMBXCLUS01.citservers.local>
Message-ID: <681a20520910021300g131cd87j7c1fa6c6c9ec7f3e@mail.gmail.com>

Hi,

for example here: http://blogs.zdnet.com/Greenfield/?p=214

We *replaced* a cluster of *10 Asterisk* servers with a *single
FreeSwitch*server, said Chris Parker, director of systems for a large
publicly traded
CLEC. Parker says hes getting several hundred concurrent calls on a single,
dual-core box thats also doing all of the media processing, a
computationally intensive task.




--
Best regards,
Dmitry Kadantsev

http://www.kadantsev.com - Home page (MS Silverlight required)
http://www.doxwox.com - Best web meeting and online collaboration tool


On Fri, Oct 2, 2009 at 9:10 PM, Ujjval Karihaloo <ujjval at simplesignal.com>wrote:

>  Is there benchmark test results on how many simultaneous calls Freeswtich
> can do (with RTP anchored through it) vs the Asterisk.
>
>
>
> For any hardware/CPU/Mem  that anyone may have performed this performance
> testing.
>
>
>
> Any numbers on average how much Freeswitch scores over the Asterisk in
> terms of capacity will help.
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Russell.Mosemann at cune.org  Fri Oct  2 14:53:54 2009
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Fri, 2 Oct 2009 16:53:54 -0500
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <87f2f3b90910021218i434f67cbjc33073f8550c8d1a@mail.gmail.com>
References: <87f2f3b90910020908p1aa5f5d4o8b69894cd4a58bef@mail.gmail.com><20091002172418.9E8AF2E19CE@mail.cune.org><87f2f3b90910021116l209a44bcg124016e58735ca37@mail.gmail.com><E2C130F6CC9B4F59B069CEF8D0B0AC42@cune.pri>
	<87f2f3b90910021218i434f67cbjc33073f8550c8d1a@mail.gmail.com>
Message-ID: <8E079E8CB37E4FA49363D5315F6E878E@cune.pri>

> cool. can you pastebin a debug log on an incoming call?
> -MC

Here you go.

http://pastebin.freeswitch.org/10570

One thing I notice is that in the second line, the caller number is missing.

2009-10-02 16:42:39.539736 [NOTICE] ozmod_libpri.c:772 -- Ring on channel 1:1 (from  to 7100)

If libpri doesn't know the number, then it's probably not being sent by the Hicomm.

--
Russell Mosemann



From msc at freeswitch.org  Fri Oct  2 17:00:08 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 2 Oct 2009 17:00:08 -0700
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <8E079E8CB37E4FA49363D5315F6E878E@cune.pri>
References: <87f2f3b90910020908p1aa5f5d4o8b69894cd4a58bef@mail.gmail.com>
	<20091002172418.9E8AF2E19CE@mail.cune.org>
	<87f2f3b90910021116l209a44bcg124016e58735ca37@mail.gmail.com>
	<E2C130F6CC9B4F59B069CEF8D0B0AC42@cune.pri>
	<87f2f3b90910021218i434f67cbjc33073f8550c8d1a@mail.gmail.com>
	<8E079E8CB37E4FA49363D5315F6E878E@cune.pri>
Message-ID: <87f2f3b90910021700h3431bd78q3b4f6ced255ecee5@mail.gmail.com>

On Fri, Oct 2, 2009 at 2:53 PM, Russell Mosemann
<Russell.Mosemann at cune.org>wrote:

> > cool. can you pastebin a debug log on an incoming call?
> > -MC
>
> Here you go.
>
> http://pastebin.freeswitch.org/10570
>
> One thing I notice is that in the second line, the caller number is
> missing.
>
> 2009-10-02 16:42:39.539736 [NOTICE] ozmod_libpri.c:772 -- Ring on channel
> 1:1 (from  to 7100)
>
> If libpri doesn't know the number, then it's probably not being sent by the
> Hicomm.
>
> Exactly. Turn on q931 debugging and try again:

oz libpri debug 1 all

PB the results again and we'll check it out.
-MC
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From Russell.Mosemann at cune.org  Fri Oct  2 17:54:34 2009
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Fri, 2 Oct 2009 19:54:34 -0500
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <87f2f3b90910021700h3431bd78q3b4f6ced255ecee5@mail.gmail.com>
References: <87f2f3b90910020908p1aa5f5d4o8b69894cd4a58bef@mail.gmail.com><20091002172418.9E8AF2E19CE@mail.cune.org><87f2f3b90910021116l209a44bcg124016e58735ca37@mail.gmail.com><E2C130F6CC9B4F59B069CEF8D0B0AC42@cune.pri><87f2f3b90910021218i434f67cbjc33073f8550c8d1a@mail.gmail.com><8E079E8CB37E4FA49363D5315F6E878E@cune.pri>
	<87f2f3b90910021700h3431bd78q3b4f6ced255ecee5@mail.gmail.com>
Message-ID: <191DBBEF043B4EFE8F680B82C1038FC7@cune.pri>

> Exactly. Turn on q931 debugging and try again:
> 
> oz libpri debug 1 all
> PB the results again and we'll check it out.
> -MC

Here's the next one. I'm not sure what to look for, but nothing pops out right away.

http://pastebin.freeswitch.org/10571

--
Russell Mosemann



From thangappan143 at gmail.com  Fri Oct  2 21:24:42 2009
From: thangappan143 at gmail.com (Thangappan.M)
Date: Sat, 3 Oct 2009 09:54:42 +0530
Subject: [Freeswitch-users] Need Help in Getting DTMF
Message-ID: <7aa29e790910022124h320de58dp16f1a19aca08bb6@mail.gmail.com>

Dear all,

          I am in the process of implementing IVR server in Perl using event
outbound socket.

Let take the following scenario.

         There are three menus in the IVR. First menu will authenticate you,
second menu get the option value from you,. third menu will give the you the
result.

           You already know  all the numbers that you could give. So when
the call answered you are giving the value in ONE SHOT.

         Is it possible to get all the DTMF values in one shot in
freeswitch?

   It should have facility to recollect DTMF values and clear the DTMF
values.

-- 
Regards,
Thangappan.M
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From vinuth.madinur at gmail.com  Fri Oct  2 22:21:17 2009
From: vinuth.madinur at gmail.com (Vinuth Madinur)
Date: Sat, 3 Oct 2009 10:51:17 +0530
Subject: [Freeswitch-users] Need Help in Getting DTMF
In-Reply-To: <7aa29e790910022124h320de58dp16f1a19aca08bb6@mail.gmail.com>
References: <7aa29e790910022124h320de58dp16f1a19aca08bb6@mail.gmail.com>
Message-ID: <910309030910022221h31a1fcd0h9f41e1aa17f0c7a3@mail.gmail.com>

You can use play_and_get_digits command or the read command.
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_read
<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_read>

Thanks,
Vinuth.


On Sat, Oct 3, 2009 at 9:54 AM, Thangappan.M <thangappan143 at gmail.com>wrote:

> Dear all,
>
>           I am in the process of implementing IVR server in Perl using
> event outbound socket.
>
> Let take the following scenario.
>
>          There are three menus in the IVR. First menu will authenticate
> you, second menu get the option value from you,. third menu will give the
> you the result.
>
>            You already know  all the numbers that you could give. So when
> the call answered you are giving the value in ONE SHOT.
>
>          Is it possible to get all the DTMF values in one shot in
> freeswitch?
>
>    It should have facility to recollect DTMF values and clear the DTMF
> values.
>
> --
> Regards,
> Thangappan.M
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From keith.wood2000 at gmail.com  Fri Oct  2 23:44:50 2009
From: keith.wood2000 at gmail.com (Keith Wood)
Date: Sat, 3 Oct 2009 14:44:50 +0800
Subject: [Freeswitch-users] wav files compression
Message-ID: <b7737fe70910022344y72cdebc4ieb0b580ce7b7905f@mail.gmail.com>

I am working on an implementation for managing thousands of IVR within an
organization.  Right now, I am storing all audio files in wav format, but it
quickly become unmanagable because the size of these wav files ( 8 bits mono
) quickly consuming a lot of the disk space.

Is there anyway I can store those audio files and still have high quality
audio for IVR?  I know mp3 is smaller but freeswitch does not support it.

any ideas?

keith
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From dujinfang at gmail.com  Sat Oct  3 00:12:57 2009
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 3 Oct 2009 15:12:57 +0800
Subject: [Freeswitch-users] wav files compression
In-Reply-To: <b7737fe70910022344y72cdebc4ieb0b580ce7b7905f@mail.gmail.com>
References: <b7737fe70910022344y72cdebc4ieb0b580ce7b7905f@mail.gmail.com>
Message-ID: <23f91030910030012k572a544tcefa32c33dc7efef@mail.gmail.com>

FS support recording to mp3 directly through mod_shout but you might not
want to use that for performance reason.

You can use lame to convert .wav to .mp3 regularly( by crontab if you on
linux) or immediately after record(by using iwatch, or listening to event
socket to see when the record is done ).

2009/10/3 Keith Wood <keith.wood2000 at gmail.com>

>
> I am working on an implementation for managing thousands of IVR within an
> organization.  Right now, I am storing all audio files in wav format, but it
> quickly become unmanagable because the size of these wav files ( 8 bits mono
> ) quickly consuming a lot of the disk space.
>
> Is there anyway I can store those audio files and still have high quality
> audio for IVR?  I know mp3 is smaller but freeswitch does not support it.
>
> any ideas?
>
> keith
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From thangappan143 at gmail.com  Sat Oct  3 03:42:55 2009
From: thangappan143 at gmail.com (Thangappan.M)
Date: Sat, 3 Oct 2009 16:12:55 +0530
Subject: [Freeswitch-users] Need Help in Getting DTMF
In-Reply-To: <7aa29e790910022124h320de58dp16f1a19aca08bb6@mail.gmail.com>
References: <7aa29e790910022124h320de58dp16f1a19aca08bb6@mail.gmail.com>
Message-ID: <7aa29e790910030342n222d7d9q8170f0a8fa537840@mail.gmail.com>

Can you please give some example?

Because I have tried it using playAndGetDigits() application only.

my need is " User can give input at any time " It should be captured.

Is there any internal mechanism avail in freeswitch?
Or

Shall we do it?


On Sat, Oct 3, 2009 at 9:54 AM, Thangappan.M <thangappan143 at gmail.com>wrote:

> Dear all,
>
>           I am in the process of implementing IVR server in Perl using
> event outbound socket.
>
> Let take the following scenario.
>
>          There are three menus in the IVR. First menu will authenticate
> you, second menu get the option value from you,. third menu will give the
> you the result.
>
>            You already know  all the numbers that you could give. So when
> the call answered you are giving the value in ONE SHOT.
>
>          Is it possible to get all the DTMF values in one shot in
> freeswitch?
>
>    It should have facility to recollect DTMF values and clear the DTMF
> values.
>
> --
> Regards,
> Thangappan.M
>



-- 
Regards,
Thangappan.M
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From brian at freeswitch.org  Sat Oct  3 07:52:54 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 3 Oct 2009 09:52:54 -0500
Subject: [Freeswitch-users] wav files compression
In-Reply-To: <b7737fe70910022344y72cdebc4ieb0b580ce7b7905f@mail.gmail.com>
References: <b7737fe70910022344y72cdebc4ieb0b580ce7b7905f@mail.gmail.com>
Message-ID: <709C5BD7-C916-4757-9AEE-C862D7F60053@freeswitch.org>

MP3 is NOT recommend and if WAV files are too large you can mosey on  
down to the local Best Buy and snag 1.5TB of disk for like $119  
dollars.  Disk is cheap.

/b

On Oct 3, 2009, at 1:44 AM, Keith Wood wrote:

>
> I am working on an implementation for managing thousands of IVR  
> within an organization.  Right now, I am storing all audio files in  
> wav format, but it quickly become unmanagable because the size of  
> these wav files ( 8 bits mono ) quickly consuming a lot of the disk  
> space.
>
> Is there anyway I can store those audio files and still have high  
> quality audio for IVR?  I know mp3 is smaller but freeswitch does  
> not support it.
>
> any ideas?
>
> keith
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From frank at carmickle.com  Sat Oct  3 09:16:36 2009
From: frank at carmickle.com (Frank Carmickle)
Date: Sat, 3 Oct 2009 12:16:36 -0400
Subject: [Freeswitch-users] voiptalk.org register 904
Message-ID: <20091003161635.GG17256@base.carmickle.com>

I hope this is more helpful.

http://carmickle.com/fs-2009-10-02.txt

<include>
      <gateway name="voiptalk.org">
        <param name="username" value="81100068"/>
        <param name="password" value="my_pass"/>
        <param name="from-user" value="81100068"/>
        <param name="from-domain" value="voiptalk.org"/>
    <param name="proxy" value="voiptalk.org"/>
        <param name="expire-seconds" value="600"/>
        <param name="register" value="true"/>
        <param name="retry-seconds" value="30"/>
        <param name="extension" value="1000"/>
        <param name="contact-params" value="domain_name=$${domain}"/>
        <param name="context" value="public"/>
      </gateway>
</include>
 
Let me know what you think.  Thanks.
--FC


From diego.viola at gmail.com  Sat Oct  3 10:07:39 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 3 Oct 2009 17:07:39 +0000
Subject: [Freeswitch-users] wav files compression
In-Reply-To: <709C5BD7-C916-4757-9AEE-C862D7F60053@freeswitch.org>
References: <b7737fe70910022344y72cdebc4ieb0b580ce7b7905f@mail.gmail.com>
	<709C5BD7-C916-4757-9AEE-C862D7F60053@freeswitch.org>
Message-ID: <86a32abc0910031007ted4b6bfk1a567574af91ecae@mail.gmail.com>

Why is not recommended?

On Sat, Oct 3, 2009 at 2:52 PM, Brian West <brian at freeswitch.org> wrote:

> MP3 is NOT recommend and if WAV files are too large you can mosey on
> down to the local Best Buy and snag 1.5TB of disk for like $119
> dollars.  Disk is cheap.
>
> /b
>
> On Oct 3, 2009, at 1:44 AM, Keith Wood wrote:
>
> >
> > I am working on an implementation for managing thousands of IVR
> > within an organization.  Right now, I am storing all audio files in
> > wav format, but it quickly become unmanagable because the size of
> > these wav files ( 8 bits mono ) quickly consuming a lot of the disk
> > space.
> >
> > Is there anyway I can store those audio files and still have high
> > quality audio for IVR?  I know mp3 is smaller but freeswitch does
> > not support it.
> >
> > any ideas?
> >
> > keith
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sat Oct  3 10:27:54 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 3 Oct 2009 12:27:54 -0500
Subject: [Freeswitch-users] wav files compression
In-Reply-To: <86a32abc0910031007ted4b6bfk1a567574af91ecae@mail.gmail.com>
References: <b7737fe70910022344y72cdebc4ieb0b580ce7b7905f@mail.gmail.com>
	<709C5BD7-C916-4757-9AEE-C862D7F60053@freeswitch.org>
	<86a32abc0910031007ted4b6bfk1a567574af91ecae@mail.gmail.com>
Message-ID: <84976809-0961-410D-8733-0C0449365B1A@freeswitch.org>

Lets see... mp3 decoding is heavy compared to wav files.. its 2009 and  
disk is cheap and fast why worry about it?  Not sure you wanna scale  
mp3 playback to the same level you can wav files.

/b

On Oct 3, 2009, at 12:07 PM, Diego Viola wrote:

> Why is not recommended?
>
> On Sat, Oct 3, 2009 at 2:52 PM, Brian West <brian at freeswitch.org>  
> wrote:
> MP3 is NOT recommend and if WAV files are too large you can mosey on
> down to the local Best Buy and snag 1.5TB of disk for like $119
> dollars.  Disk is cheap.
>
> /b

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From steveu at coppice.org  Sat Oct  3 10:47:22 2009
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 04 Oct 2009 01:47:22 +0800
Subject: [Freeswitch-users] wav files compression
In-Reply-To: <86a32abc0910031007ted4b6bfk1a567574af91ecae@mail.gmail.com>
References: <b7737fe70910022344y72cdebc4ieb0b580ce7b7905f@mail.gmail.com>	<709C5BD7-C916-4757-9AEE-C862D7F60053@freeswitch.org>
	<86a32abc0910031007ted4b6bfk1a567574af91ecae@mail.gmail.com>
Message-ID: <4AC78E2A.2090501@coppice.org>

On 10/04/2009 01:07 AM, Diego Viola wrote:
> Why is not recommended?

Square peg. Round hole.
> On Sat, Oct 3, 2009 at 2:52 PM, Brian West <brian at freeswitch.org 
> <mailto:brian at freeswitch.org>> wrote:
>
>     MP3 is NOT recommend and if WAV files are too large you can mosey on
>     down to the local Best Buy and snag 1.5TB of disk for like $119
>     dollars.  Disk is cheap.
>
>     /b
>
>     On Oct 3, 2009, at 1:44 AM, Keith Wood wrote:
>
>     >
>     > I am working on an implementation for managing thousands of IVR
>     > within an organization.  Right now, I am storing all audio files in
>     > wav format, but it quickly become unmanagable because the size of
>     > these wav files ( 8 bits mono ) quickly consuming a lot of the disk
>     > space.
>     >
>     > Is there anyway I can store those audio files and still have high
>     > quality audio for IVR?  I know mp3 is smaller but freeswitch does
>     > not support it.
>     >
>     > any ideas?
>     >
>     > keith
>
Steve



From diego.viola at gmail.com  Sat Oct  3 11:17:26 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 3 Oct 2009 18:17:26 +0000
Subject: [Freeswitch-users] wav files compression
In-Reply-To: <4AC78E2A.2090501@coppice.org>
References: <b7737fe70910022344y72cdebc4ieb0b580ce7b7905f@mail.gmail.com>
	<709C5BD7-C916-4757-9AEE-C862D7F60053@freeswitch.org>
	<86a32abc0910031007ted4b6bfk1a567574af91ecae@mail.gmail.com>
	<4AC78E2A.2090501@coppice.org>
Message-ID: <86a32abc0910031117k17109746s905d65640d548e43@mail.gmail.com>

I see, does Ogg/Vorbis have the same problem?

Diego

On Sat, Oct 3, 2009 at 5:47 PM, Steve Underwood <steveu at coppice.org> wrote:

> On 10/04/2009 01:07 AM, Diego Viola wrote:
> > Why is not recommended?
>
> Square peg. Round hole.
> > On Sat, Oct 3, 2009 at 2:52 PM, Brian West <brian at freeswitch.org
> > <mailto:brian at freeswitch.org>> wrote:
> >
> >     MP3 is NOT recommend and if WAV files are too large you can mosey on
> >     down to the local Best Buy and snag 1.5TB of disk for like $119
> >     dollars.  Disk is cheap.
> >
> >     /b
> >
> >     On Oct 3, 2009, at 1:44 AM, Keith Wood wrote:
> >
> >     >
> >     > I am working on an implementation for managing thousands of IVR
> >     > within an organization.  Right now, I am storing all audio files in
> >     > wav format, but it quickly become unmanagable because the size of
> >     > these wav files ( 8 bits mono ) quickly consuming a lot of the disk
> >     > space.
> >     >
> >     > Is there anyway I can store those audio files and still have high
> >     > quality audio for IVR?  I know mp3 is smaller but freeswitch does
> >     > not support it.
> >     >
> >     > any ideas?
> >     >
> >     > keith
> >
> Steve
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sat Oct  3 11:39:03 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 3 Oct 2009 13:39:03 -0500
Subject: [Freeswitch-users] wav files compression
In-Reply-To: <86a32abc0910031117k17109746s905d65640d548e43@mail.gmail.com>
References: <b7737fe70910022344y72cdebc4ieb0b580ce7b7905f@mail.gmail.com>
	<709C5BD7-C916-4757-9AEE-C862D7F60053@freeswitch.org>
	<86a32abc0910031007ted4b6bfk1a567574af91ecae@mail.gmail.com>
	<4AC78E2A.2090501@coppice.org>
	<86a32abc0910031117k17109746s905d65640d548e43@mail.gmail.com>
Message-ID: <635CFF1F-BFDB-4C00-B04C-D1B59F1648AE@freeswitch.org>

Yes... Why add layers of bullshit on top of audio that is going to  
traverse the public phone network?  PCM raw or ulaw/alaw are the most  
optimal formats.

/b

On Oct 3, 2009, at 1:17 PM, Diego Viola wrote:

> I see, does Ogg/Vorbis have the same problem?
>
> Diego



From steveu at coppice.org  Sat Oct  3 11:47:07 2009
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 04 Oct 2009 02:47:07 +0800
Subject: [Freeswitch-users] wav files compression
In-Reply-To: <86a32abc0910031117k17109746s905d65640d548e43@mail.gmail.com>
References: <b7737fe70910022344y72cdebc4ieb0b580ce7b7905f@mail.gmail.com>	<709C5BD7-C916-4757-9AEE-C862D7F60053@freeswitch.org>	<86a32abc0910031007ted4b6bfk1a567574af91ecae@mail.gmail.com>	<4AC78E2A.2090501@coppice.org>
	<86a32abc0910031117k17109746s905d65640d548e43@mail.gmail.com>
Message-ID: <4AC79C2B.6020505@coppice.org>

On 10/04/2009 02:17 AM, Diego Viola wrote:
> I see, does Ogg/Vorbis have the same problem?
Yep. Anything designed for general purpose audio is going to be a poor 
choice when you want to achieve compact storage of narrowband voice.

> Diego
>
> On Sat, Oct 3, 2009 at 5:47 PM, Steve Underwood <steveu at coppice.org 
> <mailto:steveu at coppice.org>> wrote:
>
>     On 10/04/2009 01:07 AM, Diego Viola wrote:
>     > Why is not recommended?
>
>     Square peg. Round hole.
>     > On Sat, Oct 3, 2009 at 2:52 PM, Brian West <brian at freeswitch.org
>     <mailto:brian at freeswitch.org>
>     > <mailto:brian at freeswitch.org <mailto:brian at freeswitch.org>>> wrote:
>     >
>     >     MP3 is NOT recommend and if WAV files are too large you can
>     mosey on
>     >     down to the local Best Buy and snag 1.5TB of disk for like $119
>     >     dollars.  Disk is cheap.
>     >
>     >     /b
>     >
>     >     On Oct 3, 2009, at 1:44 AM, Keith Wood wrote:
>     >
>     > >
>     > > I am working on an implementation for managing thousands of IVR
>     > > within an organization.  Right now, I am storing all audio
>     files in
>     > > wav format, but it quickly become unmanagable because the size of
>     > > these wav files ( 8 bits mono ) quickly consuming a lot of the
>     disk
>     > > space.
>     > >
>     > > Is there anyway I can store those audio files and still have high
>     > > quality audio for IVR?  I know mp3 is smaller but freeswitch does
>     > > not support it.
>     > >
>     > > any ideas?
>
Steve



From diego.viola at gmail.com  Sat Oct  3 14:26:20 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 3 Oct 2009 21:26:20 +0000
Subject: [Freeswitch-users] CDR page reworked
Message-ID: <86a32abc0910031426q79417376n541edf2a22740275@mail.gmail.com>

Hi FreeSWITCH community.

I just wanted to say that I have reworked this page a bit as it was a bit
poor, feel free to add anything else on it.

http://wiki.freeswitch.org/wiki/CDR

Regards,

Diego
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From tculjaga at gmail.com  Sat Oct  3 16:38:22 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Sun, 4 Oct 2009 01:38:22 +0200
Subject: [Freeswitch-users] wav files compression
In-Reply-To: <86a32abc0910031007ted4b6bfk1a567574af91ecae@mail.gmail.com>
References: <b7737fe70910022344y72cdebc4ieb0b580ce7b7905f@mail.gmail.com>
	<709C5BD7-C916-4757-9AEE-C862D7F60053@freeswitch.org>
	<86a32abc0910031007ted4b6bfk1a567574af91ecae@mail.gmail.com>
Message-ID: <65d96fc80910031638q52c3a595w8cce2209c6cf61f5@mail.gmail.com>

also, you can store files in PCMA/PCMU format and avoid transcoding at
all... and as said disk space is cheap.. go get some...

On Sat, Oct 3, 2009 at 7:07 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Why is not recommended?
>
>
> On Sat, Oct 3, 2009 at 2:52 PM, Brian West <brian at freeswitch.org> wrote:
>
>> MP3 is NOT recommend and if WAV files are too large you can mosey on
>> down to the local Best Buy and snag 1.5TB of disk for like $119
>> dollars.  Disk is cheap.
>>
>> /b
>>
>> On Oct 3, 2009, at 1:44 AM, Keith Wood wrote:
>>
>> >
>> > I am working on an implementation for managing thousands of IVR
>> > within an organization.  Right now, I am storing all audio files in
>> > wav format, but it quickly become unmanagable because the size of
>> > these wav files ( 8 bits mono ) quickly consuming a lot of the disk
>> > space.
>> >
>> > Is there anyway I can store those audio files and still have high
>> > quality audio for IVR?  I know mp3 is smaller but freeswitch does
>> > not support it.
>> >
>> > any ideas?
>> >
>> > keith
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> > users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From nandy1925 at gmail.com  Sun Oct  4 00:28:38 2009
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Sun, 4 Oct 2009 15:28:38 +0800
Subject: [Freeswitch-users] wav files compression
In-Reply-To: <65d96fc80910031638q52c3a595w8cce2209c6cf61f5@mail.gmail.com>
References: <b7737fe70910022344y72cdebc4ieb0b580ce7b7905f@mail.gmail.com>
	<709C5BD7-C916-4757-9AEE-C862D7F60053@freeswitch.org>
	<86a32abc0910031007ted4b6bfk1a567574af91ecae@mail.gmail.com>
	<65d96fc80910031638q52c3a595w8cce2209c6cf61f5@mail.gmail.com>
Message-ID: <7d0bfd8c0910040028y79ad956cxc289eec410beb8c1@mail.gmail.com>

agree that WAV/PCMA/PCMU formats are best for performance. you can use
mp3/ogg ONLY to archive recorded files.
/nandy

On Sun, Oct 4, 2009 at 7:38 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> also, you can store files in PCMA/PCMU format and avoid transcoding at
> all... and as said disk space is cheap.. go get some...
>
>
> On Sat, Oct 3, 2009 at 7:07 PM, Diego Viola <diego.viola at gmail.com> wrote:
>
>> Why is not recommended?
>>
>>
>> On Sat, Oct 3, 2009 at 2:52 PM, Brian West <brian at freeswitch.org> wrote:
>>
>>> MP3 is NOT recommend and if WAV files are too large you can mosey on
>>> down to the local Best Buy and snag 1.5TB of disk for like $119
>>> dollars.  Disk is cheap.
>>>
>>> /b
>>>
>>> On Oct 3, 2009, at 1:44 AM, Keith Wood wrote:
>>>
>>> >
>>> > I am working on an implementation for managing thousands of IVR
>>> > within an organization.  Right now, I am storing all audio files in
>>> > wav format, but it quickly become unmanagable because the size of
>>> > these wav files ( 8 bits mono ) quickly consuming a lot of the disk
>>> > space.
>>> >
>>> > Is there anyway I can store those audio files and still have high
>>> > quality audio for IVR?  I know mp3 is smaller but freeswitch does
>>> > not support it.
>>> >
>>> > any ideas?
>>> >
>>> > keith
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> > users
>>> > http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From raffaele.p.guidi at gmail.com  Sun Oct  4 03:13:05 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Sun, 4 Oct 2009 12:13:05 +0200
Subject: [Freeswitch-users]  Freeswitch as a softphone - presence?
Message-ID: <b0a8792f0910040313s3db46076sa527dd04110bc9f2@mail.gmail.com>

Hi, I was wondering how FreeSWITCH could notify presence through sofia
gateways - the basic idea is to use it as a softphone, of course.
Thanks,
   Raffaele
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From mcampbellsmith at gmail.com  Sun Oct  4 06:03:05 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sun, 4 Oct 2009 23:03:05 +1000
Subject: [Freeswitch-users] Detecting a fax
Message-ID: <33c87fa30910040603p7929df26l88f51fc7d256408f@mail.gmail.com>

Hi

I was hoping someone could help me to setup the fax detection / tone
detection application.

I want to be able to transfer an incoming fax to a specific extension.
 In my default.xml file, I have the following (extracted):

    <extension name="1000">
      <condition field="destination_number" expression="^(10[01][0-9])$">
        <action application="fax_detect"/>
        <action application="tone_detect" data="fax 1100 r +5000
transfer fax XML features" />

I can't get the fax to be detected and transferred.  Is there any way
this can be done?

Thanks!


From mike at jerris.com  Sun Oct  4 14:23:02 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 4 Oct 2009 17:23:02 -0400
Subject: [Freeswitch-users] Outbound INVITE rejected with 480 Temp
	Unavail, Reason "MANDATORY_IE_MISSING"
In-Reply-To: <6789F39C3E7544B7BDBCD0CA706B060E@greyhawk.tonecommander.com>
References: <6789F39C3E7544B7BDBCD0CA706B060E@greyhawk.tonecommander.com>
Message-ID: <0E5C8DD9-FED6-4367-AC53-62FCCE3FFFC1@jerris.com>

there is a profile param to enable 3pcc.  It should be documented in  
the default configs.

Mike

On Sep 29, 2009, at 5:22 PM, Jerry Richards wrote:

> Hello All,
>
> I have an internal extension that needs to send an INVITE without  
> SDP body
> (Content Length 0).  Freeswitch is replying with 480 Temporarily  
> Unavailable
> with reason "MANDATORY_IE_MISSING".  Would anyone know what I need  
> to do to
> enable this?



From mike at jerris.com  Sun Oct  4 14:34:01 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 4 Oct 2009 17:34:01 -0400
Subject: [Freeswitch-users] Problem with subscription expire
In-Reply-To: <4AC46DEB.3090506@ewetel.de>
References: <4AC46DEB.3090506@ewetel.de>
Message-ID: <B9D110F1-7602-4107-9151-B622F7F04A46@jerris.com>

This sounds like a bug in the snom to me, we keep changing the expire  
on to the future so it should never expire in the first place.  You  
will have to look at a longer running sip trace to see what exactly is  
going on.

Mike

On Oct 1, 2009, at 4:52 AM, Helmut Kuper wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> it seems exired subsciptions are never cleared in FS.
>
> A look into sofia_presence.c confirms explains this
>
> /* negative in exptime means keep bumping up sub time to avoid a snafu
> where every device has it's own rules about subscriptions
>                           that somehow barely resemble the RFC not  
> that
> I blame them because the RFC MAY be amibiguous and SHOULD be deleted.
>                           So to avoid the problem we keep resetting  
> the
> expiration date of the subscription so it never expires.
>
>                           Eybeam completely ignores this option and
> most other subscription-state: directives from rfc3265 and still  
> expires.
>                           Polycom is happy to keep upping the
> subscription expiry back to the original time on each new notify.
>                           The rest ... who knows...?
>
>                        */
>
> For some reasons subscriptions created by Snom phones are filling up  
> the
> sip_subscriptions table over time. This leads to some kind of DOS by  
> FS
> against the subscribing phone ... The subscribtions are  
> differentiate by
> call-id. This can be explained by RFC 3842 chapter 3.6 where expired
> subscriptions must be renewed with a NEW call-id. Because there is no
> hint about unsubscribing the old subscription I guess the clean up
> process has to be done by FS.
>
> Any way to get FS to do this job? Since there is no creation date or
> expire value which represents the expire as a timestamp I have no  
> way to
> clean up the table manually via sql and cronjob - except cleaning the
> whole table ...
>
>
> A further (but background) question is, why do the subscriptions  
> expire
> in snom phones at all ...



From mike at jerris.com  Sun Oct  4 14:40:30 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 4 Oct 2009 17:40:30 -0400
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <1b46b4e80910011229k31f7233du10b0fc661730d3f3@mail.gmail.com>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<191c3a030909291122t770af037j5b1068197b31de87@mail.gmail.com>
	<1b46b4e80909291542k71440965g75bb3bfcf38e8005@mail.gmail.com>
	<1b46b4e80910011229k31f7233du10b0fc661730d3f3@mail.gmail.com>
Message-ID: <F86F950D-9106-49C9-81CA-38B3EB2072F4@jerris.com>

can you send a link of a text sip trace please.

On Oct 1, 2009, at 3:29 PM, Nicolas Brenner wrote:

> Any ideas about this?
>
> The SIP provider is offering H323, but I'm not quite sure about  
> that, is mod_opal working right?
>
> Thanks!
>
> Nicolas
>
> On Tue, Sep 29, 2009 at 6:42 PM, Nicolas Brenner <nicolas at medularis.com 
> > wrote:
> Anthony, thanks. Below are my config files for the two gateways from  
> the sip trace. Both files are located in conf/directory/default.
>
> ---------------------
>
> redvoiss.xml (the one that works)
>
> <include>
>   <user id="gateway_redvoiss">
>     <gateways>
>       <gateway name="redvoiss-pp">
>         <param name="username" value="xxxxxxx"/>
>         <param name="password" value="xxxxxxx"/>
>         <param name="from-domain" value="pxextmy.redvoiss.net"/>
>         <param name="realm" value="pxextmy.redvoiss.net"/>
>         <param name="proxy" value="pxextmy.redvoiss.net"/>
>         <param name="from-user" value="xxxxxxx"/>
>         <param name="caller-id-in-from" value="false"/>
>         <param name="expire-seconds" value="600"/>
>         <param name="register" value="true"/>
>         <param name="retry_seconds" value="5"/>
>         <param name="extension" value="2010"/>
>         <param name="context" value="public"/>
>         <param name="codec-prefs" value="G729"/>
>         <param name="rfc2833-pt" value="101"/>
>       </gateway>
>     </gateways>
>     <params>
>       <param name="password" value="4321"/>
>     </params>
>   </user>
> </include>
>
> ---------------------
>
> orange.xml (the one that doesn't work)
>
> <include>
>   <user id="gateway_orange">
>     <gateways>
>       <gateway name="orange">
>         <param name="username" value="xxxxxxx"/>
>         <param name="password" value="xxxxxxx"/>
>         <param name="from-domain" value="216.72.10.39"/>
>         <param name="realm" value="216.72.10.39"/>
>         <param name="proxy" value="216.72.10.39"/>
>         <param name="from-user" value="xxxxxxx"/>
>         <param name="caller-id-in-from" value="false"/>
>         <param name="expire-seconds" value="600"/>
>         <param name="register" value="true"/>
>         <param name="retry_seconds" value="5"/>
>         <param name="extension" value="2011"/>
>         <param name="context" value="public"/>
>         <param name="codec-prefs" value="G729"/>
>         <param name="rfc2833-pt" value="101"/>
>       </gateway>
>     </gateways>
>     <params>
>       <param name="password" value="4321"/>
>     </params>
>   </user>
> </include>
>
> ---------------------
>
> If I remove the register=true param for the non-working gateway, I  
> don't get the registration error on the cli, but then all call  
> attempts get rejected with a 401 Unauthorized, and I get a hangup  
> cause of NORMAL_UNSPECIFIED.
>
>
> Best,
>
> Nicolas
>
>
>
> On Tue, Sep 29, 2009 at 2:22 PM, Anthony Minessale <anthony.minessale at gmail.com 
> > wrote:
> 900 level errors are sofia internal errors so probably something is  
> wrong with your gateway config xml.
> if you want to send it with any critical info replaced with XXX  
> maybe we can see the issue for you.
>
>
>
> On Tue, Sep 29, 2009 at 1:05 PM, Nicolas Brenner <nicolas at medularis.com 
> > wrote:
> Hello everyone,
>
> I am trying to add a gateway, but after configuring it just like the  
> others gateways I have, it is failing to register with a message  
> like this:
>
> 2009-09-29 12:54:40.853440 [ERR] sofia_reg.c:1402 orange  
> Registration Failed with status Operation has no matching challenge   
> [904]. failure #1
> 2009-09-29 12:54:40.906798 [WARNING] sofia_reg.c:364 orange Failed  
> Registration, setting retry to 10 seconds.
>
> I captured the sip traffic and noticed that when trying to register  
> with one gateway (the one that works), I get a "Trying" reply  
> immediately followed by a "401 Unauthorized" which contains a "WWW- 
> Authenticate: digest" with a "qop=auth" parameter. Then Freeswitch  
> replies with a second REGISTER including a large "Authorization:  
> digest" section with cnonce and nc=00000001 parameters.
>
> The gateway which doesn't register, doesn't send the "qop=auth"  
> parameter together with the "401 Unauthorized", and then Freeswitch  
> sends a "Authorization: digest" section on the second REGISTER with  
> no cnonce or nc parameters.
>
> I know very little abouth SIP, so I'm wondering what this "qop=auth"  
> parameter means and how does it affect the registration process. Is  
> there any way to do without the qop=auth parameter?
>
> Also, I tried registering with X-Lite directly to the gateway, and  
> it worked, so it appears to be a problem in the Freeswitch/gateway  
> combination. (Note: X-Lite sends an "Authorization: digest" section  
> on the _first_ REGISTER, apparently this makes a difference)
>
> Attached is a sip trace for the registration traffic when doing  
> "sofia profile external restart reloadxml" on the cli, captured with  
> "tshark -i eth0 -o "rtp.heuristic_rtp: TRUE" -w /tmp/capture.pcap -b  
> filesize:51200 -b files:100 -R 'sip or rtp or icmp or dns or rtcp or  
> t38'"
>
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From mike at jerris.com  Sun Oct  4 14:41:20 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 4 Oct 2009 17:41:20 -0400
Subject: [Freeswitch-users] register problem
In-Reply-To: <20091001212602.GC17256@base.carmickle.com>
References: <20091001212602.GC17256@base.carmickle.com>
Message-ID: <37090A74-55D8-4551-94A9-F540B60AD726@jerris.com>

Your problem is that the url below returns a Not found.


On Oct 1, 2009, at 5:26 PM, Frank Carmickle wrote:

> Can someone point out what is wrong here.  Thanks.
>
> Siptrace at http://carmickle.com/fs.txt
>
> <include>
>      <gateway name="voiptalk.org">
>        <param name="username" value="81100068"/>
>        <param name="password" value="my_pass"/>
>        <param name="from-user" value="81100068"/>
>        <param name="from-domain" value="voiptalk.org"/>
>    <param name="proxy" value="voiptalk.org"/>
>        <param name="expire-seconds" value="600"/>
>        <param name="register" value="true"/>
>        <param name="retry-seconds" value="30"/>
>        <param name="extension" value="1000"/>
>        <param name="contact-params" value="domain_name=$${domain}"/>
>        <param name="context" value="public"/>
>      </gateway>
> </include>



From mike at jerris.com  Sun Oct  4 14:45:53 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 4 Oct 2009 17:45:53 -0400
Subject: [Freeswitch-users] internal & external ip addresses of
	freeswitch
In-Reply-To: <241d382f0910020110l5f3728e7k3e521e81353b5d6f@mail.gmail.com>
References: <241d382f0910020110l5f3728e7k3e521e81353b5d6f@mail.gmail.com>
Message-ID: <C7EA8BED-2517-4E1C-94F6-148FD9C790B4@jerris.com>

You will need to setup 2 sip profiles for this setup, one for each  
interface.

Mike

On Oct 2, 2009, at 4:10 AM, Timur Irmatov wrote:

> Hi.
>
> We have a local network 192.168.1.0/24, where all the users are. Out
> FreeSWITCH server is connected to this network, and has ip address
> 192.168.1.242. Through different network card it is connected to
> external gateway, and has address 172.16.12.11 in this network.
>
> I set up a test client with softphone. When incoming call is
> deliviered to this client, call is set up normally, but client can't
> hang it up. It sends BYE to external address - 172.16.12.11 - which is
> not reachable from the client. It seems this address is coming from
> Contact: field in INVITE that FreeSWITCH sends:
>
> U 192.168.1.242:5060 -> 192.168.1.34:37169
> INVITE sip:100 at 192.168.1.34:37169 SIP/2.0.
> Via: SIP/2.0/UDP 172.16.12.11;rport;branch=z9hG4bKrvp6jm3myyaaF.
> Max-Forwards: 70.
> From: "FreeSWITCH" <sip:0000000000 at 172.16.12.11>;tag=v817pS9c6v6Fe.
> To: <sip:100 at 192.168.1.34:37169>.
> Call-ID: 797bd088-29cd-122d-9b93-0060979d54c5.
> CSeq: 121117089 INVITE.
> Contact: <sip:mod_sofia at 172.16.12.11:5060>.
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14898.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 267.
> Remote-Party-ID: "FreeSWITCH"
> <sip:0000000000 at 172.16.12.11>;party=calling;screen=yes;privacy=off.
>
> What should I tweak in freeswitch to change this behaviour?



From frank at carmickle.com  Sun Oct  4 15:00:56 2009
From: frank at carmickle.com (Frank Carmickle)
Date: Sun, 4 Oct 2009 18:00:56 -0400
Subject: [Freeswitch-users] register problem
In-Reply-To: <37090A74-55D8-4551-94A9-F540B60AD726@jerris.com>
References: <20091001212602.GC17256@base.carmickle.com>
	<37090A74-55D8-4551-94A9-F540B60AD726@jerris.com>
Message-ID: <20091004220055.GK17256@base.carmickle.com>

On Sun, Oct 04, Michael Jerris wrote:
> Your problem is that the url below returns a Not found.

I sent another message yesterday with a different link with more output.

http://carmickle.com/fs-2009-10-02.txt

Thanks
--FRank


From nicolas at medularis.com  Sun Oct  4 15:19:37 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Sun, 4 Oct 2009 18:19:37 -0400
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <F86F950D-9106-49C9-81CA-38B3EB2072F4@jerris.com>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<191c3a030909291122t770af037j5b1068197b31de87@mail.gmail.com>
	<1b46b4e80909291542k71440965g75bb3bfcf38e8005@mail.gmail.com>
	<1b46b4e80910011229k31f7233du10b0fc661730d3f3@mail.gmail.com>
	<F86F950D-9106-49C9-81CA-38B3EB2072F4@jerris.com>
Message-ID: <1b46b4e80910041519h3581d3d2x7b84041e8c18588@mail.gmail.com>

Here it is:

- http://pastebin.freeswitch.org/10582

(it is the pcap file I sent on the first email of this thread, converted to
text with 'tshark -V -r')

On Sun, Oct 4, 2009 at 5:40 PM, Michael Jerris <mike at jerris.com> wrote:

> can you send a link of a text sip trace please.
>
> On Oct 1, 2009, at 3:29 PM, Nicolas Brenner wrote:
>
> Any ideas about this?
>
> The SIP provider is offering H323, but I'm not quite sure about that, is
> mod_opal working right?
>
> Thanks!
>
> Nicolas
>
> On Tue, Sep 29, 2009 at 6:42 PM, Nicolas Brenner <nicolas at medularis.com>wrote:
>
>> Anthony, thanks. Below are my config files for the two gateways from the
>> sip trace. Both files are located in conf/directory/default.
>>
>> ---------------------
>>
>> redvoiss.xml (the one that works)
>>
>> <include>
>>   <user id="gateway_redvoiss">
>>     <gateways>
>>       <gateway name="redvoiss-pp">
>>         <param name="username" value="xxxxxxx"/>
>>         <param name="password" value="xxxxxxx"/>
>>         <param name="from-domain" value="pxextmy.redvoiss.net"/>
>>         <param name="realm" value="pxextmy.redvoiss.net"/>
>>         <param name="proxy" value="pxextmy.redvoiss.net"/>
>>         <param name="from-user" value="xxxxxxx"/>
>>         <param name="caller-id-in-from" value="false"/>
>>         <param name="expire-seconds" value="600"/>
>>         <param name="register" value="true"/>
>>         <param name="retry_seconds" value="5"/>
>>         <param name="extension" value="2010"/>
>>         <param name="context" value="public"/>
>>         <param name="codec-prefs" value="G729"/>
>>         <param name="rfc2833-pt" value="101"/>
>>       </gateway>
>>     </gateways>
>>     <params>
>>       <param name="password" value="4321"/>
>>     </params>
>>   </user>
>> </include>
>>
>> ---------------------
>>
>> orange.xml (the one that doesn't work)
>>
>> <include>
>>   <user id="gateway_orange">
>>     <gateways>
>>       <gateway name="orange">
>>         <param name="username" value="xxxxxxx"/>
>>         <param name="password" value="xxxxxxx"/>
>>         <param name="from-domain" value="216.72.10.39"/>
>>         <param name="realm" value="216.72.10.39"/>
>>         <param name="proxy" value="216.72.10.39"/>
>>         <param name="from-user" value="xxxxxxx"/>
>>         <param name="caller-id-in-from" value="false"/>
>>         <param name="expire-seconds" value="600"/>
>>         <param name="register" value="true"/>
>>         <param name="retry_seconds" value="5"/>
>>         <param name="extension" value="2011"/>
>>         <param name="context" value="public"/>
>>         <param name="codec-prefs" value="G729"/>
>>         <param name="rfc2833-pt" value="101"/>
>>       </gateway>
>>     </gateways>
>>     <params>
>>       <param name="password" value="4321"/>
>>     </params>
>>   </user>
>> </include>
>>
>> ---------------------
>>
>> If I remove the register=true param for the non-working gateway, I don't
>> get the registration error on the cli, but then all call attempts get
>> rejected with a 401 Unauthorized, and I get a hangup cause of
>> NORMAL_UNSPECIFIED.
>>
>>
>> Best,
>>
>> Nicolas
>>
>>
>>
>> On Tue, Sep 29, 2009 at 2:22 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> 900 level errors are sofia internal errors so probably something is wrong
>>> with your gateway config xml.
>>> if you want to send it with any critical info replaced with XXX maybe we
>>> can see the issue for you.
>>>
>>>
>>>
>>> On Tue, Sep 29, 2009 at 1:05 PM, Nicolas Brenner <nicolas at medularis.com>wrote:
>>>
>>>> Hello everyone,
>>>>
>>>> I am trying to add a gateway, but after configuring it just like the
>>>> others gateways I have, it is failing to register with a message like this:
>>>>
>>>> 2009-09-29 12:54:40.853440 [ERR] sofia_reg.c:1402 orange Registration
>>>> Failed with status Operation has no matching challenge  [904]. failure #1
>>>> 2009-09-29 12:54:40.906798 [WARNING] sofia_reg.c:364 orange Failed
>>>> Registration, setting retry to 10 seconds.
>>>>
>>>> I captured the sip traffic and noticed that when trying to register with
>>>> one gateway (the one that works), I get a "Trying" reply immediately
>>>> followed by a "401 Unauthorized" which contains a "WWW-Authenticate: digest"
>>>> with a "qop=auth" parameter. Then Freeswitch replies with a second REGISTER
>>>> including a large "Authorization: digest" section with cnonce and
>>>> nc=00000001 parameters.
>>>>
>>>> The gateway which doesn't register, doesn't send the "qop=auth"
>>>> parameter together with the "401 Unauthorized", and then Freeswitch sends a
>>>> "Authorization: digest" section on the second REGISTER with no cnonce or nc
>>>> parameters.
>>>>
>>>> I know very little abouth SIP, so I'm wondering what this "qop=auth"
>>>> parameter means and how does it affect the registration process. Is there
>>>> any way to do without the qop=auth parameter?
>>>>
>>>> Also, I tried registering with X-Lite directly to the gateway, and it
>>>> worked, so it appears to be a problem in the Freeswitch/gateway combination.
>>>> (Note: X-Lite sends an "Authorization: digest" section on the _first_
>>>> REGISTER, apparently this makes a difference)
>>>>
>>>> Attached is a sip trace for the registration traffic when doing "sofia
>>>> profile external restart reloadxml" on the cli, captured with "tshark -i
>>>> eth0 -o "rtp.heuristic_rtp: TRUE" -w /tmp/capture.pcap -b filesize:51200 -b
>>>> files:100 -R 'sip or rtp or icmp or dns or rtcp or t38'"
>>>>
>>>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Sun Oct  4 15:26:21 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 4 Oct 2009 18:26:21 -0400
Subject: [Freeswitch-users] register problem
In-Reply-To: <20091004220055.GK17256@base.carmickle.com>
References: <20091001212602.GC17256@base.carmickle.com>
	<37090A74-55D8-4551-94A9-F540B60AD726@jerris.com>
	<20091004220055.GK17256@base.carmickle.com>
Message-ID: <1EF6E903-1014-4042-8653-AA0B0C6C6401@jerris.com>

Is there any info of what I am looking at here, I just went through  
1000's  of lines that look like repeated good registers and a working  
call.. What exactly is not working?

Mike

On Oct 4, 2009, at 6:00 PM, Frank Carmickle wrote:

> On Sun, Oct 04, Michael Jerris wrote:
>> Your problem is that the url below returns a Not found.
>
> I sent another message yesterday with a different link with more  
> output.
>
> http://carmickle.com/fs-2009-10-02.txt
>
> Thanks
> --FRank
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From mike at jerris.com  Sun Oct  4 15:30:15 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 4 Oct 2009 18:30:15 -0400
Subject: [Freeswitch-users] How to limit the number of incoming+outgoing
	calls via specific gateway?
In-Reply-To: <65d96fc80910020532i4ba33353g7ac141624a4bb89e@mail.gmail.com>
References: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>
	<e403cddca1c90c83ce4fb6f85b026e63.squirrel@correo.nodo50.org>
	<65d96fc80910020532i4ba33353g7ac141624a4bb89e@mail.gmail.com>
Message-ID: <C1AD67AA-F969-4BA1-8415-5E381BDF18FA@jerris.com>

http://wiki.freeswitch.org/wiki/Mod_limit


On Oct 2, 2009, at 8:32 AM, Tihomir Culjaga wrote:

> what if you are running some huge traffic e.g. 2000 calls with media?
>
> a typical application for that is an IVR system handling several  
> different services. I'd like to "dedicate" some capacity for inbound  
> on per service basis.
>
>
> e.g.
>
> DID 10001 limit to 500 calls
> DID 10002 limit to 400 calls
> DID 10003 limit to 100 calls
> DID 10005 limit to 1000 calls
>
>
> This will be a total of 2000 calls.
>
>
> don't you think js is simply too weak for that? It should cont calls/ 
> channels, brake counts per service/DID and update the counters on  
> every call hit.
>
>
>
>
> in the DP you would have something like this for every DID:
>
>
> <include>
>   <extension name="MY_DID_NUM">
>     <condition field="destination_number" expression="^MY_DID_NUMBER 
> $">
>     <action application="set" data="SERVICE_LIMIT=500"/>
>
>       <!--
>              count number of active channels going towards  
> MY_DID_NUMBER and store it into COUNT_MY_DID_NUMBER
>       -->
>
>       <action application="transfer" data="do_MY_SERVICE XML public"/>
>     </condition>
>   </extension>
> </include>
>
>
>
> <include>
>   <extension name="SERVICE1">
>     <condition field="destination_number" expression="^do_MY_SERVICE 
> $"/>
>     <condition field="${COUNT_MY_DID_NUMBER}"  
> expression="^SERVICE_LIMIT$">
>
>        <!-- do your service here -->
>       <action application="playback" data="I_Accept_Your_Call.wav"/>
>       <action application="hangup" data="NORMAL_CLEARING"/>
>
>        <!-- do your limitation here -->
>       <anti-action application="respond" data="403 Forbidden"/> <=  
> put your response here!
>
>     </condition>
>   </extension>
> </include>
>
>
>
>
>
> but the question is ... how powerful a JavaScript can be? Will it be  
> enough to handle that load?
>
>
>
> Tihomir.
>
>
>
>
>
> On Fri, Oct 2, 2009 at 1:11 PM, Alberto Escudero <aep.lists at it46.se>  
> wrote:
>
> You can use the api and check that the channel is occupied with "show
> channels"?
> You can write a small javascript that checks if the channel is  
> occupied by
> means of session.execute api.
>
> /aep
> --
> Stopping junk mailers is good for the environment
>
> > My SIP provider allows only one call (incoming or outgoing) via one
> > SIP account. For FreeSWITCH I have configured it as public DID
> > extension and outgoing gateway. Now I would like to transfer to
> > another gw (or generate "limit exceded") when one tries to place an
> > outgoing call while incoming call is in progress. How tho do that?
> > Limiting the number of outgoing calls is easy (mod_limit), but how  
> to
> > take into account incoming one?
> >
> > - Dmitry Bely
> >
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From mike at jerris.com  Sun Oct  4 15:31:28 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 4 Oct 2009 18:31:28 -0400
Subject: [Freeswitch-users] configure FS: config.status: error: cannot
	find input file: Makefile.in
In-Reply-To: <65d96fc80910020505t48a6237ao19884c23fedf228d@mail.gmail.com>
References: <65d96fc80910020505t48a6237ao19884c23fedf228d@mail.gmail.com>
Message-ID: <4690D578-FBC2-4230-BDF0-C43CDF6669A0@jerris.com>

I updated the tiff lib to build better inline, try make  tiff-reconf

Mike

On Oct 2, 2009, at 8:05 AM, Tihomir Culjaga wrote:

> hello,
> i just got the last trunk and tried to compile it on one of my  
> development machines... Well configure fails on tiff-3.8.2 where it  
> is unable to find Makefile.in ... Can someone advice?
>
>
>
> checking if g++ static flag -static works... yes
> checking if g++ supports -c -o file.o... yes
> checking if g++ supports -c -o file.o... (cached) yes
> checking whether the g++ linker (/usr/bin/ld) supports shared  
> libraries... yes
> checking dynamic linker characteristics... GNU/Linux ld.so
> checking how to hardcode library paths into programs... immediate
> checking for OpenGL Utility library... no
> checking for GLUT library... no
> configure: creating ./config.status
> config.status: error: cannot find input file: Makefile.in
>
>
>
> tculjaga at subZero:~/freeswitch-trunk/libs/tiff-3.8.2$ ls -l
> total 2224
> -rw-r--r--  1 tculjaga tculjaga  23741 2009-10-02 13:19 acinclude.m4
> -rw-r--r--  1 tculjaga tculjaga 316978 2009-10-02 13:28 aclocal.m4
> -rwxr-xr-x  1 tculjaga tculjaga    121 2009-10-02 13:19 autogen.sh
> -rw-r--r--  1 tculjaga tculjaga 124047 2009-10-02 13:19 ChangeLog
> drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:28 config
> -rw-r--r--  1 tculjaga tculjaga 103926 2009-10-02 14:00 config.log
> -rwxr-xr-x  1 tculjaga tculjaga  73065 2009-10-02 14:00 config.status
> -rwxr-xr-x  1 tculjaga tculjaga 740145 2009-10-02 13:28 configure
> -rw-r--r--  1 tculjaga tculjaga  20492 2009-10-02 13:19 configure.ac
> -rwxr-xr-x  1 tculjaga tculjaga     56 2009-10-02 13:19 configure.gnu
> -rwxr-xr-x  1 tculjaga tculjaga 737794 2009-10-02 13:57  
> configure.lineno
> drwxr-xr-x 16 tculjaga tculjaga   4096 2009-10-02 13:19 contrib
> -rw-r--r--  1 tculjaga tculjaga   1146 2009-10-02 13:19 COPYRIGHT
> -rw-r--r--  1 tculjaga tculjaga   1570 2009-10-02 13:19 HOWTO-RELEASE
> drwxr-xr-x  5 tculjaga tculjaga   4096 2009-10-02 13:19 html
> drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:28 libtiff
> drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 m4
> -rw-r--r--  1 tculjaga tculjaga   1908 2009-10-02 13:19 Makefile.am
> -rw-r--r--  1 tculjaga tculjaga   1724 2009-10-02 13:19 Makefile.vc
> drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 man
> -rw-r--r--  1 tculjaga tculjaga   6270 2009-10-02 13:19 nmake.opt
> drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 port
> -rw-r--r--  1 tculjaga tculjaga   2363 2009-10-02 13:19 README
> -rw-r--r--  1 tculjaga tculjaga      9 2009-10-02 13:19 RELEASE-DATE
> -rw-r--r--  1 tculjaga tculjaga   5893 2009-10-02 13:19 SConstruct
> drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 test
> -rw-r--r--  1 tculjaga tculjaga    433 2009-10-02 13:19 TODO
> drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 tools
> -rw-r--r--  1 tculjaga tculjaga      6 2009-10-02 13:19 VERSION
> tculjaga at subZero:~/freeswitch-trunk/libs/tiff-3.8.2$
> tculjaga at subZero:~/freeswitch-trunk/libs/tiff-3.8.2$
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From mike at jerris.com  Sun Oct  4 15:35:58 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 4 Oct 2009 18:35:58 -0400
Subject: [Freeswitch-users] New to freeswitch and have a few questions
In-Reply-To: <a87d43fd0910020910l5db88c5fv980c8c8fe312f380@mail.gmail.com>
References: <4AC5534C.6050202@tx.rr.com>
	<a87d43fd0910020910l5db88c5fv980c8c8fe312f380@mail.gmail.com>
Message-ID: <8F97B3DC-C0B5-403C-9117-A6B4E64A0492@jerris.com>

Getting documentation on like this on the wiki would be awesome.

Mike

On Oct 2, 2009, at 12:10 PM, Michael Gende wrote:

> Hey Orien,
>
> I'm not using exactly your set up, but am using pfsense/FreeBSD.  
> Since you're using that, I assume you're going "dual homed". I've  
> got a starter guide that might help you out. If nothing else, I'd be  
> interested in a candid assessment of its usefulness or lack thereof,  
> especially to a guy like you.
>
> I've included it here. Its all just text at the moment so be  
> advised. Also be advised that there's a lot of great information on  
> the freeswitch site and in this group. The goal of my document was  
> so that someone just starting would have to hunt a little less.
>
> Hope its good for something, let me know either way, especially if  
> you find errors.
>
> Regards,
>
> Mike G.
>
> On Thu, Oct 1, 2009 at 8:11 PM, Orien Love <orien at tx.rr.com> wrote:
> Hello Everybody,
>    I am new to freeswitch, so forgive me if I ask stupid questions. I
> am planning a test setup consisting of:
>    1 - Pfsense router with the freeswitch package installed.
>    1 - Cisco WS-C3524-PWR-XL-EN 24 FE Switch, POE for the phones.
>    1 - LINKSYS SPA3000 to connect to my existing land line and phones.
>    2 - POLYCOM SOUNDPOINT IP601 SIP IP PHONEs
>
> The first question I have, Are the IP601 phones supported? The wiki
> lists 320, 431, 501, 550, 650 but not the 601.
>
> Second, is there a place that helps a person new to the IP phone world
> learn what is needed to set up a PBX using freeswitch at a small  
> office?
>
> Finally is my test setup a good one? is there something I am missing  
> or
> that I need to get the learning process started, I have found in the
> past, with a little information and a test system, I can learn what  
> I am
> doing by breaking and fixing the test bed.
>
> Thanks for your time
> Orien
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
> <freeswitch.txt>_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From frank at carmickle.com  Sun Oct  4 15:40:44 2009
From: frank at carmickle.com (Frank Carmickle)
Date: Sun, 4 Oct 2009 18:40:44 -0400
Subject: [Freeswitch-users] register problem
In-Reply-To: <1EF6E903-1014-4042-8653-AA0B0C6C6401@jerris.com>
References: <20091001212602.GC17256@base.carmickle.com>
	<37090A74-55D8-4551-94A9-F540B60AD726@jerris.com>
	<20091004220055.GK17256@base.carmickle.com>
	<1EF6E903-1014-4042-8653-AA0B0C6C6401@jerris.com>
Message-ID: <20091004224044.GL17256@base.carmickle.com>

On Sun, Oct 04, Michael Jerris wrote:
> Is there any info of what I am looking at here, I just went through  
> 1000's  of lines that look like repeated good registers and a working  
> call.. What exactly is not working?

The register fails and then never registers again.  See line 31268.  The failure is

2009-10-03 01:56:01.942233 [ERR] sofia_reg.c:1419 voiptalk.org Registration Failed with status Operation has no matching challenge  [904]. failure #1

Thanks for looking at this.

--FC


> 
> Mike
> 
> On Oct 4, 2009, at 6:00 PM, Frank Carmickle wrote:
> 
> > On Sun, Oct 04, Michael Jerris wrote:
> >> Your problem is that the url below returns a Not found.
> >
> > I sent another message yesterday with a different link with more  
> > output.
> >
> > http://carmickle.com/fs-2009-10-02.txt
> >
> > Thanks
> > --FRank
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> > users
> > http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From mike at jerris.com  Sun Oct  4 15:48:17 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 4 Oct 2009 18:48:17 -0400
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <1b46b4e80910041519h3581d3d2x7b84041e8c18588@mail.gmail.com>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<191c3a030909291122t770af037j5b1068197b31de87@mail.gmail.com>
	<1b46b4e80909291542k71440965g75bb3bfcf38e8005@mail.gmail.com>
	<1b46b4e80910011229k31f7233du10b0fc661730d3f3@mail.gmail.com>
	<F86F950D-9106-49C9-81CA-38B3EB2072F4@jerris.com>
	<1b46b4e80910041519h3581d3d2x7b84041e8c18588@mail.gmail.com>
Message-ID: <C887D143-F7A1-44B9-B8A2-D34A4B4E582E@jerris.com>

I've never been able to read these, why exactly do I need a text  
protocol to be decoded for me?  Ends up being too much noise so I just  
don't bother.

Mike

On Oct 4, 2009, at 6:19 PM, Nicolas Brenner wrote:

> Here it is:
>
> - http://pastebin.freeswitch.org/10582
>
> (it is the pcap file I sent on the first email of this thread,  
> converted to text with 'tshark -V -r')
>
> On Sun, Oct 4, 2009 at 5:40 PM, Michael Jerris <mike at jerris.com>  
> wrote:
> can you send a link of a text sip trace please.
>
> On Oct 1, 2009, at 3:29 PM, Nicolas Brenner wrote:
>
>> Any ideas about this?
>>
>> The SIP provider is offering H323, but I'm not quite sure about  
>> that, is mod_opal working right?
>>
>> Thanks!
>>
>> Nicolas
>>
>> On Tue, Sep 29, 2009 at 6:42 PM, Nicolas Brenner <nicolas at medularis.com 
>> > wrote:
>> Anthony, thanks. Below are my config files for the two gateways  
>> from the sip trace. Both files are located in conf/directory/default.
>>
>> ---------------------
>>
>> redvoiss.xml (the one that works)
>>
>> <include>
>>   <user id="gateway_redvoiss">
>>     <gateways>
>>       <gateway name="redvoiss-pp">
>>         <param name="username" value="xxxxxxx"/>
>>         <param name="password" value="xxxxxxx"/>
>>         <param name="from-domain" value="pxextmy.redvoiss.net"/>
>>         <param name="realm" value="pxextmy.redvoiss.net"/>
>>         <param name="proxy" value="pxextmy.redvoiss.net"/>
>>         <param name="from-user" value="xxxxxxx"/>
>>         <param name="caller-id-in-from" value="false"/>
>>         <param name="expire-seconds" value="600"/>
>>         <param name="register" value="true"/>
>>         <param name="retry_seconds" value="5"/>
>>         <param name="extension" value="2010"/>
>>         <param name="context" value="public"/>
>>         <param name="codec-prefs" value="G729"/>
>>         <param name="rfc2833-pt" value="101"/>
>>       </gateway>
>>     </gateways>
>>     <params>
>>       <param name="password" value="4321"/>
>>     </params>
>>   </user>
>> </include>
>>
>> ---------------------
>>
>> orange.xml (the one that doesn't work)
>>
>> <include>
>>   <user id="gateway_orange">
>>     <gateways>
>>       <gateway name="orange">
>>         <param name="username" value="xxxxxxx"/>
>>         <param name="password" value="xxxxxxx"/>
>>         <param name="from-domain" value="216.72.10.39"/>
>>         <param name="realm" value="216.72.10.39"/>
>>         <param name="proxy" value="216.72.10.39"/>
>>         <param name="from-user" value="xxxxxxx"/>
>>         <param name="caller-id-in-from" value="false"/>
>>         <param name="expire-seconds" value="600"/>
>>         <param name="register" value="true"/>
>>         <param name="retry_seconds" value="5"/>
>>         <param name="extension" value="2011"/>
>>         <param name="context" value="public"/>
>>         <param name="codec-prefs" value="G729"/>
>>         <param name="rfc2833-pt" value="101"/>
>>       </gateway>
>>     </gateways>
>>     <params>
>>       <param name="password" value="4321"/>
>>     </params>
>>   </user>
>> </include>
>>
>> ---------------------
>>
>> If I remove the register=true param for the non-working gateway, I  
>> don't get the registration error on the cli, but then all call  
>> attempts get rejected with a 401 Unauthorized, and I get a hangup  
>> cause of NORMAL_UNSPECIFIED.
>>
>>
>> Best,
>>
>> Nicolas
>>
>>
>>
>> On Tue, Sep 29, 2009 at 2:22 PM, Anthony Minessale <anthony.minessale at gmail.com 
>> > wrote:
>> 900 level errors are sofia internal errors so probably something is  
>> wrong with your gateway config xml.
>> if you want to send it with any critical info replaced with XXX  
>> maybe we can see the issue for you.
>>
>>
>>
>> On Tue, Sep 29, 2009 at 1:05 PM, Nicolas Brenner <nicolas at medularis.com 
>> > wrote:
>> Hello everyone,
>>
>> I am trying to add a gateway, but after configuring it just like  
>> the others gateways I have, it is failing to register with a  
>> message like this:
>>
>> 2009-09-29 12:54:40.853440 [ERR] sofia_reg.c:1402 orange  
>> Registration Failed with status Operation has no matching  
>> challenge  [904]. failure #1
>> 2009-09-29 12:54:40.906798 [WARNING] sofia_reg.c:364 orange Failed  
>> Registration, setting retry to 10 seconds.
>>
>> I captured the sip traffic and noticed that when trying to register  
>> with one gateway (the one that works), I get a "Trying" reply  
>> immediately followed by a "401 Unauthorized" which contains a "WWW- 
>> Authenticate: digest" with a "qop=auth" parameter. Then Freeswitch  
>> replies with a second REGISTER including a large "Authorization:  
>> digest" section with cnonce and nc=00000001 parameters.
>>
>> The gateway which doesn't register, doesn't send the "qop=auth"  
>> parameter together with the "401 Unauthorized", and then Freeswitch  
>> sends a "Authorization: digest" section on the second REGISTER with  
>> no cnonce or nc parameters.
>>
>> I know very little abouth SIP, so I'm wondering what this  
>> "qop=auth" parameter means and how does it affect the registration  
>> process. Is there any way to do without the qop=auth parameter?
>>
>> Also, I tried registering with X-Lite directly to the gateway, and  
>> it worked, so it appears to be a problem in the Freeswitch/gateway  
>> combination. (Note: X-Lite sends an "Authorization: digest" section  
>> on the _first_ REGISTER, apparently this makes a difference)
>>
>> Attached is a sip trace for the registration traffic when doing  
>> "sofia profile external restart reloadxml" on the cli, captured  
>> with "tshark -i eth0 -o "rtp.heuristic_rtp: TRUE" -w /tmp/ 
>> capture.pcap -b filesize:51200 -b files:100 -R 'sip or rtp or icmp  
>> or dns or rtcp or t38'"
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From mike at jerris.com  Sun Oct  4 15:56:54 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 4 Oct 2009 18:56:54 -0400
Subject: [Freeswitch-users] Detecting a fax
In-Reply-To: <33c87fa30910040603p7929df26l88f51fc7d256408f@mail.gmail.com>
References: <33c87fa30910040603p7929df26l88f51fc7d256408f@mail.gmail.com>
Message-ID: <91832961-FBA6-465F-AFD8-6A1BE604F6BC@jerris.com>

check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect

Note, you can't just have tone_detect as your last iten in the  
dialplan as the call will just get hung up.

Mike

On Oct 4, 2009, at 9:03 AM, Mark Campbell-Smith wrote:

> Hi
>
> I was hoping someone could help me to setup the fax detection / tone
> detection application.
>
> I want to be able to transfer an incoming fax to a specific extension.
> In my default.xml file, I have the following (extracted):
>
>    <extension name="1000">
>      <condition field="destination_number" expression="^(10[01][0-9]) 
> $">
>        <action application="fax_detect"/>
>        <action application="tone_detect" data="fax 1100 r +5000
> transfer fax XML features" />
>
> I can't get the fax to be detected and transferred.  Is there any way
> this can be done?
>
> Thanks!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From nicolas at medularis.com  Sun Oct  4 16:09:34 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Sun, 4 Oct 2009 19:09:34 -0400
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <C887D143-F7A1-44B9-B8A2-D34A4B4E582E@jerris.com>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<191c3a030909291122t770af037j5b1068197b31de87@mail.gmail.com>
	<1b46b4e80909291542k71440965g75bb3bfcf38e8005@mail.gmail.com>
	<1b46b4e80910011229k31f7233du10b0fc661730d3f3@mail.gmail.com>
	<F86F950D-9106-49C9-81CA-38B3EB2072F4@jerris.com>
	<1b46b4e80910041519h3581d3d2x7b84041e8c18588@mail.gmail.com>
	<C887D143-F7A1-44B9-B8A2-D34A4B4E582E@jerris.com>
Message-ID: <1b46b4e80910041609x2f41ceb9u142cb484951a21ed@mail.gmail.com>

Mike, how exactly should I format the file? I got the pcap file, how do I
convert it to text so that you can easily read it?


On Sun, Oct 4, 2009 at 6:48 PM, Michael Jerris <mike at jerris.com> wrote:

> I've never been able to read these, why exactly do I need a text protocol
> to be decoded for me?  Ends up being too much noise so I just don't bother.
> Mike
>
> On Oct 4, 2009, at 6:19 PM, Nicolas Brenner wrote:
>
> Here it is:
>
> - http://pastebin.freeswitch.org/10582
>
> (it is the pcap file I sent on the first email of this thread, converted to
> text with 'tshark -V -r')
>
> On Sun, Oct 4, 2009 at 5:40 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> can you send a link of a text sip trace please.
>>
>> On Oct 1, 2009, at 3:29 PM, Nicolas Brenner wrote:
>>
>> Any ideas about this?
>>
>> The SIP provider is offering H323, but I'm not quite sure about that, is
>> mod_opal working right?
>>
>> Thanks!
>>
>> Nicolas
>>
>> On Tue, Sep 29, 2009 at 6:42 PM, Nicolas Brenner <nicolas at medularis.com>wrote:
>>
>>> Anthony, thanks. Below are my config files for the two gateways from the
>>> sip trace. Both files are located in conf/directory/default.
>>>
>>> ---------------------
>>>
>>> redvoiss.xml (the one that works)
>>>
>>> <include>
>>>   <user id="gateway_redvoiss">
>>>     <gateways>
>>>       <gateway name="redvoiss-pp">
>>>         <param name="username" value="xxxxxxx"/>
>>>         <param name="password" value="xxxxxxx"/>
>>>         <param name="from-domain" value="pxextmy.redvoiss.net"/>
>>>         <param name="realm" value="pxextmy.redvoiss.net"/>
>>>         <param name="proxy" value="pxextmy.redvoiss.net"/>
>>>         <param name="from-user" value="xxxxxxx"/>
>>>         <param name="caller-id-in-from" value="false"/>
>>>         <param name="expire-seconds" value="600"/>
>>>         <param name="register" value="true"/>
>>>         <param name="retry_seconds" value="5"/>
>>>         <param name="extension" value="2010"/>
>>>         <param name="context" value="public"/>
>>>         <param name="codec-prefs" value="G729"/>
>>>         <param name="rfc2833-pt" value="101"/>
>>>       </gateway>
>>>     </gateways>
>>>     <params>
>>>       <param name="password" value="4321"/>
>>>     </params>
>>>   </user>
>>> </include>
>>>
>>> ---------------------
>>>
>>> orange.xml (the one that doesn't work)
>>>
>>> <include>
>>>   <user id="gateway_orange">
>>>     <gateways>
>>>       <gateway name="orange">
>>>         <param name="username" value="xxxxxxx"/>
>>>         <param name="password" value="xxxxxxx"/>
>>>         <param name="from-domain" value="216.72.10.39"/>
>>>         <param name="realm" value="216.72.10.39"/>
>>>         <param name="proxy" value="216.72.10.39"/>
>>>         <param name="from-user" value="xxxxxxx"/>
>>>         <param name="caller-id-in-from" value="false"/>
>>>         <param name="expire-seconds" value="600"/>
>>>         <param name="register" value="true"/>
>>>         <param name="retry_seconds" value="5"/>
>>>         <param name="extension" value="2011"/>
>>>         <param name="context" value="public"/>
>>>         <param name="codec-prefs" value="G729"/>
>>>         <param name="rfc2833-pt" value="101"/>
>>>       </gateway>
>>>     </gateways>
>>>     <params>
>>>       <param name="password" value="4321"/>
>>>     </params>
>>>   </user>
>>> </include>
>>>
>>> ---------------------
>>>
>>> If I remove the register=true param for the non-working gateway, I don't
>>> get the registration error on the cli, but then all call attempts get
>>> rejected with a 401 Unauthorized, and I get a hangup cause of
>>> NORMAL_UNSPECIFIED.
>>>
>>>
>>> Best,
>>>
>>> Nicolas
>>>
>>>
>>>
>>> On Tue, Sep 29, 2009 at 2:22 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> 900 level errors are sofia internal errors so probably something is
>>>> wrong with your gateway config xml.
>>>> if you want to send it with any critical info replaced with XXX maybe we
>>>> can see the issue for you.
>>>>
>>>>
>>>>
>>>> On Tue, Sep 29, 2009 at 1:05 PM, Nicolas Brenner <nicolas at medularis.com
>>>> > wrote:
>>>>
>>>>> Hello everyone,
>>>>>
>>>>> I am trying to add a gateway, but after configuring it just like the
>>>>> others gateways I have, it is failing to register with a message like this:
>>>>>
>>>>> 2009-09-29 12:54:40.853440 [ERR] sofia_reg.c:1402 orange Registration
>>>>> Failed with status Operation has no matching challenge  [904]. failure #1
>>>>> 2009-09-29 12:54:40.906798 [WARNING] sofia_reg.c:364 orange Failed
>>>>> Registration, setting retry to 10 seconds.
>>>>>
>>>>> I captured the sip traffic and noticed that when trying to register
>>>>> with one gateway (the one that works), I get a "Trying" reply immediately
>>>>> followed by a "401 Unauthorized" which contains a "WWW-Authenticate: digest"
>>>>> with a "qop=auth" parameter. Then Freeswitch replies with a second REGISTER
>>>>> including a large "Authorization: digest" section with cnonce and
>>>>> nc=00000001 parameters.
>>>>>
>>>>> The gateway which doesn't register, doesn't send the "qop=auth"
>>>>> parameter together with the "401 Unauthorized", and then Freeswitch sends a
>>>>> "Authorization: digest" section on the second REGISTER with no cnonce or nc
>>>>> parameters.
>>>>>
>>>>> I know very little abouth SIP, so I'm wondering what this "qop=auth"
>>>>> parameter means and how does it affect the registration process. Is there
>>>>> any way to do without the qop=auth parameter?
>>>>>
>>>>> Also, I tried registering with X-Lite directly to the gateway, and it
>>>>> worked, so it appears to be a problem in the Freeswitch/gateway combination.
>>>>> (Note: X-Lite sends an "Authorization: digest" section on the _first_
>>>>> REGISTER, apparently this makes a difference)
>>>>>
>>>>> Attached is a sip trace for the registration traffic when doing "sofia
>>>>> profile external restart reloadxml" on the cli, captured with "tshark -i
>>>>> eth0 -o "rtp.heuristic_rtp: TRUE" -w /tmp/capture.pcap -b filesize:51200 -b
>>>>> files:100 -R 'sip or rtp or icmp or dns or rtcp or t38'"
>>>>>
>>>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Sun Oct  4 16:14:47 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 4 Oct 2009 19:14:47 -0400
Subject: [Freeswitch-users] register problem
In-Reply-To: <20091004224044.GL17256@base.carmickle.com>
References: <20091001212602.GC17256@base.carmickle.com>
	<37090A74-55D8-4551-94A9-F540B60AD726@jerris.com>
	<20091004220055.GK17256@base.carmickle.com>
	<1EF6E903-1014-4042-8653-AA0B0C6C6401@jerris.com>
	<20091004224044.GL17256@base.carmickle.com>
Message-ID: <CD304EEF-8F23-463E-ABBB-8AAE5847FA2D@jerris.com>

It appears to be in the case of the far end sending a 100 on register,  
and no 200 or any other terminal response after that.  I sent the  
realevant part of this trace off to the developer of the sip library  
for advice.  Please file a bug on jira.freeswitch.org on this, with a  
log of a coupe good registers before the failure and a few after the  
initial failure.

Mike

On Oct 4, 2009, at 6:40 PM, Frank Carmickle wrote:

> On Sun, Oct 04, Michael Jerris wrote:
>> Is there any info of what I am looking at here, I just went through
>> 1000's  of lines that look like repeated good registers and a working
>> call.. What exactly is not working?
>
> The register fails and then never registers again.  See line 31268.   
> The failure is
>
> 2009-10-03 01:56:01.942233 [ERR] sofia_reg.c:1419 voiptalk.org  
> Registration Failed with status Operation has no matching challenge   
> [904]. failure #1
>
> Thanks for looking at this.
>
> --FC
>
>
>>
>> Mike
>>
>> On Oct 4, 2009, at 6:00 PM, Frank Carmickle wrote:
>>
>>> On Sun, Oct 04, Michael Jerris wrote:
>>>> Your problem is that the url below returns a Not found.
>>>
>>> I sent another message yesterday with a different link with more
>>> output.
>>>
>>> http://carmickle.com/fs-2009-10-02.txt
>>>
>>> Thanks
>>> --FRank
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From kjv at ken-ton.com  Sun Oct  4 16:16:07 2009
From: kjv at ken-ton.com (Karl Vesterling)
Date: Sun, 4 Oct 2009 19:16:07 -0400
Subject: [Freeswitch-users] Youtube - FreeSWITCH Promo Video
Message-ID: <9A87E77C-AD85-4F1E-8ED0-F9F06AA451EE@ken-ton.com>

Folks;

Here's something that I did playing around w/ learning Apple Motion.
It's my first Apple Motion production, so don't be too hard on the  
ratings...

http://www.youtube.com/watch?v=9Katqjx5RJ4

Best Regards,
Karl J. Vesterling
kjv at ken-ton.com
202-461-3231 x0

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From diego.viola at gmail.com  Sun Oct  4 16:51:39 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 4 Oct 2009 23:51:39 +0000
Subject: [Freeswitch-users] Youtube - FreeSWITCH Promo Video
In-Reply-To: <9A87E77C-AD85-4F1E-8ED0-F9F06AA451EE@ken-ton.com>
References: <9A87E77C-AD85-4F1E-8ED0-F9F06AA451EE@ken-ton.com>
Message-ID: <86a32abc0910041651rb6a3ff9k1f6ed7d0b1a9b440@mail.gmail.com>

Very nice :)

On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling <kjv at ken-ton.com> wrote:

> Folks;
>
> Here's something that I did playing around w/ learning Apple Motion.
> It's my first Apple Motion production, so don't be too hard on the
> ratings...
>
> http://www.youtube.com/watch?v=9Katqjx5RJ4
>
> Best Regards,
> Karl J. Vesterling
> kjv at ken-ton.com
> 202-461-3231 x0
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From srinivas.ksvreddy at gmail.com  Sun Oct  4 23:24:45 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Mon, 5 Oct 2009 11:54:45 +0530
Subject: [Freeswitch-users] Fail over
Message-ID: <f8af5740910042324t76810797hb770f86ddeeba028@mail.gmail.com>

Hi,

can any know how to implement fail over with freeswitch, please help me

Regards
-- 
Srinivasula Reddy K
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From gmaruzz at celliax.org  Sun Oct  4 23:59:34 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 5 Oct 2009 08:59:34 +0200
Subject: [Freeswitch-users] Fail over
In-Reply-To: <f8af5740910042324t76810797hb770f86ddeeba028@mail.gmail.com>
References: <f8af5740910042324t76810797hb770f86ddeeba028@mail.gmail.com>
Message-ID: <7b197bef0910042359p7dc9c89agfeaac71771ea0a4@mail.gmail.com>

On Mon, Oct 5, 2009 at 8:24 AM, srinivasula reddy
<srinivas.ksvreddy at gmail.com> wrote:

> can any know how to implement fail over with freeswitch, please help me
>

This issue has been debated many many times in the mailing lists.
(hint: no live call failover, HA with OpenSERet similia as load-balancers).

Please have a look at the archives:

http://lists.freeswitch.org/pipermail/freeswitch-dev/

http://lists.freeswitch.org/pipermail/freeswitch-users/

-gm


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From gmaruzz at celliax.org  Mon Oct  5 00:04:50 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 5 Oct 2009 09:04:50 +0200
Subject: [Freeswitch-users] Youtube - FreeSWITCH Promo Video
In-Reply-To: <86a32abc0910041651rb6a3ff9k1f6ed7d0b1a9b440@mail.gmail.com>
References: <9A87E77C-AD85-4F1E-8ED0-F9F06AA451EE@ken-ton.com> 
	<86a32abc0910041651rb6a3ff9k1f6ed7d0b1a9b440@mail.gmail.com>
Message-ID: <7b197bef0910050004v71c623c8w68a013bef9fea596@mail.gmail.com>

On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola <diego.viola at gmail.com> wrote:
> Very nice :)
>
> On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling <kjv at ken-ton.com> wrote:
>>
>> Folks;
>> Here's something that I did playing around w/ learning Apple Motion.

Me too: very nice!

-gmaruzz

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From lakindia89 at gmail.com  Mon Oct  5 00:07:25 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Mon, 5 Oct 2009 12:37:25 +0530
Subject: [Freeswitch-users] oz dump Saying error
Message-ID: <7d79b3930910050007h5af4def2u622d34592a56b6ba@mail.gmail.com>

HI all,
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From diego.viola at gmail.com  Mon Oct  5 00:13:11 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Mon, 5 Oct 2009 07:13:11 +0000
Subject: [Freeswitch-users] oz dump Saying error
In-Reply-To: <7d79b3930910050007h5af4def2u622d34592a56b6ba@mail.gmail.com>
References: <7d79b3930910050007h5af4def2u622d34592a56b6ba@mail.gmail.com>
Message-ID: <86a32abc0910050013w23c6294fma3411facf5fa7c05@mail.gmail.com>

Hello?

On Mon, Oct 5, 2009 at 7:07 AM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> HI all,
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lakindia89 at gmail.com  Mon Oct  5 00:20:37 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Mon, 5 Oct 2009 12:50:37 +0530
Subject: [Freeswitch-users] oz debug says error
Message-ID: <7d79b3930910050020i1b96541dk26c8d72c220ded8d@mail.gmail.com>

Hi all,
I've compiled the freeswitch with libpri support. But when I execute

oz libpri debug 1 all, I got the following error.

API CALL [oz(libpri debug 1 all )] output:
src/ozmod/ozmod_libpri/ozmod_libpri.c: -ERR invalid span.


Here is my openzap configurations.

openzap.conf

[span zt PRI_1]
trunk_type => e1
b-channel => 1:1-15
d-channel=> 1:16
b-channel => 1:17-31


openzap.conf.xml

<configuration name="openzap.conf" description="OpenZAP Configuration">
<settings>
<param name="debug" value="9"/>
</settings>
<pri_spans>
<span id="1" name="PRI_1">
<param name="q921loglevel" value="alert"/>
<param name="q931loglevel" value="alert"/>
<param name="mode" value="user"/>
<param name="dialect" value="q931"/>
<param name="dialplan" value="XML"/>
<param name="context" value="default"/>
</span>
</pri_spans>
</configuration>

I feel something I've missed in configurations. Please tell me how to get
rid of that error.
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From lakindia89 at gmail.com  Mon Oct  5 00:22:46 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Mon, 5 Oct 2009 12:52:46 +0530
Subject: [Freeswitch-users] oz dump Saying error
In-Reply-To: <86a32abc0910050013w23c6294fma3411facf5fa7c05@mail.gmail.com>
References: <7d79b3930910050007h5af4def2u622d34592a56b6ba@mail.gmail.com>
	<86a32abc0910050013w23c6294fma3411facf5fa7c05@mail.gmail.com>
Message-ID: <7d79b3930910050022j1a9626cau93bb229436d6a92e@mail.gmail.com>

Sorry my mail client has some problem. I've send another mail with my
question. Kindly ignore this one.

On Mon, Oct 5, 2009 at 12:43 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Hello?
>
> On Mon, Oct 5, 2009 at 7:07 AM, lakshmanan ganapathy <lakindia89 at gmail.com
> > wrote:
>
>> HI all,
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gmaruzz at celliax.org  Mon Oct  5 00:27:38 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 5 Oct 2009 09:27:38 +0200
Subject: [Freeswitch-users] Fail over
In-Reply-To: <7b197bef0910042359p7dc9c89agfeaac71771ea0a4@mail.gmail.com>
References: <f8af5740910042324t76810797hb770f86ddeeba028@mail.gmail.com> 
	<7b197bef0910042359p7dc9c89agfeaac71771ea0a4@mail.gmail.com>
Message-ID: <7b197bef0910050027v3baadd2era04e469ee34ca997@mail.gmail.com>

On Mon, Oct 5, 2009 at 8:59 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
> On Mon, Oct 5, 2009 at 8:24 AM, srinivasula reddy
> <srinivas.ksvreddy at gmail.com> wrote:
>
>> can any know how to implement fail over with freeswitch, please help me
>>
>
> This issue has been debated many many times in the mailing lists.
> (hint: no live call failover, HA with OpenSERet similia as load-balancers).
>
> Please have a look at the archives:
>
> http://lists.freeswitch.org/pipermail/freeswitch-dev/
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/
>

On Mon, Oct 5, 2009 at 9:15 AM, srinivasula reddy
<srinivas.ksvreddy at gmail.com> wrote:
> Hi Giovanni Maruzzelli
>
> Thanks for your reply,
> i am new to is there any way to do live call failover.

Srinivas, are you joking ? Please take the time to read the answer,
when you ask a question.

In my previous mail, I have replied to you:

This issue has been debated many many times in the mailing lists.

(hint: no live call failover, HA with OpenSERet similia as load-balancers).

Please have a look at the archives:

http://lists.freeswitch.org/pipermail/freeswitch-dev/

http://lists.freeswitch.org/pipermail/freeswitch-users/

-gm
-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From janvb at live.com  Mon Oct  5 02:15:24 2009
From: janvb at live.com (Jan Berger)
Date: Mon, 5 Oct 2009 11:15:24 +0200
Subject: [Freeswitch-users] Fail over
In-Reply-To: <7b197bef0910042359p7dc9c89agfeaac71771ea0a4@mail.gmail.com>
References: <f8af5740910042324t76810797hb770f86ddeeba028@mail.gmail.com>
Message-ID: <BLU127-W222B2636075E6063A62964D2CF0@phx.gbl>


hi,

 

FreeSWITCH "as is" have no live fail-over support, but this will change.

 

A live fail-over and redundancy mechanism is part of what SIGTRAN provides of added values. I am working on this, but it will take time before this is on a functional stage and available.

 

Also - SIGTRAN only provide failover and redundancy on L2/L3 signalling or higher.

 

This will however not provide failover on E1/T1 hardware level. The later can be achieved by different techniques, but the easiest is if the xternal switch is configured to re-connect a lost call on a different E1/T1. Most proper switches can provide this service. 

 

Jan


 
> From: gmaruzz at celliax.org
> Date: Mon, 5 Oct 2009 08:59:34 +0200
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Fail over
> 
> On Mon, Oct 5, 2009 at 8:24 AM, srinivasula reddy
> <srinivas.ksvreddy at gmail.com> wrote:
> 
> > can any know how to implement fail over with freeswitch, please help me
> >
> 
> This issue has been debated many many times in the mailing lists.
> (hint: no live call failover, HA with OpenSERet similia as load-balancers).
> 
> Please have a look at the archives:
> 
> http://lists.freeswitch.org/pipermail/freeswitch-dev/
> 
> http://lists.freeswitch.org/pipermail/freeswitch-users/
> 
> -gm
> 
> 
> -- 
> Sincerely,
> 
> Giovanni Maruzzelli
> Cell : +39-347-2665618
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
_________________________________________________________________
Windows Live: Make it easier for your friends to see what you?re up to on Facebook.
http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_2:092009
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From mcampbellsmith at gmail.com  Mon Oct  5 03:28:37 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Mon, 5 Oct 2009 21:28:37 +1100
Subject: [Freeswitch-users] Detecting a fax
In-Reply-To: <91832961-FBA6-465F-AFD8-6A1BE604F6BC@jerris.com>
References: <33c87fa30910040603p7929df26l88f51fc7d256408f@mail.gmail.com>
	<91832961-FBA6-465F-AFD8-6A1BE604F6BC@jerris.com>
Message-ID: <33c87fa30910050328l3d119939pd638b44b219a0140@mail.gmail.com>

Thanks for the response Mike,

I read that page and this one (among others)
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_fax_detect, but
I'm still lost.  This is an extract of my dialplan

    <extension name="Local">
      <condition field="destination_number" expression="^(10[01][0-9])$">
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
        <action application="set" data="ringback=${au-ring}"/>
        <action application="fax_detect"/>
        <action application="tone_detect" data="fax 1100 r +5000
transfer fax XML features" />
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="bridge" data="user/${dialed_extension}@${domain}"/>

I would assume that on detecting a fax, the dialplan 'fax' is called
in context features.  This never happens.

When is the fax tone detected?   Is it while the call is ringing or
can it be detected after the call is answered?  My goal is to be able
to have the same extension for a voice and fax call.  i assume that
the fax 'tones' are standardised and the ones on the wiki are correct?
 Also, I guess this doesn't work with media bypass (which I don't
use).

Thanks!


On Mon, Oct 5, 2009 at 9:56 AM, Michael Jerris <mike at jerris.com> wrote:
> check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
>
> Note, you can't just have tone_detect as your last iten in the
> dialplan as the call will just get hung up.
>
> Mike
>
> On Oct 4, 2009, at 9:03 AM, Mark Campbell-Smith wrote:
>
>> Hi
>>
>> I was hoping someone could help me to setup the fax detection / tone
>> detection application.
>>
>> I want to be able to transfer an incoming fax to a specific extension.
>> In my default.xml file, I have the following (extracted):
>>
>> ? ?<extension name="1000">
>> ? ? ?<condition field="destination_number" expression="^(10[01][0-9])
>> $">
>> ? ? ? ?<action application="fax_detect"/>
>> ? ? ? ?<action application="tone_detect" data="fax 1100 r +5000
>> transfer fax XML features" />
>>
>> I can't get the fax to be detected and transferred. ?Is there any way
>> this can be done?
>>
>> Thanks!
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From Russell.Mosemann at cune.org  Mon Oct  5 03:30:26 2009
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Mon, 5 Oct 2009 05:30:26 -0500
Subject: [Freeswitch-users] oz debug says error
In-Reply-To: <7d79b3930910050020i1b96541dk26c8d72c220ded8d@mail.gmail.com>
References: <7d79b3930910050020i1b96541dk26c8d72c220ded8d@mail.gmail.com>
Message-ID: <9F7D1E871F0C4184AA8ED441CF415C8A@cune.pri>

> On Behalf Of lakshmanan ganapathy
...
> I've compiled the freeswitch with libpri support. But when I execute
> oz libpri debug 1 all, I got the following error.
> 
> API CALL [oz(libpri debug 1 all )] output:
> src/ozmod/ozmod_libpri/ozmod_libpri.c: -ERR invalid span.

If you would start freeswitch from the command line or look at freeswitch/log/freeswitch.log, you will see during startup that libpri does not find a span (note the ozmod lines). That's because the configuration below is not for libpri.

> openzap.conf.xml
...
> <pri_spans>

You need to use a libpri span configuration.

http://wiki.freeswitch.org/wiki/Openzap.conf.xml_Examples#Using_with_PRI_.28libpri_compatibility_stack.29

--
Russell Moseman



From tculjaga at gmail.com  Mon Oct  5 04:13:04 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 5 Oct 2009 13:13:04 +0200
Subject: [Freeswitch-users] configure FS: config.status: error: cannot
	find input file: Makefile.in
In-Reply-To: <4690D578-FBC2-4230-BDF0-C43CDF6669A0@jerris.com>
References: <65d96fc80910020505t48a6237ao19884c23fedf228d@mail.gmail.com>
	<4690D578-FBC2-4230-BDF0-C43CDF6669A0@jerris.com>
Message-ID: <65d96fc80910050413je176581i5fe06b9877ada592@mail.gmail.com>

it works,

thx!


T.

On Mon, Oct 5, 2009 at 12:31 AM, Michael Jerris <mike at jerris.com> wrote:

> I updated the tiff lib to build better inline, try make  tiff-reconf
> Mike
>
> On Oct 2, 2009, at 8:05 AM, Tihomir Culjaga wrote:
>
> hello,
> i just got the last trunk and tried to compile it on one of my development
> machines... Well configure fails on tiff-3.8.2 where it is unable to find
> Makefile.in ... Can someone advice?
>
>
>
> checking if g++ static flag -static works... yes
> checking if g++ supports -c -o file.o... yes
> checking if g++ supports -c -o file.o... (cached) yes
> checking whether the g++ linker (/usr/bin/ld) supports shared libraries...
> yes
> checking dynamic linker characteristics... GNU/Linux ld.so
> checking how to hardcode library paths into programs... immediate
> checking for OpenGL Utility library... no
> checking for GLUT library... no
> configure: creating ./config.status
> config.status: error: cannot find input file: Makefile.in
>
>
>
> tculjaga at subZero:~/freeswitch-trunk/libs/tiff-3.8.2$ ls -l
> total 2224
> -rw-r--r--  1 tculjaga tculjaga  23741 2009-10-02 13:19 acinclude.m4
> -rw-r--r--  1 tculjaga tculjaga 316978 2009-10-02 13:28 aclocal.m4
> -rwxr-xr-x  1 tculjaga tculjaga    121 2009-10-02 13:19 autogen.sh
> -rw-r--r--  1 tculjaga tculjaga 124047 2009-10-02 13:19 ChangeLog
> drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:28 config
> -rw-r--r--  1 tculjaga tculjaga 103926 2009-10-02 14:00 config.log
> -rwxr-xr-x  1 tculjaga tculjaga  73065 2009-10-02 14:00 config.status
> -rwxr-xr-x  1 tculjaga tculjaga 740145 2009-10-02 13:28 configure
> -rw-r--r--  1 tculjaga tculjaga  20492 2009-10-02 13:19 configure.ac
> -rwxr-xr-x  1 tculjaga tculjaga     56 2009-10-02 13:19 configure.gnu
> -rwxr-xr-x  1 tculjaga tculjaga 737794 2009-10-02 13:57 configure.lineno
> drwxr-xr-x 16 tculjaga tculjaga   4096 2009-10-02 13:19 contrib
> -rw-r--r--  1 tculjaga tculjaga   1146 2009-10-02 13:19 COPYRIGHT
> -rw-r--r--  1 tculjaga tculjaga   1570 2009-10-02 13:19 HOWTO-RELEASE
> drwxr-xr-x  5 tculjaga tculjaga   4096 2009-10-02 13:19 html
> drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:28 libtiff
> drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 m4
> -rw-r--r--  1 tculjaga tculjaga   1908 2009-10-02 13:19 Makefile.am
> -rw-r--r--  1 tculjaga tculjaga   1724 2009-10-02 13:19 Makefile.vc
> drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 man
> -rw-r--r--  1 tculjaga tculjaga   6270 2009-10-02 13:19 nmake.opt
> drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 port
> -rw-r--r--  1 tculjaga tculjaga   2363 2009-10-02 13:19 README
> -rw-r--r--  1 tculjaga tculjaga      9 2009-10-02 13:19 RELEASE-DATE
> -rw-r--r--  1 tculjaga tculjaga   5893 2009-10-02 13:19 SConstruct
> drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 test
> -rw-r--r--  1 tculjaga tculjaga    433 2009-10-02 13:19 TODO
> drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 tools
> -rw-r--r--  1 tculjaga tculjaga      6 2009-10-02 13:19 VERSION
> tculjaga at subZero:~/freeswitch-trunk/libs/tiff-3.8.2$
> tculjaga at subZero:~/freeswitch-trunk/libs/tiff-3.8.2$
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lakindia89 at gmail.com  Mon Oct  5 05:17:30 2009
From: lakindia89 at gmail.com (lakshmanan)
Date: Mon, 5 Oct 2009 05:17:30 -0700 (PDT)
Subject: [Freeswitch-users] oz debug says error
In-Reply-To: <9F7D1E871F0C4184AA8ED441CF415C8A@cune.pri>
References: <7d79b3930910050020i1b96541dk26c8d72c220ded8d@mail.gmail.com>
	<9F7D1E871F0C4184AA8ED441CF415C8A@cune.pri>
Message-ID: <25749736.post@talk.nabble.com>


Thanks for pointing that.
I also tried that.
But in that case, I'm not able to make a call through openzap.

When I say 
  originate openzap/1/A/number number

It reported the following error
2009-10-05 17:45:47.733495 [ERR] ozmod_libpri.c:88 Can't destroy call 0!
API CALL [originate(openzap/1/1/9952248266 9952248266)] output:
-ERR INVALID_IE_CONTENTS

I also gone and looked up the hangup_cause page for the reason. But I was
unable to understand that.
Can u please tell why it is reporting this error?



Russell.Mosemann wrote:
> 
>> On Behalf Of lakshmanan ganapathy
> ...
>> I've compiled the freeswitch with libpri support. But when I execute
>> oz libpri debug 1 all, I got the following error.
>> 
>> API CALL [oz(libpri debug 1 all )] output:
>> src/ozmod/ozmod_libpri/ozmod_libpri.c: -ERR invalid span.
> 
> If you would start freeswitch from the command line or look at
> freeswitch/log/freeswitch.log, you will see during startup that libpri
> does not find a span (note the ozmod lines). That's because the
> configuration below is not for libpri.
> 
>> openzap.conf.xml
> ...
>> <pri_spans>
> 
> You need to use a libpri span configuration.
> 
> http://wiki.freeswitch.org/wiki/Openzap.conf.xml_Examples#Using_with_PRI_.28libpri_compatibility_stack.29
> 
> --
> Russell Moseman
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/oz-debug-says-error-tp25746215p25749736.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From maciej.aniserowicz at gmail.com  Mon Oct  5 00:10:59 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Mon, 5 Oct 2009 09:10:59 +0200
Subject: [Freeswitch-users] Bad sound quality while eavesdropping
Message-ID: <41A44DD027064988A914974405788C2E@procent>

Hello,
When I use eavesdropping in FreeSWITCH, the sound quality is really bad. Is there any way to improve it? Is this a known problem?
Br/
Maciej Aniserowicz
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From maciej.aniserowicz at gmail.com  Mon Oct  5 00:13:43 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Mon, 5 Oct 2009 09:13:43 +0200
Subject: [Freeswitch-users] gateway FS informs it's client FS about users
	hanged up with a long delay
Message-ID: <1A16501E57CA4727A593226A8C810308@procent>

Hi,
When I use two FreeSWITCH instances ('internal' and 'external'), all users register to the 'external' instance which acts as a gateway by 'internal' instance (which in turn is controlled by my applicaiton with commands sent by socket).
When user hangs up, the 'hanged up' event is propagated to the 'internal' instance after a long time (~3 minutes) instead of being propagated immediately.
What can cause this issue?

Br/
Maciej Aniserowicz
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From maciej.aniserowicz at gmail.com  Mon Oct  5 00:16:10 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Mon, 5 Oct 2009 09:16:10 +0200
Subject: [Freeswitch-users] Recording creates a 388-byte long file and
	deletes it
Message-ID: <4ED3AB65AFE34242AACDE97127FE1248@procent>

Hi,
When I record a call in FS, it only creates a 388-byte-long wav file. The conversation is no written there, and FS deletes the file when the session finishes.
What can cause this strange behavior?

Br/
Maciej Aniserowicz
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From mike at jerris.com  Mon Oct  5 05:32:47 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 5 Oct 2009 08:32:47 -0400
Subject: [Freeswitch-users] Detecting a fax
In-Reply-To: <33c87fa30910050328l3d119939pd638b44b219a0140@mail.gmail.com>
References: <33c87fa30910040603p7929df26l88f51fc7d256408f@mail.gmail.com>
	<91832961-FBA6-465F-AFD8-6A1BE604F6BC@jerris.com>
	<33c87fa30910050328l3d119939pd638b44b219a0140@mail.gmail.com>
Message-ID: <402585BB-EBEA-42B8-9A2A-58FBA19B066F@jerris.com>

Fax tones are not played by the remote machine until after answer, the  
tone_detect application starts a media bug that listens for the tone,  
can you confirm the tone is happening at all.  Maybe the issue here is  
the timeout, try making that longer, or doing the tone_detect in  
execute_on_answer

Mike

On Oct 5, 2009, at 6:28 AM, Mark Campbell-Smith wrote:

> Thanks for the response Mike,
>
> I read that page and this one (among others)
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_fax_detect, but
> I'm still lost.  This is an extract of my dialplan
>
>    <extension name="Local">
>      <condition field="destination_number" expression="^(10[01][0-9]) 
> $">
>        <action application="set" data="dialed_extension=$1"/>
>        <action application="export" data="dialed_extension=$1"/>
>        <action application="set" data="ringback=${au-ring}"/>
>        <action application="fax_detect"/>
>        <action application="tone_detect" data="fax 1100 r +5000
> transfer fax XML features" />
>        <action application="set" data="hangup_after_bridge=true"/>
>        <action application="set" data="continue_on_fail=true"/>
>        <action application="bridge" data="user/${dialed_extension}@$ 
> {domain}"/>
>
> I would assume that on detecting a fax, the dialplan 'fax' is called
> in context features.  This never happens.
>
> When is the fax tone detected?   Is it while the call is ringing or
> can it be detected after the call is answered?  My goal is to be able
> to have the same extension for a voice and fax call.  i assume that
> the fax 'tones' are standardised and the ones on the wiki are correct?
> Also, I guess this doesn't work with media bypass (which I don't
> use).
>
> Thanks!
>
>
> On Mon, Oct 5, 2009 at 9:56 AM, Michael Jerris <mike at jerris.com>  
> wrote:
>> check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
>>
>> Note, you can't just have tone_detect as your last iten in the
>> dialplan as the call will just get hung up.
>>
>> Mike
>>
>> On Oct 4, 2009, at 9:03 AM, Mark Campbell-Smith wrote:
>>
>>> Hi
>>>
>>> I was hoping someone could help me to setup the fax detection / tone
>>> detection application.
>>>
>>> I want to be able to transfer an incoming fax to a specific  
>>> extension.
>>> In my default.xml file, I have the following (extracted):
>>>
>>>    <extension name="1000">
>>>      <condition field="destination_number" expression="^(10[01] 
>>> [0-9])
>>> $">
>>>        <action application="fax_detect"/>
>>>        <action application="tone_detect" data="fax 1100 r +5000
>>> transfer fax XML features" />
>>>
>>> I can't get the fax to be detected and transferred.  Is there any  
>>> way
>>> this can be done?
>>>
>>> Thanks!



From xengelpublicx at gmail.com  Mon Oct  5 05:43:26 2009
From: xengelpublicx at gmail.com (Vladimir Elizarov)
Date: Mon, 05 Oct 2009 16:43:26 +0400
Subject: [Freeswitch-users] stun not working in fs 1.0.4?
Message-ID: <4AC9E9EE.8090805@gmail.com>

Hello.

I'm trying to configure stun in fs 1.0.4

vars.xml
<X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>
<X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:example.com"/>
<X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:example.com"/>

external.xml
    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <param name="ext-sip-ip" value="$${external_sip_ip}"/>
    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>

In this configuration, the address in ext_rtp_ip transmitted literally
(it is: "stun:stun.exmaple.com")

If you do not specify ext_rtp_ip stun then allegedly began to work. But
as in bug SFSIP-163
(http://jira.freeswitch.org/browse/SFSIP-163?page=com.atlassian.jira.plugin.system.issuetabpanels%3Acomment-tabpanel
) Via: IP is replaced, and SDP - no.

How do I fix this? Upgrade fs to trunk?

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From Russell.Mosemann at cune.org  Mon Oct  5 05:48:58 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Mon, 5 Oct 2009 12:48:58 -0000
Subject: [Freeswitch-users] oz debug says error
In-Reply-To: <25749736.post@talk.nabble.com>
Message-ID: <20091005124858.81857415806@mail.cune.org>

lakshmanan <lakindia89 at gmail.com> said:
> Thanks for pointing that.
> I also tried that.
> But in that case, I'm not able to make a call through openzap.

What is in openzap.conf.xml? If you start fs_cli and enter "oz list",
what does it show? Copy the ozmod lines from freeswitch.log to
pastebin.freeswitch.org and post the link here so that we can see what
openzap does when freeswitch starts.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From nagalenoj at gmail.com  Mon Oct  5 06:46:34 2009
From: nagalenoj at gmail.com (Nagalenoj)
Date: Mon, 5 Oct 2009 06:46:34 -0700 (PDT)
Subject: [Freeswitch-users] Re corded file as voicemail.
Message-ID: <25751158.post@talk.nabble.com>


Is it possible to treat a recorded voice as voice mail?

Assume that, I've recorded a conversation and I want this recorded file to
be treated like voicemail. So, I could check it like voicemail!!

-- 
View this message in context: http://www.nabble.com/Recorded-file-as-voicemail.-tp25751158p25751158.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From nagalenoj at gmail.com  Mon Oct  5 06:50:39 2009
From: nagalenoj at gmail.com (Nagalenoj)
Date: Mon, 5 Oct 2009 06:50:39 -0700 (PDT)
Subject: [Freeswitch-users]  UUID of the newly originated call?
Message-ID: <25751228.post@talk.nabble.com>


Dear friends,
     I am trying with ESL outbound socket. I'm trying to make a call when I
receive ANSWER event. Now, I would want to do something like,
     * Receive the events only for this uuid - I have done by registering
all events and filtering only for this uuid($uuid).
     * If it is CHANNEL_ANSWER, originate a new call.

Now, How can I get the uuid of the new call and receive events for this new
call.? I want to receive the events for both uuids.



-- 
View this message in context: http://www.nabble.com/UUID-of-the-newly-originated-call--tp25751228p25751228.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From anthony.minessale at gmail.com  Mon Oct  5 07:31:18 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 5 Oct 2009 09:31:18 -0500
Subject: [Freeswitch-users] Bad sound quality while eavesdropping
In-Reply-To: <41A44DD027064988A914974405788C2E@procent>
References: <41A44DD027064988A914974405788C2E@procent>
Message-ID: <191c3a030910050731m2d74979ep4598e5a1945d58ae@mail.gmail.com>

That's is a somewhat vague position.

You did not mention which version of FreeSWITCH you are running, the phones
being used in your example, your configuration, the codecs in use etc.

BTW,
I think you should only ask one question at a time on this list.  The list
is run by volunteers and it's sort of rude to expect 3 or 4 threads to be
tended to concerning the same one individual.


2009/10/5 Maciej Aniserowicz <maciej.aniserowicz at gmail.com>

>  Hello,
> When I use eavesdropping in FreeSWITCH, the sound quality is really bad. Is
> there any way to improve it? Is this a known problem?
> Br/
> Maciej Aniserowicz
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From tculjaga at gmail.com  Mon Oct  5 07:32:32 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 5 Oct 2009 16:32:32 +0200
Subject: [Freeswitch-users] Detecting a fax
In-Reply-To: <402585BB-EBEA-42B8-9A2A-58FBA19B066F@jerris.com>
References: <33c87fa30910040603p7929df26l88f51fc7d256408f@mail.gmail.com>
	<91832961-FBA6-465F-AFD8-6A1BE604F6BC@jerris.com>
	<33c87fa30910050328l3d119939pd638b44b219a0140@mail.gmail.com>
	<402585BB-EBEA-42B8-9A2A-58FBA19B066F@jerris.com>
Message-ID: <65d96fc80910050732g2f414dffs45375cfb16b08c39@mail.gmail.com>

hi Mark,

This is an inbound call leg and media channel (so far)  is open in reverse
direction only (application ringback). I'm afraid you have to answer the
call to be able to "hear" the fax tone.

T.



On Mon, Oct 5, 2009 at 2:32 PM, Michael Jerris <mike at jerris.com> wrote:

> Fax tones are not played by the remote machine until after answer, the
> tone_detect application starts a media bug that listens for the tone,
> can you confirm the tone is happening at all.  Maybe the issue here is
> the timeout, try making that longer, or doing the tone_detect in
> execute_on_answer
>
> Mike
>
> On Oct 5, 2009, at 6:28 AM, Mark Campbell-Smith wrote:
>
> > Thanks for the response Mike,
> >
> > I read that page and this one (among others)
> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_fax_detect, but
> > I'm still lost.  This is an extract of my dialplan
> >
> >    <extension name="Local">
> >      <condition field="destination_number" expression="^(10[01][0-9])
> > $">
> >        <action application="set" data="dialed_extension=$1"/>
> >        <action application="export" data="dialed_extension=$1"/>
> >        <action application="set" data="ringback=${au-ring}"/>
> >        <action application="fax_detect"/>
> >        <action application="tone_detect" data="fax 1100 r +5000
> > transfer fax XML features" />
> >        <action application="set" data="hangup_after_bridge=true"/>
> >        <action application="set" data="continue_on_fail=true"/>
> >        <action application="bridge" data="user/${dialed_extension}@$
> > {domain}"/>
> >
> > I would assume that on detecting a fax, the dialplan 'fax' is called
> > in context features.  This never happens.
> >
> > When is the fax tone detected?   Is it while the call is ringing or
> > can it be detected after the call is answered?  My goal is to be able
> > to have the same extension for a voice and fax call.  i assume that
> > the fax 'tones' are standardised and the ones on the wiki are correct?
> > Also, I guess this doesn't work with media bypass (which I don't
> > use).
> >
> > Thanks!
> >
> >
> > On Mon, Oct 5, 2009 at 9:56 AM, Michael Jerris <mike at jerris.com>
> > wrote:
> >> check out
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
> >>
> >> Note, you can't just have tone_detect as your last iten in the
> >> dialplan as the call will just get hung up.
> >>
> >> Mike
> >>
> >> On Oct 4, 2009, at 9:03 AM, Mark Campbell-Smith wrote:
> >>
> >>> Hi
> >>>
> >>> I was hoping someone could help me to setup the fax detection / tone
> >>> detection application.
> >>>
> >>> I want to be able to transfer an incoming fax to a specific
> >>> extension.
> >>> In my default.xml file, I have the following (extracted):
> >>>
> >>>    <extension name="1000">
> >>>      <condition field="destination_number" expression="^(10[01]
> >>> [0-9])
> >>> $">
> >>>        <action application="fax_detect"/>
> >>>        <action application="tone_detect" data="fax 1100 r +5000
> >>> transfer fax XML features" />
> >>>
> >>> I can't get the fax to be detected and transferred.  Is there any
> >>> way
> >>> this can be done?
> >>>
> >>> Thanks!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From rupa at rupa.com  Mon Oct  5 07:45:30 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 5 Oct 2009 08:45:30 -0600
Subject: [Freeswitch-users] stun not working in fs 1.0.4?
In-Reply-To: <4AC9E9EE.8090805@gmail.com>
References: <4AC9E9EE.8090805@gmail.com>
Message-ID: <b39273530910050745r659d8cd6vf52ec1748bff25a2@mail.gmail.com>

Yes, the stun thing was fixed after 1.4 I believe.

On Mon, Oct 5, 2009 at 6:43 AM, Vladimir Elizarov
<xengelpublicx at gmail.com> wrote:
>
> How do I fix this? Upgrade fs to trunk?



-- 
-Rupa


From woodydickson at gmail.com  Mon Oct  5 07:49:39 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Mon, 5 Oct 2009 22:49:39 +0800
Subject: [Freeswitch-users] overriding conference preference
Message-ID: <a44688a0910050749s1ae1f9a4vba96477d159d377f@mail.gmail.com>

Hi,

Is is possible to override any of the setting specified in the conference
profile?

What I want to do is to have a default profile, and be able to modify
certain fields if necessary in the dialplan.


Alternatively, I would prefer to have a dynamic profile setting for the
conference to obtain those parameters from odbc.

Is it possible?

woody
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From anthony.minessale at gmail.com  Mon Oct  5 08:02:44 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 5 Oct 2009 10:02:44 -0500
Subject: [Freeswitch-users] Youtube - FreeSWITCH Promo Video
In-Reply-To: <7b197bef0910050004v71c623c8w68a013bef9fea596@mail.gmail.com>
References: <9A87E77C-AD85-4F1E-8ED0-F9F06AA451EE@ken-ton.com>
	<86a32abc0910041651rb6a3ff9k1f6ed7d0b1a9b440@mail.gmail.com>
	<7b197bef0910050004v71c623c8w68a013bef9fea596@mail.gmail.com>
Message-ID: <191c3a030910050802v59b2a9du6d2af7ade03bc509@mail.gmail.com>

neat,

Here's some suggestions for your next ones. =p

Have them standing around the hologram trying to destroy the "Death
Star(tm)" that happens to look a lot like a giant 3d unix '*' character.
Then have one rebel say, "wait!, why are we wasting our time... watch
this... and dial a number on his cellphone as the whole thing explodes in
the background.

Have Darth Forkium face Luke ThreadSpawner in a dual.  "I see you have
fashioned your own TDM card...." vroom......
"Join me and together we can make linked lists and monolithic processes",
"NEVER!..." vroom vroom Master Coda has taught you well....."You are no
match for me...JOIN THE ORANGE SIDE OF THE FORCE"





On Mon, Oct 5, 2009 at 2:04 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola <diego.viola at gmail.com> wrote:
> > Very nice :)
> >
> > On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling <kjv at ken-ton.com>
> wrote:
> >>
> >> Folks;
> >> Here's something that I did playing around w/ learning Apple Motion.
>
> Me too: very nice!
>
> -gmaruzz
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mike at jerris.com  Mon Oct  5 08:03:04 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 5 Oct 2009 11:03:04 -0400
Subject: [Freeswitch-users] Re corded file as voicemail.
In-Reply-To: <25751158.post@talk.nabble.com>
References: <25751158.post@talk.nabble.com>
Message-ID: <BC536E8C-6813-4A69-9B80-EA11E41A79FC@jerris.com>

http://wiki.freeswitch.org/wiki/Mod_voicemail#voicemail_inject

On Oct 5, 2009, at 9:46 AM, Nagalenoj wrote:

>
> Is it possible to treat a recorded voice as voice mail?
>
> Assume that, I've recorded a conversation and I want this recorded  
> file to
> be treated like voicemail. So, I could check it like voicemail!!



From brian at freeswitch.org  Mon Oct  5 08:04:23 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Oct 2009 10:04:23 -0500
Subject: [Freeswitch-users] stun not working in fs 1.0.4?
In-Reply-To: <4AC9E9EE.8090805@gmail.com>
References: <4AC9E9EE.8090805@gmail.com>
Message-ID: <7C489261-F5F3-4B57-B807-21EC4A3947A2@freeswitch.org>

Yes!

/b

On Oct 5, 2009, at 7:43 AM, Vladimir Elizarov wrote:

> How do I fix this? Upgrade fs to trunk?



From mike at jerris.com  Mon Oct  5 08:05:15 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 5 Oct 2009 11:05:15 -0400
Subject: [Freeswitch-users] overriding conference preference
In-Reply-To: <a44688a0910050749s1ae1f9a4vba96477d159d377f@mail.gmail.com>
References: <a44688a0910050749s1ae1f9a4vba96477d159d377f@mail.gmail.com>
Message-ID: <E773DB8B-AC74-49DE-9B38-7AFC441882F5@jerris.com>


On Oct 5, 2009, at 10:49 AM, Woody Dickson wrote:

> Is is possible to override any of the setting specified in the  
> conference profile?

Just the flags you can pass per user such as pin and mute

>
> What I want to do is to have a default profile, and be able to  
> modify certain fields if necessary in the dialplan.
>
>
> Alternatively, I would prefer to have a dynamic profile setting for  
> the conference to obtain those parameters from odbc.

you can do this with mod_xml_curl

Mike



From kevin at johnnyvoip.com  Mon Oct  5 08:13:20 2009
From: kevin at johnnyvoip.com (Kevin Green)
Date: Mon, 5 Oct 2009 11:13:20 -0400
Subject: [Freeswitch-users] Gateways, Limits, & Routes
Message-ID: <ab57a3e80910050813i1e517723t1107fd70d05214af@mail.gmail.com>

It seems many people are looking for ways to control gateways, resiliency of
termination, and limit on connections easily in FS. Here are some of the
thoughts I had, and I would like to hear what others think of this.

In tradition phone hardware you would define lines, put them into a group,
and then assign a route to go through that group. For resiliency you could
group multiple routes together into a route list, if the first route failed,
was all busy, or unavailable it would start to try the second route. If you
go out a secondary route you can also play a warning tone to indicate this
might be going out a more expensive connection. For example you may have an
IP link between two boxes, then fail back to TDM if the IP links go down,
the TDM would be more expensive so you would want to warn business users so
they don't spend hours on the phone.

Inbound and outbound calls could both go on the lines so when a call comes
in or goes out a route it takes up one slot.

My thought is, why don't we create something similar to this that will allow
us to handle a lot of these cases without complex dialplans. We could create
routes that are assigned to gateways and limit the number of incoming,
outgoing, and total connections that can be on that route. We could also
specify what cases we would consider as failure to move onto the next route
if we are in a route list. Route lists would then be similar to using
multiple limits, and failover, but in this case it would simply be a list.

The following is an example of how I think this could be prgrammed.

<routes>
   <route id="route1" gateway="provider1" max-inbound="5" max-outbound="5"
max-total="5"
countinue_on_fail="NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION"/>
   <route id="route2" gateway="provider2" max-inbound="0" max-outbound="5"
max-total="5" countinue_on_fail="NORMAL_TEMPORARY_FAILURE"/>
   <route id="route3" gateway="provider3" max-inbound="5" max-outbound="5"
max-total="10" countinue_on_fail="NO_ROUTE_DESTINATION"/>
</routes>
<route_lists>
   <route_list id="list1">
          <route id="route1"/>
          <route id="route2" warning_tone="expensive.wav"/>
          <route id="route3" warning_tone="really_expensive.wav"/>
   </route_list>
</route_litsts>

Then in your dialplan you would simply put:

<extension name="outbound_rout">
   <condition field="destination_number" expression="^(\d{10})$">
      <action application="bridge" data="route/route2/$1"/>
   </condition>
</extension>

OR

<extension name="outbound_route">
   <condition field="destination_number" expression="^(\d{10})$">
      <action application="bridge" data="route_list/list1/$1"/>
   </condition>
</extension>

   The module would need to track not only outbound calls, but also inbound
calls that come in through the specified gateways. This would help track
cases where you have say only two channels that can be used for both
incoming and outgoing calls.

   I'm not 100% sure if this is all feasable or if it would be that much of
an improvement compaired to what is already there so I put it out to all of
you for feedback.

Regards,
   Kevin Green

JohnnyVoIP
350 Legget Drive
Kanata, ON, Canada
K2K 2W7

Phone:   613 271 5993 Ext 1203
Fax:       613 271 9810
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From fredyg at negosat.com  Mon Oct  5 08:29:19 2009
From: fredyg at negosat.com (Fredy Gonzales)
Date: Mon, 5 Oct 2009 10:29:19 -0500
Subject: [Freeswitch-users] Youtube - FreeSWITCH Promo Video
References: <9A87E77C-AD85-4F1E-8ED0-F9F06AA451EE@ken-ton.com>
	<86a32abc0910041651rb6a3ff9k1f6ed7d0b1a9b440@mail.gmail.com>
	<7b197bef0910050004v71c623c8w68a013bef9fea596@mail.gmail.com>
Message-ID: <11A305EDADB9418888807581F6759586@gcg.com.pe>

Espectacular!!.

FG
----- Original Message ----- 
From: "Giovanni Maruzzelli" <gmaruzz at celliax.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Monday, October 05, 2009 2:04 AM
Subject: Re: [Freeswitch-users] Youtube - FreeSWITCH Promo Video


> On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola <diego.viola at gmail.com> wrote:
>> Very nice :)
>>
>> On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling <kjv at ken-ton.com> wrote:
>>>
>>> Folks;
>>> Here's something that I did playing around w/ learning Apple Motion.
>
> Me too: very nice!
>
> -gmaruzz
>
> -- 
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From msc at freeswitch.org  Mon Oct  5 08:43:57 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 5 Oct 2009 08:43:57 -0700
Subject: [Freeswitch-users] New to freeswitch and have a few questions
In-Reply-To: <8F97B3DC-C0B5-403C-9117-A6B4E64A0492@jerris.com>
References: <4AC5534C.6050202@tx.rr.com>
	<a87d43fd0910020910l5db88c5fv980c8c8fe312f380@mail.gmail.com>
	<8F97B3DC-C0B5-403C-9117-A6B4E64A0492@jerris.com>
Message-ID: <87f2f3b90910050843m2ec3cbf9rb232ad62fa313917@mail.gmail.com>

I've got a copy of Mike G's document that I'm reviewing. Like he said, it's
not 100% complete, however at first glance it looks like it would be a
perfect fit for a config example page. I'll get with Mike G shortly and
we'll have it up on the wiki in the next few days for everyone to review.

-MC

On Sun, Oct 4, 2009 at 3:35 PM, Michael Jerris <mike at jerris.com> wrote:

> Getting documentation on like this on the wiki would be awesome.
> Mike
>
> On Oct 2, 2009, at 12:10 PM, Michael Gende wrote:
>
> Hey Orien,
>
> I'm not using exactly your set up, but am using pfsense/FreeBSD. Since
> you're using that, I assume you're going "dual homed". I've got a starter
> guide that might help you out. If nothing else, I'd be interested in a
> candid assessment of its usefulness or lack thereof, especially to a guy
> like you.
>
> I've included it here. Its all just text at the moment so be advised. Also
> be advised that there's a lot of great information on the freeswitch site
> and in this group. The goal of my document was so that someone just starting
> would have to hunt a little less.
>
> Hope its good for something, let me know either way, especially if you find
> errors.
>
> Regards,
>
> Mike G.
>
> On Thu, Oct 1, 2009 at 8:11 PM, Orien Love <orien at tx.rr.com> wrote:
>
>> Hello Everybody,
>>    I am new to freeswitch, so forgive me if I ask stupid questions. I
>> am planning a test setup consisting of:
>>    1 - Pfsense router with the freeswitch package installed.
>>    1 - Cisco WS-C3524-PWR-XL-EN 24 FE Switch, POE for the phones.
>>    1 - LINKSYS SPA3000 to connect to my existing land line and phones.
>>    2 - POLYCOM SOUNDPOINT IP601 SIP IP PHONEs
>>
>> The first question I have, Are the IP601 phones supported? The wiki
>> lists 320, 431, 501, 550, 650 but not the 601.
>>
>> Second, is there a place that helps a person new to the IP phone world
>> learn what is needed to set up a PBX using freeswitch at a small office?
>>
>> Finally is my test setup a good one? is there something I am missing or
>> that I need to get the learning process started, I have found in the
>> past, with a little information and a test system, I can learn what I am
>> doing by breaking and fixing the test bed.
>>
>> Thanks for your time
>> Orien
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> <freeswitch.txt>_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Oct  5 09:05:35 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 5 Oct 2009 09:05:35 -0700
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <191DBBEF043B4EFE8F680B82C1038FC7@cune.pri>
References: <87f2f3b90910020908p1aa5f5d4o8b69894cd4a58bef@mail.gmail.com>
	<20091002172418.9E8AF2E19CE@mail.cune.org>
	<87f2f3b90910021116l209a44bcg124016e58735ca37@mail.gmail.com>
	<E2C130F6CC9B4F59B069CEF8D0B0AC42@cune.pri>
	<87f2f3b90910021218i434f67cbjc33073f8550c8d1a@mail.gmail.com>
	<8E079E8CB37E4FA49363D5315F6E878E@cune.pri>
	<87f2f3b90910021700h3431bd78q3b4f6ced255ecee5@mail.gmail.com>
	<191DBBEF043B4EFE8F680B82C1038FC7@cune.pri>
Message-ID: <87f2f3b90910050905g68ad8ec2xb7f834ba90f40d0b@mail.gmail.com>

On Fri, Oct 2, 2009 at 5:54 PM, Russell Mosemann
<Russell.Mosemann at cune.org>wrote:

> > Exactly. Turn on q931 debugging and try again:
> >
> > oz libpri debug 1 all
> > PB the results again and we'll check it out.
> > -MC
>
> Here's the next one. I'm not sure what to look for, but nothing pops out
> right away.
>
> http://pastebin.freeswitch.org/10571
>
>
Confirmed: the Hicomm isn't sending anything at all in the SETUP message
except the usual stuff: dialed number, channel number, etc. Does the Hicomm
have any config parameters, like Caller ID presentation?

-MC
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From msc at freeswitch.org  Mon Oct  5 09:20:57 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 5 Oct 2009 09:20:57 -0700
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <1b46b4e80910041609x2f41ceb9u142cb484951a21ed@mail.gmail.com>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<191c3a030909291122t770af037j5b1068197b31de87@mail.gmail.com>
	<1b46b4e80909291542k71440965g75bb3bfcf38e8005@mail.gmail.com>
	<1b46b4e80910011229k31f7233du10b0fc661730d3f3@mail.gmail.com>
	<F86F950D-9106-49C9-81CA-38B3EB2072F4@jerris.com>
	<1b46b4e80910041519h3581d3d2x7b84041e8c18588@mail.gmail.com>
	<C887D143-F7A1-44B9-B8A2-D34A4B4E582E@jerris.com>
	<1b46b4e80910041609x2f41ceb9u142cb484951a21ed@mail.gmail.com>
Message-ID: <87f2f3b90910050920w4d17033fl4c386a0018107bdf@mail.gmail.com>

On Sun, Oct 4, 2009 at 4:09 PM, Nicolas Brenner <nicolas at medularis.com>wrote:

> Mike, how exactly should I format the file? I got the pcap file, how do I
> convert it to text so that you can easily read it?
>
>
you can open it with wireshark, follow the TCP or UDP stream, then just copy
& paste the text as needed...

-MC
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From jerry.richards at teotech.com  Mon Oct  5 09:25:08 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 5 Oct 2009 09:25:08 -0700
Subject: [Freeswitch-users] FS Does Not
	RelayPresencePUBLISHToSubscribing Phones
In-Reply-To: <b39273530910021142p546680felc6c6ec19d86c922f@mail.gmail.com>
References: <B4B4D2747EEC4852B41A7D8D2DA36006@greyhawk.tonecommander.com><191c3a030910010935s69e65569v76f4d26503a94d96@mail.gmail.com><9B5214D274044021A57C6857844C10BA@greyhawk.tonecommander.com><ebbc2d550910012014ic655d84tbf000c084a98e51d@mail.gmail.com><E407DD09040D408F8A130C2BFAFFCD17@greyhawk.tonecommander.com><191c3a030910021052y2e074b4r648024507206e25@mail.gmail.com><57502048DB624686B96B8AE76F697AA7@greyhawk.tonecommander.com>
	<b39273530910021142p546680felc6c6ec19d86c922f@mail.gmail.com>
Message-ID: <6F7998CFEDEC4CDC83915045AF03000C@greyhawk.tonecommander.com>

Okay, I added the ";" at the end of the sqlite3 "select" command and it just
returned to the "sqlite>" prompt.  No error was returned.  Do you see
anything in my database (in the pastebin) that is incorrect?  By the way,
the "select" command I put in the pastebin refers to the "external" config,
but the "internal" config does the  same thing.

Best Regards,
Jerry


-----Original Message-----
From: Rupa Schomaker [mailto:rupa at rupa.com] 
Sent: Friday, October 02, 2009 11:42 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Does Not
RelayPresencePUBLISHToSubscribing Phones

You are missing the trailing ;

On Fri, Oct 2, 2009 at 1:28 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:
> I put the sqlite3 select query in the paste bin, and prior to that, I 
> entered the .dump command.? The select command came back with a "...>"
> prompt which I don't understand.? I don't know enough about sqlite3 to 
> know what that means?
>
> Best Regards,
> Jerry


--
-Rupa





From fraunhofer.lists.freeswitch-001 at traced.net  Mon Oct  5 10:39:39 2009
From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer)
Date: Mon, 5 Oct 2009 19:39:39 +0200
Subject: [Freeswitch-users] UUID of the newly originated call?
In-Reply-To: <25751228.post@talk.nabble.com>
References: <25751228.post@talk.nabble.com>
Message-ID: <a37510170910051039i1fdeb6dcm7cc31f47e88f0c62@mail.gmail.com>

Hi,

2009/10/5 Nagalenoj <nagalenoj at gmail.com>:
> ? ? * Receive the events only for this uuid - I have done by registering
> all events and filtering only for this uuid($uuid).
> ? ? * If it is CHANNEL_ANSWER, originate a new call.

it's a "filter in", not "filter out" :)

> Now, How can I get the uuid of the new call and receive events for this new
> call.? I want to receive the events for both uuids.

You can specify the UUID of an originated call by doing the following:

    * Use create_uuid to generate a UUID to use.
    * This will allow you to kill an originated call before it is
answered by using uuid_kill.
    * The UUID of the answered call leg will not be the same UUID as
the origination_uuid specified (Each call leg always gets its own
UUID)

    originate [origination_uuid=...]user/100 at domain.name.com

shamelessly ripped from
http://wiki.freeswitch.org/wiki/Mod_commands#originate

at least it worked for me.

  Beni.


From Russell.Mosemann at cune.org  Mon Oct  5 10:50:38 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Mon, 5 Oct 2009 17:50:38 -0000
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <87f2f3b90910050905g68ad8ec2xb7f834ba90f40d0b@mail.gmail.com>
Message-ID: <20091005175038.CD7D73F642F@mail.cune.org>

Michael Collins <msc at freeswitch.org> said:

> Confirmed: the Hicomm isn't sending anything at all in the SETUP message
> except the usual stuff: dialed number, channel number, etc. Does the Hicomm
> have any config parameters, like Caller ID presentation?

I believe it does, but I don't have access to the Hicom. I have to go
through the phone guy. It's kind of a delicate situation. At least I have
something to suggest for him to investigate. I appreciate the confirmation.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From xengelpublicx at gmail.com  Mon Oct  5 10:50:37 2009
From: xengelpublicx at gmail.com (Vladimir Elizarov)
Date: Mon, 05 Oct 2009 21:50:37 +0400
Subject: [Freeswitch-users] stun not working in fs 1.0.4?
In-Reply-To: <7C489261-F5F3-4B57-B807-21EC4A3947A2@freeswitch.org>
References: <4AC9E9EE.8090805@gmail.com>
	<7C489261-F5F3-4B57-B807-21EC4A3947A2@freeswitch.org>
Message-ID: <4ACA31ED.1000002@gmail.com>

Brian West ?????:
> Yes!
>   
Ok. Brian, why fs no two branches of the stable and trunk?
> /b
>
> On Oct 5, 2009, at 7:43 AM, Vladimir Elizarov wrote:
>
>   
>> How do I fix this? Upgrade fs to trunk?
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


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From jerry.richards at teotech.com  Mon Oct  5 10:58:10 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 5 Oct 2009 10:58:10 -0700
Subject: [Freeswitch-users] SLAs and BLAs
Message-ID: <5B5F50E0B1D34BC0BCBCC05E1FE05C8A@greyhawk.tonecommander.com>


I can see how BLFs and Presence are managed, however I haven't found much
documentation on SLAs and BLAs.  What is the RFC(s) that Freeswitch used to
implement SLAs and BLAs?  Do they differ from BLFs?

Best Regards,
Jerry



From brian at freeswitch.org  Mon Oct  5 10:59:33 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Oct 2009 12:59:33 -0500
Subject: [Freeswitch-users] stun not working in fs 1.0.4?
In-Reply-To: <4ACA31ED.1000002@gmail.com>
References: <4AC9E9EE.8090805@gmail.com>
	<7C489261-F5F3-4B57-B807-21EC4A3947A2@freeswitch.org>
	<4ACA31ED.1000002@gmail.com>
Message-ID: <4AA6F4E2-2138-439C-86E8-7554A5D110D4@freeswitch.org>

Because TRUNK is stable... its only fixes going in usually and if  
things do break they don't stay broken for long.

Ask anyone our trunk is more table then most commercial products.

/b

On Oct 5, 2009, at 12:50 PM, Vladimir Elizarov wrote:

> Ok. Brian, why fs no two branches of the stable and trunk?



From brian at freeswitch.org  Mon Oct  5 11:01:38 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Oct 2009 13:01:38 -0500
Subject: [Freeswitch-users] SLAs and BLAs
In-Reply-To: <5B5F50E0B1D34BC0BCBCC05E1FE05C8A@greyhawk.tonecommander.com>
References: <5B5F50E0B1D34BC0BCBCC05E1FE05C8A@greyhawk.tonecommander.com>
Message-ID: <3F7AC380-CACB-432F-8B94-80FBA18628C0@freeswitch.org>

First off what phones are you going to be using?

/b

On Oct 5, 2009, at 12:58 PM, Jerry Richards wrote:

>
> I can see how BLFs and Presence are managed, however I haven't found  
> much
> documentation on SLAs and BLAs.  What is the RFC(s) that Freeswitch  
> used to
> implement SLAs and BLAs?  Do they differ from BLFs?
>
> Best Regards,
> Jerry



From msc at freeswitch.org  Mon Oct  5 11:25:13 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 5 Oct 2009 11:25:13 -0700
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <20091005175038.CD7D73F642F@mail.cune.org>
References: <87f2f3b90910050905g68ad8ec2xb7f834ba90f40d0b@mail.gmail.com>
	<20091005175038.CD7D73F642F@mail.cune.org>
Message-ID: <87f2f3b90910051125p5932e254yc622b1d9454e0223@mail.gmail.com>

On Mon, Oct 5, 2009 at 10:50 AM, <Russell.Mosemann at cune.org> wrote:

> Michael Collins <msc at freeswitch.org> said:
>
> > Confirmed: the Hicomm isn't sending anything at all in the SETUP message
> > except the usual stuff: dialed number, channel number, etc. Does the
> Hicomm
> > have any config parameters, like Caller ID presentation?
>
> I believe it does, but I don't have access to the Hicom. I have to go
> through the phone guy. It's kind of a delicate situation. At least I have
> something to suggest for him to investigate. I appreciate the confirmation.
>
>
If you need proverbial ammo let me know. If he speaks Q931 then your
pastebin is the ultimate proof for him that the hicom is not sending any
caller ID info. In any case, I'm here if you need assistance.
-MC
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From diego.viola at gmail.com  Mon Oct  5 11:34:31 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Mon, 5 Oct 2009 18:34:31 +0000
Subject: [Freeswitch-users] Youtube - FreeSWITCH Promo Video
In-Reply-To: <191c3a030910050802v59b2a9du6d2af7ade03bc509@mail.gmail.com>
References: <9A87E77C-AD85-4F1E-8ED0-F9F06AA451EE@ken-ton.com>
	<86a32abc0910041651rb6a3ff9k1f6ed7d0b1a9b440@mail.gmail.com>
	<7b197bef0910050004v71c623c8w68a013bef9fea596@mail.gmail.com>
	<191c3a030910050802v59b2a9du6d2af7ade03bc509@mail.gmail.com>
Message-ID: <86a32abc0910051134mbb2c39fr57e29caeb27121a5@mail.gmail.com>

Nice script Anthony, that would be amazing to have on video ;)

On Mon, Oct 5, 2009 at 3:02 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> neat,
>
> Here's some suggestions for your next ones. =p
>
> Have them standing around the hologram trying to destroy the "Death
> Star(tm)" that happens to look a lot like a giant 3d unix '*' character.
> Then have one rebel say, "wait!, why are we wasting our time... watch
> this... and dial a number on his cellphone as the whole thing explodes in
> the background.
>
> Have Darth Forkium face Luke ThreadSpawner in a dual.  "I see you have
> fashioned your own TDM card...." vroom......
> "Join me and together we can make linked lists and monolithic processes",
> "NEVER!..." vroom vroom Master Coda has taught you well....."You are no
> match for me...JOIN THE ORANGE SIDE OF THE FORCE"
>
>
>
>
>
>
> On Mon, Oct 5, 2009 at 2:04 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:
>
>> On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola <diego.viola at gmail.com>
>> wrote:
>> > Very nice :)
>> >
>> > On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling <kjv at ken-ton.com>
>> wrote:
>> >>
>> >> Folks;
>> >> Here's something that I did playing around w/ learning Apple Motion.
>>
>> Me too: very nice!
>>
>> -gmaruzz
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Oct  5 12:03:27 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 5 Oct 2009 12:03:27 -0700
Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Followup,
	Agenda For This Week
Message-ID: <87f2f3b90910051203v7bdbca5amfc82d193f3a4f1a3@mail.gmail.com>

Ladies and Gentlemen,

Thank you for calling in to the weekly FreeSWITCH conference call. Last
week's agenda was rather light, so if you have things that you would like to
have discussed please be sure to add them here:

http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_09

Here are a few updates for everyone to keep in mind:
* Starting with the upcoming meeting (Oct 9) the conference will support
48kHz CELT codec.
* When you call in, please mute your phone if you are just listening or if
you will not be speaking a lot. Background noise from one user isn't usually
too bad, but when 10 or 15 people are not muted it can get a little
distracting. :)
* We are always looking for people to help out. If you are looking for ways
to help out then please by all means call the conference. If you cannot call
the conference for some reason but still want to help out, please email me
off list. We have numerous documentation, janitorial, code review, etc.
sub-projects that FreeSWITCH users can help with.
* Be sure to update to latest SVN trunk and test test test! We are prepping
for 1.0.5 and more people testing means a more stable release delivered more
quickly.

One topic that came up was the testing of Mike van Lammerman's Ultramonkey
setup. (http://bit.ly/8zZUS) Several people agreed to try it out. We would
love to see others try it out and report back their experiences.

Lastly, we'd like to thank everyone who has been helping out with testing,
documentation, and answering questions on the email list and IRC channels. A
special thanks to Diego Viola for doing lots of wiki cleanup lately.
Everyone's efforts are appreciated.

-Michael
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From Russell.Mosemann at cune.org  Mon Oct  5 12:15:52 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Mon, 5 Oct 2009 19:15:52 -0000
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <87f2f3b90910051125p5932e254yc622b1d9454e0223@mail.gmail.com>
Message-ID: <20091005191552.0D0093BA4C5@mail.cune.org>

Michael Collins <msc at freeswitch.org> said:

> If you need proverbial ammo let me know. If he speaks Q931 then your
> pastebin is the ultimate proof for him that the hicom is not sending any
> caller ID info. In any case, I'm here if you need assistance.

Heh, I have the exact opposite problem. I don't think he configured a PRI
T1 before, and the debug output would be meaningless to him. He usually
handles everyday "my phone doesn't work" kind of issues. I try to nudge
in the right direction without being pushy. :-)

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From brian at freeswitch.org  Mon Oct  5 12:30:10 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Oct 2009 14:30:10 -0500
Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Followup,
	Agenda For This Week
In-Reply-To: <87f2f3b90910051203v7bdbca5amfc82d193f3a4f1a3@mail.gmail.com>
References: <87f2f3b90910051203v7bdbca5amfc82d193f3a4f1a3@mail.gmail.com>
Message-ID: <B5ECAE34-B846-4408-8AC1-6D94ADCC672C@freeswitch.org>

It always supported 48kHz CELT but the conference itself was running  
at 32kHz so everyone 48k had to be down sampled.  Now you all get to  
be up sampled.  w00t!

/b

On Oct 5, 2009, at 2:03 PM, Michael Collins wrote:

> * Starting with the upcoming meeting (Oct 9) the conference will  
> support 48kHz CELT codec.



From dmitry.bely at gmail.com  Mon Oct  5 12:57:47 2009
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Mon, 5 Oct 2009 23:57:47 +0400
Subject: [Freeswitch-users] How to limit the number of incoming+outgoing
	calls via specific gateway?
In-Reply-To: <C1AD67AA-F969-4BA1-8415-5E381BDF18FA@jerris.com>
References: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>
	<e403cddca1c90c83ce4fb6f85b026e63.squirrel@correo.nodo50.org>
	<65d96fc80910020532i4ba33353g7ac141624a4bb89e@mail.gmail.com>
	<C1AD67AA-F969-4BA1-8415-5E381BDF18FA@jerris.com>
Message-ID: <90823c940910051257u403108afg3478c4c13ab26283@mail.gmail.com>

On Mon, Oct 5, 2009 at 2:30 AM, Michael Jerris <mike at jerris.com> wrote:
> http://wiki.freeswitch.org/wiki/Mod_limit

Of course I've read that. But it only affects the number of outgoing
calls (at least for gateways - chapter "Using mod_limit with an
outbound gateway"). But I would like to limit the number of all calls
(incoming+outgoing) via specific gateway. Any idea?

- Dmitry Bely


From sprice at gmail.com  Mon Oct  5 13:19:13 2009
From: sprice at gmail.com (SP)
Date: Mon, 5 Oct 2009 15:19:13 -0500
Subject: [Freeswitch-users] How to limit the number of incoming+outgoing
	calls via specific gateway?
In-Reply-To: <90823c940910051257u403108afg3478c4c13ab26283@mail.gmail.com>
References: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>
	<e403cddca1c90c83ce4fb6f85b026e63.squirrel@correo.nodo50.org>
	<65d96fc80910020532i4ba33353g7ac141624a4bb89e@mail.gmail.com>
	<C1AD67AA-F969-4BA1-8415-5E381BDF18FA@jerris.com>
	<90823c940910051257u403108afg3478c4c13ab26283@mail.gmail.com>
Message-ID: <7e2ac3270910051319xfdfc9d3wa583c2da7e932339@mail.gmail.com>

Direction doesn't matter, it uses realm's and a few other vars.  Use
the same vars for both directions.

On Mon, Oct 5, 2009 at 14:57, Dmitry Bely <dmitry.bely at gmail.com> wrote:
> On Mon, Oct 5, 2009 at 2:30 AM, Michael Jerris <mike at jerris.com> wrote:
>> http://wiki.freeswitch.org/wiki/Mod_limit
>
> Of course I've read that. But it only affects the number of outgoing
> calls (at least for gateways - chapter "Using mod_limit with an
> outbound gateway"). But I would like to limit the number of all calls
> (incoming+outgoing) via specific gateway. Any idea?
>
> - Dmitry Bely
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Shannon


From monemran at gmail.com  Mon Oct  5 13:32:50 2009
From: monemran at gmail.com (M.Emran)
Date: Tue, 6 Oct 2009 02:32:50 +0600
Subject: [Freeswitch-users] How to limit the number of incoming+outgoing
	calls via specific gateway?
In-Reply-To: <7e2ac3270910051319xfdfc9d3wa583c2da7e932339@mail.gmail.com>
References: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>
	<e403cddca1c90c83ce4fb6f85b026e63.squirrel@correo.nodo50.org>
	<65d96fc80910020532i4ba33353g7ac141624a4bb89e@mail.gmail.com>
	<C1AD67AA-F969-4BA1-8415-5E381BDF18FA@jerris.com>
	<90823c940910051257u403108afg3478c4c13ab26283@mail.gmail.com>
	<7e2ac3270910051319xfdfc9d3wa583c2da7e932339@mail.gmail.com>
Message-ID: <b95af1c0910051332p3113156aj89a1a7b2fc3ae22a@mail.gmail.com>

Incoming Call-Limit http://www.howtonix.com/?p=89

Outbound Call-Limit http://www.howtonix.com/?p=86



On Tue, Oct 6, 2009 at 2:19 AM, SP <sprice at gmail.com> wrote:

> Direction doesn't matter, it uses realm's and a few other vars.  Use
> the same vars for both directions.
>
> On Mon, Oct 5, 2009 at 14:57, Dmitry Bely <dmitry.bely at gmail.com> wrote:
> > On Mon, Oct 5, 2009 at 2:30 AM, Michael Jerris <mike at jerris.com> wrote:
> >> http://wiki.freeswitch.org/wiki/Mod_limit
> >
> > Of course I've read that. But it only affects the number of outgoing
> > calls (at least for gateways - chapter "Using mod_limit with an
> > outbound gateway"). But I would like to limit the number of all calls
> > (incoming+outgoing) via specific gateway. Any idea?
> >
> > - Dmitry Bely
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> Shannon
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Regards
----------
M Emran
Managing Director
E-SOFT BILLING PTE. LTD.
Web: www.e-softbilling.com
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From dmitry.bely at gmail.com  Mon Oct  5 13:39:56 2009
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Tue, 6 Oct 2009 00:39:56 +0400
Subject: [Freeswitch-users] How to limit the number of incoming+outgoing
	calls via specific gateway?
In-Reply-To: <7e2ac3270910051319xfdfc9d3wa583c2da7e932339@mail.gmail.com>
References: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>
	<e403cddca1c90c83ce4fb6f85b026e63.squirrel@correo.nodo50.org>
	<65d96fc80910020532i4ba33353g7ac141624a4bb89e@mail.gmail.com>
	<C1AD67AA-F969-4BA1-8415-5E381BDF18FA@jerris.com>
	<90823c940910051257u403108afg3478c4c13ab26283@mail.gmail.com>
	<7e2ac3270910051319xfdfc9d3wa583c2da7e932339@mail.gmail.com>
Message-ID: <90823c940910051339h3614be97o48a6319f68da7977@mail.gmail.com>

On Tue, Oct 6, 2009 at 12:19 AM, SP <sprice at gmail.com> wrote:
> Direction doesn't matter, it uses realm's and a few other vars. ?Use
> the same vars for both directions.

Unfortunately it does.

      <action application="limit" data="$${domain} my_provider 1"/>

generates limit_exceeded for the second outbound call, but if an
incoming call is active FreeSWITCH still tries to use this gateway.

> On Mon, Oct 5, 2009 at 14:57, Dmitry Bely <dmitry.bely at gmail.com> wrote:
>> On Mon, Oct 5, 2009 at 2:30 AM, Michael Jerris <mike at jerris.com> wrote:
>>> http://wiki.freeswitch.org/wiki/Mod_limit
>>
>> Of course I've read that. But it only affects the number of outgoing
>> calls (at least for gateways - chapter "Using mod_limit with an
>> outbound gateway"). But I would like to limit the number of all calls
>> (incoming+outgoing) via specific gateway. Any idea?
>>
>> - Dmitry Bely
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Shannon
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

 Dmitry Bely


From dmitry.bely at gmail.com  Mon Oct  5 13:41:49 2009
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Tue, 6 Oct 2009 00:41:49 +0400
Subject: [Freeswitch-users] How to limit the number of incoming+outgoing
	calls via specific gateway?
In-Reply-To: <b95af1c0910051332p3113156aj89a1a7b2fc3ae22a@mail.gmail.com>
References: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>
	<e403cddca1c90c83ce4fb6f85b026e63.squirrel@correo.nodo50.org>
	<65d96fc80910020532i4ba33353g7ac141624a4bb89e@mail.gmail.com>
	<C1AD67AA-F969-4BA1-8415-5E381BDF18FA@jerris.com>
	<90823c940910051257u403108afg3478c4c13ab26283@mail.gmail.com>
	<7e2ac3270910051319xfdfc9d3wa583c2da7e932339@mail.gmail.com>
	<b95af1c0910051332p3113156aj89a1a7b2fc3ae22a@mail.gmail.com>
Message-ID: <90823c940910051341w2060e9c3m59f2b59f20ac328d@mail.gmail.com>

On Tue, Oct 6, 2009 at 12:32 AM, M.Emran <monemran at gmail.com> wrote:
> Incoming Call-Limit http://www.howtonix.com/?p=89
>
> Outbound Call-Limit http://www.howtonix.com/?p=86

But what if I need to limit the total number of calls (in my case == 1)?

> On Tue, Oct 6, 2009 at 2:19 AM, SP <sprice at gmail.com> wrote:
>>
>> Direction doesn't matter, it uses realm's and a few other vars. ?Use
>> the same vars for both directions.
>>
>> On Mon, Oct 5, 2009 at 14:57, Dmitry Bely <dmitry.bely at gmail.com> wrote:
>> > On Mon, Oct 5, 2009 at 2:30 AM, Michael Jerris <mike at jerris.com> wrote:
>> >> http://wiki.freeswitch.org/wiki/Mod_limit
>> >
>> > Of course I've read that. But it only affects the number of outgoing
>> > calls (at least for gateways - chapter "Using mod_limit with an
>> > outbound gateway"). But I would like to limit the number of all calls
>> > (incoming+outgoing) via specific gateway. Any idea?

- Dmitry Bely


From sprice at gmail.com  Mon Oct  5 13:52:51 2009
From: sprice at gmail.com (SP)
Date: Mon, 5 Oct 2009 15:52:51 -0500
Subject: [Freeswitch-users] How to limit the number of incoming+outgoing
	calls via specific gateway?
In-Reply-To: <90823c940910051339h3614be97o48a6319f68da7977@mail.gmail.com>
References: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>
	<e403cddca1c90c83ce4fb6f85b026e63.squirrel@correo.nodo50.org>
	<65d96fc80910020532i4ba33353g7ac141624a4bb89e@mail.gmail.com>
	<C1AD67AA-F969-4BA1-8415-5E381BDF18FA@jerris.com>
	<90823c940910051257u403108afg3478c4c13ab26283@mail.gmail.com>
	<7e2ac3270910051319xfdfc9d3wa583c2da7e932339@mail.gmail.com>
	<90823c940910051339h3614be97o48a6319f68da7977@mail.gmail.com>
Message-ID: <7e2ac3270910051352p7d052a5aife6f07827173767@mail.gmail.com>

did you use the application limit on the inbound call?  You'll need to
in order to account for it.

On Mon, Oct 5, 2009 at 15:39, Dmitry Bely <dmitry.bely at gmail.com> wrote:
> On Tue, Oct 6, 2009 at 12:19 AM, SP <sprice at gmail.com> wrote:
>> Direction doesn't matter, it uses realm's and a few other vars. ?Use
>> the same vars for both directions.
>
> Unfortunately it does.
>
> ? ? ?<action application="limit" data="$${domain} my_provider 1"/>
>
> generates limit_exceeded for the second outbound call, but if an
> incoming call is active FreeSWITCH still tries to use this gateway.
>
>> On Mon, Oct 5, 2009 at 14:57, Dmitry Bely <dmitry.bely at gmail.com> wrote:
>>> On Mon, Oct 5, 2009 at 2:30 AM, Michael Jerris <mike at jerris.com> wrote:
>>>> http://wiki.freeswitch.org/wiki/Mod_limit
>>>
>>> Of course I've read that. But it only affects the number of outgoing
>>> calls (at least for gateways - chapter "Using mod_limit with an
>>> outbound gateway"). But I would like to limit the number of all calls
>>> (incoming+outgoing) via specific gateway. Any idea?
>>>
>>> - Dmitry Bely
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Shannon
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> ?Dmitry Bely
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Shannon


From msc at freeswitch.org  Mon Oct  5 14:01:14 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 5 Oct 2009 14:01:14 -0700
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <20091005191552.0D0093BA4C5@mail.cune.org>
References: <87f2f3b90910051125p5932e254yc622b1d9454e0223@mail.gmail.com>
	<20091005191552.0D0093BA4C5@mail.cune.org>
Message-ID: <87f2f3b90910051401y31a1faf0v91b6a1b2443f519c@mail.gmail.com>

On Mon, Oct 5, 2009 at 12:15 PM, <Russell.Mosemann at cune.org> wrote:

> Michael Collins <msc at freeswitch.org> said:
>
> > If you need proverbial ammo let me know. If he speaks Q931 then your
> > pastebin is the ultimate proof for him that the hicom is not sending any
> > caller ID info. In any case, I'm here if you need assistance.
>
> Heh, I have the exact opposite problem. I don't think he configured a PRI
> T1 before, and the debug output would be meaningless to him. He usually
> handles everyday "my phone doesn't work" kind of issues. I try to nudge
> in the right direction without being pushy. :-)
>
> Haha, good luck w/ that. :P
-MC
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From ChristianDamianidis at globalive.com  Mon Oct  5 11:25:03 2009
From: ChristianDamianidis at globalive.com (Christian Damianidis)
Date: Mon, 5 Oct 2009 14:25:03 -0400
Subject: [Freeswitch-users] mod_xml_curl http POST is inconsistent/bug
Message-ID: <66EA3166EB339A4489B06286C0876A8A0C0C1432@mailserv.Globalive.local>

Trying to achieve dynamic binding of user directory. In short: It's not
doing the authorization properly. I can use curl in the command line and
it works perfectly, specifying BASIC auth.. however with the freeswitch
module it returns HTTP 401. 

 

So I've taken a close look at the network packets being sent and there
are some issues.

 

This is between the <bindings></bindings> tags in my xml_curl.conf.xml
(1.2.3.4 represents my webserver's IP)

 

        <binding name="users">

                <param name="gateway-url"
value="http://1.2.3.4:2000/users.aspx" bindings="directory"/>

                <param name="gateway-credentials"
value="username:password"/>

        </binding>

 

When I run "curl -basic -u username http://1.2.3.4:2000/users.aspx" it
asks me for a password and returns the correct thing. I use tshark to
monitor, and it sends a GET request, with the correct authorization
credentials in the header. I receive an HTTP 200 OK packet and the xml
follows. 

 

When I startup freeswitch, I guess the xml curl module gets to run, and
it makes the request. However this time it's a POST, and oddly DOES NOT
include the Authorization: Basic <encodedstring> line in the packet. I
get back two HTTP 401 Unauthorized responses, and then freeswitch sends
out another POST, this time includes the authorization line, and I get
back an OK with the xml.

My user directory is updated and we're all good.

 

The inconsistent POST request sent by the module causes freeswitch to
hang for 1-2 minutes during start-up.

 

Has anyone else had this issue? Is this a bug or intended functionality
(ping the server before making a real request?). I'd love to sort this
out, otherwise getting an updated directory isn't real-time, thus
defeating the purpose.

Thanks,

 

Christian

 

 

 

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From vdc1048 at tx.rr.com  Mon Oct  5 13:49:58 2009
From: vdc1048 at tx.rr.com (David Clark)
Date: Mon, 05 Oct 2009 15:49:58 -0500
Subject: [Freeswitch-users] Basic compile question.
In-Reply-To: <7e2ac3270910051319xfdfc9d3wa583c2da7e932339@mail.gmail.com
 >
References: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>
	<e403cddca1c90c83ce4fb6f85b026e63.squirrel@correo.nodo50.org>
	<65d96fc80910020532i4ba33353g7ac141624a4bb89e@mail.gmail.com>
	<C1AD67AA-F969-4BA1-8415-5E381BDF18FA@jerris.com>
	<90823c940910051257u403108afg3478c4c13ab26283@mail.gmail.com>
	<7e2ac3270910051319xfdfc9d3wa583c2da7e932339@mail.gmail.com>
Message-ID: <6.2.3.4.2.20091005154158.02b632a8@pop-server.tx.rr.com>

Ok using windows xp x64 here.  I download from the trunk as expected.
I fire up VS 2005 and I open the VS 2005 solution.  Yes it does say 
unsupported.

But I get two missing header files:
freeswitch\src\mod\endpoints\mod_opal\mod_opal.h(33) : fatal error 
C1083: Cannot open include file: 'ptlib.h': No such file or directory
.\mod_voipcodecs.c(36) : fatal error C1083: Cannot open include file: 
'spandsp.h': No such file or directory

Even if the project file is wrong or out of date I should be able to 
find the include files some place in the fileset.
I can't find either file in the freeswitch directory or below it.

Any idea what is up?

Thanks,
David Clark



From vdc1048 at tx.rr.com  Mon Oct  5 14:14:08 2009
From: vdc1048 at tx.rr.com (David Clark)
Date: Mon, 05 Oct 2009 16:14:08 -0500
Subject: [Freeswitch-users] UPDATED: Basic compile question.
Message-ID: <6.2.3.4.2.20091005161228.02b6bcc8@pop-server.tx.rr.com>

Ok I found spandsp.h.  It is a case of the project file being out of 
date.  No surprise.  ptlib.h is still not found.

------------------------------------------------------------------------------------------------------------------------------------------------------------
Ok using windows xp x64 here.  I download from the trunk as expected.
I fire up VS 2005 and I open the VS 2005 solution.  Yes it does say 
unsupported.

But I get two missing header files:
freeswitch\src\mod\endpoints\mod_opal\mod_opal.h(33) : fatal error 
C1083: Cannot open include file: 'ptlib.h': No such file or directory
.\mod_voipcodecs.c(36) : fatal error C1083: Cannot open include file: 
'spandsp.h': No such file or directory


Even if the project file is wrong or out of date I should be able to 
find the include files some place in the fileset.
I can't find either file in the freeswitch directory or below it.

Any idea what is up?

Thanks,
David Clark



From brian at freeswitch.org  Mon Oct  5 14:41:39 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Oct 2009 16:41:39 -0500
Subject: [Freeswitch-users] Basic compile question.
In-Reply-To: <6.2.3.4.2.20091005154158.02b632a8@pop-server.tx.rr.com>
References: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>
	<e403cddca1c90c83ce4fb6f85b026e63.squirrel@correo.nodo50.org>
	<65d96fc80910020532i4ba33353g7ac141624a4bb89e@mail.gmail.com>
	<C1AD67AA-F969-4BA1-8415-5E381BDF18FA@jerris.com>
	<90823c940910051257u403108afg3478c4c13ab26283@mail.gmail.com>
	<7e2ac3270910051319xfdfc9d3wa583c2da7e932339@mail.gmail.com>
	<6.2.3.4.2.20091005154158.02b632a8@pop-server.tx.rr.com>
Message-ID: <4C985F33-8B02-4F8F-95B5-36BA6B40690F@freeswitch.org>

Have you updated today?

/b

On Oct 5, 2009, at 3:49 PM, David Clark wrote:

> Any idea what is up?



From brian at freeswitch.org  Mon Oct  5 14:42:36 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Oct 2009 16:42:36 -0500
Subject: [Freeswitch-users] mod_xml_curl http POST is inconsistent/bug
In-Reply-To: <66EA3166EB339A4489B06286C0876A8A0C0C1432@mailserv.Globalive.local>
References: <66EA3166EB339A4489B06286C0876A8A0C0C1432@mailserv.Globalive.local>
Message-ID: <276FEE28-70CF-422D-9F6F-7CF747FDDB22@freeswitch.org>

Are you using something other than apache?

/b

On Oct 5, 2009, at 1:25 PM, Christian Damianidis wrote:

>
> The inconsistent POST request sent by the module causes freeswitch  
> to hang for 1-2 minutes during start-up.

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From mike at jerris.com  Mon Oct  5 14:58:51 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 5 Oct 2009 17:58:51 -0400
Subject: [Freeswitch-users] UPDATED: Basic compile question.
In-Reply-To: <6.2.3.4.2.20091005161228.02b6bcc8@pop-server.tx.rr.com>
References: <6.2.3.4.2.20091005161228.02b6bcc8@pop-server.tx.rr.com>
Message-ID: <8EA2BE7E-128C-4D56-A761-C2912ED097E0@jerris.com>

voip codecs is fixed, ptlib I can't recall if we ever did full build  
integration or if you needed to manually download the libraries, can  
someone who has done mod_opal build on windows comment?

Mike


On Oct 5, 2009, at 5:14 PM, David Clark wrote:

> Ok I found spandsp.h.  It is a case of the project file being out of
> date.  No surprise.  ptlib.h is still not found.
>
> ------------------------------------------------------------------------------------------------------------------------------------------------------------
> Ok using windows xp x64 here.  I download from the trunk as expected.
> I fire up VS 2005 and I open the VS 2005 solution.  Yes it does say
> unsupported.
>
> But I get two missing header files:
> freeswitch\src\mod\endpoints\mod_opal\mod_opal.h(33) : fatal error
> C1083: Cannot open include file: 'ptlib.h': No such file or directory
> .\mod_voipcodecs.c(36) : fatal error C1083: Cannot open include file:
> 'spandsp.h': No such file or directory
>
>
> Even if the project file is wrong or out of date I should be able to
> find the include files some place in the fileset.
> I can't find either file in the freeswitch directory or below it.



From jaybinks at gmail.com  Mon Oct  5 15:16:50 2009
From: jaybinks at gmail.com (Jay Binks)
Date: Tue, 6 Oct 2009 08:16:50 +1000
Subject: [Freeswitch-users] Youtube - FreeSWITCH Promo Video
In-Reply-To: <191c3a030910050802v59b2a9du6d2af7ade03bc509@mail.gmail.com>
References: <9A87E77C-AD85-4F1E-8ED0-F9F06AA451EE@ken-ton.com>
	<86a32abc0910041651rb6a3ff9k1f6ed7d0b1a9b440@mail.gmail.com>
	<7b197bef0910050004v71c623c8w68a013bef9fea596@mail.gmail.com>
	<191c3a030910050802v59b2a9du6d2af7ade03bc509@mail.gmail.com>
Message-ID: <73572B80-F192-48CE-A6BA-F829AF8F788B@gmail.com>

Haha classic !!!

Can't wait for the next installment in the series !!

J


On 06/10/2009, at 1:02, Anthony Minessale  
<anthony.minessale at gmail.com> wrote:

> neat,
>
> Here's some suggestions for your next ones. =p
>
> Have them standing around the hologram trying to destroy the "Death  
> Star(tm)" that happens to look a lot like a giant 3d unix '*'  
> character.  Then have one rebel say, "wait!, why are we wasting our  
> time... watch this... and dial a number on his cellphone as the  
> whole thing explodes in the background.
>
> Have Darth Forkium face Luke ThreadSpawner in a dual.  "I see you  
> have fashioned your own TDM card...." vroom......
> "Join me and together we can make linked lists and monolithic  
> processes", "NEVER!..." vroom vroom Master Coda has taught you  
> well....."You are no match for me...JOIN THE ORANGE SIDE OF THE FORCE"
>
>
>
>
>
> On Mon, Oct 5, 2009 at 2:04 AM, Giovanni Maruzzelli <gmaruzz at celliax.org 
> > wrote:
> On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola <diego.viola at gmail.com>  
> wrote:
> > Very nice :)
> >
> > On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling <kjv at ken-ton.com>  
> wrote:
> >>
> >> Folks;
> >> Here's something that I did playing around w/ learning Apple  
> Motion.
>
> Me too: very nice!
>
> -gmaruzz
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From jerry.richards at teotech.com  Mon Oct  5 15:24:07 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 5 Oct 2009 15:24:07 -0700
Subject: [Freeswitch-users] SLAs and BLAs
In-Reply-To: <3F7AC380-CACB-432F-8B94-80FBA18628C0@freeswitch.org>
References: <5B5F50E0B1D34BC0BCBCC05E1FE05C8A@greyhawk.tonecommander.com>
	<3F7AC380-CACB-432F-8B94-80FBA18628C0@freeswitch.org>
Message-ID: <910A6955DAC94B23A0CB430A4A863E33@greyhawk.tonecommander.com>

We are building our own in-house developed Teo phones.  I also have
CounterPath's Bria Professional phone.  For test purposes, I have one snom
phone and a couple Polycomm phones.

Jerry
 

-----Original Message-----
From: Brian West [mailto:brian at freeswitch.org] 
Sent: Monday, October 05, 2009 11:02 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] SLAs and BLAs

First off what phones are you going to be using?

/b

On Oct 5, 2009, at 12:58 PM, Jerry Richards wrote:

>
> I can see how BLFs and Presence are managed, however I haven't found 
> much documentation on SLAs and BLAs.  What is the RFC(s) that 
> Freeswitch used to implement SLAs and BLAs?  Do they differ from BLFs?
>
> Best Regards,
> Jerry






From msc at freeswitch.org  Mon Oct  5 15:27:41 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 5 Oct 2009 15:27:41 -0700
Subject: [Freeswitch-users] Wiki Info: Multi Homed Tutorial
Message-ID: <87f2f3b90910051527x638f55b8hda75d5302267c251@mail.gmail.com>

FYI,

For those who've been following the thread about Michael Gende's tutorial I
just wanted to let you know that I his document on the wiki. It can be found
here:

http://wiki.freeswitch.org/wiki/Multi_home_tutorial

Please feel free to get in there and try it out, make editorial changes,
etc. I'm sure you won't hurt Mike's feelings by adding your thoughts. :)

If you have any questions please reply to this thread and we'll take it from
there.

Thanks,
MC
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From gmaruzz at celliax.org  Mon Oct  5 15:35:07 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 6 Oct 2009 00:35:07 +0200
Subject: [Freeswitch-users] Youtube - FreeSWITCH Promo Video
In-Reply-To: <73572B80-F192-48CE-A6BA-F829AF8F788B@gmail.com>
References: <9A87E77C-AD85-4F1E-8ED0-F9F06AA451EE@ken-ton.com> 
	<86a32abc0910041651rb6a3ff9k1f6ed7d0b1a9b440@mail.gmail.com> 
	<7b197bef0910050004v71c623c8w68a013bef9fea596@mail.gmail.com> 
	<191c3a030910050802v59b2a9du6d2af7ade03bc509@mail.gmail.com> 
	<73572B80-F192-48CE-A6BA-F829AF8F788B@gmail.com>
Message-ID: <7b197bef0910051535w4f1a888fj6bccdfbb240c7bcd@mail.gmail.com>

"The Revenge of the Sip"

On Tue, Oct 6, 2009 at 12:16 AM, Jay Binks <jaybinks at gmail.com> wrote:
> Haha classic !!!
> Can't wait for the next installment in the series !!
>
> J
>
>
> On 06/10/2009, at 1:02, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
>
> neat,
>
> Here's some suggestions for your next ones. =p
>
> Have them standing around the hologram trying to destroy the "Death
> Star(tm)" that happens to look a lot like a giant 3d unix '*' character.
> Then have one rebel say, "wait!, why are we wasting our time... watch
> this... and dial a number on his cellphone as the whole thing explodes in
> the background.
>
> Have Darth Forkium face Luke ThreadSpawner in a dual.? "I see you have
> fashioned your own TDM card...." vroom......
> "Join me and together we can make linked lists and monolithic processes",
> "NEVER!..." vroom vroom Master Coda has taught you well....."You are no
> match for me...JOIN THE ORANGE SIDE OF THE FORCE"
>
>
>
>
>
> On Mon, Oct 5, 2009 at 2:04 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>
> wrote:
>>
>> On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola <diego.viola at gmail.com> wrote:
>> > Very nice :)
>> >
>> > On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling <kjv at ken-ton.com>
>> > wrote:
>> >>
>> >> Folks;
>> >> Here's something that I did playing around w/ learning Apple Motion.
>>
>> Me too: very nice!
>>
>> -gmaruzz
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From kjv at ken-ton.com  Mon Oct  5 16:19:52 2009
From: kjv at ken-ton.com (Karl Vesterling)
Date: Mon, 5 Oct 2009 19:19:52 -0400
Subject: [Freeswitch-users] Youtube - FreeSWITCH Promo Video
In-Reply-To: <86a32abc0910051134mbb2c39fr57e29caeb27121a5@mail.gmail.com>
References: <9A87E77C-AD85-4F1E-8ED0-F9F06AA451EE@ken-ton.com>
	<86a32abc0910041651rb6a3ff9k1f6ed7d0b1a9b440@mail.gmail.com>
	<7b197bef0910050004v71c623c8w68a013bef9fea596@mail.gmail.com>
	<191c3a030910050802v59b2a9du6d2af7ade03bc509@mail.gmail.com>
	<86a32abc0910051134mbb2c39fr57e29caeb27121a5@mail.gmail.com>
Message-ID: <E9AC4ED5-CD35-4770-A8E1-8D0DC01C1D38@ken-ton.com>


Sadly the budget of time and props can't afford such extravagance...
But OMG I'm still laughing...

I was e-mailing earlier off the list, and came up with some nice  
"names" that could be put in the "credits"...

Like:

Anthony Minnessale -as- Obi-Code-Kenobi  (But I do like "Master Coda"  
from below.)

Mike Jerris -as- Luke Skypewalker

Richard Stallman -as- cpp30

Stuff like that...

I corrected to "speech" vs "speach" (my bad, sorry...)

Now, as far as the below, I imagine we could pull that off using the  
REAL footage, and dubbing in the audio if someone can do a decent  
Darth Spencer, errr Vader voice...  But something tells me that Mr.  
Lucas might get a bit peeved at such a thing...

Best Regards,
Karl J. Vesterling
kjv at ken-ton.com
202-461-3231 x0

On Oct 5, 2009, at 2:34 PM, Diego Viola wrote:

> Nice script Anthony, that would be amazing to have on video ;)
>
> On Mon, Oct 5, 2009 at 3:02 PM, Anthony Minessale <anthony.minessale at gmail.com 
> > wrote:
> neat,
>
> Here's some suggestions for your next ones. =p
>
> Have them standing around the hologram trying to destroy the "Death  
> Star(tm)" that happens to look a lot like a giant 3d unix '*'  
> character.  Then have one rebel say, "wait!, why are we wasting our  
> time... watch this... and dial a number on his cellphone as the  
> whole thing explodes in the background.
>
> Have Darth Forkium face Luke ThreadSpawner in a dual.  "I see you  
> have fashioned your own TDM card...." vroom......
> "Join me and together we can make linked lists and monolithic  
> processes", "NEVER!..." vroom vroom Master Coda has taught you  
> well....."You are no match for me...JOIN THE ORANGE SIDE OF THE FORCE"
>
>
>
>
>
>
> On Mon, Oct 5, 2009 at 2:04 AM, Giovanni Maruzzelli <gmaruzz at celliax.org 
> > wrote:
> On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola <diego.viola at gmail.com>  
> wrote:
> > Very nice :)
> >
> > On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling <kjv at ken-ton.com>  
> wrote:
> >>
> >> Folks;
> >> Here's something that I did playing around w/ learning Apple  
> Motion.
>
> Me too: very nice!
>
> -gmaruzz
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From kjv at ken-ton.com  Mon Oct  5 16:45:17 2009
From: kjv at ken-ton.com (Karl Vesterling)
Date: Mon, 5 Oct 2009 19:45:17 -0400
Subject: [Freeswitch-users] Youtube - FreeSWITCH Promo Video
In-Reply-To: <7b197bef0910051535w4f1a888fj6bccdfbb240c7bcd@mail.gmail.com>
References: <9A87E77C-AD85-4F1E-8ED0-F9F06AA451EE@ken-ton.com>
	<86a32abc0910041651rb6a3ff9k1f6ed7d0b1a9b440@mail.gmail.com>
	<7b197bef0910050004v71c623c8w68a013bef9fea596@mail.gmail.com>
	<191c3a030910050802v59b2a9du6d2af7ade03bc509@mail.gmail.com>
	<73572B80-F192-48CE-A6BA-F829AF8F788B@gmail.com>
	<7b197bef0910051535w4f1a888fj6bccdfbb240c7bcd@mail.gmail.com>
Message-ID: <7053A8C1-0EB3-447E-BFF7-C975E8087ADD@ken-ton.com>

NOW THAT might be worth doing!


Best Regards,
Karl J. Vesterling
kjv at ken-ton.com
202-461-3231 x0

On Oct 5, 2009, at 6:35 PM, Giovanni Maruzzelli wrote:

> "The Revenge of the Sip"
>
> On Tue, Oct 6, 2009 at 12:16 AM, Jay Binks <jaybinks at gmail.com> wrote:
>> Haha classic !!!
>> Can't wait for the next installment in the series !!
>>
>> J
>>
>>
>> On 06/10/2009, at 1:02, Anthony Minessale <anthony.minessale at gmail.com 
>> >
>> wrote:
>>
>> neat,
>>
>> Here's some suggestions for your next ones. =p
>>
>> Have them standing around the hologram trying to destroy the "Death
>> Star(tm)" that happens to look a lot like a giant 3d unix '*'  
>> character.
>> Then have one rebel say, "wait!, why are we wasting our time... watch
>> this... and dial a number on his cellphone as the whole thing  
>> explodes in
>> the background.
>>
>> Have Darth Forkium face Luke ThreadSpawner in a dual.  "I see you  
>> have
>> fashioned your own TDM card...." vroom......
>> "Join me and together we can make linked lists and monolithic  
>> processes",
>> "NEVER!..." vroom vroom Master Coda has taught you well....."You  
>> are no
>> match for me...JOIN THE ORANGE SIDE OF THE FORCE"
>>
>>
>>
>>
>>
>> On Mon, Oct 5, 2009 at 2:04 AM, Giovanni Maruzzelli <gmaruzz at celliax.org 
>> >
>> wrote:
>>>
>>> On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola  
>>> <diego.viola at gmail.com> wrote:
>>>> Very nice :)
>>>>
>>>> On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling <kjv at ken-ton.com>
>>>> wrote:
>>>>>
>>>>> Folks;
>>>>> Here's something that I did playing around w/ learning Apple  
>>>>> Motion.
>>>
>>> Me too: very nice!
>>>
>>> -gmaruzz
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>>> users
>>> http://www.freeswitch.org
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>>
>
>
>
> -- 
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>



From lakindia89 at gmail.com  Mon Oct  5 20:49:33 2009
From: lakindia89 at gmail.com (lakshmanan)
Date: Mon, 5 Oct 2009 20:49:33 -0700 (PDT)
Subject: [Freeswitch-users] oz debug says error
In-Reply-To: <20091005124858.81857415806@mail.cune.org>
References: <7d79b3930910050020i1b96541dk26c8d72c220ded8d@mail.gmail.com>
	<9F7D1E871F0C4184AA8ED441CF415C8A@cune.pri>
	<25749736.post@talk.nabble.com>
	<20091005124858.81857415806@mail.cune.org>
Message-ID: <25762469.post@talk.nabble.com>


Openzap.conf.xml
<configuration name="openzap.conf" description="OpenZAP Configuration">
        <settings>
                
        </settings>
        <libpri_spans>
                
                        
                        
                        
                        
                
        </libpri_spans>
</configuration>

Output of oz list in fs_cli

span: 1 (PRI_1)
type: isdn
chan_count: 47
dialplan: XML
context: default
dial_regex:
fail_dial_regex:
hold_music:
analog_options none

freeswitch.log

http://pastebin.freeswitch.org/10604


Russell.Mosemann wrote:
> 
> lakshmanan <lakindia89 at gmail.com> said:
>> Thanks for pointing that.
>> I also tried that.
>> But in that case, I'm not able to make a call through openzap.
> 
> What is in openzap.conf.xml? If you start fs_cli and enter "oz list",
> what does it show? Copy the ozmod lines from freeswitch.log to
> pastebin.freeswitch.org and post the link here so that we can see what
> openzap does when freeswitch starts.
> 
> -- 
> Russell Mosemann
> 
> 
> 
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/oz-debug-says-error-tp25746215p25762469.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From lakindia89 at gmail.com  Mon Oct  5 20:51:08 2009
From: lakindia89 at gmail.com (lakshmanan)
Date: Mon, 5 Oct 2009 20:51:08 -0700 (PDT)
Subject: [Freeswitch-users] oz debug says error
Message-ID: <25762469.post@talk.nabble.com>


Openzap.conf.xml

<configuration name="openzap.conf" description="OpenZAP Configuration">
        <settings>
                
        </settings>
        <libpri_spans>
                
                        
                        
                        
                        
                
        </libpri_spans>
</configuration>

Output of oz list in fs_cli

span: 1 (PRI_1)
type: isdn
chan_count: 47
dialplan: XML
context: default
dial_regex:
fail_dial_regex:
hold_music:
analog_options none

freeswitch.log

http://pastebin.freeswitch.org/10604


Russell.Mosemann wrote:
> 
> lakshmanan <lakindia89 at gmail.com> said:
>> Thanks for pointing that.
>> I also tried that.
>> But in that case, I'm not able to make a call through openzap.
> 
> What is in openzap.conf.xml? If you start fs_cli and enter "oz list",
> what does it show? Copy the ozmod lines from freeswitch.log to
> pastebin.freeswitch.org and post the link here so that we can see what
> openzap does when freeswitch starts.
> 
> -- 
> Russell Mosemann
> 
> 
> 
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/oz-debug-says-error-tp25746215p25762469.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From lakindia89 at gmail.com  Mon Oct  5 21:06:36 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Tue, 6 Oct 2009 09:36:36 +0530
Subject: [Freeswitch-users] oz debug says error
In-Reply-To: <20091005124858.81857415806@mail.cune.org>
References: <25749736.post@talk.nabble.com>
	<20091005124858.81857415806@mail.cune.org>
Message-ID: <7d79b3930910052106n59cf9cbema68e4d6ccc274034@mail.gmail.com>

Openzap.conf.xml
<configuration name="openzap.conf" description="OpenZAP Configuration">
        <settings>
                <param name="debug" value="7"/>
        </settings>
        <libpri_spans>
                <span name="PRI_1">
                        <param name="node" value="cpe"/>
                        <param name="switch" value="euroisdn"/>
                        <param name="dialplan" value="XML"/>
                        <param name="context" value="default"/>
                </span>
        </libpri_spans>
</configuration>

Output of oz list in fs_cli

span: 1 (PRI_1)
type: isdn
chan_count: 47
dialplan: XML
context: default
dial_regex:
fail_dial_regex:
hold_music:
analog_options none


freeswitch.log

http://pastebin.freeswitch.org/10604





On Mon, Oct 5, 2009 at 6:18 PM, <Russell.Mosemann at cune.org> wrote:

> lakshmanan <lakindia89 at gmail.com> said:
> > Thanks for pointing that.
> > I also tried that.
> > But in that case, I'm not able to make a call through openzap.
>
> What is in openzap.conf.xml? If you start fs_cli and enter "oz list",
> what does it show? Copy the ozmod lines from freeswitch.log to
> pastebin.freeswitch.org and post the link here so that we can see what
> openzap does when freeswitch starts.
>
> --
> Russell Mosemann
>
>
>
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mcampbellsmith at gmail.com  Mon Oct  5 21:15:21 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 6 Oct 2009 15:15:21 +1100
Subject: [Freeswitch-users] Detecting a fax
In-Reply-To: <65d96fc80910050732g2f414dffs45375cfb16b08c39@mail.gmail.com>
References: <33c87fa30910040603p7929df26l88f51fc7d256408f@mail.gmail.com>
	<91832961-FBA6-465F-AFD8-6A1BE604F6BC@jerris.com>
	<33c87fa30910050328l3d119939pd638b44b219a0140@mail.gmail.com>
	<402585BB-EBEA-42B8-9A2A-58FBA19B066F@jerris.com>
	<65d96fc80910050732g2f414dffs45375cfb16b08c39@mail.gmail.com>
Message-ID: <33c87fa30910052115m268b6e48q33d90946585334f9@mail.gmail.com>

Thanks for your help Mike and Tihomir.

A little more playing around and I found that having <action
application="fax_detect"/> as well as <action
application="tone_detect" data="fax 1100 r +5000 transfer fax XML
features" /> do not work together.

Simply by removing fax_detect, the fax is detected beautifully.

My problem now is trying to email the fax.  I followed the
instructions on the wiki at http://wiki.freeswitch.org/wiki/Mod_fax,
but the dialplan <action application="system" .../> is not executed
after the rxfax command.  I know the script works because if I put the
system command in another part of the dialplan and hard code the
filename to attach, then the email is sent.

    <extension name="receivefax">
      <condition field="destination_number" expression="^receivefax$">
        <action application="answer" />
        <action application="playback" data="silence_stream://2000"/>
        <action application="rxfax"
data="//usr//local//freeswitch//storage//${caller_id_number}-${uuid}.rxfax.tiff"/>
        <action application="system"
data="/usr/local/freeswitch/scripts/emailfax.sh
/usr/local/freeswitch/storage/${caller_id_number}-${uuid}.rxfax.tiff"/>
        <action application="hangup"/>
      </condition>
    </extension>

ideas?
Thanks!

On Tue, Oct 6, 2009 at 1:32 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
> hi Mark,
>
> This is an inbound call leg and media channel (so far)? is open in reverse
> direction only (application ringback). I'm afraid you have to answer the
> call to be able to "hear" the fax tone.
>
> T.
>
>
>
> On Mon, Oct 5, 2009 at 2:32 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>> Fax tones are not played by the remote machine until after answer, the
>> tone_detect application starts a media bug that listens for the tone,
>> can you confirm the tone is happening at all. ?Maybe the issue here is
>> the timeout, try making that longer, or doing the tone_detect in
>> execute_on_answer
>>
>> Mike
>>
>> On Oct 5, 2009, at 6:28 AM, Mark Campbell-Smith wrote:
>>
>> > Thanks for the response Mike,
>> >
>> > I read that page and this one (among others)
>> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_fax_detect, but
>> > I'm still lost. ?This is an extract of my dialplan
>> >
>> > ? ?<extension name="Local">
>> > ? ? ?<condition field="destination_number" expression="^(10[01][0-9])
>> > $">
>> > ? ? ? ?<action application="set" data="dialed_extension=$1"/>
>> > ? ? ? ?<action application="export" data="dialed_extension=$1"/>
>> > ? ? ? ?<action application="set" data="ringback=${au-ring}"/>
>> > ? ? ? ?<action application="fax_detect"/>
>> > ? ? ? ?<action application="tone_detect" data="fax 1100 r +5000
>> > transfer fax XML features" />
>> > ? ? ? ?<action application="set" data="hangup_after_bridge=true"/>
>> > ? ? ? ?<action application="set" data="continue_on_fail=true"/>
>> > ? ? ? ?<action application="bridge" data="user/${dialed_extension}@$
>> > {domain}"/>
>> >
>> > I would assume that on detecting a fax, the dialplan 'fax' is called
>> > in context features. ?This never happens.
>> >
>> > When is the fax tone detected? ? Is it while the call is ringing or
>> > can it be detected after the call is answered? ?My goal is to be able
>> > to have the same extension for a voice and fax call. ?i assume that
>> > the fax 'tones' are standardised and the ones on the wiki are correct?
>> > Also, I guess this doesn't work with media bypass (which I don't
>> > use).
>> >
>> > Thanks!
>> >
>> >
>> > On Mon, Oct 5, 2009 at 9:56 AM, Michael Jerris <mike at jerris.com>
>> > wrote:
>> >> check out
>> >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
>> >>
>> >> Note, you can't just have tone_detect as your last iten in the
>> >> dialplan as the call will just get hung up.
>> >>
>> >> Mike
>> >>
>> >> On Oct 4, 2009, at 9:03 AM, Mark Campbell-Smith wrote:
>> >>
>> >>> Hi
>> >>>
>> >>> I was hoping someone could help me to setup the fax detection / tone
>> >>> detection application.
>> >>>
>> >>> I want to be able to transfer an incoming fax to a specific
>> >>> extension.
>> >>> In my default.xml file, I have the following (extracted):
>> >>>
>> >>> ? ?<extension name="1000">
>> >>> ? ? ?<condition field="destination_number" expression="^(10[01]
>> >>> [0-9])
>> >>> $">
>> >>> ? ? ? ?<action application="fax_detect"/>
>> >>> ? ? ? ?<action application="tone_detect" data="fax 1100 r +5000
>> >>> transfer fax XML features" />
>> >>>
>> >>> I can't get the fax to be detected and transferred. ?Is there any
>> >>> way
>> >>> this can be done?
>> >>>
>> >>> Thanks!
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From nicolas at medularis.com  Mon Oct  5 21:19:18 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Tue, 6 Oct 2009 00:19:18 -0400
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <87f2f3b90910050920w4d17033fl4c386a0018107bdf@mail.gmail.com>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<191c3a030909291122t770af037j5b1068197b31de87@mail.gmail.com>
	<1b46b4e80909291542k71440965g75bb3bfcf38e8005@mail.gmail.com>
	<1b46b4e80910011229k31f7233du10b0fc661730d3f3@mail.gmail.com>
	<F86F950D-9106-49C9-81CA-38B3EB2072F4@jerris.com>
	<1b46b4e80910041519h3581d3d2x7b84041e8c18588@mail.gmail.com>
	<C887D143-F7A1-44B9-B8A2-D34A4B4E582E@jerris.com>
	<1b46b4e80910041609x2f41ceb9u142cb484951a21ed@mail.gmail.com>
	<87f2f3b90910050920w4d17033fl4c386a0018107bdf@mail.gmail.com>
Message-ID: <1b46b4e80910052119me12a0d9y4653edd831f9555e@mail.gmail.com>

There was no sane way of doing that, so I ended up logging the trace from
the cli.

Here's the bad registration:

- http://pastebin.freeswitch.org/10605

Here's the good one:

- http://pastebin.freeswitch.org/10606

I am not sure if the second one is complete because for some reason the
first few packages don't appear on the console when doing 'sofia profile
external restart reloadxml' and 'sofia profile external siptrace on' or
viceversa.

Anyway, thanks for your time, and I hope those traces help in figuring out
what's going on.


Nicolas


PS: Is there anyway to get the same format from a pcap dump as with the
siptrace feature on the cli?


On Mon, Oct 5, 2009 at 12:20 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Sun, Oct 4, 2009 at 4:09 PM, Nicolas Brenner <nicolas at medularis.com>wrote:
>
>> Mike, how exactly should I format the file? I got the pcap file, how do I
>> convert it to text so that you can easily read it?
>>
>>
> you can open it with wireshark, follow the TCP or UDP stream, then just
> copy & paste the text as needed...
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mgende at gendesign.com  Mon Oct  5 21:20:55 2009
From: mgende at gendesign.com (Michael Gende)
Date: Mon, 5 Oct 2009 23:20:55 -0500
Subject: [Freeswitch-users] Wiki Info: Multi Homed Tutorial
In-Reply-To: <87f2f3b90910051527x638f55b8hda75d5302267c251@mail.gmail.com>
References: <87f2f3b90910051527x638f55b8hda75d5302267c251@mail.gmail.com>
Message-ID: <a87d43fd0910052120l76da1536ra92766174dc938d3@mail.gmail.com>

Michael,

Thanks for "wiki-fying" my text-only attempt at some user doc. I should have
done that for you. I actually have an updated version with many corrections
and the end tabs filled in. Can you point me to info on how I can amend and
append what you have kindly put up?

Mike G.

On Mon, Oct 5, 2009 at 5:27 PM, Michael Collins <msc at freeswitch.org> wrote:

> FYI,
>
> For those who've been following the thread about Michael Gende's tutorial I
> just wanted to let you know that I his document on the wiki. It can be found
> here:
>
> http://wiki.freeswitch.org/wiki/Multi_home_tutorial
>
> Please feel free to get in there and try it out, make editorial changes,
> etc. I'm sure you won't hurt Mike's feelings by adding your thoughts. :)
>
> If you have any questions please reply to this thread and we'll take it
> from there.
>
> Thanks,
> MC
>
>
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From vdc1048 at tx.rr.com  Mon Oct  5 21:30:02 2009
From: vdc1048 at tx.rr.com (David Clark)
Date: Mon, 05 Oct 2009 23:30:02 -0500
Subject: [Freeswitch-users] Basic compile question.
In-Reply-To: <4C985F33-8B02-4F8F-95B5-36BA6B40690F@freeswitch.org>
References: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>
	<e403cddca1c90c83ce4fb6f85b026e63.squirrel@correo.nodo50.org>
	<65d96fc80910020532i4ba33353g7ac141624a4bb89e@mail.gmail.com>
	<C1AD67AA-F969-4BA1-8415-5E381BDF18FA@jerris.com>
	<90823c940910051257u403108afg3478c4c13ab26283@mail.gmail.com>
	<7e2ac3270910051319xfdfc9d3wa583c2da7e932339@mail.gmail.com>
	<6.2.3.4.2.20091005154158.02b632a8@pop-server.tx.rr.com>
	<4C985F33-8B02-4F8F-95B5-36BA6B40690F@freeswitch.org>
Message-ID: <6.2.3.4.2.20091005232815.03231998@pop-server.tx.rr.com>

No I found the one header.  I added it to the include list for the 
project.  It included something else, added that.  etc.  Basically I 
think I am going to need the VC 2008
compiler  and to use the other project file.

At 04:41 PM 10/5/2009, Brian West wrote:
>Have you updated today?
>
>/b
>
>On Oct 5, 2009, at 3:49 PM, David Clark wrote:
>
> > Any idea what is up?
>
>
>_______________________________________________
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org



From nicolas at medularis.com  Mon Oct  5 21:42:32 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Tue, 6 Oct 2009 00:42:32 -0400
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <1b46b4e80910052119me12a0d9y4653edd831f9555e@mail.gmail.com>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<191c3a030909291122t770af037j5b1068197b31de87@mail.gmail.com>
	<1b46b4e80909291542k71440965g75bb3bfcf38e8005@mail.gmail.com>
	<1b46b4e80910011229k31f7233du10b0fc661730d3f3@mail.gmail.com>
	<F86F950D-9106-49C9-81CA-38B3EB2072F4@jerris.com>
	<1b46b4e80910041519h3581d3d2x7b84041e8c18588@mail.gmail.com>
	<C887D143-F7A1-44B9-B8A2-D34A4B4E582E@jerris.com>
	<1b46b4e80910041609x2f41ceb9u142cb484951a21ed@mail.gmail.com>
	<87f2f3b90910050920w4d17033fl4c386a0018107bdf@mail.gmail.com>
	<1b46b4e80910052119me12a0d9y4653edd831f9555e@mail.gmail.com>
Message-ID: <1b46b4e80910052142jaa09f08l72e5e4b2b24d10fc@mail.gmail.com>

Ignore my previous email, the traces were incomplete, got much better (and
complete) traces with ngrep (found a suggestion from Brian in the list
archive, thanks!)

The gateway that registers:

- http://pastebin.freeswitch.org/10607

The one that doesn't:

- http://pastebin.freeswitch.org/10608


Thanks again for your time and help!


Nicolas


On Tue, Oct 6, 2009 at 12:19 AM, Nicolas Brenner <nicolas at medularis.com>wrote:

> There was no sane way of doing that, so I ended up logging the trace from
> the cli.
>
> Here's the bad registration:
>
> - http://pastebin.freeswitch.org/10605
>
> Here's the good one:
>
> - http://pastebin.freeswitch.org/10606
>
> I am not sure if the second one is complete because for some reason the
> first few packages don't appear on the console when doing 'sofia profile
> external restart reloadxml' and 'sofia profile external siptrace on' or
> viceversa.
>
> Anyway, thanks for your time, and I hope those traces help in figuring out
> what's going on.
>
>
> Nicolas
>
>
> PS: Is there anyway to get the same format from a pcap dump as with the
> siptrace feature on the cli?
>
>
> On Mon, Oct 5, 2009 at 12:20 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Sun, Oct 4, 2009 at 4:09 PM, Nicolas Brenner <nicolas at medularis.com>wrote:
>>
>>> Mike, how exactly should I format the file? I got the pcap file, how do I
>>> convert it to text so that you can easily read it?
>>>
>>>
>> you can open it with wireshark, follow the TCP or UDP stream, then just
>> copy & paste the text as needed...
>>
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From mcampbellsmith at gmail.com  Mon Oct  5 21:43:41 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 6 Oct 2009 15:43:41 +1100
Subject: [Freeswitch-users] Detecting a fax
In-Reply-To: <33c87fa30910040603p7929df26l88f51fc7d256408f@mail.gmail.com>
References: <33c87fa30910040603p7929df26l88f51fc7d256408f@mail.gmail.com>
Message-ID: <33c87fa30910052143w9c9913chff5eacc2d7a2303e@mail.gmail.com>

Further playing around and everything is working fine (even the
emailing).  I'm not sure what I changed though to document it.

cheers
/M



On Mon, Oct 5, 2009 at 12:03 AM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> Hi
>
> I was hoping someone could help me to setup the fax detection / tone
> detection application.
>
> I want to be able to transfer an incoming fax to a specific extension.
> ?In my default.xml file, I have the following (extracted):
>
> ? ?<extension name="1000">
> ? ? ?<condition field="destination_number" expression="^(10[01][0-9])$">
> ? ? ? ?<action application="fax_detect"/>
> ? ? ? ?<action application="tone_detect" data="fax 1100 r +5000
> transfer fax XML features" />
>
> I can't get the fax to be detected and transferred. ?Is there any way
> this can be done?
>
> Thanks!
>


From sprice at gmail.com  Mon Oct  5 21:45:39 2009
From: sprice at gmail.com (SP)
Date: Mon, 5 Oct 2009 23:45:39 -0500
Subject: [Freeswitch-users] Detecting a fax
In-Reply-To: <33c87fa30910052115m268b6e48q33d90946585334f9@mail.gmail.com>
References: <33c87fa30910040603p7929df26l88f51fc7d256408f@mail.gmail.com>
	<91832961-FBA6-465F-AFD8-6A1BE604F6BC@jerris.com>
	<33c87fa30910050328l3d119939pd638b44b219a0140@mail.gmail.com>
	<402585BB-EBEA-42B8-9A2A-58FBA19B066F@jerris.com>
	<65d96fc80910050732g2f414dffs45375cfb16b08c39@mail.gmail.com>
	<33c87fa30910052115m268b6e48q33d90946585334f9@mail.gmail.com>
Message-ID: <7e2ac3270910052145s8c7ac0cqa9c1bd6c4549089b@mail.gmail.com>

try using the hanup hook

<action application="set" data="api_hangup_hook=system ${syscmd}"/>

On Mon, Oct 5, 2009 at 23:15, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> Thanks for your help Mike and Tihomir.
>
> A little more playing around and I found that having <action
> application="fax_detect"/> as well as <action
> application="tone_detect" data="fax 1100 r +5000 transfer fax XML
> features" /> do not work together.
>
> Simply by removing fax_detect, the fax is detected beautifully.
>
> My problem now is trying to email the fax. ?I followed the
> instructions on the wiki at http://wiki.freeswitch.org/wiki/Mod_fax,
> but the dialplan <action application="system" .../> is not executed
> after the rxfax command. ?I know the script works because if I put the
> system command in another part of the dialplan and hard code the
> filename to attach, then the email is sent.
>
> ? ?<extension name="receivefax">
> ? ? ?<condition field="destination_number" expression="^receivefax$">
> ? ? ? ?<action application="answer" />
> ? ? ? ?<action application="playback" data="silence_stream://2000"/>
> ? ? ? ?<action application="rxfax"
> data="//usr//local//freeswitch//storage//${caller_id_number}-${uuid}.rxfax.tiff"/>
> ? ? ? ?<action application="system"
> data="/usr/local/freeswitch/scripts/emailfax.sh
> /usr/local/freeswitch/storage/${caller_id_number}-${uuid}.rxfax.tiff"/>
> ? ? ? ?<action application="hangup"/>
> ? ? ?</condition>
> ? ?</extension>
>
> ideas?
> Thanks!
>
> On Tue, Oct 6, 2009 at 1:32 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>> hi Mark,
>>
>> This is an inbound call leg and media channel (so far)? is open in reverse
>> direction only (application ringback). I'm afraid you have to answer the
>> call to be able to "hear" the fax tone.
>>
>> T.
>>
>>
>>
>> On Mon, Oct 5, 2009 at 2:32 PM, Michael Jerris <mike at jerris.com> wrote:
>>>
>>> Fax tones are not played by the remote machine until after answer, the
>>> tone_detect application starts a media bug that listens for the tone,
>>> can you confirm the tone is happening at all. ?Maybe the issue here is
>>> the timeout, try making that longer, or doing the tone_detect in
>>> execute_on_answer
>>>
>>> Mike
>>>
>>> On Oct 5, 2009, at 6:28 AM, Mark Campbell-Smith wrote:
>>>
>>> > Thanks for the response Mike,
>>> >
>>> > I read that page and this one (among others)
>>> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_fax_detect, but
>>> > I'm still lost. ?This is an extract of my dialplan
>>> >
>>> > ? ?<extension name="Local">
>>> > ? ? ?<condition field="destination_number" expression="^(10[01][0-9])
>>> > $">
>>> > ? ? ? ?<action application="set" data="dialed_extension=$1"/>
>>> > ? ? ? ?<action application="export" data="dialed_extension=$1"/>
>>> > ? ? ? ?<action application="set" data="ringback=${au-ring}"/>
>>> > ? ? ? ?<action application="fax_detect"/>
>>> > ? ? ? ?<action application="tone_detect" data="fax 1100 r +5000
>>> > transfer fax XML features" />
>>> > ? ? ? ?<action application="set" data="hangup_after_bridge=true"/>
>>> > ? ? ? ?<action application="set" data="continue_on_fail=true"/>
>>> > ? ? ? ?<action application="bridge" data="user/${dialed_extension}@$
>>> > {domain}"/>
>>> >
>>> > I would assume that on detecting a fax, the dialplan 'fax' is called
>>> > in context features. ?This never happens.
>>> >
>>> > When is the fax tone detected? ? Is it while the call is ringing or
>>> > can it be detected after the call is answered? ?My goal is to be able
>>> > to have the same extension for a voice and fax call. ?i assume that
>>> > the fax 'tones' are standardised and the ones on the wiki are correct?
>>> > Also, I guess this doesn't work with media bypass (which I don't
>>> > use).
>>> >
>>> > Thanks!
>>> >
>>> >
>>> > On Mon, Oct 5, 2009 at 9:56 AM, Michael Jerris <mike at jerris.com>
>>> > wrote:
>>> >> check out
>>> >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
>>> >>
>>> >> Note, you can't just have tone_detect as your last iten in the
>>> >> dialplan as the call will just get hung up.
>>> >>
>>> >> Mike
>>> >>
>>> >> On Oct 4, 2009, at 9:03 AM, Mark Campbell-Smith wrote:
>>> >>
>>> >>> Hi
>>> >>>
>>> >>> I was hoping someone could help me to setup the fax detection / tone
>>> >>> detection application.
>>> >>>
>>> >>> I want to be able to transfer an incoming fax to a specific
>>> >>> extension.
>>> >>> In my default.xml file, I have the following (extracted):
>>> >>>
>>> >>> ? ?<extension name="1000">
>>> >>> ? ? ?<condition field="destination_number" expression="^(10[01]
>>> >>> [0-9])
>>> >>> $">
>>> >>> ? ? ? ?<action application="fax_detect"/>
>>> >>> ? ? ? ?<action application="tone_detect" data="fax 1100 r +5000
>>> >>> transfer fax XML features" />
>>> >>>
>>> >>> I can't get the fax to be detected and transferred. ?Is there any
>>> >>> way
>>> >>> this can be done?
>>> >>>
>>> >>> Thanks!
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Shannon


From xengelpublicx at gmail.com  Tue Oct  6 00:21:03 2009
From: xengelpublicx at gmail.com (Vladimir Elizarov)
Date: Tue, 06 Oct 2009 11:21:03 +0400
Subject: [Freeswitch-users] stun not working in fs 1.0.4?
In-Reply-To: <4AA6F4E2-2138-439C-86E8-7554A5D110D4@freeswitch.org>
References: <4AC9E9EE.8090805@gmail.com>	<7C489261-F5F3-4B57-B807-21EC4A3947A2@freeswitch.org>	<4ACA31ED.1000002@gmail.com>
	<4AA6F4E2-2138-439C-86E8-7554A5D110D4@freeswitch.org>
Message-ID: <4ACAEFDF.4030604@gmail.com>

Brian West ?????:
> Because TRUNK is stable... its only fixes going in usually and if  
> things do break they don't stay broken for long.
>
> Ask anyone our trunk is more table then most commercial products.
>   
This separation of the branches a very bad influence on the packaging.
That is gathered deb-package trunk 15094. Man found in the trunk bug.
Must again rebuild the package from the new trunk...
> /b
>
> On Oct 5, 2009, at 12:50 PM, Vladimir Elizarov wrote:
>
>   
>> Ok. Brian, why fs no two branches of the stable and trunk?
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


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From moizchinoy at gmail.com  Tue Oct  6 00:58:17 2009
From: moizchinoy at gmail.com (Moiz Chinoy)
Date: Tue, 6 Oct 2009 11:58:17 +0400
Subject: [Freeswitch-users] Mobile Phone As GSM Gateway....
Message-ID: <29b888f80910060058m4298a829s414a322f8548ff81@mail.gmail.com>

Hi,

Is it possible to connect a mobile phone (GSM phone) to Freeswitch and
use this as a GSM gateway?

-- 
Regards,
Moiz Chinoy.


From yehavi.bourvine at gmail.com  Tue Oct  6 01:14:31 2009
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 6 Oct 2009 10:14:31 +0200
Subject: [Freeswitch-users] Bridge application with shared lines
Message-ID: <d361ca7b0910060114g3f23a102nb3508cf6e1e196fc@mail.gmail.com>

Hello,

  We have Polycom and SNOM phones running with FreeSwitch. The Polycoms have
shared lines defined and the SNOMs have both shared lines and BLFs (defined
as extensions in the phone config). I've tried supporting both, but have
some incompatibility:


   - When calling the Bridge application with data parameter of *sofia*/*
   profile-name/number at domain* the BLF works ok, but not the shared lines
   (i.e only one of the phones  rings).
   - When calling the Bridge application with data parameter of *
   ${sofia_contact(*/*profile-name/number at domain*)} shared lines work ok but
   BLF doesn't fire up.


How do I support both? Is there a way to know whether the destination is a
shared one and then chose one of the above formats?

                                  Thanks! __Yehavi:
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From itamar at ispbrasil.com.br  Tue Oct  6 01:17:44 2009
From: itamar at ispbrasil.com.br (Itamar Reis Peixoto)
Date: Tue, 6 Oct 2009 05:17:44 -0300
Subject: [Freeswitch-users] Mobile Phone As GSM Gateway....
In-Reply-To: <29b888f80910060058m4298a829s414a322f8548ff81@mail.gmail.com>
References: <29b888f80910060058m4298a829s414a322f8548ff81@mail.gmail.com>
Message-ID: <d3ceea380910060117mc02fb80lb059e92c8d0164f7@mail.gmail.com>

a gsm phone not.

take a look

http://portech.com.tw/

I think a portech product can do what you need.




On Tue, Oct 6, 2009 at 4:58 AM, Moiz Chinoy <moizchinoy at gmail.com> wrote:
> Hi,
>
> Is it possible to connect a mobile phone (GSM phone) to Freeswitch and
> use this as a GSM gateway?
>
> --
> Regards,
> Moiz Chinoy.




-- 
------------

Itamar Reis Peixoto

e-mail/msn: itamar at ispbrasil.com.br
sip: itamar at ispbrasil.com.br
skype: itamarjp
icq: 81053601
+55 11 4063 5033
+55 34 3221 8599


From woodydickson at gmail.com  Tue Oct  6 02:00:40 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Tue, 6 Oct 2009 17:00:40 +0800
Subject: [Freeswitch-users] problem with compiling freeswith
Message-ID: <a44688a0910060200q6b6bc15cjf5100b17e63b5bd4@mail.gmail.com>

Hi,

Is this just me who is having this problem?  I can't compile the latest
freeswitch source code and here is the error:

checking for gcc option to accept ANSI C... none needed
checking for style of include used by make... GNU
checking dependency style of gcc... gcc3
checking whether gcc and cc understand -c and -o together... yes
./configure: line 3377: syntax error near unexpected token `echo'
./configure: line 3377: `echo "$as_me:$LINENO: checking for a BSD-compatible
install" >&5'
configure: error: /bin/sh './configure.gnu' failed for libs/tiff-3.8.2
[root at localhost freeswitch-snapshot]#

Does anyone know why?

woody
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From rs at runsolutions.com  Tue Oct  6 02:02:46 2009
From: rs at runsolutions.com (Raimund Sacherer)
Date: Tue, 6 Oct 2009 11:02:46 +0200
Subject: [Freeswitch-users] Youtube - FreeSWITCH Promo Video
In-Reply-To: <191c3a030910050802v59b2a9du6d2af7ade03bc509@mail.gmail.com>
References: <9A87E77C-AD85-4F1E-8ED0-F9F06AA451EE@ken-ton.com>
	<86a32abc0910041651rb6a3ff9k1f6ed7d0b1a9b440@mail.gmail.com>
	<7b197bef0910050004v71c623c8w68a013bef9fea596@mail.gmail.com>
	<191c3a030910050802v59b2a9du6d2af7ade03bc509@mail.gmail.com>
Message-ID: <91D6B861-B8B1-4559-8494-A68D90A0FB4A@runsolutions.com>

Yeah, on top of it would'nt it be nice if:

when they call the giant 3d unix '*' character with the cell phone

* switch inside the con of the giant 3d unix '*'
* People nervous and shouting about "incoming" (like in the fight szenes
    when they call incoming if missiles are fired)
* One is calling "Oh no, one of our THREADs is blocking!!"
    (THREAD = Thermal Heat REAction Device, eg, self deploying cooling  
pipe)
    (or maybe Throttle Heat REAction Device), view of a pipe on the  
point of blocking
    the heat-xchange (e.g. simulate somehow that nothing goes further  
anymore).
* A high ranked officer is shouting
    "Core Dump, Core Dump, Leave the ships through all available  
Channels!"
* switch to outer scene
* you see an anatomically hinting, but technically correct crack  
forming at the rear
    bottom side of the giant 3d unix '*'
* out of this crack comes "the dumped core"
* switch to open scene view
* you see the people leaving the giant 3d unix '*' at all available  
channels
    (e.g. light pulses which look like little ships driven by rockets  
in every direction out of the
    gicant 3d unix '*')
* When the CoreDump is finished, the giant 3d unix '*' implodes in  
itself and leaves
    nothing but the fouly stench of the dumped core (which has to  
resemble somehow
    the same stench you feel after a callcenter full of agents again  
lost all connections
    because of a fat dumb giant 3d unix '*' could not cope with it's  
pipes and dumped its core)



....

:-)

-- 
Raimund Sacherer
-
RunSolutions
     Open Source It Consulting
-
Email: rs at runsolutions.com
tel: 625 40 32 08

Parc Bit - Centro Empresarial Son Espanyol
Edificio Estel - Local 3D
07121 -  Palma de Mallorca
Baleares

On Oct 5, 2009, at 5:02 PM, Anthony Minessale wrote:

> neat,
>
> Here's some suggestions for your next ones. =p
>
> Have them standing around the hologram trying to destroy the "Death  
> Star(tm)" that happens to look a lot like a giant 3d unix '*'  
> character.  Then have one rebel say, "wait!, why are we wasting our  
> time... watch this... and dial a number on his cellphone as the  
> whole thing explodes in the background.
>
> Have Darth Forkium face Luke ThreadSpawner in a dual.  "I see you  
> have fashioned your own TDM card...." vroom......
> "Join me and together we can make linked lists and monolithic  
> processes", "NEVER!..." vroom vroom Master Coda has taught you  
> well....."You are no match for me...JOIN THE ORANGE SIDE OF THE FORCE"
>
>
>
>
>
> On Mon, Oct 5, 2009 at 2:04 AM, Giovanni Maruzzelli <gmaruzz at celliax.org 
> > wrote:
> On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola <diego.viola at gmail.com>  
> wrote:
> > Very nice :)
> >
> > On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling <kjv at ken-ton.com>  
> wrote:
> >>
> >> Folks;
> >> Here's something that I did playing around w/ learning Apple  
> Motion.
>
> Me too: very nice!
>
> -gmaruzz
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From jason at jasonjgw.net  Tue Oct  6 02:18:58 2009
From: jason at jasonjgw.net (Jason White)
Date: Tue, 6 Oct 2009 20:18:58 +1100
Subject: [Freeswitch-users] problem with compiling freeswith
In-Reply-To: <a44688a0910060200q6b6bc15cjf5100b17e63b5bd4@mail.gmail.com>
References: <a44688a0910060200q6b6bc15cjf5100b17e63b5bd4@mail.gmail.com>
Message-ID: <20091006091858.GA18854@jdc.jasonjgw.net>

Woody Dickson <woodydickson at gmail.com> wrote:
> Is this just me who is having this problem?  I can't compile the latest
> freeswitch source code and here is the error:

Try starting with a fresh checkout from the repository.

If the problem persists, please report the operating system and version
thereof, so that someone with access to a similar environment can try to
reproduce the issue.



From m.krivushin at imarto.net  Tue Oct  6 02:25:38 2009
From: m.krivushin at imarto.net (Mikhail Krivushin)
Date: Tue, 6 Oct 2009 16:25:38 +0700
Subject: [Freeswitch-users] fs_path not work
Message-ID: <5be734a50910060225mcef587dw69b12132303dadab@mail.gmail.com>

Hello!

I try to use ;fs_path in originate command, but this seems to not work:
bgapi originate
{origination_caller_id_name=qwe,origination_caller_id_number=qwe,sip_auth_username=qwe,sip_auth_password=qwe,origination_uuid=1ebf2ef8-b259-11de-b7f9-000c29cf246f,fsc_call=1ebf0432-b259-11de-b7f9-000c29cf246f,fsc_leg=legB}sofia/gateway/qwe_gw/
test05 at service.deep.com;fs_path=sip:vm3.deep.com &park

Where I am wrong?

-- 
? ?????????, ???????? ??????
?. ????? ???. +7 913 865 78 66
icq: 218 744 127
xmpp: KrivushinME at jabber.ru
skype: mkrivushin
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From m.krivushin at imarto.net  Tue Oct  6 02:41:43 2009
From: m.krivushin at imarto.net (Mikhail Krivushin)
Date: Tue, 6 Oct 2009 16:41:43 +0700
Subject: [Freeswitch-users] fs_path not work
In-Reply-To: <5be734a50910060225mcef587dw69b12132303dadab@mail.gmail.com>
References: <5be734a50910060225mcef587dw69b12132303dadab@mail.gmail.com>
Message-ID: <5be734a50910060241r707d7018m95337f464577a4fb@mail.gmail.com>

I also try to use "proxy" param in gateway, but this doesnt work too. INVITE
dont going to proxy pointed by me.

-- 
? ?????????, ???????? ??????
?. ????? ???. +7 913 865 78 66
icq: 218 744 127
xmpp: KrivushinME at jabber.ru
skype: mkrivushin
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From ahmedmunir007 at gmail.com  Tue Oct  6 03:11:03 2009
From: ahmedmunir007 at gmail.com (Ahmed Munir)
Date: Tue, 6 Oct 2009 15:11:03 +0500
Subject: [Freeswitch-users] Cannot connect to ODBC driver/database
	freeswitchdb
Message-ID: <a60fed5e0910060311w4ceb749cu520c877bb47c47b8@mail.gmail.com>

Hi,
I've installed FS on Ubuntu 9.04 and I want to run mod_nibbles on it. I
follow the steps to configure my ODBC connection with MySQL as explained in
wiki (mod_nibbles and mod_spidermonkey). But  FS, unable to connect it. The
error I got is listed below when I restart FS,

2009-10-06 15:47:21.164590 [ERR] switch_odbc.c:188 STATE: IM002 CODE 0
ERROR: [unixODBC][Driver Manager]Data source name not found, and no default
driver specified
2009-10-06 15:47:21.164617 [CRIT] mod_nibblebill.c:221 Cannot connect to
ODBC driver/database freeswitchdb (user: root / pass password)
2009-10-06 15:47:21.164650 [CONSOLE] switch_loadable_module.c:889
Successfully Loaded [mod_nibblebill]
2009-10-06 15:47:21.164664 [NOTICE] switch_loadable_module.c:248 Adding
Application 'nibblebill'
2009-10-06 15:47:21.164710 [NOTICE] switch_loadable_module.c:270 Adding API
Function 'nibblebill'

But when I use isql it accepts my odbc connection i.e. isql MySQL-freeswitch

I'm listing my settings of odbc.ini and odbcinst.ini as listed below;

odbc.ini
--------------
[MySQL-freeswitch]
Driver       = MySQL
#Driver       = /usr/lib/odbc/libodbcmyS.so
Description  = Connector/ODBC Driver DSN With FreeSwitch
SERVER       = localhost
PORT         = 3306
USER         = root
Password     = password
Database     = freeswitchdb

odbcinst.ini
-------------------
[MySQL]
Description    = ODBC for MySQL
Driver         = /usr/lib/odbc/libmyodbc.so
Setup          = /usr/lib/odbc/libodbcmyS.so
FileUsage      = 1


odbc.ini and odbcinst.ini are located at /etc/. Even I set my odbc
connection setting as I provide with this link;
http://dev.mysql.com/doc/refman/5.0/en/connector-odbc-configuration-dsn-unix.html

But unfortunately my problem is unresolved then.


Kindly advise me, how can I resolve this problem?

-- 
Regards,

Ahmed Munir
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From asannucci at gmail.com  Tue Oct  6 03:27:46 2009
From: asannucci at gmail.com (bakko)
Date: Tue, 6 Oct 2009 05:27:46 -0500
Subject: [Freeswitch-users] Cannot connect to ODBC
	driver/databasefreeswitchdb
In-Reply-To: <a60fed5e0910060311w4ceb749cu520c877bb47c47b8@mail.gmail.com>
References: <a60fed5e0910060311w4ceb749cu520c877bb47c47b8@mail.gmail.com>
Message-ID: <EA1280B80AF74924955FA379860DE26E@voztovoice>

Hi,

are you configured correctly the nibblebill.conf.xml file?

BR
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From lakindia89 at gmail.com  Tue Oct  6 03:30:03 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Tue, 6 Oct 2009 16:00:03 +0530
Subject: [Freeswitch-users] Outgoing via openzap is not working
Message-ID: <7d79b3930910060330s77a2bbbv88d8014b73f91663@mail.gmail.com>

Hi I'm using freeswitch1.0.4. This post is moreover similar to my previous
post.
When I make an outgoing call, it is saying INVALID_IE_CONTENTS.
Here are the details.
openzap.conf.xml

<configuration name="openzap.conf" description="OpenZAP Configuration">
<settings>
<param name="debug" value="7"/>
</settings>
<libpri_spans>
<span name="PRI_1">
<param name="node" value="cpe"/>
<param name="switch" value="euroisdn"/>
<param name="dialplan" value="XML"/>
<param name="context" value="default"/>
</span>
</libpri_spans>
</configuration>

openzap.conf
[span zt PRI_1]
trunk_type => e1
b-channel => 1:1-15
d-channel=> 1:16
b-channel => 1:17-31
oz libpri debug 1 all

API CALL [oz(libpri debug 1 all)] output:
src/ozmod/ozmod_libpri/ozmod_libpri.c: +OK debug set.

oz list

API CALL [oz(list)] output:
+OK
span: 1 (PRI_1)
type: isdn
chan_count: 47
dialplan: XML
context: default
dial_regex:
fail_dial_regex:
hold_music:
analog_options none

Freeswitch startup log:
http://pastebin.freeswitch.org/10609
After saying originate openzap/1/1/9952248266 openzap/1/1/9952248266
http://pastebin.freeswitch.org/10610

Please help me to solve this.
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From dujinfang at gmail.com  Tue Oct  6 05:04:58 2009
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 6 Oct 2009 20:04:58 +0800
Subject: [Freeswitch-users] Mobile Phone As GSM Gateway....
In-Reply-To: <29b888f80910060058m4298a829s414a322f8548ff81@mail.gmail.com>
References: <29b888f80910060058m4298a829s414a322f8548ff81@mail.gmail.com>
Message-ID: <23f91030910060504y3d060990l44ec913fb130954c@mail.gmail.com>

maybe you can check this: http://www.gsmopen.org/

2009/10/6 Moiz Chinoy <moizchinoy at gmail.com>

> Hi,
>
> Is it possible to connect a mobile phone (GSM phone) to Freeswitch and
> use this as a GSM gateway?
>
> --
> Regards,
> Moiz Chinoy.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From srinivas.ksvreddy at gmail.com  Tue Oct  6 05:29:25 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 6 Oct 2009 17:59:25 +0530
Subject: [Freeswitch-users] Dynamic updation of groups in default.xml
Message-ID: <f8af5740910060529v5d415d6bla4e54121d8ad2ff@mail.gmail.com>

Hi,
Can any one tell me how to add users dynamically to groups in default.xml,
with out restart the freeswitch.


Thanks
Srinivasula Reddy K
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From srinivas.ksvreddy at gmail.com  Tue Oct  6 05:31:52 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 6 Oct 2009 18:01:52 +0530
Subject: [Freeswitch-users] add users dynamically to groups in default.xml
Message-ID: <f8af5740910060531j58158057h881d9862602d360c@mail.gmail.com>

Hi,
Can any one tell me how to add users dynamically to groups in default.xml,
with out restart the freeswitch.

Thanks
Srinivasula Reddy K
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From dujinfang at gmail.com  Tue Oct  6 05:38:50 2009
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 6 Oct 2009 20:38:50 +0800
Subject: [Freeswitch-users] Dynamic updation of groups in default.xml
In-Reply-To: <f8af5740910060529v5d415d6bla4e54121d8ad2ff@mail.gmail.com>
References: <f8af5740910060529v5d415d6bla4e54121d8ad2ff@mail.gmail.com>
Message-ID: <23f91030910060538x6b1b2bddwa9f287e4e5c0251d@mail.gmail.com>

change the xml and execute "reloadxml" in FS console or fs_cli

or you can check mod_xml_curl

2009/10/6 srinivasula reddy <srinivas.ksvreddy at gmail.com>

> Hi,
> Can any one tell me how to add users dynamically to groups in default.xml,
> with out restart the freeswitch.
>
>
> Thanks
> Srinivasula Reddy K
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tomabroad at gmail.com  Tue Oct  6 05:44:51 2009
From: tomabroad at gmail.com (tom)
Date: Tue, 6 Oct 2009 08:44:51 -0400
Subject: [Freeswitch-users] Wiki Info: Multi Homed Tutorial
In-Reply-To: <a87d43fd0910052120l76da1536ra92766174dc938d3@mail.gmail.com>
References: <87f2f3b90910051527x638f55b8hda75d5302267c251@mail.gmail.com>
	<a87d43fd0910052120l76da1536ra92766174dc938d3@mail.gmail.com>
Message-ID: <6f7c60c40910060544h5d1f7251p7194bcaa944a627c@mail.gmail.com>

thx guys, that helped me alot!


On Tue, Oct 6, 2009 at 12:20 AM, Michael Gende <mgende at gendesign.com> wrote:

> Michael,
>
> Thanks for "wiki-fying" my text-only attempt at some user doc. I should
> have done that for you. I actually have an updated version with many
> corrections and the end tabs filled in. Can you point me to info on how I
> can amend and append what you have kindly put up?
>
> Mike G.
>
>
> On Mon, Oct 5, 2009 at 5:27 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> FYI,
>>
>> For those who've been following the thread about Michael Gende's tutorial
>> I just wanted to let you know that I his document on the wiki. It can be
>> found here:
>>
>> http://wiki.freeswitch.org/wiki/Multi_home_tutorial
>>
>> Please feel free to get in there and try it out, make editorial changes,
>> etc. I'm sure you won't hurt Mike's feelings by adding your thoughts. :)
>>
>> If you have any questions please reply to this thread and we'll take it
>> from there.
>>
>> Thanks,
>> MC
>>
>>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kjv at ken-ton.com  Tue Oct  6 05:51:05 2009
From: kjv at ken-ton.com (Karl Vesterling)
Date: Tue, 6 Oct 2009 08:51:05 -0400
Subject: [Freeswitch-users] Youtube - FreeSWITCH Promo Video
In-Reply-To: <91D6B861-B8B1-4559-8494-A68D90A0FB4A@runsolutions.com>
References: <9A87E77C-AD85-4F1E-8ED0-F9F06AA451EE@ken-ton.com>
	<86a32abc0910041651rb6a3ff9k1f6ed7d0b1a9b440@mail.gmail.com>
	<7b197bef0910050004v71c623c8w68a013bef9fea596@mail.gmail.com>
	<191c3a030910050802v59b2a9du6d2af7ade03bc509@mail.gmail.com>
	<91D6B861-B8B1-4559-8494-A68D90A0FB4A@runsolutions.com>
Message-ID: <E83DE12B-8728-4778-B2E6-B44E1A0A347F@ken-ton.com>


I'm flattered that you consider my abilities so capable, but time nor  
budget are available to afford such extravagance as outlined below.

Besides, bashing something doesn't really gain you any respect (but I  
do think that _*_ does ever sooo much deserve bashing).

What I have done is added to the end of the movie a Cisco 7900 series  
(top left), a Snom 360 (top right), and a Grandstream video phone  
(bottom right.)

I need more though...

I was thinking the point of the video was more along the lines to get  
people interested in FreeSWITCH, not bash such an easy target as  
(insert * here).

I think that can be done by showing various equipment from multiple  
manufacturers at the end of the movie, with another title declaring,  
"INTEROPERABILITY"...

Fade some in, fade out, fade some more in, perhaps including a sangoma  
card, over and over again...

So far I've got artwork for Cisco 7900, Snom, Grandstream...  I could  
put in a SPA-942, etc...
These are the type of "suggestions" I'm looking for... Although I  
really do like the drama with, "Oh no!  One of our threads is  
blocking!" (still chuckling...)  It did give me a few ideas too, so  
keep em' coming!

Best Regards,
Karl J. Vesterling
kjv at ken-ton.com
202-461-3231 x0

On Oct 6, 2009, at 5:02 AM, Raimund Sacherer wrote:

> Yeah, on top of it would'nt it be nice if:
>
> when they call the giant 3d unix '*' character with the cell phone
>
> * switch inside the con of the giant 3d unix '*'
> * People nervous and shouting about "incoming" (like in the fight  
> szenes
>    when they call incoming if missiles are fired)
> * One is calling "Oh no, one of our THREADs is blocking!!"
>    (THREAD = Thermal Heat REAction Device, eg, self deploying cooling
> pipe)
>    (or maybe Throttle Heat REAction Device), view of a pipe on the
> point of blocking
>    the heat-xchange (e.g. simulate somehow that nothing goes further
> anymore).
> * A high ranked officer is shouting
>    "Core Dump, Core Dump, Leave the ships through all available
> Channels!"
> * switch to outer scene
> * you see an anatomically hinting, but technically correct crack
> forming at the rear
>    bottom side of the giant 3d unix '*'
> * out of this crack comes "the dumped core"
> * switch to open scene view
> * you see the people leaving the giant 3d unix '*' at all available
> channels
>    (e.g. light pulses which look like little ships driven by rockets
> in every direction out of the
>    gicant 3d unix '*')
> * When the CoreDump is finished, the giant 3d unix '*' implodes in
> itself and leaves
>    nothing but the fouly stench of the dumped core (which has to
> resemble somehow
>    the same stench you feel after a callcenter full of agents again
> lost all connections
>    because of a fat dumb giant 3d unix '*' could not cope with it's
> pipes and dumped its core)
>
>
>
> ....
>
> :-)
>
> -- 
> Raimund Sacherer
> -
> RunSolutions
>     Open Source It Consulting
> -
> Email: rs at runsolutions.com
> tel: 625 40 32 08
>
> Parc Bit - Centro Empresarial Son Espanyol
> Edificio Estel - Local 3D
> 07121 -  Palma de Mallorca
> Baleares
>
> On Oct 5, 2009, at 5:02 PM, Anthony Minessale wrote:
>
>> neat,
>>
>> Here's some suggestions for your next ones. =p
>>
>> Have them standing around the hologram trying to destroy the "Death
>> Star(tm)" that happens to look a lot like a giant 3d unix '*'
>> character.  Then have one rebel say, "wait!, why are we wasting our
>> time... watch this... and dial a number on his cellphone as the
>> whole thing explodes in the background.
>>
>> Have Darth Forkium face Luke ThreadSpawner in a dual.  "I see you
>> have fashioned your own TDM card...." vroom......
>> "Join me and together we can make linked lists and monolithic
>> processes", "NEVER!..." vroom vroom Master Coda has taught you
>> well....."You are no match for me...JOIN THE ORANGE SIDE OF THE  
>> FORCE"
>>
>>
>>
>>
>>
>> On Mon, Oct 5, 2009 at 2:04 AM, Giovanni Maruzzelli <gmaruzz at celliax.org
>>> wrote:
>> On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola <diego.viola at gmail.com>
>> wrote:
>>> Very nice :)
>>>
>>> On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling <kjv at ken-ton.com>
>> wrote:
>>>>
>>>> Folks;
>>>> Here's something that I did playing around w/ learning Apple
>> Motion.
>>
>> Me too: very nice!
>>
>> -gmaruzz
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>>
>>
>> -- 
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:213-799-1400
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>



From Russell.Mosemann at cune.org  Tue Oct  6 05:59:50 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Tue, 6 Oct 2009 12:59:50 -0000
Subject: [Freeswitch-users] Outgoing via openzap is not working
In-Reply-To: <7d79b3930910060330s77a2bbbv88d8014b73f91663@mail.gmail.com>
Message-ID: <20091006125950.EB7073DECDF@mail.cune.org>

lakshmanan ganapathy <lakindia89 at gmail.com> said:

> When I make an outgoing call, it is saying INVALID_IE_CONTENTS.
> Here are the details.

There are a couple of things that might be OK, but they seem odd. When
ozmod starts, the first 15 channels can't be configured, because they are
busy. The output from "oz list" shows 47 channels, as if 15 channels are
added to the E1's 31 channels (and one control channel). When libpri is
checking the numbers, it shows the caller's number as '0000000000'.

The output from libpri shows that the call is being released by the other
side of the E1 because of a protocol error involving the redirecting
number (invalid contents). My guess is that you aren't providing a valid
caller number, but I could be completely wrong. Someone with more
experience will need to interpret the information. I don't see the
originate command in the debug output. Did you supply both the calling
and called number?

http://wiki.freeswitch.org/wiki/Mod_commands#originate

-- 
Russell Mosemann

________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From anthony.minessale at gmail.com  Tue Oct  6 06:49:32 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 6 Oct 2009 08:49:32 -0500
Subject: [Freeswitch-users] fs_path not work
In-Reply-To: <5be734a50910060241r707d7018m95337f464577a4fb@mail.gmail.com>
References: <5be734a50910060225mcef587dw69b12132303dadab@mail.gmail.com>
	<5be734a50910060241r707d7018m95337f464577a4fb@mail.gmail.com>
Message-ID: <191c3a030910060649o16f15cd9q7443cbf406f13041@mail.gmail.com>

gateway calls do not contain any uri data
sofia/gateway/mygw/1000
is all you can do

if you want all that other stuff you need to formulate a direct url
connection


On Tue, Oct 6, 2009 at 4:41 AM, Mikhail Krivushin <m.krivushin at imarto.net>wrote:

> I also try to use "proxy" param in gateway, but this doesnt work too.
> INVITE dont going to proxy pointed by me.
>
>
> --
> ? ?????????, ???????? ??????
> ?. ????? ???. +7 913 865 78 66
> icq: 218 744 127
> xmpp: KrivushinME at jabber.ru
> skype: mkrivushin
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From brian at freeswitch.org  Tue Oct  6 07:31:06 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 6 Oct 2009 09:31:06 -0500
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <1b46b4e80910052142jaa09f08l72e5e4b2b24d10fc@mail.gmail.com>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<191c3a030909291122t770af037j5b1068197b31de87@mail.gmail.com>
	<1b46b4e80909291542k71440965g75bb3bfcf38e8005@mail.gmail.com>
	<1b46b4e80910011229k31f7233du10b0fc661730d3f3@mail.gmail.com>
	<F86F950D-9106-49C9-81CA-38B3EB2072F4@jerris.com>
	<1b46b4e80910041519h3581d3d2x7b84041e8c18588@mail.gmail.com>
	<C887D143-F7A1-44B9-B8A2-D34A4B4E582E@jerris.com>
	<1b46b4e80910041609x2f41ceb9u142cb484951a21ed@mail.gmail.com>
	<87f2f3b90910050920w4d17033fl4c386a0018107bdf@mail.gmail.com>
	<1b46b4e80910052119me12a0d9y4653edd831f9555e@mail.gmail.com>
	<1b46b4e80910052142jaa09f08l72e5e4b2b24d10fc@mail.gmail.com>
Message-ID: <083DBCB9-48B2-4E61-81B5-B7D00A42EAB5@freeswitch.org>

This looks like you have an ALG messing with packets... notice it says  
rport 5080 but we are sending to 5060.

/b

On Oct 5, 2009, at 11:42 PM, Nicolas Brenner wrote:

> Ignore my previous email, the traces were incomplete, got much  
> better (and complete) traces with ngrep (found a suggestion from  
> Brian in the list archive, thanks!)
>
> The gateway that registers:
>
> - http://pastebin.freeswitch.org/10607
>
> The one that doesn't:
>
> - http://pastebin.freeswitch.org/10608
>
>
> Thanks again for your time and help!
>
>
> Nicolas
>
>
> On Tue, Oct 6, 2009 at 12:19 AM, Nicolas Brenner <nicolas at medularis.com 
> > wrote:
> There was no sane way of doing that, so I ended up logging the trace  
> from the cli.

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From srinivas.ksvreddy at gmail.com  Tue Oct  6 07:38:57 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 6 Oct 2009 20:08:57 +0530
Subject: [Freeswitch-users] Dynamic updation of groups in default.xml
In-Reply-To: <23f91030910060538x6b1b2bddwa9f287e4e5c0251d@mail.gmail.com>
References: <f8af5740910060529v5d415d6bla4e54121d8ad2ff@mail.gmail.com>
	<23f91030910060538x6b1b2bddwa9f287e4e5c0251d@mail.gmail.com>
Message-ID: <f8af5740910060738l111b6ab7y2c5b05c505951c4e@mail.gmail.com>

Thank u very much seven du.

Regards
Srinvas


On Tue, Oct 6, 2009 at 6:08 PM, Seven Du <dujinfang at gmail.com> wrote:

> change the xml and execute "reloadxml" in FS console or fs_cli
>
> or you can check mod_xml_curl
>
> 2009/10/6 srinivasula reddy <srinivas.ksvreddy at gmail.com>
>
>> Hi,
>> Can any one tell me how to add users dynamically to groups in
>> default.xml,  with out restart the freeswitch.
>>
>>
>> Thanks
>> Srinivasula Reddy K
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Srinivasula Reddy K
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From ChristianDamianidis at globalive.com  Tue Oct  6 07:39:27 2009
From: ChristianDamianidis at globalive.com (Christian Damianidis)
Date: Tue, 6 Oct 2009 10:39:27 -0400
Subject: [Freeswitch-users] mod_xml_curl http POST is inconsistent/bug
In-Reply-To: <276FEE28-70CF-422D-9F6F-7CF747FDDB22@freeswitch.org>
References: <66EA3166EB339A4489B06286C0876A8A0C0C1432@mailserv.Globalive.local>
	<276FEE28-70CF-422D-9F6F-7CF747FDDB22@freeswitch.org>
Message-ID: <66EA3166EB339A4489B06286C0876A8A0C0C1BA9@mailserv.Globalive.local>

This web request goes to a server running IIS on Windows Server 2003.

 

From: Brian West [mailto:brian at freeswitch.org] 
Sent: Monday, October 05, 2009 5:43 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_xml_curl http POST is
inconsistent/bug

 

Are you using something other than apache?

 

/b

 

On Oct 5, 2009, at 1:25 PM, Christian Damianidis wrote:





 

The inconsistent POST request sent by the module causes freeswitch to
hang for 1-2 minutes during start-up.

 

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From anthony.minessale at gmail.com  Tue Oct  6 08:01:46 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 6 Oct 2009 10:01:46 -0500
Subject: [Freeswitch-users] mod_xml_curl http POST is inconsistent/bug
In-Reply-To: <66EA3166EB339A4489B06286C0876A8A0C0C1BA9@mailserv.Globalive.local>
References: <66EA3166EB339A4489B06286C0876A8A0C0C1432@mailserv.Globalive.local>
	<276FEE28-70CF-422D-9F6F-7CF747FDDB22@freeswitch.org>
	<66EA3166EB339A4489B06286C0876A8A0C0C1BA9@mailserv.Globalive.local>
Message-ID: <191c3a030910060801x3ed48e98x6d69b53c6beaca69@mail.gmail.com>

My guess is that we configure the curl to support the full range of http
auth methods.
Some of them like Digest require a challenge and realm etc so it's probably
asking without auth header because it cannot create one until it gets that
data.  In the case of Basic you can send the login and pass right away but
it does not know in advance that it will be basic.

Here is a snippet from the libcurl api docs:
-------------------------------------------------------------------------------------------------------------------------------------------------------------

Both these options allow you to set multiple types (by ORing them together),
to make libcurl pick the most secure one out of the types the server/proxy
claims to support. This method does however add a round-trip since libcurl
must first ask the server what it supports:

 curl_easy_setopt(easyhandle, CURLOPT_HTTPAUTH,
 CURLAUTH_DIGEST|CURLAUTH_BASIC);
-------------------------------------------------------------------------------------------------------------------------------------------------------------

So my guess is that if we set it to only support basic, then it would work
how you expect so if you want to test it for me I can make it into a
parameter.

edit: /usr/src/freeswitch.trunk/src/mod/xml_int/mod_xml_curl/mod_xml_curl.c
line 220
change

curl_easy_setopt(curl_handle, CURLOPT_HTTPAUTH, CURLAUTH_ANY);

to

curl_easy_setopt(curl_handle, CURLOPT_HTTPAUTH, CURLAUTH_BASIC);


If this works i'll think about exposing the auth methods so you can choose
them in the config.


On Tue, Oct 6, 2009 at 9:39 AM, Christian Damianidis <
ChristianDamianidis at globalive.com> wrote:

>  This web request goes to a server running IIS on Windows Server 2003.
>
>
>
> *From:* Brian West [mailto:brian at freeswitch.org]
> *Sent:* Monday, October 05, 2009 5:43 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] mod_xml_curl http POST is
> inconsistent/bug
>
>
>
> Are you using something other than apache?
>
>
>
> /b
>
>
>
> On Oct 5, 2009, at 1:25 PM, Christian Damianidis wrote:
>
>
>
>
>
> The inconsistent POST request sent by the module causes freeswitch to hang
> for 1-2 minutes during start-up.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From woodydickson at gmail.com  Tue Oct  6 08:05:23 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Tue, 6 Oct 2009 23:05:23 +0800
Subject: [Freeswitch-users] overriding conference preference
In-Reply-To: <E773DB8B-AC74-49DE-9B38-7AFC441882F5@jerris.com>
References: <a44688a0910050749s1ae1f9a4vba96477d159d377f@mail.gmail.com>
	<E773DB8B-AC74-49DE-9B38-7AFC441882F5@jerris.com>
Message-ID: <a44688a0910060805o49d549b8vbd4d545b229c75d@mail.gmail.com>

Hi,

Is there anyway of using curl without having to setup a standalone http
service?  Is it possible to generate curl xml using scripts?

woody

On Mon, Oct 5, 2009 at 11:05 PM, Michael Jerris <mike at jerris.com> wrote:

>
> On Oct 5, 2009, at 10:49 AM, Woody Dickson wrote:
>
> > Is is possible to override any of the setting specified in the
> > conference profile?
>
> Just the flags you can pass per user such as pin and mute
>
> >
> > What I want to do is to have a default profile, and be able to
> > modify certain fields if necessary in the dialplan.
> >
> >
> > Alternatively, I would prefer to have a dynamic profile setting for
> > the conference to obtain those parameters from odbc.
>
> you can do this with mod_xml_curl
>
> Mike
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From todd.baumgartner at gmail.com  Tue Oct  6 07:31:06 2009
From: todd.baumgartner at gmail.com (Todd Baumgartner)
Date: Tue, 6 Oct 2009 10:31:06 -0400
Subject: [Freeswitch-users] Cannot connect to ODBC driver/database
	freeswitchdb
In-Reply-To: <a60fed5e0910060311w4ceb749cu520c877bb47c47b8@mail.gmail.com>
References: <a60fed5e0910060311w4ceb749cu520c877bb47c47b8@mail.gmail.com>
Message-ID: <dc8dcf060910060731j2e3961a1ra65f580101ec411d@mail.gmail.com>

Ahmed,

I believe you need to specify the database name as it is configured in the
odbc.ini

I am assuming you have something like this in your nibblebill.conf.xml

    <param name="db_dsn" value="freeswitchdb"/>

Try changing it to this as it is named in the odbc.ini:

    <param name="db_dsn" value="MySQL-freeswitch"/>


Thanks,

Todd



On Tue, Oct 6, 2009 at 6:11 AM, Ahmed Munir <ahmedmunir007 at gmail.com> wrote:

> Hi,
> I've installed FS on Ubuntu 9.04 and I want to run mod_nibbles on it. I
> follow the steps to configure my ODBC connection with MySQL as explained in
> wiki (mod_nibbles and mod_spidermonkey). But  FS, unable to connect it. The
> error I got is listed below when I restart FS,
>
> 2009-10-06 15:47:21.164590 [ERR] switch_odbc.c:188 STATE: IM002 CODE 0
> ERROR: [unixODBC][Driver Manager]Data source name not found, and no default
> driver specified
> 2009-10-06 15:47:21.164617 [CRIT] mod_nibblebill.c:221 Cannot connect to
> ODBC driver/database freeswitchdb (user: root / pass password)
> 2009-10-06 15:47:21.164650 [CONSOLE] switch_loadable_module.c:889
> Successfully Loaded [mod_nibblebill]
> 2009-10-06 15:47:21.164664 [NOTICE] switch_loadable_module.c:248 Adding
> Application 'nibblebill'
> 2009-10-06 15:47:21.164710 [NOTICE] switch_loadable_module.c:270 Adding API
> Function 'nibblebill'
>
> But when I use isql it accepts my odbc connection i.e. isql
> MySQL-freeswitch
>
> I'm listing my settings of odbc.ini and odbcinst.ini as listed below;
>
> odbc.ini
> --------------
> [MySQL-freeswitch]
> Driver       = MySQL
> #Driver       = /usr/lib/odbc/libodbcmyS.so
> Description  = Connector/ODBC Driver DSN With FreeSwitch
> SERVER       = localhost
> PORT         = 3306
> USER         = root
> Password     = password
> Database     = freeswitchdb
>
> odbcinst.ini
> -------------------
> [MySQL]
> Description    = ODBC for MySQL
> Driver         = /usr/lib/odbc/libmyodbc.so
> Setup          = /usr/lib/odbc/libodbcmyS.so
> FileUsage      = 1
>
>
> odbc.ini and odbcinst.ini are located at /etc/. Even I set my odbc
> connection setting as I provide with this link;
> http://dev.mysql.com/doc/refman/5.0/en/connector-odbc-configuration-dsn-unix.html
>
> But unfortunately my problem is unresolved then.
>
>
> Kindly advise me, how can I resolve this problem?
>
> --
> Regards,
>
> Ahmed Munir
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Tue Oct  6 08:12:58 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 6 Oct 2009 11:12:58 -0400
Subject: [Freeswitch-users] Basic compile question.
In-Reply-To: <6.2.3.4.2.20091005232815.03231998@pop-server.tx.rr.com>
References: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>
	<e403cddca1c90c83ce4fb6f85b026e63.squirrel@correo.nodo50.org>
	<65d96fc80910020532i4ba33353g7ac141624a4bb89e@mail.gmail.com>
	<C1AD67AA-F969-4BA1-8415-5E381BDF18FA@jerris.com>
	<90823c940910051257u403108afg3478c4c13ab26283@mail.gmail.com>
	<7e2ac3270910051319xfdfc9d3wa583c2da7e932339@mail.gmail.com>
	<6.2.3.4.2.20091005154158.02b632a8@pop-server.tx.rr.com>
	<4C985F33-8B02-4F8F-95B5-36BA6B40690F@freeswitch.org>
	<6.2.3.4.2.20091005232815.03231998@pop-server.tx.rr.com>
Message-ID: <B5B67EEE-6103-4BEF-A064-A2D6110C77E4@jerris.com>

As I said in the duplicate thread, the voip codecs issue has been  
resolved in trunk, I had a change 1/2 done waiting for testing and it  
is now complete.

Mike

On Oct 6, 2009, at 12:30 AM, David Clark wrote:

> No I found the one header.  I added it to the include list for the
> project.  It included something else, added that.  etc.  Basically I
> think I am going to need the VC 2008
> compiler  and to use the other project file.
>
> At 04:41 PM 10/5/2009, Brian West wrote:
>> Have you updated today?
>>
>> /b
>>
>> On Oct 5, 2009, at 3:49 PM, David Clark wrote:
>>
>>> Any idea what is up?
>>



From mike at jerris.com  Tue Oct  6 08:15:10 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 6 Oct 2009 11:15:10 -0400
Subject: [Freeswitch-users] stun not working in fs 1.0.4?
In-Reply-To: <4ACAEFDF.4030604@gmail.com>
References: <4AC9E9EE.8090805@gmail.com>	<7C489261-F5F3-4B57-B807-21EC4A3947A2@freeswitch.org>	<4ACA31ED.1000002@gmail.com>
	<4AA6F4E2-2138-439C-86E8-7554A5D110D4@freeswitch.org>
	<4ACAEFDF.4030604@gmail.com>
Message-ID: <177D2C13-C1CC-4C00-BD6B-18A8F2F69366@jerris.com>

I am not sure what you mean, do you think that fixes from today should  
somehow go somewhere else before we do a release?


On Oct 6, 2009, at 3:21 AM, Vladimir Elizarov wrote:

> Brian West ?????:
>> Because TRUNK is stable... its only fixes going in usually and if
>> things do break they don't stay broken for long.
>>
>> Ask anyone our trunk is more table then most commercial products.
>>
> This separation of the branches a very bad influence on the packaging.
> That is gathered deb-package trunk 15094. Man found in the trunk bug.
> Must again rebuild the package from the new trunk...
>> /b
>>
>> On Oct 5, 2009, at 12:50 PM, Vladimir Elizarov wrote:
>>
>>
>>> Ok. Brian, why fs no two branches of the stable and trunk?
>>>



From mike at jerris.com  Tue Oct  6 08:19:27 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 6 Oct 2009 11:19:27 -0400
Subject: [Freeswitch-users] overriding conference preference
In-Reply-To: <a44688a0910060805o49d549b8vbd4d545b229c75d@mail.gmail.com>
References: <a44688a0910050749s1ae1f9a4vba96477d159d377f@mail.gmail.com>
	<E773DB8B-AC74-49DE-9B38-7AFC441882F5@jerris.com>
	<a44688a0910060805o49d549b8vbd4d545b229c75d@mail.gmail.com>
Message-ID: <CF2AF67F-68EF-4AD5-9E83-7C53B054379A@jerris.com>

There are xml hooks in several of the embedded including mod_perl and  
mod_lua.  Not sure how well those scale as I have not seen anyone use  
them heavily.

Mike

On Oct 6, 2009, at 11:05 AM, Woody Dickson wrote:

> Hi,
>
> Is there anyway of using curl without having to setup a standalone  
> http service?  Is it possible to generate curl xml using scripts?
>
> woody
>
> On Mon, Oct 5, 2009 at 11:05 PM, Michael Jerris <mike at jerris.com>  
> wrote:
>
> On Oct 5, 2009, at 10:49 AM, Woody Dickson wrote:
>
> > Is is possible to override any of the setting specified in the
> > conference profile?
>
> Just the flags you can pass per user such as pin and mute
>
> >
> > What I want to do is to have a default profile, and be able to
> > modify certain fields if necessary in the dialplan.
> >
> >
> > Alternatively, I would prefer to have a dynamic profile setting for
> > the conference to obtain those parameters from odbc.
>
> you can do this with mod_xml_curl

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From brian at freeswitch.org  Tue Oct  6 07:31:06 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 6 Oct 2009 09:31:06 -0500
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <1b46b4e80910052142jaa09f08l72e5e4b2b24d10fc@mail.gmail.com>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<191c3a030909291122t770af037j5b1068197b31de87@mail.gmail.com>
	<1b46b4e80909291542k71440965g75bb3bfcf38e8005@mail.gmail.com>
	<1b46b4e80910011229k31f7233du10b0fc661730d3f3@mail.gmail.com>
	<F86F950D-9106-49C9-81CA-38B3EB2072F4@jerris.com>
	<1b46b4e80910041519h3581d3d2x7b84041e8c18588@mail.gmail.com>
	<C887D143-F7A1-44B9-B8A2-D34A4B4E582E@jerris.com>
	<1b46b4e80910041609x2f41ceb9u142cb484951a21ed@mail.gmail.com>
	<87f2f3b90910050920w4d17033fl4c386a0018107bdf@mail.gmail.com>
	<1b46b4e80910052119me12a0d9y4653edd831f9555e@mail.gmail.com>
	<1b46b4e80910052142jaa09f08l72e5e4b2b24d10fc@mail.gmail.com>
Message-ID: <083DBCB9-48B2-4E61-81B5-B7D00A42EAB5@freeswitch.org>

This looks like you have an ALG messing with packets... notice it says  
rport 5080 but we are sending to 5060.

/b

On Oct 5, 2009, at 11:42 PM, Nicolas Brenner wrote:

> Ignore my previous email, the traces were incomplete, got much  
> better (and complete) traces with ngrep (found a suggestion from  
> Brian in the list archive, thanks!)
>
> The gateway that registers:
>
> - http://pastebin.freeswitch.org/10607
>
> The one that doesn't:
>
> - http://pastebin.freeswitch.org/10608
>
>
> Thanks again for your time and help!
>
>
> Nicolas
>
>
> On Tue, Oct 6, 2009 at 12:19 AM, Nicolas Brenner <nicolas at medularis.com 
> > wrote:
> There was no sane way of doing that, so I ended up logging the trace  
> from the cli.

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From larclap at yahoo.com  Tue Oct  6 08:41:29 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Tue, 6 Oct 2009 08:41:29 -0700
Subject: [Freeswitch-users] Upgrading causes no answer
Message-ID: <00ea01ca469b$7929faf0$6b7df0d0$@com>

http://pastebin.freeswitch.org/10612

 

I having been running v14996 OK for a while. I have upgraded a couple of
times after, but every time, an inbound call is hung up on. The only thing
that has changed is the upgrade. This morning I upgraded to v15098 and the
problem persists.

 

I believe it has to do with a lua script I use for inbound calls. Reading
from the log, just after the script is launched, the following two lines
appear:

 

switch_cpp.cpp:1116  session not ready

switch_cpp.cpp:925 destroy/unlink session from object

 

Has something changed recently with lua processing? Is there something in
the lua script which is causing the problem?

 

I would appreciate any help.

 

Thanks, Lars

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From msc at freeswitch.org  Tue Oct  6 09:06:02 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 6 Oct 2009 09:06:02 -0700
Subject: [Freeswitch-users] oz debug says error
In-Reply-To: <7d79b3930910052106n59cf9cbema68e4d6ccc274034@mail.gmail.com>
References: <25749736.post@talk.nabble.com>
	<20091005124858.81857415806@mail.cune.org>
	<7d79b3930910052106n59cf9cbema68e4d6ccc274034@mail.gmail.com>
Message-ID: <87f2f3b90910060906y4b087d7coa91de38d78a3456b@mail.gmail.com>

Pastebin your openzap.conf file. Also, is this Sangoma or zaptel-based
hardware? If it's Sangoma, pastebin your wanpipe1.conf file. If zaptel,
please paste your zaptel.conf file.

-MC

On Mon, Oct 5, 2009 at 9:06 PM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Openzap.conf.xml
> <configuration name="openzap.conf" description="OpenZAP Configuration">
>         <settings>
>                 <param name="debug" value="7"/>
>         </settings>
>         <libpri_spans>
>                 <span name="PRI_1">
>                         <param name="node" value="cpe"/>
>                         <param name="switch" value="euroisdn"/>
>                         <param name="dialplan" value="XML"/>
>                         <param name="context" value="default"/>
>                 </span>
>         </libpri_spans>
> </configuration>
>
> Output of oz list in fs_cli
>
> span: 1 (PRI_1)
> type: isdn
> chan_count: 47
> dialplan: XML
> context: default
> dial_regex:
> fail_dial_regex:
> hold_music:
> analog_options none
>
>
> freeswitch.log
>
> http://pastebin.freeswitch.org/10604
>
>
>
>
>
> On Mon, Oct 5, 2009 at 6:18 PM, <Russell.Mosemann at cune.org> wrote:
>
>> lakshmanan <lakindia89 at gmail.com> said:
>> > Thanks for pointing that.
>> > I also tried that.
>> > But in that case, I'm not able to make a call through openzap.
>>
>> What is in openzap.conf.xml? If you start fs_cli and enter "oz list",
>> what does it show? Copy the ozmod lines from freeswitch.log to
>> pastebin.freeswitch.org and post the link here so that we can see what
>> openzap does when freeswitch starts.
>>
>> --
>> Russell Mosemann
>>
>>
>>
>> ________________________________________________________
>> Concordia University, Nebraska
>> See http://www.cune.edu/ for the latest news and events!
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Oct  6 09:15:42 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 6 Oct 2009 09:15:42 -0700
Subject: [Freeswitch-users] Wiki Info: Multi Homed Tutorial
In-Reply-To: <a87d43fd0910052120l76da1536ra92766174dc938d3@mail.gmail.com>
References: <87f2f3b90910051527x638f55b8hda75d5302267c251@mail.gmail.com>
	<a87d43fd0910052120l76da1536ra92766174dc938d3@mail.gmail.com>
Message-ID: <87f2f3b90910060915g167bafa7jd8b0ed5e1b3b2b20@mail.gmail.com>

On Mon, Oct 5, 2009 at 9:20 PM, Michael Gende <mgende at gendesign.com> wrote:

> Michael,
>
> Thanks for "wiki-fying" my text-only attempt at some user doc. I should
> have done that for you. I actually have an updated version with many
> corrections and the end tabs filled in. Can you point me to info on how I
> can amend and append what you have kindly put up?
>
> Mike G.
>
> First, go to wiki.freeswitch.org and create a wiki account. Then, go to
the Multi_home_tutorial page and click edit (top of the page). You'll see
that there is wiki markup to learn. If you have questions let me know. Just
edit the wiki text and click Show Preview to see what the real thing looks
like. Then click Save Page to save your changes.

Welcome to the world of MediaWiki and thanks for your help!
-MC
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From msc at freeswitch.org  Tue Oct  6 09:18:32 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 6 Oct 2009 09:18:32 -0700
Subject: [Freeswitch-users] add users dynamically to groups in
	default.xml
In-Reply-To: <f8af5740910060531j58158057h881d9862602d360c@mail.gmail.com>
References: <f8af5740910060531j58158057h881d9862602d360c@mail.gmail.com>
Message-ID: <87f2f3b90910060918q7d5d4d4fu454031120d098174@mail.gmail.com>

On Tue, Oct 6, 2009 at 5:31 AM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

> Hi,
> Can any one tell me how to add users dynamically to groups in default.xml,
> with out restart the freeswitch.
>
> Thanks
> Srinivasula Reddy K
>
>
Changes to the dialplan xml files get updated with a simple 'reloadxml'
command at the CLI. If you want truly dynamic configuration then you'll want
to investigate mod_xml_curl.
-MC
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From msc at freeswitch.org  Tue Oct  6 09:40:33 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 6 Oct 2009 09:40:33 -0700
Subject: [Freeswitch-users] Outgoing via openzap is not working
In-Reply-To: <7d79b3930910060330s77a2bbbv88d8014b73f91663@mail.gmail.com>
References: <7d79b3930910060330s77a2bbbv88d8014b73f91663@mail.gmail.com>
Message-ID: <87f2f3b90910060940k543bfd35rc7d6b2505b6d853@mail.gmail.com>

On Tue, Oct 6, 2009 at 3:30 AM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Hi I'm using freeswitch1.0.4. This post is moreover similar to my previous
> post.
> When I make an outgoing call, it is saying INVALID_IE_CONTENTS.
> Here are the details.
> openzap.conf.xml
>
> <configuration name="openzap.conf" description="OpenZAP Configuration">
> <settings>
> <param name="debug" value="7"/>
> </settings>
> <libpri_spans>
> <span name="PRI_1">
> <param name="node" value="cpe"/>
> <param name="switch" value="euroisdn"/>
> <param name="dialplan" value="XML"/>
> <param name="context" value="default"/>
> </span>
> </libpri_spans>
> </configuration>
>
> openzap.conf
> [span zt PRI_1]
> trunk_type => e1
> b-channel => 1:1-15
> d-channel=> 1:16
> b-channel => 1:17-31
> oz libpri debug 1 all
>
> API CALL [oz(libpri debug 1 all)] output:
> src/ozmod/ozmod_libpri/ozmod_libpri.c: +OK debug set.
>
> oz list
>
> API CALL [oz(list)] output:
> +OK
> span: 1 (PRI_1)
> type: isdn
> chan_count: 47
> dialplan: XML
> context: default
> dial_regex:
> fail_dial_regex:
> hold_music:
> analog_options none
>
> Freeswitch startup log:
> http://pastebin.freeswitch.org/10609
> After saying originate openzap/1/1/9952248266 openzap/1/1/9952248266
>

Can you confirm your originate line? What you type above is incorrect
syntax. Correct syntax:

openzap/1/a/99522448266 1234

Where the 'a' means select first available b chan and the 1234 is just an
extension number. You can put any number that works for your dialplan. The
originate command originates a call leg and then connects it to a second
call leg in the dialplan. Make sure that you are using this command properly
before continuing your debugging.
-MC



> http://pastebin.freeswitch.org/10610
>
> Please help me to solve this.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Oct  6 09:45:13 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 6 Oct 2009 09:45:13 -0700
Subject: [Freeswitch-users] overriding conference preference
In-Reply-To: <a44688a0910060805o49d549b8vbd4d545b229c75d@mail.gmail.com>
References: <a44688a0910050749s1ae1f9a4vba96477d159d377f@mail.gmail.com>
	<E773DB8B-AC74-49DE-9B38-7AFC441882F5@jerris.com>
	<a44688a0910060805o49d549b8vbd4d545b229c75d@mail.gmail.com>
Message-ID: <87f2f3b90910060945ge3f2a44ocf836b83b038736c@mail.gmail.com>

On Tue, Oct 6, 2009 at 8:05 AM, Woody Dickson <woodydickson at gmail.com>wrote:

> Hi,
>
> Is there anyway of using curl without having to setup a standalone http
> service?  Is it possible to generate curl xml using scripts?
>
> woody
>
>
Check out this page on the wiki:
http://wiki.freeswitch.org/wiki/Mod_xml_curl#Two_types_of_data_sources

Depending on your needs you might find an option that helps, such as using a
static file (which you can update in a 3rd party process).
-MC


> On Mon, Oct 5, 2009 at 11:05 PM, Michael Jerris <mike at jerris.com> wrote:
>
>>
>> On Oct 5, 2009, at 10:49 AM, Woody Dickson wrote:
>>
>> > Is is possible to override any of the setting specified in the
>> > conference profile?
>>
>> Just the flags you can pass per user such as pin and mute
>>
>> >
>> > What I want to do is to have a default profile, and be able to
>> > modify certain fields if necessary in the dialplan.
>> >
>> >
>> > Alternatively, I would prefer to have a dynamic profile setting for
>> > the conference to obtain those parameters from odbc.
>>
>> you can do this with mod_xml_curl
>>
>> Mike
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ChristianDamianidis at globalive.com  Tue Oct  6 09:55:05 2009
From: ChristianDamianidis at globalive.com (Christian Damianidis)
Date: Tue, 6 Oct 2009 12:55:05 -0400
Subject: [Freeswitch-users] mod_xml_curl http POST is inconsistent/bug
In-Reply-To: <191c3a030910060801x3ed48e98x6d69b53c6beaca69@mail.gmail.com>
References: <66EA3166EB339A4489B06286C0876A8A0C0C1432@mailserv.Globalive.local><276FEE28-70CF-422D-9F6F-7CF747FDDB22@freeswitch.org><66EA3166EB339A4489B06286C0876A8A0C0C1BA9@mailserv.Globalive.local>
	<191c3a030910060801x3ed48e98x6d69b53c6beaca69@mail.gmail.com>
Message-ID: <66EA3166EB339A4489B06286C0876A8A0C0C1F04@mailserv.Globalive.local>

I've tested this and making the change from ANY to BASIC worked. Thanks
for the help. 

It no longer sends the initial post without auth.

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Tuesday, October 06, 2009 11:02 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_xml_curl http POST is
inconsistent/bug

 

My guess is that we configure the curl to support the full range of http
auth methods.
Some of them like Digest require a challenge and realm etc so it's
probably asking without auth header because it cannot create one until
it gets that data.  In the case of Basic you can send the login and pass
right away but it does not know in advance that it will be basic.

Here is a snippet from the libcurl api docs:
------------------------------------------------------------------------
------------------------------------------------------------------------
-------------

Both these options allow you to set multiple types (by ORing them
together), to make libcurl pick the most secure one out of the types the
server/proxy claims to support. This method does however add a
round-trip since libcurl must first ask the server what it supports: 

 curl_easy_setopt(easyhandle, CURLOPT_HTTPAUTH,
CURLAUTH_DIGEST|CURLAUTH_BASIC); 

------------------------------------------------------------------------
------------------------------------------------------------------------
-------------

So my guess is that if we set it to only support basic, then it would
work how you expect so if you want to test it for me I can make it into
a parameter.

edit:
/usr/src/freeswitch.trunk/src/mod/xml_int/mod_xml_curl/mod_xml_curl.c
line 220
change

curl_easy_setopt(curl_handle, CURLOPT_HTTPAUTH, CURLAUTH_ANY);

to

curl_easy_setopt(curl_handle, CURLOPT_HTTPAUTH, CURLAUTH_BASIC);


If this works i'll think about exposing the auth methods so you can
choose them in the config.



On Tue, Oct 6, 2009 at 9:39 AM, Christian Damianidis
<ChristianDamianidis at globalive.com> wrote:

This web request goes to a server running IIS on Windows Server 2003.

 

From: Brian West [mailto:brian at freeswitch.org] 
Sent: Monday, October 05, 2009 5:43 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_xml_curl http POST is
inconsistent/bug

 

Are you using something other than apache?

 

/b

 

On Oct 5, 2009, at 1:25 PM, Christian Damianidis wrote:

 

 

The inconsistent POST request sent by the module causes freeswitch to
hang for 1-2 minutes during start-up.

 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:213-799-1400

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From tculjaga at gmail.com  Tue Oct  6 10:37:08 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 6 Oct 2009 19:37:08 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
Message-ID: <65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>

hello guys,


i was playing with mod_opal to see if i can make it working ... well it
seems SIP-H323 interworking is not tuned at all.

I have a call from a registered sip user (1001) to PSTN via mod_opal


<include>
  <extension name="EMERGENCY">
    <condition field="destination_number" expression="^0(112|9[23456])$">
      <action application="set"
data="effective_caller_id_number=1001282122"/>
      <action application="set" data="NCX_IP=10.4.4.254"/>
      <action application="set" data="call_timeout=30"/>
      <action application="set" data="hangup_after_bridge=true"/>

      <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
    </condition>
  </extension>

  <extension name="SPECIAL_SERVICES">
    <condition field="destination_number" expression="^0(9[01789]\d{3,4})$">
      <action application="set"
data="effective_caller_id_number=1001282122"/>
      <action application="set" data="NCX_IP=10.4.4.254"/>
      <action application="set" data="call_timeout=30"/>
      <action application="set" data="hangup_after_bridge=true"/>

      <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
    </condition>
  </extension>

  <extension name="ENYTHING_ELSE">
    <condition field="destination_number"
expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
      <action application="set"
data="effective_caller_id_number=1001282122"/>
      <action application="set" data="NCX_IP=10.4.4.254"/>
      <action application="set" data="call_timeout=30"/>
      <action application="set" data="hangup_after_bridge=true"/>

      <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
    </condition>
  </extension>
</include>



One of the many issues i sow is that FS connects the call on SIP leg before
it actually receives H.225 connect from H323 leg... as it is configured to
send 200 OK on the 1st H.225 message containing a FastStart element/OLC.


Attached is the tcpdump i took on FS machine... just use this filter: "h225
or h245 or q931 or sip"
Also, you can check the attac CDR,,,, this is an unanswered call i placed to
PSTN and FS billed it 23 seconds.



Can anyone tell where i can do correct SIP - H323 message mappings to avoid
this?



T.
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From dome at tel.co.th  Tue Oct  6 10:51:27 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Wed, 7 Oct 2009 00:51:27 +0700
Subject: [Freeswitch-users] Memory Question ?
Message-ID: <8ccbff060910061051p12b39c67kddf201113dda7c48@mail.gmail.com>

Dear All,
             why freeswitch use more memory after send and receuve
call. i attach htop capture screen.
you can compare to asterisk it use 0.7% for long time. but FS use 7.7%
(start from 1.2%)  after running about 4 hr.

Best Regards.

Dome C.
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From quentusrex at gmail.com  Tue Oct  6 10:57:40 2009
From: quentusrex at gmail.com (William King)
Date: Tue, 06 Oct 2009 10:57:40 -0700
Subject: [Freeswitch-users] Memory Question ?
In-Reply-To: <8ccbff060910061051p12b39c67kddf201113dda7c48@mail.gmail.com>
References: <8ccbff060910061051p12b39c67kddf201113dda7c48@mail.gmail.com>
Message-ID: <4ACB8514.2050201@gmail.com>

What operating system?

-William King

Dome Charoenyost wrote:
> Dear All,
>              why freeswitch use more memory after send and receuve
> call. i attach htop capture screen.
> you can compare to asterisk it use 0.7% for long time. but FS use 7.7%
> (start from 1.2%)  after running about 4 hr.
>
> Best Regards.
>
> Dome C.
>   
>
> ------------------------------------------------------------------------
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From brian at freeswitch.org  Tue Oct  6 11:07:05 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 6 Oct 2009 13:07:05 -0500
Subject: [Freeswitch-users] Memory Question ?
In-Reply-To: <8ccbff060910061051p12b39c67kddf201113dda7c48@mail.gmail.com>
References: <8ccbff060910061051p12b39c67kddf201113dda7c48@mail.gmail.com>
Message-ID: <DB335ACC-EEC9-41A0-83E4-8829E12FE6A8@freeswitch.org>

We use memory pools and its not uncommon to use what you displayed.

/b

On Oct 6, 2009, at 12:51 PM, Dome Charoenyost wrote:

> Dear All,
>             why freeswitch use more memory after send and receuve
> call. i attach htop capture screen.
> you can compare to asterisk it use 0.7% for long time. but FS use 7.7%
> (start from 1.2%)  after running about 4 hr.
>
> Best Regards.
>
> Dome C.
> <Screenshot-1.png>_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From msc at freeswitch.org  Tue Oct  6 11:12:02 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 6 Oct 2009 11:12:02 -0700
Subject: [Freeswitch-users] Memory Question ?
In-Reply-To: <8ccbff060910061051p12b39c67kddf201113dda7c48@mail.gmail.com>
References: <8ccbff060910061051p12b39c67kddf201113dda7c48@mail.gmail.com>
Message-ID: <87f2f3b90910061112h221d37b2p23d82342b9ac9a09@mail.gmail.com>

On Tue, Oct 6, 2009 at 10:51 AM, Dome Charoenyost <dome at tel.co.th> wrote:

> Dear All,
>             why freeswitch use more memory after send and receuve
> call. i attach htop capture screen.
> you can compare to asterisk it use 0.7% for long time. but FS use 7.7%
> (start from 1.2%)  after running about 4 hr.
>
> Best Regards.
>
> Dome C.
>
> Do you have lots of modules loaded? The default config has lots of stuff
loaded. If memory usage is a big issue then check your modules.conf.xml file
and don't load unnecessary modules. Of course, you could use Asterisk but
then you save memory at the expense of fewer features and more crashes...

-MC
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From dome at tel.co.th  Tue Oct  6 11:43:10 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Wed, 7 Oct 2009 01:43:10 +0700
Subject: [Freeswitch-users] Memory Question ?
In-Reply-To: <4ACB8514.2050201@gmail.com>
References: <8ccbff060910061051p12b39c67kddf201113dda7c48@mail.gmail.com>
	<4ACB8514.2050201@gmail.com>
Message-ID: <8ccbff060910061143gc0d0edcl46a2b5274a74c876@mail.gmail.com>

Debian Squeeze  i386 32bit
And Debian Lenny are same

Dome C.

2009/10/7 William King <quentusrex at gmail.com>:
> What operating system?
>
> -William King
>
> Dome Charoenyost wrote:
>> Dear All,
>> ? ? ? ? ? ? ?why freeswitch use more memory after send and receuve
>> call. i attach htop capture screen.
>> you can compare to asterisk it use 0.7% for long time. but FS use 7.7%
>> (start from 1.2%) ?after running about 4 hr.
>>
>> Best Regards.
>>
>> Dome C.
>>
>>
>> ------------------------------------------------------------------------
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From dome at tel.co.th  Tue Oct  6 11:50:25 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Wed, 7 Oct 2009 01:50:25 +0700
Subject: [Freeswitch-users] Memory Question ?
In-Reply-To: <87f2f3b90910061112h221d37b2p23d82342b9ac9a09@mail.gmail.com>
References: <8ccbff060910061051p12b39c67kddf201113dda7c48@mail.gmail.com>
	<87f2f3b90910061112h221d37b2p23d82342b9ac9a09@mail.gmail.com>
Message-ID: <8ccbff060910061150h2ca1c495u339e2ee046b8c21a@mail.gmail.com>

2009/10/7 Michael Collins <msc at freeswitch.org>:
>
>
> On Tue, Oct 6, 2009 at 10:51 AM, Dome Charoenyost <dome at tel.co.th> wrote:
>>
>> Dear All,
>> ? ? ? ? ? ? why freeswitch use more memory after send and receuve
>> call. i attach htop capture screen.
>> you can compare to asterisk it use 0.7% for long time. but FS use 7.7%
>> (start from 1.2%) ?after running about 4 hr.
>>
>> Best Regards.
>>
>> Dome C.
>>
> Do you have lots of modules loaded? The default config has lots of stuff
I use LCR , nibblebill , odbcquery.
> loaded. If memory usage is a big issue then check your modules.conf.xml file
My question is why FS need more memory when start about 4 hr. is is
normal or got problem ?
So what's happen when running about 1 weeks ?
some time my FS crash and i report bug in jira. and bug maintainer ask
me is memory run out ?
i think yes.

> and don't load unnecessary modules. Of course, you could use Asterisk but
> then you save memory at the expense of fewer features and more crashes...
No. i move all from asterisk to FS right now :)

>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From quentusrex at gmail.com  Tue Oct  6 11:50:49 2009
From: quentusrex at gmail.com (William King)
Date: Tue, 06 Oct 2009 11:50:49 -0700
Subject: [Freeswitch-users] Memory Question ?
In-Reply-To: <8ccbff060910061143gc0d0edcl46a2b5274a74c876@mail.gmail.com>
References: <8ccbff060910061051p12b39c67kddf201113dda7c48@mail.gmail.com>	<4ACB8514.2050201@gmail.com>
	<8ccbff060910061143gc0d0edcl46a2b5274a74c876@mail.gmail.com>
Message-ID: <4ACB9189.5070504@gmail.com>

Which modules do you have loaded?

-William King

Dome Charoenyost wrote:
> Debian Squeeze  i386 32bit
> And Debian Lenny are same
>
> Dome C.
>
> 2009/10/7 William King <quentusrex at gmail.com>:
>   
>> What operating system?
>>
>> -William King
>>
>> Dome Charoenyost wrote:
>>     
>>> Dear All,
>>>              why freeswitch use more memory after send and receuve
>>> call. i attach htop capture screen.
>>> you can compare to asterisk it use 0.7% for long time. but FS use 7.7%
>>> (start from 1.2%)  after running about 4 hr.
>>>
>>> Best Regards.
>>>
>>> Dome C.
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From dome at tel.co.th  Tue Oct  6 11:57:49 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Wed, 7 Oct 2009 01:57:49 +0700
Subject: [Freeswitch-users] Memory Question ?
In-Reply-To: <4ACB9189.5070504@gmail.com>
References: <8ccbff060910061051p12b39c67kddf201113dda7c48@mail.gmail.com>
	<4ACB8514.2050201@gmail.com>
	<8ccbff060910061143gc0d0edcl46a2b5274a74c876@mail.gmail.com>
	<4ACB9189.5070504@gmail.com>
Message-ID: <8ccbff060910061157g2220a8fay1585593303bf593e@mail.gmail.com>

2009/10/7 William King <quentusrex at gmail.com>:
> Which modules do you have loaded?
default config and nibllebill , lcr , odbcquery

>
> -William King
>
> Dome Charoenyost wrote:
>> Debian Squeeze ?i386 32bit
>> And Debian Lenny are same
>>
>> Dome C.
>>
>> 2009/10/7 William King <quentusrex at gmail.com>:
>>
>>> What operating system?
>>>
>>> -William King
>>>
>>> Dome Charoenyost wrote:
>>>
>>>> Dear All,
>>>> ? ? ? ? ? ? ?why freeswitch use more memory after send and receuve
>>>> call. i attach htop capture screen.
>>>> you can compare to asterisk it use 0.7% for long time. but FS use 7.7%
>>>> (start from 1.2%) ?after running about 4 hr.
>>>>
>>>> Best Regards.
>>>>
>>>> Dome C.
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From xengelpublicx at gmail.com  Tue Oct  6 12:06:01 2009
From: xengelpublicx at gmail.com (Vladimir Elizarov)
Date: Tue, 06 Oct 2009 23:06:01 +0400
Subject: [Freeswitch-users] linksys spa-3102 fax t38 + freeswitch media proxy
Message-ID: <4ACB9519.9000805@gmail.com>

Hello.

I want to set up faxing via the gateway linksys spa-3102 (with support
for t38) via SIP.

SIP-client -> linksys spa3102 -> fs -> provider

    <gateway name="callwithus">
        ....
        <param name="inbound-late-negotiation" value="true"/>
        <param name="stun-enabled" value="true"/>
    </gateway>

<extension name="Fax">
    <condition field="destination_number" expression="^(30)(\d{11})$">
            <action application="set" data="proxy_media=true"/>
            <action application="set" data="bypass_media=false"/>
            <action application="bridge"
data="sofia/gateway/callwithus/*4087$2 at sip.callwithus.com"/>
        </condition>
</extension>

I make a call, press start on the fax. Fax not sent. Get this in the log
(UA freeswitch not trunk-12805):

   ------------------------------------------------------------------------
   INVITE sip:*408774957558438 at sip.callwithus.com SIP/2.0
   Via: SIP/2.0/UDP 77.239.230.202:5080;rport;branch=z9hG4bK8yjrHeUprX2Hr
   Max-Forwards: 99
   From: "Mihailova Ludmila"
<sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
   To: <sip:*408774957558438 at sip.callwithus.com>
   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
   CSeq: 121302430 INVITE
   Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 250
   Remote-Party-ID: "Mihailova Ludmila"
<sip:214 at sip.callwithus.com>;party=calling;screen=yes;privacy=off
   
   v=0
   o=- 216896551 216896551 IN IP4 192.168.50.51
   s=-
   c=IN IP4 77.239.230.202
   t=0 0
   m=audio 32404 RTP/AVP 0 100 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:100 NSE/8000
   a=fmtp:100 192-193
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=ptime:30

   ------------------------------------------------------------------------
send 1333 bytes to udp/[38.99.70.232]:5060 at 15:05:34.485226:
   ------------------------------------------------------------------------
   INVITE sip:*408774957558438 at sip.callwithus.com SIP/2.0
   Via: SIP/2.0/UDP 77.239.230.202:5080;rport;branch=z9hG4bK97BHK9BtN6r4K
   Max-Forwards: 99
   From: "Mihailova Ludmila"
<sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
   To: <sip:*408774957558438 at sip.callwithus.com>
   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
   CSeq: 121302431 INVITE
   Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
   Expires: 3600
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 250
   Remote-Party-ID: "Mihailova Ludmila"
<sip:214 at sip.callwithus.com>;party=calling;screen=yes;privacy=off
   
   v=0
   o=- 216896551 216896551 IN IP4 192.168.50.51
   s=-
   c=IN IP4 77.239.230.202
   t=0 0
   m=audio 32404 RTP/AVP 0 100 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:100 NSE/8000
   a=fmtp:100 192-193
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=ptime:30
   ------------------------------------------------------------------------
   SIP/2.0 100 Giving a try
   Via: SIP/2.0/UDP
192.168.50.10:5080;rport=5080;branch=z9hG4bK97BHK9BtN6r4K
   From: "Mihailova Ludmila"
<sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
   To: <sip:*408774957558438 at sip.callwithus.com>
   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
   CSeq: 121302431 INVITE
   Server: CWU SIP GW
   Content-Length: 0
   
   ------------------------------------------------------------------------
recv 836 bytes from udp/[38.99.70.232]:5060 at 15:05:49.641312:
   ------------------------------------------------------------------------
   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP
192.168.50.10:5080;rport=5080;branch=z9hG4bK97BHK9BtN6r4K
   Record-Route:
<sip:38.99.70.232;lr=on;ftag=HXjrDKpHHX8ZN;did=e8e.089fc2b6>
   From: "Mihailova Ludmila"
<sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
   To: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
   CSeq: 121302431 INVITE
   User-Agent: CWU SIP-GW
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
   Supported: replaces
   Contact: <sip:*408774957558438 at 204.74.213.1:5062>
   Content-Type: application/sdp
   Content-Length: 209
   
   v=0
   o=root 31797 31797 IN IP4 4.55.17.66
   s=session
   c=IN IP4 4.55.17.66
   t=0 0
   m=audio 14104 RTP/AVP 0 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   a=sendrecv
   ------------------------------------------------------------------------
v=0
o=root 31797 31797 IN IP4 4.55.17.66
s=session
c=IN IP4 4.55.17.66
t=0 0
m=audio 14104 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

2009-10-06 19:03:31.112988 [NOTICE] sofia.c:3449 Pre-Answer
sofia/external/*408774957558438 at sip.callwithus.com!
2009-10-06 19:03:31.112988 [DEBUG] switch_channel.c:1822 Send signal
sofia/internal/214 at pbx0.tssec.lan [BREAK]
2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2095 Set Codec
sofia/external/*408774957558438 at sip.callwithus.com PROXY/8000 20 ms 160
samples
2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2372 PROXY AUDIO RTP
[sofia/external/*408774957558438 at sip.callwithus.com]
192.168.50.10:32404->4.55.17.66:14104 codec: 0 ms: 20
2009-10-06 19:03:31.112988 [DEBUG] switch_rtp.c:1163 Not using a timer
2009-10-06 19:03:31.112988 [DEBUG] switch_ivr_originate.c:2154
sofia/internal/214 at pbx0.tssec.lan receive message [PROGRESS]
2009-10-06 19:03:31.112988 [INFO] switch_ivr_originate.c:2154 Sending
early media
2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:1322
sofia/internal/214 at pbx0.tssec.lan Patched SDP
---
v=0
o=root 31797 31797 IN IP4 4.55.17.66
s=session
c=IN IP4 4.55.17.66
t=0 0
m=audio 14104 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

+++
v=0
o=root 31797 31797 IN IP4 4.55.17.66
s=session
c=IN IP4 192.168.50.10
t=0 0
m=audio 32698 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2372 PROXY AUDIO RTP
[sofia/internal/214 at pbx0.tssec.lan]
192.168.50.10:32698->192.168.50.51:16474 codec: 0 ms: 30
2009-10-06 19:03:31.112988 [DEBUG] switch_rtp.c:1163 Not using a timer
2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2508 Set comfort noise
payload to 13
2009-10-06 19:03:31.112988 [DEBUG] sofia.c:3462
sofia/internal/214 at pbx0.tssec.lan receive message [PROGRESS]
2009-10-06 19:03:31.116989 [NOTICE] mod_sofia.c:1521 Pre-Answer
sofia/internal/214 at pbx0.tssec.lan!
2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal
sofia/internal/214 at pbx0.tssec.lan [BREAK]
2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_originate.c:2196 Originate
Resulted in Success: [sofia/external/*408774957558438 at sip.callwithus.com]
2009-10-06 19:03:31.116989 [DEBUG] switch_channel.c:182
sofia/external/*408774957558438 at sip.callwithus.com receive message
[AUDIO_SYNC]
2009-10-06 19:03:31.116989 [DEBUG] switch_channel.c:182
sofia/internal/214 at pbx0.tssec.lan receive message [AUDIO_SYNC]
2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_bridge.c:911
sofia/external/*408774957558438 at sip.callwithus.com receive message [BRIDGE]
2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal
sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_bridge.c:918
sofia/internal/214 at pbx0.tssec.lan receive message [BRIDGE]
2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal
sofia/internal/214 at pbx0.tssec.lan [BREAK]
2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_bridge.c:962
(sofia/external/*408774957558438 at sip.callwithus.com) State Change
CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2009-10-06 19:03:31.116989 [DEBUG] sofia.c:3359 Channel
sofia/internal/214 at pbx0.tssec.lan entering state [early][183]
2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal
sofia/internal/214 at pbx0.tssec.lan [BREAK]
2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
2009-10-06 19:03:31.116989 [DEBUG] switch_core_state_machine.c:306
(sofia/external/*408774957558438 at sip.callwithus.com) Running State
Change CS_EXCHANGE_MEDIA
2009-10-06 19:03:31.116989 [DEBUG] switch_core_state_machine.c:343
(sofia/external/*408774957558438 at sip.callwithus.com) State EXCHANGE_MEDIA
2009-10-06 19:03:31.116989 [DEBUG] mod_sofia.c:430 SOFIA LOOPBACK
recv 822 bytes from udp/[38.99.70.232]:5060 at 15:05:58.503701:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP
192.168.50.10:5080;rport=5080;branch=z9hG4bK97BHK9BtN6r4K
   Record-Route:
<sip:38.99.70.232;lr=on;ftag=HXjrDKpHHX8ZN;did=e8e.089fc2b6>
   From: "Mihailova Ludmila"
<sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
   To: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
   CSeq: 121302431 INVITE
   User-Agent: CWU SIP-GW
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
   Supported: replaces
   Contact: <sip:*408774957558438 at 204.74.213.1:5062>
   Content-Type: application/sdp
   Content-Length: 209
   
   v=0
   o=root 31797 31798 IN IP4 4.55.17.66
   s=session
   c=IN IP4 4.55.17.66
   t=0 0
   m=audio 14104 RTP/AVP 0 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   a=sendrecv
   ------------------------------------------------------------------------
send 762 bytes to udp/[38.99.70.232]:5060 at 15:05:58.507702:
   ------------------------------------------------------------------------
   ACK sip:*408774957558438 at 204.74.213.1:5062 SIP/2.0
   Via: SIP/2.0/UDP 192.168.50.10:5080;rport;branch=z9hG4bKaH59m4vXjFFQF
   Route: <sip:38.99.70.232;lr=on;ftag=HXjrDKpHHX8ZN;did=e8e.089fc2b6>
   Max-Forwards: 70
   From: "Mihailova Ludmila"
<sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
   To: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
   CSeq: 121302431 ACK
   Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
   Content-Length: 0
   
   ------------------------------------------------------------------------
2009-10-06 19:03:39.975377 [DEBUG] sofia.c:3359 Channel
sofia/external/*408774957558438 at sip.callwithus.com entering state
[ready][200]
2009-10-06 19:03:39.975377 [DEBUG] switch_channel.c:1935 Send signal
sofia/internal/214 at pbx0.tssec.lan [BREAK]
2009-10-06 19:03:39.975377 [NOTICE] sofia.c:3814 Channel
[sofia/external/*408774957558438 at sip.callwithus.com] has been answered
2009-10-06 19:03:39.975377 [DEBUG] switch_channel.c:182
sofia/external/*408774957558438 at sip.callwithus.com receive message
[AUDIO_SYNC]
2009-10-06 19:03:39.975377 [DEBUG] sofia.c:3819
sofia/internal/214 at pbx0.tssec.lan receive message [ANSWER]
2009-10-06 19:03:39.975377 [DEBUG] sofia_glue.c:1322
sofia/internal/214 at pbx0.tssec.lan Patched SDP
---
v=0
o=root 31797 31797 IN IP4 4.55.17.66
s=session
c=IN IP4 4.55.17.66
t=0 0
m=audio 14104 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

+++
v=0
o=root 31797 31797 IN IP4 4.55.17.66
s=session
c=IN IP4 192.168.50.10
t=0 0
m=audio 32698 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

2009-10-06 19:03:39.975377 [DEBUG] switch_core_session.c:630 Send signal
sofia/internal/214 at pbx0.tssec.lan [BREAK]
2009-10-06 19:03:39.975377 [NOTICE] sofia.c:3819 Channel
[sofia/internal/214 at pbx0.tssec.lan] has been answered
2009-10-06 19:03:39.975377 [DEBUG] switch_channel.c:182
sofia/internal/214 at pbx0.tssec.lan receive message [AUDIO_SYNC]
2009-10-06 19:03:39.975377 [DEBUG] sofia.c:3359 Channel
sofia/internal/214 at pbx0.tssec.lan entering state [completed][200]
2009-10-06 19:03:39.991381 [DEBUG] sofia.c:3359 Channel
sofia/internal/214 at pbx0.tssec.lan entering state [ready][200]
recv 1054 bytes from udp/[38.99.70.232]:5060 at 15:06:05.577607:
   ------------------------------------------------------------------------
   INVITE sip:gw+callwithus at 192.168.50.10:5080;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.0
   Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK41d48157;rport=5062
   From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
   To: "Mihailova Ludmila"
<sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
   Contact: <sip:*408774957558438 at 204.74.213.1:5062>
   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
   CSeq: 102 INVITE
   User-Agent: CWU SIP-GW
   Max-Forwards: 69
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
   Supported: replaces
   X-asterisk-info: SIP re-invite (T38 switchover)
   Content-Type: application/sdp
   Content-Length: 350
   
   v=0
   o=root 31797 31799 IN IP4 32.50.63.192
   s=session
   c=IN IP4 32.50.63.192
   t=0 0
   m=image 14104 udptl t38
   a=T38FaxVersion:0
   a=T38MaxBitRate:9600
   a=T38FaxFillBitRemoval:0
   a=T38FaxTranscodingMMR:0
   a=T38FaxTranscodingJBIG:0
   a=T38FaxRateManagement:transferredTCF
   a=T38FaxMaxBuffer:176
   a=T38FaxMaxDatagram:176
   a=T38FaxUdpEC:t38UDPRedundancy
   ------------------------------------------------------------------------
send 443 bytes to udp/[38.99.70.232]:5060 at 15:06:05.577607:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.0
   Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK41d48157;rport=5062
   From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
   To: "Mihailova Ludmila"
<sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
   CSeq: 102 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
   Content-Length: 0
   
   ------------------------------------------------------------------------
2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3359 Channel
sofia/external/*408774957558438 at sip.callwithus.com entering state
[received][100]
2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3366 Remote SDP:
v=0
o=root 31797 31799 IN IP4 32.50.63.192
s=session
c=IN IP4 32.50.63.192
t=0 0
m=image 14104 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:176
a=T38FaxMaxDatagram:176
a=T38FaxUdpEC:t38UDPRedundancy

2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3683 Passing SDP to other leg.
v=0
o=root 31797 31799 IN IP4 32.50.63.192
s=session
c=IN IP4 32.50.63.192
t=0 0
m=image 14104 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:176
a=T38FaxMaxDatagram:176
a=T38FaxUdpEC:t38UDPRedundancy

2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3696
sofia/internal/214 at pbx0.tssec.lan receive message [MEDIA_REDIRECT]
2009-10-06 19:03:47.049284 [DEBUG] mod_sofia.c:1195
sofia/internal/214 at pbx0.tssec.lan Sending media re-direct:
v=0
o=root 31797 31799 IN IP4 32.50.63.192
s=session
c=IN IP4 32.50.63.192
t=0 0
m=image 14104 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:176
a=T38FaxMaxDatagram:176
a=T38FaxUdpEC:t38UDPRedundancy

2009-10-06 19:03:47.049284 [DEBUG] sofia_glue.c:1101 Remote address:port
[192.168.50.51:16474] has not changed.
2009-10-06 19:03:47.049284 [DEBUG] sofia_glue.c:1322
sofia/internal/214 at pbx0.tssec.lan Patched SDP
---
v=0
o=root 31797 31799 IN IP4 32.50.63.192
s=session
c=IN IP4 32.50.63.192
t=0 0
m=image 14104 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:176
a=T38FaxMaxDatagram:176
a=T38FaxUdpEC:t38UDPRedundancy

+++
v=0
o=root 31797 31799 IN IP4 32.50.63.192
s=session
c=IN IP4 192.168.50.10
t=0 0
m=image 32698 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:176
a=T38FaxMaxDatagram:176
a=T38FaxUdpEC:t38UDPRedundancy

2009-10-06 19:03:47.049284 [DEBUG] switch_core_session.c:630 Send signal
sofia/internal/214 at pbx0.tssec.lan [BREAK]
2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3359 Channel
sofia/internal/214 at pbx0.tssec.lan entering state [calling][0]
2009-10-06 19:03:47.077291 [DEBUG] sofia_glue.c:1101 Remote address:port
[192.168.50.51:16474] has not changed.
2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3158 Passing 200 OK to other leg
2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3170
sofia/external/*408774957558438 at sip.callwithus.com receive message [RESPOND]
2009-10-06 19:03:47.077291 [DEBUG] mod_sofia.c:1427 Responding with 200 [OK]
2009-10-06 19:03:47.077291 [DEBUG] sofia_glue.c:1322
sofia/external/*408774957558438 at sip.callwithus.com Patched SDP
---
v=0
o=- 216899808 216899808 IN IP4 192.168.50.51
s=-
c=IN IP4 192.168.50.51
t=0 0
m=image 16474 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:200
a=T38FaxUdpEC:t38UDPRedundancy

+++
v=0
o=- 216899808 216899808 IN IP4 192.168.50.51
s=-
c=IN IP4 77.239.230.202
t=0 0
m=image 32404 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:200
a=T38FaxUdpEC:t38UDPRedundancy

2009-10-06 19:03:47.077291 [DEBUG] sofia_glue.c:1110 AUDIO RTP CHANGING
DEST TO: [32.50.63.192:14104]
2009-10-06 19:03:47.077291 [DEBUG] switch_core_session.c:630 Send signal
sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3359 Channel
sofia/internal/214 at pbx0.tssec.lan entering state [ready][200]
2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3366 Remote SDP:
v=0
o=- 216899808 216899808 IN IP4 192.168.50.51
s=-
c=IN IP4 192.168.50.51
t=0 0
m=image 16474 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:200
a=T38FaxUdpEC:t38UDPRedundancy

send 1003 bytes to udp/[38.99.70.232]:5060 at 15:06:05.605615:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.0
   Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK41d48157;rport=5062
   From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
   To: "Mihailova Ludmila"
<sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
   CSeq: 102 INVITE
   Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 274
   
   v=0
   o=- 216896551 216896552 IN IP4 192.168.50.51
   s=-
   c=IN IP4 77.239.230.202
   t=0 0
   m=image 32404 udptl t38
   a=T38FaxVersion:0
   a=T38MaxBitRate:14400
   a=T38FaxRateManagement:transferredTCF
   a=T38FaxMaxBuffer:200
   a=T38FaxMaxDatagram:200
   a=T38FaxUdpEC:t38UDPRedundancy
   ------------------------------------------------------------------------
2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3359 Channel
sofia/external/*408774957558438 at sip.callwithus.com entering state
[completed][200]
2009-10-06 19:03:47.285347 [INFO] switch_rtp.c:1905 Auto Changing port
from 32.50.63.192:14104 to 4.55.17.66:14104
recv 527 bytes from udp/[38.99.70.232]:5060 at 15:06:05.817672:
   ------------------------------------------------------------------------
   ACK sip:gw+callwithus at 192.168.50.10:5080;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.2
   Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK5ffd72cd;rport=5062
   From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
   To: "Mihailova Ludmila"
<sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
   Contact: <sip:*408774957558438 at 204.74.213.1:5062>
   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
   CSeq: 102 ACK
   User-Agent: CWU SIP-GW
   Max-Forwards: 69
   Content-Length: 0
   
   ------------------------------------------------------------------------
2009-10-06 19:03:47.289349 [DEBUG] sofia.c:3359 Channel
sofia/external/*408774957558438 at sip.callwithus.com entering state
[ready][200]
recv 476 bytes from udp/[38.99.70.232]:5060 at 15:06:47.588933:
   ------------------------------------------------------------------------
   BYE sip:gw+callwithus at 192.168.50.10:5080;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKd406.f8553e14.0
   Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK0f2cff01;rport=5062
   From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
   To: "Mihailova Ludmila"
<sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
   CSeq: 103 BYE
   User-Agent: CWU SIP-GW
   Max-Forwards: 69
   Content-Length: 0
   
   ------------------------------------------------------------------------
2009-10-06 19:04:29.060609 [NOTICE] sofia.c:328 Hangup
sofia/external/*408774957558438 at sip.callwithus.com [CS_EXCHANGE_MEDIA]
[NORMAL_CLEARING]
2009-10-06 19:04:29.060609 [DEBUG] switch_channel.c:1726 Send signal
sofia/external/*408774957558438 at sip.callwithus.com [KILL]
2009-10-06 19:04:29.060609 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
2009-10-06 19:04:29.060609 [DEBUG] switch_core_state_machine.c:437
thread mismatch skipping state handler.
send 574 bytes to udp/[38.99.70.232]:5060 at 15:06:47.588933:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKd406.f8553e14.0
   Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK0f2cff01;rport=5062
   From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
   To: "Mihailova Ludmila"
<sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
   CSeq: 103 BYE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Content-Length: 0
   
   ------------------------------------------------------------------------


-------------- next part --------------
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From quentusrex at gmail.com  Tue Oct  6 12:14:09 2009
From: quentusrex at gmail.com (William King)
Date: Tue, 06 Oct 2009 12:14:09 -0700
Subject: [Freeswitch-users] Memory Question ?
In-Reply-To: <8ccbff060910061157g2220a8fay1585593303bf593e@mail.gmail.com>
References: <8ccbff060910061051p12b39c67kddf201113dda7c48@mail.gmail.com>	<4ACB8514.2050201@gmail.com>	<8ccbff060910061143gc0d0edcl46a2b5274a74c876@mail.gmail.com>	<4ACB9189.5070504@gmail.com>
	<8ccbff060910061157g2220a8fay1585593303bf593e@mail.gmail.com>
Message-ID: <4ACB9701.2010707@gmail.com>

It will use enough ram to load all of your modules, and applications.
That is the initial ram usage on startup.

So how much ram is FS using on startup? less than 250-300 MB of ram on
initial load with all the modules and applications isn't unreasonable. I
think by default it loads tons of stuff you probably wouldn't use in
production.

Next up is how much ram is used per call. This should hardly be anything.

And last up, if you have freeswitch running for days and you see the
memory usage grow from ~150MB to ~800MB, even if it takes weeks. Then
there might be an issue.

If you want to start finding where the RAM usage is located, then unload
some modules and see if it does it still. Also, do you have any custom
scripts in the dialplan? they might not be ending properly. Thus they
stick around when you didn't intend them to.

-William King

Dome Charoenyost wrote:
> 2009/10/7 William King <quentusrex at gmail.com>:
>   
>> Which modules do you have loaded?
>>     
> default config and nibllebill , lcr , odbcquery
>
>   
>> -William King
>>
>> Dome Charoenyost wrote:
>>     
>>> Debian Squeeze  i386 32bit
>>> And Debian Lenny are same
>>>
>>> Dome C.
>>>
>>> 2009/10/7 William King <quentusrex at gmail.com>:
>>>
>>>       
>>>> What operating system?
>>>>
>>>> -William King
>>>>
>>>> Dome Charoenyost wrote:
>>>>
>>>>         
>>>>> Dear All,
>>>>>              why freeswitch use more memory after send and receuve
>>>>> call. i attach htop capture screen.
>>>>> you can compare to asterisk it use 0.7% for long time. but FS use 7.7%
>>>>> (start from 1.2%)  after running about 4 hr.
>>>>>
>>>>> Best Regards.
>>>>>
>>>>> Dome C.
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>           
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>         
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From anthony.minessale at gmail.com  Tue Oct  6 12:25:03 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 6 Oct 2009 14:25:03 -0500
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
Message-ID: <191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>

pcap is not as useful as FS console log on debug with:
sofia profile internal siptrace on

you should be reporting issues to jira under mod_opal not to the mailing
list.
http://jira.freeswitch.org

FYI
There is little financial support from the community for h323 which prevents
the mod_opal from getting much attention.
We actually have to contract the author of opal to help with these issues
including the original writing of the module that he did with very little
funding and nobody ever wants to pay him to improve it.

That does not mean your issue will not be addressed but there is no promise
how fast it will be.



On Tue, Oct 6, 2009 at 12:37 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> hello guys,
>
>
> i was playing with mod_opal to see if i can make it working ... well it
> seems SIP-H323 interworking is not tuned at all.
>
> I have a call from a registered sip user (1001) to PSTN via mod_opal
>
>
> <include>
>   <extension name="EMERGENCY">
>     <condition field="destination_number" expression="^0(112|9[23456])$">
>       <action application="set"
> data="effective_caller_id_number=1001282122"/>
>       <action application="set" data="NCX_IP=10.4.4.254"/>
>       <action application="set" data="call_timeout=30"/>
>       <action application="set" data="hangup_after_bridge=true"/>
>
>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>     </condition>
>   </extension>
>
>   <extension name="SPECIAL_SERVICES">
>     <condition field="destination_number"
> expression="^0(9[01789]\d{3,4})$">
>       <action application="set"
> data="effective_caller_id_number=1001282122"/>
>       <action application="set" data="NCX_IP=10.4.4.254"/>
>       <action application="set" data="call_timeout=30"/>
>       <action application="set" data="hangup_after_bridge=true"/>
>
>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>     </condition>
>   </extension>
>
>   <extension name="ENYTHING_ELSE">
>     <condition field="destination_number"
> expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
>       <action application="set"
> data="effective_caller_id_number=1001282122"/>
>       <action application="set" data="NCX_IP=10.4.4.254"/>
>       <action application="set" data="call_timeout=30"/>
>       <action application="set" data="hangup_after_bridge=true"/>
>
>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>     </condition>
>   </extension>
> </include>
>
>
>
> One of the many issues i sow is that FS connects the call on SIP leg before
> it actually receives H.225 connect from H323 leg... as it is configured to
> send 200 OK on the 1st H.225 message containing a FastStart element/OLC.
>
>
> Attached is the tcpdump i took on FS machine... just use this filter: "h225
> or h245 or q931 or sip"
> Also, you can check the attac CDR,,,, this is an unanswered call i placed
> to PSTN and FS billed it 23 seconds.
>
>
>
> Can anyone tell where i can do correct SIP - H323 message mappings to avoid
> this?
>
>
>
> T.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From tculjaga at gmail.com  Tue Oct  6 13:47:19 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 6 Oct 2009 22:47:19 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
Message-ID: <65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>

hi Anthony,

it is somewhere here:

         switch_status_t
FSConnection::receive_message(switch_core_session_message_t *msg)


anyhow, i will open an issue jira of course.


I understand your financial point of view, but anyhow while the entire world
is wants sip and trying to move to sip, the reality is quite different. The
majority of voice traffic exchanged via IP is still H323. This means a
working SIP - H323 interworking is really needed... pity nobody wants/has
time to work in this direction to produce a decent mod_h323.



T.





On Tue, Oct 6, 2009 at 9:25 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> pcap is not as useful as FS console log on debug with:
> sofia profile internal siptrace on
>
> you should be reporting issues to jira under mod_opal not to the mailing
> list.
> http://jira.freeswitch.org
>
> FYI
> There is little financial support from the community for h323 which
> prevents the mod_opal from getting much attention.
> We actually have to contract the author of opal to help with these issues
> including the original writing of the module that he did with very little
> funding and nobody ever wants to pay him to improve it.
>
> That does not mean your issue will not be addressed but there is no promise
> how fast it will be.
>
>
>
> On Tue, Oct 6, 2009 at 12:37 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> hello guys,
>>
>>
>> i was playing with mod_opal to see if i can make it working ... well it
>> seems SIP-H323 interworking is not tuned at all.
>>
>> I have a call from a registered sip user (1001) to PSTN via mod_opal
>>
>>
>> <include>
>>   <extension name="EMERGENCY">
>>     <condition field="destination_number" expression="^0(112|9[23456])$">
>>       <action application="set"
>> data="effective_caller_id_number=1001282122"/>
>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>       <action application="set" data="call_timeout=30"/>
>>       <action application="set" data="hangup_after_bridge=true"/>
>>
>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>     </condition>
>>   </extension>
>>
>>   <extension name="SPECIAL_SERVICES">
>>     <condition field="destination_number"
>> expression="^0(9[01789]\d{3,4})$">
>>       <action application="set"
>> data="effective_caller_id_number=1001282122"/>
>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>       <action application="set" data="call_timeout=30"/>
>>       <action application="set" data="hangup_after_bridge=true"/>
>>
>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>     </condition>
>>   </extension>
>>
>>   <extension name="ENYTHING_ELSE">
>>     <condition field="destination_number"
>> expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
>>       <action application="set"
>> data="effective_caller_id_number=1001282122"/>
>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>       <action application="set" data="call_timeout=30"/>
>>       <action application="set" data="hangup_after_bridge=true"/>
>>
>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>     </condition>
>>   </extension>
>> </include>
>>
>>
>>
>> One of the many issues i sow is that FS connects the call on SIP leg
>> before it actually receives H.225 connect from H323 leg... as it is
>> configured to send 200 OK on the 1st H.225 message containing a FastStart
>> element/OLC.
>>
>>
>> Attached is the tcpdump i took on FS machine... just use this filter:
>> "h225 or h245 or q931 or sip"
>> Also, you can check the attac CDR,,,, this is an unanswered call i placed
>> to PSTN and FS billed it 23 seconds.
>>
>>
>>
>> Can anyone tell where i can do correct SIP - H323 message mappings to
>> avoid this?
>>
>>
>>
>> T.
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From nicolas at medularis.com  Tue Oct  6 14:22:31 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Tue, 6 Oct 2009 17:22:31 -0400
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <083DBCB9-48B2-4E61-81B5-B7D00A42EAB5@freeswitch.org>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<1b46b4e80910011229k31f7233du10b0fc661730d3f3@mail.gmail.com>
	<F86F950D-9106-49C9-81CA-38B3EB2072F4@jerris.com>
	<1b46b4e80910041519h3581d3d2x7b84041e8c18588@mail.gmail.com>
	<C887D143-F7A1-44B9-B8A2-D34A4B4E582E@jerris.com>
	<1b46b4e80910041609x2f41ceb9u142cb484951a21ed@mail.gmail.com>
	<87f2f3b90910050920w4d17033fl4c386a0018107bdf@mail.gmail.com>
	<1b46b4e80910052119me12a0d9y4653edd831f9555e@mail.gmail.com>
	<1b46b4e80910052142jaa09f08l72e5e4b2b24d10fc@mail.gmail.com>
	<083DBCB9-48B2-4E61-81B5-B7D00A42EAB5@freeswitch.org>
Message-ID: <1b46b4e80910061422q7be2a21fg5d2caaef7e307abb@mail.gmail.com>

That happens with both gateways though, one works and the other doesn't.
Would the rport have anything to do with the registration failing?

The big difference to me is that the working gateway replies a 401
Unauthorized containing:

 WWW-Authenticate: Digest realm="pxextmy.redvoiss.net",
nonce="4acac8fe248a9075a13773274684392a65a40240", qop="auth".

Whereas the non-working gateway's 401 has:

WWW-Authenticate: Digest realm="216.72.10.39",
nonce="4acac08249c439decb2bea539282faf755c80b0c".


What does the qop parameter stand for? Apparently because of that parameter,
FS sends a new REGISTER including this:

Authorization: Digest username="xxxxxxxxx", realm="pxextmy.redvoiss.net",
nonce="4acac8fe248a9075a13773274684392a65a40240",
cnonce="h1DCSizTEi2eMQAdCe9KJA", algorithm=MD5, uri="sip:
pxextmy.redvoiss.net", response="05adb2a7f9d7772e57dc846257484f5d",
qop=auth, nc=00000001.

Instead, on the non-working gateway case, FS sends a REGISTER with this:

Authorization: Digest username="yyyyyyyyy", realm="216.72.10.39",
nonce="4acac08249c439decb2bea539282faf755c80b0c", algorithm=MD5, uri="sip:
216.72.10.39", response="8311db7666779df89d5223e16a611826".


Notice the absence of the qop and nc parameters. I'm guessing the lack of
those parameters causes the gateway (SIP server) to use another nonce and
hence reject the mismatching REGISTER.

BTW, registration from an X-Lite softphone works.


Thanks!

Nicolas


On Tue, Oct 6, 2009 at 10:31 AM, Brian West <brian at freeswitch.org> wrote:

> This looks like you have an ALG messing with packets... notice it says
> rport 5080 but we are sending to 5060.
> /b
>
> On Oct 5, 2009, at 11:42 PM, Nicolas Brenner wrote:
>
> Ignore my previous email, the traces were incomplete, got much better (and
> complete) traces with ngrep (found a suggestion from Brian in the list
> archive, thanks!)
>
> The gateway that registers:
>
> - http://pastebin.freeswitch.org/10607
>
> The one that doesn't:
>
> - http://pastebin.freeswitch.org/10608
>
>
> Thanks again for your time and help!
>
>
> Nicolas
>
>
> On Tue, Oct 6, 2009 at 12:19 AM, Nicolas Brenner <nicolas at medularis.com>wrote:
>
>> There was no sane way of doing that, so I ended up logging the trace from
>> the cli.
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Tue Oct  6 14:45:10 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 6 Oct 2009 16:45:10 -0500
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <1b46b4e80910061422q7be2a21fg5d2caaef7e307abb@mail.gmail.com>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<1b46b4e80910011229k31f7233du10b0fc661730d3f3@mail.gmail.com>
	<F86F950D-9106-49C9-81CA-38B3EB2072F4@jerris.com>
	<1b46b4e80910041519h3581d3d2x7b84041e8c18588@mail.gmail.com>
	<C887D143-F7A1-44B9-B8A2-D34A4B4E582E@jerris.com>
	<1b46b4e80910041609x2f41ceb9u142cb484951a21ed@mail.gmail.com>
	<87f2f3b90910050920w4d17033fl4c386a0018107bdf@mail.gmail.com>
	<1b46b4e80910052119me12a0d9y4653edd831f9555e@mail.gmail.com>
	<1b46b4e80910052142jaa09f08l72e5e4b2b24d10fc@mail.gmail.com>
	<083DBCB9-48B2-4E61-81B5-B7D00A42EAB5@freeswitch.org>
	<1b46b4e80910061422q7be2a21fg5d2caaef7e307abb@mail.gmail.com>
Message-ID: <9C8FB847-FBFB-4382-8C60-865C9FB5CAE7@freeswitch.org>

First off you have to fully understand how SIP authentication works  
the two authorization line are different because one is for a  
challenge and one is a response to a challenge.

http://en.wikipedia.org/wiki/Digest_access_authentication


On Oct 6, 2009, at 4:22 PM, Nicolas Brenner wrote:

> That happens with both gateways though, one works and the other  
> doesn't. Would the rport have anything to do with the registration  
> failing?
>
> The big difference to me is that the working gateway replies a 401  
> Unauthorized containing:
>
>  WWW-Authenticate: Digest realm="pxextmy.redvoiss.net",  
> nonce="4acac8fe248a9075a13773274684392a65a40240", qop="auth".
>
> Whereas the non-working gateway's 401 has:
>
> WWW-Authenticate: Digest realm="216.72.10.39",  
> nonce="4acac08249c439decb2bea539282faf755c80b0c".

What is this gateway? You might actually put the realm param INTO the  
gateway config for this gateway.


> What does the qop parameter stand for? Apparently because of that  
> parameter, FS sends a new REGISTER including this:

Quality of Protection, qop is assumed auth if excluded.

>
> Authorization: Digest username="xxxxxxxxx", realm="pxextmy.redvoiss.net 
> ", nonce="4acac8fe248a9075a13773274684392a65a40240",  
> cnonce="h1DCSizTEi2eMQAdCe9KJA", algorithm=MD5, uri="sip:pxextmy.redvoiss.net 
> ", response="05adb2a7f9d7772e57dc846257484f5d", qop=auth, nc=00000001.

This is a response to a challenge.

> Instead, on the non-working gateway case, FS sends a REGISTER with  
> this:
>
> Authorization: Digest username="yyyyyyyyy", realm="216.72.10.39",  
> nonce="4acac08249c439decb2bea539282faf755c80b0c", algorithm=MD5,  
> uri="sip:216.72.10.39", response="8311db7666779df89d5223e16a611826".

This is a challenge.

> Notice the absence of the qop and nc parameters. I'm guessing the  
> lack of those parameters causes the gateway (SIP server) to use  
> another nonce and hence reject the mismatching REGISTER.

Again challenge vs response.

>
> BTW, registration from an X-Lite softphone works.
>
>
> Thanks!
>
> Nicolas

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From diego.viola at gmail.com  Tue Oct  6 14:45:25 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 6 Oct 2009 21:45:25 +0000
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
Message-ID: <86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>

Instead of complaining and demanding things for free, people should start to
put their money where their mouth is.

Diego

On Tue, Oct 6, 2009 at 8:47 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> hi Anthony,
>
> it is somewhere here:
>
>          switch_status_t
> FSConnection::receive_message(switch_core_session_message_t *msg)
>
>
> anyhow, i will open an issue jira of course.
>
>
> I understand your financial point of view, but anyhow while the entire
> world is wants sip and trying to move to sip, the reality is quite
> different. The majority of voice traffic exchanged via IP is still H323.
> This means a working SIP - H323 interworking is really needed... pity nobody
> wants/has time to work in this direction to produce a decent mod_h323.
>
>
>
> T.
>
>
>
>
>
>
> On Tue, Oct 6, 2009 at 9:25 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> pcap is not as useful as FS console log on debug with:
>> sofia profile internal siptrace on
>>
>> you should be reporting issues to jira under mod_opal not to the mailing
>> list.
>> http://jira.freeswitch.org
>>
>> FYI
>> There is little financial support from the community for h323 which
>> prevents the mod_opal from getting much attention.
>> We actually have to contract the author of opal to help with these issues
>> including the original writing of the module that he did with very little
>> funding and nobody ever wants to pay him to improve it.
>>
>> That does not mean your issue will not be addressed but there is no
>> promise how fast it will be.
>>
>>
>>
>> On Tue, Oct 6, 2009 at 12:37 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>
>>> hello guys,
>>>
>>>
>>> i was playing with mod_opal to see if i can make it working ... well it
>>> seems SIP-H323 interworking is not tuned at all.
>>>
>>> I have a call from a registered sip user (1001) to PSTN via mod_opal
>>>
>>>
>>> <include>
>>>   <extension name="EMERGENCY">
>>>     <condition field="destination_number" expression="^0(112|9[23456])$">
>>>       <action application="set"
>>> data="effective_caller_id_number=1001282122"/>
>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>       <action application="set" data="call_timeout=30"/>
>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>
>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>     </condition>
>>>   </extension>
>>>
>>>   <extension name="SPECIAL_SERVICES">
>>>     <condition field="destination_number"
>>> expression="^0(9[01789]\d{3,4})$">
>>>       <action application="set"
>>> data="effective_caller_id_number=1001282122"/>
>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>       <action application="set" data="call_timeout=30"/>
>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>
>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>     </condition>
>>>   </extension>
>>>
>>>   <extension name="ENYTHING_ELSE">
>>>     <condition field="destination_number"
>>> expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
>>>       <action application="set"
>>> data="effective_caller_id_number=1001282122"/>
>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>       <action application="set" data="call_timeout=30"/>
>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>
>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>     </condition>
>>>   </extension>
>>> </include>
>>>
>>>
>>>
>>> One of the many issues i sow is that FS connects the call on SIP leg
>>> before it actually receives H.225 connect from H323 leg... as it is
>>> configured to send 200 OK on the 1st H.225 message containing a FastStart
>>> element/OLC.
>>>
>>>
>>> Attached is the tcpdump i took on FS machine... just use this filter:
>>> "h225 or h245 or q931 or sip"
>>> Also, you can check the attac CDR,,,, this is an unanswered call i placed
>>> to PSTN and FS billed it 23 seconds.
>>>
>>>
>>>
>>> Can anyone tell where i can do correct SIP - H323 message mappings to
>>> avoid this?
>>>
>>>
>>>
>>> T.
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
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From brian at freeswitch.org  Tue Oct  6 14:46:59 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 6 Oct 2009 16:46:59 -0500
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <1b46b4e80910061422q7be2a21fg5d2caaef7e307abb@mail.gmail.com>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<1b46b4e80910011229k31f7233du10b0fc661730d3f3@mail.gmail.com>
	<F86F950D-9106-49C9-81CA-38B3EB2072F4@jerris.com>
	<1b46b4e80910041519h3581d3d2x7b84041e8c18588@mail.gmail.com>
	<C887D143-F7A1-44B9-B8A2-D34A4B4E582E@jerris.com>
	<1b46b4e80910041609x2f41ceb9u142cb484951a21ed@mail.gmail.com>
	<87f2f3b90910050920w4d17033fl4c386a0018107bdf@mail.gmail.com>
	<1b46b4e80910052119me12a0d9y4653edd831f9555e@mail.gmail.com>
	<1b46b4e80910052142jaa09f08l72e5e4b2b24d10fc@mail.gmail.com>
	<083DBCB9-48B2-4E61-81B5-B7D00A42EAB5@freeswitch.org>
	<1b46b4e80910061422q7be2a21fg5d2caaef7e307abb@mail.gmail.com>
Message-ID: <9106C8E1-498F-4AE8-8D34-7F6AAD4DC8AE@freeswitch.org>

btw My mistake it doesn't assume auth it just calculates the response  
hash differently on this case where qop isn't present.

/b

On Oct 6, 2009, at 4:22 PM, Nicolas Brenner wrote:

>
> What does the qop parameter stand for? Apparently because of that  
> parameter, FS sends a new REGISTER including this:



From jason at jasonjgw.net  Tue Oct  6 15:58:24 2009
From: jason at jasonjgw.net (Jason White)
Date: Wed, 7 Oct 2009 09:58:24 +1100
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
Message-ID: <20091006225824.GA7451@jdc.jasonjgw.net>

Tihomir Culjaga <tculjaga at gmail.com> wrote:
 
> 
> I understand your financial point of view, but anyhow while the entire world
> is wants sip and trying to move to sip, the reality is quite different. The
> majority of voice traffic exchanged via IP is still H323. 

Is there any evidence in support of the above assertion (e.g., survey results
of VoIP traffic)? I've heard of H323 but I don't know anyone who uses it, or
any phones that implement it.

The lack of interest in this forum and the absence of financial support to
improve the H323 support in FreeSWITCH suggest that the level of demand for
this is quite low, relative to SIP.

Of course, improvements are always welcome, so if you're interested in funding
better H323 support, or helping with the module I'm sure the FreeSWITCH
community would welcome your efforts.



From tculjaga at gmail.com  Tue Oct  6 16:41:13 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 7 Oct 2009 01:41:13 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
Message-ID: <65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>

Diego,

what i'm pointing here is the situation where you have a great product that
lacks in one of most common protocol. It is true H323 is going to disappear
(eventually), it is true that the community prefers SIP/IAX instead ... but
the reality still remains. H323 is going to be used for quite a long time to
exchange a lot of traffic while FS will be left aside. Today, when you setup
an IP peering interconnection 80% of carriers will prefer H323.

Of course, developing something costs "time" (and we all know what time
stands for...) and as i said, i understand the financial point of view and i
really understand if nobody is going to work on that, but let's face it FS
doesn't have any usable module to reliably handle H323 protocol.


said that, i don't intend to offend anyone... just facing the reality.


regarding the h323 module, we don't have any issue fixing the existing or
developing a new one... but before we go developing something it is always
better check if the thing you want already exists in an usable state or
not... that's what i did today.


So, I'm interested in a reliable module handling H323v4... anyone else?


T.





On Tue, Oct 6, 2009 at 11:45 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Instead of complaining and demanding things for free, people should start
> to put their money where their mouth is.
>
> Diego
>
>
> On Tue, Oct 6, 2009 at 8:47 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> hi Anthony,
>>
>> it is somewhere here:
>>
>>          switch_status_t
>> FSConnection::receive_message(switch_core_session_message_t *msg)
>>
>>
>> anyhow, i will open an issue jira of course.
>>
>>
>> I understand your financial point of view, but anyhow while the entire
>> world is wants sip and trying to move to sip, the reality is quite
>> different. The majority of voice traffic exchanged via IP is still H323.
>> This means a working SIP - H323 interworking is really needed... pity nobody
>> wants/has time to work in this direction to produce a decent mod_h323.
>>
>>
>>
>> T.
>>
>>
>>
>>
>>
>>
>> On Tue, Oct 6, 2009 at 9:25 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> pcap is not as useful as FS console log on debug with:
>>> sofia profile internal siptrace on
>>>
>>> you should be reporting issues to jira under mod_opal not to the mailing
>>> list.
>>> http://jira.freeswitch.org
>>>
>>> FYI
>>> There is little financial support from the community for h323 which
>>> prevents the mod_opal from getting much attention.
>>> We actually have to contract the author of opal to help with these issues
>>> including the original writing of the module that he did with very little
>>> funding and nobody ever wants to pay him to improve it.
>>>
>>> That does not mean your issue will not be addressed but there is no
>>> promise how fast it will be.
>>>
>>>
>>>
>>> On Tue, Oct 6, 2009 at 12:37 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>>
>>>> hello guys,
>>>>
>>>>
>>>> i was playing with mod_opal to see if i can make it working ... well it
>>>> seems SIP-H323 interworking is not tuned at all.
>>>>
>>>> I have a call from a registered sip user (1001) to PSTN via mod_opal
>>>>
>>>>
>>>> <include>
>>>>   <extension name="EMERGENCY">
>>>>     <condition field="destination_number"
>>>> expression="^0(112|9[23456])$">
>>>>       <action application="set"
>>>> data="effective_caller_id_number=1001282122"/>
>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>>       <action application="set" data="call_timeout=30"/>
>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>
>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>>     </condition>
>>>>   </extension>
>>>>
>>>>   <extension name="SPECIAL_SERVICES">
>>>>     <condition field="destination_number"
>>>> expression="^0(9[01789]\d{3,4})$">
>>>>       <action application="set"
>>>> data="effective_caller_id_number=1001282122"/>
>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>>       <action application="set" data="call_timeout=30"/>
>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>
>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>>     </condition>
>>>>   </extension>
>>>>
>>>>   <extension name="ENYTHING_ELSE">
>>>>     <condition field="destination_number"
>>>> expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
>>>>       <action application="set"
>>>> data="effective_caller_id_number=1001282122"/>
>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>>       <action application="set" data="call_timeout=30"/>
>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>
>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>>     </condition>
>>>>   </extension>
>>>> </include>
>>>>
>>>>
>>>>
>>>> One of the many issues i sow is that FS connects the call on SIP leg
>>>> before it actually receives H.225 connect from H323 leg... as it is
>>>> configured to send 200 OK on the 1st H.225 message containing a FastStart
>>>> element/OLC.
>>>>
>>>>
>>>> Attached is the tcpdump i took on FS machine... just use this filter:
>>>> "h225 or h245 or q931 or sip"
>>>> Also, you can check the attac CDR,,,, this is an unanswered call i
>>>> placed to PSTN and FS billed it 23 seconds.
>>>>
>>>>
>>>>
>>>> Can anyone tell where i can do correct SIP - H323 message mappings to
>>>> avoid this?
>>>>
>>>>
>>>>
>>>> T.
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:213-799-1400
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Tue Oct  6 17:09:25 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 7 Oct 2009 02:09:25 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <20091006225824.GA7451@jdc.jasonjgw.net>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<20091006225824.GA7451@jdc.jasonjgw.net>
Message-ID: <65d96fc80910061709t2bc9aa5ex8769ef7aa3e27671@mail.gmail.com>

thanks for your e-mail,

H323 is mainly used for trunking purpose, inter-carrier traffic exchange...
it is not used to control IP phones :P

well, believe me, I've heard enough of H323 that i'm sick of it :P
What i can tell you comes from my own experience on daily activities i'm
doing for living... Of course, there might be part of the world where H323
dispersed completely but over here in Europe things tend to stick on
tradition :P

Yep, you are right... the forum wants SIP and that's understandable...

anyhow you might check this:
http://www.dailypayload.com/content/3111


T.


On Wed, Oct 7, 2009 at 12:58 AM, Jason White <jason at jasonjgw.net> wrote:

> Tihomir Culjaga <tculjaga at gmail.com> wrote:
>
> >
> > I understand your financial point of view, but anyhow while the entire
> world
> > is wants sip and trying to move to sip, the reality is quite different.
> The
> > majority of voice traffic exchanged via IP is still H323.
>
> Is there any evidence in support of the above assertion (e.g., survey
> results
> of VoIP traffic)? I've heard of H323 but I don't know anyone who uses it,
> or
> any phones that implement it.
>
> The lack of interest in this forum and the absence of financial support to
> improve the H323 support in FreeSWITCH suggest that the level of demand for
> this is quite low, relative to SIP.
>
> Of course, improvements are always welcome, so if you're interested in
> funding
> better H323 support, or helping with the module I'm sure the FreeSWITCH
> community would welcome your efforts.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091007/23f45625/attachment-0001.html 

From diego.viola at gmail.com  Tue Oct  6 17:22:13 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Wed, 7 Oct 2009 00:22:13 +0000
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
Message-ID: <86a32abc0910061722t67ff23ddr62e4f62365539bd4@mail.gmail.com>

Yeah I understand your point of view, but saying "I want a H.323 module" or
"I want a Ferrari" wont magically make it happen.

We need to work on it ourselves or pay to the people that knows how to do it
to do it for us.

There is no other way I think.

Diego



On Tue, Oct 6, 2009 at 11:41 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> Diego,
>
> what i'm pointing here is the situation where you have a great product that
> lacks in one of most common protocol. It is true H323 is going to disappear
> (eventually), it is true that the community prefers SIP/IAX instead ... but
> the reality still remains. H323 is going to be used for quite a long time to
> exchange a lot of traffic while FS will be left aside. Today, when you setup
> an IP peering interconnection 80% of carriers will prefer H323.
>
> Of course, developing something costs "time" (and we all know what time
> stands for...) and as i said, i understand the financial point of view and i
> really understand if nobody is going to work on that, but let's face it FS
> doesn't have any usable module to reliably handle H323 protocol.
>
>
> said that, i don't intend to offend anyone... just facing the reality.
>
>
> regarding the h323 module, we don't have any issue fixing the existing or
> developing a new one... but before we go developing something it is always
> better check if the thing you want already exists in an usable state or
> not... that's what i did today.
>
>
> So, I'm interested in a reliable module handling H323v4... anyone else?
>
>
> T.
>
>
>
>
>
>
> On Tue, Oct 6, 2009 at 11:45 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> Instead of complaining and demanding things for free, people should start
>> to put their money where their mouth is.
>>
>> Diego
>>
>>
>> On Tue, Oct 6, 2009 at 8:47 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>
>>> hi Anthony,
>>>
>>> it is somewhere here:
>>>
>>>          switch_status_t
>>> FSConnection::receive_message(switch_core_session_message_t *msg)
>>>
>>>
>>> anyhow, i will open an issue jira of course.
>>>
>>>
>>> I understand your financial point of view, but anyhow while the entire
>>> world is wants sip and trying to move to sip, the reality is quite
>>> different. The majority of voice traffic exchanged via IP is still H323.
>>> This means a working SIP - H323 interworking is really needed... pity nobody
>>> wants/has time to work in this direction to produce a decent mod_h323.
>>>
>>>
>>>
>>> T.
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Tue, Oct 6, 2009 at 9:25 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> pcap is not as useful as FS console log on debug with:
>>>> sofia profile internal siptrace on
>>>>
>>>> you should be reporting issues to jira under mod_opal not to the mailing
>>>> list.
>>>> http://jira.freeswitch.org
>>>>
>>>> FYI
>>>> There is little financial support from the community for h323 which
>>>> prevents the mod_opal from getting much attention.
>>>> We actually have to contract the author of opal to help with these
>>>> issues including the original writing of the module that he did with very
>>>> little funding and nobody ever wants to pay him to improve it.
>>>>
>>>> That does not mean your issue will not be addressed but there is no
>>>> promise how fast it will be.
>>>>
>>>>
>>>>
>>>> On Tue, Oct 6, 2009 at 12:37 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>>>
>>>>> hello guys,
>>>>>
>>>>>
>>>>> i was playing with mod_opal to see if i can make it working ... well it
>>>>> seems SIP-H323 interworking is not tuned at all.
>>>>>
>>>>> I have a call from a registered sip user (1001) to PSTN via mod_opal
>>>>>
>>>>>
>>>>> <include>
>>>>>   <extension name="EMERGENCY">
>>>>>     <condition field="destination_number"
>>>>> expression="^0(112|9[23456])$">
>>>>>       <action application="set"
>>>>> data="effective_caller_id_number=1001282122"/>
>>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>>>       <action application="set" data="call_timeout=30"/>
>>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>>
>>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>>>     </condition>
>>>>>   </extension>
>>>>>
>>>>>   <extension name="SPECIAL_SERVICES">
>>>>>     <condition field="destination_number"
>>>>> expression="^0(9[01789]\d{3,4})$">
>>>>>       <action application="set"
>>>>> data="effective_caller_id_number=1001282122"/>
>>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>>>       <action application="set" data="call_timeout=30"/>
>>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>>
>>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>>>     </condition>
>>>>>   </extension>
>>>>>
>>>>>   <extension name="ENYTHING_ELSE">
>>>>>     <condition field="destination_number"
>>>>> expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
>>>>>       <action application="set"
>>>>> data="effective_caller_id_number=1001282122"/>
>>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>>>       <action application="set" data="call_timeout=30"/>
>>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>>
>>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>>>     </condition>
>>>>>   </extension>
>>>>> </include>
>>>>>
>>>>>
>>>>>
>>>>> One of the many issues i sow is that FS connects the call on SIP leg
>>>>> before it actually receives H.225 connect from H323 leg... as it is
>>>>> configured to send 200 OK on the 1st H.225 message containing a FastStart
>>>>> element/OLC.
>>>>>
>>>>>
>>>>> Attached is the tcpdump i took on FS machine... just use this filter:
>>>>> "h225 or h245 or q931 or sip"
>>>>> Also, you can check the attac CDR,,,, this is an unanswered call i
>>>>> placed to PSTN and FS billed it 23 seconds.
>>>>>
>>>>>
>>>>>
>>>>> Can anyone tell where i can do correct SIP - H323 message mappings to
>>>>> avoid this?
>>>>>
>>>>>
>>>>>
>>>>> T.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:213-799-1400
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Tue Oct  6 17:58:20 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 6 Oct 2009 19:58:20 -0500
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <86a32abc0910061722t67ff23ddr62e4f62365539bd4@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<86a32abc0910061722t67ff23ddr62e4f62365539bd4@mail.gmail.com>
Message-ID: <191c3a030910061758r2ddea5fand388f0af2c4dfc33@mail.gmail.com>

I didn't mean to start anything.  I'm just saying we work very long hours
and barely get anybody asking about h.323.
I wanted to support it and that's why we took up a collection to get funding
for mod_opal but when only 1 donor showed any interest we were forced to
proceed in our spare time which is very limited.

The developers of opal are not part of our project and they need financial
compensation to be motivated to work on it.  Its not even related to me its
only fair that an outside developer who makes his living as a consultant
would want money to integrate his work into our project.

Like I said, I will do my best to point your issue to the opal devs but I
cannot force them to work on it.




On Tue, Oct 6, 2009 at 7:22 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Yeah I understand your point of view, but saying "I want a H.323 module" or
> "I want a Ferrari" wont magically make it happen.
>
> We need to work on it ourselves or pay to the people that knows how to do
> it to do it for us.
>
> There is no other way I think.
>
> Diego
>
>
>
>
> On Tue, Oct 6, 2009 at 11:41 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> Diego,
>>
>> what i'm pointing here is the situation where you have a great product
>> that lacks in one of most common protocol. It is true H323 is going to
>> disappear (eventually), it is true that the community prefers SIP/IAX
>> instead ... but the reality still remains. H323 is going to be used for
>> quite a long time to exchange a lot of traffic while FS will be left aside.
>> Today, when you setup an IP peering interconnection 80% of carriers will
>> prefer H323.
>>
>> Of course, developing something costs "time" (and we all know what time
>> stands for...) and as i said, i understand the financial point of view and i
>> really understand if nobody is going to work on that, but let's face it FS
>> doesn't have any usable module to reliably handle H323 protocol.
>>
>>
>> said that, i don't intend to offend anyone... just facing the reality.
>>
>>
>> regarding the h323 module, we don't have any issue fixing the existing or
>> developing a new one... but before we go developing something it is always
>> better check if the thing you want already exists in an usable state or
>> not... that's what i did today.
>>
>>
>> So, I'm interested in a reliable module handling H323v4... anyone else?
>>
>>
>> T.
>>
>>
>>
>>
>>
>>
>> On Tue, Oct 6, 2009 at 11:45 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>
>>> Instead of complaining and demanding things for free, people should start
>>> to put their money where their mouth is.
>>>
>>> Diego
>>>
>>>
>>> On Tue, Oct 6, 2009 at 8:47 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>>
>>>> hi Anthony,
>>>>
>>>> it is somewhere here:
>>>>
>>>>          switch_status_t
>>>> FSConnection::receive_message(switch_core_session_message_t *msg)
>>>>
>>>>
>>>> anyhow, i will open an issue jira of course.
>>>>
>>>>
>>>> I understand your financial point of view, but anyhow while the entire
>>>> world is wants sip and trying to move to sip, the reality is quite
>>>> different. The majority of voice traffic exchanged via IP is still H323.
>>>> This means a working SIP - H323 interworking is really needed... pity nobody
>>>> wants/has time to work in this direction to produce a decent mod_h323.
>>>>
>>>>
>>>>
>>>> T.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Tue, Oct 6, 2009 at 9:25 PM, Anthony Minessale <
>>>> anthony.minessale at gmail.com> wrote:
>>>>
>>>>> pcap is not as useful as FS console log on debug with:
>>>>> sofia profile internal siptrace on
>>>>>
>>>>> you should be reporting issues to jira under mod_opal not to the
>>>>> mailing list.
>>>>> http://jira.freeswitch.org
>>>>>
>>>>> FYI
>>>>> There is little financial support from the community for h323 which
>>>>> prevents the mod_opal from getting much attention.
>>>>> We actually have to contract the author of opal to help with these
>>>>> issues including the original writing of the module that he did with very
>>>>> little funding and nobody ever wants to pay him to improve it.
>>>>>
>>>>> That does not mean your issue will not be addressed but there is no
>>>>> promise how fast it will be.
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Oct 6, 2009 at 12:37 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>>>>
>>>>>> hello guys,
>>>>>>
>>>>>>
>>>>>> i was playing with mod_opal to see if i can make it working ... well
>>>>>> it seems SIP-H323 interworking is not tuned at all.
>>>>>>
>>>>>> I have a call from a registered sip user (1001) to PSTN via mod_opal
>>>>>>
>>>>>>
>>>>>> <include>
>>>>>>   <extension name="EMERGENCY">
>>>>>>     <condition field="destination_number"
>>>>>> expression="^0(112|9[23456])$">
>>>>>>       <action application="set"
>>>>>> data="effective_caller_id_number=1001282122"/>
>>>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>>>>       <action application="set" data="call_timeout=30"/>
>>>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>>>
>>>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>>>>     </condition>
>>>>>>   </extension>
>>>>>>
>>>>>>   <extension name="SPECIAL_SERVICES">
>>>>>>     <condition field="destination_number"
>>>>>> expression="^0(9[01789]\d{3,4})$">
>>>>>>       <action application="set"
>>>>>> data="effective_caller_id_number=1001282122"/>
>>>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>>>>       <action application="set" data="call_timeout=30"/>
>>>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>>>
>>>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>>>>     </condition>
>>>>>>   </extension>
>>>>>>
>>>>>>   <extension name="ENYTHING_ELSE">
>>>>>>     <condition field="destination_number"
>>>>>> expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
>>>>>>       <action application="set"
>>>>>> data="effective_caller_id_number=1001282122"/>
>>>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>>>>       <action application="set" data="call_timeout=30"/>
>>>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>>>
>>>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>>>>     </condition>
>>>>>>   </extension>
>>>>>> </include>
>>>>>>
>>>>>>
>>>>>>
>>>>>> One of the many issues i sow is that FS connects the call on SIP leg
>>>>>> before it actually receives H.225 connect from H323 leg... as it is
>>>>>> configured to send 200 OK on the 1st H.225 message containing a FastStart
>>>>>> element/OLC.
>>>>>>
>>>>>>
>>>>>> Attached is the tcpdump i took on FS machine... just use this filter:
>>>>>> "h225 or h245 or q931 or sip"
>>>>>> Also, you can check the attac CDR,,,, this is an unanswered call i
>>>>>> placed to PSTN and FS billed it 23 seconds.
>>>>>>
>>>>>>
>>>>>>
>>>>>> Can anyone tell where i can do correct SIP - H323 message mappings to
>>>>>> avoid this?
>>>>>>
>>>>>>
>>>>>>
>>>>>> T.
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>> iax:guest at conference.freeswitch.org/888
>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>> pstn:213-799-1400
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mgende at gendesign.com  Tue Oct  6 18:52:40 2009
From: mgende at gendesign.com (Michael Gende)
Date: Tue, 6 Oct 2009 20:52:40 -0500
Subject: [Freeswitch-users] Wiki Info: Multi Homed Tutorial
In-Reply-To: <87f2f3b90910060915g167bafa7jd8b0ed5e1b3b2b20@mail.gmail.com>
References: <87f2f3b90910051527x638f55b8hda75d5302267c251@mail.gmail.com>
	<a87d43fd0910052120l76da1536ra92766174dc938d3@mail.gmail.com>
	<87f2f3b90910060915g167bafa7jd8b0ed5e1b3b2b20@mail.gmail.com>
Message-ID: <a87d43fd0910061852y3542f23p2bab39365b4b350b@mail.gmail.com>

Thanks Michael,

I found this right after posting you and have already begun
adding/correcting the guide. Thanks again for getting it up there in the
first place.

Mike G.

On Tue, Oct 6, 2009 at 11:15 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Oct 5, 2009 at 9:20 PM, Michael Gende <mgende at gendesign.com>wrote:
>
>> Michael,
>>
>> Thanks for "wiki-fying" my text-only attempt at some user doc. I should
>> have done that for you. I actually have an updated version with many
>> corrections and the end tabs filled in. Can you point me to info on how I
>> can amend and append what you have kindly put up?
>>
>> Mike G.
>>
>> First, go to wiki.freeswitch.org and create a wiki account. Then, go to
> the Multi_home_tutorial page and click edit (top of the page). You'll see
> that there is wiki markup to learn. If you have questions let me know. Just
> edit the wiki text and click Show Preview to see what the real thing looks
> like. Then click Save Page to save your changes.
>
> Welcome to the world of MediaWiki and thanks for your help!
> -MC
>
>
>
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From mgende at gendesign.com  Tue Oct  6 18:53:14 2009
From: mgende at gendesign.com (Michael Gende)
Date: Tue, 6 Oct 2009 20:53:14 -0500
Subject: [Freeswitch-users] Wiki Info: Multi Homed Tutorial
In-Reply-To: <6f7c60c40910060544h5d1f7251p7194bcaa944a627c@mail.gmail.com>
References: <87f2f3b90910051527x638f55b8hda75d5302267c251@mail.gmail.com>
	<a87d43fd0910052120l76da1536ra92766174dc938d3@mail.gmail.com>
	<6f7c60c40910060544h5d1f7251p7194bcaa944a627c@mail.gmail.com>
Message-ID: <a87d43fd0910061853k1d78e687jdc5d4e7249a1c69f@mail.gmail.com>

Sounds good.

On Tue, Oct 6, 2009 at 7:44 AM, tom <tomabroad at gmail.com> wrote:

> thx guys, that helped me alot!
>
>
> On Tue, Oct 6, 2009 at 12:20 AM, Michael Gende <mgende at gendesign.com>wrote:
>
>> Michael,
>>
>> Thanks for "wiki-fying" my text-only attempt at some user doc. I should
>> have done that for you. I actually have an updated version with many
>> corrections and the end tabs filled in. Can you point me to info on how I
>> can amend and append what you have kindly put up?
>>
>> Mike G.
>>
>>
>> On Mon, Oct 5, 2009 at 5:27 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> FYI,
>>>
>>> For those who've been following the thread about Michael Gende's tutorial
>>> I just wanted to let you know that I his document on the wiki. It can be
>>> found here:
>>>
>>> http://wiki.freeswitch.org/wiki/Multi_home_tutorial
>>>
>>> Please feel free to get in there and try it out, make editorial changes,
>>> etc. I'm sure you won't hurt Mike's feelings by adding your thoughts. :)
>>>
>>> If you have any questions please reply to this thread and we'll take it
>>> from there.
>>>
>>> Thanks,
>>> MC
>>>
>>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Tue Oct  6 20:04:23 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 6 Oct 2009 23:04:23 -0400
Subject: [Freeswitch-users] mod_xml_curl http POST is inconsistent/bug
In-Reply-To: <66EA3166EB339A4489B06286C0876A8A0C0C1F04@mailserv.Globalive.local>
References: <66EA3166EB339A4489B06286C0876A8A0C0C1432@mailserv.Globalive.local><276FEE28-70CF-422D-9F6F-7CF747FDDB22@freeswitch.org><66EA3166EB339A4489B06286C0876A8A0C0C1BA9@mailserv.Globalive.local>
	<191c3a030910060801x3ed48e98x6d69b53c6beaca69@mail.gmail.com>
	<66EA3166EB339A4489B06286C0876A8A0C0C1F04@mailserv.Globalive.local>
Message-ID: <1CD47AE5-ED16-47AE-9730-BA22551BDF31@jerris.com>

Could you open a bug on jira.freeswitch.org as a feature request to  
make this a configurable param.  (patches that do it even better)

Mike

On Oct 6, 2009, at 12:55 PM, Christian Damianidis wrote:

> I?ve tested this and making the change from ANY to BASIC worked.  
> Thanks for the help.
> It no longer sends the initial post without auth.
> From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
> Sent: Tuesday, October 06, 2009 11:02 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] mod_xml_curl http POST is  
> inconsistent/bug
>
> My guess is that we configure the curl to support the full range of  
> http auth methods.
> Some of them like Digest require a challenge and realm etc so it's  
> probably asking without auth header because it cannot create one  
> until it gets that data.  In the case of Basic you can send the  
> login and pass right away but it does not know in advance that it  
> will be basic.
>
> Here is a snippet from the libcurl api docs:
> -------------------------------------------------------------------------------------------------------------------------------------------------------------
> Both these options allow you to set multiple types (by ORing them  
> together), to make libcurl pick the most secure one out of the types  
> the server/proxy claims to support. This method does however add a  
> round-trip since libcurl must first ask the server what it supports:
>
>  curl_easy_setopt(easyhandle, CURLOPT_HTTPAUTH,  CURLAUTH_DIGEST| 
> CURLAUTH_BASIC);
>
> -------------------------------------------------------------------------------------------------------------------------------------------------------------
>
> So my guess is that if we set it to only support basic, then it  
> would work how you expect so if you want to test it for me I can  
> make it into a parameter.
>
> edit: /usr/src/freeswitch.trunk/src/mod/xml_int/mod_xml_curl/ 
> mod_xml_curl.c line 220
> change
>
> curl_easy_setopt(curl_handle, CURLOPT_HTTPAUTH, CURLAUTH_ANY);
>
> to
>
> curl_easy_setopt(curl_handle, CURLOPT_HTTPAUTH, CURLAUTH_BASIC);
>
>
> If this works i'll think about exposing the auth methods so you can  
> choose them in the config.
>
>
>
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From shaheryarkh at googlemail.com  Tue Oct  6 21:24:31 2009
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Wed, 7 Oct 2009 10:24:31 +0600
Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Followup,
	Agenda For This Week
In-Reply-To: <B5ECAE34-B846-4408-8AC1-6D94ADCC672C@freeswitch.org>
References: <87f2f3b90910051203v7bdbca5amfc82d193f3a4f1a3@mail.gmail.com>
	<B5ECAE34-B846-4408-8AC1-6D94ADCC672C@freeswitch.org>
Message-ID: <b16156850910062124h4bae532enc3624e18d1b20b2f@mail.gmail.com>

I got a lot of problem last week for making conference call. I was at home
(conference call starts at 2200hours PKST, my time) and unable to make SIP
call since the government has blocked it. So my only choice was Skype, but
unfortunately DTMF wasn't working, i get connected on Skypiax5 for about 7-8
times, but DTMF problem never let me get in to conference. Is there any
solution for this?

I really need to discuss a lot of things about FS documentation in
conference, like what FS community can expect from it and what not? how i
have planned it? how much progress has done? what are the problems me and my
team are facing (which has slow us down considerably)? etc. etc.

I have a solution for it but that needs testing. The plan is to use one of
my FS servers to connect my jingle calls from GTalk to conference server
over SIP. How can i test this setup with conference server, any ideas?

If any one else also interested in getting connected to weekly conference
call through this setup then i can also extend this setup as needed.

Thank you.


On Tue, Oct 6, 2009 at 1:30 AM, Brian West <brian at freeswitch.org> wrote:

> It always supported 48kHz CELT but the conference itself was running
> at 32kHz so everyone 48k had to be down sampled.  Now you all get to
> be up sampled.  w00t!
>
> /b
>
> On Oct 5, 2009, at 2:03 PM, Michael Collins wrote:
>
> > * Starting with the upcoming meeting (Oct 9) the conference will
> > support 48kHz CELT codec.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
________________________________________________________
|
                     |
| FATAL ERROR                                                           ---
O X |
|_______________________________________________________|
|                        You have moved the mouse.
 |
| Windows must be restarted for the changes to take effect.   |
|                                        <OK>
               |
####################################/


Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From srinivas.ksvreddy at gmail.com  Tue Oct  6 23:25:34 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Wed, 7 Oct 2009 11:55:34 +0530
Subject: [Freeswitch-users] mod_xml_curl
Message-ID: <f8af5740910062325u47da2813sc2ffc0cf9d688e44@mail.gmail.com>

Hi,

i want use mod_xml_curl, the xml files also there in my local system, i dont
want to take from any other system,
can any please tell me how to configure mox_xml.conf.xml,  how can i use
<bindings> to local folders

Thanks
Srinivasula Reddy K
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From srinivas.ksvreddy at gmail.com  Tue Oct  6 23:43:12 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Wed, 7 Oct 2009 12:13:12 +0530
Subject: [Freeswitch-users] mod_xml_curl
Message-ID: <f8af5740910062343q463fc2dt7e08cf610d28e351@mail.gmail.com>

Hi,

i want use mod_xml_curl, the xml files also there in my local system, i dont
want to take from any other system,
can any please tell me how to configure mox_xml.conf.xml,  how can i use
<bindings> to local folders

Thanks
Srinivasula Reddy K
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From brian at freeswitch.org  Tue Oct  6 23:52:16 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 7 Oct 2009 01:52:16 -0500
Subject: [Freeswitch-users] mod_xml_curl
In-Reply-To: <f8af5740910062343q463fc2dt7e08cf610d28e351@mail.gmail.com>
References: <f8af5740910062343q463fc2dt7e08cf610d28e351@mail.gmail.com>
Message-ID: <6F6B493F-6233-4249-B1AE-1628AB6333B6@freeswitch.org>

http://wiki.freeswitch.org/wiki/Mod_xml_curl  Is a great start.

/b

On Oct 7, 2009, at 1:43 AM, srinivasula reddy wrote:

> Hi,
>
> i want use mod_xml_curl, the xml files also there in my local  
> system, i dont want to take from any other system,
> can any please tell me how to configure mox_xml.conf.xml,  how can i  
> use <bindings> to local folders
>
> Thanks
> Srinivasula Reddy K _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From srinivas.ksvreddy at gmail.com  Wed Oct  7 00:28:37 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Wed, 7 Oct 2009 12:58:37 +0530
Subject: [Freeswitch-users] mod_xml_curl
In-Reply-To: <6F6B493F-6233-4249-B1AE-1628AB6333B6@freeswitch.org>
References: <f8af5740910062343q463fc2dt7e08cf610d28e351@mail.gmail.com>
	<6F6B493F-6233-4249-B1AE-1628AB6333B6@freeswitch.org>
Message-ID: <f8af5740910070028k3342b55fkc04e859008c212c5@mail.gmail.com>

Hi brain west,

Thank you very much for your reply,  i have gone through the link what u
have sent me, they have given how to access remote system through webserver,
but they have not given how to access from local system,

if u know please hellp me

Thanks

On Wed, Oct 7, 2009 at 12:22 PM, Brian West <brian at freeswitch.org> wrote:

> http://wiki.freeswitch.org/wiki/Mod_xml_curl  Is a great start.
> /b
>
> On Oct 7, 2009, at 1:43 AM, srinivasula reddy wrote:
>
> Hi,
>
> i want use mod_xml_curl, the xml files also there in my local system, i
> dont want to take from any other system,
> can any please tell me how to configure mox_xml.conf.xml,  how can i use
> <bindings> to local folders
>
> Thanks
> Srinivasula Reddy K _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Srinivasula Reddy K
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From jason at jasonjgw.net  Wed Oct  7 00:49:50 2009
From: jason at jasonjgw.net (Jason White)
Date: Wed, 7 Oct 2009 18:49:50 +1100
Subject: [Freeswitch-users] mod_xml_curl
In-Reply-To: <f8af5740910070028k3342b55fkc04e859008c212c5@mail.gmail.com>
References: <f8af5740910062343q463fc2dt7e08cf610d28e351@mail.gmail.com>
	<6F6B493F-6233-4249-B1AE-1628AB6333B6@freeswitch.org>
	<f8af5740910070028k3342b55fkc04e859008c212c5@mail.gmail.com>
Message-ID: <20091007074950.GA30931@jdc.jasonjgw.net>

srinivasula reddy <srinivas.ksvreddy at gmail.com> wrote:
> 
> Thank you very much for your reply,  i have gone through the link what u
> have sent me, they have given how to access remote system through webserver,
> but they have not given how to access from local system,

On that page, there is an example of how to use mod_curl to access a local
system. Do you know what localhost means?

You can't use a file:// URL, you have to use an HTTP URL and set up a (small)
HTTP server on localhost.



From moizchinoy at gmail.com  Wed Oct  7 00:53:02 2009
From: moizchinoy at gmail.com (Moiz Chinoy)
Date: Wed, 7 Oct 2009 11:53:02 +0400
Subject: [Freeswitch-users] Mobile Phone As GSM Gateway....
In-Reply-To: <23f91030910060504y3d060990l44ec913fb130954c@mail.gmail.com>
References: <29b888f80910060058m4298a829s414a322f8548ff81@mail.gmail.com>
	<23f91030910060504y3d060990l44ec913fb130954c@mail.gmail.com>
Message-ID: <29b888f80910070053j75b344c3n99c2d53045c08e00@mail.gmail.com>

Thanks for your replies....

gsmopen,org seems interesting but it does not have any documentation.
Can anyone point me where I can find information regarding this
project.


On Tue, Oct 6, 2009 at 4:04 PM, Seven Du <dujinfang at gmail.com> wrote:
> maybe you can check this: http://www.gsmopen.org/
>
> 2009/10/6 Moiz Chinoy <moizchinoy at gmail.com>
>>
>> Hi,
>>
>> Is it possible to connect a mobile phone (GSM phone) to Freeswitch and
>> use this as a GSM gateway?
>>
>> --
>> Regards,
>> Moiz Chinoy.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Regards,
Moiz Chinoy.


From gmaruzz at celliax.org  Wed Oct  7 01:16:18 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 7 Oct 2009 10:16:18 +0200
Subject: [Freeswitch-users] Mobile Phone As GSM Gateway....
In-Reply-To: <29b888f80910070053j75b344c3n99c2d53045c08e00@mail.gmail.com>
References: <29b888f80910060058m4298a829s414a322f8548ff81@mail.gmail.com> 
	<23f91030910060504y3d060990l44ec913fb130954c@mail.gmail.com> 
	<29b888f80910070053j75b344c3n99c2d53045c08e00@mail.gmail.com>
Message-ID: <7b197bef0910070116l7703f0fbk57244b0b763c933a@mail.gmail.com>

On Wed, Oct 7, 2009 at 9:53 AM, Moiz Chinoy <moizchinoy at gmail.com> wrote:
> Thanks for your replies....
>
> gsmopen,org seems interesting but it does not have any documentation.
> Can anyone point me where I can find information regarding this
> project.
>

:)
it is prealpha now, will available as alpha in couple week or so...

Docs will change a lot before being alpha, but... in the spirit of
openness... this is what is in the works :
http://wiki.freeswitch.org/wiki/GSMopen

>
> On Tue, Oct 6, 2009 at 4:04 PM, Seven Du <dujinfang at gmail.com> wrote:
>> maybe you can check this: http://www.gsmopen.org/
>>
>> 2009/10/6 Moiz Chinoy <moizchinoy at gmail.com>
>>>
>>> Hi,
>>>
>>> Is it possible to connect a mobile phone (GSM phone) to Freeswitch and
>>> use this as a GSM gateway?
>>>
>>> --
>>> Regards,
>>> Moiz Chinoy.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Regards,
> Moiz Chinoy.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From lyncker at lyth.de  Wed Oct  7 01:30:54 2009
From: lyncker at lyth.de (Filip Lyncker)
Date: Wed, 07 Oct 2009 10:30:54 +0200
Subject: [Freeswitch-users] how to Conference  with picup?
Message-ID: <4ACC51BE.6000201@lyth.de>

Hi List,

maybe someone can give me some hints to get faster into this stuff. Here 
the use case:

I am Ext A and talking to Ext B. Now Ext C is calling to another number 
: D.


How can I

1. Pic up the calling Ext C ?
2. Include it to my current call to Ext A ?


I would need some infos wich elements in my dialplan i have to use and 
how to configure them ... maybe you guys know some documentation else 
about this issue ...

thanks a lot


regards,

Filip

-- 
_________________________________
Filip Lyncker, Dipl.-Inform. (FH)


Lyncker & Theis GmbH
Wilhelmstr. 16
65185 Wiesbaden
Germany

Fon +49 611/9006951
Fax +49 611/9406125


Handelsregister: HRB 23156 Amtsgericht Wiesbaden
Steuernummer: 4023897051
USt-IdNr.: DE255806399

Gesch?ftsf?hrer:
Filip Lyncker,
Armin Theis 




From tculjaga at gmail.com  Wed Oct  7 01:37:50 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 7 Oct 2009 10:37:50 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <191c3a030910061758r2ddea5fand388f0af2c4dfc33@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<86a32abc0910061722t67ff23ddr62e4f62365539bd4@mail.gmail.com>
	<191c3a030910061758r2ddea5fand388f0af2c4dfc33@mail.gmail.com>
Message-ID: <65d96fc80910070137k52be89e3g846e55db72a8de5d@mail.gmail.com>

Anthony,

of course, nobody wants to start anything... we are all here to help making
FS a better product.

so, regarding the founding for mod_opal ... what is the amount you need?


Tihomir.



On Wed, Oct 7, 2009 at 2:58 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> I didn't mean to start anything.  I'm just saying we work very long hours
> and barely get anybody asking about h.323.
> I wanted to support it and that's why we took up a collection to get
> funding for mod_opal but when only 1 donor showed any interest we were
> forced to proceed in our spare time which is very limited.
>
> The developers of opal are not part of our project and they need financial
> compensation to be motivated to work on it.  Its not even related to me its
> only fair that an outside developer who makes his living as a consultant
> would want money to integrate his work into our project.
>
> Like I said, I will do my best to point your issue to the opal devs but I
> cannot force them to work on it.
>
>
>
>
>
> On Tue, Oct 6, 2009 at 7:22 PM, Diego Viola <diego.viola at gmail.com> wrote:
>
>> Yeah I understand your point of view, but saying "I want a H.323 module"
>> or "I want a Ferrari" wont magically make it happen.
>>
>> We need to work on it ourselves or pay to the people that knows how to do
>> it to do it for us.
>>
>> There is no other way I think.
>>
>> Diego
>>
>>
>>
>>
>> On Tue, Oct 6, 2009 at 11:41 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>
>>> Diego,
>>>
>>> what i'm pointing here is the situation where you have a great product
>>> that lacks in one of most common protocol. It is true H323 is going to
>>> disappear (eventually), it is true that the community prefers SIP/IAX
>>> instead ... but the reality still remains. H323 is going to be used for
>>> quite a long time to exchange a lot of traffic while FS will be left aside.
>>> Today, when you setup an IP peering interconnection 80% of carriers will
>>> prefer H323.
>>>
>>> Of course, developing something costs "time" (and we all know what time
>>> stands for...) and as i said, i understand the financial point of view and i
>>> really understand if nobody is going to work on that, but let's face it FS
>>> doesn't have any usable module to reliably handle H323 protocol.
>>>
>>>
>>> said that, i don't intend to offend anyone... just facing the reality.
>>>
>>>
>>> regarding the h323 module, we don't have any issue fixing the existing or
>>> developing a new one... but before we go developing something it is always
>>> better check if the thing you want already exists in an usable state or
>>> not... that's what i did today.
>>>
>>>
>>> So, I'm interested in a reliable module handling H323v4... anyone else?
>>>
>>>
>>> T.
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Tue, Oct 6, 2009 at 11:45 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>
>>>> Instead of complaining and demanding things for free, people should
>>>> start to put their money where their mouth is.
>>>>
>>>> Diego
>>>>
>>>>
>>>> On Tue, Oct 6, 2009 at 8:47 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>>>
>>>>> hi Anthony,
>>>>>
>>>>> it is somewhere here:
>>>>>
>>>>>          switch_status_t
>>>>> FSConnection::receive_message(switch_core_session_message_t *msg)
>>>>>
>>>>>
>>>>> anyhow, i will open an issue jira of course.
>>>>>
>>>>>
>>>>> I understand your financial point of view, but anyhow while the entire
>>>>> world is wants sip and trying to move to sip, the reality is quite
>>>>> different. The majority of voice traffic exchanged via IP is still H323.
>>>>> This means a working SIP - H323 interworking is really needed... pity nobody
>>>>> wants/has time to work in this direction to produce a decent mod_h323.
>>>>>
>>>>>
>>>>>
>>>>> T.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Oct 6, 2009 at 9:25 PM, Anthony Minessale <
>>>>> anthony.minessale at gmail.com> wrote:
>>>>>
>>>>>> pcap is not as useful as FS console log on debug with:
>>>>>> sofia profile internal siptrace on
>>>>>>
>>>>>> you should be reporting issues to jira under mod_opal not to the
>>>>>> mailing list.
>>>>>> http://jira.freeswitch.org
>>>>>>
>>>>>> FYI
>>>>>> There is little financial support from the community for h323 which
>>>>>> prevents the mod_opal from getting much attention.
>>>>>> We actually have to contract the author of opal to help with these
>>>>>> issues including the original writing of the module that he did with very
>>>>>> little funding and nobody ever wants to pay him to improve it.
>>>>>>
>>>>>> That does not mean your issue will not be addressed but there is no
>>>>>> promise how fast it will be.
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Tue, Oct 6, 2009 at 12:37 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>>>>>
>>>>>>> hello guys,
>>>>>>>
>>>>>>>
>>>>>>> i was playing with mod_opal to see if i can make it working ... well
>>>>>>> it seems SIP-H323 interworking is not tuned at all.
>>>>>>>
>>>>>>> I have a call from a registered sip user (1001) to PSTN via mod_opal
>>>>>>>
>>>>>>>
>>>>>>> <include>
>>>>>>>   <extension name="EMERGENCY">
>>>>>>>     <condition field="destination_number"
>>>>>>> expression="^0(112|9[23456])$">
>>>>>>>       <action application="set"
>>>>>>> data="effective_caller_id_number=1001282122"/>
>>>>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>>>>>       <action application="set" data="call_timeout=30"/>
>>>>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>>>>
>>>>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>>>>>     </condition>
>>>>>>>   </extension>
>>>>>>>
>>>>>>>   <extension name="SPECIAL_SERVICES">
>>>>>>>     <condition field="destination_number"
>>>>>>> expression="^0(9[01789]\d{3,4})$">
>>>>>>>       <action application="set"
>>>>>>> data="effective_caller_id_number=1001282122"/>
>>>>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>>>>>       <action application="set" data="call_timeout=30"/>
>>>>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>>>>
>>>>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>>>>>     </condition>
>>>>>>>   </extension>
>>>>>>>
>>>>>>>   <extension name="ENYTHING_ELSE">
>>>>>>>     <condition field="destination_number"
>>>>>>> expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
>>>>>>>       <action application="set"
>>>>>>> data="effective_caller_id_number=1001282122"/>
>>>>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>>>>>       <action application="set" data="call_timeout=30"/>
>>>>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>>>>
>>>>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>>>>>     </condition>
>>>>>>>   </extension>
>>>>>>> </include>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> One of the many issues i sow is that FS connects the call on SIP leg
>>>>>>> before it actually receives H.225 connect from H323 leg... as it is
>>>>>>> configured to send 200 OK on the 1st H.225 message containing a FastStart
>>>>>>> element/OLC.
>>>>>>>
>>>>>>>
>>>>>>> Attached is the tcpdump i took on FS machine... just use this filter:
>>>>>>> "h225 or h245 or q931 or sip"
>>>>>>> Also, you can check the attac CDR,,,, this is an unanswered call i
>>>>>>> placed to PSTN and FS billed it 23 seconds.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Can anyone tell where i can do correct SIP - H323 message mappings to
>>>>>>> avoid this?
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> T.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Anthony Minessale II
>>>>>>
>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>> ClueCon http://www.cluecon.com/
>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>
>>>>>> AIM: anthm
>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>
>>>>>> FreeSWITCH Developer Conference
>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>> pstn:213-799-1400
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
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From lakindia89 at gmail.com  Wed Oct  7 03:00:24 2009
From: lakindia89 at gmail.com (lakshmanan)
Date: Wed, 7 Oct 2009 03:00:24 -0700 (PDT)
Subject: [Freeswitch-users] oz debug says error
In-Reply-To: <87f2f3b90910060906y4b087d7coa91de38d78a3456b@mail.gmail.com>
References: <7d79b3930910050020i1b96541dk26c8d72c220ded8d@mail.gmail.com>
	<9F7D1E871F0C4184AA8ED441CF415C8A@cune.pri>
	<25749736.post@talk.nabble.com>
	<20091005124858.81857415806@mail.cune.org>
	<7d79b3930910052106n59cf9cbema68e4d6ccc274034@mail.gmail.com>
	<87f2f3b90910060906y4b087d7coa91de38d78a3456b@mail.gmail.com>
Message-ID: <25783750.post@talk.nabble.com>


Here are the details

openzap.conf
[span zt PRI_1]
trunk_type => e1
b-channel => 1:1-15
d-channel=> 1:16
b-channel => 1:17-31

Zaptel.conf

span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16

loadzone        = us
defaultzone     = us



mercutioviz wrote:
> 
> Pastebin your openzap.conf file. Also, is this Sangoma or zaptel-based
> hardware? If it's Sangoma, pastebin your wanpipe1.conf file. If zaptel,
> please paste your zaptel.conf file.
> 
> -MC
> 
> On Mon, Oct 5, 2009 at 9:06 PM, lakshmanan ganapathy
> <lakindia89 at gmail.com>wrote:
> 
>> Openzap.conf.xml
>> <configuration name="openzap.conf" description="OpenZAP Configuration">
>>         <settings>
>>                 
>>         </settings>
>>         <libpri_spans>
>>                 
>>                         
>>                         
>>                         
>>                         
>>                 
>>         </libpri_spans>
>> </configuration>
>>
>> Output of oz list in fs_cli
>>
>> span: 1 (PRI_1)
>> type: isdn
>> chan_count: 47
>> dialplan: XML
>> context: default
>> dial_regex:
>> fail_dial_regex:
>> hold_music:
>> analog_options none
>>
>>
>> freeswitch.log
>>
>> http://pastebin.freeswitch.org/10604
>>
>>
>>
>>
>>
>> On Mon, Oct 5, 2009 at 6:18 PM, <Russell.Mosemann at cune.org> wrote:
>>
>>> lakshmanan <lakindia89 at gmail.com> said:
>>> > Thanks for pointing that.
>>> > I also tried that.
>>> > But in that case, I'm not able to make a call through openzap.
>>>
>>> What is in openzap.conf.xml? If you start fs_cli and enter "oz list",
>>> what does it show? Copy the ozmod lines from freeswitch.log to
>>> pastebin.freeswitch.org and post the link here so that we can see what
>>> openzap does when freeswitch starts.
>>>
>>> --
>>> Russell Mosemann
>>>
>>>
>>>
>>> ________________________________________________________
>>> Concordia University, Nebraska
>>> See http://www.cune.edu/ for the latest news and events!
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/oz-debug-says-error-tp25746215p25783750.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From lakindia89 at gmail.com  Wed Oct  7 03:34:50 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Wed, 7 Oct 2009 16:04:50 +0530
Subject: [Freeswitch-users] openzap Failure opening channel error
Message-ID: <7d79b3930910070334g53cef0c5m63037e264f1a9233@mail.gmail.com>

Hi,
Again I was struck in a problem, Here is the scenario.

On incomming call, I just call an event outboud socket. But what happens is,
for the first 15 call, it is working fine. But from the 16th call to 30th
call, it says the below error.

2009-10-07 15:07:48.201846 [WARNING] ozmod_libpri.c:761 --Failure opening
channel 1:17 (ignored)
2009-10-07 15:07:55.381861 [WARNING] ozmod_libpri.c:761 --Failure opening
channel 1:18 (ignored)
2009-10-07 15:07:58.569774 [WARNING] ozmod_libpri.c:761 --Failure opening
channel 1:19 (ignored)
2009-10-07 15:08:01.37824 [WARNING] ozmod_libpri.c:761 --Failure opening
channel 1:20 (ignored)
2009-10-07 15:08:03.129846 [WARNING] ozmod_libpri.c:761 --Failure opening
channel 1:21 (ignored)
2009-10-07 15:08:04.825851 [WARNING] ozmod_libpri.c:761 --Failure opening
channel 1:22 (ignored)
2009-10-07 15:08:06.289977 [WARNING] ozmod_libpri.c:761 --Failure opening
channel 1:23 (ignored)
2009-10-07 15:08:07.761961 [WARNING] ozmod_libpri.c:761 --Failure opening
channel 1:24 (ignored)
2009-10-07 15:08:09.737944 [WARNING] ozmod_libpri.c:761 --Failure opening
channel 1:25 (ignored)
2009-10-07 15:08:11.462018 [WARNING] ozmod_libpri.c:761 --Failure opening
channel 1:26 (ignored)
2009-10-07 15:08:13.566024 [WARNING] ozmod_libpri.c:761 --Failure opening
channel 1:27 (ignored)
2009-10-07 15:08:15.430163 [WARNING] ozmod_libpri.c:761 --Failure opening
channel 1:28 (ignored)
2009-10-07 15:08:17.446103 [WARNING] ozmod_libpri.c:761 --Failure opening
channel 1:29 (ignored)
2009-10-07 15:08:19.430118 [WARNING] ozmod_libpri.c:761 --Failure opening
channel 1:30 (ignored)
2009-10-07 15:08:21.358121 [WARNING] ozmod_libpri.c:761 --Failure opening
channel 1:31 (ignored)


But for the next call, when it is opening channel 1:1, it is executing my
dial plans.
I don't know why it failed when it is choosing 1:17-1:31. Any one has any
idea.

Below are my configuration details:

openzap.conf
[span zt PRI_1]
trunk_type => e1
b-channel => 1:1-15
d-channel=> 1:16
b-channel => 1:17-31

openzap.conf.xml
<configuration name="openzap.conf" description="OpenZAP Configuration">
        <settings>
                <param name="debug" value="7"/>
        </settings>
        <libpri_spans>
                <span name="PRI_1">
                        <param name="node" value="cpe"/>
                        <param name="switch" value="euroisdn"/>
                        <param name="dialplan" value="XML"/>
                        <param name="context" value="default"/>
                </span>
        </libpri_spans>
</configuration>

zaptel.conf

span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16

loadzone        = us
defaultzone     = us

oz list

span: 1 (PRI_1)
type: isdn
chan_count: 47
dialplan: XML
context: default
dial_regex:
fail_dial_regex:
hold_music:
analog_options none
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From woodydickson at gmail.com  Wed Oct  7 03:40:24 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Wed, 7 Oct 2009 18:40:24 +0800
Subject: [Freeswitch-users] unable to configure Digium TDM400P
Message-ID: <a44688a0910070340p4209e7b5yec9218b8c4fc692c@mail.gmail.com>

Hi,

I am trying to setup a Digium TDM400P following the instruction on the
wiki.
It is a 1 fxo and 1 fxs card, so I tried
loadzone=in
defaultzone=in
fxsks=2
fxoks=1

and

loadzone=in
defaultzone=in
fxsks=1
fxoks=2

None works.  Does anyone know how it should be configured?

Here is what I get by following the wiki.

[root at localhost zaptel]# ztcfg -vv

Zaptel Version: SVN-branch-1.4-r4629M
Echo Canceller: MG2
Configuration
======================


Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Slaves: 02)

2 channels to configure.

Changing signalling on channel 1 from FXO Kewlstart to FXS Kewlstart
Changing signalling on channel 2 from FXS Kewlstart to FXO Kewlstart


[root at localhost zaptel]# ztcfg -vv

Zaptel Version: SVN-branch-1.4-r4629M
Echo Canceller: MG2
Configuration
======================


Channel map:

Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)

2 channels to configure.

Changing signalling on channel 1 from FXS Kewlstart to FXO Kewlstart
Changing signalling on channel 2 from FXO Kewlstart to FXS Kewlstart


[root at localhost zaptel]# lspci
00:14.0 Ethernet controller: Digium, Inc. TDM400P (rev 11)
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From Russell.Mosemann at cune.org  Wed Oct  7 04:59:39 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Wed, 7 Oct 2009 11:59:39 -0000
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <7d79b3930910070334g53cef0c5m63037e264f1a9233@mail.gmail.com>
Message-ID: <20091007115939.4856239DEA0@mail.cune.org>

lakshmanan ganapathy <lakindia89 at gmail.com> said:

> On incomming call, I just call an event outboud socket. But what
happens is,
> for the first 15 call, it is working fine. But from the 16th call to 30th
> call, it says the below error.

What is displayed with

  ztcfg -vv

What is displayed with

  cat /proc/zaptel/1

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From lakindia89 at gmail.com  Wed Oct  7 05:12:23 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Wed, 7 Oct 2009 17:42:23 +0530
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <20091007115939.4856239DEA0@mail.cune.org>
References: <7d79b3930910070334g53cef0c5m63037e264f1a9233@mail.gmail.com>
	<20091007115939.4856239DEA0@mail.cune.org>
Message-ID: <7d79b3930910070512n5f2f795fob19eece48e2a872@mail.gmail.com>

[debian :~]# ztcfg -vvv
Zaptel Version: 1.4.11
Echo Canceller: MG2
Configuration
======================

SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)

Channel map:

Channel 01: Clear channel (Default) (Slaves: 01)
Channel 02: Clear channel (Default) (Slaves: 02)
Channel 03: Clear channel (Default) (Slaves: 03)
Channel 04: Clear channel (Default) (Slaves: 04)
Channel 05: Clear channel (Default) (Slaves: 05)
Channel 06: Clear channel (Default) (Slaves: 06)
Channel 07: Clear channel (Default) (Slaves: 07)
Channel 08: Clear channel (Default) (Slaves: 08)
Channel 09: Clear channel (Default) (Slaves: 09)
Channel 10: Clear channel (Default) (Slaves: 10)
Channel 11: Clear channel (Default) (Slaves: 11)
Channel 12: Clear channel (Default) (Slaves: 12)
Channel 13: Clear channel (Default) (Slaves: 13)
Channel 14: Clear channel (Default) (Slaves: 14)
Channel 15: Clear channel (Default) (Slaves: 15)
Channel 16: D-channel (Default) (Slaves: 16)
Channel 17: Clear channel (Default) (Slaves: 17)
Channel 18: Clear channel (Default) (Slaves: 18)
Channel 19: Clear channel (Default) (Slaves: 19)
Channel 20: Clear channel (Default) (Slaves: 20)
Channel 21: Clear channel (Default) (Slaves: 21)
Channel 22: Clear channel (Default) (Slaves: 22)
Channel 23: Clear channel (Default) (Slaves: 23)
Channel 24: Clear channel (Default) (Slaves: 24)
Channel 25: Clear channel (Default) (Slaves: 25)
Channel 26: Clear channel (Default) (Slaves: 26)
Channel 27: Clear channel (Default) (Slaves: 27)
Channel 28: Clear channel (Default) (Slaves: 28)
Channel 29: Clear channel (Default) (Slaves: 29)
Channel 30: Clear channel (Default) (Slaves: 30)
Channel 31: Clear channel (Default) (Slaves: 31)

31 channels to configure.

[debian :~]# cat /proc/zaptel/1
+ cat /proc/zaptel/1
Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS ClockSource
        Timing slips: 1

        1 TE2/0/1/1 Clear
        2 TE2/0/1/2 Clear
        3 TE2/0/1/3 Clear
        4 TE2/0/1/4 Clear
        5 TE2/0/1/5 Clear
        6 TE2/0/1/6 Clear
        7 TE2/0/1/7 Clear
        8 TE2/0/1/8 Clear
        9 TE2/0/1/9 Clear
        10 TE2/0/1/10 Clear
        11 TE2/0/1/11 Clear
        12 TE2/0/1/12 Clear
        13 TE2/0/1/13 Clear
        14 TE2/0/1/14 Clear
        15 TE2/0/1/15 Clear
        16 TE2/0/1/16 HDLCFCS
        17 TE2/0/1/17 Clear
        18 TE2/0/1/18 Clear
        19 TE2/0/1/19 Clear
        20 TE2/0/1/20 Clear
        21 TE2/0/1/21 Clear
        22 TE2/0/1/22 Clear
        23 TE2/0/1/23 Clear
        24 TE2/0/1/24 Clear
        25 TE2/0/1/25 Clear
        26 TE2/0/1/26 Clear
        27 TE2/0/1/27 Clear
        28 TE2/0/1/28 Clear
        29 TE2/0/1/29 Clear
        30 TE2/0/1/30 Clear
        31 TE2/0/1/31 Clear


On Wed, Oct 7, 2009 at 5:29 PM, <Russell.Mosemann at cune.org> wrote:

> lakshmanan ganapathy <lakindia89 at gmail.com> said:
>
> > On incomming call, I just call an event outboud socket. But what
> happens is,
> > for the first 15 call, it is working fine. But from the 16th call to 30th
> > call, it says the below error.
>
> What is displayed with
>
>  ztcfg -vv
>
> What is displayed with
>
>  cat /proc/zaptel/1
>
> --
> Russell Mosemann
>
>
>
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ahmedmunir007 at gmail.com  Wed Oct  7 05:34:54 2009
From: ahmedmunir007 at gmail.com (Ahmed Munir)
Date: Wed, 7 Oct 2009 17:34:54 +0500
Subject: [Freeswitch-users] Configuring Nibble Bill
Message-ID: <a60fed5e0910070534v6e00bad9r518c07c002e0ee6e@mail.gmail.com>

Hi,

Thanks for replying. The advise u gave resolved my problem. I want to ask
three questions related to nibble bills, as I 'm listing down below;

1- Can we select/use dynamic tables for billing using nibble bill?
2- Can we define more than two tables and attributes in nibblebill.conf.xml?
3- As Nibble bill is use to deduct amount of user account, Can we deduct
minutes instead of cash? Because my case is, if a user buy a package and I
only want to deducts his/her minutes. How we can resolve it by nibble bill?
/ What other way we can resolve it?

Kindly advise soon.



>
> ---------- Forwarded message ----------
> From: Todd Baumgartner <todd.baumgartner at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Tue, 6 Oct 2009 10:31:06 -0400
> Subject: Re: [Freeswitch-users] Cannot connect to ODBC driver/database
> freeswitchdb
> Ahmed,
>
> I believe you need to specify the database name as it is configured in the
> odbc.ini
>
> I am assuming you have something like this in your nibblebill.conf.xml
>
>     <param name="db_dsn" value="freeswitchdb"/>
>
> Try changing it to this as it is named in the odbc.ini:
>
>     <param name="db_dsn" value="MySQL-freeswitch"/>
>
>
> Thanks,
>
> Todd
>
>
>
> On Tue, Oct 6, 2009 at 6:11 AM, Ahmed Munir <ahmedmunir007 at gmail.com>wrote:
>
>> Hi,
>> I've installed FS on Ubuntu 9.04 and I want to run mod_nibbles on it. I
>> follow the steps to configure my ODBC connection with MySQL as explained in
>> wiki (mod_nibbles and mod_spidermonkey). But  FS, unable to connect it. The
>> error I got is listed below when I restart FS,
>>
>> 2009-10-06 15:47:21.164590 [ERR] switch_odbc.c:188 STATE: IM002 CODE 0
>> ERROR: [unixODBC][Driver Manager]Data source name not found, and no default
>> driver specified
>> 2009-10-06 15:47:21.164617 [CRIT] mod_nibblebill.c:221 Cannot connect to
>> ODBC driver/database freeswitchdb (user: root / pass password)
>> 2009-10-06 15:47:21.164650 [CONSOLE] switch_loadable_module.c:889
>> Successfully Loaded [mod_nibblebill]
>> 2009-10-06 15:47:21.164664 [NOTICE] switch_loadable_module.c:248 Adding
>> Application 'nibblebill'
>> 2009-10-06 15:47:21.164710 [NOTICE] switch_loadable_module.c:270 Adding
>> API Function 'nibblebill'
>>
>> But when I use isql it accepts my odbc connection i.e. isql
>> MySQL-freeswitch
>>
>> I'm listing my settings of odbc.ini and odbcinst.ini as listed below;
>>
>> odbc.ini
>> --------------
>> [MySQL-freeswitch]
>> Driver       = MySQL
>> #Driver       = /usr/lib/odbc/libodbcmyS.so
>> Description  = Connector/ODBC Driver DSN With FreeSwitch
>> SERVER       = localhost
>> PORT         = 3306
>> USER         = root
>> Password     = password
>> Database     = freeswitchdb
>>
>> odbcinst.ini
>> -------------------
>> [MySQL]
>> Description    = ODBC for MySQL
>> Driver         = /usr/lib/odbc/libmyodbc.so
>> Setup          = /usr/lib/odbc/libodbcmyS.so
>> FileUsage      = 1
>>
>>
>> odbc.ini and odbcinst.ini are located at /etc/. Even I set my odbc
>> connection setting as I provide with this link;
>> http://dev.mysql.com/doc/refman/5.0/en/connector-odbc-configuration-dsn-unix.html
>>
>> But unfortunately my problem is unresolved then.
>>
>>
>> Kindly advise me, how can I resolve this problem?
>>
>> --
>> Regards,
>>
>> Ahmed Munir
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,

Ahmed Munir
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From maciej.aniserowicz at gmail.com  Wed Oct  7 00:39:52 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Wed, 7 Oct 2009 00:39:52 -0700 (PDT)
Subject: [Freeswitch-users] Bad sound quality while eavesdropping
In-Reply-To: <191c3a030910050731m2d74979ep4598e5a1945d58ae@mail.gmail.com>
References: <41A44DD027064988A914974405788C2E@procent>
	<191c3a030910050731m2d74979ep4598e5a1945d58ae@mail.gmail.com>
Message-ID: <1254901192035-3780245.post@n2.nabble.com>


Sorry about posting several questions at once, I wasn't aware it's "rude". 
Let's concentrate on this issue then. 

I use FS rev 14994. Phones on extensions: 
1) x-lite 
2) cisco sip phone 
3) audio played by fs to the extension being eavesdropped 

I did not change any codec configuration, I just use the standard one that
comes with both FS and the phones. 
Some time ago someone on FS irc channel told me that this is just how FS
eavesdropping works... from your response I understand that this is not
entirely true? 

Maciej Aniserowicz



Anthony Minessale wrote:
> 
> That's is a somewhat vague position.
> 
> You did not mention which version of FreeSWITCH you are running, the
> phones
> being used in your example, your configuration, the codecs in use etc.
> 
> BTW,
> I think you should only ask one question at a time on this list.  The list
> is run by volunteers and it's sort of rude to expect 3 or 4 threads to be
> tended to concerning the same one individual.
> 
> 
> 2009/10/5 Maciej Aniserowicz <maciej.aniserowicz at gmail.com>
> 
>>  Hello,
>> When I use eavesdropping in FreeSWITCH, the sound quality is really bad.
>> Is
>> there any way to improve it? Is this a known problem?
>> Br/
>> Maciej Aniserowicz
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3780245.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From claudiu at globtel.ro  Wed Oct  7 05:40:57 2009
From: claudiu at globtel.ro (Claudiu Filip)
Date: Wed, 7 Oct 2009 15:40:57 +0300
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910070137k52be89e3g846e55db72a8de5d@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<86a32abc0910061722t67ff23ddr62e4f62365539bd4@mail.gmail.com>
	<191c3a030910061758r2ddea5fand388f0af2c4dfc33@mail.gmail.com>
	<65d96fc80910070137k52be89e3g846e55db72a8de5d@mail.gmail.com>
Message-ID: <14480284.20091007154057@globtel.ro>




Hi Tihomir,


   I've  done  some  tests  to  see  how  suitable  is freeswitch as a
   SIP/H323  translator  and  you  are  right about the fact that H323
   'alert+open  logical  channel'  will generate a SIP '200 OK'. I was
   able  to fix that with a couple of changes in mod_opal.cpp, however
   some  things  were changed on mod_sofia in the latest svn. (on this
   particular  issue,  open_logical_channel  is  processed  BEFORE the
   alerting,   so   the   call   is   in   SetupPhase  when  the  proc
   OnOpenMediaStream is triggered)
   
   
   The  most  important  problem  I'm having right now is that G729 is
   still  not  working (poor quality due to high buffering). Even with
   the  latest  Opal,  which  includes  the last week patch for jitter
   buffer.  If  you  dont  need G729, I could send you a patch for the
   latest svns (freeswitch, opal, ptlib), ofc no founding needed.
   
   There are a couple of bugs in opal itself and h323ing freeswitch
   with opal will bring them in. On the other hand, mod_opal is
   already there, it just needs a few adjustments.

   
Best wishes,

Claudiu Filip




Wednesday, October 7, 2009, 11:37:50 AM, you wrote:
Tihomir> Anthony, 

Tihomir> of course, nobody wants to start anything... we are all here
Tihomir> to help making FS a better product.

Tihomir> so, regarding the founding for mod_opal ... what is the amount you need?


Tihomir> Tihomir.


Tihomir>  
Tihomir>  On Wed, Oct 7, 2009 at 2:58 AM, Anthony Minessale
Tihomir> <anthony.minessale at gmail.com> wrote:
Tihomir>  I didn't mean to start anything.? I'm just saying we work
Tihomir> very long hours and barely get anybody asking about h.323.
Tihomir> I wanted to support it and that's why we took up a
Tihomir> collection to get funding for mod_opal but when only 1 donor
Tihomir> showed any interest we were forced to proceed in our spare time which is very limited.
Tihomir>   
Tihomir> The developers of opal are not part of our project and they
Tihomir> need financial compensation to be motivated to work on it.?
Tihomir> Its not even related to me its only fair that an outside
Tihomir> developer who makes his living as a consultant would want
Tihomir> money to integrate his work into our project.
Tihomir>   
Tihomir> Like I said, I will do my best to point your issue to the
Tihomir> opal devs but I cannot force them to work on it.





Tihomir> On Tue, Oct 6, 2009 at 7:22 PM, Diego Viola
Tihomir> <diego.viola at gmail.com> wrote:
Tihomir>   
Tihomir> Yeah I understand your point of view, but saying "I want a
Tihomir> H.323 module" or "I want a Ferrari" wont magically make it happen.
Tihomir>   
Tihomir> We need to work on it ourselves or pay to the people that
Tihomir> knows how to do it to do it for us.
Tihomir>  
Tihomir> There is no other way I think.

Tihomir> Diego




Tihomir> On Tue, Oct 6, 2009 at 11:41 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
Tihomir>   
Tihomir> Diego,

Tihomir> what i'm pointing here is the situation where you have a
Tihomir> great product that lacks in one of most common protocol. It
Tihomir> is true H323 is going to disappear (eventually), it is true
Tihomir> that the community prefers SIP/IAX instead ... but the
Tihomir> reality still remains. H323 is going to be used for quite a
Tihomir> long time to exchange a lot of traffic while FS will be left
Tihomir> aside. Today, when you setup an IP peering interconnection
Tihomir> 80% of carriers will prefer H323.
Tihomir>   
Tihomir> Of course, developing something costs "time" (and we all
Tihomir> know what time stands for...) and as i said, i understand the
Tihomir> financial point of view and i really understand if nobody is
Tihomir> going to work on that, but let's face it FS doesn't have any
Tihomir> usable module to reliably handle H323 protocol. 
Tihomir>   

Tihomir> said that, i don't intend to offend anyone... just facing the reality.


Tihomir> regarding the h323 module, we don't have any issue fixing
Tihomir> the existing or developing a new one... but before we go
Tihomir> developing something it is always better check if the thing
Tihomir> you want already exists in an usable state or not... that's what i did today.
Tihomir>   

Tihomir> So, I'm interested in a reliable module handling H323v4... anyone else?


Tihomir> T.






Tihomir> On Tue, Oct 6, 2009 at 11:45 PM, Diego Viola
Tihomir> <diego.viola at gmail.com> wrote:
Tihomir>   
Tihomir> Instead of complaining and demanding things for free, people
Tihomir> should start to put their money where their mouth is.
Tihomir>   
Tihomir> Diego


Tihomir> On Tue, Oct 6, 2009 at 8:47 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
Tihomir>   
Tihomir> hi Anthony,

Tihomir> it is somewhere here:

Tihomir> ? ? ? ?? switch_status_t
Tihomir> FSConnection::receive_message(switch_core_session_message_t *msg)
Tihomir>   

Tihomir> anyhow, i will open an issue jira of course.


Tihomir> I understand your financial point of view, but anyhow while
Tihomir> the entire world is wants sip and trying to move to sip, the
Tihomir> reality is quite different. The majority of voice traffic
Tihomir> exchanged via IP is still H323. This means a working SIP -
Tihomir> H323 interworking is really needed... pity nobody wants/has
Tihomir> time to work in this direction to produce a decent mod_h323. 
Tihomir>    


Tihomir> T.



Tihomir> ?


Tihomir> On Tue, Oct 6, 2009 at 9:25 PM, Anthony Minessale
Tihomir> <anthony.minessale at gmail.com> wrote:
Tihomir>   
Tihomir> pcap is not as useful as FS console log on debug with:
Tihomir> sofia profile internal siptrace on
Tihomir>   
Tihomir> you should be reporting issues to jira under mod_opal not to the mailing list.
Tihomir> http://jira.freeswitch.org
Tihomir>  
Tihomir> FYI
Tihomir> There is little financial support from the community for
Tihomir> h323 which prevents the mod_opal from getting much attention.
Tihomir> We actually have to contract the author of opal to help with
Tihomir> these issues including the original writing of the module
Tihomir> that he did with very little funding and nobody ever wants to pay him to improve it.
Tihomir>   
Tihomir> That does not mean your issue will not be addressed but
Tihomir> there is no promise how fast it will be.



Tihomir> On Tue, Oct 6, 2009 at 12:37 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
Tihomir>   
Tihomir> hello guys,


Tihomir>  i was playing with mod_opal to see if i can make it working
Tihomir> ... well it seems SIP-H323 interworking is not tuned at all.
Tihomir>  
Tihomir> I have a call from a registered sip user (1001) to PSTN via mod_opal
Tihomir>   

Tihomir> <include>
Tihomir> ? <extension name="EMERGENCY">
Tihomir> ??? <condition field="destination_number"
Tihomir> expression="^0(112|9[23456])$">
Tihomir> ????? <action application="set"
Tihomir> data="effective_caller_id_number=1001282122"/>
Tihomir>  ????? <action application="set" data="NCX_IP=10.4.4.254"/>
Tihomir> ????? <action application="set" data="call_timeout=30"/>
Tihomir> ????? <action application="set" data="hangup_after_bridge=true"/>
Tihomir>   
Tihomir> ????? <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
Tihomir> ??? </condition>
Tihomir> ? </extension>

Tihomir> ? <extension name="SPECIAL_SERVICES">
Tihomir> ??? <condition field="destination_number"
Tihomir> expression="^0(9[01789]\d{3,4})$">
Tihomir>  ????? <action application="set"
Tihomir> data="effective_caller_id_number=1001282122"/>
Tihomir> ????? <action application="set" data="NCX_IP=10.4.4.254"/>
Tihomir> ????? <action application="set" data="call_timeout=30"/>
Tihomir>  ????? <action application="set" data="hangup_after_bridge=true"/>

Tihomir> ????? <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
Tihomir> ??? </condition>
Tihomir>  ? </extension>

Tihomir> ? <extension name="ENYTHING_ELSE">
Tihomir> ??? <condition field="destination_number"
Tihomir> expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
Tihomir> ????? <action application="set"
Tihomir> data="effective_caller_id_number=1001282122"/>
Tihomir>  ????? <action application="set" data="NCX_IP=10.4.4.254"/>
Tihomir> ????? <action application="set" data="call_timeout=30"/>
Tihomir> ????? <action application="set" data="hangup_after_bridge=true"/>
Tihomir>   
Tihomir> ????? <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
Tihomir> ??? </condition>
Tihomir> ? </extension>
Tihomir> </include>



Tihomir> One of the many issues i sow is that FS connects the call on
Tihomir> SIP leg before it actually receives H.225 connect from H323
Tihomir> leg... as it is configured to send 200 OK on the 1st H.225
Tihomir> message containing a FastStart element/OLC.
Tihomir>   

Tihomir> Attached is the tcpdump i took on FS machine... just use
Tihomir> this filter: "h225 or h245 or q931 or sip"
Tihomir> Also, you can check the attac CDR,,,, this is an unanswered
Tihomir> call i placed to PSTN and FS billed it 23 seconds.
Tihomir>   


Tihomir> Can anyone tell where i can do correct SIP - H323 message mappings to avoid this?



Tihomir> T.


Tihomir>  

Tihomir>  
Tihomir> _______________________________________________
Tihomir>  FreeSWITCH-users mailing list
Tihomir>  FreeSWITCH-users at lists.freeswitch.org
Tihomir>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Tihomir> 
Tihomir> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Tihomir>  http://www.freeswitch.org
Tihomir>  




Tihomir> -- 
Tihomir> Anthony Minessale II

Tihomir> FreeSWITCH http://www.freeswitch.org/
Tihomir> ClueCon http://www.cluecon.com/
Tihomir>  Twitter: http://twitter.com/FreeSWITCH_wire

Tihomir> AIM: anthm
Tihomir> MSN:anthony_minessale at hotmail.com
Tihomir>  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
Tihomir>  IRC: irc.freenode.net #freeswitch

Tihomir> FreeSWITCH Developer Conference
Tihomir> sip:888 at conference.freeswitch.org
Tihomir>   iax:guest at conference.freeswitch.org/888
Tihomir>  googletalk:conf+888 at conference.freeswitch.org
Tihomir> pstn:213-799-1400
Tihomir>  
Tihomir> _______________________________________________
Tihomir>  FreeSWITCH-users mailing list
Tihomir>  FreeSWITCH-users at lists.freeswitch.org
Tihomir>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Tihomir> 
Tihomir> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Tihomir>  http://www.freeswitch.org
Tihomir>  


Tihomir>  

Tihomir> _______________________________________________
Tihomir>  FreeSWITCH-users mailing list
Tihomir>  FreeSWITCH-users at lists.freeswitch.org
Tihomir>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Tihomir> 
Tihomir> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Tihomir>  http://www.freeswitch.org
Tihomir>  


Tihomir>  

Tihomir> _______________________________________________
Tihomir>  FreeSWITCH-users mailing list
Tihomir>  FreeSWITCH-users at lists.freeswitch.org
Tihomir>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Tihomir> 
Tihomir> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Tihomir>  http://www.freeswitch.org
Tihomir>  


Tihomir>  

Tihomir> _______________________________________________
Tihomir>  FreeSWITCH-users mailing list
Tihomir>  FreeSWITCH-users at lists.freeswitch.org
Tihomir>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Tihomir> 
Tihomir> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Tihomir>  http://www.freeswitch.org
Tihomir>  


Tihomir>  

Tihomir> _______________________________________________
Tihomir>  FreeSWITCH-users mailing list
Tihomir>  FreeSWITCH-users at lists.freeswitch.org
Tihomir>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Tihomir> 
Tihomir> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Tihomir>  http://www.freeswitch.org
Tihomir>  







From bottleman at icf.org.ru  Wed Oct  7 05:50:47 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Wed, 7 Oct 2009 16:50:47 +0400 (MSD)
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>

On 2009-10-07 01:41 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

We are developing module to handle h323 proto now, we try to use mod_opal and try improve it, but no luck, 
there is many issues in libopal, and finaly we now move to h323plus library.

TC>Diego,
TC>
TC>what i'm pointing here is the situation where you have a great product that
TC>lacks in one of most common protocol. It is true H323 is going to disappear
TC>(eventually), it is true that the community prefers SIP/IAX instead ... but
TC>the reality still remains. H323 is going to be used for quite a long time to
TC>exchange a lot of traffic while FS will be left aside. Today, when you setup
TC>an IP peering interconnection 80% of carriers will prefer H323.
TC>
TC>Of course, developing something costs "time" (and we all know what time
TC>stands for...) and as i said, i understand the financial point of view and i
TC>really understand if nobody is going to work on that, but let's face it FS
TC>doesn't have any usable module to reliably handle H323 protocol.
TC>
TC>
TC>said that, i don't intend to offend anyone... just facing the reality.
TC>
TC>
TC>regarding the h323 module, we don't have any issue fixing the existing or
TC>developing a new one... but before we go developing something it is always
TC>better check if the thing you want already exists in an usable state or
TC>not... that's what i did today.
TC>
TC>
TC>So, I'm interested in a reliable module handling H323v4... anyone else?
TC>
TC>
TC>T.
TC>
TC>
TC>
TC>
TC>
TC>On Tue, Oct 6, 2009 at 11:45 PM, Diego Viola <diego.viola at gmail.com> wrote:
TC>
TC>> Instead of complaining and demanding things for free, people should start
TC>> to put their money where their mouth is.
TC>>
TC>> Diego
TC>>
TC>>
TC>> On Tue, Oct 6, 2009 at 8:47 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
TC>>
TC>>> hi Anthony,
TC>>>
TC>>> it is somewhere here:
TC>>>
TC>>>          switch_status_t
TC>>> FSConnection::receive_message(switch_core_session_message_t *msg)
TC>>>
TC>>>
TC>>> anyhow, i will open an issue jira of course.
TC>>>
TC>>>
TC>>> I understand your financial point of view, but anyhow while the entire
TC>>> world is wants sip and trying to move to sip, the reality is quite
TC>>> different. The majority of voice traffic exchanged via IP is still H323.
TC>>> This means a working SIP - H323 interworking is really needed... pity nobody
TC>>> wants/has time to work in this direction to produce a decent mod_h323.
TC>>>
TC>>>
TC>>>
TC>>> T.
TC>>>
TC>>>
TC>>>
TC>>>
TC>>>
TC>>>
TC>>> On Tue, Oct 6, 2009 at 9:25 PM, Anthony Minessale <
TC>>> anthony.minessale at gmail.com> wrote:
TC>>>
TC>>>> pcap is not as useful as FS console log on debug with:
TC>>>> sofia profile internal siptrace on
TC>>>>
TC>>>> you should be reporting issues to jira under mod_opal not to the mailing
TC>>>> list.
TC>>>> http://jira.freeswitch.org
TC>>>>
TC>>>> FYI
TC>>>> There is little financial support from the community for h323 which
TC>>>> prevents the mod_opal from getting much attention.
TC>>>> We actually have to contract the author of opal to help with these issues
TC>>>> including the original writing of the module that he did with very little
TC>>>> funding and nobody ever wants to pay him to improve it.
TC>>>>
TC>>>> That does not mean your issue will not be addressed but there is no
TC>>>> promise how fast it will be.
TC>>>>
TC>>>>
TC>>>>
TC>>>> On Tue, Oct 6, 2009 at 12:37 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
TC>>>>
TC>>>>> hello guys,
TC>>>>>
TC>>>>>
TC>>>>> i was playing with mod_opal to see if i can make it working ... well it
TC>>>>> seems SIP-H323 interworking is not tuned at all.
TC>>>>>
TC>>>>> I have a call from a registered sip user (1001) to PSTN via mod_opal
TC>>>>>
TC>>>>>
TC>>>>> <include>
TC>>>>>   <extension name="EMERGENCY">
TC>>>>>     <condition field="destination_number"
TC>>>>> expression="^0(112|9[23456])$">
TC>>>>>       <action application="set"
TC>>>>> data="effective_caller_id_number=1001282122"/>
TC>>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
TC>>>>>       <action application="set" data="call_timeout=30"/>
TC>>>>>       <action application="set" data="hangup_after_bridge=true"/>
TC>>>>>
TC>>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
TC>>>>>     </condition>
TC>>>>>   </extension>
TC>>>>>
TC>>>>>   <extension name="SPECIAL_SERVICES">
TC>>>>>     <condition field="destination_number"
TC>>>>> expression="^0(9[01789]\d{3,4})$">
TC>>>>>       <action application="set"
TC>>>>> data="effective_caller_id_number=1001282122"/>
TC>>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
TC>>>>>       <action application="set" data="call_timeout=30"/>
TC>>>>>       <action application="set" data="hangup_after_bridge=true"/>
TC>>>>>
TC>>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
TC>>>>>     </condition>
TC>>>>>   </extension>
TC>>>>>
TC>>>>>   <extension name="ENYTHING_ELSE">
TC>>>>>     <condition field="destination_number"
TC>>>>> expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
TC>>>>>       <action application="set"
TC>>>>> data="effective_caller_id_number=1001282122"/>
TC>>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
TC>>>>>       <action application="set" data="call_timeout=30"/>
TC>>>>>       <action application="set" data="hangup_after_bridge=true"/>
TC>>>>>
TC>>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
TC>>>>>     </condition>
TC>>>>>   </extension>
TC>>>>> </include>
TC>>>>>
TC>>>>>
TC>>>>>
TC>>>>> One of the many issues i sow is that FS connects the call on SIP leg
TC>>>>> before it actually receives H.225 connect from H323 leg... as it is
TC>>>>> configured to send 200 OK on the 1st H.225 message containing a FastStart
TC>>>>> element/OLC.
TC>>>>>
TC>>>>>
TC>>>>> Attached is the tcpdump i took on FS machine... just use this filter:
TC>>>>> "h225 or h245 or q931 or sip"
TC>>>>> Also, you can check the attac CDR,,,, this is an unanswered call i
TC>>>>> placed to PSTN and FS billed it 23 seconds.
TC>>>>>
TC>>>>>
TC>>>>>
TC>>>>> Can anyone tell where i can do correct SIP - H323 message mappings to
TC>>>>> avoid this?
TC>>>>>
TC>>>>>
TC>>>>>
TC>>>>> T.
TC>>>>>
TC>>>>>
TC>>>>>
TC>>>>>
TC>>>>> _______________________________________________
TC>>>>> FreeSWITCH-users mailing list
TC>>>>> FreeSWITCH-users at lists.freeswitch.org
TC>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>>>>> UNSUBSCRIBE:
TC>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>>>>> http://www.freeswitch.org
TC>>>>>
TC>>>>>
TC>>>>
TC>>>>
TC>>>> --
TC>>>> Anthony Minessale II
TC>>>>
TC>>>> FreeSWITCH http://www.freeswitch.org/
TC>>>> ClueCon http://www.cluecon.com/
TC>>>> Twitter: http://twitter.com/FreeSWITCH_wire
TC>>>>
TC>>>> AIM: anthm
TC>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
TC>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
TC>>>> IRC: irc.freenode.net #freeswitch
TC>>>>
TC>>>> FreeSWITCH Developer Conference
TC>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
TC>>>> iax:guest at conference.freeswitch.org/888
TC>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
TC>>>> pstn:213-799-1400
TC>>>>
TC>>>> _______________________________________________
TC>>>> FreeSWITCH-users mailing list
TC>>>> FreeSWITCH-users at lists.freeswitch.org
TC>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>>>> http://www.freeswitch.org
TC>>>>
TC>>>>
TC>>>
TC>>> _______________________________________________
TC>>> FreeSWITCH-users mailing list
TC>>> FreeSWITCH-users at lists.freeswitch.org
TC>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>>> http://www.freeswitch.org
TC>>>
TC>>>
TC>>
TC>> _______________________________________________
TC>> FreeSWITCH-users mailing list
TC>> FreeSWITCH-users at lists.freeswitch.org
TC>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> http://www.freeswitch.org
TC>>
TC>>
TC>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From mike at jerris.com  Wed Oct  7 06:06:26 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 7 Oct 2009 09:06:26 -0400
Subject: [Freeswitch-users] Bad sound quality while eavesdropping
In-Reply-To: <1254901192035-3780245.post@n2.nabble.com>
References: <41A44DD027064988A914974405788C2E@procent>
	<191c3a030910050731m2d74979ep4598e5a1945d58ae@mail.gmail.com>
	<1254901192035-3780245.post@n2.nabble.com>
Message-ID: <8437F5BC-7AFF-4A74-B8CD-C5B8219021F6@jerris.com>

What codecs are all the call legs using, also, please try current svn  
trunk.

Mike

On Oct 7, 2009, at 3:39 AM, Maciej Aniserowicz wrote:

>
> Sorry about posting several questions at once, I wasn't aware it's  
> "rude".
> Let's concentrate on this issue then.
>
> I use FS rev 14994. Phones on extensions:
> 1) x-lite
> 2) cisco sip phone
> 3) audio played by fs to the extension being eavesdropped
>
> I did not change any codec configuration, I just use the standard  
> one that
> comes with both FS and the phones.
> Some time ago someone on FS irc channel told me that this is just  
> how FS
> eavesdropping works... from your response I understand that this is  
> not
> entirely true?
>
> Maciej Aniserowicz
>
>
>
> Anthony Minessale wrote:
>>
>> That's is a somewhat vague position.
>>
>> You did not mention which version of FreeSWITCH you are running, the
>> phones
>> being used in your example, your configuration, the codecs in use  
>> etc.
>>
>> BTW,
>> I think you should only ask one question at a time on this list.   
>> The list
>> is run by volunteers and it's sort of rude to expect 3 or 4 threads  
>> to be
>> tended to concerning the same one individual.
>>
>>
>> 2009/10/5 Maciej Aniserowicz <maciej.aniserowicz at gmail.com>
>>
>>> Hello,
>>> When I use eavesdropping in FreeSWITCH, the sound quality is  
>>> really bad.
>>> Is
>>> there any way to improve it? Is this a known problem?
>>> Br/
>>> Maciej Aniserowicz
>>>



From andy at fabulous4.co.uk  Wed Oct  7 06:11:24 2009
From: andy at fabulous4.co.uk (Andy)
Date: Wed, 7 Oct 2009 14:11:24 +0100
Subject: [Freeswitch-users] NAT problems - sorry
Message-ID: <5BE289D2AE22492BA71A6919C8A440B3@D810>

Hi folks,
 
I know this problem comes up all the time so sorry to bring it up again but
I can't seem to find the answer in previous posts.
 
I have my freeswitch installation behind a DLink firewall so the freeswitch
server is natted. I have added what I believe are all the necessary rules to
the firewall to handle SIP correctly and the firewall isn't showing any
blocked traffic when calls come in. Calls are forwarded to my public ip
address from my VOIP provider Voiptalk. Here's the problem:
 
Incoming calls are successfully forwarded to the freeswitch server and
answered.
I can hear the IVR message played down the phone line so outgoing audio is
ok.
When I try to send DTMF key presses from my phone none of these register on
the switch.
When I hangup the call the call hangup is also not picked up by the switch.
 
I have followed the auto-nat instrcutions and have verified that the switch
correctly grabs and sets the external ip address. I have also tried setting
the external ip address manually but without success.
 
Can anyone offer any guidance as to what might be causing this problem or
what I could try to resolve it? I can post log/config snippets if that
helps.
 
Many thanks
Andy
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From claudiu at globtel.ro  Wed Oct  7 06:14:40 2009
From: claudiu at globtel.ro (Claudiu Filip)
Date: Wed, 7 Oct 2009 16:14:40 +0300
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <191c3a030910061758r2ddea5fand388f0af2c4dfc33@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<86a32abc0910061722t67ff23ddr62e4f62365539bd4@mail.gmail.com>
	<191c3a030910061758r2ddea5fand388f0af2c4dfc33@mail.gmail.com>
Message-ID: <716784809.20091007161440@globtel.ro>


Hi,

Wednesday, October 7, 2009, 3:58:20 AM, Anthony M. wrote:
> barely get anybody asking about h.323.


H323 may not be popular for small ITSPs or small/medium PBXes, but
it's widely used by the big players.. and freeswitch doesnt share the
same goals with asterisk.



Best wishes,


Claudiu Filip





From chris.chen2004 at gmail.com  Wed Oct  7 06:15:31 2009
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Wed, 7 Oct 2009 09:15:31 -0400
Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Followup,
	Agenda For This Week
In-Reply-To: <b16156850910062124h4bae532enc3624e18d1b20b2f@mail.gmail.com>
References: <87f2f3b90910051203v7bdbca5amfc82d193f3a4f1a3@mail.gmail.com>
	<B5ECAE34-B846-4408-8AC1-6D94ADCC672C@freeswitch.org>
	<b16156850910062124h4bae532enc3624e18d1b20b2f@mail.gmail.com>
Message-ID: <507898380910070615k9fb3c9ha2d040e31c0035ad@mail.gmail.com>

Hi Muhammad, the simple and reliable solution for you where SIP is being
blocked is add conf+888 at conference.freeswitch.org<conf%2B888 at conference.freeswitch.org>to
your Goolgetalk buddy list, and you can call from there to join the
conference, simple and straightforward.

Chris

On Wed, Oct 7, 2009 at 12:24 AM, Muhammad Shahzad <
shaheryarkh at googlemail.com> wrote:

> I got a lot of problem last week for making conference call. I was at home
> (conference call starts at 2200hours PKST, my time) and unable to make SIP
> call since the government has blocked it. So my only choice was Skype, but
> unfortunately DTMF wasn't working, i get connected on Skypiax5 for about 7-8
> times, but DTMF problem never let me get in to conference. Is there any
> solution for this?
>
> I really need to discuss a lot of things about FS documentation in
> conference, like what FS community can expect from it and what not? how i
> have planned it? how much progress has done? what are the problems me and my
> team are facing (which has slow us down considerably)? etc. etc.
>
> I have a solution for it but that needs testing. The plan is to use one of
> my FS servers to connect my jingle calls from GTalk to conference server
> over SIP. How can i test this setup with conference server, any ideas?
>
> If any one else also interested in getting connected to weekly conference
> call through this setup then i can also extend this setup as needed.
>
> Thank you.
>
>
>
> On Tue, Oct 6, 2009 at 1:30 AM, Brian West <brian at freeswitch.org> wrote:
>
>> It always supported 48kHz CELT but the conference itself was running
>> at 32kHz so everyone 48k had to be down sampled.  Now you all get to
>> be up sampled.  w00t!
>>
>> /b
>>
>> On Oct 5, 2009, at 2:03 PM, Michael Collins wrote:
>>
>> > * Starting with the upcoming meeting (Oct 9) the conference will
>> > support 48kHz CELT codec.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> ________________________________________________________
> |
>                      |
> | FATAL ERROR                                                           ---
> O X |
> |_______________________________________________________|
> |                        You have moved the mouse.
>    |
> | Windows must be restarted for the changes to take effect.   |
> |                                        <OK>
>                |
> ####################################/
>
>
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Wed Oct  7 06:21:51 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 7 Oct 2009 09:21:51 -0400
Subject: [Freeswitch-users] Recording creates a 388-byte long file and
	deletes it
In-Reply-To: <4ED3AB65AFE34242AACDE97127FE1248@procent>
References: <4ED3AB65AFE34242AACDE97127FE1248@procent>
Message-ID: <74D80E7B-119B-4CC1-82DC-9A370F940BA5@jerris.com>

switch_ivr_async.c:480

On Oct 5, 2009, at 3:16 AM, Maciej Aniserowicz wrote:

> Hi,
> When I record a call in FS, it only creates a 388-byte-long wav  
> file. The conversation is no written there, and FS deletes the file  
> when the session finishes.
> What can cause this strange behavior?

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From mike at jerris.com  Wed Oct  7 06:23:33 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 7 Oct 2009 09:23:33 -0400
Subject: [Freeswitch-users] gateway FS informs it's client FS about
	users hanged up with a long delay
In-Reply-To: <1A16501E57CA4727A593226A8C810308@procent>
References: <1A16501E57CA4727A593226A8C810308@procent>
Message-ID: <98F72CCB-88B9-47ED-AED3-B6BA6DE648C0@jerris.com>

Incorrect NAT configuration so one of the boxes is not actually  
getting a BYE.


On Oct 5, 2009, at 3:13 AM, Maciej Aniserowicz wrote:

> Hi,
> When I use two FreeSWITCH instances ('internal' and 'external'), all  
> users register to the 'external' instance which acts as a gateway by  
> 'internal' instance (which in turn is controlled by my applicaiton  
> with commands sent by socket).
> When user hangs up, the 'hanged up' event is propagated to the  
> 'internal' instance after a long time (~3 minutes) instead of being  
> propagated immediately.
> What can cause this issue?

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From mike at jerris.com  Wed Oct  7 06:26:31 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 7 Oct 2009 09:26:31 -0400
Subject: [Freeswitch-users] Bridge application with shared lines
In-Reply-To: <d361ca7b0910060114g3f23a102nb3508cf6e1e196fc@mail.gmail.com>
References: <d361ca7b0910060114g3f23a102nb3508cf6e1e196fc@mail.gmail.com>
Message-ID: <7F6D4A24-DCC7-4367-9EEE-73968A42DE98@jerris.com>


On Oct 6, 2009, at 4:14 AM, Yehavi Bourvine wrote:

> Hello,
>
>   We have Polycom and SNOM phones running with FreeSwitch. The  
> Polycoms have shared lines defined and the SNOMs have both shared  
> lines and BLFs (defined as extensions in the phone config). I've  
> tried supporting both, but have some incompatibility:
>
> When calling the Bridge application with data parameter of sofia/ 
> profile-name/number at domain the BLF works ok, but not the shared  
> lines (i.e only one of the phones  rings).
> When calling the Bridge application with data parameter of $ 
> {sofia_contact(/profile-name/number at domain)} shared lines work ok  
> but BLF doesn't fire up.
>
> How do I support both? Is there a way to know whether the  
> destination is a shared one and then chose one of the above formats?

You should probably always be using the second method or using a %  
instead of the @ in the first method to get the registered contact.   
can you provide more information about why the BLF "doesn't fire up" .

Mike

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From shaheryarkh at googlemail.com  Wed Oct  7 06:37:52 2009
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Wed, 7 Oct 2009 18:37:52 +0500
Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Followup,
	Agenda For This Week
In-Reply-To: <507898380910070615k9fb3c9ha2d040e31c0035ad@mail.gmail.com>
References: <87f2f3b90910051203v7bdbca5amfc82d193f3a4f1a3@mail.gmail.com>
	<B5ECAE34-B846-4408-8AC1-6D94ADCC672C@freeswitch.org>
	<b16156850910062124h4bae532enc3624e18d1b20b2f@mail.gmail.com>
	<507898380910070615k9fb3c9ha2d040e31c0035ad@mail.gmail.com>
Message-ID: <b16156850910070637w7c2b1f27s33ee87949f719d45@mail.gmail.com>

Great. Just added it. Is there any user limit on this?

Thank you.

On Wed, Oct 7, 2009 at 6:15 PM, Chris Chen <chris.chen2004 at gmail.com> wrote:

> Hi Muhammad, the simple and reliable solution for you where SIP is being
> blocked is add conf+888 at conference.freeswitch.org<conf%2B888 at conference.freeswitch.org>to your Goolgetalk buddy list, and you can call from there to join the
> conference, simple and straightforward.
>
> Chris
>
>
> On Wed, Oct 7, 2009 at 12:24 AM, Muhammad Shahzad <
> shaheryarkh at googlemail.com> wrote:
>
>> I got a lot of problem last week for making conference call. I was at home
>> (conference call starts at 2200hours PKST, my time) and unable to make SIP
>> call since the government has blocked it. So my only choice was Skype, but
>> unfortunately DTMF wasn't working, i get connected on Skypiax5 for about 7-8
>> times, but DTMF problem never let me get in to conference. Is there any
>> solution for this?
>>
>> I really need to discuss a lot of things about FS documentation in
>> conference, like what FS community can expect from it and what not? how i
>> have planned it? how much progress has done? what are the problems me and my
>> team are facing (which has slow us down considerably)? etc. etc.
>>
>> I have a solution for it but that needs testing. The plan is to use one of
>> my FS servers to connect my jingle calls from GTalk to conference server
>> over SIP. How can i test this setup with conference server, any ideas?
>>
>> If any one else also interested in getting connected to weekly conference
>> call through this setup then i can also extend this setup as needed.
>>
>> Thank you.
>>
>>
>>
>> On Tue, Oct 6, 2009 at 1:30 AM, Brian West <brian at freeswitch.org> wrote:
>>
>>> It always supported 48kHz CELT but the conference itself was running
>>> at 32kHz so everyone 48k had to be down sampled.  Now you all get to
>>> be up sampled.  w00t!
>>>
>>> /b
>>>
>>> On Oct 5, 2009, at 2:03 PM, Michael Collins wrote:
>>>
>>> > * Starting with the upcoming meeting (Oct 9) the conference will
>>> > support 48kHz CELT codec.
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> ________________________________________________________
>> |
>>                        |
>> | FATAL ERROR
>> --- O X |
>> |_______________________________________________________|
>> |                        You have moved the mouse.
>>    |
>> | Windows must be restarted for the changes to take effect.   |
>> |                                        <OK>
>>                  |
>> ####################################/
>>
>>
>> Muhammad Shahzad
>> -----------------------------------
>> CISCO Rich Media Communication Specialist (CRMCS)
>> CISCO Certified Network Associate (CCNA)
>> Cell: +92 334 422 40 88
>> MSN: shari_786pk at hotmail.com
>> Email: shaheryarkh at googlemail.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
________________________________________________________
|
                     |
| FATAL ERROR                                                           ---
O X |
|_______________________________________________________|
|                        You have moved the mouse.
 |
| Windows must be restarted for the changes to take effect.   |
|                                        <OK>
               |
####################################/


Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From nicolas at medularis.com  Wed Oct  7 06:48:59 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Wed, 7 Oct 2009 09:48:59 -0400
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <9106C8E1-498F-4AE8-8D34-7F6AAD4DC8AE@freeswitch.org>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<1b46b4e80910041519h3581d3d2x7b84041e8c18588@mail.gmail.com>
	<C887D143-F7A1-44B9-B8A2-D34A4B4E582E@jerris.com>
	<1b46b4e80910041609x2f41ceb9u142cb484951a21ed@mail.gmail.com>
	<87f2f3b90910050920w4d17033fl4c386a0018107bdf@mail.gmail.com>
	<1b46b4e80910052119me12a0d9y4653edd831f9555e@mail.gmail.com>
	<1b46b4e80910052142jaa09f08l72e5e4b2b24d10fc@mail.gmail.com>
	<083DBCB9-48B2-4E61-81B5-B7D00A42EAB5@freeswitch.org>
	<1b46b4e80910061422q7be2a21fg5d2caaef7e307abb@mail.gmail.com>
	<9106C8E1-498F-4AE8-8D34-7F6AAD4DC8AE@freeswitch.org>
Message-ID: <1b46b4e80910070648n622c3f9haacc22c2328b44ff@mail.gmail.com>

Is there some way to make FS register with the gateway that is rejecting the
authentication? is it FS or the SIP server at fault? Why would X-Lite work
and FS not?

Thanks again for your time and help.


On Tue, Oct 6, 2009 at 5:46 PM, Brian West <brian at freeswitch.org> wrote:

> btw My mistake it doesn't assume auth it just calculates the response
> hash differently on this case where qop isn't present.
>
> /b
>
> On Oct 6, 2009, at 4:22 PM, Nicolas Brenner wrote:
>
> >
> > What does the qop parameter stand for? Apparently because of that
> > parameter, FS sends a new REGISTER including this:
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Wed Oct  7 07:09:32 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 7 Oct 2009 09:09:32 -0500
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <716784809.20091007161440@globtel.ro>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<86a32abc0910061722t67ff23ddr62e4f62365539bd4@mail.gmail.com>
	<191c3a030910061758r2ddea5fand388f0af2c4dfc33@mail.gmail.com>
	<716784809.20091007161440@globtel.ro>
Message-ID: <191c3a030910070709r2f63cf1dm9f41fff66ab76d3@mail.gmail.com>

I am not commenting on how popular it is overall.
I am commenting on the specific demand presented to us.
I don't know what else to say to explain that I am completely neutral when
it comes to this topic.

One more time:

1) We are an open source project who volunteer most of our time as well as
paid commercial support.
2) We made mod_opal anyway despite only a small handful of requests.
3) The devs from OPAL are consultants for hire and like us, they will look
at what they can for free but you can hire them to heighten their attention
span.

This is not a discussion on the merits of h323, I am more than happy to
support it.  The OPAL guys are some of the best open source telephony guys
out there and they have been doing this for over a decade.  We Just can't
promise a whole lot of man hours towards it when we already work day and
night managing jira with only a few managers.  We also cannot make any
promises on the free time the OPAL devs have to devote to it either.




On Wed, Oct 7, 2009 at 8:14 AM, Claudiu Filip <claudiu at globtel.ro> wrote:

>
> Hi,
>
> Wednesday, October 7, 2009, 3:58:20 AM, Anthony M. wrote:
> > barely get anybody asking about h.323.
>
>
> H323 may not be popular for small ITSPs or small/medium PBXes, but
> it's widely used by the big players.. and freeswitch doesnt share the
> same goals with asterisk.
>
>
>
> Best wishes,
>
>
> Claudiu Filip
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Wed Oct  7 07:10:51 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 7 Oct 2009 09:10:51 -0500
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910070137k52be89e3g846e55db72a8de5d@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<86a32abc0910061722t67ff23ddr62e4f62365539bd4@mail.gmail.com>
	<191c3a030910061758r2ddea5fand388f0af2c4dfc33@mail.gmail.com>
	<65d96fc80910070137k52be89e3g846e55db72a8de5d@mail.gmail.com>
Message-ID: <191c3a030910070710s6722c2b5j4a7c6fd2b1bcc648@mail.gmail.com>

I think the way to determine the funding is to get all the most important
issues up on jira, try to deal with them and see if we need to put bounties
on any of them to get them done faster.


On Wed, Oct 7, 2009 at 3:37 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> Anthony,
>
> of course, nobody wants to start anything... we are all here to help making
> FS a better product.
>
> so, regarding the founding for mod_opal ... what is the amount you need?
>
>
> Tihomir.
>
>
>
>
> On Wed, Oct 7, 2009 at 2:58 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> I didn't mean to start anything.  I'm just saying we work very long hours
>> and barely get anybody asking about h.323.
>> I wanted to support it and that's why we took up a collection to get
>> funding for mod_opal but when only 1 donor showed any interest we were
>> forced to proceed in our spare time which is very limited.
>>
>> The developers of opal are not part of our project and they need financial
>> compensation to be motivated to work on it.  Its not even related to me its
>> only fair that an outside developer who makes his living as a consultant
>> would want money to integrate his work into our project.
>>
>> Like I said, I will do my best to point your issue to the opal devs but I
>> cannot force them to work on it.
>>
>>
>>
>>
>>
>> On Tue, Oct 6, 2009 at 7:22 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>
>>> Yeah I understand your point of view, but saying "I want a H.323 module"
>>> or "I want a Ferrari" wont magically make it happen.
>>>
>>> We need to work on it ourselves or pay to the people that knows how to do
>>> it to do it for us.
>>>
>>> There is no other way I think.
>>>
>>> Diego
>>>
>>>
>>>
>>>
>>> On Tue, Oct 6, 2009 at 11:41 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>>
>>>> Diego,
>>>>
>>>> what i'm pointing here is the situation where you have a great product
>>>> that lacks in one of most common protocol. It is true H323 is going to
>>>> disappear (eventually), it is true that the community prefers SIP/IAX
>>>> instead ... but the reality still remains. H323 is going to be used for
>>>> quite a long time to exchange a lot of traffic while FS will be left aside.
>>>> Today, when you setup an IP peering interconnection 80% of carriers will
>>>> prefer H323.
>>>>
>>>> Of course, developing something costs "time" (and we all know what time
>>>> stands for...) and as i said, i understand the financial point of view and i
>>>> really understand if nobody is going to work on that, but let's face it FS
>>>> doesn't have any usable module to reliably handle H323 protocol.
>>>>
>>>>
>>>> said that, i don't intend to offend anyone... just facing the reality.
>>>>
>>>>
>>>> regarding the h323 module, we don't have any issue fixing the existing
>>>> or developing a new one... but before we go developing something it is
>>>> always better check if the thing you want already exists in an usable state
>>>> or not... that's what i did today.
>>>>
>>>>
>>>> So, I'm interested in a reliable module handling H323v4... anyone else?
>>>>
>>>>
>>>> T.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Tue, Oct 6, 2009 at 11:45 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>>
>>>>> Instead of complaining and demanding things for free, people should
>>>>> start to put their money where their mouth is.
>>>>>
>>>>> Diego
>>>>>
>>>>>
>>>>> On Tue, Oct 6, 2009 at 8:47 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>>>>
>>>>>> hi Anthony,
>>>>>>
>>>>>> it is somewhere here:
>>>>>>
>>>>>>          switch_status_t
>>>>>> FSConnection::receive_message(switch_core_session_message_t *msg)
>>>>>>
>>>>>>
>>>>>> anyhow, i will open an issue jira of course.
>>>>>>
>>>>>>
>>>>>> I understand your financial point of view, but anyhow while the entire
>>>>>> world is wants sip and trying to move to sip, the reality is quite
>>>>>> different. The majority of voice traffic exchanged via IP is still H323.
>>>>>> This means a working SIP - H323 interworking is really needed... pity nobody
>>>>>> wants/has time to work in this direction to produce a decent mod_h323.
>>>>>>
>>>>>>
>>>>>>
>>>>>> T.
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Tue, Oct 6, 2009 at 9:25 PM, Anthony Minessale <
>>>>>> anthony.minessale at gmail.com> wrote:
>>>>>>
>>>>>>> pcap is not as useful as FS console log on debug with:
>>>>>>> sofia profile internal siptrace on
>>>>>>>
>>>>>>> you should be reporting issues to jira under mod_opal not to the
>>>>>>> mailing list.
>>>>>>> http://jira.freeswitch.org
>>>>>>>
>>>>>>> FYI
>>>>>>> There is little financial support from the community for h323 which
>>>>>>> prevents the mod_opal from getting much attention.
>>>>>>> We actually have to contract the author of opal to help with these
>>>>>>> issues including the original writing of the module that he did with very
>>>>>>> little funding and nobody ever wants to pay him to improve it.
>>>>>>>
>>>>>>> That does not mean your issue will not be addressed but there is no
>>>>>>> promise how fast it will be.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Tue, Oct 6, 2009 at 12:37 PM, Tihomir Culjaga <tculjaga at gmail.com
>>>>>>> > wrote:
>>>>>>>
>>>>>>>> hello guys,
>>>>>>>>
>>>>>>>>
>>>>>>>> i was playing with mod_opal to see if i can make it working ... well
>>>>>>>> it seems SIP-H323 interworking is not tuned at all.
>>>>>>>>
>>>>>>>> I have a call from a registered sip user (1001) to PSTN via mod_opal
>>>>>>>>
>>>>>>>>
>>>>>>>> <include>
>>>>>>>>   <extension name="EMERGENCY">
>>>>>>>>     <condition field="destination_number"
>>>>>>>> expression="^0(112|9[23456])$">
>>>>>>>>       <action application="set"
>>>>>>>> data="effective_caller_id_number=1001282122"/>
>>>>>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>>>>>>       <action application="set" data="call_timeout=30"/>
>>>>>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>>>>>
>>>>>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>>>>>>     </condition>
>>>>>>>>   </extension>
>>>>>>>>
>>>>>>>>   <extension name="SPECIAL_SERVICES">
>>>>>>>>     <condition field="destination_number"
>>>>>>>> expression="^0(9[01789]\d{3,4})$">
>>>>>>>>       <action application="set"
>>>>>>>> data="effective_caller_id_number=1001282122"/>
>>>>>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>>>>>>       <action application="set" data="call_timeout=30"/>
>>>>>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>>>>>
>>>>>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>>>>>>     </condition>
>>>>>>>>   </extension>
>>>>>>>>
>>>>>>>>   <extension name="ENYTHING_ELSE">
>>>>>>>>     <condition field="destination_number"
>>>>>>>> expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
>>>>>>>>       <action application="set"
>>>>>>>> data="effective_caller_id_number=1001282122"/>
>>>>>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>>>>>>       <action application="set" data="call_timeout=30"/>
>>>>>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>>>>>
>>>>>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>>>>>>     </condition>
>>>>>>>>   </extension>
>>>>>>>> </include>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> One of the many issues i sow is that FS connects the call on SIP leg
>>>>>>>> before it actually receives H.225 connect from H323 leg... as it is
>>>>>>>> configured to send 200 OK on the 1st H.225 message containing a FastStart
>>>>>>>> element/OLC.
>>>>>>>>
>>>>>>>>
>>>>>>>> Attached is the tcpdump i took on FS machine... just use this
>>>>>>>> filter: "h225 or h245 or q931 or sip"
>>>>>>>> Also, you can check the attac CDR,,,, this is an unanswered call i
>>>>>>>> placed to PSTN and FS billed it 23 seconds.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Can anyone tell where i can do correct SIP - H323 message mappings
>>>>>>>> to avoid this?
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> T.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Anthony Minessale II
>>>>>>>
>>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>>> ClueCon http://www.cluecon.com/
>>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>>
>>>>>>> AIM: anthm
>>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>>
>>>>>>> FreeSWITCH Developer Conference
>>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>>> pstn:213-799-1400
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From brian at freeswitch.org  Wed Oct  7 07:20:20 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 7 Oct 2009 09:20:20 -0500
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <1b46b4e80910070648n622c3f9haacc22c2328b44ff@mail.gmail.com>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<1b46b4e80910041519h3581d3d2x7b84041e8c18588@mail.gmail.com>
	<C887D143-F7A1-44B9-B8A2-D34A4B4E582E@jerris.com>
	<1b46b4e80910041609x2f41ceb9u142cb484951a21ed@mail.gmail.com>
	<87f2f3b90910050920w4d17033fl4c386a0018107bdf@mail.gmail.com>
	<1b46b4e80910052119me12a0d9y4653edd831f9555e@mail.gmail.com>
	<1b46b4e80910052142jaa09f08l72e5e4b2b24d10fc@mail.gmail.com>
	<083DBCB9-48B2-4E61-81B5-B7D00A42EAB5@freeswitch.org>
	<1b46b4e80910061422q7be2a21fg5d2caaef7e307abb@mail.gmail.com>
	<9106C8E1-498F-4AE8-8D34-7F6AAD4DC8AE@freeswitch.org>
	<1b46b4e80910070648n622c3f9haacc22c2328b44ff@mail.gmail.com>
Message-ID: <8750A70D-6891-4C07-B4DF-C7F4ABA58D67@freeswitch.org>

I would suspect its a PEBKAC.  I mean if you could register to a  
gateway that rejected auth... what purpose would auth serve in the  
first place?

/b

On Oct 7, 2009, at 8:48 AM, Nicolas Brenner wrote:

> Is there some way to make FS register with the gateway that is  
> rejecting the authentication? is it FS or the SIP server at fault?  
> Why would X-Lite work and FS not?
>
> Thanks again for your time and help.



From brian at freeswitch.org  Wed Oct  7 07:20:42 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 7 Oct 2009 09:20:42 -0500
Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Followup,
	Agenda For This Week
In-Reply-To: <b16156850910070637w7c2b1f27s33ee87949f719d45@mail.gmail.com>
References: <87f2f3b90910051203v7bdbca5amfc82d193f3a4f1a3@mail.gmail.com>
	<B5ECAE34-B846-4408-8AC1-6D94ADCC672C@freeswitch.org>
	<b16156850910062124h4bae532enc3624e18d1b20b2f@mail.gmail.com>
	<507898380910070615k9fb3c9ha2d040e31c0035ad@mail.gmail.com>
	<b16156850910070637w7c2b1f27s33ee87949f719d45@mail.gmail.com>
Message-ID: <9FF9A0D6-0D40-4C62-9244-6607172BAEA7@freeswitch.org>

No!

/b

On Oct 7, 2009, at 8:37 AM, Muhammad Shahzad wrote:

> Great. Just added it. Is there any user limit on this?
>
> Thank you.



From brian at freeswitch.org  Wed Oct  7 07:21:43 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 7 Oct 2009 09:21:43 -0500
Subject: [Freeswitch-users] NAT problems - sorry
In-Reply-To: <5BE289D2AE22492BA71A6919C8A440B3@D810>
References: <5BE289D2AE22492BA71A6919C8A440B3@D810>
Message-ID: <5A70A0D4-70FA-43B2-BAF5-F8D6ECC64838@freeswitch.org>

Bet you its inband dtmf and you need to start the dtmf detector.

/b

On Oct 7, 2009, at 8:11 AM, Andy wrote:

> I can hear the IVR message played down the phone line so outgoing  
> audio is ok.

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From brian at freeswitch.org  Wed Oct  7 07:24:00 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 7 Oct 2009 09:24:00 -0500
Subject: [Freeswitch-users] Bridge application with shared lines
In-Reply-To: <7F6D4A24-DCC7-4367-9EEE-73968A42DE98@jerris.com>
References: <d361ca7b0910060114g3f23a102nb3508cf6e1e196fc@mail.gmail.com>
	<7F6D4A24-DCC7-4367-9EEE-73968A42DE98@jerris.com>
Message-ID: <09EC098B-3883-4FF3-B7F2-08360E39D374@freeswitch.org>

You'll need to set presence_id so it can work properly. SEE the  
default config that does exactly that with sofia_contact in the dial- 
string on the domain.

/b

On Oct 7, 2009, at 8:26 AM, Michael Jerris wrote:

>> When calling the Bridge application with data parameter of sofia/ 
>> profile-name/number at domain the BLF works ok, but not the shared  
>> lines (i.e only one of the phones  rings).
>> When calling the Bridge application with data parameter of $ 
>> {sofia_contact(/profile-name/number at domain)} shared lines work ok  
>> but BLF doesn't fire up.

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From nicolas at medularis.com  Wed Oct  7 07:43:44 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Wed, 7 Oct 2009 10:43:44 -0400
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <8750A70D-6891-4C07-B4DF-C7F4ABA58D67@freeswitch.org>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<1b46b4e80910041609x2f41ceb9u142cb484951a21ed@mail.gmail.com>
	<87f2f3b90910050920w4d17033fl4c386a0018107bdf@mail.gmail.com>
	<1b46b4e80910052119me12a0d9y4653edd831f9555e@mail.gmail.com>
	<1b46b4e80910052142jaa09f08l72e5e4b2b24d10fc@mail.gmail.com>
	<083DBCB9-48B2-4E61-81B5-B7D00A42EAB5@freeswitch.org>
	<1b46b4e80910061422q7be2a21fg5d2caaef7e307abb@mail.gmail.com>
	<9106C8E1-498F-4AE8-8D34-7F6AAD4DC8AE@freeswitch.org>
	<1b46b4e80910070648n622c3f9haacc22c2328b44ff@mail.gmail.com>
	<8750A70D-6891-4C07-B4DF-C7F4ABA58D67@freeswitch.org>
Message-ID: <1b46b4e80910070743h38d39e5ax362ff7944754067e@mail.gmail.com>

You are missing the point, it is only rejecting auth for FS, Asterisk and
X-Lite work fine with the same config for that gateway.


On Wed, Oct 7, 2009 at 10:20 AM, Brian West <brian at freeswitch.org> wrote:

> I would suspect its a PEBKAC.  I mean if you could register to a
> gateway that rejected auth... what purpose would auth serve in the
> first place?
>
> /b
>
> On Oct 7, 2009, at 8:48 AM, Nicolas Brenner wrote:
>
> > Is there some way to make FS register with the gateway that is
> > rejecting the authentication? is it FS or the SIP server at fault?
> > Why would X-Lite work and FS not?
> >
> > Thanks again for your time and help.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From srinivas.ksvreddy at gmail.com  Wed Oct  7 07:48:14 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Wed, 7 Oct 2009 20:18:14 +0530
Subject: [Freeswitch-users] mod_sofia.c registered calls how to know
Message-ID: <f8af5740910070748g1efee71ft5841b6fa4d2412c8@mail.gmail.com>

Hi

can any please tell me where registered calls are stored, so when incoming
call came to mod_sofia.c how it will check it is registered or not?\\

-- 
Srinivasula Reddy K
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From vhatz at kinetix.gr  Wed Oct  7 06:37:58 2009
From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI))
Date: Wed, 07 Oct 2009 16:37:58 +0300
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <716784809.20091007161440@globtel.ro>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>	<86a32abc0910061722t67ff23ddr62e4f62365539bd4@mail.gmail.com>	<191c3a030910061758r2ddea5fand388f0af2c4dfc33@mail.gmail.com>
	<716784809.20091007161440@globtel.ro>
Message-ID: <4ACC99B6.4070706@kinetix.gr>


Claudiu Filip wrote:
> Hi,
> 
> Wednesday, October 7, 2009, 3:58:20 AM, Anthony M. wrote:
>> barely get anybody asking about h.323.
> 
> 
> H323 may not be popular for small ITSPs or small/medium PBXes, but
> it's widely used by the big players.. and freeswitch doesnt share the
> same goals with asterisk.
> 
> 
> 
> Best wishes,
> 
> 
> Claudiu Filip
> 

I have to agree with Filip, H323 is not popular (or sometimes even 
known) to the VoIP end user and small PBX community. In fact many VoIP 
users that I've talked with even believe that VoIP started with SIP, 
which is not true.

H323 although diminishing in usage is still used among voice carriers 
because it better resembles telephony environments and is not prone to 
errors and problems of SIP (example: if there is a network error in 
mid-call H323 uses TCP for signaling and therefore almost immediately 
detects the error to hang-up the call, while SIP _typically_ uses UDP 
and cannot detect such issues via signaling but only if a media gateway 
monitors RTP activity etc.).

As it is already pointed out, many carriers switch to SIP steadily, but 
still H323 is important for carrier-to carrier interconnection.

FreeSWITCH is positioned/advertised as a high-capacity softswitch (not 
just a SME PBX) which implies interconnection with large carriers (who 
still prefer H323).

There are many SIP platforms out there, open source and closed source 
that can do SIP (most of them pretty good, some of them excellent).

There are many closed source platforms that can do SIP and H323 pretty 
good (and some of them excellent).

However, there is no open source platform that can do BOTH SIP and H323 
REALLY good. That could be a competitive advantage for an open source 
platform...

NOTE: All the above is just academic discussion. A reply to the above in 
the spirit "put your money where your mouth is" is not really 
constructive and will be countered by a series of valid arguments 
against it.

Best regards,
Vlasis Hatzistavrou.


From msc at freeswitch.org  Wed Oct  7 08:49:44 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 7 Oct 2009 08:49:44 -0700
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <7d79b3930910070334g53cef0c5m63037e264f1a9233@mail.gmail.com>
References: <7d79b3930910070334g53cef0c5m63037e264f1a9233@mail.gmail.com>
Message-ID: <87f2f3b90910070849t79b7cb90x4150bf1dd8501ca4@mail.gmail.com>

On Wed, Oct 7, 2009 at 3:34 AM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Hi,
> Again I was struck in a problem, Here is the scenario.
>
> On incomming call, I just call an event outboud socket. But what happens
> is, for the first 15 call, it is working fine. But from the 16th call to
> 30th call, it says the below error.
>
> 2009-10-07 15:07:48.201846 [WARNING] ozmod_libpri.c:761 --Failure opening
> channel 1:17 (ignored)
> 2009-10-07 15:07:55.381861 [WARNING] ozmod_libpri.c:761 --Failure opening
> channel 1:18 (ignored)
> 2009-10-07 15:07:58.569774 [WARNING] ozmod_libpri.c:761 --Failure opening
> channel 1:19 (ignored)
> 2009-10-07 15:08:01.37824 [WARNING] ozmod_libpri.c:761 --Failure opening
> channel 1:20 (ignored)
> 2009-10-07 15:08:03.129846 [WARNING] ozmod_libpri.c:761 --Failure opening
> channel 1:21 (ignored)
> 2009-10-07 15:08:04.825851 [WARNING] ozmod_libpri.c:761 --Failure opening
> channel 1:22 (ignored)
> 2009-10-07 15:08:06.289977 [WARNING] ozmod_libpri.c:761 --Failure opening
> channel 1:23 (ignored)
> 2009-10-07 15:08:07.761961 [WARNING] ozmod_libpri.c:761 --Failure opening
> channel 1:24 (ignored)
> 2009-10-07 15:08:09.737944 [WARNING] ozmod_libpri.c:761 --Failure opening
> channel 1:25 (ignored)
> 2009-10-07 15:08:11.462018 [WARNING] ozmod_libpri.c:761 --Failure opening
> channel 1:26 (ignored)
> 2009-10-07 15:08:13.566024 [WARNING] ozmod_libpri.c:761 --Failure opening
> channel 1:27 (ignored)
> 2009-10-07 15:08:15.430163 [WARNING] ozmod_libpri.c:761 --Failure opening
> channel 1:28 (ignored)
> 2009-10-07 15:08:17.446103 [WARNING] ozmod_libpri.c:761 --Failure opening
> channel 1:29 (ignored)
> 2009-10-07 15:08:19.430118 [WARNING] ozmod_libpri.c:761 --Failure opening
> channel 1:30 (ignored)
> 2009-10-07 15:08:21.358121 [WARNING] ozmod_libpri.c:761 --Failure opening
> channel 1:31 (ignored)
>
>
> But for the next call, when it is opening channel 1:1, it is executing my
> dial plans.
> I don't know why it failed when it is choosing 1:17-1:31. Any one has any
> idea.
>
> Below are my configuration details:
>
> openzap.conf
> [span zt PRI_1]
> trunk_type => e1
> b-channel => 1:1-15
> d-channel=> 1:16
> b-channel => 1:17-31
>

Just a hunch, but could you try this in openzap.conf:
b-channel => 1:1-15,17-31
d-channel => 1:16

I want to see what happens. I don't have an E1 setup to test with right now
otherwise I'd do it myself. Please report back. Thanks,
MC


>
> openzap.conf.xml
> <configuration name="openzap.conf" description="OpenZAP Configuration">
>         <settings>
>                 <param name="debug" value="7"/>
>         </settings>
>         <libpri_spans>
>                 <span name="PRI_1">
>                         <param name="node" value="cpe"/>
>                         <param name="switch" value="euroisdn"/>
>                         <param name="dialplan" value="XML"/>
>                         <param name="context" value="default"/>
>                 </span>
>         </libpri_spans>
> </configuration>
>
> zaptel.conf
>
> span=1,1,0,ccs,hdb3
> bchan=1-15,17-31
> dchan=16
>
> loadzone        = us
> defaultzone     = us
>
> oz list
>
> span: 1 (PRI_1)
> type: isdn
> chan_count: 47
> dialplan: XML
> context: default
> dial_regex:
> fail_dial_regex:
> hold_music:
> analog_options none
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Wed Oct  7 08:50:12 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 7 Oct 2009 17:50:12 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <14480284.20091007154057@globtel.ro>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<86a32abc0910061722t67ff23ddr62e4f62365539bd4@mail.gmail.com>
	<191c3a030910061758r2ddea5fand388f0af2c4dfc33@mail.gmail.com>
	<65d96fc80910070137k52be89e3g846e55db72a8de5d@mail.gmail.com>
	<14480284.20091007154057@globtel.ro>
Message-ID: <65d96fc80910070850h4c3735b7s7a7577b25b71e1f6@mail.gmail.com>

On Wed, Oct 7, 2009 at 2:40 PM, Claudiu Filip <claudiu at globtel.ro> wrote:

>
>
>
> Hi Tihomir,
>
>
>   I've  done  some  tests  to  see  how  suitable  is freeswitch as a
>   SIP/H323  translator  and  you  are  right about the fact that H323
>   'alert+open  logical  channel'  will generate a SIP '200 OK'. I was
>   able  to fix that with a couple of changes in mod_opal.cpp, however
>   some  things  were changed on mod_sofia in the latest svn. (on this
>   particular  issue,  open_logical_channel  is  processed  BEFORE the
>   alerting,   so   the   call   is   in   SetupPhase  when  the  proc
>   OnOpenMediaStream is triggered)
>
>
>


yep, thats correct ... i was just wondering why it hangs in SetUpPhase


2009-10-07 16:50:11.690451 [DEBUG] manager.cxx:718 OnOpenMediaStream
Call[n03f409711]-EP<h323>[localhost/3263],OpalRTPMediaStream-Source-G.711-ALaw-64k
2009-10-07 16:50:11.690451 [INFO] mod_opal.cpp:1283 opal/
h323:05492122 at 10.4.4.254 <h323%3A05492122 at 10.4.4.254> initialise
opal/h323:05492122 at 10.4.4.254read audio codec G.711-ALaw-64k for connection
FSMediaStream-Sink-G.711-ALaw-64k
2009-10-07 16:50:11.690451 [DEBUG] mod_opal.cpp:1313 Set read audio codec to
<< G.711-ALaw-64k for connection FSMediaStream-Sink-G.711-ALaw-64k
2009-10-07 16:50:11.691525 [DEBUG] manager.cxx:718 OnOpenMediaStream
Call[n03f409711]-EP<local>[1],FSMediaStream-Sink-G.711-ALaw-64k
*2009-10-07 16:50:11.691525 [CONSOLE] mod_opal.cpp:852 SetUpPhase =>
GetPhase() = '1'*
2009-10-07 16:50:11.691525 [DEBUG] connection.cxx:561 Opened sink stream
n03f409711_1 with format G.711-ALaw-64k
2009-10-07 16:50:11.691525 [DEBUG] patch.cxx:341 Created Sink:
format=G.711-ALaw-64k
2009-10-07 16:50:11.691525 [DEBUG] mediastrm.cxx:666 RTP data size cannot be
changed to 160, fixed at 2048
2009-10-07 16:50:11.691525 [DEBUG] patch.cxx:179 Added direct media stream
sink FSMediaStream-Sink-G.711-ALaw-64k



this is the original code, and it never triggers eraly_media as never
reaches AlertingPhase.

if (GetMediaStream(stream.GetSessionID(), stream.IsSink()) != NULL) {
        // Have open media in both directions.
        if (GetPhase() == AlertingPhase) {
            switch_channel_mark_pre_answered(m_fsChannel);
            switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CONSOLE, "LOG
==>\t Alerting => GetPhase() = '%d'\n",GetPhase());


        } else if (GetPhase() < ReleasingPhase) {
            switch_channel_mark_answered(m_fsChannel);
            switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CONSOLE, "LOG
==>\t GetPhase() = '%d'\n",GetPhase());
        }
    }




I tried this, it works for early media but i still need to open a full media
path and say the call actually connected ....


   if (GetMediaStream(stream.GetSessionID(), stream.IsSink()) != NULL) {

        // Have open media in both directions.
        if (GetPhase() < ConnectedPhase) {
            switch_channel_mark_pre_answered(m_fsChannel);
            switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CONSOLE, "EARLY
MEDIA => GetPhase() = '%d'\n",GetPhase());

        } else if (GetPhase() < ReleasingPhase) {
            switch_channel_mark_answered(m_fsChannel);
            switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CONSOLE, "FULL
MEDIA => GetPhase() = '%d'\n",GetPhase());

        }
    }






this is when i'm dong early_media:


2009-10-07 17:45:26.788082 [DEBUG] manager.cxx:718 OnOpenMediaStream
Call[c8dce50981]-EP<h323>[localhost/26906],OpalRTPMediaStream-Source-G.711-ALaw-64k
2009-10-07 17:45:26.789158 [INFO] mod_opal.cpp:1279 opal/
h323:05492122 at 10.4.4.254 <h323%3A05492122 at 10.4.4.254> initialise
opal/h323:05492122 at 10.4.4.254read audio codec G.711-ALaw-64k for connection
FSMediaStream-Sink-G.711-ALaw-64k
2009-10-07 17:45:26.789158 [DEBUG] mod_opal.cpp:1309 Set read audio codec to
<< G.711-ALaw-64k for connection FSMediaStream-Sink-G.711-ALaw-64k
2009-10-07 17:45:26.789158 [DEBUG] manager.cxx:718 OnOpenMediaStream
Call[c8dce50981]-EP<local>[1],FSMediaStream-Sink-G.711-ALaw-64k
2009-10-07 17:45:26.789158 [NOTICE] mod_opal.cpp:887 Pre-Answer opal/
h323:05492122 at 10.4.4.254 <h323%3A05492122 at 10.4.4.254>!
2009-10-07 17:45:26.789158 [DEBUG] switch_channel.c:1822 Send signal
sofia/internal/1001 at 10.4.62.7 [BREAK]
*2009-10-07 17:45:26.789158 [CONSOLE] mod_opal.cpp:888 EARLY MEDIA =>
GetPhase() = '1'*
2009-10-07 17:45:26.789158 [DEBUG] connection.cxx:561 Opened sink stream
c8dce50981_1 with format G.711-ALaw-64k
2009-10-07 17:45:26.789158 [DEBUG] patch.cxx:341 Created Sink:
format=G.711-ALaw-64k
2009-10-07 17:45:26.790236 [DEBUG] switch_ivr_originate.c:2154
sofia/internal/1001 at 10.4.62.7 receive message [PROGRESS]
2009-10-07 17:45:26.790236 [INFO] switch_ivr_originate.c:2154 Sending early
media
2009-10-07 17:45:26.790236 [DEBUG] sofia_glue.c:2329 AUDIO RTP
[sofia/internal/1001 at 10.4.62.7] 10.4.62.7 port 19594 -> 10.4.62.89 port 5004
codec: 8 ms: 20
2009-10-07 17:45:26.790236 [DEBUG] switch_rtp.c:1155 Starting timer [soft]
160 bytes per 20ms
2009-10-07 17:45:26.790236 [DEBUG] mediastrm.cxx:666 RTP data size cannot be
changed to 160, fixed at 2048
2009-10-07 17:45:26.790236 [DEBUG] patch.cxx:179 Added direct media stream
sink FSMediaStream-Sink-G.711-ALaw-64k
2009-10-07 17:45:26.790236 [DEBUG] connection.cxx:728 Sink stream of
connection Call[c8dce50981]-EP<local>[1] uses patch Patch
OpalRTPMediaStream-Source-G.711-ALaw-64k ->
FSMediaStream-Sink-G.711-ALaw-64k
2009-10-07 17:45:26.791311 [DEBUG] connection.cxx:728 Source stream of
connection Call[c8dce50981]-EP<h323>[localhost/26906] uses patch Patch
OpalRTPMediaStream-Source-G.711-ALaw-64k ->
FSMediaStream-Sink-G.711-ALaw-64k
2009-10-07 17:45:26.791311 [DEBUG] rtpconn.cxx:249 Adding RFC2833 receive
handler
2009-10-07 17:45:26.791311 [DEBUG] rtpconn.cxx:254 Adding Cisco NSE receive
handler
2009-10-07 17:45:26.791311 [DEBUG] h323ep.cxx:1118 Started receiving logical
channel: G.711-ALaw-64k <2>
2009-10-07 17:45:26.791311 [DEBUG] rtpconn.cxx:496 Releasing session 1
2009-10-07 17:45:26.792387 [DEBUG] rtpconn.cxx:496 Releasing session 1
2009-10-07 17:45:26.792387 [DEBUG] h323.cxx:2450 Fast starting 2 channels
2009-10-07 17:45:26.792387 [DEBUG] h323ep.cxx:1024 Received alerting PDU.
2009-10-07 17:45:26.792387 [DEBUG] manager.cxx:636 OnAlerting
Call[c8dce50981]-EP<h323>[localhost/26906]
2009-10-07 17:45:26.792387 [DEBUG] call.cxx:184 OnAlerting
Call[c8dce50981]-EP<h323>[localhost/26906]
2009-10-07 17:45:26.792387 [DEBUG] localep.cxx:214 SetAlerting(10.4.4.254)
2009-10-07 17:45:26.792387 [INFO] mod_sofia.c:1518 Ring SDP:
v=0
o=FreeSWITCH 1254910732 1254910733 IN IP4 10.4.62.7
s=FreeSWITCH
c=IN IP4 10.4.62.7
t=0 0
m=audio 19594 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2009-10-07 17:45:26.792387 [DEBUG] connection.cxx:1115 SetPhase from
SetUpPhase to AlertingPhase for Call[c8dce50981]-EP<local>[1]
2009-10-07 17:45:26.792387 [NOTICE] mod_sofia.c:1521 Pre-Answer
sofia/internal/1001 at 10.4.62.7!
2009-10-07 17:45:26.793458 [DEBUG] switch_core_session.c:630 Send signal
sofia/internal/1001 at 10.4.62.7 [BREAK]
2009-10-07 17:45:26.793458 [DEBUG] switch_ivr_originate.c:2196 Originate
Resulted in Success: [opal/h323:05492122 at 10.4.4.254<h323%3A05492122 at 10.4.4.254>
]
2009-10-07 17:45:26.793458 [DEBUG] switch_channel.c:182 opal/
h323:05492122 at 10.4.4.254 <h323%3A05492122 at 10.4.4.254> receive message
[AUDIO_SYNC]
2009-10-07 17:45:26.793458 [DEBUG] mod_opal.cpp:1156 Received message 20 on
connection Call[c8dce50981]-EP<local>[1]
2009-10-07 17:45:26.793458 [DEBUG] switch_channel.c:182 sofia/internal/
1001 at 10.4.62.7 receive message [AUDIO_SYNC]
2009-10-07 17:45:26.793458 [DEBUG] switch_ivr_bridge.c:911 opal/
h323:05492122 at 10.4.4.254 <h323%3A05492122 at 10.4.4.254> receive message
[BRIDGE]
2009-10-07 17:45:26.793458 [DEBUG] sofia.c:3359 Channel sofia/internal/
1001 at 10.4.62.7 entering state [early][183]
2009-10-07 17:45:26.793458 [DEBUG] mod_opal.cpp:1156 Received message 4 on
connection Call[c8dce50981]-EP<local>[1]
2009-10-07 17:45:26.793458 [DEBUG] switch_core_session.c:630 Send signal
opal/h323:05492122 at 10.4.4.254 <h323%3A05492122 at 10.4.4.254> [BREAK]
2009-10-07 17:45:26.794816 [DEBUG] mod_opal.cpp:999 Kill 3 on connection
Call[c8dce50981]-EP<local>[1]
2009-10-07 17:45:26.794816 [DEBUG] switch_ivr_bridge.c:918 sofia/internal/
1001 at 10.4.62.7 receive message [BRIDGE]
2009-10-07 17:45:26.794816 [DEBUG] switch_core_session.c:630 Send signal
sofia/internal/1001 at 10.4.62.7 [BREAK]
2009-10-07 17:45:26.794816 [DEBUG] switch_ivr_bridge.c:962 (opal/
h323:05492122 at 10.4.4.254 <h323%3A05492122 at 10.4.4.254>) State Change
CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA

< ----------------- snip ------------------>


 1 entries {
            [0]=dialedDigits "1001282122"
          }
        }
        h245Tunneling = true
      }
    }
  }
2009-10-07 17:45:31.892521 [DEBUG] h323.cxx:582 Handling PDU: Connect
callRef=26906
2009-10-07 17:45:31.892521 [DEBUG] connection.cxx:1115 SetPhase from
AlertingPhase to ConnectedPhase for
Call[c8dce50981]-EP<h323>[localhost/26906]
2009-10-07 17:45:31.892521 [DEBUG] connection.cxx:513 OnConnected for
Call[c8dce50981]-EP<h323>[localhost/26906]
2009-10-07 17:45:31.892521 [DEBUG] manager.cxx:652 OnConnected
Call[c8dce50981]-EP<h323>[localhost/26906]
2009-10-07 17:45:31.892521 [DEBUG] call.cxx:214 OnConnected
Call[c8dce50981]-EP<h323>[localhost/26906]
2009-10-07 17:45:31.892521 [DEBUG] connection.cxx:480 SetConnected for
Call[c8dce50981]-EP<local>[1]
2009-10-07 17:45:31.892521 [DEBUG] connection.cxx:1115 SetPhase from
AlertingPhase to ConnectedPhase for Call[c8dce50981]-EP<local>[1]
2009-10-07 17:45:31.892521 [DEBUG] connection.cxx:1115 SetPhase from
ConnectedPhase to EstablishedPhase for Call[c8dce50981]-EP<local>[1]
2009-10-07 17:45:31.892521 [DEBUG] connection.cxx:520 OnEstablished
Call[c8dce50981]-EP<local>[1]
2009-10-07 17:45:31.892521 [DEBUG] connection.cxx:617 Media stream threads
started.
2009-10-07 17:45:31.892521 [DEBUG] manager.cxx:660 OnEstablished
Call[c8dce50981]-EP<local>[1]
2009-10-07 17:45:31.892521 [DEBUG] call.cxx:245 OnEstablished
Call[c8dce50981]-EP<local>[1]
2009-10-07 17:45:31.892521 [DEBUG] connection.cxx:617 Media stream threads
started.
2009-10-07 17:45:31.892521 [DEBUG] connection.cxx:1115 SetPhase from
ConnectedPhase to EstablishedPhase for
Call[c8dce50981]-EP<h323>[localhost/26906]
2009-10-07 17:45:31.892521 [DEBUG] connection.cxx:520 OnEstablished
Call[c8dce50981]-EP<h323>[localhost/26906]
2009-10-07 17:45:31.892521 [DEBUG] connection.cxx:617 Media stream threads
started.
2009-10-07 17:45:31.893601 [DEBUG] manager.cxx:660 OnEstablished
Call[c8dce50981]-EP<h323>[localhost/26906]
2009-10-07 17:45:31.893601 [DEBUG] call.cxx:245 OnEstablished
Call[c8dce50981]-EP<h323>[localhost/26906]
2009-10-07 17:45:31.893601 [DEBUG] connection.cxx:617 Media stream threads
started.
2009-10-07 17:45:31.893601 [DEBUG] h323.cxx:945 Set protocol version to 5
2009-10-07 17:45:31.893601 [DEBUG] h323.cxx:1261 Set remote party name:
"10.4.4.254"
2009-10-07 17:45:31.893601 [DEBUG] h323.cxx:1269 Set remote application
name: "Avaya Multivantage R014x.00.1.731.2 181/19540 Vox Lucida Pty. Ltd."
2009-10-07 17:45:31.893601 [DEBUG] h323ep.cxx:1087 Received connect PDU.
2009-10-07 17:45:31.893601 [DEBUG] h323.cxx:2367 Fast start response with no
channels to open
2009-10-07 17:45:31.893601 [DEBUG] h323.cxx:3543
InternalEstablishedConnectionCheck: connectionState=HasExecutedSignalConnect
fastStartState=FastStartAcknowledged H.245 is ready



as ConnectMessage was never sent from opal to FS :P




what was your workaround?






>   The  most  important  problem  I'm having right now is that G729 is
>   still  not  working (poor quality due to high buffering). Even with
>   the  latest  Opal,  which  includes  the last week patch for jitter
>   buffer.  If  you  dont  need G729, I could send you a patch for the
>   latest svns (freeswitch, opal, ptlib), ofc no founding needed.
>
>   There are a couple of bugs in opal itself and h323ing freeswitch
>   with opal will bring them in. On the other hand, mod_opal is
>   already there, it just needs a few adjustments.
>
>
> Best wishes,
>
> Claudiu Filip
>
>
>
>
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From msc at freeswitch.org  Wed Oct  7 09:56:41 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 7 Oct 2009 09:56:41 -0700
Subject: [Freeswitch-users] unable to configure Digium TDM400P
In-Reply-To: <a44688a0910070340p4209e7b5yec9218b8c4fc692c@mail.gmail.com>
References: <a44688a0910070340p4209e7b5yec9218b8c4fc692c@mail.gmail.com>
Message-ID: <87f2f3b90910070956l520b2c51p7f33463128bf68b2@mail.gmail.com>

The ztcfg seems okay since it set the signaling to kewlstart. What part is
not working? Be sure to pastebin your openzap.conf and openzap.conf.xml
files as well as a full debug of what happens when loading mod_openzap.
-MC

On Wed, Oct 7, 2009 at 3:40 AM, Woody Dickson <woodydickson at gmail.com>wrote:

> Hi,
>
> I am trying to setup a Digium TDM400P following the instruction on the
> wiki.
> It is a 1 fxo and 1 fxs card, so I tried
> loadzone=in
> defaultzone=in
> fxsks=2
> fxoks=1
>
> and
>
> loadzone=in
> defaultzone=in
> fxsks=1
> fxoks=2
>
> None works.  Does anyone know how it should be configured?
>
> Here is what I get by following the wiki.
>
> [root at localhost zaptel]# ztcfg -vv
>
> Zaptel Version: SVN-branch-1.4-r4629M
> Echo Canceller: MG2
> Configuration
> ======================
>
>
> Channel map:
>
> Channel 01: FXS Kewlstart (Default) (Slaves: 01)
> Channel 02: FXO Kewlstart (Default) (Slaves: 02)
>
> 2 channels to configure.
>
> Changing signalling on channel 1 from FXO Kewlstart to FXS Kewlstart
> Changing signalling on channel 2 from FXS Kewlstart to FXO Kewlstart
>
>
> [root at localhost zaptel]# ztcfg -vv
>
> Zaptel Version: SVN-branch-1.4-r4629M
> Echo Canceller: MG2
> Configuration
> ======================
>
>
> Channel map:
>
> Channel 01: FXO Kewlstart (Default) (Slaves: 01)
> Channel 02: FXS Kewlstart (Default) (Slaves: 02)
>
> 2 channels to configure.
>
> Changing signalling on channel 1 from FXS Kewlstart to FXO Kewlstart
> Changing signalling on channel 2 from FXO Kewlstart to FXS Kewlstart
>
>
> [root at localhost zaptel]# lspci
> 00:14.0 Ethernet controller: Digium, Inc. TDM400P (rev 11)
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Prometheus001 at gmx.net  Wed Oct  7 10:11:27 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Wed, 07 Oct 2009 19:11:27 +0200
Subject: [Freeswitch-users] xml_curl configuration for failover cluster
Message-ID: <4ACCCBBF.4010308@gmx.net>

Hello,

I read in the wiki that binding blocks are processed in sequential order
in a failover matter.

So I  created the following bindings for XML-Curl:
However grepping the network traffic I can see that Freewitch always
fetches both servers fo one binding. So there is no real failover.
How can I avoid that?

Best regards
Peter


<configuration name="xml_curl.conf" description="CURL XML Gateway">
  <bindings>

    <!-- Configuration -->
    <!-- FIRST Application server -->
    <binding name="configuration">
      <param name="gateway-url"
value="http://localhost/xml_curls/configuration" bindings="configuration"/>
    </binding>
    <!-- SECOND Application server -->
    <binding name="configuration_backup">
      <param name="gateway-url"
value="http://10.0.0.104/xml_curls/configuration" bindings="configuration"/>
    </binding>

    <!-- Directory -->
    <!-- FIRST Application server -->
    <binding name="directory">
      <param name="gateway-url"
value="http://localhost/xml_curls/directory" bindings="directory"/>
    </binding>
    <!-- SECOND Application server -->
    <binding name="directory_backup">
      <param name="gateway-url"
value="http://10.0.0.104/xml_curls/directory" bindings="directory"/>
    </binding>

    <!-- Dialplan -->
    <!-- FIRST Application server -->
    <binding name="dialplan">
      <param name="gateway-url"
value="http://localhost/xml_curls/dialplan" bindings="dialplan"/>
    </binding>
    <!-- SECOND Application server -->
    <binding name="dialpla_backup">
      <param name="gateway-url"
value="http://10.0.0.104/xml_curls/dialplan" bindings="dialplan"/>
    </binding>

  </bindings>
</configuration>


From andy at fabulous4.co.uk  Wed Oct  7 10:44:00 2009
From: andy at fabulous4.co.uk (Andy)
Date: Wed, 7 Oct 2009 18:44:00 +0100
Subject: [Freeswitch-users] NAT problems - sorry
In-Reply-To: <5A70A0D4-70FA-43B2-BAF5-F8D6ECC64838@freeswitch.org>
References: <5BE289D2AE22492BA71A6919C8A440B3@D810>
	<5A70A0D4-70FA-43B2-BAF5-F8D6ECC64838@freeswitch.org>
Message-ID: <60E56B6B355C45BCA0D31F4425B9173A@D810>

Thanks Brian, sorry should have pre-empted that one as I've issue before.
start_dtmf is in the dialplan and occurs at the start of every call.
 
On closer inspection however it appears that only part of the nat setup is
taking place.
 
sofia status gives:
 
Name          Type                               Data     State
============================================================================
=====================
                 internal       profile       sip:mod_sofia at 10.10.0.2:5080
RUNNING (0)
                 external       profile       sip:mod_sofia at 10.10.0.2:5060
RUNNING (0)
                  default         alias                           internal
ALIASED
                10.10.0.2         alias                           internal
ALIASED
============================================================================
=====================
2 profiles 2 aliases

 
sofia status profile external gives (the external ip addresses are correct):

API CALL [sofia(status profile external)] output:
============================================================================
=====================
Name                    external
Domain Name             N/A
DBName                  sofia_reg_external
Pres Hosts
Dialplan                XML,enum
Context                 default
Challenge Realm         auto_to
RTP-IP                  10.10.0.2
Ext-RTP-IP              82.5.159.138
SIP-IP                  10.10.0.2
Ext-SIP-IP              82.5.159.138
URL                     sip:mod_sofia at 10.10.0.2:5060
BIND-URL                sip:mod_sofia at 10.10.0.2:5060
HOLD-MUSIC              N/A
OUTBOUND-PROXY          N/A
CODECS                  PCMU,PCMA,GSM
TEL-EVENT               101
DTMF-MODE               rfc2833
CNG                     13
SESSION-TO              0
MAX-DIALOG              0
NOMEDIA                 false
LATE-NEG                false
PROXY-MEDIA             false
AGGRESSIVENAT           false
STUN-ENABLED            true
STUN-AUTO-DISABLE       false
CALLS-IN                0
FAILED-CALLS-IN         0
CALLS-OUT               0
FAILED-CALLS-OUT        0
 
BUT......
 
nat_map status, gives:

API CALL [nat_map(status)] output:
false

And there is no mention of nat detection in the startup log.
 
Is this because I'm using port 5060 externally?
 
Cheers
Andy 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: 07 October 2009 15:22
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] NAT problems - sorry


Bet you its inband dtmf and you need to start the dtmf detector. 


/b


On Oct 7, 2009, at 8:11 AM, Andy wrote:


I can hear the IVR message played down the phone line so outgoing audio is
ok.


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From brian at freeswitch.org  Wed Oct  7 10:49:24 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 7 Oct 2009 12:49:24 -0500
Subject: [Freeswitch-users] NAT problems - sorry
In-Reply-To: <60E56B6B355C45BCA0D31F4425B9173A@D810>
References: <5BE289D2AE22492BA71A6919C8A440B3@D810>
	<5A70A0D4-70FA-43B2-BAF5-F8D6ECC64838@freeswitch.org>
	<60E56B6B355C45BCA0D31F4425B9173A@D810>
Message-ID: <6B2B7FFD-75BC-44CE-A11C-2B26447DA889@freeswitch.org>

No its because you're not behind a upnp/nat-pmp router so you'll have  
to manually forward everything... All the info you showed displaying  
the profile status is correct.

/b

On Oct 7, 2009, at 12:44 PM, Andy wrote:

> Is this because I'm using port 5060 externally?
>
> Cheers
> Andy

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From dome at tel.co.th  Wed Oct  7 10:52:35 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Thu, 8 Oct 2009 00:52:35 +0700
Subject: [Freeswitch-users] mod_sofia.c registered calls how to know
In-Reply-To: <f8af5740910070748g1efee71ft5841b6fa4d2412c8@mail.gmail.com>
References: <f8af5740910070748g1efee71ft5841b6fa4d2412c8@mail.gmail.com>
Message-ID: <8ccbff060910071052x5c27c7c0ub15c69c5c39ff83e@mail.gmail.com>

2009/10/7 srinivasula reddy <srinivas.ksvreddy at gmail.com>:
>
> Hi
>
> can any please tell me where registered calls are stored, so when incoming
> call came to mod_sofia.c how it will check it is registered or not?\\
I use 2 profile
   internal profile require register before make a call.
   external i use odbcquery  check ipaddress before process call. (in dialplan)

Dome C.
>
> --
> Srinivasula Reddy K
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From andy at fabulous4.co.uk  Wed Oct  7 10:58:19 2009
From: andy at fabulous4.co.uk (Andy)
Date: Wed, 7 Oct 2009 18:58:19 +0100
Subject: [Freeswitch-users] NAT problems - sorry
In-Reply-To: <6B2B7FFD-75BC-44CE-A11C-2B26447DA889@freeswitch.org>
References: <5BE289D2AE22492BA71A6919C8A440B3@D810><5A70A0D4-70FA-43B2-BAF5-F8D6ECC64838@freeswitch.org><60E56B6B355C45BCA0D31F4425B9173A@D810>
	<6B2B7FFD-75BC-44CE-A11C-2B26447DA889@freeswitch.org>
Message-ID: <703EEB9AC9AD43898C6C46F82904E6A9@D810>

Many thanks Brian, the firewall docs assure me it is uPnp but is probably
lying or a poor implementation. Could you point me to the right section of
the Wiki to tell me how to do this manually as I've been scouting for some
time and can;t seem to find the right thing. sorry if I'm being blind.

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: 07 October 2009 18:49
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] NAT problems - sorry


No its because you're not behind a upnp/nat-pmp router so you'll have to
manually forward everything... All the info you showed displaying the
profile status is correct. 

/b

On Oct 7, 2009, at 12:44 PM, Andy wrote:



Is this because I'm using port 5060 externally?
 
Cheers
Andy 


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From brian at freeswitch.org  Wed Oct  7 11:06:37 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 7 Oct 2009 13:06:37 -0500
Subject: [Freeswitch-users] NAT problems - sorry
In-Reply-To: <703EEB9AC9AD43898C6C46F82904E6A9@D810>
References: <5BE289D2AE22492BA71A6919C8A440B3@D810><5A70A0D4-70FA-43B2-BAF5-F8D6ECC64838@freeswitch.org><60E56B6B355C45BCA0D31F4425B9173A@D810>
	<6B2B7FFD-75BC-44CE-A11C-2B26447DA889@freeswitch.org>
	<703EEB9AC9AD43898C6C46F82904E6A9@D810>
Message-ID: <0A2A7A27-611E-40FA-8268-D2C451AA8B77@freeswitch.org>

s/auto-nat/$realip/ then forward the rtp ports and sip ports.

/b
PS chances are you have to ENABLE upnp.
On Oct 7, 2009, at 12:58 PM, Andy wrote:

> Many thanks Brian, the firewall docs assure me it is uPnp but is  
> probably lying or a poor implementation. Could you point me to the  
> right section of the Wiki to tell me how to do this manually as I've  
> been scouting for some time and can;t seem to find the right thing.  
> sorry if I'm being blind.

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From andy at fabulous4.co.uk  Wed Oct  7 11:48:33 2009
From: andy at fabulous4.co.uk (Andy)
Date: Wed, 7 Oct 2009 19:48:33 +0100
Subject: [Freeswitch-users] NAT problems - sorry
In-Reply-To: <0A2A7A27-611E-40FA-8268-D2C451AA8B77@freeswitch.org>
References: <5BE289D2AE22492BA71A6919C8A440B3@D810><5A70A0D4-70FA-43B2-BAF5-F8D6ECC64838@freeswitch.org><60E56B6B355C45BCA0D31F4425B9173A@D810><6B2B7FFD-75BC-44CE-A11C-2B26447DA889@freeswitch.org><703EEB9AC9AD43898C6C46F82904E6A9@D810>
	<0A2A7A27-611E-40FA-8268-D2C451AA8B77@freeswitch.org>
Message-ID: <1B629AB98E50412496D2673E1FE160D9@D810>

Thanks Brian,
 
I've now set the external ips manually to be my external ip and have forward
all ports through my firewall to the FS server. It's actually set up as a
DMZ to everything is being forwarded without restriction but sadly DTMF and
HANGUP messages are still not getting through. Have I misunderstood what is
required. Is there some additional forwarding within FS required. I'm really
sorry to keep coming back but I've been wrestling with this for a long time
now and not getting anywhere.
 
Many thanks
Andy 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: 07 October 2009 19:07
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] NAT problems - sorry


s/auto-nat/$realip/ then forward the rtp ports and sip ports. 

/b
PS chances are you have to ENABLE upnp.

On Oct 7, 2009, at 12:58 PM, Andy wrote:


Many thanks Brian, the firewall docs assure me it is uPnp but is probably
lying or a poor implementation. Could you point me to the right section of
the Wiki to tell me how to do this manually as I've been scouting for some
time and can;t seem to find the right thing. sorry if I'm being blind.



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From rupa at rupa.com  Wed Oct  7 12:30:16 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 7 Oct 2009 13:30:16 -0600
Subject: [Freeswitch-users] NAT problems - sorry
In-Reply-To: <1B629AB98E50412496D2673E1FE160D9@D810>
References: <5BE289D2AE22492BA71A6919C8A440B3@D810>
	<5A70A0D4-70FA-43B2-BAF5-F8D6ECC64838@freeswitch.org>
	<60E56B6B355C45BCA0D31F4425B9173A@D810>
	<6B2B7FFD-75BC-44CE-A11C-2B26447DA889@freeswitch.org>
	<703EEB9AC9AD43898C6C46F82904E6A9@D810>
	<0A2A7A27-611E-40FA-8268-D2C451AA8B77@freeswitch.org>
	<1B629AB98E50412496D2673E1FE160D9@D810>
Message-ID: <b39273530910071230x1e5c3ddawa72239545c1d86c6@mail.gmail.com>

Double check your firewall and:

1) ensure you've actually enabled UPNP
and
2) Ensure that any mention of a SIP ALG (application level gateway) is
turned off.  SIP ALGs tend to really screw things up.

On Wed, Oct 7, 2009 at 12:48 PM, Andy <andy at fabulous4.co.uk> wrote:
> I've now set the external ips manually to be my external ip and have forward
> all ports through my firewall to the FS server. It's actually set up as a
> DMZ to everything is?being forwarded without restriction but sadly DTMF and
> HANGUP messages are still not getting through. Have I misunderstood what is
> required. Is there some additional forwarding within FS required. I'm really
> sorry to keep?coming back but I've been wrestling with this?for a long time
> now and not?getting anywhere.
>



-- 
-Rupa


From kristian.kielhofner at gmail.com  Wed Oct  7 12:35:56 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Wed, 7 Oct 2009 15:35:56 -0400
Subject: [Freeswitch-users] linksys spa-3102 fax t38 + freeswitch media
	proxy
In-Reply-To: <4ACB9519.9000805@gmail.com>
References: <4ACB9519.9000805@gmail.com>
Message-ID: <2d9149cd0910071235k27c1429v2bc555c0e7397fc4@mail.gmail.com>

Since no one else has responded I'll chime in with some general advice.

It's troubling to see that your "provider" is using Asterisk to face
you (the customer).

I've never had any luck getting T.38 to work (at all, in any mode)
using Asterisk.  I've heard of other people making it work but
consider the chain here:

Their provider (they appear to be using DIDx, hmmm) -> Callwithus
(using Asterisk) -> You (FreeSWITCH) -> Your device

I don't see anything specific in your case but there are many, many
places for T.38 to go wrong in this scenario.  I'd recommend
attempting T.38 against a local gateway or known Tier 1 provider
supporting T.38 with known equipment.

Otherwise, do you have any updates for us?

On Tue, Oct 6, 2009 at 3:06 PM, Vladimir Elizarov
<xengelpublicx at gmail.com> wrote:
> Hello.
>
> I want to set up faxing via the gateway linksys spa-3102 (with support
> for t38) via SIP.
>
> SIP-client -> linksys spa3102 -> fs -> provider
>
> ? ?<gateway name="callwithus">
> ? ? ? ?....
> ? ? ? ?<param name="inbound-late-negotiation" value="true"/>
> ? ? ? ?<param name="stun-enabled" value="true"/>
> ? ?</gateway>
>
> <extension name="Fax">
> ? ?<condition field="destination_number" expression="^(30)(\d{11})$">
> ? ? ? ? ? ?<action application="set" data="proxy_media=true"/>
> ? ? ? ? ? ?<action application="set" data="bypass_media=false"/>
> ? ? ? ? ? ?<action application="bridge"
> data="sofia/gateway/callwithus/*4087$2 at sip.callwithus.com"/>
> ? ? ? ?</condition>
> </extension>
>
> I make a call, press start on the fax. Fax not sent. Get this in the log
> (UA freeswitch not trunk-12805):
>
> ? ------------------------------------------------------------------------
> ? INVITE sip:*408774957558438 at sip.callwithus.com SIP/2.0
> ? Via: SIP/2.0/UDP 77.239.230.202:5080;rport;branch=z9hG4bK8yjrHeUprX2Hr
> ? Max-Forwards: 99
> ? From: "Mihailova Ludmila"
> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
> ? To: <sip:*408774957558438 at sip.callwithus.com>
> ? Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
> ? CSeq: 121302430 INVITE
> ? Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
> ? Supported: timer, precondition, path, replaces
> ? Allow-Events: talk, refer
> ? Content-Type: application/sdp
> ? Content-Disposition: session
> ? Content-Length: 250
> ? Remote-Party-ID: "Mihailova Ludmila"
> <sip:214 at sip.callwithus.com>;party=calling;screen=yes;privacy=off
>
> ? v=0
> ? o=- 216896551 216896551 IN IP4 192.168.50.51
> ? s=-
> ? c=IN IP4 77.239.230.202
> ? t=0 0
> ? m=audio 32404 RTP/AVP 0 100 101
> ? a=rtpmap:0 PCMU/8000
> ? a=rtpmap:100 NSE/8000
> ? a=fmtp:100 192-193
> ? a=rtpmap:101 telephone-event/8000
> ? a=fmtp:101 0-15
> ? a=ptime:30
>
> ? ------------------------------------------------------------------------
> send 1333 bytes to udp/[38.99.70.232]:5060 at 15:05:34.485226:
> ? ------------------------------------------------------------------------
> ? INVITE sip:*408774957558438 at sip.callwithus.com SIP/2.0
> ? Via: SIP/2.0/UDP 77.239.230.202:5080;rport;branch=z9hG4bK97BHK9BtN6r4K
> ? Max-Forwards: 99
> ? From: "Mihailova Ludmila"
> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
> ? To: <sip:*408774957558438 at sip.callwithus.com>
> ? Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
> ? CSeq: 121302431 INVITE
> ? Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
> ? Expires: 3600
> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
> ? Supported: timer, precondition, path, replaces
> ? Allow-Events: talk, refer
> ? Content-Type: application/sdp
> ? Content-Disposition: session
> ? Content-Length: 250
> ? Remote-Party-ID: "Mihailova Ludmila"
> <sip:214 at sip.callwithus.com>;party=calling;screen=yes;privacy=off
>
> ? v=0
> ? o=- 216896551 216896551 IN IP4 192.168.50.51
> ? s=-
> ? c=IN IP4 77.239.230.202
> ? t=0 0
> ? m=audio 32404 RTP/AVP 0 100 101
> ? a=rtpmap:0 PCMU/8000
> ? a=rtpmap:100 NSE/8000
> ? a=fmtp:100 192-193
> ? a=rtpmap:101 telephone-event/8000
> ? a=fmtp:101 0-15
> ? a=ptime:30
> ? ------------------------------------------------------------------------
> ? SIP/2.0 100 Giving a try
> ? Via: SIP/2.0/UDP
> 192.168.50.10:5080;rport=5080;branch=z9hG4bK97BHK9BtN6r4K
> ? From: "Mihailova Ludmila"
> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
> ? To: <sip:*408774957558438 at sip.callwithus.com>
> ? Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
> ? CSeq: 121302431 INVITE
> ? Server: CWU SIP GW
> ? Content-Length: 0
>
> ? ------------------------------------------------------------------------
> recv 836 bytes from udp/[38.99.70.232]:5060 at 15:05:49.641312:
> ? ------------------------------------------------------------------------
> ? SIP/2.0 183 Session Progress
> ? Via: SIP/2.0/UDP
> 192.168.50.10:5080;rport=5080;branch=z9hG4bK97BHK9BtN6r4K
> ? Record-Route:
> <sip:38.99.70.232;lr=on;ftag=HXjrDKpHHX8ZN;did=e8e.089fc2b6>
> ? From: "Mihailova Ludmila"
> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
> ? To: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
> ? Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
> ? CSeq: 121302431 INVITE
> ? User-Agent: CWU SIP-GW
> ? Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> ? Supported: replaces
> ? Contact: <sip:*408774957558438 at 204.74.213.1:5062>
> ? Content-Type: application/sdp
> ? Content-Length: 209
>
> ? v=0
> ? o=root 31797 31797 IN IP4 4.55.17.66
> ? s=session
> ? c=IN IP4 4.55.17.66
> ? t=0 0
> ? m=audio 14104 RTP/AVP 0 101
> ? a=rtpmap:0 PCMU/8000
> ? a=rtpmap:101 telephone-event/8000
> ? a=fmtp:101 0-16
> ? a=ptime:20
> ? a=sendrecv
> ? ------------------------------------------------------------------------
> v=0
> o=root 31797 31797 IN IP4 4.55.17.66
> s=session
> c=IN IP4 4.55.17.66
> t=0 0
> m=audio 14104 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
> 2009-10-06 19:03:31.112988 [NOTICE] sofia.c:3449 Pre-Answer
> sofia/external/*408774957558438 at sip.callwithus.com!
> 2009-10-06 19:03:31.112988 [DEBUG] switch_channel.c:1822 Send signal
> sofia/internal/214 at pbx0.tssec.lan [BREAK]
> 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2095 Set Codec
> sofia/external/*408774957558438 at sip.callwithus.com PROXY/8000 20 ms 160
> samples
> 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2372 PROXY AUDIO RTP
> [sofia/external/*408774957558438 at sip.callwithus.com]
> 192.168.50.10:32404->4.55.17.66:14104 codec: 0 ms: 20
> 2009-10-06 19:03:31.112988 [DEBUG] switch_rtp.c:1163 Not using a timer
> 2009-10-06 19:03:31.112988 [DEBUG] switch_ivr_originate.c:2154
> sofia/internal/214 at pbx0.tssec.lan receive message [PROGRESS]
> 2009-10-06 19:03:31.112988 [INFO] switch_ivr_originate.c:2154 Sending
> early media
> 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:1322
> sofia/internal/214 at pbx0.tssec.lan Patched SDP
> ---
> v=0
> o=root 31797 31797 IN IP4 4.55.17.66
> s=session
> c=IN IP4 4.55.17.66
> t=0 0
> m=audio 14104 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
> +++
> v=0
> o=root 31797 31797 IN IP4 4.55.17.66
> s=session
> c=IN IP4 192.168.50.10
> t=0 0
> m=audio 32698 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
> 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2372 PROXY AUDIO RTP
> [sofia/internal/214 at pbx0.tssec.lan]
> 192.168.50.10:32698->192.168.50.51:16474 codec: 0 ms: 30
> 2009-10-06 19:03:31.112988 [DEBUG] switch_rtp.c:1163 Not using a timer
> 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2508 Set comfort noise
> payload to 13
> 2009-10-06 19:03:31.112988 [DEBUG] sofia.c:3462
> sofia/internal/214 at pbx0.tssec.lan receive message [PROGRESS]
> 2009-10-06 19:03:31.116989 [NOTICE] mod_sofia.c:1521 Pre-Answer
> sofia/internal/214 at pbx0.tssec.lan!
> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal
> sofia/internal/214 at pbx0.tssec.lan [BREAK]
> 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_originate.c:2196 Originate
> Resulted in Success: [sofia/external/*408774957558438 at sip.callwithus.com]
> 2009-10-06 19:03:31.116989 [DEBUG] switch_channel.c:182
> sofia/external/*408774957558438 at sip.callwithus.com receive message
> [AUDIO_SYNC]
> 2009-10-06 19:03:31.116989 [DEBUG] switch_channel.c:182
> sofia/internal/214 at pbx0.tssec.lan receive message [AUDIO_SYNC]
> 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_bridge.c:911
> sofia/external/*408774957558438 at sip.callwithus.com receive message [BRIDGE]
> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal
> sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
> 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_bridge.c:918
> sofia/internal/214 at pbx0.tssec.lan receive message [BRIDGE]
> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal
> sofia/internal/214 at pbx0.tssec.lan [BREAK]
> 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_bridge.c:962
> (sofia/external/*408774957558438 at sip.callwithus.com) State Change
> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
> 2009-10-06 19:03:31.116989 [DEBUG] sofia.c:3359 Channel
> sofia/internal/214 at pbx0.tssec.lan entering state [early][183]
> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal
> sofia/internal/214 at pbx0.tssec.lan [BREAK]
> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/*408774957558438 at sip.callwithus.com) Running State
> Change CS_EXCHANGE_MEDIA
> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_state_machine.c:343
> (sofia/external/*408774957558438 at sip.callwithus.com) State EXCHANGE_MEDIA
> 2009-10-06 19:03:31.116989 [DEBUG] mod_sofia.c:430 SOFIA LOOPBACK
> recv 822 bytes from udp/[38.99.70.232]:5060 at 15:05:58.503701:
> ? ------------------------------------------------------------------------
> ? SIP/2.0 200 OK
> ? Via: SIP/2.0/UDP
> 192.168.50.10:5080;rport=5080;branch=z9hG4bK97BHK9BtN6r4K
> ? Record-Route:
> <sip:38.99.70.232;lr=on;ftag=HXjrDKpHHX8ZN;did=e8e.089fc2b6>
> ? From: "Mihailova Ludmila"
> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
> ? To: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
> ? Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
> ? CSeq: 121302431 INVITE
> ? User-Agent: CWU SIP-GW
> ? Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> ? Supported: replaces
> ? Contact: <sip:*408774957558438 at 204.74.213.1:5062>
> ? Content-Type: application/sdp
> ? Content-Length: 209
>
> ? v=0
> ? o=root 31797 31798 IN IP4 4.55.17.66
> ? s=session
> ? c=IN IP4 4.55.17.66
> ? t=0 0
> ? m=audio 14104 RTP/AVP 0 101
> ? a=rtpmap:0 PCMU/8000
> ? a=rtpmap:101 telephone-event/8000
> ? a=fmtp:101 0-16
> ? a=ptime:20
> ? a=sendrecv
> ? ------------------------------------------------------------------------
> send 762 bytes to udp/[38.99.70.232]:5060 at 15:05:58.507702:
> ? ------------------------------------------------------------------------
> ? ACK sip:*408774957558438 at 204.74.213.1:5062 SIP/2.0
> ? Via: SIP/2.0/UDP 192.168.50.10:5080;rport;branch=z9hG4bKaH59m4vXjFFQF
> ? Route: <sip:38.99.70.232;lr=on;ftag=HXjrDKpHHX8ZN;did=e8e.089fc2b6>
> ? Max-Forwards: 70
> ? From: "Mihailova Ludmila"
> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
> ? To: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
> ? Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
> ? CSeq: 121302431 ACK
> ? Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
> ? Content-Length: 0
>
> ? ------------------------------------------------------------------------
> 2009-10-06 19:03:39.975377 [DEBUG] sofia.c:3359 Channel
> sofia/external/*408774957558438 at sip.callwithus.com entering state
> [ready][200]
> 2009-10-06 19:03:39.975377 [DEBUG] switch_channel.c:1935 Send signal
> sofia/internal/214 at pbx0.tssec.lan [BREAK]
> 2009-10-06 19:03:39.975377 [NOTICE] sofia.c:3814 Channel
> [sofia/external/*408774957558438 at sip.callwithus.com] has been answered
> 2009-10-06 19:03:39.975377 [DEBUG] switch_channel.c:182
> sofia/external/*408774957558438 at sip.callwithus.com receive message
> [AUDIO_SYNC]
> 2009-10-06 19:03:39.975377 [DEBUG] sofia.c:3819
> sofia/internal/214 at pbx0.tssec.lan receive message [ANSWER]
> 2009-10-06 19:03:39.975377 [DEBUG] sofia_glue.c:1322
> sofia/internal/214 at pbx0.tssec.lan Patched SDP
> ---
> v=0
> o=root 31797 31797 IN IP4 4.55.17.66
> s=session
> c=IN IP4 4.55.17.66
> t=0 0
> m=audio 14104 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
> +++
> v=0
> o=root 31797 31797 IN IP4 4.55.17.66
> s=session
> c=IN IP4 192.168.50.10
> t=0 0
> m=audio 32698 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
> 2009-10-06 19:03:39.975377 [DEBUG] switch_core_session.c:630 Send signal
> sofia/internal/214 at pbx0.tssec.lan [BREAK]
> 2009-10-06 19:03:39.975377 [NOTICE] sofia.c:3819 Channel
> [sofia/internal/214 at pbx0.tssec.lan] has been answered
> 2009-10-06 19:03:39.975377 [DEBUG] switch_channel.c:182
> sofia/internal/214 at pbx0.tssec.lan receive message [AUDIO_SYNC]
> 2009-10-06 19:03:39.975377 [DEBUG] sofia.c:3359 Channel
> sofia/internal/214 at pbx0.tssec.lan entering state [completed][200]
> 2009-10-06 19:03:39.991381 [DEBUG] sofia.c:3359 Channel
> sofia/internal/214 at pbx0.tssec.lan entering state [ready][200]
> recv 1054 bytes from udp/[38.99.70.232]:5060 at 15:06:05.577607:
> ? ------------------------------------------------------------------------
> ? INVITE sip:gw+callwithus at 192.168.50.10:5080;transport=udp SIP/2.0
> ? Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.0
> ? Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK41d48157;rport=5062
> ? From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
> ? To: "Mihailova Ludmila"
> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
> ? Contact: <sip:*408774957558438 at 204.74.213.1:5062>
> ? Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
> ? CSeq: 102 INVITE
> ? User-Agent: CWU SIP-GW
> ? Max-Forwards: 69
> ? Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> ? Supported: replaces
> ? X-asterisk-info: SIP re-invite (T38 switchover)
> ? Content-Type: application/sdp
> ? Content-Length: 350
>
> ? v=0
> ? o=root 31797 31799 IN IP4 32.50.63.192
> ? s=session
> ? c=IN IP4 32.50.63.192
> ? t=0 0
> ? m=image 14104 udptl t38
> ? a=T38FaxVersion:0
> ? a=T38MaxBitRate:9600
> ? a=T38FaxFillBitRemoval:0
> ? a=T38FaxTranscodingMMR:0
> ? a=T38FaxTranscodingJBIG:0
> ? a=T38FaxRateManagement:transferredTCF
> ? a=T38FaxMaxBuffer:176
> ? a=T38FaxMaxDatagram:176
> ? a=T38FaxUdpEC:t38UDPRedundancy
> ? ------------------------------------------------------------------------
> send 443 bytes to udp/[38.99.70.232]:5060 at 15:06:05.577607:
> ? ------------------------------------------------------------------------
> ? SIP/2.0 100 Trying
> ? Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.0
> ? Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK41d48157;rport=5062
> ? From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
> ? To: "Mihailova Ludmila"
> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
> ? Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
> ? CSeq: 102 INVITE
> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
> ? Content-Length: 0
>
> ? ------------------------------------------------------------------------
> 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3359 Channel
> sofia/external/*408774957558438 at sip.callwithus.com entering state
> [received][100]
> 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3366 Remote SDP:
> v=0
> o=root 31797 31799 IN IP4 32.50.63.192
> s=session
> c=IN IP4 32.50.63.192
> t=0 0
> m=image 14104 udptl t38
> a=T38FaxVersion:0
> a=T38MaxBitRate:9600
> a=T38FaxFillBitRemoval:0
> a=T38FaxTranscodingMMR:0
> a=T38FaxTranscodingJBIG:0
> a=T38FaxRateManagement:transferredTCF
> a=T38FaxMaxBuffer:176
> a=T38FaxMaxDatagram:176
> a=T38FaxUdpEC:t38UDPRedundancy
>
> 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3683 Passing SDP to other leg.
> v=0
> o=root 31797 31799 IN IP4 32.50.63.192
> s=session
> c=IN IP4 32.50.63.192
> t=0 0
> m=image 14104 udptl t38
> a=T38FaxVersion:0
> a=T38MaxBitRate:9600
> a=T38FaxFillBitRemoval:0
> a=T38FaxTranscodingMMR:0
> a=T38FaxTranscodingJBIG:0
> a=T38FaxRateManagement:transferredTCF
> a=T38FaxMaxBuffer:176
> a=T38FaxMaxDatagram:176
> a=T38FaxUdpEC:t38UDPRedundancy
>
> 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3696
> sofia/internal/214 at pbx0.tssec.lan receive message [MEDIA_REDIRECT]
> 2009-10-06 19:03:47.049284 [DEBUG] mod_sofia.c:1195
> sofia/internal/214 at pbx0.tssec.lan Sending media re-direct:
> v=0
> o=root 31797 31799 IN IP4 32.50.63.192
> s=session
> c=IN IP4 32.50.63.192
> t=0 0
> m=image 14104 udptl t38
> a=T38FaxVersion:0
> a=T38MaxBitRate:9600
> a=T38FaxFillBitRemoval:0
> a=T38FaxTranscodingMMR:0
> a=T38FaxTranscodingJBIG:0
> a=T38FaxRateManagement:transferredTCF
> a=T38FaxMaxBuffer:176
> a=T38FaxMaxDatagram:176
> a=T38FaxUdpEC:t38UDPRedundancy
>
> 2009-10-06 19:03:47.049284 [DEBUG] sofia_glue.c:1101 Remote address:port
> [192.168.50.51:16474] has not changed.
> 2009-10-06 19:03:47.049284 [DEBUG] sofia_glue.c:1322
> sofia/internal/214 at pbx0.tssec.lan Patched SDP
> ---
> v=0
> o=root 31797 31799 IN IP4 32.50.63.192
> s=session
> c=IN IP4 32.50.63.192
> t=0 0
> m=image 14104 udptl t38
> a=T38FaxVersion:0
> a=T38MaxBitRate:9600
> a=T38FaxFillBitRemoval:0
> a=T38FaxTranscodingMMR:0
> a=T38FaxTranscodingJBIG:0
> a=T38FaxRateManagement:transferredTCF
> a=T38FaxMaxBuffer:176
> a=T38FaxMaxDatagram:176
> a=T38FaxUdpEC:t38UDPRedundancy
>
> +++
> v=0
> o=root 31797 31799 IN IP4 32.50.63.192
> s=session
> c=IN IP4 192.168.50.10
> t=0 0
> m=image 32698 udptl t38
> a=T38FaxVersion:0
> a=T38MaxBitRate:9600
> a=T38FaxFillBitRemoval:0
> a=T38FaxTranscodingMMR:0
> a=T38FaxTranscodingJBIG:0
> a=T38FaxRateManagement:transferredTCF
> a=T38FaxMaxBuffer:176
> a=T38FaxMaxDatagram:176
> a=T38FaxUdpEC:t38UDPRedundancy
>
> 2009-10-06 19:03:47.049284 [DEBUG] switch_core_session.c:630 Send signal
> sofia/internal/214 at pbx0.tssec.lan [BREAK]
> 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3359 Channel
> sofia/internal/214 at pbx0.tssec.lan entering state [calling][0]
> 2009-10-06 19:03:47.077291 [DEBUG] sofia_glue.c:1101 Remote address:port
> [192.168.50.51:16474] has not changed.
> 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3158 Passing 200 OK to other leg
> 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3170
> sofia/external/*408774957558438 at sip.callwithus.com receive message [RESPOND]
> 2009-10-06 19:03:47.077291 [DEBUG] mod_sofia.c:1427 Responding with 200 [OK]
> 2009-10-06 19:03:47.077291 [DEBUG] sofia_glue.c:1322
> sofia/external/*408774957558438 at sip.callwithus.com Patched SDP
> ---
> v=0
> o=- 216899808 216899808 IN IP4 192.168.50.51
> s=-
> c=IN IP4 192.168.50.51
> t=0 0
> m=image 16474 udptl t38
> a=T38FaxVersion:0
> a=T38MaxBitRate:14400
> a=T38FaxRateManagement:transferredTCF
> a=T38FaxMaxBuffer:200
> a=T38FaxMaxDatagram:200
> a=T38FaxUdpEC:t38UDPRedundancy
>
> +++
> v=0
> o=- 216899808 216899808 IN IP4 192.168.50.51
> s=-
> c=IN IP4 77.239.230.202
> t=0 0
> m=image 32404 udptl t38
> a=T38FaxVersion:0
> a=T38MaxBitRate:14400
> a=T38FaxRateManagement:transferredTCF
> a=T38FaxMaxBuffer:200
> a=T38FaxMaxDatagram:200
> a=T38FaxUdpEC:t38UDPRedundancy
>
> 2009-10-06 19:03:47.077291 [DEBUG] sofia_glue.c:1110 AUDIO RTP CHANGING
> DEST TO: [32.50.63.192:14104]
> 2009-10-06 19:03:47.077291 [DEBUG] switch_core_session.c:630 Send signal
> sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
> 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3359 Channel
> sofia/internal/214 at pbx0.tssec.lan entering state [ready][200]
> 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3366 Remote SDP:
> v=0
> o=- 216899808 216899808 IN IP4 192.168.50.51
> s=-
> c=IN IP4 192.168.50.51
> t=0 0
> m=image 16474 udptl t38
> a=T38FaxVersion:0
> a=T38MaxBitRate:14400
> a=T38FaxRateManagement:transferredTCF
> a=T38FaxMaxBuffer:200
> a=T38FaxMaxDatagram:200
> a=T38FaxUdpEC:t38UDPRedundancy
>
> send 1003 bytes to udp/[38.99.70.232]:5060 at 15:06:05.605615:
> ? ------------------------------------------------------------------------
> ? SIP/2.0 200 OK
> ? Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.0
> ? Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK41d48157;rport=5062
> ? From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
> ? To: "Mihailova Ludmila"
> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
> ? Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
> ? CSeq: 102 INVITE
> ? Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
> ? Accept: application/sdp
> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
> ? Supported: timer, precondition, path, replaces
> ? Content-Type: application/sdp
> ? Content-Disposition: session
> ? Content-Length: 274
>
> ? v=0
> ? o=- 216896551 216896552 IN IP4 192.168.50.51
> ? s=-
> ? c=IN IP4 77.239.230.202
> ? t=0 0
> ? m=image 32404 udptl t38
> ? a=T38FaxVersion:0
> ? a=T38MaxBitRate:14400
> ? a=T38FaxRateManagement:transferredTCF
> ? a=T38FaxMaxBuffer:200
> ? a=T38FaxMaxDatagram:200
> ? a=T38FaxUdpEC:t38UDPRedundancy
> ? ------------------------------------------------------------------------
> 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3359 Channel
> sofia/external/*408774957558438 at sip.callwithus.com entering state
> [completed][200]
> 2009-10-06 19:03:47.285347 [INFO] switch_rtp.c:1905 Auto Changing port
> from 32.50.63.192:14104 to 4.55.17.66:14104
> recv 527 bytes from udp/[38.99.70.232]:5060 at 15:06:05.817672:
> ? ------------------------------------------------------------------------
> ? ACK sip:gw+callwithus at 192.168.50.10:5080;transport=udp SIP/2.0
> ? Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.2
> ? Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK5ffd72cd;rport=5062
> ? From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
> ? To: "Mihailova Ludmila"
> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
> ? Contact: <sip:*408774957558438 at 204.74.213.1:5062>
> ? Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
> ? CSeq: 102 ACK
> ? User-Agent: CWU SIP-GW
> ? Max-Forwards: 69
> ? Content-Length: 0
>
> ? ------------------------------------------------------------------------
> 2009-10-06 19:03:47.289349 [DEBUG] sofia.c:3359 Channel
> sofia/external/*408774957558438 at sip.callwithus.com entering state
> [ready][200]
> recv 476 bytes from udp/[38.99.70.232]:5060 at 15:06:47.588933:
> ? ------------------------------------------------------------------------
> ? BYE sip:gw+callwithus at 192.168.50.10:5080;transport=udp SIP/2.0
> ? Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKd406.f8553e14.0
> ? Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK0f2cff01;rport=5062
> ? From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
> ? To: "Mihailova Ludmila"
> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
> ? Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
> ? CSeq: 103 BYE
> ? User-Agent: CWU SIP-GW
> ? Max-Forwards: 69
> ? Content-Length: 0
>
> ? ------------------------------------------------------------------------
> 2009-10-06 19:04:29.060609 [NOTICE] sofia.c:328 Hangup
> sofia/external/*408774957558438 at sip.callwithus.com [CS_EXCHANGE_MEDIA]
> [NORMAL_CLEARING]
> 2009-10-06 19:04:29.060609 [DEBUG] switch_channel.c:1726 Send signal
> sofia/external/*408774957558438 at sip.callwithus.com [KILL]
> 2009-10-06 19:04:29.060609 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
> 2009-10-06 19:04:29.060609 [DEBUG] switch_core_state_machine.c:437
> thread mismatch skipping state handler.
> send 574 bytes to udp/[38.99.70.232]:5060 at 15:06:47.588933:
> ? ------------------------------------------------------------------------
> ? SIP/2.0 200 OK
> ? Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKd406.f8553e14.0
> ? Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK0f2cff01;rport=5062
> ? From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
> ? To: "Mihailova Ludmila"
> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
> ? Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
> ? CSeq: 103 BYE
> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
> ? Supported: timer, precondition, path, replaces
> ? Content-Length: 0
>
> ? ------------------------------------------------------------------------
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From brian at freeswitch.org  Wed Oct  7 13:09:45 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 7 Oct 2009 15:09:45 -0500
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
Message-ID: <82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>

 From what I have been told h323plus is a based/fork of OpenH323 which  
OPAL is just a continuation of OpenH323.  So why not support the  
developers of OPAL/OpenH323 ?

/b

On Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote:

> We are developing module to handle h323 proto now, we try to use  
> mod_opal and try improve it, but no luck,
> there is many issues in libopal, and finaly we now move to h323plus  
> library.



From xengelpublicx at gmail.com  Wed Oct  7 14:26:06 2009
From: xengelpublicx at gmail.com (Vladimir Elizarov)
Date: Thu, 08 Oct 2009 01:26:06 +0400
Subject: [Freeswitch-users] linksys spa-3102 fax t38 + freeswitch media
 proxy
In-Reply-To: <2d9149cd0910071235k27c1429v2bc555c0e7397fc4@mail.gmail.com>
References: <4ACB9519.9000805@gmail.com>
	<2d9149cd0910071235k27c1429v2bc555c0e7397fc4@mail.gmail.com>
Message-ID: <4ACD076E.2080107@gmail.com>

Kristian Kielhofner ?????:
> Since no one else has responded I'll chime in with some general advice.
>
> It's troubling to see that your "provider" is using Asterisk to face
> you (the customer).
>
> I've never had any luck getting T.38 to work (at all, in any mode)
> using Asterisk.  I've heard of other people making it work but
> consider the chain here:
>
> Their provider (they appear to be using DIDx, hmmm) -> Callwithus
> (using Asterisk) -> You (FreeSWITCH) -> Your device
>
> I don't see anything specific in your case but there are many, many
> places for T.38 to go wrong in this scenario.  I'd recommend
> attempting T.38 against a local gateway or known Tier 1 provider
> supporting T.38 with known equipment.
>   
Can you advise sip-providers offering t38?
> Otherwise, do you have any updates for us?
>   
No. I can not add anything.


> On Tue, Oct 6, 2009 at 3:06 PM, Vladimir Elizarov
> <xengelpublicx at gmail.com> wrote:
>   
>> Hello.
>>
>> I want to set up faxing via the gateway linksys spa-3102 (with support
>> for t38) via SIP.
>>
>> SIP-client -> linksys spa3102 -> fs -> provider
>>
>>    <gateway name="callwithus">
>>        ....
>>        <param name="inbound-late-negotiation" value="true"/>
>>        <param name="stun-enabled" value="true"/>
>>    </gateway>
>>
>> <extension name="Fax">
>>    <condition field="destination_number" expression="^(30)(\d{11})$">
>>            <action application="set" data="proxy_media=true"/>
>>            <action application="set" data="bypass_media=false"/>
>>            <action application="bridge"
>> data="sofia/gateway/callwithus/*4087$2 at sip.callwithus.com"/>
>>        </condition>
>> </extension>
>>
>> I make a call, press start on the fax. Fax not sent. Get this in the log
>> (UA freeswitch not trunk-12805):
>>
>>   ------------------------------------------------------------------------
>>   INVITE sip:*408774957558438 at sip.callwithus.com SIP/2.0
>>   Via: SIP/2.0/UDP 77.239.230.202:5080;rport;branch=z9hG4bK8yjrHeUprX2Hr
>>   Max-Forwards: 99
>>   From: "Mihailova Ludmila"
>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>   To: <sip:*408774957558438 at sip.callwithus.com>
>>   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>   CSeq: 121302430 INVITE
>>   Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
>>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY
>>   Supported: timer, precondition, path, replaces
>>   Allow-Events: talk, refer
>>   Content-Type: application/sdp
>>   Content-Disposition: session
>>   Content-Length: 250
>>   Remote-Party-ID: "Mihailova Ludmila"
>> <sip:214 at sip.callwithus.com>;party=calling;screen=yes;privacy=off
>>
>>   v=0
>>   o=- 216896551 216896551 IN IP4 192.168.50.51
>>   s=-
>>   c=IN IP4 77.239.230.202
>>   t=0 0
>>   m=audio 32404 RTP/AVP 0 100 101
>>   a=rtpmap:0 PCMU/8000
>>   a=rtpmap:100 NSE/8000
>>   a=fmtp:100 192-193
>>   a=rtpmap:101 telephone-event/8000
>>   a=fmtp:101 0-15
>>   a=ptime:30
>>
>>   ------------------------------------------------------------------------
>> send 1333 bytes to udp/[38.99.70.232]:5060 at 15:05:34.485226:
>>   ------------------------------------------------------------------------
>>   INVITE sip:*408774957558438 at sip.callwithus.com SIP/2.0
>>   Via: SIP/2.0/UDP 77.239.230.202:5080;rport;branch=z9hG4bK97BHK9BtN6r4K
>>   Max-Forwards: 99
>>   From: "Mihailova Ludmila"
>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>   To: <sip:*408774957558438 at sip.callwithus.com>
>>   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>   CSeq: 121302431 INVITE
>>   Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
>>   Expires: 3600
>>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY
>>   Supported: timer, precondition, path, replaces
>>   Allow-Events: talk, refer
>>   Content-Type: application/sdp
>>   Content-Disposition: session
>>   Content-Length: 250
>>   Remote-Party-ID: "Mihailova Ludmila"
>> <sip:214 at sip.callwithus.com>;party=calling;screen=yes;privacy=off
>>
>>   v=0
>>   o=- 216896551 216896551 IN IP4 192.168.50.51
>>   s=-
>>   c=IN IP4 77.239.230.202
>>   t=0 0
>>   m=audio 32404 RTP/AVP 0 100 101
>>   a=rtpmap:0 PCMU/8000
>>   a=rtpmap:100 NSE/8000
>>   a=fmtp:100 192-193
>>   a=rtpmap:101 telephone-event/8000
>>   a=fmtp:101 0-15
>>   a=ptime:30
>>   ------------------------------------------------------------------------
>>   SIP/2.0 100 Giving a try
>>   Via: SIP/2.0/UDP
>> 192.168.50.10:5080;rport=5080;branch=z9hG4bK97BHK9BtN6r4K
>>   From: "Mihailova Ludmila"
>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>   To: <sip:*408774957558438 at sip.callwithus.com>
>>   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>   CSeq: 121302431 INVITE
>>   Server: CWU SIP GW
>>   Content-Length: 0
>>
>>   ------------------------------------------------------------------------
>> recv 836 bytes from udp/[38.99.70.232]:5060 at 15:05:49.641312:
>>   ------------------------------------------------------------------------
>>   SIP/2.0 183 Session Progress
>>   Via: SIP/2.0/UDP
>> 192.168.50.10:5080;rport=5080;branch=z9hG4bK97BHK9BtN6r4K
>>   Record-Route:
>> <sip:38.99.70.232;lr=on;ftag=HXjrDKpHHX8ZN;did=e8e.089fc2b6>
>>   From: "Mihailova Ludmila"
>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>   To: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>   CSeq: 121302431 INVITE
>>   User-Agent: CWU SIP-GW
>>   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>>   Supported: replaces
>>   Contact: <sip:*408774957558438 at 204.74.213.1:5062>
>>   Content-Type: application/sdp
>>   Content-Length: 209
>>
>>   v=0
>>   o=root 31797 31797 IN IP4 4.55.17.66
>>   s=session
>>   c=IN IP4 4.55.17.66
>>   t=0 0
>>   m=audio 14104 RTP/AVP 0 101
>>   a=rtpmap:0 PCMU/8000
>>   a=rtpmap:101 telephone-event/8000
>>   a=fmtp:101 0-16
>>   a=ptime:20
>>   a=sendrecv
>>   ------------------------------------------------------------------------
>> v=0
>> o=root 31797 31797 IN IP4 4.55.17.66
>> s=session
>> c=IN IP4 4.55.17.66
>> t=0 0
>> m=audio 14104 RTP/AVP 0 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>>
>> 2009-10-06 19:03:31.112988 [NOTICE] sofia.c:3449 Pre-Answer
>> sofia/external/*408774957558438 at sip.callwithus.com!
>> 2009-10-06 19:03:31.112988 [DEBUG] switch_channel.c:1822 Send signal
>> sofia/internal/214 at pbx0.tssec.lan [BREAK]
>> 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2095 Set Codec
>> sofia/external/*408774957558438 at sip.callwithus.com PROXY/8000 20 ms 160
>> samples
>> 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2372 PROXY AUDIO RTP
>> [sofia/external/*408774957558438 at sip.callwithus.com]
>> 192.168.50.10:32404->4.55.17.66:14104 codec: 0 ms: 20
>> 2009-10-06 19:03:31.112988 [DEBUG] switch_rtp.c:1163 Not using a timer
>> 2009-10-06 19:03:31.112988 [DEBUG] switch_ivr_originate.c:2154
>> sofia/internal/214 at pbx0.tssec.lan receive message [PROGRESS]
>> 2009-10-06 19:03:31.112988 [INFO] switch_ivr_originate.c:2154 Sending
>> early media
>> 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:1322
>> sofia/internal/214 at pbx0.tssec.lan Patched SDP
>> ---
>> v=0
>> o=root 31797 31797 IN IP4 4.55.17.66
>> s=session
>> c=IN IP4 4.55.17.66
>> t=0 0
>> m=audio 14104 RTP/AVP 0 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>>
>> +++
>> v=0
>> o=root 31797 31797 IN IP4 4.55.17.66
>> s=session
>> c=IN IP4 192.168.50.10
>> t=0 0
>> m=audio 32698 RTP/AVP 0 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>>
>> 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2372 PROXY AUDIO RTP
>> [sofia/internal/214 at pbx0.tssec.lan]
>> 192.168.50.10:32698->192.168.50.51:16474 codec: 0 ms: 30
>> 2009-10-06 19:03:31.112988 [DEBUG] switch_rtp.c:1163 Not using a timer
>> 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2508 Set comfort noise
>> payload to 13
>> 2009-10-06 19:03:31.112988 [DEBUG] sofia.c:3462
>> sofia/internal/214 at pbx0.tssec.lan receive message [PROGRESS]
>> 2009-10-06 19:03:31.116989 [NOTICE] mod_sofia.c:1521 Pre-Answer
>> sofia/internal/214 at pbx0.tssec.lan!
>> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal
>> sofia/internal/214 at pbx0.tssec.lan [BREAK]
>> 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_originate.c:2196 Originate
>> Resulted in Success: [sofia/external/*408774957558438 at sip.callwithus.com]
>> 2009-10-06 19:03:31.116989 [DEBUG] switch_channel.c:182
>> sofia/external/*408774957558438 at sip.callwithus.com receive message
>> [AUDIO_SYNC]
>> 2009-10-06 19:03:31.116989 [DEBUG] switch_channel.c:182
>> sofia/internal/214 at pbx0.tssec.lan receive message [AUDIO_SYNC]
>> 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_bridge.c:911
>> sofia/external/*408774957558438 at sip.callwithus.com receive message [BRIDGE]
>> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal
>> sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
>> 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_bridge.c:918
>> sofia/internal/214 at pbx0.tssec.lan receive message [BRIDGE]
>> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal
>> sofia/internal/214 at pbx0.tssec.lan [BREAK]
>> 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_bridge.c:962
>> (sofia/external/*408774957558438 at sip.callwithus.com) State Change
>> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
>> 2009-10-06 19:03:31.116989 [DEBUG] sofia.c:3359 Channel
>> sofia/internal/214 at pbx0.tssec.lan entering state [early][183]
>> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal
>> sofia/internal/214 at pbx0.tssec.lan [BREAK]
>> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
>> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/*408774957558438 at sip.callwithus.com) Running State
>> Change CS_EXCHANGE_MEDIA
>> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_state_machine.c:343
>> (sofia/external/*408774957558438 at sip.callwithus.com) State EXCHANGE_MEDIA
>> 2009-10-06 19:03:31.116989 [DEBUG] mod_sofia.c:430 SOFIA LOOPBACK
>> recv 822 bytes from udp/[38.99.70.232]:5060 at 15:05:58.503701:
>>   ------------------------------------------------------------------------
>>   SIP/2.0 200 OK
>>   Via: SIP/2.0/UDP
>> 192.168.50.10:5080;rport=5080;branch=z9hG4bK97BHK9BtN6r4K
>>   Record-Route:
>> <sip:38.99.70.232;lr=on;ftag=HXjrDKpHHX8ZN;did=e8e.089fc2b6>
>>   From: "Mihailova Ludmila"
>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>   To: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>   CSeq: 121302431 INVITE
>>   User-Agent: CWU SIP-GW
>>   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>>   Supported: replaces
>>   Contact: <sip:*408774957558438 at 204.74.213.1:5062>
>>   Content-Type: application/sdp
>>   Content-Length: 209
>>
>>   v=0
>>   o=root 31797 31798 IN IP4 4.55.17.66
>>   s=session
>>   c=IN IP4 4.55.17.66
>>   t=0 0
>>   m=audio 14104 RTP/AVP 0 101
>>   a=rtpmap:0 PCMU/8000
>>   a=rtpmap:101 telephone-event/8000
>>   a=fmtp:101 0-16
>>   a=ptime:20
>>   a=sendrecv
>>   ------------------------------------------------------------------------
>> send 762 bytes to udp/[38.99.70.232]:5060 at 15:05:58.507702:
>>   ------------------------------------------------------------------------
>>   ACK sip:*408774957558438 at 204.74.213.1:5062 SIP/2.0
>>   Via: SIP/2.0/UDP 192.168.50.10:5080;rport;branch=z9hG4bKaH59m4vXjFFQF
>>   Route: <sip:38.99.70.232;lr=on;ftag=HXjrDKpHHX8ZN;did=e8e.089fc2b6>
>>   Max-Forwards: 70
>>   From: "Mihailova Ludmila"
>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>   To: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>   CSeq: 121302431 ACK
>>   Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
>>   Content-Length: 0
>>
>>   ------------------------------------------------------------------------
>> 2009-10-06 19:03:39.975377 [DEBUG] sofia.c:3359 Channel
>> sofia/external/*408774957558438 at sip.callwithus.com entering state
>> [ready][200]
>> 2009-10-06 19:03:39.975377 [DEBUG] switch_channel.c:1935 Send signal
>> sofia/internal/214 at pbx0.tssec.lan [BREAK]
>> 2009-10-06 19:03:39.975377 [NOTICE] sofia.c:3814 Channel
>> [sofia/external/*408774957558438 at sip.callwithus.com] has been answered
>> 2009-10-06 19:03:39.975377 [DEBUG] switch_channel.c:182
>> sofia/external/*408774957558438 at sip.callwithus.com receive message
>> [AUDIO_SYNC]
>> 2009-10-06 19:03:39.975377 [DEBUG] sofia.c:3819
>> sofia/internal/214 at pbx0.tssec.lan receive message [ANSWER]
>> 2009-10-06 19:03:39.975377 [DEBUG] sofia_glue.c:1322
>> sofia/internal/214 at pbx0.tssec.lan Patched SDP
>> ---
>> v=0
>> o=root 31797 31797 IN IP4 4.55.17.66
>> s=session
>> c=IN IP4 4.55.17.66
>> t=0 0
>> m=audio 14104 RTP/AVP 0 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>>
>> +++
>> v=0
>> o=root 31797 31797 IN IP4 4.55.17.66
>> s=session
>> c=IN IP4 192.168.50.10
>> t=0 0
>> m=audio 32698 RTP/AVP 0 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>>
>> 2009-10-06 19:03:39.975377 [DEBUG] switch_core_session.c:630 Send signal
>> sofia/internal/214 at pbx0.tssec.lan [BREAK]
>> 2009-10-06 19:03:39.975377 [NOTICE] sofia.c:3819 Channel
>> [sofia/internal/214 at pbx0.tssec.lan] has been answered
>> 2009-10-06 19:03:39.975377 [DEBUG] switch_channel.c:182
>> sofia/internal/214 at pbx0.tssec.lan receive message [AUDIO_SYNC]
>> 2009-10-06 19:03:39.975377 [DEBUG] sofia.c:3359 Channel
>> sofia/internal/214 at pbx0.tssec.lan entering state [completed][200]
>> 2009-10-06 19:03:39.991381 [DEBUG] sofia.c:3359 Channel
>> sofia/internal/214 at pbx0.tssec.lan entering state [ready][200]
>> recv 1054 bytes from udp/[38.99.70.232]:5060 at 15:06:05.577607:
>>   ------------------------------------------------------------------------
>>   INVITE sip:gw+callwithus at 192.168.50.10:5080;transport=udp SIP/2.0
>>   Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.0
>>   Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK41d48157;rport=5062
>>   From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>   To: "Mihailova Ludmila"
>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>   Contact: <sip:*408774957558438 at 204.74.213.1:5062>
>>   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>   CSeq: 102 INVITE
>>   User-Agent: CWU SIP-GW
>>   Max-Forwards: 69
>>   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>>   Supported: replaces
>>   X-asterisk-info: SIP re-invite (T38 switchover)
>>   Content-Type: application/sdp
>>   Content-Length: 350
>>
>>   v=0
>>   o=root 31797 31799 IN IP4 32.50.63.192
>>   s=session
>>   c=IN IP4 32.50.63.192
>>   t=0 0
>>   m=image 14104 udptl t38
>>   a=T38FaxVersion:0
>>   a=T38MaxBitRate:9600
>>   a=T38FaxFillBitRemoval:0
>>   a=T38FaxTranscodingMMR:0
>>   a=T38FaxTranscodingJBIG:0
>>   a=T38FaxRateManagement:transferredTCF
>>   a=T38FaxMaxBuffer:176
>>   a=T38FaxMaxDatagram:176
>>   a=T38FaxUdpEC:t38UDPRedundancy
>>   ------------------------------------------------------------------------
>> send 443 bytes to udp/[38.99.70.232]:5060 at 15:06:05.577607:
>>   ------------------------------------------------------------------------
>>   SIP/2.0 100 Trying
>>   Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.0
>>   Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK41d48157;rport=5062
>>   From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>   To: "Mihailova Ludmila"
>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>   CSeq: 102 INVITE
>>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
>>   Content-Length: 0
>>
>>   ------------------------------------------------------------------------
>> 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3359 Channel
>> sofia/external/*408774957558438 at sip.callwithus.com entering state
>> [received][100]
>> 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3366 Remote SDP:
>> v=0
>> o=root 31797 31799 IN IP4 32.50.63.192
>> s=session
>> c=IN IP4 32.50.63.192
>> t=0 0
>> m=image 14104 udptl t38
>> a=T38FaxVersion:0
>> a=T38MaxBitRate:9600
>> a=T38FaxFillBitRemoval:0
>> a=T38FaxTranscodingMMR:0
>> a=T38FaxTranscodingJBIG:0
>> a=T38FaxRateManagement:transferredTCF
>> a=T38FaxMaxBuffer:176
>> a=T38FaxMaxDatagram:176
>> a=T38FaxUdpEC:t38UDPRedundancy
>>
>> 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3683 Passing SDP to other leg.
>> v=0
>> o=root 31797 31799 IN IP4 32.50.63.192
>> s=session
>> c=IN IP4 32.50.63.192
>> t=0 0
>> m=image 14104 udptl t38
>> a=T38FaxVersion:0
>> a=T38MaxBitRate:9600
>> a=T38FaxFillBitRemoval:0
>> a=T38FaxTranscodingMMR:0
>> a=T38FaxTranscodingJBIG:0
>> a=T38FaxRateManagement:transferredTCF
>> a=T38FaxMaxBuffer:176
>> a=T38FaxMaxDatagram:176
>> a=T38FaxUdpEC:t38UDPRedundancy
>>
>> 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3696
>> sofia/internal/214 at pbx0.tssec.lan receive message [MEDIA_REDIRECT]
>> 2009-10-06 19:03:47.049284 [DEBUG] mod_sofia.c:1195
>> sofia/internal/214 at pbx0.tssec.lan Sending media re-direct:
>> v=0
>> o=root 31797 31799 IN IP4 32.50.63.192
>> s=session
>> c=IN IP4 32.50.63.192
>> t=0 0
>> m=image 14104 udptl t38
>> a=T38FaxVersion:0
>> a=T38MaxBitRate:9600
>> a=T38FaxFillBitRemoval:0
>> a=T38FaxTranscodingMMR:0
>> a=T38FaxTranscodingJBIG:0
>> a=T38FaxRateManagement:transferredTCF
>> a=T38FaxMaxBuffer:176
>> a=T38FaxMaxDatagram:176
>> a=T38FaxUdpEC:t38UDPRedundancy
>>
>> 2009-10-06 19:03:47.049284 [DEBUG] sofia_glue.c:1101 Remote address:port
>> [192.168.50.51:16474] has not changed.
>> 2009-10-06 19:03:47.049284 [DEBUG] sofia_glue.c:1322
>> sofia/internal/214 at pbx0.tssec.lan Patched SDP
>> ---
>> v=0
>> o=root 31797 31799 IN IP4 32.50.63.192
>> s=session
>> c=IN IP4 32.50.63.192
>> t=0 0
>> m=image 14104 udptl t38
>> a=T38FaxVersion:0
>> a=T38MaxBitRate:9600
>> a=T38FaxFillBitRemoval:0
>> a=T38FaxTranscodingMMR:0
>> a=T38FaxTranscodingJBIG:0
>> a=T38FaxRateManagement:transferredTCF
>> a=T38FaxMaxBuffer:176
>> a=T38FaxMaxDatagram:176
>> a=T38FaxUdpEC:t38UDPRedundancy
>>
>> +++
>> v=0
>> o=root 31797 31799 IN IP4 32.50.63.192
>> s=session
>> c=IN IP4 192.168.50.10
>> t=0 0
>> m=image 32698 udptl t38
>> a=T38FaxVersion:0
>> a=T38MaxBitRate:9600
>> a=T38FaxFillBitRemoval:0
>> a=T38FaxTranscodingMMR:0
>> a=T38FaxTranscodingJBIG:0
>> a=T38FaxRateManagement:transferredTCF
>> a=T38FaxMaxBuffer:176
>> a=T38FaxMaxDatagram:176
>> a=T38FaxUdpEC:t38UDPRedundancy
>>
>> 2009-10-06 19:03:47.049284 [DEBUG] switch_core_session.c:630 Send signal
>> sofia/internal/214 at pbx0.tssec.lan [BREAK]
>> 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3359 Channel
>> sofia/internal/214 at pbx0.tssec.lan entering state [calling][0]
>> 2009-10-06 19:03:47.077291 [DEBUG] sofia_glue.c:1101 Remote address:port
>> [192.168.50.51:16474] has not changed.
>> 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3158 Passing 200 OK to other leg
>> 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3170
>> sofia/external/*408774957558438 at sip.callwithus.com receive message [RESPOND]
>> 2009-10-06 19:03:47.077291 [DEBUG] mod_sofia.c:1427 Responding with 200 [OK]
>> 2009-10-06 19:03:47.077291 [DEBUG] sofia_glue.c:1322
>> sofia/external/*408774957558438 at sip.callwithus.com Patched SDP
>> ---
>> v=0
>> o=- 216899808 216899808 IN IP4 192.168.50.51
>> s=-
>> c=IN IP4 192.168.50.51
>> t=0 0
>> m=image 16474 udptl t38
>> a=T38FaxVersion:0
>> a=T38MaxBitRate:14400
>> a=T38FaxRateManagement:transferredTCF
>> a=T38FaxMaxBuffer:200
>> a=T38FaxMaxDatagram:200
>> a=T38FaxUdpEC:t38UDPRedundancy
>>
>> +++
>> v=0
>> o=- 216899808 216899808 IN IP4 192.168.50.51
>> s=-
>> c=IN IP4 77.239.230.202
>> t=0 0
>> m=image 32404 udptl t38
>> a=T38FaxVersion:0
>> a=T38MaxBitRate:14400
>> a=T38FaxRateManagement:transferredTCF
>> a=T38FaxMaxBuffer:200
>> a=T38FaxMaxDatagram:200
>> a=T38FaxUdpEC:t38UDPRedundancy
>>
>> 2009-10-06 19:03:47.077291 [DEBUG] sofia_glue.c:1110 AUDIO RTP CHANGING
>> DEST TO: [32.50.63.192:14104]
>> 2009-10-06 19:03:47.077291 [DEBUG] switch_core_session.c:630 Send signal
>> sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
>> 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3359 Channel
>> sofia/internal/214 at pbx0.tssec.lan entering state [ready][200]
>> 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3366 Remote SDP:
>> v=0
>> o=- 216899808 216899808 IN IP4 192.168.50.51
>> s=-
>> c=IN IP4 192.168.50.51
>> t=0 0
>> m=image 16474 udptl t38
>> a=T38FaxVersion:0
>> a=T38MaxBitRate:14400
>> a=T38FaxRateManagement:transferredTCF
>> a=T38FaxMaxBuffer:200
>> a=T38FaxMaxDatagram:200
>> a=T38FaxUdpEC:t38UDPRedundancy
>>
>> send 1003 bytes to udp/[38.99.70.232]:5060 at 15:06:05.605615:
>>   ------------------------------------------------------------------------
>>   SIP/2.0 200 OK
>>   Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.0
>>   Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK41d48157;rport=5062
>>   From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>   To: "Mihailova Ludmila"
>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>   CSeq: 102 INVITE
>>   Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
>>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
>>   Accept: application/sdp
>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY
>>   Supported: timer, precondition, path, replaces
>>   Content-Type: application/sdp
>>   Content-Disposition: session
>>   Content-Length: 274
>>
>>   v=0
>>   o=- 216896551 216896552 IN IP4 192.168.50.51
>>   s=-
>>   c=IN IP4 77.239.230.202
>>   t=0 0
>>   m=image 32404 udptl t38
>>   a=T38FaxVersion:0
>>   a=T38MaxBitRate:14400
>>   a=T38FaxRateManagement:transferredTCF
>>   a=T38FaxMaxBuffer:200
>>   a=T38FaxMaxDatagram:200
>>   a=T38FaxUdpEC:t38UDPRedundancy
>>   ------------------------------------------------------------------------
>> 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3359 Channel
>> sofia/external/*408774957558438 at sip.callwithus.com entering state
>> [completed][200]
>> 2009-10-06 19:03:47.285347 [INFO] switch_rtp.c:1905 Auto Changing port
>> from 32.50.63.192:14104 to 4.55.17.66:14104
>> recv 527 bytes from udp/[38.99.70.232]:5060 at 15:06:05.817672:
>>   ------------------------------------------------------------------------
>>   ACK sip:gw+callwithus at 192.168.50.10:5080;transport=udp SIP/2.0
>>   Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.2
>>   Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK5ffd72cd;rport=5062
>>   From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>   To: "Mihailova Ludmila"
>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>   Contact: <sip:*408774957558438 at 204.74.213.1:5062>
>>   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>   CSeq: 102 ACK
>>   User-Agent: CWU SIP-GW
>>   Max-Forwards: 69
>>   Content-Length: 0
>>
>>   ------------------------------------------------------------------------
>> 2009-10-06 19:03:47.289349 [DEBUG] sofia.c:3359 Channel
>> sofia/external/*408774957558438 at sip.callwithus.com entering state
>> [ready][200]
>> recv 476 bytes from udp/[38.99.70.232]:5060 at 15:06:47.588933:
>>   ------------------------------------------------------------------------
>>   BYE sip:gw+callwithus at 192.168.50.10:5080;transport=udp SIP/2.0
>>   Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKd406.f8553e14.0
>>   Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK0f2cff01;rport=5062
>>   From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>   To: "Mihailova Ludmila"
>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>   CSeq: 103 BYE
>>   User-Agent: CWU SIP-GW
>>   Max-Forwards: 69
>>   Content-Length: 0
>>
>>   ------------------------------------------------------------------------
>> 2009-10-06 19:04:29.060609 [NOTICE] sofia.c:328 Hangup
>> sofia/external/*408774957558438 at sip.callwithus.com [CS_EXCHANGE_MEDIA]
>> [NORMAL_CLEARING]
>> 2009-10-06 19:04:29.060609 [DEBUG] switch_channel.c:1726 Send signal
>> sofia/external/*408774957558438 at sip.callwithus.com [KILL]
>> 2009-10-06 19:04:29.060609 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
>> 2009-10-06 19:04:29.060609 [DEBUG] switch_core_state_machine.c:437
>> thread mismatch skipping state handler.
>> send 574 bytes to udp/[38.99.70.232]:5060 at 15:06:47.588933:
>>   ------------------------------------------------------------------------
>>   SIP/2.0 200 OK
>>   Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKd406.f8553e14.0
>>   Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK0f2cff01;rport=5062
>>   From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>   To: "Mihailova Ludmila"
>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>   CSeq: 103 BYE
>>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY
>>   Supported: timer, precondition, path, replaces
>>   Content-Length: 0
>>
>>   ------------------------------------------------------------------------
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>     
>
>
>
>   


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From rob4manhere at gmail.com  Wed Oct  7 17:11:54 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Wed, 7 Oct 2009 19:11:54 -0500
Subject: [Freeswitch-users] linksys spa-3102 fax t38 + freeswitch media
	proxy
In-Reply-To: <4ACD076E.2080107@gmail.com>
References: <4ACB9519.9000805@gmail.com>
	<2d9149cd0910071235k27c1429v2bc555c0e7397fc4@mail.gmail.com>
	<4ACD076E.2080107@gmail.com>
Message-ID: <5B417827-6CCB-4EE4-9524-5CD45AC9E28F@gmail.com>

> Can you advise sip-providers offering t38?

Gafachi has T38 fax support.


On Oct 7, 2009, at 4:26 PM, Vladimir Elizarov wrote:

> Kristian Kielhofner ?????:
>> Since no one else has responded I'll chime in with some general  
>> advice.
>>
>> It's troubling to see that your "provider" is using Asterisk to face
>> you (the customer).
>>
>> I've never had any luck getting T.38 to work (at all, in any mode)
>> using Asterisk.  I've heard of other people making it work but
>> consider the chain here:
>>
>> Their provider (they appear to be using DIDx, hmmm) -> Callwithus
>> (using Asterisk) -> You (FreeSWITCH) -> Your device
>>
>> I don't see anything specific in your case but there are many, many
>> places for T.38 to go wrong in this scenario.  I'd recommend
>> attempting T.38 against a local gateway or known Tier 1 provider
>> supporting T.38 with known equipment.
>>
> Can you advise sip-providers offering t38?
>> Otherwise, do you have any updates for us?
>>
> No. I can not add anything.
>
>
>> On Tue, Oct 6, 2009 at 3:06 PM, Vladimir Elizarov
>> <xengelpublicx at gmail.com> wrote:
>>
>>> Hello.
>>>
>>> I want to set up faxing via the gateway linksys spa-3102 (with  
>>> support
>>> for t38) via SIP.
>>>
>>> SIP-client -> linksys spa3102 -> fs -> provider
>>>
>>>   <gateway name="callwithus">
>>>       ....
>>>       <param name="inbound-late-negotiation" value="true"/>
>>>       <param name="stun-enabled" value="true"/>
>>>   </gateway>
>>>
>>> <extension name="Fax">
>>>   <condition field="destination_number" expression="^(30)(\d{11})$">
>>>           <action application="set" data="proxy_media=true"/>
>>>           <action application="set" data="bypass_media=false"/>
>>>           <action application="bridge"
>>> data="sofia/gateway/callwithus/*4087$2 at sip.callwithus.com"/>
>>>       </condition>
>>> </extension>
>>>
>>> I make a call, press start on the fax. Fax not sent. Get this in  
>>> the log
>>> (UA freeswitch not trunk-12805):
>>>
>>>   
>>> ------------------------------------------------------------------------
>>>  INVITE sip:*408774957558438 at sip.callwithus.com SIP/2.0
>>>  Via: SIP/2.0/UDP  
>>> 77.239.230.202:5080;rport;branch=z9hG4bK8yjrHeUprX2Hr
>>>  Max-Forwards: 99
>>>  From: "Mihailova Ludmila"
>>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>>  To: <sip:*408774957558438 at sip.callwithus.com>
>>>  Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>>  CSeq: 121302430 INVITE
>>>  Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
>>>  User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
>>>  Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>> REGISTER, REFER, NOTIFY
>>>  Supported: timer, precondition, path, replaces
>>>  Allow-Events: talk, refer
>>>  Content-Type: application/sdp
>>>  Content-Disposition: session
>>>  Content-Length: 250
>>>  Remote-Party-ID: "Mihailova Ludmila"
>>> <sip:214 at sip.callwithus.com>;party=calling;screen=yes;privacy=off
>>>
>>>  v=0
>>>  o=- 216896551 216896551 IN IP4 192.168.50.51
>>>  s=-
>>>  c=IN IP4 77.239.230.202
>>>  t=0 0
>>>  m=audio 32404 RTP/AVP 0 100 101
>>>  a=rtpmap:0 PCMU/8000
>>>  a=rtpmap:100 NSE/8000
>>>  a=fmtp:100 192-193
>>>  a=rtpmap:101 telephone-event/8000
>>>  a=fmtp:101 0-15
>>>  a=ptime:30
>>>
>>>   
>>> ------------------------------------------------------------------------
>>> send 1333 bytes to udp/[38.99.70.232]:5060 at 15:05:34.485226:
>>>   
>>> ------------------------------------------------------------------------
>>>  INVITE sip:*408774957558438 at sip.callwithus.com SIP/2.0
>>>  Via: SIP/2.0/UDP  
>>> 77.239.230.202:5080;rport;branch=z9hG4bK97BHK9BtN6r4K
>>>  Max-Forwards: 99
>>>  From: "Mihailova Ludmila"
>>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>>  To: <sip:*408774957558438 at sip.callwithus.com>
>>>  Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>>  CSeq: 121302431 INVITE
>>>  Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
>>>  Expires: 3600
>>>  User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
>>>  Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>> REGISTER, REFER, NOTIFY
>>>  Supported: timer, precondition, path, replaces
>>>  Allow-Events: talk, refer
>>>  Content-Type: application/sdp
>>>  Content-Disposition: session
>>>  Content-Length: 250
>>>  Remote-Party-ID: "Mihailova Ludmila"
>>> <sip:214 at sip.callwithus.com>;party=calling;screen=yes;privacy=off
>>>
>>>  v=0
>>>  o=- 216896551 216896551 IN IP4 192.168.50.51
>>>  s=-
>>>  c=IN IP4 77.239.230.202
>>>  t=0 0
>>>  m=audio 32404 RTP/AVP 0 100 101
>>>  a=rtpmap:0 PCMU/8000
>>>  a=rtpmap:100 NSE/8000
>>>  a=fmtp:100 192-193
>>>  a=rtpmap:101 telephone-event/8000
>>>  a=fmtp:101 0-15
>>>  a=ptime:30
>>>   
>>> ------------------------------------------------------------------------
>>>  SIP/2.0 100 Giving a try
>>>  Via: SIP/2.0/UDP
>>> 192.168.50.10:5080;rport=5080;branch=z9hG4bK97BHK9BtN6r4K
>>>  From: "Mihailova Ludmila"
>>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>>  To: <sip:*408774957558438 at sip.callwithus.com>
>>>  Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>>  CSeq: 121302431 INVITE
>>>  Server: CWU SIP GW
>>>  Content-Length: 0
>>>
>>>   
>>> ------------------------------------------------------------------------
>>> recv 836 bytes from udp/[38.99.70.232]:5060 at 15:05:49.641312:
>>>   
>>> ------------------------------------------------------------------------
>>>  SIP/2.0 183 Session Progress
>>>  Via: SIP/2.0/UDP
>>> 192.168.50.10:5080;rport=5080;branch=z9hG4bK97BHK9BtN6r4K
>>>  Record-Route:
>>> <sip:38.99.70.232;lr=on;ftag=HXjrDKpHHX8ZN;did=e8e.089fc2b6>
>>>  From: "Mihailova Ludmila"
>>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>>  To: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>>  Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>>  CSeq: 121302431 INVITE
>>>  User-Agent: CWU SIP-GW
>>>  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>>>  Supported: replaces
>>>  Contact: <sip:*408774957558438 at 204.74.213.1:5062>
>>>  Content-Type: application/sdp
>>>  Content-Length: 209
>>>
>>>  v=0
>>>  o=root 31797 31797 IN IP4 4.55.17.66
>>>  s=session
>>>  c=IN IP4 4.55.17.66
>>>  t=0 0
>>>  m=audio 14104 RTP/AVP 0 101
>>>  a=rtpmap:0 PCMU/8000
>>>  a=rtpmap:101 telephone-event/8000
>>>  a=fmtp:101 0-16
>>>  a=ptime:20
>>>  a=sendrecv
>>>   
>>> ------------------------------------------------------------------------
>>> v=0
>>> o=root 31797 31797 IN IP4 4.55.17.66
>>> s=session
>>> c=IN IP4 4.55.17.66
>>> t=0 0
>>> m=audio 14104 RTP/AVP 0 101
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>>
>>> 2009-10-06 19:03:31.112988 [NOTICE] sofia.c:3449 Pre-Answer
>>> sofia/external/*408774957558438 at sip.callwithus.com!
>>> 2009-10-06 19:03:31.112988 [DEBUG] switch_channel.c:1822 Send signal
>>> sofia/internal/214 at pbx0.tssec.lan [BREAK]
>>> 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2095 Set Codec
>>> sofia/external/*408774957558438 at sip.callwithus.com PROXY/8000 20  
>>> ms 160
>>> samples
>>> 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2372 PROXY AUDIO RTP
>>> [sofia/external/*408774957558438 at sip.callwithus.com]
>>> 192.168.50.10:32404->4.55.17.66:14104 codec: 0 ms: 20
>>> 2009-10-06 19:03:31.112988 [DEBUG] switch_rtp.c:1163 Not using a  
>>> timer
>>> 2009-10-06 19:03:31.112988 [DEBUG] switch_ivr_originate.c:2154
>>> sofia/internal/214 at pbx0.tssec.lan receive message [PROGRESS]
>>> 2009-10-06 19:03:31.112988 [INFO] switch_ivr_originate.c:2154  
>>> Sending
>>> early media
>>> 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:1322
>>> sofia/internal/214 at pbx0.tssec.lan Patched SDP
>>> ---
>>> v=0
>>> o=root 31797 31797 IN IP4 4.55.17.66
>>> s=session
>>> c=IN IP4 4.55.17.66
>>> t=0 0
>>> m=audio 14104 RTP/AVP 0 101
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>>
>>> +++
>>> v=0
>>> o=root 31797 31797 IN IP4 4.55.17.66
>>> s=session
>>> c=IN IP4 192.168.50.10
>>> t=0 0
>>> m=audio 32698 RTP/AVP 0 101
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>>
>>> 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2372 PROXY AUDIO RTP
>>> [sofia/internal/214 at pbx0.tssec.lan]
>>> 192.168.50.10:32698->192.168.50.51:16474 codec: 0 ms: 30
>>> 2009-10-06 19:03:31.112988 [DEBUG] switch_rtp.c:1163 Not using a  
>>> timer
>>> 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2508 Set comfort  
>>> noise
>>> payload to 13
>>> 2009-10-06 19:03:31.112988 [DEBUG] sofia.c:3462
>>> sofia/internal/214 at pbx0.tssec.lan receive message [PROGRESS]
>>> 2009-10-06 19:03:31.116989 [NOTICE] mod_sofia.c:1521 Pre-Answer
>>> sofia/internal/214 at pbx0.tssec.lan!
>>> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send  
>>> signal
>>> sofia/internal/214 at pbx0.tssec.lan [BREAK]
>>> 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_originate.c:2196  
>>> Originate
>>> Resulted in Success: [sofia/external/*408774957558438 at sip.callwithus.com 
>>> ]
>>> 2009-10-06 19:03:31.116989 [DEBUG] switch_channel.c:182
>>> sofia/external/*408774957558438 at sip.callwithus.com receive message
>>> [AUDIO_SYNC]
>>> 2009-10-06 19:03:31.116989 [DEBUG] switch_channel.c:182
>>> sofia/internal/214 at pbx0.tssec.lan receive message [AUDIO_SYNC]
>>> 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_bridge.c:911
>>> sofia/external/*408774957558438 at sip.callwithus.com receive message  
>>> [BRIDGE]
>>> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send  
>>> signal
>>> sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
>>> 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_bridge.c:918
>>> sofia/internal/214 at pbx0.tssec.lan receive message [BRIDGE]
>>> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send  
>>> signal
>>> sofia/internal/214 at pbx0.tssec.lan [BREAK]
>>> 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_bridge.c:962
>>> (sofia/external/*408774957558438 at sip.callwithus.com) State Change
>>> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
>>> 2009-10-06 19:03:31.116989 [DEBUG] sofia.c:3359 Channel
>>> sofia/internal/214 at pbx0.tssec.lan entering state [early][183]
>>> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send  
>>> signal
>>> sofia/internal/214 at pbx0.tssec.lan [BREAK]
>>> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:932 Send  
>>> signal
>>> sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
>>> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/*408774957558438 at sip.callwithus.com) Running State
>>> Change CS_EXCHANGE_MEDIA
>>> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_state_machine.c:343
>>> (sofia/external/*408774957558438 at sip.callwithus.com) State  
>>> EXCHANGE_MEDIA
>>> 2009-10-06 19:03:31.116989 [DEBUG] mod_sofia.c:430 SOFIA LOOPBACK
>>> recv 822 bytes from udp/[38.99.70.232]:5060 at 15:05:58.503701:
>>>   
>>> ------------------------------------------------------------------------
>>>  SIP/2.0 200 OK
>>>  Via: SIP/2.0/UDP
>>> 192.168.50.10:5080;rport=5080;branch=z9hG4bK97BHK9BtN6r4K
>>>  Record-Route:
>>> <sip:38.99.70.232;lr=on;ftag=HXjrDKpHHX8ZN;did=e8e.089fc2b6>
>>>  From: "Mihailova Ludmila"
>>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>>  To: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>>  Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>>  CSeq: 121302431 INVITE
>>>  User-Agent: CWU SIP-GW
>>>  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>>>  Supported: replaces
>>>  Contact: <sip:*408774957558438 at 204.74.213.1:5062>
>>>  Content-Type: application/sdp
>>>  Content-Length: 209
>>>
>>>  v=0
>>>  o=root 31797 31798 IN IP4 4.55.17.66
>>>  s=session
>>>  c=IN IP4 4.55.17.66
>>>  t=0 0
>>>  m=audio 14104 RTP/AVP 0 101
>>>  a=rtpmap:0 PCMU/8000
>>>  a=rtpmap:101 telephone-event/8000
>>>  a=fmtp:101 0-16
>>>  a=ptime:20
>>>  a=sendrecv
>>>   
>>> ------------------------------------------------------------------------
>>> send 762 bytes to udp/[38.99.70.232]:5060 at 15:05:58.507702:
>>>   
>>> ------------------------------------------------------------------------
>>>  ACK sip:*408774957558438 at 204.74.213.1:5062 SIP/2.0
>>>  Via: SIP/2.0/UDP  
>>> 192.168.50.10:5080;rport;branch=z9hG4bKaH59m4vXjFFQF
>>>  Route: <sip:38.99.70.232;lr=on;ftag=HXjrDKpHHX8ZN;did=e8e.089fc2b6>
>>>  Max-Forwards: 70
>>>  From: "Mihailova Ludmila"
>>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>>  To: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>>  Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>>  CSeq: 121302431 ACK
>>>  Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
>>>  Content-Length: 0
>>>
>>>   
>>> ------------------------------------------------------------------------
>>> 2009-10-06 19:03:39.975377 [DEBUG] sofia.c:3359 Channel
>>> sofia/external/*408774957558438 at sip.callwithus.com entering state
>>> [ready][200]
>>> 2009-10-06 19:03:39.975377 [DEBUG] switch_channel.c:1935 Send signal
>>> sofia/internal/214 at pbx0.tssec.lan [BREAK]
>>> 2009-10-06 19:03:39.975377 [NOTICE] sofia.c:3814 Channel
>>> [sofia/external/*408774957558438 at sip.callwithus.com] has been  
>>> answered
>>> 2009-10-06 19:03:39.975377 [DEBUG] switch_channel.c:182
>>> sofia/external/*408774957558438 at sip.callwithus.com receive message
>>> [AUDIO_SYNC]
>>> 2009-10-06 19:03:39.975377 [DEBUG] sofia.c:3819
>>> sofia/internal/214 at pbx0.tssec.lan receive message [ANSWER]
>>> 2009-10-06 19:03:39.975377 [DEBUG] sofia_glue.c:1322
>>> sofia/internal/214 at pbx0.tssec.lan Patched SDP
>>> ---
>>> v=0
>>> o=root 31797 31797 IN IP4 4.55.17.66
>>> s=session
>>> c=IN IP4 4.55.17.66
>>> t=0 0
>>> m=audio 14104 RTP/AVP 0 101
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>>
>>> +++
>>> v=0
>>> o=root 31797 31797 IN IP4 4.55.17.66
>>> s=session
>>> c=IN IP4 192.168.50.10
>>> t=0 0
>>> m=audio 32698 RTP/AVP 0 101
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>>
>>> 2009-10-06 19:03:39.975377 [DEBUG] switch_core_session.c:630 Send  
>>> signal
>>> sofia/internal/214 at pbx0.tssec.lan [BREAK]
>>> 2009-10-06 19:03:39.975377 [NOTICE] sofia.c:3819 Channel
>>> [sofia/internal/214 at pbx0.tssec.lan] has been answered
>>> 2009-10-06 19:03:39.975377 [DEBUG] switch_channel.c:182
>>> sofia/internal/214 at pbx0.tssec.lan receive message [AUDIO_SYNC]
>>> 2009-10-06 19:03:39.975377 [DEBUG] sofia.c:3359 Channel
>>> sofia/internal/214 at pbx0.tssec.lan entering state [completed][200]
>>> 2009-10-06 19:03:39.991381 [DEBUG] sofia.c:3359 Channel
>>> sofia/internal/214 at pbx0.tssec.lan entering state [ready][200]
>>> recv 1054 bytes from udp/[38.99.70.232]:5060 at 15:06:05.577607:
>>>   
>>> ------------------------------------------------------------------------
>>>  INVITE sip:gw+callwithus at 192.168.50.10:5080;transport=udp SIP/2.0
>>>  Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.0
>>>  Via: SIP/2.0/UDP  
>>> 204.74.213.1:5062;branch=z9hG4bK41d48157;rport=5062
>>>  From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>>  To: "Mihailova Ludmila"
>>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>>  Contact: <sip:*408774957558438 at 204.74.213.1:5062>
>>>  Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>>  CSeq: 102 INVITE
>>>  User-Agent: CWU SIP-GW
>>>  Max-Forwards: 69
>>>  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>>>  Supported: replaces
>>>  X-asterisk-info: SIP re-invite (T38 switchover)
>>>  Content-Type: application/sdp
>>>  Content-Length: 350
>>>
>>>  v=0
>>>  o=root 31797 31799 IN IP4 32.50.63.192
>>>  s=session
>>>  c=IN IP4 32.50.63.192
>>>  t=0 0
>>>  m=image 14104 udptl t38
>>>  a=T38FaxVersion:0
>>>  a=T38MaxBitRate:9600
>>>  a=T38FaxFillBitRemoval:0
>>>  a=T38FaxTranscodingMMR:0
>>>  a=T38FaxTranscodingJBIG:0
>>>  a=T38FaxRateManagement:transferredTCF
>>>  a=T38FaxMaxBuffer:176
>>>  a=T38FaxMaxDatagram:176
>>>  a=T38FaxUdpEC:t38UDPRedundancy
>>>   
>>> ------------------------------------------------------------------------
>>> send 443 bytes to udp/[38.99.70.232]:5060 at 15:06:05.577607:
>>>   
>>> ------------------------------------------------------------------------
>>>  SIP/2.0 100 Trying
>>>  Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.0
>>>  Via: SIP/2.0/UDP  
>>> 204.74.213.1:5062;branch=z9hG4bK41d48157;rport=5062
>>>  From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>>  To: "Mihailova Ludmila"
>>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>>  Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>>  CSeq: 102 INVITE
>>>  User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
>>>  Content-Length: 0
>>>
>>>   
>>> ------------------------------------------------------------------------
>>> 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3359 Channel
>>> sofia/external/*408774957558438 at sip.callwithus.com entering state
>>> [received][100]
>>> 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3366 Remote SDP:
>>> v=0
>>> o=root 31797 31799 IN IP4 32.50.63.192
>>> s=session
>>> c=IN IP4 32.50.63.192
>>> t=0 0
>>> m=image 14104 udptl t38
>>> a=T38FaxVersion:0
>>> a=T38MaxBitRate:9600
>>> a=T38FaxFillBitRemoval:0
>>> a=T38FaxTranscodingMMR:0
>>> a=T38FaxTranscodingJBIG:0
>>> a=T38FaxRateManagement:transferredTCF
>>> a=T38FaxMaxBuffer:176
>>> a=T38FaxMaxDatagram:176
>>> a=T38FaxUdpEC:t38UDPRedundancy
>>>
>>> 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3683 Passing SDP to  
>>> other leg.
>>> v=0
>>> o=root 31797 31799 IN IP4 32.50.63.192
>>> s=session
>>> c=IN IP4 32.50.63.192
>>> t=0 0
>>> m=image 14104 udptl t38
>>> a=T38FaxVersion:0
>>> a=T38MaxBitRate:9600
>>> a=T38FaxFillBitRemoval:0
>>> a=T38FaxTranscodingMMR:0
>>> a=T38FaxTranscodingJBIG:0
>>> a=T38FaxRateManagement:transferredTCF
>>> a=T38FaxMaxBuffer:176
>>> a=T38FaxMaxDatagram:176
>>> a=T38FaxUdpEC:t38UDPRedundancy
>>>
>>> 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3696
>>> sofia/internal/214 at pbx0.tssec.lan receive message [MEDIA_REDIRECT]
>>> 2009-10-06 19:03:47.049284 [DEBUG] mod_sofia.c:1195
>>> sofia/internal/214 at pbx0.tssec.lan Sending media re-direct:
>>> v=0
>>> o=root 31797 31799 IN IP4 32.50.63.192
>>> s=session
>>> c=IN IP4 32.50.63.192
>>> t=0 0
>>> m=image 14104 udptl t38
>>> a=T38FaxVersion:0
>>> a=T38MaxBitRate:9600
>>> a=T38FaxFillBitRemoval:0
>>> a=T38FaxTranscodingMMR:0
>>> a=T38FaxTranscodingJBIG:0
>>> a=T38FaxRateManagement:transferredTCF
>>> a=T38FaxMaxBuffer:176
>>> a=T38FaxMaxDatagram:176
>>> a=T38FaxUdpEC:t38UDPRedundancy
>>>
>>> 2009-10-06 19:03:47.049284 [DEBUG] sofia_glue.c:1101 Remote  
>>> address:port
>>> [192.168.50.51:16474] has not changed.
>>> 2009-10-06 19:03:47.049284 [DEBUG] sofia_glue.c:1322
>>> sofia/internal/214 at pbx0.tssec.lan Patched SDP
>>> ---
>>> v=0
>>> o=root 31797 31799 IN IP4 32.50.63.192
>>> s=session
>>> c=IN IP4 32.50.63.192
>>> t=0 0
>>> m=image 14104 udptl t38
>>> a=T38FaxVersion:0
>>> a=T38MaxBitRate:9600
>>> a=T38FaxFillBitRemoval:0
>>> a=T38FaxTranscodingMMR:0
>>> a=T38FaxTranscodingJBIG:0
>>> a=T38FaxRateManagement:transferredTCF
>>> a=T38FaxMaxBuffer:176
>>> a=T38FaxMaxDatagram:176
>>> a=T38FaxUdpEC:t38UDPRedundancy
>>>
>>> +++
>>> v=0
>>> o=root 31797 31799 IN IP4 32.50.63.192
>>> s=session
>>> c=IN IP4 192.168.50.10
>>> t=0 0
>>> m=image 32698 udptl t38
>>> a=T38FaxVersion:0
>>> a=T38MaxBitRate:9600
>>> a=T38FaxFillBitRemoval:0
>>> a=T38FaxTranscodingMMR:0
>>> a=T38FaxTranscodingJBIG:0
>>> a=T38FaxRateManagement:transferredTCF
>>> a=T38FaxMaxBuffer:176
>>> a=T38FaxMaxDatagram:176
>>> a=T38FaxUdpEC:t38UDPRedundancy
>>>
>>> 2009-10-06 19:03:47.049284 [DEBUG] switch_core_session.c:630 Send  
>>> signal
>>> sofia/internal/214 at pbx0.tssec.lan [BREAK]
>>> 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3359 Channel
>>> sofia/internal/214 at pbx0.tssec.lan entering state [calling][0]
>>> 2009-10-06 19:03:47.077291 [DEBUG] sofia_glue.c:1101 Remote  
>>> address:port
>>> [192.168.50.51:16474] has not changed.
>>> 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3158 Passing 200 OK to  
>>> other leg
>>> 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3170
>>> sofia/external/*408774957558438 at sip.callwithus.com receive message  
>>> [RESPOND]
>>> 2009-10-06 19:03:47.077291 [DEBUG] mod_sofia.c:1427 Responding  
>>> with 200 [OK]
>>> 2009-10-06 19:03:47.077291 [DEBUG] sofia_glue.c:1322
>>> sofia/external/*408774957558438 at sip.callwithus.com Patched SDP
>>> ---
>>> v=0
>>> o=- 216899808 216899808 IN IP4 192.168.50.51
>>> s=-
>>> c=IN IP4 192.168.50.51
>>> t=0 0
>>> m=image 16474 udptl t38
>>> a=T38FaxVersion:0
>>> a=T38MaxBitRate:14400
>>> a=T38FaxRateManagement:transferredTCF
>>> a=T38FaxMaxBuffer:200
>>> a=T38FaxMaxDatagram:200
>>> a=T38FaxUdpEC:t38UDPRedundancy
>>>
>>> +++
>>> v=0
>>> o=- 216899808 216899808 IN IP4 192.168.50.51
>>> s=-
>>> c=IN IP4 77.239.230.202
>>> t=0 0
>>> m=image 32404 udptl t38
>>> a=T38FaxVersion:0
>>> a=T38MaxBitRate:14400
>>> a=T38FaxRateManagement:transferredTCF
>>> a=T38FaxMaxBuffer:200
>>> a=T38FaxMaxDatagram:200
>>> a=T38FaxUdpEC:t38UDPRedundancy
>>>
>>> 2009-10-06 19:03:47.077291 [DEBUG] sofia_glue.c:1110 AUDIO RTP  
>>> CHANGING
>>> DEST TO: [32.50.63.192:14104]
>>> 2009-10-06 19:03:47.077291 [DEBUG] switch_core_session.c:630 Send  
>>> signal
>>> sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
>>> 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3359 Channel
>>> sofia/internal/214 at pbx0.tssec.lan entering state [ready][200]
>>> 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3366 Remote SDP:
>>> v=0
>>> o=- 216899808 216899808 IN IP4 192.168.50.51
>>> s=-
>>> c=IN IP4 192.168.50.51
>>> t=0 0
>>> m=image 16474 udptl t38
>>> a=T38FaxVersion:0
>>> a=T38MaxBitRate:14400
>>> a=T38FaxRateManagement:transferredTCF
>>> a=T38FaxMaxBuffer:200
>>> a=T38FaxMaxDatagram:200
>>> a=T38FaxUdpEC:t38UDPRedundancy
>>>
>>> send 1003 bytes to udp/[38.99.70.232]:5060 at 15:06:05.605615:
>>>   
>>> ------------------------------------------------------------------------
>>>  SIP/2.0 200 OK
>>>  Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.0
>>>  Via: SIP/2.0/UDP  
>>> 204.74.213.1:5062;branch=z9hG4bK41d48157;rport=5062
>>>  From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>>  To: "Mihailova Ludmila"
>>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>>  Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>>  CSeq: 102 INVITE
>>>  Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
>>>  User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
>>>  Accept: application/sdp
>>>  Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>> REGISTER, REFER, NOTIFY
>>>  Supported: timer, precondition, path, replaces
>>>  Content-Type: application/sdp
>>>  Content-Disposition: session
>>>  Content-Length: 274
>>>
>>>  v=0
>>>  o=- 216896551 216896552 IN IP4 192.168.50.51
>>>  s=-
>>>  c=IN IP4 77.239.230.202
>>>  t=0 0
>>>  m=image 32404 udptl t38
>>>  a=T38FaxVersion:0
>>>  a=T38MaxBitRate:14400
>>>  a=T38FaxRateManagement:transferredTCF
>>>  a=T38FaxMaxBuffer:200
>>>  a=T38FaxMaxDatagram:200
>>>  a=T38FaxUdpEC:t38UDPRedundancy
>>>   
>>> ------------------------------------------------------------------------
>>> 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3359 Channel
>>> sofia/external/*408774957558438 at sip.callwithus.com entering state
>>> [completed][200]
>>> 2009-10-06 19:03:47.285347 [INFO] switch_rtp.c:1905 Auto Changing  
>>> port
>>> from 32.50.63.192:14104 to 4.55.17.66:14104
>>> recv 527 bytes from udp/[38.99.70.232]:5060 at 15:06:05.817672:
>>>   
>>> ------------------------------------------------------------------------
>>>  ACK sip:gw+callwithus at 192.168.50.10:5080;transport=udp SIP/2.0
>>>  Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.2
>>>  Via: SIP/2.0/UDP  
>>> 204.74.213.1:5062;branch=z9hG4bK5ffd72cd;rport=5062
>>>  From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>>  To: "Mihailova Ludmila"
>>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>>  Contact: <sip:*408774957558438 at 204.74.213.1:5062>
>>>  Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>>  CSeq: 102 ACK
>>>  User-Agent: CWU SIP-GW
>>>  Max-Forwards: 69
>>>  Content-Length: 0
>>>
>>>   
>>> ------------------------------------------------------------------------
>>> 2009-10-06 19:03:47.289349 [DEBUG] sofia.c:3359 Channel
>>> sofia/external/*408774957558438 at sip.callwithus.com entering state
>>> [ready][200]
>>> recv 476 bytes from udp/[38.99.70.232]:5060 at 15:06:47.588933:
>>>   
>>> ------------------------------------------------------------------------
>>>  BYE sip:gw+callwithus at 192.168.50.10:5080;transport=udp SIP/2.0
>>>  Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKd406.f8553e14.0
>>>  Via: SIP/2.0/UDP  
>>> 204.74.213.1:5062;branch=z9hG4bK0f2cff01;rport=5062
>>>  From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>>  To: "Mihailova Ludmila"
>>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>>  Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>>  CSeq: 103 BYE
>>>  User-Agent: CWU SIP-GW
>>>  Max-Forwards: 69
>>>  Content-Length: 0
>>>
>>>   
>>> ------------------------------------------------------------------------
>>> 2009-10-06 19:04:29.060609 [NOTICE] sofia.c:328 Hangup
>>> sofia/external/*408774957558438 at sip.callwithus.com  
>>> [CS_EXCHANGE_MEDIA]
>>> [NORMAL_CLEARING]
>>> 2009-10-06 19:04:29.060609 [DEBUG] switch_channel.c:1726 Send signal
>>> sofia/external/*408774957558438 at sip.callwithus.com [KILL]
>>> 2009-10-06 19:04:29.060609 [DEBUG] switch_core_session.c:932 Send  
>>> signal
>>> sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
>>> 2009-10-06 19:04:29.060609 [DEBUG] switch_core_state_machine.c:437
>>> thread mismatch skipping state handler.
>>> send 574 bytes to udp/[38.99.70.232]:5060 at 15:06:47.588933:
>>>   
>>> ------------------------------------------------------------------------
>>>  SIP/2.0 200 OK
>>>  Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKd406.f8553e14.0
>>>  Via: SIP/2.0/UDP  
>>> 204.74.213.1:5062;branch=z9hG4bK0f2cff01;rport=5062
>>>  From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>>  To: "Mihailova Ludmila"
>>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>>  Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>>  CSeq: 103 BYE
>>>  User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
>>>  Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>> REGISTER, REFER, NOTIFY
>>>  Supported: timer, precondition, path, replaces
>>>  Content-Length: 0
>>>
>>>   
>>> ------------------------------------------------------------------------
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>
>>
>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From Mailings at kh-dev.de  Wed Oct  7 18:46:10 2009
From: Mailings at kh-dev.de (Klaus Hochlehnert)
Date: Thu, 8 Oct 2009 03:46:10 +0200
Subject: [Freeswitch-users] Changing callerid before/after call pickup
In-Reply-To: <5B417827-6CCB-4EE4-9524-5CD45AC9E28F@gmail.com>
References: <4ACB9519.9000805@gmail.com>
	<2d9149cd0910071235k27c1429v2bc555c0e7397fc4@mail.gmail.com>
	<4ACD076E.2080107@gmail.com>
	<5B417827-6CCB-4EE4-9524-5CD45AC9E28F@gmail.com>
Message-ID: <A2F1C0E212BBBA4E8C0E7BE568361F04027F0233024A@srv02.khdev.corp>

Hi,

currently I'm playing around with call pickup and Snom phones.
I'm using the intercept function for that.

My "problem" is now that after the call pickup (which works fine) I don't see the caller id of the original call.
Instead I see the pickup code, e.g. *820

I've tried to change nearly every channel variable (before doing the call pickup), but the phone won't display the caller id.

Does anybody know how to change that?

Thanks, Klaus



From jason at jasonjgw.net  Wed Oct  7 19:05:57 2009
From: jason at jasonjgw.net (Jason White)
Date: Thu, 8 Oct 2009 13:05:57 +1100
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
Message-ID: <20091008020557.GA23391@jdc.jasonjgw.net>

Brian West <brian at freeswitch.org> wrote:
>  From what I have been told h323plus is a based/fork of OpenH323 which  
> OPAL is just a continuation of OpenH323.  

>From a quick search of gmane.org, the situation seems a little more
complicated.

http://article.gmane.org/gmane.comp.telephony.openh323.general/10930/match=h+460+opal



From srinivas.ksvreddy at gmail.com  Wed Oct  7 22:20:41 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Thu, 8 Oct 2009 10:50:41 +0530
Subject: [Freeswitch-users] sofia_reg_handle_register
Message-ID: <f8af5740910072220j384799fcob42ffcfa3dca1f1a@mail.gmail.com>

Hi,

sofia_reg_handle_register, once it got executed, actuallyl where it will
maintain the registred users adn groups data,

Thanks
Srinivasula Reddy K
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From woodydickson at gmail.com  Wed Oct  7 22:45:08 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Thu, 8 Oct 2009 13:45:08 +0800
Subject: [Freeswitch-users] unable to configure Digium TDM400P
In-Reply-To: <87f2f3b90910070956l520b2c51p7f33463128bf68b2@mail.gmail.com>
References: <a44688a0910070340p4209e7b5yec9218b8c4fc692c@mail.gmail.com>
	<87f2f3b90910070956l520b2c51p7f33463128bf68b2@mail.gmail.com>
Message-ID: <a44688a0910072245v7f496385of411552f0098ba90@mail.gmail.com>

Hi Michael

Is the ztcfg output supposed to say something like 2 channels configured?

I have not set up openzap yet because I don't know if the openzap.config is
good or not.

Can you give me any suggestion if the ztcfg output I am getting is proper or
not?

thx,
woody


-MC
>
> On Wed, Oct 7, 2009 at 3:40 AM, Woody Dickson <woodydickson at gmail.com>wrote:
>
>> Hi,
>>
>> I am trying to setup a Digium TDM400P following the instruction on the
>> wiki.
>> It is a 1 fxo and 1 fxs card, so I tried
>> loadzone=in
>> defaultzone=in
>> fxsks=2
>> fxoks=1
>>
>> and
>>
>> loadzone=in
>> defaultzone=in
>> fxsks=1
>> fxoks=2
>>
>> None works.  Does anyone know how it should be configured?
>>
>> Here is what I get by following the wiki.
>>
>> [root at localhost zaptel]# ztcfg -vv
>>
>> Zaptel Version: SVN-branch-1.4-r4629M
>> Echo Canceller: MG2
>> Configuration
>> ======================
>>
>>
>> Channel map:
>>
>> Channel 01: FXS Kewlstart (Default) (Slaves: 01)
>> Channel 02: FXO Kewlstart (Default) (Slaves: 02)
>>
>> 2 channels to configure.
>>
>> Changing signalling on channel 1 from FXO Kewlstart to FXS Kewlstart
>> Changing signalling on channel 2 from FXS Kewlstart to FXO Kewlstart
>>
>>
>> [root at localhost zaptel]# ztcfg -vv
>>
>> Zaptel Version: SVN-branch-1.4-r4629M
>> Echo Canceller: MG2
>> Configuration
>> ======================
>>
>>
>> Channel map:
>>
>> Channel 01: FXO Kewlstart (Default) (Slaves: 01)
>> Channel 02: FXS Kewlstart (Default) (Slaves: 02)
>>
>> 2 channels to configure.
>>
>> Changing signalling on channel 1 from FXS Kewlstart to FXO Kewlstart
>> Changing signalling on channel 2 from FXO Kewlstart to FXS Kewlstart
>>
>>
>> [root at localhost zaptel]# lspci
>> 00:14.0 Ethernet controller: Digium, Inc. TDM400P (rev 11)
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lakindia89 at gmail.com  Thu Oct  8 02:41:08 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Thu, 8 Oct 2009 15:11:08 +0530
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <87f2f3b90910070849t79b7cb90x4150bf1dd8501ca4@mail.gmail.com>
References: <7d79b3930910070334g53cef0c5m63037e264f1a9233@mail.gmail.com>
	<87f2f3b90910070849t79b7cb90x4150bf1dd8501ca4@mail.gmail.com>
Message-ID: <7d79b3930910080241q4eeb374an2ec34642bf450818@mail.gmail.com>

Hi I tried with the following openzap.conf
[span zt PRI_1]
trunk_type => e1
b-channel => 1:1-15,17-31
d-channel => 1:16

openzap.conf.xml
<configuration name="openzap.conf" description="OpenZAP Configuration">
        <settings>
                <param name="debug" value="7"/>
        </settings>
        <libpri_spans>
                <span name="PRI_1">
                        <param name="node" value="cpe"/>
                        <param name="switch" value="euroisdn"/>
                        <param name="dialplan" value="XML"/>
                        <param name="context" value="default"/>
                </span>
        </libpri_spans>
</configuration>

But as soon as I start the freeswitch I got the following error

2009-10-08 15:08:35.496270 [DEBUG] ozmod_libpri.c:885 Event Failure! 1
2009-10-08 15:08:37.601135 [DEBUG] ozmod_libpri.c:817 EVENT [ONHOOK][1][1:1]
STATE [DOWN]
2009-10-08 15:08:37.701176 [DEBUG] ozmod_libpri.c:885 Event Failure! 1
2009-10-08 15:08:39.802004 [DEBUG] ozmod_libpri.c:817 EVENT [DTMF][0][1:1]
STATE [DOWN]
2009-10-08 15:08:39.902034 [DEBUG] ozmod_libpri.c:885 Event Failure! 1
2009-10-08 15:08:42.010983 [DEBUG] ozmod_libpri.c:885 Event Failure! 1
2009-10-08 15:08:44.115874 [DEBUG] ozmod_libpri.c:817 EVENT [ONHOOK][1][1:1]
STATE [DOWN]
2009-10-08 15:08:44.215915 [DEBUG] ozmod_libpri.c:885 Event Failure! 1
2009-10-08 15:08:46.312748 [DEBUG] ozmod_libpri.c:885 Event Failure! 1
2009-10-08 15:08:48.417552 [DEBUG] ozmod_libpri.c:817 EVENT [DTMF][0][1:1]
STATE [DOWN]
2009-10-08 15:08:48.517587 [DEBUG] ozmod_libpri.c:885 Event Failure! 1
2009-10-08 15:08:50.626485 [DEBUG] ozmod_libpri.c:885 Event Failure! 1
2009-10-08 15:08:52.735325 [DEBUG] ozmod_libpri.c:817 EVENT [ONHOOK][1][1:1]
STATE [DOWN]
2009-10-08 15:08:52.835357 [DEBUG] ozmod_libpri.c:885 Event Failure! 1
2009-10-08 15:08:54.944190 [DEBUG] ozmod_libpri.c:885 Event Failure! 1
2009-10-08 15:08:57.049109 [DEBUG] ozmod_libpri.c:817 EVENT [DTMF][0][1:1]
STATE [DOWN]
2009-10-08 15:08:57.149148 [DEBUG] ozmod_libpri.c:885 Event Failure! 1
2009-10-08 15:08:59.253979 [DEBUG] ozmod_libpri.c:885 Event Failure! 1

I also got the below error in the middle

2009-10-08 15:09:48.992613 [CRIT] lpwrap_pri.c:154 span 1 D-WRITE FAIL! []
2009-10-08 15:09:48.992613 [ERR] ozmod_libpri.c:88 Short write: -1/5
(Invalid or incomplete multibyte or wide character)

Please help.

On Wed, Oct 7, 2009 at 9:19 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Oct 7, 2009 at 3:34 AM, lakshmanan ganapathy <lakindia89 at gmail.com
> > wrote:
>
>> Hi,
>> Again I was struck in a problem, Here is the scenario.
>>
>> On incomming call, I just call an event outboud socket. But what happens
>> is, for the first 15 call, it is working fine. But from the 16th call to
>> 30th call, it says the below error.
>>
>> 2009-10-07 15:07:48.201846 [WARNING] ozmod_libpri.c:761 --Failure opening
>> channel 1:17 (ignored)
>> 2009-10-07 15:07:55.381861 [WARNING] ozmod_libpri.c:761 --Failure opening
>> channel 1:18 (ignored)
>> 2009-10-07 15:07:58.569774 [WARNING] ozmod_libpri.c:761 --Failure opening
>> channel 1:19 (ignored)
>> 2009-10-07 15:08:01.37824 [WARNING] ozmod_libpri.c:761 --Failure opening
>> channel 1:20 (ignored)
>> 2009-10-07 15:08:03.129846 [WARNING] ozmod_libpri.c:761 --Failure opening
>> channel 1:21 (ignored)
>> 2009-10-07 15:08:04.825851 [WARNING] ozmod_libpri.c:761 --Failure opening
>> channel 1:22 (ignored)
>> 2009-10-07 15:08:06.289977 [WARNING] ozmod_libpri.c:761 --Failure opening
>> channel 1:23 (ignored)
>> 2009-10-07 15:08:07.761961 [WARNING] ozmod_libpri.c:761 --Failure opening
>> channel 1:24 (ignored)
>> 2009-10-07 15:08:09.737944 [WARNING] ozmod_libpri.c:761 --Failure opening
>> channel 1:25 (ignored)
>> 2009-10-07 15:08:11.462018 [WARNING] ozmod_libpri.c:761 --Failure opening
>> channel 1:26 (ignored)
>> 2009-10-07 15:08:13.566024 [WARNING] ozmod_libpri.c:761 --Failure opening
>> channel 1:27 (ignored)
>> 2009-10-07 15:08:15.430163 [WARNING] ozmod_libpri.c:761 --Failure opening
>> channel 1:28 (ignored)
>> 2009-10-07 15:08:17.446103 [WARNING] ozmod_libpri.c:761 --Failure opening
>> channel 1:29 (ignored)
>> 2009-10-07 15:08:19.430118 [WARNING] ozmod_libpri.c:761 --Failure opening
>> channel 1:30 (ignored)
>> 2009-10-07 15:08:21.358121 [WARNING] ozmod_libpri.c:761 --Failure opening
>> channel 1:31 (ignored)
>>
>>
>> But for the next call, when it is opening channel 1:1, it is executing my
>> dial plans.
>> I don't know why it failed when it is choosing 1:17-1:31. Any one has any
>> idea.
>>
>> Below are my configuration details:
>>
>> openzap.conf
>> [span zt PRI_1]
>> trunk_type => e1
>> b-channel => 1:1-15
>> d-channel=> 1:16
>> b-channel => 1:17-31
>>
>
> Just a hunch, but could you try this in openzap.conf:
> b-channel => 1:1-15,17-31
> d-channel => 1:16
>
> I want to see what happens. I don't have an E1 setup to test with right now
> otherwise I'd do it myself. Please report back. Thanks,
> MC
>
>
>>
>> openzap.conf.xml
>> <configuration name="openzap.conf" description="OpenZAP Configuration">
>>         <settings>
>>                 <param name="debug" value="7"/>
>>         </settings>
>>         <libpri_spans>
>>                 <span name="PRI_1">
>>                         <param name="node" value="cpe"/>
>>                         <param name="switch" value="euroisdn"/>
>>                         <param name="dialplan" value="XML"/>
>>                         <param name="context" value="default"/>
>>                 </span>
>>         </libpri_spans>
>> </configuration>
>>
>> zaptel.conf
>>
>> span=1,1,0,ccs,hdb3
>> bchan=1-15,17-31
>> dchan=16
>>
>> loadzone        = us
>> defaultzone     = us
>>
>> oz list
>>
>> span: 1 (PRI_1)
>> type: isdn
>> chan_count: 47
>> dialplan: XML
>> context: default
>> dial_regex:
>> fail_dial_regex:
>> hold_music:
>> analog_options none
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bottleman at icf.org.ru  Thu Oct  8 02:42:51 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Thu, 8 Oct 2009 13:42:51 +0400 (MSD)
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
Message-ID: <Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>

On 2009-10-07 15:09 -0500, Brian West wrote freeswitch-users at lists.freeswit...:

opal have addition abstraction layer called opalmgr, and it implementation is not so good in 
this case, for example to implemet pre_answer in mod_opal i need patch libopal, because 
there is no way to send progress inicator throuch opalmgr. and there is many another issues like
this in that layer. ftp://srv.icf.org.ru/pub/soft/f/freeswitch/ - there is my work on mod_opal before 
i start moving to h323plus, may be this help somebody there.

BW> From what I have been told h323plus is a based/fork of OpenH323 which  
BW>OPAL is just a continuation of OpenH323.  So why not support the  
BW>developers of OPAL/OpenH323 ?
BW>
BW>/b
BW>
BW>On Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote:
BW>
BW>> We are developing module to handle h323 proto now, we try to use  
BW>> mod_opal and try improve it, but no luck,
BW>> there is many issues in libopal, and finaly we now move to h323plus  
BW>> library.
BW>
BW>
BW>_______________________________________________
BW>FreeSWITCH-users mailing list
BW>FreeSWITCH-users at lists.freeswitch.org
BW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
BW>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
BW>http://www.freeswitch.org
BW>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From vhatz at kinetix.gr  Thu Oct  8 02:47:47 2009
From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI))
Date: Thu, 08 Oct 2009 12:47:47 +0300
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
Message-ID: <4ACDB543.3070906@kinetix.gr>

Brian West wrote:
>  From what I have been told h323plus is a based/fork of OpenH323 which  
> OPAL is just a continuation of OpenH323.  So why not support the  
> developers of OPAL/OpenH323 ?

H323Plus incorporates many if not all new features/additions etc from 
OPAL (for H323 that is) and a few more features of its own. It is not 
just a continuation of the antiquated OpenH323, so, IMHO it is a good 
option for H323 development.

mod_opal is mostly developed for use with H323 (SIP and IAX support is 
already there in FS in different modules), so it doesn't make much 
difference if it uses H323Plus, OPAL or any other H323 lib.

It will make a difference however if the underlying library leads to 
faster mod_opal/mod_h323 development.

So, yes it would be great to support OPAL and its developers, but 
sometimes the end result (and how fast you obtain it) is what matters 
most.

Therefore, if there are people out there who have already built a usable 
and stable mod for H323 (compared to the current state of mod_opal) I 
think we should give it a try, even if it uses a different library.

At least I'm more than willing to test it in lab and production and help 
in bug reports, or even patching if I can.

Regards,
Vlasis.



From juanbackson at gmail.com  Thu Oct  8 03:20:38 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Thu, 8 Oct 2009 18:20:38 +0800
Subject: [Freeswitch-users] directory variables not set correct with the
	latest trunk
Message-ID: <27c25bc40910080320x7b3b380eu933797197e334cb8@mail.gmail.com>

Hi,

My application fails to set the appropriate variables using directory xml
after using the latest trunk as of yesterday.

My curl looks like:



<document type="freeswitch/xml">
  <section name="directory">
    <domain name="192.168.1.102">
      <user id="200002">
        <params>
          <param name="password" value="123456"></param>
          <param name="dial-string"
value="sofia/internal/sip:200002 at 192.168.1.11:29440
;rinstance=0245b2a59ddff837"></param>
        </params>
        <variables>
          <variable name="account_id" value="20"></variable>
          <variable name="vm-code" value="1000"></variable>
        </variables>
      </user>
    </domain>
  </section>
</document>



In the info app, I am not seeing account-id and vm-code anymore.

How to fix that?

juan
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From srinivas.ksvreddy at gmail.com  Thu Oct  8 04:29:03 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Thu, 8 Oct 2009 16:59:03 +0530
Subject: [Freeswitch-users] sofia_Reg_find_url
Message-ID: <f8af5740910080429y42a9bf05wb626d806e826fc8a@mail.gmail.com>

Hi,

can any one please tell me how sofia_reg_fine_url, and
sofia_glue_execute_sql_callback will work?
which format data is storing and retriving.

Thanks
-- 
Srinivasula Reddy K
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From maciej.aniserowicz at gmail.com  Thu Oct  8 06:18:04 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Thu, 8 Oct 2009 06:18:04 -0700 (PDT)
Subject: [Freeswitch-users] gateway FS informs it's client FS about
 users hanged up with a long delay
In-Reply-To: <98F72CCB-88B9-47ED-AED3-B6BA6DE648C0@jerris.com>
References: <1A16501E57CA4727A593226A8C810308@procent>
	<98F72CCB-88B9-47ED-AED3-B6BA6DE648C0@jerris.com>
Message-ID: <1255007884165-3787956.post@n2.nabble.com>


Both of the instances are run on the same machine, i just changed the default
ports they use. Can anything else cause this strange behavior?
MA



Michael Jerris wrote:
> 
> Incorrect NAT configuration so one of the boxes is not actually  
> getting a BYE.
> 
> 
> On Oct 5, 2009, at 3:13 AM, Maciej Aniserowicz wrote:
> 
>> Hi,
>> When I use two FreeSWITCH instances ('internal' and 'external'), all  
>> users register to the 'external' instance which acts as a gateway by  
>> 'internal' instance (which in turn is controlled by my applicaiton  
>> with commands sent by socket).
>> When user hangs up, the 'hanged up' event is propagated to the  
>> 'internal' instance after a long time (~3 minutes) instead of being  
>> propagated immediately.
>> What can cause this issue?
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://n2.nabble.com/gateway-FS-informs-it-s-client-FS-about-users-hanged-up-with-a-long-delay-tp3768540p3787956.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From d.kochmashev at enforta.com  Wed Oct  7 20:50:13 2009
From: d.kochmashev at enforta.com (Denis Kochmashev "Enforta")
Date: Thu, 8 Oct 2009 09:50:13 +0600
Subject: [Freeswitch-users] [Spam] Re: mod_opal - call charged before
	H.225 connect
In-Reply-To: <20091008020557.GA23391@jdc.jasonjgw.net>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com><65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com><191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com><65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com><86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com><65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com><Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru><82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<20091008020557.GA23391@jdc.jasonjgw.net>
Message-ID: <C723CF32A20B4B61939DB5CD38DB6A06@kochmashev>

It's better to read this -
http://lists.packetizer.com/pipermail/h323plus/2007-October/000020.html 

> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On 
> Behalf Of Jason White
> Sent: Thursday, October 08, 2009 8:06 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Spam] Re: [Freeswitch-users] mod_opal - call 
> charged before H.225 connect
> 
> Brian West <brian at freeswitch.org> wrote:
> >  From what I have been told h323plus is a based/fork of 
> OpenH323 which  
> > OPAL is just a continuation of OpenH323.  
> 
> >From a quick search of gmane.org, the situation seems a little more
> complicated.
> 
> http://article.gmane.org/gmane.comp.telephony.openh323.general
> /10930/match=h+460+opal
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
> itch-users
> http://www.freeswitch.org



From maciej.aniserowicz at gmail.com  Thu Oct  8 06:19:31 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Thu, 8 Oct 2009 06:19:31 -0700 (PDT)
Subject: [Freeswitch-users] Recording creates a 388-byte long file and
 deletes it
In-Reply-To: <74D80E7B-119B-4CC1-82DC-9A370F940BA5@jerris.com>
References: <4ED3AB65AFE34242AACDE97127FE1248@procent>
	<74D80E7B-119B-4CC1-82DC-9A370F940BA5@jerris.com>
Message-ID: <1255007971021-3787968.post@n2.nabble.com>


Yes, I know that FS deletes short files. I just don't know why the file is so
small... it is always 388 bytes, no matter how long the session lasts.
MA



Michael Jerris wrote:
> 
> switch_ivr_async.c:480
> 
> On Oct 5, 2009, at 3:16 AM, Maciej Aniserowicz wrote:
> 
>> Hi,
>> When I record a call in FS, it only creates a 388-byte-long wav  
>> file. The conversation is no written there, and FS deletes the file  
>> when the session finishes.
>> What can cause this strange behavior?
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://n2.nabble.com/Recording-creates-a-388-byte-long-file-and-deletes-it-tp3768541p3787968.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From maciej.aniserowicz at gmail.com  Thu Oct  8 06:27:07 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Thu, 8 Oct 2009 06:27:07 -0700 (PDT)
Subject: [Freeswitch-users] Bad sound quality while eavesdropping
In-Reply-To: <8437F5BC-7AFF-4A74-B8CD-C5B8219021F6@jerris.com>
References: <41A44DD027064988A914974405788C2E@procent>
	<191c3a030910050731m2d74979ep4598e5a1945d58ae@mail.gmail.com>
	<1254901192035-3780245.post@n2.nabble.com>
	<8437F5BC-7AFF-4A74-B8CD-C5B8219021F6@jerris.com>
Message-ID: <1255008427639-3788019.post@n2.nabble.com>


It's the same on the trunk (the last rev I used was not so old anyway).

Codecs are the same on both legs:
read codec/read rate: PCMU	8000
write codec/write rate: PCMU	8000	

MA




Michael Jerris wrote:
> 
> What codecs are all the call legs using, also, please try current svn  
> trunk.
> 
> Mike
> 
> On Oct 7, 2009, at 3:39 AM, Maciej Aniserowicz wrote:
> 
>>
>> Sorry about posting several questions at once, I wasn't aware it's  
>> "rude".
>> Let's concentrate on this issue then.
>>
>> I use FS rev 14994. Phones on extensions:
>> 1) x-lite
>> 2) cisco sip phone
>> 3) audio played by fs to the extension being eavesdropped
>>
>> I did not change any codec configuration, I just use the standard  
>> one that
>> comes with both FS and the phones.
>> Some time ago someone on FS irc channel told me that this is just  
>> how FS
>> eavesdropping works... from your response I understand that this is  
>> not
>> entirely true?
>>
>> Maciej Aniserowicz
>>
>>
>>
>> Anthony Minessale wrote:
>>>
>>> That's is a somewhat vague position.
>>>
>>> You did not mention which version of FreeSWITCH you are running, the
>>> phones
>>> being used in your example, your configuration, the codecs in use  
>>> etc.
>>>
>>> BTW,
>>> I think you should only ask one question at a time on this list.   
>>> The list
>>> is run by volunteers and it's sort of rude to expect 3 or 4 threads  
>>> to be
>>> tended to concerning the same one individual.
>>>
>>>
>>> 2009/10/5 Maciej Aniserowicz <maciej.aniserowicz at gmail.com>
>>>
>>>> Hello,
>>>> When I use eavesdropping in FreeSWITCH, the sound quality is  
>>>> really bad.
>>>> Is
>>>> there any way to improve it? Is this a known problem?
>>>> Br/
>>>> Maciej Aniserowicz
>>>>
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3788019.html
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From tculjaga at gmail.com  Thu Oct  8 07:20:44 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 8 Oct 2009 16:20:44 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
Message-ID: <65d96fc80910080720x5a9e9756tab7c08ec6b67f163@mail.gmail.com>

Hi Yuriy, did you manage to do something with H323plus and FS ?

btw: have you checked Objective OpenH323
http://www.obj-sys.com/telephony-objective.shtml ?
This looks better to me as it is lighter and can be easily customized.

T.


2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-07 15:09 -0500, Brian West wrote
> freeswitch-users at lists.freeswit...:
>
> opal have addition abstraction layer called opalmgr, and it implementation
> is not so good in
> this case, for example to implemet pre_answer in mod_opal i need patch
> libopal, because
> there is no way to send progress inicator throuch opalmgr. and there is
> many another issues like
> this in that layer. ftp://srv.icf.org.ru/pub/soft/f/freeswitch/ - there is
> my work on mod_opal before
> i start moving to h323plus, may be this help somebody there.
>
> BW> From what I have been told h323plus is a based/fork of OpenH323 which
> BW>OPAL is just a continuation of OpenH323.  So why not support the
> BW>developers of OPAL/OpenH323 ?
> BW>
> BW>/b
> BW>
> BW>On Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote:
> BW>
> BW>> We are developing module to handle h323 proto now, we try to use
> BW>> mod_opal and try improve it, but no luck,
> BW>> there is many issues in libopal, and finaly we now move to h323plus
> BW>> library.
> BW>
> BW>
> BW>_______________________________________________
> BW>FreeSWITCH-users mailing list
> BW>FreeSWITCH-users at lists.freeswitch.org
> BW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> BW>UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> BW>http://www.freeswitch.org
> BW>
>
> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129-RIPE
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rupa at rupa.com  Thu Oct  8 07:40:09 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 8 Oct 2009 08:40:09 -0600
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910080720x5a9e9756tab7c08ec6b67f163@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<65d96fc80910080720x5a9e9756tab7c08ec6b67f163@mail.gmail.com>
Message-ID: <b39273530910080740u7671454eg36cab6bee8339296@mail.gmail.com>

That code is GPL, not license compatible. ;(

2009/10/8 Tihomir Culjaga <tculjaga at gmail.com>:
> Hi Yuriy, did you manage to do something with H323plus and FS ?
>
> btw: have you checked Objective OpenH323
> http://www.obj-sys.com/telephony-objective.shtml ?
> This looks better to me as it is lighter and can be easily customized.
>
> T.


-- 
-Rupa


From dome at tel.co.th  Thu Oct  8 07:39:46 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Thu, 8 Oct 2009 21:39:46 +0700
Subject: [Freeswitch-users] max-sessions change when running about 2 hr
Message-ID: <8ccbff060910080739h14143bf8lb43060a5ec6393a0@mail.gmail.com>

Dear sir,
   I'm running freeswitch with option
/opt/freeswitch/bin/freeswitch -nf -waste -nonat -nc
I'm setting up max-sessions in switch.conf to 10000

    <param name="max-sessions" value="10000"/>
    <!--Most channels to create per second -->
    <param name="sessions-per-second" value="50"/>
But when fs running about 2 hr. max session down to 299
freeswitch at internal> status
UP 0 years, 0 days, 2 hours, 11 minutes, 10 seconds, 434 milliseconds,
251 microseconds
20502 session(s) since startup
271 session(s) 0/50
299 session(s) max

Can somone tell me what's happen ?

Dome C.


From gabe at gundy.org  Thu Oct  8 07:42:22 2009
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 8 Oct 2009 08:42:22 -0600
Subject: [Freeswitch-users] FS Slide deck?
Message-ID: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>

Hey all,

The 3rd annual UTOSC (Utah Open Source Conference) [1] starts today.
It's a conference that gives Open Source users/hackers a chance gather
together to mingle and share.  It's a lot of fun.  Anyway, this year
I've offered to work with a friend of mine to lead a BoF (Birds of a
Feather) discussion on Open Source Telephony [2].

While it should mostly be group discussion, you never know when people
are going to get shy and clam up.  *If* they do, it would be nice to
have some slides to fill that time.  Rather than make some from
scratch (that are unlikely to get used), I thought I'd ask here and
see if anyone has a FS slide deck that they wouldn't mind sharing.

Thanks,
Gabe


1) http://2009.utosc.com/pages/home/
2) http://2009.utosc.com/presentation/136/


From vhatz at kinetix.gr  Thu Oct  8 07:44:55 2009
From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI))
Date: Thu, 08 Oct 2009 17:44:55 +0300
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910080720x5a9e9756tab7c08ec6b67f163@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<65d96fc80910080720x5a9e9756tab7c08ec6b67f163@mail.gmail.com>
Message-ID: <4ACDFAE7.1000409@kinetix.gr>


Tihomir Culjaga wrote:
> Hi Yuriy, did you manage to do something with H323plus and FS ?
> 
> btw: have you checked Objective OpenH323 
> http://www.obj-sys.com/telephony-objective.shtml ?
> This looks better to me as it is lighter and can be easily customized.
> 
> T.

AFAIK Objective Systems H323 library does not support fax or video, it 
does only voice (at least the last time I checked it which was many 
months ago). Using it would lead to a mod for H323 which would never be 
as flexible or feature-rich as mod_sofia...

Regards,
Vlasis.


From bottleman at icf.org.ru  Thu Oct  8 07:44:09 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Thu, 8 Oct 2009 18:44:09 +0400 (MSD)
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910080720x5a9e9756tab7c08ec6b67f163@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<65d96fc80910080720x5a9e9756tab7c08ec6b67f163@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910081830380.11841@srv.icf.org.ru>

On 2009-10-08 16:20 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>Hi Yuriy, did you manage to do something with H323plus and FS ?

i already doing it, but now it not in usable state.

TC>btw: have you checked Objective OpenH323
TC>http://www.obj-sys.com/telephony-objective.shtml ?
TC>This looks better to me as it is lighter and can be easily customized.

i see this library later in asterisk module, h323plus is a successor of opanh323, i use it many yars and 
i think it more complete mature and stable than objective systems stack, and finally h323plus not depend 
in its development from some kinde of Objective System Inc/any other xxx Inc.

TC>
TC>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
TC>
TC>> On 2009-10-07 15:09 -0500, Brian West wrote
TC>> freeswitch-users at lists.freeswit...:
TC>>
TC>> opal have addition abstraction layer called opalmgr, and it implementation
TC>> is not so good in
TC>> this case, for example to implemet pre_answer in mod_opal i need patch
TC>> libopal, because
TC>> there is no way to send progress inicator throuch opalmgr. and there is
TC>> many another issues like
TC>> this in that layer. ftp://srv.icf.org.ru/pub/soft/f/freeswitch/ - there is
TC>> my work on mod_opal before
TC>> i start moving to h323plus, may be this help somebody there.
TC>>
TC>> BW> From what I have been told h323plus is a based/fork of OpenH323 which
TC>> BW>OPAL is just a continuation of OpenH323.  So why not support the
TC>> BW>developers of OPAL/OpenH323 ?
TC>> BW>
TC>> BW>/b
TC>> BW>
TC>> BW>On Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote:
TC>> BW>
TC>> BW>> We are developing module to handle h323 proto now, we try to use
TC>> BW>> mod_opal and try improve it, but no luck,
TC>> BW>> there is many issues in libopal, and finaly we now move to h323plus
TC>> BW>> library.
TC>> BW>
TC>> BW>
TC>> BW>_______________________________________________
TC>> BW>FreeSWITCH-users mailing list
TC>> BW>FreeSWITCH-users at lists.freeswitch.org
TC>> BW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> BW>UNSUBSCRIBE:
TC>> http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> BW>http://www.freeswitch.org
TC>> BW>
TC>>
TC>> C ?????????                       With Best Regards
TC>> ???????????? ????.                Georgiewskiy Yuriy
TC>> +7 4872 711666                    +7 4872 711666
TC>> ???? +7 4872 711143               fax +7 4872 711143
TC>> ???????? ??? "?? ?? ??????"       IT Service Ltd
TC>> http://nkoort.ru                  http://nkoort.ru
TC>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
TC>> YG129-RIPE                        YG129-RIPE
TC>>
TC>> _______________________________________________
TC>> FreeSWITCH-users mailing list
TC>> FreeSWITCH-users at lists.freeswitch.org
TC>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> http://www.freeswitch.org
TC>>
TC>>
TC>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From dome at tel.co.th  Thu Oct  8 07:46:44 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Thu, 8 Oct 2009 21:46:44 +0700
Subject: [Freeswitch-users] max-sessions change when running about 2 hr
In-Reply-To: <8ccbff060910080739h14143bf8lb43060a5ec6393a0@mail.gmail.com>
References: <8ccbff060910080739h14143bf8lb43060a5ec6393a0@mail.gmail.com>
Message-ID: <8ccbff060910080746p30333c13t2a11a70d0b961e3@mail.gmail.com>

When i try
fsctl max_sessions 10000
in CLI . FS set it back to 2xx again

Dome C.
2009/10/8 Dome Charoenyost <dome at tel.co.th>:
> Dear sir,
> ? I'm running freeswitch with option
> /opt/freeswitch/bin/freeswitch -nf -waste -nonat -nc
> I'm setting up max-sessions in switch.conf to 10000
>
> ? ?<param name="max-sessions" value="10000"/>
> ? ?<!--Most channels to create per second -->
> ? ?<param name="sessions-per-second" value="50"/>
> But when fs running about 2 hr. max session down to 299
> freeswitch at internal> status
> UP 0 years, 0 days, 2 hours, 11 minutes, 10 seconds, 434 milliseconds,
> 251 microseconds
> 20502 session(s) since startup
> 271 session(s) 0/50
> 299 session(s) max
>
> Can somone tell me what's happen ?
>
> Dome C.
>


From brian at freeswitch.org  Thu Oct  8 07:59:24 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 8 Oct 2009 09:59:24 -0500
Subject: [Freeswitch-users] max-sessions change when running about 2 hr
In-Reply-To: <8ccbff060910080746p30333c13t2a11a70d0b961e3@mail.gmail.com>
References: <8ccbff060910080739h14143bf8lb43060a5ec6393a0@mail.gmail.com>
	<8ccbff060910080746p30333c13t2a11a70d0b961e3@mail.gmail.com>
Message-ID: <75EBC642-1912-4CFA-8774-40DF1FA0152E@freeswitch.org>

How much ram do you have and what distro are you running on, 32bit or  
64bit?

/b

On Oct 8, 2009, at 9:46 AM, Dome Charoenyost wrote:

> When i try
> fsctl max_sessions 10000
> in CLI . FS set it back to 2xx again
>
> Dome C.



From tculjaga at gmail.com  Thu Oct  8 08:00:30 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 8 Oct 2009 17:00:30 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910081830380.11841@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<65d96fc80910080720x5a9e9756tab7c08ec6b67f163@mail.gmail.com>
	<Pine.LNX.4.64.0910081830380.11841@srv.icf.org.ru>
Message-ID: <65d96fc80910080800t5427abc5m2012ff2a5c3326c0@mail.gmail.com>

yep, you made the point :P

T.


2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-08 16:20 +0200, Tihomir Culjaga wrote
> freeswitch-users at lists.fre...:
>
> TC>Hi Yuriy, did you manage to do something with H323plus and FS ?
>
> i already doing it, but now it not in usable state.
>
> TC>btw: have you checked Objective OpenH323
> TC>http://www.obj-sys.com/telephony-objective.shtml ?
> TC>This looks better to me as it is lighter and can be easily customized.
>
> i see this library later in asterisk module, h323plus is a successor of
> opanh323, i use it many yars and
> i think it more complete mature and stable than objective systems stack,
> and finally h323plus not depend
> in its development from some kinde of Objective System Inc/any other xxx
> Inc.
>
> TC>
> TC>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
> TC>
> TC>> On 2009-10-07 15:09 -0500, Brian West wrote
> TC>> freeswitch-users at lists.freeswit...:
> TC>>
> TC>> opal have addition abstraction layer called opalmgr, and it
> implementation
> TC>> is not so good in
> TC>> this case, for example to implemet pre_answer in mod_opal i need patch
> TC>> libopal, because
> TC>> there is no way to send progress inicator throuch opalmgr. and there
> is
> TC>> many another issues like
> TC>> this in that layer. ftp://srv.icf.org.ru/pub/soft/f/freeswitch/ -
> there is
> TC>> my work on mod_opal before
> TC>> i start moving to h323plus, may be this help somebody there.
> TC>>
> TC>> BW> From what I have been told h323plus is a based/fork of OpenH323
> which
> TC>> BW>OPAL is just a continuation of OpenH323.  So why not support the
> TC>> BW>developers of OPAL/OpenH323 ?
> TC>> BW>
> TC>> BW>/b
> TC>> BW>
> TC>> BW>On Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote:
> TC>> BW>
> TC>> BW>> We are developing module to handle h323 proto now, we try to use
> TC>> BW>> mod_opal and try improve it, but no luck,
> TC>> BW>> there is many issues in libopal, and finaly we now move to
> h323plus
> TC>> BW>> library.
> TC>> BW>
> TC>> BW>
> TC>> BW>_______________________________________________
> TC>> BW>FreeSWITCH-users mailing list
> TC>> BW>FreeSWITCH-users at lists.freeswitch.org
> TC>> BW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> TC>> BW>UNSUBSCRIBE:
> TC>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> TC>> BW>http://www.freeswitch.org
> TC>> BW>
> TC>>
> TC>> C ?????????                       With Best Regards
> TC>> ???????????? ????.                Georgiewskiy Yuriy
> TC>> +7 4872 711666                    +7 4872 711666
> TC>> ???? +7 4872 711143               fax +7 4872 711143
> TC>> ???????? ??? "?? ?? ??????"       IT Service Ltd
> TC>> http://nkoort.ru                  http://nkoort.ru
> TC>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> TC>> YG129-RIPE                        YG129-RIPE
> TC>>
> TC>> _______________________________________________
> TC>> FreeSWITCH-users mailing list
> TC>> FreeSWITCH-users at lists.freeswitch.org
> TC>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> TC>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> TC>> http://www.freeswitch.org
> TC>>
> TC>>
> TC>
>
> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129-RIPE
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From moises.silva at gmail.com  Thu Oct  8 08:02:26 2009
From: moises.silva at gmail.com (Moises Silva)
Date: Thu, 8 Oct 2009 11:02:26 -0400
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <7d79b3930910080241q4eeb374an2ec34642bf450818@mail.gmail.com>
References: <7d79b3930910070334g53cef0c5m63037e264f1a9233@mail.gmail.com>
	<87f2f3b90910070849t79b7cb90x4150bf1dd8501ca4@mail.gmail.com>
	<7d79b3930910080241q4eeb374an2ec34642bf450818@mail.gmail.com>
Message-ID: <c4d05cbe0910080802t7ebf8007m333cbd6e74efd286@mail.gmail.com>

On Thu, Oct 8, 2009 at 5:41 AM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Hi I tried with the following openzap.conf
> [span zt PRI_1]
> trunk_type => e1
> b-channel => 1:1-15,17-31
> d-channel => 1:16
>

This does not look like a healthy config to me. You are using
<span>:<channels> notation in zaptel spans, zaptel channels increment across
spans, read point number 5 of this web page
http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2 , this is for another type of
signaling, but the span/channel numbering concepts are the same.

-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From dome at tel.co.th  Thu Oct  8 08:14:43 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Thu, 8 Oct 2009 22:14:43 +0700
Subject: [Freeswitch-users] max-sessions change when running about 2 hr
In-Reply-To: <75EBC642-1912-4CFA-8774-40DF1FA0152E@freeswitch.org>
References: <8ccbff060910080739h14143bf8lb43060a5ec6393a0@mail.gmail.com>
	<8ccbff060910080746p30333c13t2a11a70d0b961e3@mail.gmail.com>
	<75EBC642-1912-4CFA-8774-40DF1FA0152E@freeswitch.org>
Message-ID: <8ccbff060910080814v56dbd129nccb1b99353a367cf@mail.gmail.com>

2009/10/8 Brian West <brian at freeswitch.org>:
> How much ram do you have and what distro are you running on, 32bit or
> 64bit?

2GB  32bit  debian squeeze
(swap 6 GB)

Dome C.


>
> /b
>
> On Oct 8, 2009, at 9:46 AM, Dome Charoenyost wrote:
>
>> When i try
>> fsctl max_sessions 10000
>> in CLI . FS set it back to 2xx again
>>
>> Dome C.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From dome at tel.co.th  Thu Oct  8 08:15:42 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Thu, 8 Oct 2009 22:15:42 +0700
Subject: [Freeswitch-users] max-sessions change when running about 2 hr
In-Reply-To: <8ccbff060910080814v56dbd129nccb1b99353a367cf@mail.gmail.com>
References: <8ccbff060910080739h14143bf8lb43060a5ec6393a0@mail.gmail.com>
	<8ccbff060910080746p30333c13t2a11a70d0b961e3@mail.gmail.com>
	<75EBC642-1912-4CFA-8774-40DF1FA0152E@freeswitch.org>
	<8ccbff060910080814v56dbd129nccb1b99353a367cf@mail.gmail.com>
Message-ID: <8ccbff060910080815h7f79f68v992270146a747cfe@mail.gmail.com>

may be -waste option effect  ?

2009/10/8 Dome Charoenyost <dome at tel.co.th>:
> 2009/10/8 Brian West <brian at freeswitch.org>:
>> How much ram do you have and what distro are you running on, 32bit or
>> 64bit?
>
> 2GB ?32bit ?debian squeeze
> (swap 6 GB)
>
> Dome C.
>
>
>>
>> /b
>>
>> On Oct 8, 2009, at 9:46 AM, Dome Charoenyost wrote:
>>
>>> When i try
>>> fsctl max_sessions 10000
>>> in CLI . FS set it back to 2xx again
>>>
>>> Dome C.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>


From anthony.minessale at gmail.com  Thu Oct  8 08:21:14 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 8 Oct 2009 10:21:14 -0500
Subject: [Freeswitch-users] directory variables not set correct with the
	latest trunk
In-Reply-To: <27c25bc40910080320x7b3b380eu933797197e334cb8@mail.gmail.com>
References: <27c25bc40910080320x7b3b380eu933797197e334cb8@mail.gmail.com>
Message-ID: <191c3a030910080821w691aa78aw272d71dee7675ff1@mail.gmail.com>

they only set if you authenticate on sip or run the set_user app


On Thu, Oct 8, 2009 at 5:20 AM, Juan Backson <juanbackson at gmail.com> wrote:

> Hi,
>
> My application fails to set the appropriate variables using directory xml
> after using the latest trunk as of yesterday.
>
> My curl looks like:
>
>
>
> <document type="freeswitch/xml">
>   <section name="directory">
>     <domain name="192.168.1.102">
>       <user id="200002">
>         <params>
>           <param name="password" value="123456"></param>
>           <param name="dial-string"
> value="sofia/internal/sip:200002 at 192.168.1.11:29440
> ;rinstance=0245b2a59ddff837"></param>
>         </params>
>         <variables>
>           <variable name="account_id" value="20"></variable>
>           <variable name="vm-code" value="1000"></variable>
>         </variables>
>       </user>
>     </domain>
>   </section>
> </document>
>
>
>
> In the info app, I am not seeing account-id and vm-code anymore.
>
> How to fix that?
>
> juan
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

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From anthony.minessale at gmail.com  Thu Oct  8 08:23:13 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 8 Oct 2009 10:23:13 -0500
Subject: [Freeswitch-users] Bad sound quality while eavesdropping
In-Reply-To: <1255008427639-3788019.post@n2.nabble.com>
References: <41A44DD027064988A914974405788C2E@procent>
	<191c3a030910050731m2d74979ep4598e5a1945d58ae@mail.gmail.com>
	<1254901192035-3780245.post@n2.nabble.com>
	<8437F5BC-7AFF-4A74-B8CD-C5B8219021F6@jerris.com>
	<1255008427639-3788019.post@n2.nabble.com>
Message-ID: <191c3a030910080823g79c7c596x1cd887e1538ce2e1@mail.gmail.com>

you probably have some device lying about ptime everywhere
look at a sip trace an pay especially close attention to ptime:x param in
sdp
if you don't understand this just attach it here

execute the following at the cli
sofia profile internal siptrace on
sofila loglevel debug



On Thu, Oct 8, 2009 at 8:27 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> It's the same on the trunk (the last rev I used was not so old anyway).
>
> Codecs are the same on both legs:
> read codec/read rate: PCMU      8000
> write codec/write rate: PCMU    8000
>
> MA
>
>
>
>
> Michael Jerris wrote:
> >
> > What codecs are all the call legs using, also, please try current svn
> > trunk.
> >
> > Mike
> >
> > On Oct 7, 2009, at 3:39 AM, Maciej Aniserowicz wrote:
> >
> >>
> >> Sorry about posting several questions at once, I wasn't aware it's
> >> "rude".
> >> Let's concentrate on this issue then.
> >>
> >> I use FS rev 14994. Phones on extensions:
> >> 1) x-lite
> >> 2) cisco sip phone
> >> 3) audio played by fs to the extension being eavesdropped
> >>
> >> I did not change any codec configuration, I just use the standard
> >> one that
> >> comes with both FS and the phones.
> >> Some time ago someone on FS irc channel told me that this is just
> >> how FS
> >> eavesdropping works... from your response I understand that this is
> >> not
> >> entirely true?
> >>
> >> Maciej Aniserowicz
> >>
> >>
> >>
> >> Anthony Minessale wrote:
> >>>
> >>> That's is a somewhat vague position.
> >>>
> >>> You did not mention which version of FreeSWITCH you are running, the
> >>> phones
> >>> being used in your example, your configuration, the codecs in use
> >>> etc.
> >>>
> >>> BTW,
> >>> I think you should only ask one question at a time on this list.
> >>> The list
> >>> is run by volunteers and it's sort of rude to expect 3 or 4 threads
> >>> to be
> >>> tended to concerning the same one individual.
> >>>
> >>>
> >>> 2009/10/5 Maciej Aniserowicz <maciej.aniserowicz at gmail.com>
> >>>
> >>>> Hello,
> >>>> When I use eavesdropping in FreeSWITCH, the sound quality is
> >>>> really bad.
> >>>> Is
> >>>> there any way to improve it? Is this a known problem?
> >>>> Br/
> >>>> Maciej Aniserowicz
> >>>>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> --
> View this message in context:
> http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3788019.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Thu Oct  8 08:29:21 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 8 Oct 2009 10:29:21 -0500
Subject: [Freeswitch-users] max-sessions change when running about 2 hr
In-Reply-To: <8ccbff060910080815h7f79f68v992270146a747cfe@mail.gmail.com>
References: <8ccbff060910080739h14143bf8lb43060a5ec6393a0@mail.gmail.com>
	<8ccbff060910080746p30333c13t2a11a70d0b961e3@mail.gmail.com>
	<75EBC642-1912-4CFA-8774-40DF1FA0152E@freeswitch.org>
	<8ccbff060910080814v56dbd129nccb1b99353a367cf@mail.gmail.com>
	<8ccbff060910080815h7f79f68v992270146a747cfe@mail.gmail.com>
Message-ID: <191c3a030910080829s7886f664m73f6e504562bb38d@mail.gmail.com>

you are running out of threads and it's fixing itself. one of many reasons
you should upgrade to 64 bit OS

execute this in your shell before you start FS

ulimit -c unlimited
ulimit -d unlimited
ulimit -f unlimited
ulimit -i unlimited
ulimit -n 999999
ulimit -q unlimited
ulimit -u unlimited
ulimit -v unlimited
ulimit -x unlimited
ulimit -s 244
ulimit -l unlimited



On Thu, Oct 8, 2009 at 10:15 AM, Dome Charoenyost <dome at tel.co.th> wrote:

> may be -waste option effect  ?
>
> 2009/10/8 Dome Charoenyost <dome at tel.co.th>:
> > 2009/10/8 Brian West <brian at freeswitch.org>:
> >> How much ram do you have and what distro are you running on, 32bit or
> >> 64bit?
> >
> > 2GB  32bit  debian squeeze
> > (swap 6 GB)
> >
> > Dome C.
> >
> >
> >>
> >> /b
> >>
> >> On Oct 8, 2009, at 9:46 AM, Dome Charoenyost wrote:
> >>
> >>> When i try
> >>> fsctl max_sessions 10000
> >>> in CLI . FS set it back to 2xx again
> >>>
> >>> Dome C.
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From anthony.minessale at gmail.com  Thu Oct  8 08:43:14 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 8 Oct 2009 10:43:14 -0500
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
Message-ID: <191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>

If you are going to make that alternate module are you going to host it in
the FS tree along side mod_opal?
also if were working on mod_opal why did you not try to involve us and the
opal team?
How far away from what is in tree are these patches you have?

2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-07 15:09 -0500, Brian West wrote
> freeswitch-users at lists.freeswit...:
>
> opal have addition abstraction layer called opalmgr, and it implementation
> is not so good in
> this case, for example to implemet pre_answer in mod_opal i need patch
> libopal, because
> there is no way to send progress inicator throuch opalmgr. and there is
> many another issues like
> this in that layer. ftp://srv.icf.org.ru/pub/soft/f/freeswitch/ - there is
> my work on mod_opal before
> i start moving to h323plus, may be this help somebody there.
>
> BW> From what I have been told h323plus is a based/fork of OpenH323 which
> BW>OPAL is just a continuation of OpenH323.  So why not support the
> BW>developers of OPAL/OpenH323 ?
> BW>
> BW>/b
> BW>
> BW>On Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote:
> BW>
> BW>> We are developing module to handle h323 proto now, we try to use
> BW>> mod_opal and try improve it, but no luck,
> BW>> there is many issues in libopal, and finaly we now move to h323plus
> BW>> library.
> BW>
> BW>
> BW>_______________________________________________
> BW>FreeSWITCH-users mailing list
> BW>FreeSWITCH-users at lists.freeswitch.org
> BW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> BW>UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> BW>http://www.freeswitch.org
> BW>
>
> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129-RIPE
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From dome at tel.co.th  Thu Oct  8 08:50:36 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Thu, 8 Oct 2009 22:50:36 +0700
Subject: [Freeswitch-users] max-sessions change when running about 2 hr
In-Reply-To: <191c3a030910080829s7886f664m73f6e504562bb38d@mail.gmail.com>
References: <8ccbff060910080739h14143bf8lb43060a5ec6393a0@mail.gmail.com>
	<8ccbff060910080746p30333c13t2a11a70d0b961e3@mail.gmail.com>
	<75EBC642-1912-4CFA-8774-40DF1FA0152E@freeswitch.org>
	<8ccbff060910080814v56dbd129nccb1b99353a367cf@mail.gmail.com>
	<8ccbff060910080815h7f79f68v992270146a747cfe@mail.gmail.com>
	<191c3a030910080829s7886f664m73f6e504562bb38d@mail.gmail.com>
Message-ID: <8ccbff060910080850k2a849842u450a303b92d27619@mail.gmail.com>

Thanks.  I'll try 64 bit :)

2009/10/8 Anthony Minessale <anthony.minessale at gmail.com>:
> you are running out of threads and it's fixing itself. one of many reasons
> you should upgrade to 64 bit OS
>
> execute this in your shell before you start FS
>
> ulimit -c unlimited
> ulimit -d unlimited
> ulimit -f unlimited
>
> ulimit -i unlimited
> ulimit -n 999999
> ulimit -q unlimited
> ulimit -u unlimited
> ulimit -v unlimited
> ulimit -x unlimited
> ulimit -s 244
> ulimit -l unlimited
>
>
> On Thu, Oct 8, 2009 at 10:15 AM, Dome Charoenyost <dome at tel.co.th> wrote:
>>
>> may be -waste option effect ??
>>
>> 2009/10/8 Dome Charoenyost <dome at tel.co.th>:
>> > 2009/10/8 Brian West <brian at freeswitch.org>:
>> >> How much ram do you have and what distro are you running on, 32bit or
>> >> 64bit?
>> >
>> > 2GB ?32bit ?debian squeeze
>> > (swap 6 GB)
>> >
>> > Dome C.
>> >
>> >
>> >>
>> >> /b
>> >>
>> >> On Oct 8, 2009, at 9:46 AM, Dome Charoenyost wrote:
>> >>
>> >>> When i try
>> >>> fsctl max_sessions 10000
>> >>> in CLI . FS set it back to 2xx again
>> >>>
>> >>> Dome C.
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From bottleman at icf.org.ru  Thu Oct  8 09:03:00 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Thu, 8 Oct 2009 20:03:00 +0400 (MSD)
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>

On 2009-10-08 10:43 -0500, Anthony Minessale wrote freeswitch-users at lists.f...:

AM>If you are going to make that alternate module are you going to host it in
AM>the FS tree along side mod_opal?

Yes, but then it be useful, now i have working only signaling part and some 
kinde of not working rtp part :)

AM>also if were working on mod_opal why did you not try to involve us and the
AM>opal team?

Because i made patches for libopal, one is a bugfix in rtp part, there is a race condition 
in inicialisation in jitter buffer, another patch implements method to send progress indicator, 
and i don't wont spent my time to incorporate this changes into libopal. without this changes
my work on mod_opal in freeswitch don't useful at all, i provide link to my work with all 
patches, if somebody wont incorporate it in libopal tree and fs - go on, but i think 
better and more elegant make new module based on h323plus.

AM>How far away from what is in tree are these patches you have?
AM>
AM>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
AM>
AM>> On 2009-10-07 15:09 -0500, Brian West wrote
AM>> freeswitch-users at lists.freeswit...:
AM>>
AM>> opal have addition abstraction layer called opalmgr, and it implementation
AM>> is not so good in
AM>> this case, for example to implemet pre_answer in mod_opal i need patch
AM>> libopal, because
AM>> there is no way to send progress inicator throuch opalmgr. and there is
AM>> many another issues like
AM>> this in that layer. ftp://srv.icf.org.ru/pub/soft/f/freeswitch/ - there is
AM>> my work on mod_opal before
AM>> i start moving to h323plus, may be this help somebody there.
AM>>
AM>> BW> From what I have been told h323plus is a based/fork of OpenH323 which
AM>> BW>OPAL is just a continuation of OpenH323.  So why not support the
AM>> BW>developers of OPAL/OpenH323 ?
AM>> BW>
AM>> BW>/b
AM>> BW>
AM>> BW>On Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote:
AM>> BW>
AM>> BW>> We are developing module to handle h323 proto now, we try to use
AM>> BW>> mod_opal and try improve it, but no luck,
AM>> BW>> there is many issues in libopal, and finaly we now move to h323plus
AM>> BW>> library.
AM>> BW>
AM>> BW>
AM>> BW>_______________________________________________
AM>> BW>FreeSWITCH-users mailing list
AM>> BW>FreeSWITCH-users at lists.freeswitch.org
AM>> BW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
AM>> BW>UNSUBSCRIBE:
AM>> http://lists.freeswitch.org/mailman/options/freeswitch-users
AM>> BW>http://www.freeswitch.org
AM>> BW>
AM>>
AM>> C ?????????                       With Best Regards
AM>> ???????????? ????.                Georgiewskiy Yuriy
AM>> +7 4872 711666                    +7 4872 711666
AM>> ???? +7 4872 711143               fax +7 4872 711143
AM>> ???????? ??? "?? ?? ??????"       IT Service Ltd
AM>> http://nkoort.ru                  http://nkoort.ru
AM>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
AM>> YG129-RIPE                        YG129-RIPE
AM>>
AM>> _______________________________________________
AM>> FreeSWITCH-users mailing list
AM>> FreeSWITCH-users at lists.freeswitch.org
AM>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
AM>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
AM>> http://www.freeswitch.org
AM>>
AM>>
AM>
AM>
AM>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From msc at freeswitch.org  Thu Oct  8 09:47:04 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Oct 2009 09:47:04 -0700
Subject: [Freeswitch-users] unable to configure Digium TDM400P
In-Reply-To: <a44688a0910072245v7f496385of411552f0098ba90@mail.gmail.com>
References: <a44688a0910070340p4209e7b5yec9218b8c4fc692c@mail.gmail.com>
	<87f2f3b90910070956l520b2c51p7f33463128bf68b2@mail.gmail.com>
	<a44688a0910072245v7f496385of411552f0098ba90@mail.gmail.com>
Message-ID: <87f2f3b90910080947h3cebf253s6e011dd8ddda8504@mail.gmail.com>

On Wed, Oct 7, 2009 at 10:45 PM, Woody Dickson <woodydickson at gmail.com>wrote:

>
> Hi Michael
>
> Is the ztcfg output supposed to say something like 2 channels configured?
>

Yes. If you had the signaling incorrect it would complain.


>
> I have not set up openzap yet because I don't know if the openzap.config is
> good or not.
>
>
You need to make sure that zaptel is set up first. Running ztcfg and getting
no errors or warnings, plus a message that says "2 channels configured" is a
pretty good sign that you can move on to the OpenZAP stuff.


> Can you give me any suggestion if the ztcfg output I am getting is proper
> or not?
>
> No obvious problems, so move on to the OpenZAP configs.
-MC

> thx,
> woody
>
>
>
> -MC
>>
>> On Wed, Oct 7, 2009 at 3:40 AM, Woody Dickson <woodydickson at gmail.com>wrote:
>>
>>>  Hi,
>>>
>>> I am trying to setup a Digium TDM400P following the instruction on the
>>> wiki.
>>> It is a 1 fxo and 1 fxs card, so I tried
>>> loadzone=in
>>> defaultzone=in
>>> fxsks=2
>>> fxoks=1
>>>
>>> and
>>>
>>> loadzone=in
>>> defaultzone=in
>>> fxsks=1
>>> fxoks=2
>>>
>>> None works.  Does anyone know how it should be configured?
>>>
>>> Here is what I get by following the wiki.
>>>
>>> [root at localhost zaptel]# ztcfg -vv
>>>
>>> Zaptel Version: SVN-branch-1.4-r4629M
>>> Echo Canceller: MG2
>>> Configuration
>>> ======================
>>>
>>>
>>> Channel map:
>>>
>>> Channel 01: FXS Kewlstart (Default) (Slaves: 01)
>>> Channel 02: FXO Kewlstart (Default) (Slaves: 02)
>>>
>>> 2 channels to configure.
>>>
>>> Changing signalling on channel 1 from FXO Kewlstart to FXS Kewlstart
>>> Changing signalling on channel 2 from FXS Kewlstart to FXO Kewlstart
>>>
>>>
>>> [root at localhost zaptel]# ztcfg -vv
>>>
>>> Zaptel Version: SVN-branch-1.4-r4629M
>>> Echo Canceller: MG2
>>> Configuration
>>> ======================
>>>
>>>
>>> Channel map:
>>>
>>> Channel 01: FXO Kewlstart (Default) (Slaves: 01)
>>> Channel 02: FXS Kewlstart (Default) (Slaves: 02)
>>>
>>> 2 channels to configure.
>>>
>>> Changing signalling on channel 1 from FXS Kewlstart to FXO Kewlstart
>>> Changing signalling on channel 2 from FXO Kewlstart to FXS Kewlstart
>>>
>>>
>>> [root at localhost zaptel]# lspci
>>> 00:14.0 Ethernet controller: Digium, Inc. TDM400P (rev 11)
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vhatz at kinetix.gr  Thu Oct  8 09:53:28 2009
From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI))
Date: Thu, 08 Oct 2009 19:53:28 +0300
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
Message-ID: <4ACE1908.9040004@kinetix.gr>

Georgiewskiy Yuriy wrote:
> On 2009-10-08 10:43 -0500, Anthony Minessale wrote freeswitch-users at lists.f...:
> 
> AM>If you are going to make that alternate module are you going to host it in
> AM>the FS tree along side mod_opal?
> 
> Yes, but then it be useful, now i have working only signaling part and some 
> kinde of not working rtp part :)


Hello Yuriy,

Since FS already has its own RTP stack (for pass-through and 
transcoding) wouldn't it be best if your H323 implementation with OPAL 
was used for signaling only and rely on FS RTP for the media? Perhaps 
that would save you time and effort, too?


Best regards,
Vlasis Hatzistavrou.


From msc at freeswitch.org  Thu Oct  8 09:54:18 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Oct 2009 09:54:18 -0700
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <c4d05cbe0910080802t7ebf8007m333cbd6e74efd286@mail.gmail.com>
References: <7d79b3930910070334g53cef0c5m63037e264f1a9233@mail.gmail.com>
	<87f2f3b90910070849t79b7cb90x4150bf1dd8501ca4@mail.gmail.com>
	<7d79b3930910080241q4eeb374an2ec34642bf450818@mail.gmail.com>
	<c4d05cbe0910080802t7ebf8007m333cbd6e74efd286@mail.gmail.com>
Message-ID: <87f2f3b90910080954l1ecb867dn8a025f5f0f701eb4@mail.gmail.com>

On Thu, Oct 8, 2009 at 8:02 AM, Moises Silva <moises.silva at gmail.com> wrote:

> On Thu, Oct 8, 2009 at 5:41 AM, lakshmanan ganapathy <lakindia89 at gmail.com
> > wrote:
>
>> Hi I tried with the following openzap.conf
>> [span zt PRI_1]
>> trunk_type => e1
>> b-channel => 1:1-15,17-31
>> d-channel => 1:16
>>
>
> This does not look like a healthy config to me. You are using
> <span>:<channels> notation in zaptel spans, zaptel channels increment across
> spans, read point number 5 of this web page
> http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2 , this is for another type
> of signaling, but the span/channel numbering concepts are the same.
>
>
Moy,

As usual you are right on the money. I'm so used to doing Sangoma configs
that I forget about the zaptel syntax. I recommend this config:

[span zt PRI_1]
trunk_type => e1
b-channel => 1-15
d-channel => 16
b-channel => 17-31


Give that a try and report back if you have issues.
-MC

> --
> Moises Silva
> Software Developer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Oct  8 09:58:12 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Oct 2009 09:58:12 -0700
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
Message-ID: <87f2f3b90910080958i1f5cfccax621fd12100a05e3e@mail.gmail.com>

On Thu, Oct 8, 2009 at 7:42 AM, Gabriel Gunderson <gabe at gundy.org> wrote:

> Hey all,
>
> The 3rd annual UTOSC (Utah Open Source Conference) [1] starts today.
> It's a conference that gives Open Source users/hackers a chance gather
> together to mingle and share.  It's a lot of fun.  Anyway, this year
> I've offered to work with a friend of mine to lead a BoF (Birds of a
> Feather) discussion on Open Source Telephony [2].
>
> While it should mostly be group discussion, you never know when people
> are going to get shy and clam up.  *If* they do, it would be nice to
> have some slides to fill that time.  Rather than make some from
> scratch (that are unlikely to get used), I thought I'd ask here and
> see if anyone has a FS slide deck that they wouldn't mind sharing.
>
> Thanks,
> Gabe
>
>
> 1) http://2009.utosc.com/pages/home/
> 2) http://2009.utosc.com/presentation/136/
>

There are numerous slides and such from ClueCon. Go to
http://www.cluecon.com and you can download both the video files as well as
the actual presentations. The presentation files are in various formats such
as PPT(X), PDF, and Keynote.
The second presentation on day 1 is Anthony Minessale and he talks a lot
about expanding one's thinking when it comes to applying FreeSWITCH in
production. For example, some people will jump through crazy hoops to solve
a problem "because that's how they've always done it." Anyway, check out
that video and presentation file for some great ideas.
-MC


>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Thu Oct  8 10:00:12 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Oct 2009 10:00:12 -0700
Subject: [Freeswitch-users] gateway FS informs it's client FS about
	users hanged up with a long delay
In-Reply-To: <1255007884165-3787956.post@n2.nabble.com>
References: <1A16501E57CA4727A593226A8C810308@procent>
	<98F72CCB-88B9-47ED-AED3-B6BA6DE648C0@jerris.com>
	<1255007884165-3787956.post@n2.nabble.com>
Message-ID: <87f2f3b90910081000v5b505f4ao53104211b620ea77@mail.gmail.com>

On Thu, Oct 8, 2009 at 6:18 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> Both of the instances are run on the same machine, i just changed the
> default
> ports they use. Can anything else cause this strange behavior?
> MA
>
Did a packet capture yield any clues? That is, were you able to confirm that
each instance sent and received all the packets that you believe they should
have sent and received? The reason I ask is so that you don't end up chasing
a ghost because you made an assumption somewhere in your troubleshooting.

-MC


>
>
>
> Michael Jerris wrote:
> >
> > Incorrect NAT configuration so one of the boxes is not actually
> > getting a BYE.
> >
> >
> > On Oct 5, 2009, at 3:13 AM, Maciej Aniserowicz wrote:
> >
> >> Hi,
> >> When I use two FreeSWITCH instances ('internal' and 'external'), all
> >> users register to the 'external' instance which acts as a gateway by
> >> 'internal' instance (which in turn is controlled by my applicaiton
> >> with commands sent by socket).
> >> When user hangs up, the 'hanged up' event is propagated to the
> >> 'internal' instance after a long time (~3 minutes) instead of being
> >> propagated immediately.
> >> What can cause this issue?
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> --
> View this message in context:
> http://n2.nabble.com/gateway-FS-informs-it-s-client-FS-about-users-hanged-up-with-a-long-delay-tp3768540p3787956.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bottleman at icf.org.ru  Thu Oct  8 10:03:25 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Thu, 8 Oct 2009 21:03:25 +0400 (MSD)
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <4ACE1908.9040004@kinetix.gr>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<4ACE1908.9040004@kinetix.gr>
Message-ID: <Pine.LNX.4.64.0910082058320.31576@srv.icf.org.ru>

On 2009-10-08 19:53 +0300, Vlasis Hatzistavrou (KTI) wrote freeswitch-users...:

VHK>Georgiewskiy Yuriy wrote:
VHK>> On 2009-10-08 10:43 -0500, Anthony Minessale wrote freeswitch-users at lists.f...:
VHK>> 
VHK>> AM>If you are going to make that alternate module are you going to host it in
VHK>> AM>the FS tree along side mod_opal?
VHK>> 
VHK>> Yes, but then it be useful, now i have working only signaling part and some 
VHK>> kinde of not working rtp part :)
VHK>
VHK>
VHK>Hello Yuriy,
VHK>
VHK>Since FS already has its own RTP stack (for pass-through and 
VHK>transcoding) wouldn't it be best if your H323 implementation with OPAL 
VHK>was used for signaling only and rely on FS RTP for the media? Perhaps 
VHK>that would save you time and effort, too?

it already use fs rtp stack, and my implementation based on h323plus, not on opal, 
but fs rtp stack not so complete, there need write some code to send/receive packets 
because fs stac only encode/decode packets, not send/receive, or i don't known how to 
do it with it. i don't find any example for this.


C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From anthony.minessale at gmail.com  Thu Oct  8 10:12:47 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 8 Oct 2009 12:12:47 -0500
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910082058320.31576@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<4ACE1908.9040004@kinetix.gr>
	<Pine.LNX.4.64.0910082058320.31576@srv.icf.org.ru>
Message-ID: <191c3a030910081012o63634e3bg1e510df510199541@mail.gmail.com>

hmm, not following that very well.
FS RTP stack only sends and receives and has no understanding of
encode/decode......

mod_opal uses OPAL rtp stack.



2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-08 19:53 +0300, Vlasis Hatzistavrou (KTI) wrote
> freeswitch-users...:
>
> VHK>Georgiewskiy Yuriy wrote:
> VHK>> On 2009-10-08 10:43 -0500, Anthony Minessale wrote
> freeswitch-users at lists.f...:
> VHK>>
> VHK>> AM>If you are going to make that alternate module are you going to
> host it in
> VHK>> AM>the FS tree along side mod_opal?
> VHK>>
> VHK>> Yes, but then it be useful, now i have working only signaling part
> and some
> VHK>> kinde of not working rtp part :)
> VHK>
> VHK>
> VHK>Hello Yuriy,
> VHK>
> VHK>Since FS already has its own RTP stack (for pass-through and
> VHK>transcoding) wouldn't it be best if your H323 implementation with OPAL
> VHK>was used for signaling only and rely on FS RTP for the media? Perhaps
> VHK>that would save you time and effort, too?
>
> it already use fs rtp stack, and my implementation based on h323plus, not
> on opal,
> but fs rtp stack not so complete, there need write some code to
> send/receive packets
> because fs stac only encode/decode packets, not send/receive, or i don't
> known how to
> do it with it. i don't find any example for this.
>
>
> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129-RIPE
>
> _______________________________________________
> FreeSWITCH-users mailing list
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
>


-- 
Anthony Minessale II

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From bottleman at icf.org.ru  Thu Oct  8 10:22:43 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Thu, 8 Oct 2009 21:22:43 +0400 (MSD)
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <191c3a030910081012o63634e3bg1e510df510199541@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<4ACE1908.9040004@kinetix.gr>
	<Pine.LNX.4.64.0910082058320.31576@srv.icf.org.ru>
	<191c3a030910081012o63634e3bg1e510df510199541@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910082117120.757@srv.icf.org.ru>

On 2009-10-08 12:12 -0500, Anthony Minessale wrote freeswitch-users at lists.f...:

AM>hmm, not following that very well.
AM>FS RTP stack only sends and receives and has no understanding of
AM>encode/decode......

hmm, give me some example how to use fs stack, or link there i can read about this.

AM>
AM>mod_opal uses OPAL rtp stack.
AM>
AM>
AM>
AM>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
AM>
AM>> On 2009-10-08 19:53 +0300, Vlasis Hatzistavrou (KTI) wrote
AM>> freeswitch-users...:
AM>>
AM>> VHK>Georgiewskiy Yuriy wrote:
AM>> VHK>> On 2009-10-08 10:43 -0500, Anthony Minessale wrote
AM>> freeswitch-users at lists.f...:
AM>> VHK>>
AM>> VHK>> AM>If you are going to make that alternate module are you going to
AM>> host it in
AM>> VHK>> AM>the FS tree along side mod_opal?
AM>> VHK>>
AM>> VHK>> Yes, but then it be useful, now i have working only signaling part
AM>> and some
AM>> VHK>> kinde of not working rtp part :)
AM>> VHK>
AM>> VHK>
AM>> VHK>Hello Yuriy,
AM>> VHK>
AM>> VHK>Since FS already has its own RTP stack (for pass-through and
AM>> VHK>transcoding) wouldn't it be best if your H323 implementation with OPAL
AM>> VHK>was used for signaling only and rely on FS RTP for the media? Perhaps
AM>> VHK>that would save you time and effort, too?
AM>>
AM>> it already use fs rtp stack, and my implementation based on h323plus, not
AM>> on opal,
AM>> but fs rtp stack not so complete, there need write some code to
AM>> send/receive packets
AM>> because fs stac only encode/decode packets, not send/receive, or i don't
AM>> known how to
AM>> do it with it. i don't find any example for this.
AM>>
AM>>
AM>> C ?????????                       With Best Regards
AM>> ???????????? ????.                Georgiewskiy Yuriy
AM>> +7 4872 711666                    +7 4872 711666
AM>> ???? +7 4872 711143               fax +7 4872 711143
AM>> ???????? ??? "?? ?? ??????"       IT Service Ltd
AM>> http://nkoort.ru                  http://nkoort.ru
AM>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
AM>> YG129-RIPE                        YG129-RIPE
AM>>
AM>> _______________________________________________
AM>> FreeSWITCH-users mailing list
AM>> FreeSWITCH-users at lists.freeswitch.org
AM>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
AM>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
AM>> http://www.freeswitch.org
AM>>
AM>>
AM>
AM>
AM>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From tuyanozipek at gmail.com  Thu Oct  8 10:25:13 2009
From: tuyanozipek at gmail.com (=?ISO-8859-1?Q?Tuyan_=D6zipek?=)
Date: Thu, 8 Oct 2009 13:25:13 -0400
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
Message-ID: <881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>

Hi,

2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>:
> On 2009-10-08 10:43 -0500, Anthony Minessale wrote freeswitch-users at lists.f...:
>
> AM>If you are going to make that alternate module are you going to host it in
> AM>the FS tree along side mod_opal?
>
> Yes, but then it be useful, now i have working only signaling part and some
> kinde of not working rtp part :)

If you dont use fs rtp stack, its unlikely that it will be accepted
into the tree.

>
> AM>also if were working on mod_opal why did you not try to involve us and the
> AM>opal team?
>
> Because i made patches for libopal, one is a bugfix in rtp part, there is a race condition
> in inicialisation in jitter buffer, another patch implements method to send progress indicator,
> and i don't wont spent my time to incorporate this changes into libopal.

Thats bad.
Any bugfixes from fs, goes to upstream on any of the used libraries.
You should be doing the same.
And Opal developers, will either include or refuse your patches. If
they refuse it, they will give you the reason.

> without this changes
> my work on mod_opal in freeswitch don't useful at all, i provide link to my work with all
> patches, if somebody wont incorporate it in libopal tree and fs - go on, but i think
> better and more elegant make new module based on h323plus.

If you dont publish your changes, all those you are trying to achieve,
wont happen.

>
> AM>How far away from what is in tree are these patches you have?
> AM>
> AM>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
> AM>
> AM>> On 2009-10-07 15:09 -0500, Brian West wrote
> AM>> freeswitch-users at lists.freeswit...:
> AM>>
> AM>> opal have addition abstraction layer called opalmgr, and it implementation
> AM>> is not so good in
> AM>> this case, for example to implemet pre_answer in mod_opal i need patch
> AM>> libopal, because

The patch you have in there, adds a method to the OpalCall,  it does
not touch any parts of OpalManager
so, i dont understand why opalmanager would be the cause of your pain?

> AM>> there is no way to send progress inicator throuch opalmgr. and there is
> AM>> many another issues like
> AM>> this in that layer.

Please point me to the issues you have in opal, their bug reports , traces etc.
I dont think any of the opal people has psychic abilities to detect
-your- problems
and solve them.

>ftp://srv.icf.org.ru/pub/soft/f/freeswitch/ - there is
> AM>> my work on mod_opal before
> AM>> i start moving to h323plus, may be this help somebody there.
> AM>>
> AM>> BW> From what I have been told h323plus is a based/fork of OpenH323 which
> AM>> BW>OPAL is just a continuation of OpenH323. ?So why not support the
> AM>> BW>developers of OPAL/OpenH323 ?
> AM>> BW>
> AM>> BW>/b
> AM>> BW>
> AM>> BW>On Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote:
> AM>> BW>
> AM>> BW>> We are developing module to handle h323 proto now, we try to use
> AM>> BW>> mod_opal and try improve it, but no luck,
> AM>> BW>> there is many issues in libopal, and finaly we now move to h323plus
> AM>> BW>> library.

Did any of you try to report those issues?

Regards

/tyn

> AM>> BW>
> AM>> BW>
> AM>> BW>_______________________________________________
> AM>> BW>FreeSWITCH-users mailing list
> AM>> BW>FreeSWITCH-users at lists.freeswitch.org
> AM>> BW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> AM>> BW>UNSUBSCRIBE:
> AM>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> AM>> BW>http://www.freeswitch.org
> AM>> BW>
> AM>>
> AM>> C ????????? ? ? ? ? ? ? ? ? ? ? ? With Best Regards
> AM>> ???????????? ????. ? ? ? ? ? ? ? ?Georgiewskiy Yuriy
> AM>> +7 4872 711666 ? ? ? ? ? ? ? ? ? ?+7 4872 711666
> AM>> ???? +7 4872 711143 ? ? ? ? ? ? ? fax +7 4872 711143
> AM>> ???????? ??? "?? ?? ??????" ? ? ? IT Service Ltd
> AM>> http://nkoort.ru ? ? ? ? ? ? ? ? ?http://nkoort.ru
> AM>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> AM>> YG129-RIPE ? ? ? ? ? ? ? ? ? ? ? ?YG129-RIPE
> AM>>
> AM>> _______________________________________________
> AM>> FreeSWITCH-users mailing list
> AM>> FreeSWITCH-users at lists.freeswitch.org
> AM>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> AM>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> AM>> http://www.freeswitch.org
> AM>>
> AM>>
> AM>
> AM>
> AM>
>
> C ????????? ? ? ? ? ? ? ? ? ? ? ? With Best Regards
> ???????????? ????. ? ? ? ? ? ? ? ?Georgiewskiy Yuriy
> +7 4872 711666 ? ? ? ? ? ? ? ? ? ?+7 4872 711666
> ???? +7 4872 711143 ? ? ? ? ? ? ? fax +7 4872 711143
> ???????? ??? "?? ?? ??????" ? ? ? IT Service Ltd
> http://nkoort.ru ? ? ? ? ? ? ? ? ?http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE ? ? ? ? ? ? ? ? ? ? ? ?YG129-RIPE
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From bottleman at icf.org.ru  Thu Oct  8 10:41:09 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Thu, 8 Oct 2009 21:41:09 +0400 (MSD)
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>

On 2009-10-08 13:25 -0400, Tuyan ?zipek wrote freeswitch-users at lists.freesw...:

Tz>Hi,
Tz>
Tz>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>:
Tz>> On 2009-10-08 10:43 -0500, Anthony Minessale wrote freeswitch-users at lists.f...:
Tz>>
Tz>> AM>If you are going to make that alternate module are you going to host it in
Tz>> AM>the FS tree along side mod_opal?
Tz>>
Tz>> Yes, but then it be useful, now i have working only signaling part and some
Tz>> kinde of not working rtp part :)
Tz>
Tz>If you dont use fs rtp stack, its unlikely that it will be accepted
Tz>into the tree.
Tz>
Tz>>
Tz>> AM>also if were working on mod_opal why did you not try to involve us and the
Tz>> AM>opal team?
Tz>>
Tz>> Because i made patches for libopal, one is a bugfix in rtp part, there is a race condition
Tz>> in inicialisation in jitter buffer, another patch implements method to send progress indicator,
Tz>> and i don't wont spent my time to incorporate this changes into libopal.
Tz>
Tz>Thats bad.
Tz>Any bugfixes from fs, goes to upstream on any of the used libraries.
Tz>You should be doing the same.
Tz>And Opal developers, will either include or refuse your patches. If
Tz>they refuse it, they will give you the reason.

i make this fix only to freeze my current mod_opal work on working state, while it now work for me i work on 
my new implementation of h323 proto for fs, i think opal developers will fix this rtp bug himself becouse 
it crashes and make library unuseful.

Tz>
Tz>> without this changes
Tz>> my work on mod_opal in freeswitch don't useful at all, i provide link to my work with all
Tz>> patches, if somebody wont incorporate it in libopal tree and fs - go on, but i think
Tz>> better and more elegant make new module based on h323plus.
Tz>
Tz>If you dont publish your changes, all those you are trying to achieve,
Tz>wont happen.
Tz>
Tz>>
Tz>> AM>How far away from what is in tree are these patches you have?
Tz>> AM>
Tz>> AM>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
Tz>> AM>
Tz>> AM>> On 2009-10-07 15:09 -0500, Brian West wrote
Tz>> AM>> freeswitch-users at lists.freeswit...:
Tz>> AM>>
Tz>> AM>> opal have addition abstraction layer called opalmgr, and it implementation
Tz>> AM>> is not so good in
Tz>> AM>> this case, for example to implemet pre_answer in mod_opal i need patch
Tz>> AM>> libopal, because
Tz>
Tz>The patch you have in there, adds a method to the OpalCall,  it does
Tz>not touch any parts of OpalManager
Tz>so, i dont understand why opalmanager would be the cause of your pain?
Tz>
Tz>> AM>> there is no way to send progress inicator throuch opalmgr. and there is
Tz>> AM>> many another issues like
Tz>> AM>> this in that layer.
Tz>
Tz>Please point me to the issues you have in opal, their bug reports , traces etc.
Tz>I dont think any of the opal people has psychic abilities to detect
Tz>-your- problems
Tz>and solve them.
Tz>
Tz>>ftp://srv.icf.org.ru/pub/soft/f/freeswitch/ - there is
Tz>> AM>> my work on mod_opal before
Tz>> AM>> i start moving to h323plus, may be this help somebody there.
Tz>> AM>>
Tz>> AM>> BW> From what I have been told h323plus is a based/fork of OpenH323 which
Tz>> AM>> BW>OPAL is just a continuation of OpenH323. ?So why not support the
Tz>> AM>> BW>developers of OPAL/OpenH323 ?
Tz>> AM>> BW>
Tz>> AM>> BW>/b
Tz>> AM>> BW>
Tz>> AM>> BW>On Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote:
Tz>> AM>> BW>
Tz>> AM>> BW>> We are developing module to handle h323 proto now, we try to use
Tz>> AM>> BW>> mod_opal and try improve it, but no luck,
Tz>> AM>> BW>> there is many issues in libopal, and finaly we now move to h323plus
Tz>> AM>> BW>> library.
Tz>
Tz>Did any of you try to report those issues?
Tz>
Tz>Regards
Tz>
Tz>/tyn
Tz>
Tz>> AM>> BW>
Tz>> AM>> BW>
Tz>> AM>> BW>_______________________________________________
Tz>> AM>> BW>FreeSWITCH-users mailing list
Tz>> AM>> BW>FreeSWITCH-users at lists.freeswitch.org
Tz>> AM>> BW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Tz>> AM>> BW>UNSUBSCRIBE:
Tz>> AM>> http://lists.freeswitch.org/mailman/options/freeswitch-users
Tz>> AM>> BW>http://www.freeswitch.org
Tz>> AM>> BW>
Tz>> AM>>
Tz>> AM>> C ????????? ? ? ? ? ? ? ? ? ? ? ? With Best Regards
Tz>> AM>> ???????????? ????. ? ? ? ? ? ? ? ?Georgiewskiy Yuriy
Tz>> AM>> +7 4872 711666 ? ? ? ? ? ? ? ? ? ?+7 4872 711666
Tz>> AM>> ???? +7 4872 711143 ? ? ? ? ? ? ? fax +7 4872 711143
Tz>> AM>> ???????? ??? "?? ?? ??????" ? ? ? IT Service Ltd
Tz>> AM>> http://nkoort.ru ? ? ? ? ? ? ? ? ?http://nkoort.ru
Tz>> AM>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
Tz>> AM>> YG129-RIPE ? ? ? ? ? ? ? ? ? ? ? ?YG129-RIPE
Tz>> AM>>
Tz>> AM>> _______________________________________________
Tz>> AM>> FreeSWITCH-users mailing list
Tz>> AM>> FreeSWITCH-users at lists.freeswitch.org
Tz>> AM>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Tz>> AM>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Tz>> AM>> http://www.freeswitch.org
Tz>> AM>>
Tz>> AM>>
Tz>> AM>
Tz>> AM>
Tz>> AM>
Tz>>
Tz>> C ????????? ? ? ? ? ? ? ? ? ? ? ? With Best Regards
Tz>> ???????????? ????. ? ? ? ? ? ? ? ?Georgiewskiy Yuriy
Tz>> +7 4872 711666 ? ? ? ? ? ? ? ? ? ?+7 4872 711666
Tz>> ???? +7 4872 711143 ? ? ? ? ? ? ? fax +7 4872 711143
Tz>> ???????? ??? "?? ?? ??????" ? ? ? IT Service Ltd
Tz>> http://nkoort.ru ? ? ? ? ? ? ? ? ?http://nkoort.ru
Tz>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
Tz>> YG129-RIPE ? ? ? ? ? ? ? ? ? ? ? ?YG129-RIPE
Tz>> _______________________________________________
Tz>> FreeSWITCH-users mailing list
Tz>> FreeSWITCH-users at lists.freeswitch.org
Tz>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Tz>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Tz>> http://www.freeswitch.org
Tz>>
Tz>>
Tz>
Tz>_______________________________________________
Tz>FreeSWITCH-users mailing list
Tz>FreeSWITCH-users at lists.freeswitch.org
Tz>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Tz>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Tz>http://www.freeswitch.org
Tz>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From tculjaga at gmail.com  Thu Oct  8 11:32:40 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 8 Oct 2009 20:32:40 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
Message-ID: <65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>

Hi Yuriy,

can you share what you have so far, I'm sure we can help with RTP part...

T.

2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-08 13:25 -0400, Tuyan ?zipek wrote
> freeswitch-users at lists.freesw...:
>
> Tz>Hi,
> Tz>
> Tz>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>:
> Tz>> On 2009-10-08 10:43 -0500, Anthony Minessale wrote
> freeswitch-users at lists.f...:
> Tz>>
> Tz>> AM>If you are going to make that alternate module are you going to
> host it in
> Tz>> AM>the FS tree along side mod_opal?
> Tz>>
> Tz>> Yes, but then it be useful, now i have working only signaling part and
> some
> Tz>> kinde of not working rtp part :)
> Tz>
> Tz>If you dont use fs rtp stack, its unlikely that it will be accepted
> Tz>into the tree.
> Tz>
> Tz>>
> Tz>> AM>also if were working on mod_opal why did you not try to involve us
> and the
> Tz>> AM>opal team?
> Tz>>
> Tz>> Because i made patches for libopal, one is a bugfix in rtp part, there
> is a race condition
> Tz>> in inicialisation in jitter buffer, another patch implements method to
> send progress indicator,
> Tz>> and i don't wont spent my time to incorporate this changes into
> libopal.
> Tz>
> Tz>Thats bad.
> Tz>Any bugfixes from fs, goes to upstream on any of the used libraries.
> Tz>You should be doing the same.
> Tz>And Opal developers, will either include or refuse your patches. If
> Tz>they refuse it, they will give you the reason.
>
> i make this fix only to freeze my current mod_opal work on working state,
> while it now work for me i work on
> my new implementation of h323 proto for fs, i think opal developers will
> fix this rtp bug himself becouse
> it crashes and make library unuseful.
>
> Tz>
> Tz>> without this changes
> Tz>> my work on mod_opal in freeswitch don't useful at all, i provide link
> to my work with all
> Tz>> patches, if somebody wont incorporate it in libopal tree and fs - go
> on, but i think
> Tz>> better and more elegant make new module based on h323plus.
> Tz>
> Tz>If you dont publish your changes, all those you are trying to achieve,
> Tz>wont happen.
> Tz>
> Tz>>
> Tz>> AM>How far away from what is in tree are these patches you have?
> Tz>> AM>
> Tz>> AM>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
> Tz>> AM>
> Tz>> AM>> On 2009-10-07 15:09 -0500, Brian West wrote
> Tz>> AM>> freeswitch-users at lists.freeswit...:
> Tz>> AM>>
> Tz>> AM>> opal have addition abstraction layer called opalmgr, and it
> implementation
> Tz>> AM>> is not so good in
> Tz>> AM>> this case, for example to implemet pre_answer in mod_opal i need
> patch
> Tz>> AM>> libopal, because
> Tz>
> Tz>The patch you have in there, adds a method to the OpalCall,  it does
> Tz>not touch any parts of OpalManager
> Tz>so, i dont understand why opalmanager would be the cause of your pain?
> Tz>
> Tz>> AM>> there is no way to send progress inicator throuch opalmgr. and
> there is
> Tz>> AM>> many another issues like
> Tz>> AM>> this in that layer.
> Tz>
> Tz>Please point me to the issues you have in opal, their bug reports ,
> traces etc.
> Tz>I dont think any of the opal people has psychic abilities to detect
> Tz>-your- problems
> Tz>and solve them.
> Tz>
> Tz>>ftp://srv.icf.org.ru/pub/soft/f/freeswitch/ - there is
> Tz>> AM>> my work on mod_opal before
> Tz>> AM>> i start moving to h323plus, may be this help somebody there.
> Tz>> AM>>
> Tz>> AM>> BW> From what I have been told h323plus is a based/fork of
> OpenH323 which
> Tz>> AM>> BW>OPAL is just a continuation of OpenH323.  So why not support
> the
> Tz>> AM>> BW>developers of OPAL/OpenH323 ?
> Tz>> AM>> BW>
> Tz>> AM>> BW>/b
> Tz>> AM>> BW>
> Tz>> AM>> BW>On Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote:
> Tz>> AM>> BW>
> Tz>> AM>> BW>> We are developing module to handle h323 proto now, we try to
> use
> Tz>> AM>> BW>> mod_opal and try improve it, but no luck,
> Tz>> AM>> BW>> there is many issues in libopal, and finaly we now move to
> h323plus
> Tz>> AM>> BW>> library.
> Tz>
> Tz>Did any of you try to report those issues?
> Tz>
> Tz>Regards
> Tz>
> Tz>/tyn
> Tz>
> Tz>> AM>> BW>
> Tz>> AM>> BW>
> Tz>> AM>> BW>_______________________________________________
> Tz>> AM>> BW>FreeSWITCH-users mailing list
> Tz>> AM>> BW>FreeSWITCH-users at lists.freeswitch.org
> Tz>> AM>> BW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> Tz>> AM>> BW>UNSUBSCRIBE:
> Tz>> AM>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> Tz>> AM>> BW>http://www.freeswitch.org
> Tz>> AM>> BW>
> Tz>> AM>>
> Tz>> AM>> C ?????????                       With Best Regards
> Tz>> AM>> ???????????? ????.                Georgiewskiy Yuriy
> Tz>> AM>> +7 4872 711666                    +7 4872 711666
> Tz>> AM>> ???? +7 4872 711143               fax +7 4872 711143
> Tz>> AM>> ???????? ??? "?? ?? ??????"       IT Service Ltd
> Tz>> AM>> http://nkoort.ru                  http://nkoort.ru
> Tz>> AM>> JID: GHhost at jabber.tula-ix.net.ru JID:
> GHhost at jabber.tula-ix.net.ru
> Tz>> AM>> YG129-RIPE                        YG129-RIPE
> Tz>> AM>>
> Tz>> AM>> _______________________________________________
> Tz>> AM>> FreeSWITCH-users mailing list
> Tz>> AM>> FreeSWITCH-users at lists.freeswitch.org
> Tz>> AM>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> Tz>> AM>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> Tz>> AM>> http://www.freeswitch.org
> Tz>> AM>>
> Tz>> AM>>
> Tz>> AM>
> Tz>> AM>
> Tz>> AM>
> Tz>>
> Tz>> C ?????????                       With Best Regards
> Tz>> ???????????? ????.                Georgiewskiy Yuriy
> Tz>> +7 4872 711666                    +7 4872 711666
> Tz>> ???? +7 4872 711143               fax +7 4872 711143
> Tz>> ???????? ??? "?? ?? ??????"       IT Service Ltd
> Tz>> http://nkoort.ru                  http://nkoort.ru
> Tz>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> Tz>> YG129-RIPE                        YG129-RIPE
> Tz>> _______________________________________________
> Tz>> FreeSWITCH-users mailing list
> Tz>> FreeSWITCH-users at lists.freeswitch.org
> Tz>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> Tz>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> Tz>> http://www.freeswitch.org
> Tz>>
> Tz>>
> Tz>
> Tz>_______________________________________________
> Tz>FreeSWITCH-users mailing list
> Tz>FreeSWITCH-users at lists.freeswitch.org
> Tz>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> Tz>UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> Tz>http://www.freeswitch.org
> Tz>
>
> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129-RIPE
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Oct  8 11:38:37 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Oct 2009 11:38:37 -0700
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <1b46b4e80910070743h38d39e5ax362ff7944754067e@mail.gmail.com>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<87f2f3b90910050920w4d17033fl4c386a0018107bdf@mail.gmail.com>
	<1b46b4e80910052119me12a0d9y4653edd831f9555e@mail.gmail.com>
	<1b46b4e80910052142jaa09f08l72e5e4b2b24d10fc@mail.gmail.com>
	<083DBCB9-48B2-4E61-81B5-B7D00A42EAB5@freeswitch.org>
	<1b46b4e80910061422q7be2a21fg5d2caaef7e307abb@mail.gmail.com>
	<9106C8E1-498F-4AE8-8D34-7F6AAD4DC8AE@freeswitch.org>
	<1b46b4e80910070648n622c3f9haacc22c2328b44ff@mail.gmail.com>
	<8750A70D-6891-4C07-B4DF-C7F4ABA58D67@freeswitch.org>
	<1b46b4e80910070743h38d39e5ax362ff7944754067e@mail.gmail.com>
Message-ID: <87f2f3b90910081138i32d1c28dmf229344f4b5a4e84@mail.gmail.com>

On Wed, Oct 7, 2009 at 7:43 AM, Nicolas Brenner <nicolas at medularis.com>wrote:

> You are missing the point, it is only rejecting auth for FS, Asterisk and
> X-Lite work fine with the same config for that gateway.
>
> Did you figure this one out yet? If not, snag some SIP captures of working
vs. non-working REGISTRATIONs and pastebin them along with your failed reg
and your redacted gateway config.
-MC


>
>
> On Wed, Oct 7, 2009 at 10:20 AM, Brian West <brian at freeswitch.org> wrote:
>
>> I would suspect its a PEBKAC.  I mean if you could register to a
>> gateway that rejected auth... what purpose would auth serve in the
>> first place?
>>
>> /b
>>
>> On Oct 7, 2009, at 8:48 AM, Nicolas Brenner wrote:
>>
>> > Is there some way to make FS register with the gateway that is
>> > rejecting the authentication? is it FS or the SIP server at fault?
>> > Why would X-Lite work and FS not?
>> >
>> > Thanks again for your time and help.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bottleman at icf.org.ru  Thu Oct  8 11:44:38 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Thu, 8 Oct 2009 22:44:38 +0400 (MSD)
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910082242570.16063@srv.icf.org.ru>

On 2009-10-08 20:32 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>Hi Yuriy,
TC>
TC>can you share what you have so far, I'm sure we can help with RTP part...

I think there is a few days and i make it work, after this i start to test and share it.

TC>
TC>T.
TC>
TC>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
TC>
TC>> On 2009-10-08 13:25 -0400, Tuyan ?zipek wrote
TC>> freeswitch-users at lists.freesw...:
TC>>
TC>> Tz>Hi,
TC>> Tz>
TC>> Tz>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>:
TC>> Tz>> On 2009-10-08 10:43 -0500, Anthony Minessale wrote
TC>> freeswitch-users at lists.f...:
TC>> Tz>>
TC>> Tz>> AM>If you are going to make that alternate module are you going to
TC>> host it in
TC>> Tz>> AM>the FS tree along side mod_opal?
TC>> Tz>>
TC>> Tz>> Yes, but then it be useful, now i have working only signaling part and
TC>> some
TC>> Tz>> kinde of not working rtp part :)
TC>> Tz>
TC>> Tz>If you dont use fs rtp stack, its unlikely that it will be accepted
TC>> Tz>into the tree.
TC>> Tz>
TC>> Tz>>
TC>> Tz>> AM>also if were working on mod_opal why did you not try to involve us
TC>> and the
TC>> Tz>> AM>opal team?
TC>> Tz>>
TC>> Tz>> Because i made patches for libopal, one is a bugfix in rtp part, there
TC>> is a race condition
TC>> Tz>> in inicialisation in jitter buffer, another patch implements method to
TC>> send progress indicator,
TC>> Tz>> and i don't wont spent my time to incorporate this changes into
TC>> libopal.
TC>> Tz>
TC>> Tz>Thats bad.
TC>> Tz>Any bugfixes from fs, goes to upstream on any of the used libraries.
TC>> Tz>You should be doing the same.
TC>> Tz>And Opal developers, will either include or refuse your patches. If
TC>> Tz>they refuse it, they will give you the reason.
TC>>
TC>> i make this fix only to freeze my current mod_opal work on working state,
TC>> while it now work for me i work on
TC>> my new implementation of h323 proto for fs, i think opal developers will
TC>> fix this rtp bug himself becouse
TC>> it crashes and make library unuseful.
TC>>
TC>> Tz>
TC>> Tz>> without this changes
TC>> Tz>> my work on mod_opal in freeswitch don't useful at all, i provide link
TC>> to my work with all
TC>> Tz>> patches, if somebody wont incorporate it in libopal tree and fs - go
TC>> on, but i think
TC>> Tz>> better and more elegant make new module based on h323plus.
TC>> Tz>
TC>> Tz>If you dont publish your changes, all those you are trying to achieve,
TC>> Tz>wont happen.
TC>> Tz>
TC>> Tz>>
TC>> Tz>> AM>How far away from what is in tree are these patches you have?
TC>> Tz>> AM>
TC>> Tz>> AM>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
TC>> Tz>> AM>
TC>> Tz>> AM>> On 2009-10-07 15:09 -0500, Brian West wrote
TC>> Tz>> AM>> freeswitch-users at lists.freeswit...:
TC>> Tz>> AM>>
TC>> Tz>> AM>> opal have addition abstraction layer called opalmgr, and it
TC>> implementation
TC>> Tz>> AM>> is not so good in
TC>> Tz>> AM>> this case, for example to implemet pre_answer in mod_opal i need
TC>> patch
TC>> Tz>> AM>> libopal, because
TC>> Tz>
TC>> Tz>The patch you have in there, adds a method to the OpalCall,  it does
TC>> Tz>not touch any parts of OpalManager
TC>> Tz>so, i dont understand why opalmanager would be the cause of your pain?
TC>> Tz>
TC>> Tz>> AM>> there is no way to send progress inicator throuch opalmgr. and
TC>> there is
TC>> Tz>> AM>> many another issues like
TC>> Tz>> AM>> this in that layer.
TC>> Tz>
TC>> Tz>Please point me to the issues you have in opal, their bug reports ,
TC>> traces etc.
TC>> Tz>I dont think any of the opal people has psychic abilities to detect
TC>> Tz>-your- problems
TC>> Tz>and solve them.
TC>> Tz>
TC>> Tz>>ftp://srv.icf.org.ru/pub/soft/f/freeswitch/ - there is
TC>> Tz>> AM>> my work on mod_opal before
TC>> Tz>> AM>> i start moving to h323plus, may be this help somebody there.
TC>> Tz>> AM>>
TC>> Tz>> AM>> BW> From what I have been told h323plus is a based/fork of
TC>> OpenH323 which
TC>> Tz>> AM>> BW>OPAL is just a continuation of OpenH323.  So why not support
TC>> the
TC>> Tz>> AM>> BW>developers of OPAL/OpenH323 ?
TC>> Tz>> AM>> BW>
TC>> Tz>> AM>> BW>/b
TC>> Tz>> AM>> BW>
TC>> Tz>> AM>> BW>On Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote:
TC>> Tz>> AM>> BW>
TC>> Tz>> AM>> BW>> We are developing module to handle h323 proto now, we try to
TC>> use
TC>> Tz>> AM>> BW>> mod_opal and try improve it, but no luck,
TC>> Tz>> AM>> BW>> there is many issues in libopal, and finaly we now move to
TC>> h323plus
TC>> Tz>> AM>> BW>> library.
TC>> Tz>
TC>> Tz>Did any of you try to report those issues?
TC>> Tz>
TC>> Tz>Regards
TC>> Tz>
TC>> Tz>/tyn
TC>> Tz>
TC>> Tz>> AM>> BW>
TC>> Tz>> AM>> BW>
TC>> Tz>> AM>> BW>_______________________________________________
TC>> Tz>> AM>> BW>FreeSWITCH-users mailing list
TC>> Tz>> AM>> BW>FreeSWITCH-users at lists.freeswitch.org
TC>> Tz>> AM>> BW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> Tz>> AM>> BW>UNSUBSCRIBE:
TC>> Tz>> AM>> http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> Tz>> AM>> BW>http://www.freeswitch.org
TC>> Tz>> AM>> BW>
TC>> Tz>> AM>>
TC>> Tz>> AM>> C ?????????                       With Best Regards
TC>> Tz>> AM>> ???????????? ????.                Georgiewskiy Yuriy
TC>> Tz>> AM>> +7 4872 711666                    +7 4872 711666
TC>> Tz>> AM>> ???? +7 4872 711143               fax +7 4872 711143
TC>> Tz>> AM>> ???????? ??? "?? ?? ??????"       IT Service Ltd
TC>> Tz>> AM>> http://nkoort.ru                  http://nkoort.ru
TC>> Tz>> AM>> JID: GHhost at jabber.tula-ix.net.ru JID:
TC>> GHhost at jabber.tula-ix.net.ru
TC>> Tz>> AM>> YG129-RIPE                        YG129-RIPE
TC>> Tz>> AM>>
TC>> Tz>> AM>> _______________________________________________
TC>> Tz>> AM>> FreeSWITCH-users mailing list
TC>> Tz>> AM>> FreeSWITCH-users at lists.freeswitch.org
TC>> Tz>> AM>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> Tz>> AM>> UNSUBSCRIBE:
TC>> http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> Tz>> AM>> http://www.freeswitch.org
TC>> Tz>> AM>>
TC>> Tz>> AM>>
TC>> Tz>> AM>
TC>> Tz>> AM>
TC>> Tz>> AM>
TC>> Tz>>
TC>> Tz>> C ?????????                       With Best Regards
TC>> Tz>> ???????????? ????.                Georgiewskiy Yuriy
TC>> Tz>> +7 4872 711666                    +7 4872 711666
TC>> Tz>> ???? +7 4872 711143               fax +7 4872 711143
TC>> Tz>> ???????? ??? "?? ?? ??????"       IT Service Ltd
TC>> Tz>> http://nkoort.ru                  http://nkoort.ru
TC>> Tz>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
TC>> Tz>> YG129-RIPE                        YG129-RIPE
TC>> Tz>> _______________________________________________
TC>> Tz>> FreeSWITCH-users mailing list
TC>> Tz>> FreeSWITCH-users at lists.freeswitch.org
TC>> Tz>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> Tz>> UNSUBSCRIBE:
TC>> http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> Tz>> http://www.freeswitch.org
TC>> Tz>>
TC>> Tz>>
TC>> Tz>
TC>> Tz>_______________________________________________
TC>> Tz>FreeSWITCH-users mailing list
TC>> Tz>FreeSWITCH-users at lists.freeswitch.org
TC>> Tz>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> Tz>UNSUBSCRIBE:
TC>> http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> Tz>http://www.freeswitch.org
TC>> Tz>
TC>>
TC>> C ?????????                       With Best Regards
TC>> ???????????? ????.                Georgiewskiy Yuriy
TC>> +7 4872 711666                    +7 4872 711666
TC>> ???? +7 4872 711143               fax +7 4872 711143
TC>> ???????? ??? "?? ?? ??????"       IT Service Ltd
TC>> http://nkoort.ru                  http://nkoort.ru
TC>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
TC>> YG129-RIPE                        YG129-RIPE
TC>>
TC>> _______________________________________________
TC>> FreeSWITCH-users mailing list
TC>> FreeSWITCH-users at lists.freeswitch.org
TC>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> http://www.freeswitch.org
TC>>
TC>>
TC>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From gabe at gundy.org  Thu Oct  8 12:01:35 2009
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 8 Oct 2009 13:01:35 -0600
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <87f2f3b90910080958i1f5cfccax621fd12100a05e3e@mail.gmail.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<87f2f3b90910080958i1f5cfccax621fd12100a05e3e@mail.gmail.com>
Message-ID: <903da5680910081201p427a8358n768edb61a96674a4@mail.gmail.com>

Naturally the wireless is jacked at the conf.  It's no match for the
100's of geeks sporting laptops.  Thanks for the link. Looks like
we'll be shooting from the hip. :)

>From my phone,
Gabe

On Thursday, October 8, 2009, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Thu, Oct 8, 2009 at 7:42 AM, Gabriel Gunderson <gabe at gundy.org?<javascript:_e({}, 'cvml', 'gabe at gundy.org');>> wrote:
>
> Hey all,
>
> The 3rd annual UTOSC (Utah Open Source Conference) [1] starts today.
> It's a conference that gives Open Source users/hackers a chance gather
> together to mingle and share. ?It's a lot of fun. ?Anyway, this year
> I've offered to work with a friend of mine to lead a BoF (Birds of a
> Feather) discussion on Open Source Telephony [2].
>
> While it should mostly be group discussion, you never know when people
> are going to get shy and clam up. ?*If* they do, it would be nice to
> have some slides to fill that time. ?Rather than make some from
> scratch (that are unlikely to get used), I thought I'd ask here and
> see if anyone has a FS slide deck that they wouldn't mind sharing.
>
> Thanks,
> Gabe
>
>
> 1) http://2009.utosc.com/pages/home/
> 2) http://2009.utosc.com/presentation/136/
>
> There are numerous slides and such from ClueCon. Go to http://www.cluecon.com and you can download both the video files as well as the actual presentations. The presentation files are in various formats such as PPT(X), PDF, and Keynote.
> The second presentation on day 1 is Anthony Minessale and he talks a lot about expanding one's thinking when it comes to applying FreeSWITCH in production. For example, some people will jump through crazy hoops to solve a problem "because that's how they've always done it." Anyway, check out that video and presentation file for some great ideas.
> -MC
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Thu Oct  8 12:02:43 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Oct 2009 12:02:43 -0700
Subject: [Freeswitch-users] REMINDER: Weekly FreeSWITCH Conference Scheduled
	for October 9, 11AM 	CST (GMT -6)
Message-ID: <87f2f3b90910081202m5dd3f567v240541dd2009c9a8@mail.gmail.com>

Just a friendly reminder that we will be having the weekly FreeSWITCH
conference call tomorrow, October 9th, at 11AM Central. The agenda is
updated:
http://bit.ly/lzEYy

The conference lasts for six hours, so feel free to dial in at any time. We
usually start going over the agenda about 15 minutes after the conference
starts. The agenda itself only takes about an hour or two, after which we
spend most of the time discussing whatever is on the minds of the community
members. Remember, the core FreeSWITCH development team of Anthony Minessale
(primary author, lead architect), Mike Jerris (build master), and Brian West
(all around VoIP expert, FS configuration guru) are present for the whole
conference. This is a great time to ask questions about the project. Please
join us!

Call in options include SIP, Skype, PSTN, and Jabber. Please see the agenda
for more information.

Looking forward to speaking with you all tomorrow,
Michael
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From tculjaga at gmail.com  Thu Oct  8 12:03:36 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 8 Oct 2009 21:03:36 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910082242570.16063@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910082242570.16063@srv.icf.org.ru>
Message-ID: <65d96fc80910081203x5a8dfc73n57ebf78cc532b920@mail.gmail.com>

k

2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-08 20:32 +0200, Tihomir Culjaga wrote
> freeswitch-users at lists.fre...:
>
> TC>Hi Yuriy,
> TC>
> TC>can you share what you have so far, I'm sure we can help with RTP
> part...
>
> I think there is a few days and i make it work, after this i start to test
> and share it.
>
> TC>
> TC>T.
> TC>
> TC>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
> TC>
> TC>> On 2009-10-08 13:25 -0400, Tuyan ?zipek wrote
> TC>> freeswitch-users at lists.freesw...:
> TC>>
> TC>> Tz>Hi,
> TC>> Tz>
> TC>> Tz>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>:
> TC>> Tz>> On 2009-10-08 10:43 -0500, Anthony Minessale wrote
> TC>> freeswitch-users at lists.f...:
> TC>> Tz>>
> TC>> Tz>> AM>If you are going to make that alternate module are you going
> to
> TC>> host it in
> TC>> Tz>> AM>the FS tree along side mod_opal?
> TC>> Tz>>
> TC>> Tz>> Yes, but then it be useful, now i have working only signaling
> part and
> TC>> some
> TC>> Tz>> kinde of not working rtp part :)
> TC>> Tz>
> TC>> Tz>If you dont use fs rtp stack, its unlikely that it will be accepted
> TC>> Tz>into the tree.
> TC>> Tz>
> TC>> Tz>>
> TC>> Tz>> AM>also if were working on mod_opal why did you not try to
> involve us
> TC>> and the
> TC>> Tz>> AM>opal team?
> TC>> Tz>>
> TC>> Tz>> Because i made patches for libopal, one is a bugfix in rtp part,
> there
> TC>> is a race condition
> TC>> Tz>> in inicialisation in jitter buffer, another patch implements
> method to
> TC>> send progress indicator,
> TC>> Tz>> and i don't wont spent my time to incorporate this changes into
> TC>> libopal.
> TC>> Tz>
> TC>> Tz>Thats bad.
> TC>> Tz>Any bugfixes from fs, goes to upstream on any of the used
> libraries.
> TC>> Tz>You should be doing the same.
> TC>> Tz>And Opal developers, will either include or refuse your patches. If
> TC>> Tz>they refuse it, they will give you the reason.
> TC>>
> TC>> i make this fix only to freeze my current mod_opal work on working
> state,
> TC>> while it now work for me i work on
> TC>> my new implementation of h323 proto for fs, i think opal developers
> will
> TC>> fix this rtp bug himself becouse
> TC>> it crashes and make library unuseful.
> TC>>
> TC>> Tz>
> TC>> Tz>> without this changes
> TC>> Tz>> my work on mod_opal in freeswitch don't useful at all, i provide
> link
> TC>> to my work with all
> TC>> Tz>> patches, if somebody wont incorporate it in libopal tree and fs -
> go
> TC>> on, but i think
> TC>> Tz>> better and more elegant make new module based on h323plus.
> TC>> Tz>
> TC>> Tz>If you dont publish your changes, all those you are trying to
> achieve,
> TC>> Tz>wont happen.
> TC>> Tz>
> TC>> Tz>>
> TC>> Tz>> AM>How far away from what is in tree are these patches you have?
> TC>> Tz>> AM>
> TC>> Tz>> AM>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
> TC>> Tz>> AM>
> TC>> Tz>> AM>> On 2009-10-07 15:09 -0500, Brian West wrote
> TC>> Tz>> AM>> freeswitch-users at lists.freeswit...:
> TC>> Tz>> AM>>
> TC>> Tz>> AM>> opal have addition abstraction layer called opalmgr, and it
> TC>> implementation
> TC>> Tz>> AM>> is not so good in
> TC>> Tz>> AM>> this case, for example to implemet pre_answer in mod_opal i
> need
> TC>> patch
> TC>> Tz>> AM>> libopal, because
> TC>> Tz>
> TC>> Tz>The patch you have in there, adds a method to the OpalCall,  it
> does
> TC>> Tz>not touch any parts of OpalManager
> TC>> Tz>so, i dont understand why opalmanager would be the cause of your
> pain?
> TC>> Tz>
> TC>> Tz>> AM>> there is no way to send progress inicator throuch opalmgr.
> and
> TC>> there is
> TC>> Tz>> AM>> many another issues like
> TC>> Tz>> AM>> this in that layer.
> TC>> Tz>
> TC>> Tz>Please point me to the issues you have in opal, their bug reports ,
> TC>> traces etc.
> TC>> Tz>I dont think any of the opal people has psychic abilities to detect
> TC>> Tz>-your- problems
> TC>> Tz>and solve them.
> TC>> Tz>
> TC>> Tz>>ftp://srv.icf.org.ru/pub/soft/f/freeswitch/ - there is
> TC>> Tz>> AM>> my work on mod_opal before
> TC>> Tz>> AM>> i start moving to h323plus, may be this help somebody there.
> TC>> Tz>> AM>>
> TC>> Tz>> AM>> BW> From what I have been told h323plus is a based/fork of
> TC>> OpenH323 which
> TC>> Tz>> AM>> BW>OPAL is just a continuation of OpenH323.  So why not
> support
> TC>> the
> TC>> Tz>> AM>> BW>developers of OPAL/OpenH323 ?
> TC>> Tz>> AM>> BW>
> TC>> Tz>> AM>> BW>/b
> TC>> Tz>> AM>> BW>
> TC>> Tz>> AM>> BW>On Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote:
> TC>> Tz>> AM>> BW>
> TC>> Tz>> AM>> BW>> We are developing module to handle h323 proto now, we
> try to
> TC>> use
> TC>> Tz>> AM>> BW>> mod_opal and try improve it, but no luck,
> TC>> Tz>> AM>> BW>> there is many issues in libopal, and finaly we now move
> to
> TC>> h323plus
> TC>> Tz>> AM>> BW>> library.
> TC>> Tz>
> TC>> Tz>Did any of you try to report those issues?
> TC>> Tz>
> TC>> Tz>Regards
> TC>> Tz>
> TC>> Tz>/tyn
> TC>> Tz>
> TC>> Tz>> AM>> BW>
> TC>> Tz>> AM>> BW>
> TC>> Tz>> AM>> BW>_______________________________________________
> TC>> Tz>> AM>> BW>FreeSWITCH-users mailing list
> TC>> Tz>> AM>> BW>FreeSWITCH-users at lists.freeswitch.org
> TC>> Tz>> AM>> BW>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> TC>> Tz>> AM>> BW>UNSUBSCRIBE:
> TC>> Tz>> AM>>
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> TC>> Tz>> AM>> BW>http://www.freeswitch.org
> TC>> Tz>> AM>> BW>
> TC>> Tz>> AM>>
> TC>> Tz>> AM>> C ?????????                       With Best Regards
> TC>> Tz>> AM>> ???????????? ????.                Georgiewskiy Yuriy
> TC>> Tz>> AM>> +7 4872 711666                    +7 4872 711666
> TC>> Tz>> AM>> ???? +7 4872 711143               fax +7 4872 711143
> TC>> Tz>> AM>> ???????? ??? "?? ?? ??????"       IT Service Ltd
> TC>> Tz>> AM>> http://nkoort.ru                  http://nkoort.ru
> TC>> Tz>> AM>> JID: GHhost at jabber.tula-ix.net.ru JID:
> TC>> GHhost at jabber.tula-ix.net.ru
> TC>> Tz>> AM>> YG129-RIPE                        YG129-RIPE
> TC>> Tz>> AM>>
> TC>> Tz>> AM>> _______________________________________________
> TC>> Tz>> AM>> FreeSWITCH-users mailing list
> TC>> Tz>> AM>> FreeSWITCH-users at lists.freeswitch.org
> TC>> Tz>> AM>>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> TC>> Tz>> AM>> UNSUBSCRIBE:
> TC>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> TC>> Tz>> AM>> http://www.freeswitch.org
> TC>> Tz>> AM>>
> TC>> Tz>> AM>>
> TC>> Tz>> AM>
> TC>> Tz>> AM>
> TC>> Tz>> AM>
> TC>> Tz>>
> TC>> Tz>> C ?????????                       With Best Regards
> TC>> Tz>> ???????????? ????.                Georgiewskiy Yuriy
> TC>> Tz>> +7 4872 711666                    +7 4872 711666
> TC>> Tz>> ???? +7 4872 711143               fax +7 4872 711143
> TC>> Tz>> ???????? ??? "?? ?? ??????"       IT Service Ltd
> TC>> Tz>> http://nkoort.ru                  http://nkoort.ru
> TC>> Tz>> JID: GHhost at jabber.tula-ix.net.ru JID:
> GHhost at jabber.tula-ix.net.ru
> TC>> Tz>> YG129-RIPE                        YG129-RIPE
> TC>> Tz>> _______________________________________________
> TC>> Tz>> FreeSWITCH-users mailing list
> TC>> Tz>> FreeSWITCH-users at lists.freeswitch.org
> TC>> Tz>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> TC>> Tz>> UNSUBSCRIBE:
> TC>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> TC>> Tz>> http://www.freeswitch.org
> TC>> Tz>>
> TC>> Tz>>
> TC>> Tz>
> TC>> Tz>_______________________________________________
> TC>> Tz>FreeSWITCH-users mailing list
> TC>> Tz>FreeSWITCH-users at lists.freeswitch.org
> TC>> Tz>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> TC>> Tz>UNSUBSCRIBE:
> TC>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> TC>> Tz>http://www.freeswitch.org
> TC>> Tz>
> TC>>
> TC>> C ?????????                       With Best Regards
> TC>> ???????????? ????.                Georgiewskiy Yuriy
> TC>> +7 4872 711666                    +7 4872 711666
> TC>> ???? +7 4872 711143               fax +7 4872 711143
> TC>> ???????? ??? "?? ?? ??????"       IT Service Ltd
> TC>> http://nkoort.ru                  http://nkoort.ru
> TC>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> TC>> YG129-RIPE                        YG129-RIPE
> TC>>
> TC>> _______________________________________________
> TC>> FreeSWITCH-users mailing list
> TC>> FreeSWITCH-users at lists.freeswitch.org
> TC>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> TC>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> TC>> http://www.freeswitch.org
> TC>>
> TC>>
> TC>
>
> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129-RIPE
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From xengelpublicx at gmail.com  Thu Oct  8 12:50:44 2009
From: xengelpublicx at gmail.com (Vladimir Elizarov)
Date: Thu, 08 Oct 2009 23:50:44 +0400
Subject: [Freeswitch-users] freeswitch distributed
Message-ID: <4ACE4294.3030402@gmail.com>

I want to make freeswitch distributed in such a situation. Several
offices in each of fs. We need a unified dialplan, a single
entrance/exit on the SIP external gateways, in the case of
unavailability of one of the fs others work without his participation.

How to understand. It should be the following. Base presence to keep in
a separate database (pgpoll + pg). Dialplan, configure pull xml_curl.
What can you tell from the web-application (wikipbx, fs2web etc)? The
main task of which will impact dialplan, configure, directory?

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From dmitry.bely at gmail.com  Thu Oct  8 13:23:32 2009
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Fri, 9 Oct 2009 00:23:32 +0400
Subject: [Freeswitch-users] How to limit the number of incoming+outgoing
	calls via specific gateway?
In-Reply-To: <7e2ac3270910051352p7d052a5aife6f07827173767@mail.gmail.com>
References: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>
	<e403cddca1c90c83ce4fb6f85b026e63.squirrel@correo.nodo50.org>
	<65d96fc80910020532i4ba33353g7ac141624a4bb89e@mail.gmail.com>
	<C1AD67AA-F969-4BA1-8415-5E381BDF18FA@jerris.com>
	<90823c940910051257u403108afg3478c4c13ab26283@mail.gmail.com>
	<7e2ac3270910051319xfdfc9d3wa583c2da7e932339@mail.gmail.com>
	<90823c940910051339h3614be97o48a6319f68da7977@mail.gmail.com>
	<7e2ac3270910051352p7d052a5aife6f07827173767@mail.gmail.com>
Message-ID: <90823c940910081323j73e63385kdaab8cf3ac77d139@mail.gmail.com>

On Tue, Oct 6, 2009 at 12:52 AM, SP <sprice at gmail.com> wrote:
> did you use the application limit on the inbound call?
>?You'll need to in order to account for it.

Can you provide more detail? I have DID and gateway

  <extension name="public_did">
    <condition field="destination_number" expression="^(1234567)$">
      <action application="transfer" data="incoming_call XML default"/>
    </condition>
  </extension>

  <gateway name="my_provider">
    <param name="username" value="1234567"/>
(...)
  </gateway>

  <extension name="my_provider_local">
    <condition field="${toll_allow}" expression="local"/>
(...)
      <action application="bridge" data="sofia/gateway/my_provider/$1"/>
    </condition>
  </extension>

that correspond to the same provider account. Then an incoming call is
active, outgoing one is not possible (provider will reject the it).
How to generate "limit_exceeded" then?

- Dmitry Bely


From msc at freeswitch.org  Thu Oct  8 15:38:10 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Oct 2009 15:38:10 -0700
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <903da5680910081201p427a8358n768edb61a96674a4@mail.gmail.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<87f2f3b90910080958i1f5cfccax621fd12100a05e3e@mail.gmail.com>
	<903da5680910081201p427a8358n768edb61a96674a4@mail.gmail.com>
Message-ID: <87f2f3b90910081538l6e4e4b7dv3aedb2c4d1af58bc@mail.gmail.com>

On Thu, Oct 8, 2009 at 12:01 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> Naturally the wireless is jacked at the conf.  It's no match for the
> 100's of geeks sporting laptops.  Thanks for the link. Looks like
> we'll be shooting from the hip. :)
>
> We experienced something similar at ClueCon, but it was that the hotel's
DHCP server was a total wuss. :)
-MC
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From msc at freeswitch.org  Thu Oct  8 15:44:49 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Oct 2009 15:44:49 -0700
Subject: [Freeswitch-users] Changing callerid before/after call pickup
In-Reply-To: <A2F1C0E212BBBA4E8C0E7BE568361F04027F0233024A@srv02.khdev.corp>
References: <4ACB9519.9000805@gmail.com>
	<2d9149cd0910071235k27c1429v2bc555c0e7397fc4@mail.gmail.com>
	<4ACD076E.2080107@gmail.com>
	<5B417827-6CCB-4EE4-9524-5CD45AC9E28F@gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F04027F0233024A@srv02.khdev.corp>
Message-ID: <87f2f3b90910081544h3f86404fid50dfb644cbbb9f5@mail.gmail.com>

On Wed, Oct 7, 2009 at 6:46 PM, Klaus Hochlehnert <Mailings at kh-dev.de>wrote:

> Hi,
>
> currently I'm playing around with call pickup and Snom phones.
> I'm using the intercept function for that.
>
> My "problem" is now that after the call pickup (which works fine) I don't
> see the caller id of the original call.
> Instead I see the pickup code, e.g. *820
>
> I've tried to change nearly every channel variable (before doing the call
> pickup), but the phone won't display the caller id.
>
> Does anybody know how to change that?
>
Can you paste a debug log of the call flow? Also, are you using just the
default dialplan or have you made any changes? Lastly, what mods, if any,
did you make to your Snom configs? I have a bricked Snom that I'm waiting
for TFTP to fix but when it does I will see about testing this.

-MC
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From msc at freeswitch.org  Thu Oct  8 15:50:42 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Oct 2009 15:50:42 -0700
Subject: [Freeswitch-users] freeswitch distributed
In-Reply-To: <4ACE4294.3030402@gmail.com>
References: <4ACE4294.3030402@gmail.com>
Message-ID: <87f2f3b90910081550r7596fd11v71c406304e15478e@mail.gmail.com>

On Thu, Oct 8, 2009 at 12:50 PM, Vladimir Elizarov
<xengelpublicx at gmail.com>wrote:

> I want to make freeswitch distributed in such a situation. Several
> offices in each of fs. We need a unified dialplan, a single
> entrance/exit on the SIP external gateways, in the case of
> unavailability of one of the fs others work without his participation.
>
> How to understand. It should be the following. Base presence to keep in
> a separate database (pgpoll + pg). Dialplan, configure pull xml_curl.
> What can you tell from the web-application (wikipbx, fs2web etc)? The
> main task of which will impact dialplan, configure, directory?
>
>
This is quite an in depth question. It's not really something that can be
answered on a mailing list because of all of the specific questions that
need to be answered. I think your best bet is to email
consulting at freeswitch.org and let them discuss it with you. They can get you
hooked up with a FreeSWITCH expert whom you can hire to assist with this
project.
-MC
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From msc at freeswitch.org  Thu Oct  8 15:54:23 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Oct 2009 15:54:23 -0700
Subject: [Freeswitch-users] Changing callerid before/after call pickup
In-Reply-To: <A2F1C0E212BBBA4E8C0E7BE568361F04027F0233024A@srv02.khdev.corp>
References: <4ACB9519.9000805@gmail.com>
	<2d9149cd0910071235k27c1429v2bc555c0e7397fc4@mail.gmail.com>
	<4ACD076E.2080107@gmail.com>
	<5B417827-6CCB-4EE4-9524-5CD45AC9E28F@gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F04027F0233024A@srv02.khdev.corp>
Message-ID: <87f2f3b90910081554g187e1412hac62fa0a387a4c1a@mail.gmail.com>

Also,

I failed to mention two things:
#1 - Please don't hijack threads. :)
#2 - Update to the latest SVN and try again. You might be pleasantly
surprised.

-MC

On Wed, Oct 7, 2009 at 6:46 PM, Klaus Hochlehnert <Mailings at kh-dev.de>wrote:

> Hi,
>
> currently I'm playing around with call pickup and Snom phones.
> I'm using the intercept function for that.
>
> My "problem" is now that after the call pickup (which works fine) I don't
> see the caller id of the original call.
> Instead I see the pickup code, e.g. *820
>
> I've tried to change nearly every channel variable (before doing the call
> pickup), but the phone won't display the caller id.
>
> Does anybody know how to change that?
>
> Thanks, Klaus
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Mailings at kh-dev.de  Thu Oct  8 16:23:25 2009
From: Mailings at kh-dev.de (Klaus Hochlehnert)
Date: Fri, 9 Oct 2009 01:23:25 +0200
Subject: [Freeswitch-users] Changing callerid before/after call pickup
In-Reply-To: <87f2f3b90910081554g187e1412hac62fa0a387a4c1a@mail.gmail.com>
References: <4ACB9519.9000805@gmail.com>
	<2d9149cd0910071235k27c1429v2bc555c0e7397fc4@mail.gmail.com>
	<4ACD076E.2080107@gmail.com>
	<5B417827-6CCB-4EE4-9524-5CD45AC9E28F@gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F04027F0233024A@srv02.khdev.corp>
	<87f2f3b90910081554g187e1412hac62fa0a387a4c1a@mail.gmail.com>
Message-ID: <A2F1C0E212BBBA4E8C0E7BE568361F04027F0233024D@srv02.khdev.corp>

Sorry for hijacking. Didn't know that the mailing list system finds out about that... ;-)

Will do the update in the next days and try again.

Today I tried with the command  send_display.
This workaround currently works for me.

Thanks, Klaus


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Friday, October 09, 2009 12:54 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Changing callerid before/after call pickup

Also,

I failed to mention two things:
#1 - Please don't hijack threads. :)
#2 - Update to the latest SVN and try again. You might be pleasantly surprised.

-MC
On Wed, Oct 7, 2009 at 6:46 PM, Klaus Hochlehnert <Mailings at kh-dev.de<mailto:Mailings at kh-dev.de>> wrote:
Hi,

currently I'm playing around with call pickup and Snom phones.
I'm using the intercept function for that.

My "problem" is now that after the call pickup (which works fine) I don't see the caller id of the original call.
Instead I see the pickup code, e.g. *820

I've tried to change nearly every channel variable (before doing the call pickup), but the phone won't display the caller id.

Does anybody know how to change that?

Thanks, Klaus


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From brian at freeswitch.org  Thu Oct  8 16:44:14 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 8 Oct 2009 18:44:14 -0500
Subject: [Freeswitch-users] Changing callerid before/after call pickup
In-Reply-To: <A2F1C0E212BBBA4E8C0E7BE568361F04027F0233024D@srv02.khdev.corp>
References: <4ACB9519.9000805@gmail.com>
	<2d9149cd0910071235k27c1429v2bc555c0e7397fc4@mail.gmail.com>
	<4ACD076E.2080107@gmail.com>
	<5B417827-6CCB-4EE4-9524-5CD45AC9E28F@gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F04027F0233024A@srv02.khdev.corp>
	<87f2f3b90910081554g187e1412hac62fa0a387a4c1a@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F04027F0233024D@srv02.khdev.corp>
Message-ID: <9B081533-60AD-4D25-858C-D9814D2AF5FE@freeswitch.org>

If you're starting a new topic/thread please click new message.  If  
you click reply, delete the subject and body and change it.. the  
thread is now hijacked.

/b

On Oct 8, 2009, at 6:23 PM, Klaus Hochlehnert wrote:

> Sorry for hijacking. Didn?t know that the mailing list system finds  
> out about that... ;-)
>
> Will do the update in the next days and try again.
>
> Today I tried with the command  send_display.
> This workaround currently works for me.
>
> Thanks, Klaus

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From msc at freeswitch.org  Thu Oct  8 18:42:59 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Oct 2009 18:42:59 -0700
Subject: [Freeswitch-users] FreeSWITCH Update: Valet Parking
Message-ID: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>

FYI,

The FreeSWITCH devs have added valet parking! Check it out:
http://www.freeswitch.org/node/207

Let us know what you think.

Thanks,
Michael
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From diego.viola at gmail.com  Thu Oct  8 19:56:39 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 9 Oct 2009 02:56:39 +0000
Subject: [Freeswitch-users] Heartbeat question
Message-ID: <86a32abc0910081956k91b3600sb8c556e589f5b089@mail.gmail.com>

Hi everyone,

I have a question about FreeSWITCH heartbeat, I have this on my dialplan:

<extension name="test">
<condition field="destination_number" expression="^8084$">
<action application="answer"/>
<action application="enable_heartbeat" data="1"/>
</condition>
</extension>

But when I do "event plain all" I see the heartbeats are being fired every
20 seconds... what I'm doing wrong?

Thanks,

Diego
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From diego.viola at gmail.com  Thu Oct  8 20:03:43 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 9 Oct 2009 03:03:43 +0000
Subject: [Freeswitch-users] Heartbeat question
In-Reply-To: <86a32abc0910081956k91b3600sb8c556e589f5b089@mail.gmail.com>
References: <86a32abc0910081956k91b3600sb8c556e589f5b089@mail.gmail.com>
Message-ID: <86a32abc0910082003v3f5d4e6dua8e7eae18fd2f939@mail.gmail.com>

I want it to fire every 1 second...

On Fri, Oct 9, 2009 at 2:56 AM, Diego Viola <diego.viola at gmail.com> wrote:

> Hi everyone,
>
> I have a question about FreeSWITCH heartbeat, I have this on my dialplan:
>
> <extension name="test">
> <condition field="destination_number" expression="^8084$">
> <action application="answer"/>
> <action application="enable_heartbeat" data="1"/>
> </condition>
> </extension>
>
> But when I do "event plain all" I see the heartbeats are being fired every
> 20 seconds... what I'm doing wrong?
>
> Thanks,
>
> Diego
>
>
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From mcampbellsmith at gmail.com  Thu Oct  8 20:06:37 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Fri, 9 Oct 2009 14:06:37 +1100
Subject: [Freeswitch-users] mod_fax compile fails
Message-ID: <33c87fa30910082006v1e91849fjbf4282541e7ac377@mail.gmail.com>

HI all,

I just tried to update to the latest svn and I get these errors right
at the end after issuing a 'make current'.  I am using Debian Lenny.

making all mod_fax
make[5]: Entering directory `/home/mark/freeswitch/src/mod/applications/mod_fax'
make[6]: Entering directory `/home/mark/freeswitch/src/mod/applications/mod_fax'
make[7]: Entering directory `/home/mark/freeswitch/libs/tiff-3.8.2'
running /bin/sh ./configure  --prefix=/usr/local/freeswitch
--cache-file=/dev/null --srcdir=. --disable-shared --with-pic
--no-create --no-recursion
configure: error: cannot run /bin/sh config/config.sub
make[7]: *** [config.status] Error 1
make[7]: Leaving directory `/home/mark/freeswitch/libs/tiff-3.8.2'
make[6]: *** [../../../../libs/tiff-3.8.2/libtiff/libtiff.la] Error 2
make[6]: Leaving directory `/home/mark/freeswitch/src/mod/applications/mod_fax'
make[5]: *** [all] Error 1
make[5]: Leaving directory `/home/mark/freeswitch/src/mod/applications/mod_fax'
make[4]: *** [mod_fax-all] Error 1
make[4]: Leaving directory `/home/mark/freeswitch/src/mod'
make[3]: *** [all-recursive] Error 1
make[3]: Leaving directory `/home/mark/freeswitch/src'
Making all in build
make[3]: Entering directory `/home/mark/freeswitch/build'
 +-------- FreeSWITCH Build Complete -----------+
 + FreeSWITCH has been successfully built.      +
 + Install by running:                          +
 +                                              +
 +               make install                   +
 +----------------------------------------------+
make[3]: Leaving directory `/home/mark/freeswitch/build'
make[2]: *** [all-recursive] Error 1
make[2]: Leaving directory `/home/mark/freeswitch'
make[1]: *** [all] Error 2
make[1]: Leaving directory `/home/mark/freeswitch'
make: *** [current] Error 2

Also, are the 'Leaving directory / all-recursive' errors going to
cause a problem?

Thanks!
Any ideas what the cause is?


From william.suffill at gmail.com  Thu Oct  8 20:36:15 2009
From: william.suffill at gmail.com (William Suffill)
Date: Thu, 8 Oct 2009 23:36:15 -0400
Subject: [Freeswitch-users] Heartbeat question
In-Reply-To: <86a32abc0910082003v3f5d4e6dua8e7eae18fd2f939@mail.gmail.com>
References: <86a32abc0910081956k91b3600sb8c556e589f5b089@mail.gmail.com> 
	<86a32abc0910082003v3f5d4e6dua8e7eae18fd2f939@mail.gmail.com>
Message-ID: <6b65470d0910082036g204d5dcbw26e1e43ef80c7c84@mail.gmail.com>

Why do you need it every second? If you want real time channel counts
you would  be able to track each create/destroy even instead of
relying on the heartbeat summary.

-- W


From rob4manhere at gmail.com  Thu Oct  8 20:39:10 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Thu, 8 Oct 2009 22:39:10 -0500
Subject: [Freeswitch-users] mod_fax compile fails
In-Reply-To: <33c87fa30910082006v1e91849fjbf4282541e7ac377@mail.gmail.com>
References: <33c87fa30910082006v1e91849fjbf4282541e7ac377@mail.gmail.com>
Message-ID: <343C884A-9AB5-4755-9C5F-42ED705D05E7@gmail.com>

I had that issue too where make current failed on mod_fax (under libs/ 
tiff).  And yeah, it caused a problem where a bunch of modules  
wouldn't load.  You'll want to get it resolved before installing.  I  
ended up moving the existing source aside and re-checked out the  
trunk, which compiled fine.


On Oct 8, 2009, at 10:06 PM, Mark Campbell-Smith wrote:

> HI all,
>
> I just tried to update to the latest svn and I get these errors right
> at the end after issuing a 'make current'.  I am using Debian Lenny.
>
> making all mod_fax
> make[5]: Entering directory `/home/mark/freeswitch/src/mod/ 
> applications/mod_fax'
> make[6]: Entering directory `/home/mark/freeswitch/src/mod/ 
> applications/mod_fax'
> make[7]: Entering directory `/home/mark/freeswitch/libs/tiff-3.8.2'
> running /bin/sh ./configure  --prefix=/usr/local/freeswitch
> --cache-file=/dev/null --srcdir=. --disable-shared --with-pic
> --no-create --no-recursion
> configure: error: cannot run /bin/sh config/config.sub
> make[7]: *** [config.status] Error 1
> make[7]: Leaving directory `/home/mark/freeswitch/libs/tiff-3.8.2'
> make[6]: *** [../../../../libs/tiff-3.8.2/libtiff/libtiff.la] Error 2
> make[6]: Leaving directory `/home/mark/freeswitch/src/mod/ 
> applications/mod_fax'
> make[5]: *** [all] Error 1
> make[5]: Leaving directory `/home/mark/freeswitch/src/mod/ 
> applications/mod_fax'
> make[4]: *** [mod_fax-all] Error 1
> make[4]: Leaving directory `/home/mark/freeswitch/src/mod'
> make[3]: *** [all-recursive] Error 1
> make[3]: Leaving directory `/home/mark/freeswitch/src'
> Making all in build
> make[3]: Entering directory `/home/mark/freeswitch/build'
> +-------- FreeSWITCH Build Complete -----------+
> + FreeSWITCH has been successfully built.      +
> + Install by running:                          +
> +                                              +
> +               make install                   +
> +----------------------------------------------+
> make[3]: Leaving directory `/home/mark/freeswitch/build'
> make[2]: *** [all-recursive] Error 1
> make[2]: Leaving directory `/home/mark/freeswitch'
> make[1]: *** [all] Error 2
> make[1]: Leaving directory `/home/mark/freeswitch'
> make: *** [current] Error 2
>
> Also, are the 'Leaving directory / all-recursive' errors going to
> cause a problem?
>
> Thanks!
> Any ideas what the cause is?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From ahmedmunir007 at gmail.com  Thu Oct  8 20:39:24 2009
From: ahmedmunir007 at gmail.com (Ahmed Munir)
Date: Fri, 9 Oct 2009 08:39:24 +0500
Subject: [Freeswitch-users] Questions regarding to mod_nibble
Message-ID: <a60fed5e0910082039yc8b7ef8x24260041a7c11d31@mail.gmail.com>

 I want to ask three questions related to mod_nibble bills, as I'm listing
down below;

1- Can we select/use dynamic tables for billing using nibble bill?
2- Can we define more than two tables and attributes in nibblebill.conf.xml?
3- As Nibble bill is use to deduct amount of user account, Can we deduct
minutes instead of cash? Because my case is, if a user buy a package and I
only want to deducts his/her minutes. How we can resolve it by nibble bill?
/ What other way we can resolve it?

Kindly advise soon.

-- 
Regards,

Ahmed Munir
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From hads at nice.net.nz  Thu Oct  8 21:54:41 2009
From: hads at nice.net.nz (Hadley Rich)
Date: Fri, 09 Oct 2009 17:54:41 +1300
Subject: [Freeswitch-users] FreeSWITCH Update: Valet Parking
In-Reply-To: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>
References: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>
Message-ID: <1255064081.2669.1.camel@sodium>

On Thu, 2009-10-08 at 18:42 -0700, Michael Collins wrote:
> FYI,
> 
> The FreeSWITCH devs have added valet parking! Check it out:
> http://www.freeswitch.org/node/207
> 
> Let us know what you think.

I think it's awesome. Love your work as always guys.

hads

-- 
http://nicegear.co.nz
New Zealand's Open Source Hardware Supplier



From msc at freeswitch.org  Thu Oct  8 22:13:37 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Oct 2009 22:13:37 -0700
Subject: [Freeswitch-users] Heartbeat question
In-Reply-To: <86a32abc0910081956k91b3600sb8c556e589f5b089@mail.gmail.com>
References: <86a32abc0910081956k91b3600sb8c556e589f5b089@mail.gmail.com>
Message-ID: <87f2f3b90910082213j1fb95654mf482c1ca8cf4cdf@mail.gmail.com>

On Thu, Oct 8, 2009 at 7:56 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Hi everyone,
>
> I have a question about FreeSWITCH heartbeat, I have this on my dialplan:
>
> <extension name="test">
> <condition field="destination_number" expression="^8084$">
> <action application="answer"/>
> <action application="enable_heartbeat" data="1"/>
> </condition>
> </extension>
>
> But when I do "event plain all" I see the heartbeats are being fired every
> 20 seconds... what I'm doing wrong?
>
>
On the console do you see the call to switch_core_session_enable_heartbeat?
switch_core_session.c:1041:
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "%s
setting session heartbeat to %u second(s).\n",

Confirm that it is actually calling that.

BTW, it would sure be nice if this app were documented on the wiki, hint
hint. ;)
-MC


> Thanks,
>
> Diego
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gabe at gundy.org  Thu Oct  8 22:38:56 2009
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 8 Oct 2009 23:38:56 -0600
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <87f2f3b90910081538l6e4e4b7dv3aedb2c4d1af58bc@mail.gmail.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<87f2f3b90910080958i1f5cfccax621fd12100a05e3e@mail.gmail.com>
	<903da5680910081201p427a8358n768edb61a96674a4@mail.gmail.com>
	<87f2f3b90910081538l6e4e4b7dv3aedb2c4d1af58bc@mail.gmail.com>
Message-ID: <903da5680910082238x27bff476s8bea0e74ab516a8d@mail.gmail.com>

On Thu, Oct 8, 2009 at 4:38 PM, Michael Collins <msc at freeswitch.org> wrote:
> On Thu, Oct 8, 2009 at 12:01 PM, Gabriel Gunderson <gabe at gundy.org> wrote:
>> Naturally the wireless is jacked at the conf. ?It's no match for the
>> 100's of geeks sporting laptops. ?Thanks for the link. Looks like
>> we'll be shooting from the hip. :)
>>
> We experienced something similar at ClueCon, but it was that the hotel's
> DHCP server was a total wuss. :)

So, I just got back from the first day of the conference and there was
plenty of interest in FreeSWITCH from all the Asterisk users.  We had
a good time and answered lots of questions.  It was interesting to
hear what their perceptions about FS were.

Gabe


From diego.viola at gmail.com  Thu Oct  8 22:39:46 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 9 Oct 2009 05:39:46 +0000
Subject: [Freeswitch-users] Heartbeat question
In-Reply-To: <87f2f3b90910082213j1fb95654mf482c1ca8cf4cdf@mail.gmail.com>
References: <86a32abc0910081956k91b3600sb8c556e589f5b089@mail.gmail.com>
	<87f2f3b90910082213j1fb95654mf482c1ca8cf4cdf@mail.gmail.com>
Message-ID: <86a32abc0910082239s68f9df39ocf1548b57f79fe3f@mail.gmail.com>

Yes, I have called to my extension and saw this:

2009-10-09 01:25:44.820677 [INFO] switch_core_session.c:1041 sofia/internal/
1000 at 192.168.0.2 setting session heartbeat to 1 second(s).

But I still see the heartbeat events being fired after 20 seconds...

I have made this script that tells me the seconds when a heartbeat is being
fired...

#!/usr/bin/env ruby

require 'rubygems'
require 'fsr'
require "fsr/listener/inbound"

def custom_channel_heartbeat_handler(event)
  puts Time.now.strftime('%S')
end
FSL::Inbound.add_event_hook(:HEARTBEAT) {|event|
custom_channel_heartbeat_handler(event) }
FSR.start_ies!(FSL::Inbound, :host => "localhost", :port => 8021)


This is the output:

[diego at myhost ~]$ ruby test.rb
No log4r found, falling back to standard ruby library Logger
I, [2009-10-09T01:25:51.797012 #3292]  INFO -- : *** FreeSWITCHer Inbound
EventSocket Listener connected to localhost:8021 ***
I, [2009-10-09T01:25:51.797291 #3292]  INFO -- : ***
http://code.rubyists.com/projects/fs
59
19
39
59
19
39
59


On Fri, Oct 9, 2009 at 5:13 AM, Michael Collins <msc at freeswitch.org> wrote:

>
> On Thu, Oct 8, 2009 at 7:56 PM, Diego Viola <diego.viola at gmail.com> wrote:
>
>> Hi everyone,
>>
>> I have a question about FreeSWITCH heartbeat, I have this on my dialplan:
>>
>> <extension name="test">
>> <condition field="destination_number" expression="^8084$">
>> <action application="answer"/>
>> <action application="enable_heartbeat" data="1"/>
>> </condition>
>> </extension>
>>
>> But when I do "event plain all" I see the heartbeats are being fired every
>> 20 seconds... what I'm doing wrong?
>>
>>
> On the console do you see the call to switch_core_session_enable_heartbeat?
>
> switch_core_session.c:1041:
> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "%s
> setting session heartbeat to %u second(s).\n",
>
> Confirm that it is actually calling that.
>
> BTW, it would sure be nice if this app were documented on the wiki, hint
> hint. ;)
> -MC
>
>
>> Thanks,
>>
>> Diego
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mcampbellsmith at gmail.com  Thu Oct  8 22:43:39 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Fri, 9 Oct 2009 16:43:39 +1100
Subject: [Freeswitch-users] mod_fax compile fails
In-Reply-To: <343C884A-9AB5-4755-9C5F-42ED705D05E7@gmail.com>
References: <33c87fa30910082006v1e91849fjbf4282541e7ac377@mail.gmail.com>
	<343C884A-9AB5-4755-9C5F-42ED705D05E7@gmail.com>
Message-ID: <33c87fa30910082243h6c689816i23368bedfdfd14c0@mail.gmail.com>

Thanks Rob,

Is this a fault in the svn update process?  if so, should/has it been
bug reported?


On Fri, Oct 9, 2009 at 2:39 PM, Rob Forman <rob4manhere at gmail.com> wrote:
> I had that issue too where make current failed on mod_fax (under libs/
> tiff). ?And yeah, it caused a problem where a bunch of modules
> wouldn't load. ?You'll want to get it resolved before installing. ?I
> ended up moving the existing source aside and re-checked out the
> trunk, which compiled fine.
>
>
> On Oct 8, 2009, at 10:06 PM, Mark Campbell-Smith wrote:
>
>> HI all,
>>
>> I just tried to update to the latest svn and I get these errors right
>> at the end after issuing a 'make current'. ?I am using Debian Lenny.
>>
>> making all mod_fax
>> make[5]: Entering directory `/home/mark/freeswitch/src/mod/
>> applications/mod_fax'
>> make[6]: Entering directory `/home/mark/freeswitch/src/mod/
>> applications/mod_fax'
>> make[7]: Entering directory `/home/mark/freeswitch/libs/tiff-3.8.2'
>> running /bin/sh ./configure ?--prefix=/usr/local/freeswitch
>> --cache-file=/dev/null --srcdir=. --disable-shared --with-pic
>> --no-create --no-recursion
>> configure: error: cannot run /bin/sh config/config.sub
>> make[7]: *** [config.status] Error 1
>> make[7]: Leaving directory `/home/mark/freeswitch/libs/tiff-3.8.2'
>> make[6]: *** [../../../../libs/tiff-3.8.2/libtiff/libtiff.la] Error 2
>> make[6]: Leaving directory `/home/mark/freeswitch/src/mod/
>> applications/mod_fax'
>> make[5]: *** [all] Error 1
>> make[5]: Leaving directory `/home/mark/freeswitch/src/mod/
>> applications/mod_fax'
>> make[4]: *** [mod_fax-all] Error 1
>> make[4]: Leaving directory `/home/mark/freeswitch/src/mod'
>> make[3]: *** [all-recursive] Error 1
>> make[3]: Leaving directory `/home/mark/freeswitch/src'
>> Making all in build
>> make[3]: Entering directory `/home/mark/freeswitch/build'
>> +-------- FreeSWITCH Build Complete -----------+
>> + FreeSWITCH has been successfully built. ? ? ?+
>> + Install by running: ? ? ? ? ? ? ? ? ? ? ? ? ?+
>> + ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?+
>> + ? ? ? ? ? ? ? make install ? ? ? ? ? ? ? ? ? +
>> +----------------------------------------------+
>> make[3]: Leaving directory `/home/mark/freeswitch/build'
>> make[2]: *** [all-recursive] Error 1
>> make[2]: Leaving directory `/home/mark/freeswitch'
>> make[1]: *** [all] Error 2
>> make[1]: Leaving directory `/home/mark/freeswitch'
>> make: *** [current] Error 2
>>
>> Also, are the 'Leaving directory / all-recursive' errors going to
>> cause a problem?
>>
>> Thanks!
>> Any ideas what the cause is?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From diego.viola at gmail.com  Thu Oct  8 22:45:47 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 9 Oct 2009 05:45:47 +0000
Subject: [Freeswitch-users] Heartbeat question
In-Reply-To: <6b65470d0910082036g204d5dcbw26e1e43ef80c7c84@mail.gmail.com>
References: <86a32abc0910081956k91b3600sb8c556e589f5b089@mail.gmail.com>
	<86a32abc0910082003v3f5d4e6dua8e7eae18fd2f939@mail.gmail.com>
	<6b65470d0910082036g204d5dcbw26e1e43ef80c7c84@mail.gmail.com>
Message-ID: <86a32abc0910082245j36b8b3e9ncaf1cdd358fce319@mail.gmail.com>

Nope, I was just wondering why it didn't work at 1 second exactly...

On Fri, Oct 9, 2009 at 3:36 AM, William Suffill
<william.suffill at gmail.com>wrote:

> Why do you need it every second? If you want real time channel counts
> you would  be able to track each create/destroy even instead of
> relying on the heartbeat summary.
>
> -- W
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Thu Oct  8 23:08:19 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Oct 2009 23:08:19 -0700
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <903da5680910082238x27bff476s8bea0e74ab516a8d@mail.gmail.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<87f2f3b90910080958i1f5cfccax621fd12100a05e3e@mail.gmail.com>
	<903da5680910081201p427a8358n768edb61a96674a4@mail.gmail.com>
	<87f2f3b90910081538l6e4e4b7dv3aedb2c4d1af58bc@mail.gmail.com>
	<903da5680910082238x27bff476s8bea0e74ab516a8d@mail.gmail.com>
Message-ID: <87f2f3b90910082308i7f6144d9kc887596be54dd609@mail.gmail.com>

On Thu, Oct 8, 2009 at 10:38 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Thu, Oct 8, 2009 at 4:38 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> > On Thu, Oct 8, 2009 at 12:01 PM, Gabriel Gunderson <gabe at gundy.org>
> wrote:
> >> Naturally the wireless is jacked at the conf.  It's no match for the
> >> 100's of geeks sporting laptops.  Thanks for the link. Looks like
> >> we'll be shooting from the hip. :)
> >>
> > We experienced something similar at ClueCon, but it was that the hotel's
> > DHCP server was a total wuss. :)
>
> So, I just got back from the first day of the conference and there was
> plenty of interest in FreeSWITCH from all the Asterisk users.  We had
> a good time and answered lots of questions.  It was interesting to
> hear what their perceptions about FS were.
>

Thanks for reporting back. Please let all the Asterisk users know that they
are welcome to join us in #freeswitch on irc.freenode.net and that they will
not be abused like people do in other less friendly IRC channels.

-MC


>
> Gabe
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jason at jasonjgw.net  Thu Oct  8 23:11:30 2009
From: jason at jasonjgw.net (Jason White)
Date: Fri, 9 Oct 2009 17:11:30 +1100
Subject: [Freeswitch-users] mod_fax compile fails
In-Reply-To: <33c87fa30910082243h6c689816i23368bedfdfd14c0@mail.gmail.com>
References: <33c87fa30910082006v1e91849fjbf4282541e7ac377@mail.gmail.com>
	<343C884A-9AB5-4755-9C5F-42ED705D05E7@gmail.com>
	<33c87fa30910082243h6c689816i23368bedfdfd14c0@mail.gmail.com>
Message-ID: <20091009061130.GA7509@jdc.jasonjgw.net>

Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
> Thanks Rob,
> 
> Is this a fault in the svn update process?  

Usually, no, it's rather that if you build in a directory that you've checked
out and then run svn update, not everything gets cleaned up properly the next
time you build.

I find it's faster to build in a tmpfs file system anyway, e.g.,

mkdir /tmp/fs && sudo mount -t tmpfs tmpfs /tmpfs && svn export . /tmp/fs
then go into /tmp/fs and run the compilation process, or the package building
command, or whatever you need.

This way, my original svn tree is never altered and it will always update
cleanly.



From gabe at gundy.org  Thu Oct  8 23:41:48 2009
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 9 Oct 2009 00:41:48 -0600
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <87f2f3b90910082308i7f6144d9kc887596be54dd609@mail.gmail.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<87f2f3b90910080958i1f5cfccax621fd12100a05e3e@mail.gmail.com>
	<903da5680910081201p427a8358n768edb61a96674a4@mail.gmail.com>
	<87f2f3b90910081538l6e4e4b7dv3aedb2c4d1af58bc@mail.gmail.com>
	<903da5680910082238x27bff476s8bea0e74ab516a8d@mail.gmail.com>
	<87f2f3b90910082308i7f6144d9kc887596be54dd609@mail.gmail.com>
Message-ID: <903da5680910082341r485092vb0cd10c43c4a2534@mail.gmail.com>

On Fri, Oct 9, 2009 at 12:08 AM, Michael Collins <msc at freeswitch.org> wrote:
> Thanks for reporting back. Please let all the Asterisk users know that they
> are welcome to join us in #freeswitch on irc.freenode.net and that they will
> not be abused like people do in other less friendly IRC channels.

Funny you mention this.  Many people report that the way the FS
community refers to Asterisk in docs/wikis/irc/whatever makes the FS
camp seem *less* welcoming to them.  After all, they identify as
Asterisk Users and take the criticism as being kinda harsh.  Most of
them acknowledge the shortcomings of Asterisk but are put off when
someone else points them out.  It's crazy, I know.  The thing is, I
remember thinking that too.  After getting to know FS better, I didn't
notice it as much.  Nobody likes to hear their baby is ugly --even if
they know it is.

At our session, and in general, I've noticed people are more
interested in hearing about FS when you don't make direct comparisons
to Asterisk.  Besides, FS stands on it's own merit.

Just what I've observed *and* my 2 additional cents.

Gabe


From gabe at gundy.org  Thu Oct  8 23:45:53 2009
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 9 Oct 2009 00:45:53 -0600
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910082242570.16063@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910082242570.16063@srv.icf.org.ru>
Message-ID: <903da5680910082345y12c11fa8h65c559ffdbf3df23@mail.gmail.com>

Doesn't anyone trim on this list?

Gabe

P.S.  Sorry about the top post, but I was worried that if I bottom
posted, nobody would find it.


2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>:
> On 2009-10-08 20:32 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:
>
> TC>Hi Yuriy,
> TC>
> TC>can you share what you have so far, I'm sure we can help with RTP part...
>
> I think there is a few days and i make it work, after this i start to test and share it.
>
> TC>
> TC>T.
> TC>
> TC>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
> TC>
> TC>> On 2009-10-08 13:25 -0400, Tuyan ?zipek wrote
> TC>> freeswitch-users at lists.freesw...:
> TC>>
> TC>> Tz>Hi,
> TC>> Tz>
> TC>> Tz>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>:
> TC>> Tz>> On 2009-10-08 10:43 -0500, Anthony Minessale wrote
> TC>> freeswitch-users at lists.f...:
> TC>> Tz>>
> TC>> Tz>> AM>If you are going to make that alternate module are you going to
> TC>> host it in
> TC>> Tz>> AM>the FS tree along side mod_opal?
> TC>> Tz>>
> TC>> Tz>> Yes, but then it be useful, now i have working only signaling part and
> TC>> some
> TC>> Tz>> kinde of not working rtp part :)
> TC>> Tz>
> TC>> Tz>If you dont use fs rtp stack, its unlikely that it will be accepted
> TC>> Tz>into the tree.
> TC>> Tz>
> TC>> Tz>>
> TC>> Tz>> AM>also if were working on mod_opal why did you not try to involve us
> TC>> and the
> TC>> Tz>> AM>opal team?
> TC>> Tz>>
> TC>> Tz>> Because i made patches for libopal, one is a bugfix in rtp part, there
> TC>> is a race condition
> TC>> Tz>> in inicialisation in jitter buffer, another patch implements method to
> TC>> send progress indicator,
> TC>> Tz>> and i don't wont spent my time to incorporate this changes into
> TC>> libopal.
> TC>> Tz>
> TC>> Tz>Thats bad.
> TC>> Tz>Any bugfixes from fs, goes to upstream on any of the used libraries.
> TC>> Tz>You should be doing the same.
> TC>> Tz>And Opal developers, will either include or refuse your patches. If
> TC>> Tz>they refuse it, they will give you the reason.
> TC>>
> TC>> i make this fix only to freeze my current mod_opal work on working state,
> TC>> while it now work for me i work on
> TC>> my new implementation of h323 proto for fs, i think opal developers will
> TC>> fix this rtp bug himself becouse
> TC>> it crashes and make library unuseful.
> TC>>
> TC>> Tz>
> TC>> Tz>> without this changes
> TC>> Tz>> my work on mod_opal in freeswitch don't useful at all, i provide link
> TC>> to my work with all
> TC>> Tz>> patches, if somebody wont incorporate it in libopal tree and fs - go
> TC>> on, but i think
> TC>> Tz>> better and more elegant make new module based on h323plus.
> TC>> Tz>
> TC>> Tz>If you dont publish your changes, all those you are trying to achieve,
> TC>> Tz>wont happen.
> TC>> Tz>
> TC>> Tz>>
> TC>> Tz>> AM>How far away from what is in tree are these patches you have?
> TC>> Tz>> AM>
> TC>> Tz>> AM>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
> TC>> Tz>> AM>
> TC>> Tz>> AM>> On 2009-10-07 15:09 -0500, Brian West wrote
> TC>> Tz>> AM>> freeswitch-users at lists.freeswit...:
> TC>> Tz>> AM>>
> TC>> Tz>> AM>> opal have addition abstraction layer called opalmgr, and it
> TC>> implementation
> TC>> Tz>> AM>> is not so good in
> TC>> Tz>> AM>> this case, for example to implemet pre_answer in mod_opal i need
> TC>> patch
> TC>> Tz>> AM>> libopal, because
> TC>> Tz>
> TC>> Tz>The patch you have in there, adds a method to the OpalCall, ?it does
> TC>> Tz>not touch any parts of OpalManager
> TC>> Tz>so, i dont understand why opalmanager would be the cause of your pain?
> TC>> Tz>
> TC>> Tz>> AM>> there is no way to send progress inicator throuch opalmgr. and
> TC>> there is
> TC>> Tz>> AM>> many another issues like
> TC>> Tz>> AM>> this in that layer.
> TC>> Tz>
> TC>> Tz>Please point me to the issues you have in opal, their bug reports ,
> TC>> traces etc.
> TC>> Tz>I dont think any of the opal people has psychic abilities to detect
> TC>> Tz>-your- problems
> TC>> Tz>and solve them.
> TC>> Tz>
> TC>> Tz>>ftp://srv.icf.org.ru/pub/soft/f/freeswitch/ - there is
> TC>> Tz>> AM>> my work on mod_opal before
> TC>> Tz>> AM>> i start moving to h323plus, may be this help somebody there.
> TC>> Tz>> AM>>
> TC>> Tz>> AM>> BW> From what I have been told h323plus is a based/fork of
> TC>> OpenH323 which
> TC>> Tz>> AM>> BW>OPAL is just a continuation of OpenH323. ?So why not support
> TC>> the
> TC>> Tz>> AM>> BW>developers of OPAL/OpenH323 ?
> TC>> Tz>> AM>> BW>
> TC>> Tz>> AM>> BW>/b
> TC>> Tz>> AM>> BW>
> TC>> Tz>> AM>> BW>On Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote:
> TC>> Tz>> AM>> BW>
> TC>> Tz>> AM>> BW>> We are developing module to handle h323 proto now, we try to
> TC>> use
> TC>> Tz>> AM>> BW>> mod_opal and try improve it, but no luck,
> TC>> Tz>> AM>> BW>> there is many issues in libopal, and finaly we now move to
> TC>> h323plus
> TC>> Tz>> AM>> BW>> library.
> TC>> Tz>
> TC>> Tz>Did any of you try to report those issues?
> TC>> Tz>
> TC>> Tz>Regards
> TC>> Tz>
> TC>> Tz>/tyn
> TC>> Tz>
> TC>> Tz>> AM>> BW>
> TC>> Tz>> AM>> BW>
> TC>> Tz>> AM>> BW>_______________________________________________
> TC>> Tz>> AM>> BW>FreeSWITCH-users mailing list
> TC>> Tz>> AM>> BW>FreeSWITCH-users at lists.freeswitch.org
> TC>> Tz>> AM>> BW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> TC>> Tz>> AM>> BW>UNSUBSCRIBE:
> TC>> Tz>> AM>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> TC>> Tz>> AM>> BW>http://www.freeswitch.org
> TC>> Tz>> AM>> BW>
> TC>> Tz>> AM>>
> TC>> Tz>> AM>> C ????????? ? ? ? ? ? ? ? ? ? ? ? With Best Regards
> TC>> Tz>> AM>> ???????????? ????. ? ? ? ? ? ? ? ?Georgiewskiy Yuriy
> TC>> Tz>> AM>> +7 4872 711666 ? ? ? ? ? ? ? ? ? ?+7 4872 711666
> TC>> Tz>> AM>> ???? +7 4872 711143 ? ? ? ? ? ? ? fax +7 4872 711143
> TC>> Tz>> AM>> ???????? ??? "?? ?? ??????" ? ? ? IT Service Ltd
> TC>> Tz>> AM>> http://nkoort.ru ? ? ? ? ? ? ? ? ?http://nkoort.ru
> TC>> Tz>> AM>> JID: GHhost at jabber.tula-ix.net.ru JID:
> TC>> GHhost at jabber.tula-ix.net.ru
> TC>> Tz>> AM>> YG129-RIPE ? ? ? ? ? ? ? ? ? ? ? ?YG129-RIPE
> TC>> Tz>> AM>>
> TC>> Tz>> AM>> _______________________________________________
> TC>> Tz>> AM>> FreeSWITCH-users mailing list
> TC>> Tz>> AM>> FreeSWITCH-users at lists.freeswitch.org
> TC>> Tz>> AM>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> TC>> Tz>> AM>> UNSUBSCRIBE:
> TC>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> TC>> Tz>> AM>> http://www.freeswitch.org
> TC>> Tz>> AM>>
> TC>> Tz>> AM>>
> TC>> Tz>> AM>
> TC>> Tz>> AM>
> TC>> Tz>> AM>
> TC>> Tz>>
> TC>> Tz>> C ????????? ? ? ? ? ? ? ? ? ? ? ? With Best Regards
> TC>> Tz>> ???????????? ????. ? ? ? ? ? ? ? ?Georgiewskiy Yuriy
> TC>> Tz>> +7 4872 711666 ? ? ? ? ? ? ? ? ? ?+7 4872 711666
> TC>> Tz>> ???? +7 4872 711143 ? ? ? ? ? ? ? fax +7 4872 711143
> TC>> Tz>> ???????? ??? "?? ?? ??????" ? ? ? IT Service Ltd
> TC>> Tz>> http://nkoort.ru ? ? ? ? ? ? ? ? ?http://nkoort.ru
> TC>> Tz>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> TC>> Tz>> YG129-RIPE ? ? ? ? ? ? ? ? ? ? ? ?YG129-RIPE
> TC>> Tz>> _______________________________________________
> TC>> Tz>> FreeSWITCH-users mailing list
> TC>> Tz>> FreeSWITCH-users at lists.freeswitch.org
> TC>> Tz>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> TC>> Tz>> UNSUBSCRIBE:
> TC>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> TC>> Tz>> http://www.freeswitch.org
> TC>> Tz>>
> TC>> Tz>>
> TC>> Tz>
> TC>> Tz>_______________________________________________
> TC>> Tz>FreeSWITCH-users mailing list
> TC>> Tz>FreeSWITCH-users at lists.freeswitch.org
> TC>> Tz>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> TC>> Tz>UNSUBSCRIBE:
> TC>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> TC>> Tz>http://www.freeswitch.org
> TC>> Tz>
> TC>>
> TC>> C ????????? ? ? ? ? ? ? ? ? ? ? ? With Best Regards
> TC>> ???????????? ????. ? ? ? ? ? ? ? ?Georgiewskiy Yuriy
> TC>> +7 4872 711666 ? ? ? ? ? ? ? ? ? ?+7 4872 711666
> TC>> ???? +7 4872 711143 ? ? ? ? ? ? ? fax +7 4872 711143
> TC>> ???????? ??? "?? ?? ??????" ? ? ? IT Service Ltd
> TC>> http://nkoort.ru ? ? ? ? ? ? ? ? ?http://nkoort.ru
> TC>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> TC>> YG129-RIPE ? ? ? ? ? ? ? ? ? ? ? ?YG129-RIPE
> TC>>
> TC>> _______________________________________________
> TC>> FreeSWITCH-users mailing list
> TC>> FreeSWITCH-users at lists.freeswitch.org
> TC>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> TC>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> TC>> http://www.freeswitch.org
> TC>>
> TC>>
> TC>
>
> C ????????? ? ? ? ? ? ? ? ? ? ? ? With Best Regards
> ???????????? ????. ? ? ? ? ? ? ? ?Georgiewskiy Yuriy
> +7 4872 711666 ? ? ? ? ? ? ? ? ? ?+7 4872 711666
> ???? +7 4872 711143 ? ? ? ? ? ? ? fax +7 4872 711143
> ???????? ??? "?? ?? ??????" ? ? ? IT Service Ltd
> http://nkoort.ru ? ? ? ? ? ? ? ? ?http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE ? ? ? ? ? ? ? ? ? ? ? ?YG129-RIPE
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Fri Oct  9 00:05:39 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Oct 2009 00:05:39 -0700
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <903da5680910082341r485092vb0cd10c43c4a2534@mail.gmail.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<87f2f3b90910080958i1f5cfccax621fd12100a05e3e@mail.gmail.com>
	<903da5680910081201p427a8358n768edb61a96674a4@mail.gmail.com>
	<87f2f3b90910081538l6e4e4b7dv3aedb2c4d1af58bc@mail.gmail.com>
	<903da5680910082238x27bff476s8bea0e74ab516a8d@mail.gmail.com>
	<87f2f3b90910082308i7f6144d9kc887596be54dd609@mail.gmail.com>
	<903da5680910082341r485092vb0cd10c43c4a2534@mail.gmail.com>
Message-ID: <87f2f3b90910090005i51dbfc60t20dd77789872bc9e@mail.gmail.com>

On Thu, Oct 8, 2009 at 11:41 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Fri, Oct 9, 2009 at 12:08 AM, Michael Collins <msc at freeswitch.org>
> wrote:
> > Thanks for reporting back. Please let all the Asterisk users know that
> they
> > are welcome to join us in #freeswitch on irc.freenode.net and that they
> will
> > not be abused like people do in other less friendly IRC channels.
>
> Funny you mention this.  Many people report that the way the FS
> community refers to Asterisk in docs/wikis/irc/whatever makes the FS
> camp seem *less* welcoming to them.  After all, they identify as
> Asterisk Users and take the criticism as being kinda harsh.  Most of
> them acknowledge the shortcomings of Asterisk but are put off when
> someone else points them out.  It's crazy, I know.  The thing is, I
> remember thinking that too.  After getting to know FS better, I didn't
> notice it as much.  Nobody likes to hear their baby is ugly --even if
> they know it is.
>

Interesting. I'd like to know what specifically turned them off. It could be
node 117, but that was written by Anthony in response to dozens of questions
about the subject. In any case, if the Asterisk guys look at the first
edition of the Starfish/TFOT book they'll see glowing praise of Anthony
Minessale and Brian West, so maybe they'll be willing to listen to those
guys since they've got so much Asterisk knowledge. The other thing is that
many people in the FreeSWITCH community are Asterisk refugees. They left
Asterisk for FreeSWITCH because they had some sort of trouble with Asterisk.
If that comes across in the docs then we'll have to see about cleaning that
up.

>
> At our session, and in general, I've noticed people are more
> interested in hearing about FS when you don't make direct comparisons
> to Asterisk.  Besides, FS stands on it's own merit.
>

Agreed. Direct comparisons can always be slanted in any direction. Besides,
FreeSWITCH stacks up very well against a lot of modern (i.e. expensive)
equipment put out by the big boys (Cisco, Avaya, NEC, ShoreTel, etc.).

>
> Just what I've observed *and* my 2 additional cents.
>
> Gabe
>
Thanks,
MC
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From mcampbellsmith at gmail.com  Fri Oct  9 03:38:58 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Fri, 9 Oct 2009 21:38:58 +1100
Subject: [Freeswitch-users] [ERR] mod_dingaling.c:980 Stun Failed!
In-Reply-To: <b16156850910020300x6cd7dd9cm2348942b04c79030@mail.gmail.com>
References: <33c87fa30909290415y178b6506x120a6e5c39839ce0@mail.gmail.com>
	<33c87fa30910020158q7589e18fj53bd4cc95ff3926f@mail.gmail.com>
	<b16156850910020300x6cd7dd9cm2348942b04c79030@mail.gmail.com>
Message-ID: <33c87fa30910090338w7d81b8dehc2f47059e1bf32f@mail.gmail.com>

It was the svn revision I was using... I have updated now and it is
working again.

On Fri, Oct 2, 2009 at 9:00 PM, Muhammad Shahzad
<shaheryarkh at googlemail.com> wrote:
> Yes, i had same problem, then i changed stun server to one of our own
> servers. You may try some of public stun servers listed on below link,
>
> http://www.voip-info.org/wiki/view/STUN
>
> Thank you.
>
>
> On Fri, Oct 2, 2009 at 2:58 PM, Mark Campbell-Smith
> <mcampbellsmith at gmail.com> wrote:
>>
>> Anyone have this issue?
>>
>> On Tue, Sep 29, 2009 at 9:15 PM, Mark Campbell-Smith
>> <mcampbellsmith at gmail.com> wrote:
>> > Hi!
>> >
>> > I have just started to use dingaling again, and noticed I constantly
>> > get a stun error.
>> >
>> > 2009-09-29 21:11:03.175002 [ERR] mod_dingaling.c:980 Stun Failed!
>> > stun.fwdnet.net:3478 [Remote Address Error!]
>> >
>> > I have tried with stun.freeswitch.org and stun.fwdnet.net stun servers
>> > and keep getting this error with dingaling. ?I have no problems with
>> > inbound sip calls, so I don't think ?its the actual stun server.
>> >
>> > Has anyone else seen this? ?I am using: FreeSWITCH Version 1.0.trunk
>> > (14952)
>> >
>> > Thanks!
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> ________________________________________________________
> |
> ? ? ? ? ? ? ? ? ? ? ?|
> | FATAL ERROR ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ---
> O X |
> |_______________________________________________________|
> | ? ? ? ? ? ? ? ? ? ? ? ?You have moved the mouse.
> ?|
> | Windows must be restarted for the changes to take effect. ? |
> | ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?<OK>
> ? ? ? ? ? ? ? ?|
> ####################################/
>
>
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From lakindia89 at gmail.com  Fri Oct  9 04:07:35 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Fri, 9 Oct 2009 16:37:35 +0530
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <87f2f3b90910080954l1ecb867dn8a025f5f0f701eb4@mail.gmail.com>
References: <7d79b3930910070334g53cef0c5m63037e264f1a9233@mail.gmail.com>
	<87f2f3b90910070849t79b7cb90x4150bf1dd8501ca4@mail.gmail.com>
	<7d79b3930910080241q4eeb374an2ec34642bf450818@mail.gmail.com>
	<c4d05cbe0910080802t7ebf8007m333cbd6e74efd286@mail.gmail.com>
	<87f2f3b90910080954l1ecb867dn8a025f5f0f701eb4@mail.gmail.com>
Message-ID: <7d79b3930910090407m5dc81d12pf2b0a47f6017804@mail.gmail.com>

Thanks all u guys. It got worked when I replaced with the following
configuration and the incoming has no problem with that.
[span zt PRI_1]
trunk_type => e1
b-channel => 1-15
d-channel => 16
b-channel => 17-31

But still I'm facing problem with the outgoing call. It says
INVALID_IE_CONTENTS.
What might be the issue? Even I tried the following dialplan to call by
using bridge.
<extension name="outgoing-fxo">
                  <condition field="destination_number"
expression="^(\d{10})$">
                          <action application="answer"/>
                          <action application="bridge"
data="openzap/1/1/${dialed_ext}"/>
                  </condition>
          </extension>
But it prints the same error.
What might be the issue?


On Thu, Oct 8, 2009 at 10:24 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Thu, Oct 8, 2009 at 8:02 AM, Moises Silva <moises.silva at gmail.com>wrote:
>
>> On Thu, Oct 8, 2009 at 5:41 AM, lakshmanan ganapathy <
>> lakindia89 at gmail.com> wrote:
>>
>>> Hi I tried with the following openzap.conf
>>> [span zt PRI_1]
>>> trunk_type => e1
>>> b-channel => 1:1-15,17-31
>>> d-channel => 1:16
>>>
>>
>> This does not look like a healthy config to me. You are using
>> <span>:<channels> notation in zaptel spans, zaptel channels increment across
>> spans, read point number 5 of this web page
>> http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2 , this is for another type
>> of signaling, but the span/channel numbering concepts are the same.
>>
>>
> Moy,
>
> As usual you are right on the money. I'm so used to doing Sangoma configs
> that I forget about the zaptel syntax. I recommend this config:
>
> [span zt PRI_1]
> trunk_type => e1
> b-channel => 1-15
> d-channel => 16
> b-channel => 17-31
>
>
> Give that a try and report back if you have issues.
> -MC
>
>>  --
>> Moises Silva
>> Software Developer
>> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
>> 9T3 Canada
>> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From srinivas.ksvreddy at gmail.com  Fri Oct  9 04:58:28 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Fri, 9 Oct 2009 17:28:28 +0530
Subject: [Freeswitch-users] apr_queue
Message-ID: <f8af5740910090458w6540db9fg972ad1ba8a691c30@mail.gmail.com>

Hi all,

does any know about How apr_queue is maintaing and retriving all registered
and all stuff....


-- 
Srinivasula Reddy K
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From lakindia89 at gmail.com  Fri Oct  9 04:59:50 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Fri, 9 Oct 2009 17:29:50 +0530
Subject: [Freeswitch-users] libpri_span vs prispans
Message-ID: <7d79b3930910090459v2b07e3aapd715c7618f11ada0@mail.gmail.com>

Hi all,

What difference it will make, if I use pri_span configuration and lib_pri
span configuration in openzap.conf.xml.
I'm preety much confused on the difference between this.
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From rob4manhere at gmail.com  Fri Oct  9 05:22:35 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Fri, 9 Oct 2009 07:22:35 -0500
Subject: [Freeswitch-users] mod_fax compile fails
In-Reply-To: <20091009061130.GA7509@jdc.jasonjgw.net>
References: <33c87fa30910082006v1e91849fjbf4282541e7ac377@mail.gmail.com>
	<343C884A-9AB5-4755-9C5F-42ED705D05E7@gmail.com>
	<33c87fa30910082243h6c689816i23368bedfdfd14c0@mail.gmail.com>
	<20091009061130.GA7509@jdc.jasonjgw.net>
Message-ID: <826997E7-F76D-440C-BF70-ECE21B9F1E63@gmail.com>

I was in a production window so when the latest trunk worked I moved  
on.  I went back to troubleshoot later when I saw your email but I  
couldn't reproduce it.

I like the tmpfs build- thanks for the tip Jason.


On Oct 9, 2009, at 1:11 AM, Jason White wrote:

> Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
>> Thanks Rob,
>>
>> Is this a fault in the svn update process?
>
> Usually, no, it's rather that if you build in a directory that  
> you've checked
> out and then run svn update, not everything gets cleaned up properly  
> the next
> time you build.
>
> I find it's faster to build in a tmpfs file system anyway, e.g.,
>
> mkdir /tmp/fs && sudo mount -t tmpfs tmpfs /tmpfs && svn export . / 
> tmp/fs
> then go into /tmp/fs and run the compilation process, or the package  
> building
> command, or whatever you need.
>
> This way, my original svn tree is never altered and it will always  
> update
> cleanly.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From Russell.Mosemann at cune.org  Fri Oct  9 05:36:15 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Fri, 9 Oct 2009 12:36:15 -0000
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <7d79b3930910090407m5dc81d12pf2b0a47f6017804@mail.gmail.com>
Message-ID: <20091009123615.BC50A3FA6E9@mail.cune.org>

lakshmanan ganapathy <lakindia89 at gmail.com> said:

> But still I'm facing problem with the outgoing call. It says
> INVALID_IE_CONTENTS.
> What might be the issue? Even I tried the following dialplan to call by
> using bridge.
> <extension name="outgoing-fxo">
>                   <condition field="destination_number"
> expression="^(\d{10})$">
>                           <action application="answer"/>
>                           <action application="bridge"
> data="openzap/1/1/${dialed_ext}"/>

Does "answer" need to be called here? I haven't used an fxo. So, I don't
know. What value does $dialed_ext have? If you want to use the number
matched in the condition, then it should be

openzap/1/1/$1

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From dujinfang at gmail.com  Fri Oct  9 05:43:45 2009
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 9 Oct 2009 20:43:45 +0800
Subject: [Freeswitch-users] FreeSWITCH Update: Valet Parking
In-Reply-To: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>
References: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>
Message-ID: <23f91030910090543o1f70c997t7d860d4d65b2a175@mail.gmail.com>

It's very cool. But think about the following scenario:

Alice answers a call from Bob.
Alice transfers Bob to 6001 and hangs up.
Bob is now in parking stall 6001, hearing MOH.
Alice calls Charlie and tells him that he has a call parked in 6001.

  Dee call in and answered by Alice  # at that time, Alice don't know the
lot is empty or not
  If Alice transfer Dee to 6001 # then Dee will talk to Bob, that's not
expected

Charlie calls 6000, waits for prompt, then dials 6001#.
Charlie Listening to music.............

Event it's rarely happen, I do think Alice would like to now the if lot is
empty or not before transfer. Make sense?

2009/10/9 Michael Collins <msc at freeswitch.org>

> FYI,
>
> The FreeSWITCH devs have added valet parking! Check it out:
> http://www.freeswitch.org/node/207
>
> Let us know what you think.
>
> Thanks,
> Michael
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Russell.Mosemann at cune.org  Fri Oct  9 05:57:14 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Fri, 9 Oct 2009 12:57:14 -0000
Subject: [Freeswitch-users] libpri_span vs prispans
In-Reply-To: <7d79b3930910090459v2b07e3aapd715c7618f11ada0@mail.gmail.com>
Message-ID: <20091009125714.E0FF73043D7@mail.cune.org>

lakshmanan ganapathy <lakindia89 at gmail.com> said:
> What difference it will make, if I use pri_span configuration and lib_pri
> span configuration in openzap.conf.xml.
> I'm preety much confused on the difference between this.

Openzap can handle a T1/E1 itself, or openzap can use libpri to do that.
libpri is more mature than openzap. In my case openzap was not able to
handle everything invovling a T1 here, and using libpri fixed it.

A libpri_span configuration is handled by libpri. A pri_span
configuration is handled natively by openzap. Use the one that works for you.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From rupa at rupa.com  Fri Oct  9 06:24:39 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 9 Oct 2009 07:24:39 -0600
Subject: [Freeswitch-users] FreeSWITCH Update: Valet Parking
In-Reply-To: <23f91030910090543o1f70c997t7d860d4d65b2a175@mail.gmail.com>
References: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>
	<23f91030910090543o1f70c997t7d860d4d65b2a175@mail.gmail.com>
Message-ID: <b39273530910090624l5aa8a507s2be37f833e19a5a1@mail.gmail.com>

Perhaps a "put in next empty slot" mode and then a say of the slot so
Alice and make a note of the lot #?

On Fri, Oct 9, 2009 at 6:43 AM, Seven Du <dujinfang at gmail.com> wrote:
> It's very cool. But think about the following scenario:
>
> Alice answers a call from Bob.
> Alice transfers Bob to 6001 and hangs up.
> Bob is now in parking stall 6001, hearing MOH.
> Alice calls Charlie and tells him that he has a call parked in 6001.
>
> ? Dee call in and answered by Alice? # at that time, Alice don't know the
> lot is empty or not
> ? If Alice transfer Dee to 6001 # then Dee will talk to Bob, that's not
> expected
>
> Charlie calls 6000, waits for prompt, then dials 6001#.
> Charlie Listening to music.............
>
> Event it's rarely happen, I do think Alice would like to now the if lot is
> empty or not before transfer. Make sense?
>
> 2009/10/9 Michael Collins <msc at freeswitch.org>
>>
>> FYI,
>>
>> The FreeSWITCH devs have added valet parking! Check it out:
>> http://www.freeswitch.org/node/207
>>
>> Let us know what you think.
>>
>> Thanks,
>> Michael
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa


From andy at fabulous4.co.uk  Fri Oct  9 07:12:51 2009
From: andy at fabulous4.co.uk (Andy)
Date: Fri, 9 Oct 2009 15:12:51 +0100
Subject: [Freeswitch-users] NAT problems - sorry
In-Reply-To: <0A2A7A27-611E-40FA-8268-D2C451AA8B77@freeswitch.org>
References: <5BE289D2AE22492BA71A6919C8A440B3@D810><5A70A0D4-70FA-43B2-BAF5-F8D6ECC64838@freeswitch.org><60E56B6B355C45BCA0D31F4425B9173A@D810><6B2B7FFD-75BC-44CE-A11C-2B26447DA889@freeswitch.org><703EEB9AC9AD43898C6C46F82904E6A9@D810>
	<0A2A7A27-611E-40FA-8268-D2C451AA8B77@freeswitch.org>
Message-ID: <9B8EDDA782464750A4AAC1F947B56A4C@D810>

Hi folks,
 
Just thought I'd post the solution to this problem. In my case the problem
was that my config was based on an older version and so was missing a
crucial parameter that exists in the default config wirth a new install.
 
The external.xml profile needed to contain the parameter:
 
<param name="local-network-acl" value="localnet.auto"/>
 
With this in place, DTMF and hangup messages traverse the nat firewall
correctly. Without it they don't. I searched the Wiki and couldn't find any
info on this parameter. Can anyone provide a description of what it does and
why it's significant that can be added to the WIKI?
 
Cheers
Andy

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: 07 October 2009 19:07
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] NAT problems - sorry


s/auto-nat/$realip/ then forward the rtp ports and sip ports. 

/b
PS chances are you have to ENABLE upnp.

On Oct 7, 2009, at 12:58 PM, Andy wrote:


Many thanks Brian, the firewall docs assure me it is uPnp but is probably
lying or a poor implementation. Could you point me to the right section of
the Wiki to tell me how to do this manually as I've been scouting for some
time and can;t seem to find the right thing. sorry if I'm being blind.



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From anthony.minessale at gmail.com  Fri Oct  9 07:25:33 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 9 Oct 2009 09:25:33 -0500
Subject: [Freeswitch-users] FreeSWITCH Update: Valet Parking
In-Reply-To: <23f91030910090543o1f70c997t7d860d4d65b2a175@mail.gmail.com>
References: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>
	<23f91030910090543o1f70c997t7d860d4d65b2a175@mail.gmail.com>
Message-ID: <191c3a030910090725o4150a17x56b93303bf0b8c2b@mail.gmail.com>

or twist that into a feature... =D

alice is bob's secretary so she calls him on his cell when he has an appt
and parks him.
Then she calls the other guy and says "I have bob on line for you" and xfers
him to the same parking space =p


On Fri, Oct 9, 2009 at 7:43 AM, Seven Du <dujinfang at gmail.com> wrote:

> It's very cool. But think about the following scenario:
>
> Alice answers a call from Bob.
> Alice transfers Bob to 6001 and hangs up.
> Bob is now in parking stall 6001, hearing MOH.
> Alice calls Charlie and tells him that he has a call parked in 6001.
>
>   Dee call in and answered by Alice  # at that time, Alice don't know the
> lot is empty or not
>   If Alice transfer Dee to 6001 # then Dee will talk to Bob, that's not
> expected
>
> Charlie calls 6000, waits for prompt, then dials 6001#.
> Charlie Listening to music.............
>
> Event it's rarely happen, I do think Alice would like to now the if lot is
> empty or not before transfer. Make sense?
>
> 2009/10/9 Michael Collins <msc at freeswitch.org>
>
>> FYI,
>>
>> The FreeSWITCH devs have added valet parking! Check it out:
>> http://www.freeswitch.org/node/207
>>
>> Let us know what you think.
>>
>> Thanks,
>> Michael
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From maciej.aniserowicz at gmail.com  Fri Oct  9 07:40:32 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Fri, 9 Oct 2009 07:40:32 -0700 (PDT)
Subject: [Freeswitch-users] gateway FS informs it's client FS about
 users hanged up with a long delay
In-Reply-To: <87f2f3b90910081000v5b505f4ao53104211b620ea77@mail.gmail.com>
References: <1A16501E57CA4727A593226A8C810308@procent>
	<98F72CCB-88B9-47ED-AED3-B6BA6DE648C0@jerris.com>
	<1255007884165-3787956.post@n2.nabble.com>
	<87f2f3b90910081000v5b505f4ao53104211b620ea77@mail.gmail.com>
Message-ID: <1255099232783-3795011.post@n2.nabble.com>


Hello,
The issue is resolved. I feel stupid, because Michael Jerris was right the first time. Setting external_rtp_ip and external_sip_ip to $${local_ip_v4} made it work.
But the strange thing is: it SOMETIMES worked before without any delay, which 'should not be possible', because the original IP was my external ip and the BYE message was sent straight to it. And there is no way it could reach the target 'internal' FS, because it runs on virtual machine, and no ports are forwarded on my router.
Any thoughts? Why this could (rarely) work even with the previous config?

Thanks to both of you for your answers.

MA


  ----- Original Message ----- 
  From: mercutioviz (via Nabble) 
  To: Maciej Aniserowicz 
  Sent: Thursday, October 08, 2009 7:06 PM
  Subject: Re: [Freeswitch-users] gateway FS informs it's client FS about users hanged up with a long delay





  On Thu, Oct 8, 2009 at 6:18 AM, Maciej Aniserowicz <[hidden email]> wrote:


    Both of the instances are run on the same machine, i just changed the default
    ports they use. Can anything else cause this strange behavior?
    MA

  Did a packet capture yield any clues? That is, were you able to confirm that each instance sent and received all the packets that you believe they should have sent and received? The reason I ask is so that you don't end up chasing a ghost because you made an assumption somewhere in your troubleshooting.

  -MC
  ? 




    Michael Jerris wrote:
    >
    > Incorrect NAT configuration so one of the boxes is not actually
    > getting a BYE.
    >
    >
    > On Oct 5, 2009, at 3:13 AM, Maciej Aniserowicz wrote:
    >
    >> Hi,
    >> When I use two FreeSWITCH instances ('internal' and 'external'), all
    >> users register to the 'external' instance which acts as a gateway by
    >> 'internal' instance (which in turn is controlled by my applicaiton
    >> with commands sent by socket).
    >> When user hangs up, the 'hanged up' event is propagated to the
    >> 'internal' instance after a long time (~3 minutes) instead of being
    >> propagated immediately.
    >> What can cause this issue?
    >
    >
    > _______________________________________________
    > FreeSWITCH-users mailing list
    > [hidden email]
    > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    > http://www.freeswitch.org
    >
    >

    --
    View this message in context: http://n2.nabble.com/gateway-FS-informs-it-s-client-FS-about-users-hanged-up-with-a-long-delay-tp3768540p3787956.html
    Sent from the freeswitch-users mailing list archive at Nabble.com.

    _______________________________________________
    FreeSWITCH-users mailing list
    [hidden email]
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org



  _______________________________________________ 
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  [hidden email] 
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org



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  To unsubscribe from Re: gateway FS informs it's client FS about users hanged up with a long delay, click here. 


-- 
View this message in context: http://n2.nabble.com/gateway-FS-informs-it-s-client-FS-about-users-hanged-up-with-a-long-delay-tp3768540p3795011.html
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From rupa at rupa.com  Fri Oct  9 07:50:33 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 9 Oct 2009 08:50:33 -0600
Subject: [Freeswitch-users] FreeSWITCH Update: Valet Parking
In-Reply-To: <191c3a030910090725o4150a17x56b93303bf0b8c2b@mail.gmail.com>
References: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>
	<23f91030910090543o1f70c997t7d860d4d65b2a175@mail.gmail.com>
	<191c3a030910090725o4150a17x56b93303bf0b8c2b@mail.gmail.com>
Message-ID: <b39273530910090750ga9debecp783d374560e8c577@mail.gmail.com>

Oh, definitely.  I think both use cases (existing, and park in next
avail slot and read back slot #) are useful.

On Fri, Oct 9, 2009 at 8:25 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> or twist that into a feature... =D
>
> alice is bob's secretary so she calls him on his cell when he has an appt
> and parks him.
> Then she calls the other guy and says "I have bob on line for you" and xfers
> him to the same parking space =p



-- 
-Rupa


From leo.zibi at gmail.com  Fri Oct  9 00:35:23 2009
From: leo.zibi at gmail.com (leo.zibi at gmail.com)
Date: Fri, 09 Oct 2009 09:35:23 +0200
Subject: [Freeswitch-users] mod_fax compile fails
In-Reply-To: <33c87fa30910082006v1e91849fjbf4282541e7ac377@mail.gmail.com>
References: <33c87fa30910082006v1e91849fjbf4282541e7ac377@mail.gmail.com>
Message-ID: <4ACEE7BB.1040606@gmail.com>

Hi,

./bootstrap.sh
./configure
make
make install

--
Regards,
Leo



Mark Campbell-Smith wrote:
> HI all,
>
> I just tried to update to the latest svn and I get these errors right
> at the end after issuing a 'make current'.  I am using Debian Lenny.
>
> making all mod_fax
> make[5]: Entering directory `/home/mark/freeswitch/src/mod/applications/mod_fax'
> make[6]: Entering directory `/home/mark/freeswitch/src/mod/applications/mod_fax'
> make[7]: Entering directory `/home/mark/freeswitch/libs/tiff-3.8.2'
> running /bin/sh ./configure  --prefix=/usr/local/freeswitch
> --cache-file=/dev/null --srcdir=. --disable-shared --with-pic
> --no-create --no-recursion
> configure: error: cannot run /bin/sh config/config.sub
> make[7]: *** [config.status] Error 1
> make[7]: Leaving directory `/home/mark/freeswitch/libs/tiff-3.8.2'
> make[6]: *** [../../../../libs/tiff-3.8.2/libtiff/libtiff.la] Error 2
> make[6]: Leaving directory `/home/mark/freeswitch/src/mod/applications/mod_fax'
> make[5]: *** [all] Error 1
> make[5]: Leaving directory `/home/mark/freeswitch/src/mod/applications/mod_fax'
> make[4]: *** [mod_fax-all] Error 1
> make[4]: Leaving directory `/home/mark/freeswitch/src/mod'
> make[3]: *** [all-recursive] Error 1
> make[3]: Leaving directory `/home/mark/freeswitch/src'
> Making all in build
> make[3]: Entering directory `/home/mark/freeswitch/build'
>  +-------- FreeSWITCH Build Complete -----------+
>  + FreeSWITCH has been successfully built.      +
>  + Install by running:                          +
>  +                                              +
>  +               make install                   +
>  +----------------------------------------------+
> make[3]: Leaving directory `/home/mark/freeswitch/build'
> make[2]: *** [all-recursive] Error 1
> make[2]: Leaving directory `/home/mark/freeswitch'
> make[1]: *** [all] Error 2
> make[1]: Leaving directory `/home/mark/freeswitch'
> make: *** [current] Error 2
>
> Also, are the 'Leaving directory / all-recursive' errors going to
> cause a problem?
>
> Thanks!
> Any ideas what the cause is?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   



From jerry.richards at teotech.com  Fri Oct  9 08:23:27 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 9 Oct 2009 08:23:27 -0700
Subject: [Freeswitch-users] SLAs and BLAs
References: <5B5F50E0B1D34BC0BCBCC05E1FE05C8A@greyhawk.tonecommander.com>
	<3F7AC380-CACB-432F-8B94-80FBA18628C0@freeswitch.org> 
Message-ID: <5CC2E91025314615ACBB65E029D5CB12@greyhawk.tonecommander.com>


I gather from the mailing archive that BLAs are implemented using the
draft-anil-sipping-bla-04.txt document.  According to the draft, the
Appearance Agent is supposed to initiate a SUBSCRIBE request, but I don't
see FS doing this.

What phone types/models are known to work with the FS BLA implementation?

Best Regards,
Jerry
 

-----Original Message-----
From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Monday, October 05, 2009 3:24 PM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: [Freeswitch-users] SLAs and BLAs

We are building our own in-house developed Teo phones.  I also have
CounterPath's Bria Professional phone.  For test purposes, I have one snom
phone and a couple Polycomm phones.

Jerry
 

-----Original Message-----
From: Brian West [mailto:brian at freeswitch.org]
Sent: Monday, October 05, 2009 11:02 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] SLAs and BLAs

First off what phones are you going to be using?

/b

On Oct 5, 2009, at 12:58 PM, Jerry Richards wrote:

>
> I can see how BLFs and Presence are managed, however I haven't found 
> much documentation on SLAs and BLAs.  What is the RFC(s) that 
> Freeswitch used to implement SLAs and BLAs?  Do they differ from BLFs?
>
> Best Regards,
> Jerry






From anthony.minessale at gmail.com  Fri Oct  9 09:02:24 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 9 Oct 2009 11:02:24 -0500
Subject: [Freeswitch-users] Heartbeat question
In-Reply-To: <86a32abc0910082245j36b8b3e9ncaf1cdd358fce319@mail.gmail.com>
References: <86a32abc0910081956k91b3600sb8c556e589f5b089@mail.gmail.com>
	<86a32abc0910082003v3f5d4e6dua8e7eae18fd2f939@mail.gmail.com>
	<6b65470d0910082036g204d5dcbw26e1e43ef80c7c84@mail.gmail.com>
	<86a32abc0910082245j36b8b3e9ncaf1cdd358fce319@mail.gmail.com>
Message-ID: <191c3a030910090902t38282f72l6b6d1733a758836@mail.gmail.com>

Update to trunk and try it with fs_cli it for sure will let you do every 1
second

in fs_cli type

/events plain all

if you make that call you will see one every 1 second


On Fri, Oct 9, 2009 at 12:45 AM, Diego Viola <diego.viola at gmail.com> wrote:

> Nope, I was just wondering why it didn't work at 1 second exactly...
>
>
> On Fri, Oct 9, 2009 at 3:36 AM, William Suffill <william.suffill at gmail.com
> > wrote:
>
>> Why do you need it every second? If you want real time channel counts
>> you would  be able to track each create/destroy even instead of
>> relying on the heartbeat summary.
>>
>> -- W
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Fri Oct  9 09:07:56 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Oct 2009 09:07:56 -0700
Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Starting,
	Please Join Us!
Message-ID: <87f2f3b90910090907v6a01bae2rd74f577d482dddf6@mail.gmail.com>

Agenda and call-in info: http://bit.ly/lzEYy
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From mike at jerris.com  Fri Oct  9 10:25:40 2009
From: mike at jerris.com (Michael Jerris)
Date: Fri, 9 Oct 2009 13:25:40 -0400
Subject: [Freeswitch-users] apr_queue
In-Reply-To: <f8af5740910090458w6540db9fg972ad1ba8a691c30@mail.gmail.com>
References: <f8af5740910090458w6540db9fg972ad1ba8a691c30@mail.gmail.com>
Message-ID: <91049B87-E571-4181-B436-6DAD065A6516@jerris.com>

On Oct 9, 2009, at 7:58 AM, srinivasula reddy wrote:

> Hi all,
>
> does any know about How apr_queue is maintaing and retriving all  
> registered and all stuff....
>

parse error



From msc at freeswitch.org  Fri Oct  9 10:57:04 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Oct 2009 10:57:04 -0700
Subject: [Freeswitch-users] libpri_span vs prispans
In-Reply-To: <20091009125714.E0FF73043D7@mail.cune.org>
References: <7d79b3930910090459v2b07e3aapd715c7618f11ada0@mail.gmail.com>
	<20091009125714.E0FF73043D7@mail.cune.org>
Message-ID: <87f2f3b90910091057q3a033548j462446cf424ad257@mail.gmail.com>

On Fri, Oct 9, 2009 at 5:57 AM, <Russell.Mosemann at cune.org> wrote:

> lakshmanan ganapathy <lakindia89 at gmail.com> said:
> > What difference it will make, if I use pri_span configuration and lib_pri
> > span configuration in openzap.conf.xml.
> > I'm preety much confused on the difference between this.
>
> Openzap can handle a T1/E1 itself, or openzap can use libpri to do that.
> libpri is more mature than openzap. In my case openzap was not able to
> handle everything invovling a T1 here, and using libpri fixed it.
>
> A libpri_span configuration is handled by libpri. A pri_span
> configuration is handled natively by openzap. Use the one that works for
> you.
>

Right now the libpri stack is more mature than the ozmod_isdn stuff. See the
OpenZAP page on the wiki for instructions on how to set up ozmod_libpri.
-MC
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From msc at freeswitch.org  Fri Oct  9 11:01:34 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Oct 2009 11:01:34 -0700
Subject: [Freeswitch-users] FreeSWITCH Update: Valet Parking
In-Reply-To: <b39273530910090750ga9debecp783d374560e8c577@mail.gmail.com>
References: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>
	<23f91030910090543o1f70c997t7d860d4d65b2a175@mail.gmail.com>
	<191c3a030910090725o4150a17x56b93303bf0b8c2b@mail.gmail.com>
	<b39273530910090750ga9debecp783d374560e8c577@mail.gmail.com>
Message-ID: <87f2f3b90910091101x33bfbd8vf504b8bb3186da08@mail.gmail.com>

On Fri, Oct 9, 2009 at 7:50 AM, Rupa Schomaker <rupa at rupa.com> wrote:

> Oh, definitely.  I think both use cases (existing, and park in next
> avail slot and read back slot #) are useful.
>
>
They do have some usefulness. The most important reason for this feature,
though, is for the person who handles calls all day and moves them around.
That person will most likely have a pretty good idea of which parking stalls
are available. However, I do like the idea of a webby interface showing what
stalls are in use...
-MC


> On Fri, Oct 9, 2009 at 8:25 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > or twist that into a feature... =D
> >
> > alice is bob's secretary so she calls him on his cell when he has an appt
> > and parks him.
> > Then she calls the other guy and says "I have bob on line for you" and
> xfers
> > him to the same parking space =p
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kristian.kielhofner at gmail.com  Fri Oct  9 12:10:36 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 9 Oct 2009 15:10:36 -0400
Subject: [Freeswitch-users] On the handling of SIP headers
Message-ID: <2d9149cd0910091210s33fb2bb9oc0c94a64aa87e0be@mail.gmail.com>

Hello everyone,

  In using FS for various scenarios I've noticed some behavior that
I'm not sure is completely "proper".  Given that this probably lives
in mod_sofia who knows what's really "proper".  It is SIP after all...

  So the issue comes up when using FreeSWITCH as a B2BUA and bridging
between endpoints (very common).  Should FreeSWITCH copy the X-
headers (possibly others) as it does now?  I'd like to think it
shouldn't by default and the behavior should be one of:

1)  Don't pass X-* (or anything else, really) from one leg to another.
 If you want to pass specific X- headers (or anything else), set them
explicitly on the outbound leg.
2)  Make the behavior configurable with a channel variable and/or
sofia config option:

{sip_pass_headers=all|none|X-MyCustomHeaderByName}

  Thoughts?

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From mattdfong at gmail.com  Fri Oct  9 12:34:01 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Sat, 10 Oct 2009 02:34:01 +0700
Subject: [Freeswitch-users] Sending an Event to a Session for onInput
Message-ID: <4256bf830910091234t11f4f4d0w28283f341bd1fb06@mail.gmail.com>

I'm used to using the onInput callbacks inside lua and javascript to listen
for dtmf and other events and perform a task accordingly. I'm wondering if
there is a way to send an event to a session or channel that can be caught
using the setInputCallback inside lua from outside the session program.
Maybe an API command that can generate an event for a specific UUID. Does a
mechanism exist to do this that I'm over looking? Thanks.
--matt
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From jerry.richards at teotech.com  Fri Oct  9 13:06:43 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 9 Oct 2009 13:06:43 -0700
Subject: [Freeswitch-users] FW: FS Does Not Relay
	PresencePUBLISHToSubscribing Phones
Message-ID: <6E290A3B014040CFA97E8BE97BC36BAC@greyhawk.tonecommander.com>

I put the sqlite3 select query in the paste bin again, and prior to that, I
entered the .dump command.  The select command came back with the sqlite3
prompt, which I guess means it didn't find an entry.  How do I go about
isolating this problem?  I'm using CounterPath's Bria Professional
softphone.  They are the same company that make the Eyebeam.
 
Any ideas?
 
Best Regards,
Jerry


  _____  

From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Friday, October 02, 2009 11:28 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: [Freeswitch-users] FS Does Not Relay
PresencePUBLISHToSubscribing Phones


I put the sqlite3 select query in the paste bin, and prior to that, I
entered the .dump command.  The select command came back with a "...>"
prompt which I don't understand.  I don't know enough about sqlite3 to know
what that means?
 
Best Regards,
Jerry


  _____  

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Friday, October 02, 2009 10:52 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Does Not Relay
PresencePUBLISHToSubscribing Phones


connect to sqlite directly with sqlite3 app and try that sql stmt and see
why it doesn't match anything.

sqlite3 /usr/local/freeswitch/db/sofia_reg_internal.db

select
sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_hos
t,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscripti
ons.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscripti
ons.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subsc
riptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,
'Away','away','192.168.72.38',sip_presence.status,sip_presence.rpid from
sip_subscriptions left join sip_presence on
(sip_subscriptions.sub_to_user=sip_presence.sip_user and
sip_subscriptions.sub_to_host=sip_presence.sip_host and
sip_subscriptions.profile_name=sip_presence.profile_name) where
(event='presence' or event='presence') and sub_to_user='1001' and
(sub_to_host='192.168.72.38' or presence_hosts like '%192.168.72.38%') and
(sip_subscriptions.profile_name = 'external' or
sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host)



On Fri, Oct 2, 2009 at 12:12 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:


Okay, I put a log up on the pastebin that shows the PUBLISH event coming
from a CounterPath Bria Professional phone.  For some reason, FS is getting
an error and not relaying the presence status to the subscriber.
 
Best Regards,
Jerry


  _____  

From: Jo?o Mesquita [mailto:jmesquita at freeswitch.org] 
Sent: Thursday, October 01, 2009 8:14 PM 

To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Does Not Relay Presence
PUBLISHToSubscribing Phones


Piece of advice, don't ask, just do it. ;)

jmesquita


On Thu, Oct 1, 2009 at 3:29 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:


If you have time to take a look, I could put a trace in the pastebin?
 
Jerry


  _____  


From: Jerry Richards [mailto:jerry.richards at teotech.com] 

Sent: Thursday, October 01, 2009 10:29 AM 

To: 'freeswitch-users at lists.freeswitch.org'

Subject: RE: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
ToSubscribing Phones


I am using two Bria Professional Version 2.5.4 Build 54835 softphones.
 
Thanks,
Jerry


  _____  


From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 

Sent: Thursday, October 01, 2009 9:36 AM 

To: freeswitch-users at lists.freeswitch.org

Subject: Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
ToSubscribing Phones


which phone is it,
we tested it with eyebeam and it appears to work for us.



On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards <jerry.richards at teotech.com>
wrote:



By the way, I see the following lines at the FS console, which might be a
clue as to why this is happening.  Could someone point me toward what might
cause this?  I set the "manage-presence" parameter to "true" in each XML
file where I saw it defined.

[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[WARNING] sofia_presence.c:565  192.168.72.38 is an alias, skipping


Best Regards,
Jerry



-----Original Message-----
From: Jerry Richards [mailto:jerry.richards at teotech.com]
Sent: Wednesday, September 30, 2009 9:12 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones

I have two phones configured to subscribe to each other's presence status.
When I change the presence status in one phone, I see the SIP PUBLISH
message going to FS, but I don't see FS relaying that presence status to the
subscribing phone.  Does anyone know why?

Best Regards,
Jerry


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:213-799-1400



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org






-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:213-799-1400


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From andrew at hijacked.us  Fri Oct  9 13:24:10 2009
From: andrew at hijacked.us (Andrew Thompson)
Date: Fri, 9 Oct 2009 16:24:10 -0400
Subject: [Freeswitch-users] FreeSWITCH Update: Valet Parking
In-Reply-To: <87f2f3b90910091101x33bfbd8vf504b8bb3186da08@mail.gmail.com>
References: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>
	<23f91030910090543o1f70c997t7d860d4d65b2a175@mail.gmail.com>
	<191c3a030910090725o4150a17x56b93303bf0b8c2b@mail.gmail.com>
	<b39273530910090750ga9debecp783d374560e8c577@mail.gmail.com>
	<87f2f3b90910091101x33bfbd8vf504b8bb3186da08@mail.gmail.com>
Message-ID: <20091009202409.GQ4572@hijacked.us>

On Fri, Oct 09, 2009 at 11:01:34AM -0700, Michael Collins wrote:
> On Fri, Oct 9, 2009 at 7:50 AM, Rupa Schomaker <rupa at rupa.com> wrote:
> 
> > Oh, definitely.  I think both use cases (existing, and park in next
> > avail slot and read back slot #) are useful.
> >
> >
> They do have some usefulness. The most important reason for this feature,
> though, is for the person who handles calls all day and moves them around.
> That person will most likely have a pretty good idea of which parking stalls
> are available. However, I do like the idea of a webby interface showing what
> stalls are in use...

mod_snom and the programmable LEDs would be a handy way to do it too. I
implemented a poor-man's version of a key system using that, but I think
I'll replace the parking portion with the valet parking stuff when I
get a chance.

Andrew


From diego.viola at gmail.com  Fri Oct  9 14:27:20 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 9 Oct 2009 21:27:20 +0000
Subject: [Freeswitch-users] Heartbeat question
In-Reply-To: <191c3a030910090902t38282f72l6b6d1733a758836@mail.gmail.com>
References: <86a32abc0910081956k91b3600sb8c556e589f5b089@mail.gmail.com>
	<86a32abc0910082003v3f5d4e6dua8e7eae18fd2f939@mail.gmail.com>
	<6b65470d0910082036g204d5dcbw26e1e43ef80c7c84@mail.gmail.com>
	<86a32abc0910082245j36b8b3e9ncaf1cdd358fce319@mail.gmail.com>
	<191c3a030910090902t38282f72l6b6d1733a758836@mail.gmail.com>
Message-ID: <86a32abc0910091427i2395fd80t3bc557186ece3972@mail.gmail.com>

Thanks Anthony, this solved it. You rock :)

My program now outputs:

Got a SESSION_HEARTBEAT at 17:14:59
Got a SESSION_HEARTBEAT at 17:15:00
Got a SESSION_HEARTBEAT at 17:15:02
Got a SESSION_HEARTBEAT at 17:15:03
Got a SESSION_HEARTBEAT at 17:15:04
Got a SESSION_HEARTBEAT at 17:15:05
Got a SESSION_HEARTBEAT at 17:15:06
Got a SESSION_HEARTBEAT at 17:15:07
Got a SESSION_HEARTBEAT at 17:15:08
Got a SESSION_HEARTBEAT at 17:15:09
Got a SESSION_HEARTBEAT at 17:15:10
Got a SESSION_HEARTBEAT at 17:15:11
Got a SESSION_HEARTBEAT at 17:15:12
Got a SESSION_HEARTBEAT at 17:15:13
Got a SESSION_HEARTBEAT at 17:15:14
Got a SESSION_HEARTBEAT at 17:15:15
Got a SESSION_HEARTBEAT at 17:15:16
Got a SESSION_HEARTBEAT at 17:15:17
Got a SESSION_HEARTBEAT at 17:15:18
Got a SESSION_HEARTBEAT at 17:15:19
Got a SESSION_HEARTBEAT at 17:15:20
Got a SESSION_HEARTBEAT at 17:15:21
Got a SESSION_HEARTBEAT at 17:15:22
Got a SESSION_HEARTBEAT at 17:15:23
Got a SESSION_HEARTBEAT at 17:15:24
Got a SESSION_HEARTBEAT at 17:15:25
Got a SESSION_HEARTBEAT at 17:15:26
Got a SESSION_HEARTBEAT at 17:15:27
Got a SESSION_HEARTBEAT at 17:15:28
Got a SESSION_HEARTBEAT at 17:15:29
Got a SESSION_HEARTBEAT at 17:15:30




On Fri, Oct 9, 2009 at 4:02 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Update to trunk and try it with fs_cli it for sure will let you do every 1
> second
>
> in fs_cli type
>
> /events plain all
>
> if you make that call you will see one every 1 second
>
>
>
> On Fri, Oct 9, 2009 at 12:45 AM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> Nope, I was just wondering why it didn't work at 1 second exactly...
>>
>>
>> On Fri, Oct 9, 2009 at 3:36 AM, William Suffill <
>> william.suffill at gmail.com> wrote:
>>
>>> Why do you need it every second? If you want real time channel counts
>>> you would  be able to track each create/destroy even instead of
>>> relying on the heartbeat summary.
>>>
>>> -- W
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From diego.viola at gmail.com  Fri Oct  9 14:30:12 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 9 Oct 2009 21:30:12 +0000
Subject: [Freeswitch-users] Heartbeat question
In-Reply-To: <86a32abc0910091427i2395fd80t3bc557186ece3972@mail.gmail.com>
References: <86a32abc0910081956k91b3600sb8c556e589f5b089@mail.gmail.com>
	<86a32abc0910082003v3f5d4e6dua8e7eae18fd2f939@mail.gmail.com>
	<6b65470d0910082036g204d5dcbw26e1e43ef80c7c84@mail.gmail.com>
	<86a32abc0910082245j36b8b3e9ncaf1cdd358fce319@mail.gmail.com>
	<191c3a030910090902t38282f72l6b6d1733a758836@mail.gmail.com>
	<86a32abc0910091427i2395fd80t3bc557186ece3972@mail.gmail.com>
Message-ID: <86a32abc0910091430hc151ed8of085fcc3af2f1de4@mail.gmail.com>

Here is on two seconds ;)

Got a SESSION_HEARTBEAT at 17:17:13
Got a SESSION_HEARTBEAT at 17:17:15
Got a SESSION_HEARTBEAT at 17:17:17
Got a SESSION_HEARTBEAT at 17:17:19
Got a SESSION_HEARTBEAT at 17:17:21
Got a SESSION_HEARTBEAT at 17:17:23
Got a SESSION_HEARTBEAT at 17:17:25
Got a SESSION_HEARTBEAT at 17:17:27
Got a SESSION_HEARTBEAT at 17:17:29
Got a SESSION_HEARTBEAT at 17:17:31
Got a SESSION_HEARTBEAT at 17:17:33
Got a SESSION_HEARTBEAT at 17:17:35
Got a SESSION_HEARTBEAT at 17:17:37
Got a SESSION_HEARTBEAT at 17:17:39
Got a SESSION_HEARTBEAT at 17:17:41
Got a SESSION_HEARTBEAT at 17:17:43
Got a SESSION_HEARTBEAT at 17:17:45
Got a SESSION_HEARTBEAT at 17:17:47
Got a SESSION_HEARTBEAT at 17:17:49
Got a SESSION_HEARTBEAT at 17:17:51
Got a SESSION_HEARTBEAT at 17:17:53
Got a SESSION_HEARTBEAT at 17:17:55
Got a SESSION_HEARTBEAT at 17:17:57
Got a SESSION_HEARTBEAT at 17:17:59
Got a SESSION_HEARTBEAT at 17:18:01
Got a SESSION_HEARTBEAT at 17:18:03
Got a SESSION_HEARTBEAT at 17:18:05
Got a SESSION_HEARTBEAT at 17:18:07
Got a SESSION_HEARTBEAT at 17:18:09
Got a SESSION_HEARTBEAT at 17:18:11
Got a SESSION_HEARTBEAT at 17:18:13
Got a SESSION_HEARTBEAT at 17:18:15
Got a SESSION_HEARTBEAT at 17:18:17


On Fri, Oct 9, 2009 at 9:27 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Thanks Anthony, this solved it. You rock :)
>
> My program now outputs:
>
> Got a SESSION_HEARTBEAT at 17:14:59
> Got a SESSION_HEARTBEAT at 17:15:00
> Got a SESSION_HEARTBEAT at 17:15:02
> Got a SESSION_HEARTBEAT at 17:15:03
> Got a SESSION_HEARTBEAT at 17:15:04
> Got a SESSION_HEARTBEAT at 17:15:05
> Got a SESSION_HEARTBEAT at 17:15:06
> Got a SESSION_HEARTBEAT at 17:15:07
> Got a SESSION_HEARTBEAT at 17:15:08
> Got a SESSION_HEARTBEAT at 17:15:09
> Got a SESSION_HEARTBEAT at 17:15:10
> Got a SESSION_HEARTBEAT at 17:15:11
> Got a SESSION_HEARTBEAT at 17:15:12
> Got a SESSION_HEARTBEAT at 17:15:13
> Got a SESSION_HEARTBEAT at 17:15:14
> Got a SESSION_HEARTBEAT at 17:15:15
> Got a SESSION_HEARTBEAT at 17:15:16
> Got a SESSION_HEARTBEAT at 17:15:17
> Got a SESSION_HEARTBEAT at 17:15:18
> Got a SESSION_HEARTBEAT at 17:15:19
> Got a SESSION_HEARTBEAT at 17:15:20
> Got a SESSION_HEARTBEAT at 17:15:21
> Got a SESSION_HEARTBEAT at 17:15:22
> Got a SESSION_HEARTBEAT at 17:15:23
> Got a SESSION_HEARTBEAT at 17:15:24
> Got a SESSION_HEARTBEAT at 17:15:25
> Got a SESSION_HEARTBEAT at 17:15:26
> Got a SESSION_HEARTBEAT at 17:15:27
> Got a SESSION_HEARTBEAT at 17:15:28
> Got a SESSION_HEARTBEAT at 17:15:29
> Got a SESSION_HEARTBEAT at 17:15:30
>
>
>
>
>
> On Fri, Oct 9, 2009 at 4:02 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Update to trunk and try it with fs_cli it for sure will let you do every 1
>> second
>>
>> in fs_cli type
>>
>> /events plain all
>>
>> if you make that call you will see one every 1 second
>>
>>
>>
>> On Fri, Oct 9, 2009 at 12:45 AM, Diego Viola <diego.viola at gmail.com>wrote:
>>
>>> Nope, I was just wondering why it didn't work at 1 second exactly...
>>>
>>>
>>> On Fri, Oct 9, 2009 at 3:36 AM, William Suffill <
>>> william.suffill at gmail.com> wrote:
>>>
>>>> Why do you need it every second? If you want real time channel counts
>>>> you would  be able to track each create/destroy even instead of
>>>> relying on the heartbeat summary.
>>>>
>>>> -- W
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From diego.viola at gmail.com  Fri Oct  9 15:01:46 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 9 Oct 2009 22:01:46 +0000
Subject: [Freeswitch-users] Heartbeat question
In-Reply-To: <86a32abc0910091430hc151ed8of085fcc3af2f1de4@mail.gmail.com>
References: <86a32abc0910081956k91b3600sb8c556e589f5b089@mail.gmail.com>
	<86a32abc0910082003v3f5d4e6dua8e7eae18fd2f939@mail.gmail.com>
	<6b65470d0910082036g204d5dcbw26e1e43ef80c7c84@mail.gmail.com>
	<86a32abc0910082245j36b8b3e9ncaf1cdd358fce319@mail.gmail.com>
	<191c3a030910090902t38282f72l6b6d1733a758836@mail.gmail.com>
	<86a32abc0910091427i2395fd80t3bc557186ece3972@mail.gmail.com>
	<86a32abc0910091430hc151ed8of085fcc3af2f1de4@mail.gmail.com>
Message-ID: <86a32abc0910091501y40dd34c4r1bca3eeb2b6fa693@mail.gmail.com>

Here's my heartbeat script now.

#!/usr/bin/env ruby

require 'rubygems'
require 'fsr'
require "fsr/listener/inbound"

def custom_channel_heartbeat_handler(event)
  puts "Got a SESSION_HEARTBEAT at #{Time.now.strftime('%H:%M:%S')}"
end
FSL::Inbound.add_event_hook(:SESSION_HEARTBEAT) {|event|
custom_channel_heartbeat_handler(event) }
FSR.start_ies!(FSL::Inbound, :host => "localhost", :port => 8021)

Thanks again.

Diego

On Fri, Oct 9, 2009 at 9:30 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Here is on two seconds ;)
>
> Got a SESSION_HEARTBEAT at 17:17:13
> Got a SESSION_HEARTBEAT at 17:17:15
> Got a SESSION_HEARTBEAT at 17:17:17
> Got a SESSION_HEARTBEAT at 17:17:19
> Got a SESSION_HEARTBEAT at 17:17:21
> Got a SESSION_HEARTBEAT at 17:17:23
> Got a SESSION_HEARTBEAT at 17:17:25
> Got a SESSION_HEARTBEAT at 17:17:27
> Got a SESSION_HEARTBEAT at 17:17:29
> Got a SESSION_HEARTBEAT at 17:17:31
> Got a SESSION_HEARTBEAT at 17:17:33
> Got a SESSION_HEARTBEAT at 17:17:35
> Got a SESSION_HEARTBEAT at 17:17:37
> Got a SESSION_HEARTBEAT at 17:17:39
> Got a SESSION_HEARTBEAT at 17:17:41
> Got a SESSION_HEARTBEAT at 17:17:43
> Got a SESSION_HEARTBEAT at 17:17:45
> Got a SESSION_HEARTBEAT at 17:17:47
> Got a SESSION_HEARTBEAT at 17:17:49
> Got a SESSION_HEARTBEAT at 17:17:51
> Got a SESSION_HEARTBEAT at 17:17:53
> Got a SESSION_HEARTBEAT at 17:17:55
> Got a SESSION_HEARTBEAT at 17:17:57
> Got a SESSION_HEARTBEAT at 17:17:59
> Got a SESSION_HEARTBEAT at 17:18:01
> Got a SESSION_HEARTBEAT at 17:18:03
> Got a SESSION_HEARTBEAT at 17:18:05
> Got a SESSION_HEARTBEAT at 17:18:07
> Got a SESSION_HEARTBEAT at 17:18:09
> Got a SESSION_HEARTBEAT at 17:18:11
> Got a SESSION_HEARTBEAT at 17:18:13
> Got a SESSION_HEARTBEAT at 17:18:15
> Got a SESSION_HEARTBEAT at 17:18:17
>
>
>
> On Fri, Oct 9, 2009 at 9:27 PM, Diego Viola <diego.viola at gmail.com> wrote:
>
>> Thanks Anthony, this solved it. You rock :)
>>
>> My program now outputs:
>>
>> Got a SESSION_HEARTBEAT at 17:14:59
>> Got a SESSION_HEARTBEAT at 17:15:00
>> Got a SESSION_HEARTBEAT at 17:15:02
>> Got a SESSION_HEARTBEAT at 17:15:03
>> Got a SESSION_HEARTBEAT at 17:15:04
>> Got a SESSION_HEARTBEAT at 17:15:05
>> Got a SESSION_HEARTBEAT at 17:15:06
>> Got a SESSION_HEARTBEAT at 17:15:07
>> Got a SESSION_HEARTBEAT at 17:15:08
>> Got a SESSION_HEARTBEAT at 17:15:09
>> Got a SESSION_HEARTBEAT at 17:15:10
>> Got a SESSION_HEARTBEAT at 17:15:11
>> Got a SESSION_HEARTBEAT at 17:15:12
>> Got a SESSION_HEARTBEAT at 17:15:13
>> Got a SESSION_HEARTBEAT at 17:15:14
>> Got a SESSION_HEARTBEAT at 17:15:15
>> Got a SESSION_HEARTBEAT at 17:15:16
>> Got a SESSION_HEARTBEAT at 17:15:17
>> Got a SESSION_HEARTBEAT at 17:15:18
>> Got a SESSION_HEARTBEAT at 17:15:19
>> Got a SESSION_HEARTBEAT at 17:15:20
>> Got a SESSION_HEARTBEAT at 17:15:21
>> Got a SESSION_HEARTBEAT at 17:15:22
>> Got a SESSION_HEARTBEAT at 17:15:23
>> Got a SESSION_HEARTBEAT at 17:15:24
>> Got a SESSION_HEARTBEAT at 17:15:25
>> Got a SESSION_HEARTBEAT at 17:15:26
>> Got a SESSION_HEARTBEAT at 17:15:27
>> Got a SESSION_HEARTBEAT at 17:15:28
>> Got a SESSION_HEARTBEAT at 17:15:29
>> Got a SESSION_HEARTBEAT at 17:15:30
>>
>>
>>
>>
>>
>> On Fri, Oct 9, 2009 at 4:02 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> Update to trunk and try it with fs_cli it for sure will let you do every
>>> 1 second
>>>
>>> in fs_cli type
>>>
>>> /events plain all
>>>
>>> if you make that call you will see one every 1 second
>>>
>>>
>>>
>>> On Fri, Oct 9, 2009 at 12:45 AM, Diego Viola <diego.viola at gmail.com>wrote:
>>>
>>>> Nope, I was just wondering why it didn't work at 1 second exactly...
>>>>
>>>>
>>>> On Fri, Oct 9, 2009 at 3:36 AM, William Suffill <
>>>> william.suffill at gmail.com> wrote:
>>>>
>>>>> Why do you need it every second? If you want real time channel counts
>>>>> you would  be able to track each create/destroy even instead of
>>>>> relying on the heartbeat summary.
>>>>>
>>>>> -- W
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:213-799-1400
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
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From anthony.minessale at gmail.com  Fri Oct  9 15:53:13 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 9 Oct 2009 17:53:13 -0500
Subject: [Freeswitch-users] FW: FS Does Not Relay
	PresencePUBLISHToSubscribing Phones
In-Reply-To: <6E290A3B014040CFA97E8BE97BC36BAC@greyhawk.tonecommander.com>
References: <6E290A3B014040CFA97E8BE97BC36BAC@greyhawk.tonecommander.com>
Message-ID: <191c3a030910091553s3d16a992x2b0116abe686408a@mail.gmail.com>

yah what i was getting at was between the sql statement and what was
actually in the table you should be able to tell what's wrong.


On Fri, Oct 9, 2009 at 3:06 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>  I put the sqlite3 select query in the paste bin again, and prior to that,
> I entered the .dump command.  The select command came back with the
> sqlite3 prompt, which I guess means it didn't find an entry.  How do I go
> about isolating this problem?  I'm using CounterPath's Bria Professional
> softphone.  They are the same company that make the Eyebeam.
>
> Any ideas?
>
> Best Regards,
> Jerry
>
>
>  ------------------------------
> *From:* Jerry Richards [mailto:jerry.richards at teotech.com]
> *Sent:* Friday, October 02, 2009 11:28 AM
> *To:* 'freeswitch-users at lists.freeswitch.org'
> *Subject:* RE: [Freeswitch-users] FS Does Not Relay
> PresencePUBLISHToSubscribing Phones
>
>  I put the sqlite3 select query in the paste bin, and prior to that, I
> entered the .dump command.  The select command came back with a "...>"
> prompt which I don't understand.  I don't know enough about sqlite3 to know
> what that means?
>
> Best Regards,
> Jerry
>
>  ------------------------------
> *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com]
> *Sent:* Friday, October 02, 2009 10:52 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] FS Does Not Relay
> PresencePUBLISHToSubscribing Phones
>
> connect to sqlite directly with sqlite3 app and try that sql stmt and see
> why it doesn't match anything.
>
> sqlite3 /usr/local/freeswitch/db/sofia_reg_internal.db
>
> select
> sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Away','away','
> 192.168.72.38',sip_presence.status,sip_presence.rpid from
> sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user
> and sip_subscriptions.sub_to_host=sip_presence.sip_host and
> sip_subscriptions.profile_name=sip_presence.profile_name) where (event='presence'
> or event='presence') and sub_to_user='1001' and (sub_to_host='192.168.72
> .38' or presence_hosts like '%192.168.72.38%') and (sip_subscriptions.profile_name
> = 'external' or sip_subscriptions.presence_hosts !=
> sip_subscriptions.sub_to_host)
>
>
> On Fri, Oct 2, 2009 at 12:12 PM, Jerry Richards <
> jerry.richards at teotech.com> wrote:
>
>>  Okay, I put a log up on the pastebin that shows the PUBLISH event coming
>> from a CounterPath Bria Professional phone.  For some reason, FS is getting
>> an error and not relaying the presence status to the subscriber.
>>
>> Best Regards,
>> Jerry
>>
>>  ------------------------------
>> *From:* Jo?o Mesquita [mailto:jmesquita at freeswitch.org]
>> *Sent:* Thursday, October 01, 2009 8:14 PM
>>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* Re: [Freeswitch-users] FS Does Not Relay Presence
>> PUBLISHToSubscribing Phones
>>
>>   Piece of advice, don't ask, just do it. ;)
>>
>> jmesquita
>>
>> On Thu, Oct 1, 2009 at 3:29 PM, Jerry Richards <
>> jerry.richards at teotech.com> wrote:
>>
>>>  If you have time to take a look, I could put a trace in the pastebin?
>>>
>>> Jerry
>>>
>>>  ------------------------------
>>>  *From:* Jerry Richards [mailto:jerry.richards at teotech.com]
>>> *Sent:* Thursday, October 01, 2009 10:29 AM
>>> *To:* 'freeswitch-users at lists.freeswitch.org'
>>> *Subject:* RE: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
>>> ToSubscribing Phones
>>>
>>>  I am using two Bria Professional Version 2.5.4 Build 54835 softphones.
>>>
>>> Thanks,
>>> Jerry
>>>
>>>  ------------------------------
>>>  *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com]
>>> *Sent:* Thursday, October 01, 2009 9:36 AM
>>> *To:* freeswitch-users at lists.freeswitch.org
>>> *Subject:* Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
>>> ToSubscribing Phones
>>>
>>>   which phone is it,
>>> we tested it with eyebeam and it appears to work for us.
>>>
>>>
>>> On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards <
>>> jerry.richards at teotech.com> wrote:
>>>
>>>>
>>>> By the way, I see the following lines at the FS console, which might be
>>>> a
>>>> clue as to why this is happening.  Could someone point me toward what
>>>> might
>>>> cause this?  I set the "manage-presence" parameter to "true" in each XML
>>>> file where I saw it defined.
>>>>
>>>> [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal)
>>>> [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
>>>>   ...
>>>> [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6)
>>>> [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
>>>>   ...
>>>> [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external)
>>>> [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
>>>>   ...
>>>> [WARNING] sofia_presence.c:565  192.168.72.38 is an alias, skipping
>>>>
>>>>
>>>> Best Regards,
>>>> Jerry
>>>>
>>>>
>>>> -----Original Message-----
>>>> From: Jerry Richards [mailto:jerry.richards at teotech.com]
>>>> Sent: Wednesday, September 30, 2009 9:12 AM
>>>> To: 'freeswitch-users at lists.freeswitch.org'
>>>> Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones
>>>>
>>>> I have two phones configured to subscribe to each other's presence
>>>> status.
>>>> When I change the presence status in one phone, I see the SIP PUBLISH
>>>> message going to FS, but I don't see FS relaying that presence status to
>>>> the
>>>> subscribing phone.  Does anyone know why?
>>>>
>>>> Best Regards,
>>>> Jerry
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:213-799-1400
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From nagalenoj at gmail.com  Fri Oct  9 20:50:28 2009
From: nagalenoj at gmail.com (Nagalenoj)
Date: Fri, 9 Oct 2009 20:50:28 -0700 (PDT)
Subject: [Freeswitch-users] Re  corded file as voicemail.
In-Reply-To: <BC536E8C-6813-4A69-9B80-EA11E41A79FC@jerris.com>
References: <25751158.post@talk.nabble.com>
	<BC536E8C-6813-4A69-9B80-EA11E41A79FC@jerris.com>
Message-ID: <25830785.post@talk.nabble.com>


No, When I do voicemail_inject it is reporting the following error.

'Stereo is currently not supported, please downsample to mono.'

Any other way to do this.??


Michael Jerris wrote:
> 
> http://wiki.freeswitch.org/wiki/Mod_voicemail#voicemail_inject
> 
> On Oct 5, 2009, at 9:46 AM, Nagalenoj wrote:
> 
>>
>> Is it possible to treat a recorded voice as voice mail?
>>
>> Assume that, I've recorded a conversation and I want this recorded  
>> file to
>> be treated like voicemail. So, I could check it like voicemail!!
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/Recorded-file-as-voicemail.-tp25751158p25830785.html
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From nagalenoj at gmail.com  Fri Oct  9 20:51:30 2009
From: nagalenoj at gmail.com (Nagalenoj)
Date: Fri, 9 Oct 2009 20:51:30 -0700 (PDT)
Subject: [Freeswitch-users] Re  corded file as voicemail.
Message-ID: <25830785.post@talk.nabble.com>


No, When I do voicemail_inject and check through voicemail, it is not playing
the file instead reporting the following error.

'Stereo is currently not supported, please downsample to mono.'

Any other way to do this.??


Michael Jerris wrote:
> 
> http://wiki.freeswitch.org/wiki/Mod_voicemail#voicemail_inject
> 
> On Oct 5, 2009, at 9:46 AM, Nagalenoj wrote:
> 
>>
>> Is it possible to treat a recorded voice as voice mail?
>>
>> Assume that, I've recorded a conversation and I want this recorded  
>> file to
>> be treated like voicemail. So, I could check it like voicemail!!
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/Recorded-file-as-voicemail.-tp25751158p25830785.html
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From jason at jasonjgw.net  Fri Oct  9 21:05:39 2009
From: jason at jasonjgw.net (Jason White)
Date: Sat, 10 Oct 2009 15:05:39 +1100
Subject: [Freeswitch-users] Re  corded file as voicemail.
In-Reply-To: <25830785.post@talk.nabble.com>
References: <25830785.post@talk.nabble.com>
Message-ID: <20091010040539.GB23188@jdc.jasonjgw.net>

Nagalenoj <nagalenoj at gmail.com> wrote:
> 
> No, When I do voicemail_inject and check through voicemail, it is not playing
> the file instead reporting the following error.
> 
> 'Stereo is currently not supported, please downsample to mono.'

This has been discussed on the list before. I think the solution was to use
sox to convert the files to mono. You might need a script to do this before
injecting them into the voicemail.



From lakindia89 at gmail.com  Fri Oct  9 21:39:07 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Sat, 10 Oct 2009 10:09:07 +0530
Subject: [Freeswitch-users] libpri_span vs prispans
In-Reply-To: <87f2f3b90910091057q3a033548j462446cf424ad257@mail.gmail.com>
References: <7d79b3930910090459v2b07e3aapd715c7618f11ada0@mail.gmail.com>
	<20091009125714.E0FF73043D7@mail.cune.org>
	<87f2f3b90910091057q3a033548j462446cf424ad257@mail.gmail.com>
Message-ID: <7d79b3930910092139v3589b761oab0bae46598118f@mail.gmail.com>

Where can I found entire configurations for a libpri spans. Right now I've
only 4 things in my libpri configuration. They are Switchtype, Node,
Contect, Dialplan, which is given as example in wiki.

But I think libprispan has more configuration. Where can I find those
things?



On Fri, Oct 9, 2009 at 11:27 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Oct 9, 2009 at 5:57 AM, <Russell.Mosemann at cune.org> wrote:
>
>> lakshmanan ganapathy <lakindia89 at gmail.com> said:
>> > What difference it will make, if I use pri_span configuration and
>> lib_pri
>> > span configuration in openzap.conf.xml.
>> > I'm preety much confused on the difference between this.
>>
>> Openzap can handle a T1/E1 itself, or openzap can use libpri to do that.
>> libpri is more mature than openzap. In my case openzap was not able to
>> handle everything invovling a T1 here, and using libpri fixed it.
>>
>> A libpri_span configuration is handled by libpri. A pri_span
>> configuration is handled natively by openzap. Use the one that works for
>> you.
>>
>
> Right now the libpri stack is more mature than the ozmod_isdn stuff. See
> the OpenZAP page on the wiki for instructions on how to set up ozmod_libpri.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Fri Oct  9 22:40:18 2009
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 10 Oct 2009 13:40:18 +0800
Subject: [Freeswitch-users] Re corded file as voicemail.
In-Reply-To: <20091010040539.GB23188@jdc.jasonjgw.net>
References: <25830785.post@talk.nabble.com>
	<20091010040539.GB23188@jdc.jasonjgw.net>
Message-ID: <23f91030910092240u53b0cc51jd5b1670901f20e21@mail.gmail.com>

Yes, it's discussed before.

http://wiki.freeswitch.org/wiki/Channel_Variables#RECORD_STEREO

set that var to false before you record.


2009/10/10 Jason White <jason at jasonjgw.net>

> Nagalenoj <nagalenoj at gmail.com> wrote:
> >
> > No, When I do voicemail_inject and check through voicemail, it is not
> playing
> > the file instead reporting the following error.
> >
> > 'Stereo is currently not supported, please downsample to mono.'
>
> This has been discussed on the list before. I think the solution was to use
> sox to convert the files to mono. You might need a script to do this before
> injecting them into the voicemail.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From velu.technical at gmail.com  Fri Oct  9 23:51:43 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Sat, 10 Oct 2009 12:21:43 +0530
Subject: [Freeswitch-users] Difference between park and valet_park
Message-ID: <1452e2980910092351v766cad5bpcb55b59ad5fee6d0@mail.gmail.com>

Dear All,
     Could you please any one explain the difference between parking and
valet parking?
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From jason at jasonjgw.net  Sat Oct 10 00:00:03 2009
From: jason at jasonjgw.net (Jason White)
Date: Sat, 10 Oct 2009 18:00:03 +1100
Subject: [Freeswitch-users] Status of ubuntu/debian packages.
In-Reply-To: <4ABA47FB.2050100@gmail.com>
References: <4ABA47FB.2050100@gmail.com>
Message-ID: <20091010070003.GA16076@jdc.jasonjgw.net>

for those who are working on the Debian packages, the following ITP bug may be
of interest.
http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=513606
which also refers to an older entry.

Thanks are due to all who are contributing to the packaging effort.



From dujinfang at gmail.com  Sat Oct 10 00:32:37 2009
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 10 Oct 2009 15:32:37 +0800
Subject: [Freeswitch-users] Difference between park and valet_park
In-Reply-To: <1452e2980910092351v766cad5bpcb55b59ad5fee6d0@mail.gmail.com>
References: <1452e2980910092351v766cad5bpcb55b59ad5fee6d0@mail.gmail.com>
Message-ID: <23f91030910100032k41513eecp2db3d32a79dfe724@mail.gmail.com>

search this list, just has been discussed.

2009/10/10 velusamy velu <velu.technical at gmail.com>

> Dear All,
>      Could you please any one explain the difference between parking and
> valet parking?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From velu.technical at gmail.com  Sat Oct 10 02:37:20 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Sat, 10 Oct 2009 15:07:20 +0530
Subject: [Freeswitch-users] Play music on hold after parking the call
Message-ID: <1452e2980910100237u317d6138w1d13bd3cfb250353@mail.gmail.com>

Dear All,
      I am using ESL.pm module to control the dial plan application. I want
to play some music while executing the some external scripts. I executed
park after then I executed the playback the music didn't play.

    Could any one please explain how can I solve this problem without using
async mode in socket application?

Thanks,
Velusamy.
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From wangdq.no1 at gmail.com  Sat Oct 10 02:53:19 2009
From: wangdq.no1 at gmail.com (daqiang wang)
Date: Sat, 10 Oct 2009 17:53:19 +0800
Subject: [Freeswitch-users] how to match '#' in XML dialplan ?
Message-ID: <ecd05ce30910100253q1424cd4an9be93ab434912b3f@mail.gmail.com>

hello every one :
   I want to match the # in XML dialplan , how to do  ?
   example :
      1#5555#6666 . how to do ?
     I do this :
     <condition field="destination_number" expression="^1#(d+)#(d+)$"/>
    but it's not work
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From maciej.aniserowicz at gmail.com  Sat Oct 10 03:04:04 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Sat, 10 Oct 2009 03:04:04 -0700 (PDT)
Subject: [Freeswitch-users] Bad sound quality while eavesdropping
In-Reply-To: <191c3a030910080823g79c7c596x1cd887e1538ce2e1@mail.gmail.com>
References: <41A44DD027064988A914974405788C2E@procent>
	<191c3a030910050731m2d74979ep4598e5a1945d58ae@mail.gmail.com>
	<1254901192035-3780245.post@n2.nabble.com>
	<8437F5BC-7AFF-4A74-B8CD-C5B8219021F6@jerris.com>
	<1255008427639-3788019.post@n2.nabble.com>
	<191c3a030910080823g79c7c596x1cd887e1538ce2e1@mail.gmail.com>
Message-ID: <1255169044209-3799274.post@n2.nabble.com>


Hi,
Here are the messages with a:ptime parameter. All the calls are started by
commands sent through socket.
I'm not sure if this is all information you need, please let me know if
something is missing here and I'll post that.

1) starting connection with x-lite (number 2003, the eavesdropper):

   INVITE sip:2003 at 192.168.3.100:60188;rinstance=80b8f8d92af87cd2 SIP/2.0
   Via: SIP/2.0/UDP 192.168.3.159:15060;rport;branch=z9hG4bKD9meHQN5XQ88K
   Max-Forwards: 69
   From: "MyApp" <sip:0000000000 at 192.168.3.159>;tag=jpQ6D7D2jUXvF
   To: <sip:2003 at 192.168.3.100:60188;rinstance=80b8f8d92af87cd2>
   Call-ID: 663c4500-3024-122d-57a2-9be8d85a4cff
   CSeq: 121465610 INVITE
   Contact: <sip:mod_sofia at 192.168.3.159:15060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 447
   Remote-Party-ID: "MyApp"
<sip:0000000000 at 192.168.3.159>;party=calling;screen=yes;privacy=off
   
   v=0
   o=FreeSWITCH 2223565947735016740 3096553520713245589 IN IP4 192.168.3.159
   s=FreeSWITCH
   c=IN IP4 192.168.3.159
   t=0 0
   m=audio 29966 RTP/AVP 0 115 107 9 8 3 101 13
   a=rtpmap:0 PCMU/8000
   a=rtpmap:115 G7221/32000
   a=fmtp:115 bitrate=48000
   a=rtpmap:107 G7221/16000
   a=fmtp:107 bitrate=32000
   a=rtpmap:9 G722/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:3 GSM/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:13 CN/8000
   a=ptime:20


2) starting connection with cisco ip phone (number 2006, first leg of
eavesdropped session):

   INVITE sip:2006 at 192.168.2.106:5060;user=phone SIP/2.0
   Via: SIP/2.0/UDP 192.168.3.159:15060;rport;branch=z9hG4bKg40rp87FNjB1p
   Max-Forwards: 69
   From: "MyApp" <sip:0000000000 at 192.168.3.159>;tag=Q3N2pe2K47ctS
   To: <sip:2006 at 192.168.2.106:5060;user=phone>
   Call-ID: 6e6fd8a0-3024-122d-57a2-9be8d85a4cff
   CSeq: 121465616 INVITE
   Contact: <sip:mod_sofia at 192.168.3.159:15060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 447
   Remote-Party-ID: "MyApp"
<sip:0000000000 at 192.168.3.159>;party=calling;screen=yes;privacy=off
   
   v=0
   o=FreeSWITCH 8453497903781974949 2294419114567885490 IN IP4 192.168.3.159
   s=FreeSWITCH
   c=IN IP4 192.168.3.159
   t=0 0
   m=audio 17670 RTP/AVP 0 115 107 9 8 3 101 13
   a=rtpmap:0 PCMU/8000
   a=rtpmap:115 G7221/32000
   a=fmtp:115 bitrate=48000
   a=rtpmap:107 G7221/16000
   a=fmtp:107 bitrate=32000
   a=rtpmap:9 G722/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:3 GSM/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:13 CN/8000
   a=ptime:20


3) starting connection with extension playing a file (number 9999, second
leg of eavesdropped session):

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP
192.168.3.159:5080;rport=5080;branch=z9hG4bKv6Dg7myj4tvBS
   From: "FreeSWITCH" <sip:myuser at mydomain;transport=udp>;tag=091j2Q0Fre8vp
   To: <sip:9999 at 192.168.3.159:15060>;tag=U7t5Xt51rB64Q
   Call-ID: 7551b7d0-3024-122d-b3a6-156c4939c4f8
   CSeq: 121465623 INVITE
   Contact: <sip:mod_sofia at 192.168.3.159:15060;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 263
   
   v=0
   o=FreeSWITCH 5614997529598779838 636912243381649698 IN IP4 192.168.3.159
   s=FreeSWITCH
   c=IN IP4 192.168.3.159
   t=0 0
   m=audio 30086 RTP/AVP 0 101 13
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:13 CN/8000
   a=ptime:20




Anthony Minessale wrote:
> 
> you probably have some device lying about ptime everywhere
> look at a sip trace an pay especially close attention to ptime:x param in
> sdp
> if you don't understand this just attach it here
> 
> execute the following at the cli
> sofia profile internal siptrace on
> sofila loglevel debug
> 
> 
> 
> On Thu, Oct 8, 2009 at 8:27 AM, Maciej Aniserowicz <
> maciej.aniserowicz at gmail.com> wrote:
> 
>>
>> It's the same on the trunk (the last rev I used was not so old anyway).
>>
>> Codecs are the same on both legs:
>> read codec/read rate: PCMU      8000
>> write codec/write rate: PCMU    8000
>>
>> MA
>>
>>
>>
>>
>> Michael Jerris wrote:
>> >
>> > What codecs are all the call legs using, also, please try current svn
>> > trunk.
>> >
>> > Mike
>> >
>> > On Oct 7, 2009, at 3:39 AM, Maciej Aniserowicz wrote:
>> >
>> >>
>> >> Sorry about posting several questions at once, I wasn't aware it's
>> >> "rude".
>> >> Let's concentrate on this issue then.
>> >>
>> >> I use FS rev 14994. Phones on extensions:
>> >> 1) x-lite
>> >> 2) cisco sip phone
>> >> 3) audio played by fs to the extension being eavesdropped
>> >>
>> >> I did not change any codec configuration, I just use the standard
>> >> one that
>> >> comes with both FS and the phones.
>> >> Some time ago someone on FS irc channel told me that this is just
>> >> how FS
>> >> eavesdropping works... from your response I understand that this is
>> >> not
>> >> entirely true?
>> >>
>> >> Maciej Aniserowicz
>> >>
>> >>
>> >>
>> >> Anthony Minessale wrote:
>> >>>
>> >>> That's is a somewhat vague position.
>> >>>
>> >>> You did not mention which version of FreeSWITCH you are running, the
>> >>> phones
>> >>> being used in your example, your configuration, the codecs in use
>> >>> etc.
>> >>>
>> >>> BTW,
>> >>> I think you should only ask one question at a time on this list.
>> >>> The list
>> >>> is run by volunteers and it's sort of rude to expect 3 or 4 threads
>> >>> to be
>> >>> tended to concerning the same one individual.
>> >>>
>> >>>
>> >>> 2009/10/5 Maciej Aniserowicz <maciej.aniserowicz at gmail.com>
>> >>>
>> >>>> Hello,
>> >>>> When I use eavesdropping in FreeSWITCH, the sound quality is
>> >>>> really bad.
>> >>>> Is
>> >>>> there any way to improve it? Is this a known problem?
>> >>>> Br/
>> >>>> Maciej Aniserowicz
>> >>>>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> --
>> View this message in context:
>> http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3788019.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3799274.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From juanbackson at gmail.com  Sat Oct 10 03:05:21 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Sat, 10 Oct 2009 18:05:21 +0800
Subject: [Freeswitch-users] directory variables not set correct with the
	latest trunk
In-Reply-To: <191c3a030910080821w691aa78aw272d71dee7675ff1@mail.gmail.com>
References: <27c25bc40910080320x7b3b380eu933797197e334cb8@mail.gmail.com>
	<191c3a030910080821w691aa78aw272d71dee7675ff1@mail.gmail.com>
Message-ID: <27c25bc40910100305q5ae87ddbjc9999852724470d8@mail.gmail.com>

Hi,

My sip phone did authentication ( sending in SIP REGISTER etc ), but it is
still not showing up.  How come?

It was working before.

thanks,
juan

On Thu, Oct 8, 2009 at 11:21 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> they only set if you authenticate on sip or run the set_user app
>
>
> On Thu, Oct 8, 2009 at 5:20 AM, Juan Backson <juanbackson at gmail.com>wrote:
>
>> Hi,
>>
>> My application fails to set the appropriate variables using directory xml
>> after using the latest trunk as of yesterday.
>>
>> My curl looks like:
>>
>>
>>
>> <document type="freeswitch/xml">
>>   <section name="directory">
>>     <domain name="192.168.1.102">
>>       <user id="200002">
>>         <params>
>>           <param name="password" value="123456"></param>
>>           <param name="dial-string"
>> value="sofia/internal/sip:200002 at 192.168.1.11:29440
>> ;rinstance=0245b2a59ddff837"></param>
>>         </params>
>>         <variables>
>>           <variable name="account_id" value="20"></variable>
>>           <variable name="vm-code" value="1000"></variable>
>>         </variables>
>>       </user>
>>     </domain>
>>   </section>
>> </document>
>>
>>
>>
>> In the info app, I am not seeing account-id and vm-code anymore.
>>
>> How to fix that?
>>
>> juan
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tayeb.meftah at gmail.com  Sat Oct 10 05:32:11 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sat, 10 Oct 2009 12:32:11 +0000
Subject: [Freeswitch-users] Anyone intairaisted?
Message-ID: <4AD07ECB.5070602@gmail.com>

dear all
anyone intairaisted in porting/integrating some billing engine/software 
to freeswitch?
we are building a new Community for that
no for billing only but billing is the primary focus
if anyone intairaisted please mail me to:
tayeb.meftah at gmail.com
i'm waiting for you to join!
the web site is not yet ready, is:
http://www.awesomevoip.org
we have also a hosted PBX to do conferencing... but we need a conference 
bridge
thanks!


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4494 (20091009) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com




From testeador01 at gmail.com  Sat Oct 10 06:24:20 2009
From: testeador01 at gmail.com (Milena)
Date: Sat, 10 Oct 2009 08:24:20 -0500
Subject: [Freeswitch-users] how to match '#' in XML dialplan ?
In-Reply-To: <ecd05ce30910100253q1424cd4an9be93ab434912b3f@mail.gmail.com>
References: <ecd05ce30910100253q1424cd4an9be93ab434912b3f@mail.gmail.com>
Message-ID: <c29b930b0910100624n7d5dd257h526d2815705703d3@mail.gmail.com>

escape character is '\'try
  <condition field="destination_number" expression="^1\#(d+)\#(d+)$"/>
2009/10/10 daqiang wang <wangdq.no1 at gmail.com>

> hello every one :
>    I want to match the # in XML dialplan , how to do  ?
>    example :
>       1#5555#6666 . how to do ?
>      I do this :
>      <condition field="destination_number" expression="^1#(d+)#(d+)$"/>
>     but it's not work
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lakindia89 at gmail.com  Sat Oct 10 06:47:00 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Sat, 10 Oct 2009 19:17:00 +0530
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <20091009123615.BC50A3FA6E9@mail.cune.org>
References: <7d79b3930910090407m5dc81d12pf2b0a47f6017804@mail.gmail.com>
	<20091009123615.BC50A3FA6E9@mail.cune.org>
Message-ID: <7d79b3930910100647u7fd11413ucec0a535d68b3e3a@mail.gmail.com>

Hi,
I've tried with the following dialplan(After making the changes that are
recommended). But still I got INVALID_IE_CONTENTS error.

Another thing is, if I use prispan configuration, I'm able to make outgoing
calls.
But if I use the libpri span configuration I'm not able to make outgoing
calls, which says INVALID_IE_CONTENTS.

I've been struck with this problem for the past 1 week.
Any solution to this???


On Fri, Oct 9, 2009 at 6:06 PM, <Russell.Mosemann at cune.org> wrote:

> lakshmanan ganapathy <lakindia89 at gmail.com> said:
>
> > But still I'm facing problem with the outgoing call. It says
> > INVALID_IE_CONTENTS.
> > What might be the issue? Even I tried the following dialplan to call by
> > using bridge.
> > <extension name="outgoing-fxo">
> >                   <condition field="destination_number"
> > expression="^(\d{10})$">
> >                           <action application="answer"/>
> >                           <action application="bridge"
> > data="openzap/1/1/${dialed_ext}"/>
>
> Does "answer" need to be called here? I haven't used an fxo. So, I don't
> know. What value does $dialed_ext have? If you want to use the number
> matched in the condition, then it should be
>
> openzap/1/1/$1
>
> --
> Russell Mosemann
>
>
>
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jmesquita at freeswitch.org  Sat Oct 10 07:30:05 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 10 Oct 2009 11:30:05 -0300
Subject: [Freeswitch-users] Questions regarding to mod_nibble
In-Reply-To: <a60fed5e0910082039yc8b7ef8x24260041a7c11d31@mail.gmail.com>
References: <a60fed5e0910082039yc8b7ef8x24260041a7c11d31@mail.gmail.com>
Message-ID: <ebbc2d550910100730o6abae5b5p254b72cf1c3869c4@mail.gmail.com>

I am testing the latest version of nibblebill so let me see if I can help
you with your questions.

jm

On Fri, Oct 9, 2009 at 12:39 AM, Ahmed Munir <ahmedmunir007 at gmail.com>wrote:

> I want to ask three questions related to mod_nibble bills, as I'm listing
> down below;
>
> 1- Can we select/use dynamic tables for billing using nibble bill?
>
What do you mean for dynamic tables? Like LCR does that you can specify your
own SQL statement to be executed? If that's what you are asking, no, but it
would be a nice todo.


> 2- Can we define more than two tables and attributes in
> nibblebill.conf.xml?
>
What else do you want to define and how do you imagine it to behave?


> 3- As Nibble bill is use to deduct amount of user account, Can we deduct
> minutes instead of cash? Because my case is, if a user buy a package and I
> only want to deducts his/her minutes. How we can resolve it by nibble bill?
> / What other way we can resolve it?
>
When we say cash on the column, we are really saying just a number that is
being deducted, that's it. If you deduct 1 every 60 seconds, you will have
your "cash" converted to minutes, won't you?


>
> Kindly advise soon.
>
> --
> Regards,
>
> Ahmed Munir
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dome at tel.co.th  Sat Oct 10 08:02:42 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Sat, 10 Oct 2009 22:02:42 +0700
Subject: [Freeswitch-users] Questions regarding to mod_nibble
In-Reply-To: <a60fed5e0910082039yc8b7ef8x24260041a7c11d31@mail.gmail.com>
References: <a60fed5e0910082039yc8b7ef8x24260041a7c11d31@mail.gmail.com>
Message-ID: <8ccbff060910100802t71a606aeo42c75008c3054794@mail.gmail.com>

2009/10/9 Ahmed Munir <ahmedmunir007 at gmail.com>:
> I want to ask three questions related to mod_nibble bills, as I'm listing
> down below;
>
> 1- Can we select/use dynamic tables for billing using nibble bill?
> 2- Can we define more than two tables and attributes in nibblebill.conf.xml?
> 3- As Nibble bill is use to deduct amount of user account, Can we deduct
> minutes instead of cash? Because my case is, if a user buy a package and I
> only want to deducts his/her minutes. How we can resolve it by nibble bill?
> / What other way we can resolve it?
My solution (Postgresql 8.4 with postgresql-prefix).
1. modify mod_nibblebill.c  change sqlupdate command to update
duration in my cdr table (insert blank data when start billing)
2. create trigger to deduct balance.


Dome C.
>
> Kindly advise soon.
>
> --
> Regards,
>
> Ahmed Munir
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Sat Oct 10 10:09:04 2009
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 10 Oct 2009 10:09:04 -0700
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <7d79b3930910100647u7fd11413ucec0a535d68b3e3a@mail.gmail.com>
References: <7d79b3930910090407m5dc81d12pf2b0a47f6017804@mail.gmail.com>
	<20091009123615.BC50A3FA6E9@mail.cune.org>
	<7d79b3930910100647u7fd11413ucec0a535d68b3e3a@mail.gmail.com>
Message-ID: <87f2f3b90910101009j1295d812jd98cae91e30b1eff@mail.gmail.com>

Okay, please go to pastebin.freeswitch.org and paste your openzap.conf.xml
file. Also, paste the ENTIRE debug log from a call from start to finish.
Telling us that you see INVALID_IE_CONTENTS doesn't help if we don't know
what the information element contains. Finally, turn on PRI debugging and
make another test call and pastebin that debug log as well. The debug will
show details about the communications between your machine and the carrier.
Instructions for turning on debugging for libpri are found in the OpenZAP
wiki page in the same place where the libpri instructions are located.

Put the pastebin number in this email thread and then we'll go have a look.

Thanks,
MC

On Sat, Oct 10, 2009 at 6:47 AM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Hi,
> I've tried with the following dialplan(After making the changes that are
> recommended). But still I got INVALID_IE_CONTENTS error.
>
> Another thing is, if I use prispan configuration, I'm able to make outgoing
> calls.
> But if I use the libpri span configuration I'm not able to make outgoing
> calls, which says INVALID_IE_CONTENTS.
>
> I've been struck with this problem for the past 1 week.
> Any solution to this???
>
>
>
> On Fri, Oct 9, 2009 at 6:06 PM, <Russell.Mosemann at cune.org> wrote:
>
>> lakshmanan ganapathy <lakindia89 at gmail.com> said:
>>
>> > But still I'm facing problem with the outgoing call. It says
>> > INVALID_IE_CONTENTS.
>> > What might be the issue? Even I tried the following dialplan to call by
>> > using bridge.
>> > <extension name="outgoing-fxo">
>> >                   <condition field="destination_number"
>> > expression="^(\d{10})$">
>> >                           <action application="answer"/>
>> >                           <action application="bridge"
>> > data="openzap/1/1/${dialed_ext}"/>
>>
>> Does "answer" need to be called here? I haven't used an fxo. So, I don't
>> know. What value does $dialed_ext have? If you want to use the number
>> matched in the condition, then it should be
>>
>> openzap/1/1/$1
>>
>> --
>> Russell Mosemann
>>
>>
>>
>> ________________________________________________________
>> Concordia University, Nebraska
>> See http://www.cune.edu/ for the latest news and events!
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Sat Oct 10 10:22:08 2009
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 10 Oct 2009 10:22:08 -0700
Subject: [Freeswitch-users] Difference between park and valet_park
In-Reply-To: <1452e2980910092351v766cad5bpcb55b59ad5fee6d0@mail.gmail.com>
References: <1452e2980910092351v766cad5bpcb55b59ad5fee6d0@mail.gmail.com>
Message-ID: <87f2f3b90910101022v52153d0dhd1ed774af84ecd22@mail.gmail.com>

On Fri, Oct 9, 2009 at 11:51 PM, velusamy velu <velu.technical at gmail.com>wrote:

> Dear All,
>      Could you please any one explain the difference between parking and
> valet parking?
>

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_valet_park

Like the wiki says, valet_park puts calls into numbered parking stalls. A
call that is parked in a numbered stall is retrieved by calling (or
transferring) a second call into that numbered stall. That makes it easy to
pass calls around. The regular park app puts calls in a special park state
that is like being on hold, but the only way to get them out of park is to
uuid_bridge a second leg to the uuid of the parked call leg.

Read the wiki entry on valet_park and then read the one on park and you'll
see that they are similar apps but the valet_park app is a bit easier for
phone users to use for moving calls in and out of the park state.

-MC


> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Sat Oct 10 10:30:50 2009
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 10 Oct 2009 10:30:50 -0700
Subject: [Freeswitch-users] how to match '#' in XML dialplan ?
In-Reply-To: <c29b930b0910100624n7d5dd257h526d2815705703d3@mail.gmail.com>
References: <ecd05ce30910100253q1424cd4an9be93ab434912b3f@mail.gmail.com>
	<c29b930b0910100624n7d5dd257h526d2815705703d3@mail.gmail.com>
Message-ID: <87f2f3b90910101030y7d1142b5kbdac5ffdbc12334b@mail.gmail.com>

Some characters need a backslash to match in a regular expression. However,
# is not one of them. I think your regex is wrong:
<condition field="destination_number" expression="^1#(d+)#(d+)$"/>

It should probably be:
<condition field="destination_number" expression="^1#(\d+)#(\d+)$"/>

Note the backslashes in front of the d+ entries. \d means "match a digit"
whereas a bare d means "make a lowercase d character".

Hope that helps.
-MC

P.S. - The * character does need to be escaped in regexes. See the
default.xml dialplan file for some obvious examples.

On Sat, Oct 10, 2009 at 6:24 AM, Milena <testeador01 at gmail.com> wrote:

> escape character is '\'try
>   <condition field="destination_number" expression="^1\#(d+)\#(d+)$"/>
> 2009/10/10 daqiang wang <wangdq.no1 at gmail.com>
>
>> hello every one :
>>    I want to match the # in XML dialplan , how to do  ?
>>    example :
>>       1#5555#6666 . how to do ?
>>      I do this :
>>      <condition field="destination_number" expression="^1#(d+)#(d+)$"/>
>>     but it's not work
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Sat Oct 10 10:36:09 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 10 Oct 2009 12:36:09 -0500
Subject: [Freeswitch-users] Difference between park and valet_park
In-Reply-To: <87f2f3b90910101022v52153d0dhd1ed774af84ecd22@mail.gmail.com>
References: <1452e2980910092351v766cad5bpcb55b59ad5fee6d0@mail.gmail.com>
	<87f2f3b90910101022v52153d0dhd1ed774af84ecd22@mail.gmail.com>
Message-ID: <191c3a030910101036y64c68a37i5a65bc73ba8aee69@mail.gmail.com>

park is an app you can never return from that just sits idle so you can
remote control the channel.
valet_park is an app to be used from the phones as a way to park and
retrieve calls.



On Sat, Oct 10, 2009 at 12:22 PM, Michael Collins <msc at freeswitch.org>wrote:

>
>
> On Fri, Oct 9, 2009 at 11:51 PM, velusamy velu <velu.technical at gmail.com>wrote:
>
>> Dear All,
>>      Could you please any one explain the difference between parking and
>> valet parking?
>>
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_valet_park
>
> Like the wiki says, valet_park puts calls into numbered parking stalls. A
> call that is parked in a numbered stall is retrieved by calling (or
> transferring) a second call into that numbered stall. That makes it easy to
> pass calls around. The regular park app puts calls in a special park state
> that is like being on hold, but the only way to get them out of park is to
> uuid_bridge a second leg to the uuid of the parked call leg.
>
> Read the wiki entry on valet_park and then read the one on park and you'll
> see that they are similar apps but the valet_park app is a bit easier for
> phone users to use for moving calls in and out of the park state.
>
> -MC
>
>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From moises.silva at gmail.com  Sat Oct 10 10:40:16 2009
From: moises.silva at gmail.com (Moises Silva)
Date: Sat, 10 Oct 2009 13:40:16 -0400
Subject: [Freeswitch-users] libpri_span vs prispans
In-Reply-To: <7d79b3930910092139v3589b761oab0bae46598118f@mail.gmail.com>
References: <7d79b3930910090459v2b07e3aapd715c7618f11ada0@mail.gmail.com>
	<20091009125714.E0FF73043D7@mail.cune.org>
	<87f2f3b90910091057q3a033548j462446cf424ad257@mail.gmail.com>
	<7d79b3930910092139v3589b761oab0bae46598118f@mail.gmail.com>
Message-ID: <c4d05cbe0910101040m4dea6a3bx7e7e2c2c983141ef@mail.gmail.com>

>From libs/openzap/mod_openzap/mod_openzap.c and
libs/openzap/src/ozmod_libpri/ozmod_libpri.c

node=cpe|net
switch=ni1|ni2|dms100|lucent5|5ess|att4ess|4ess|euroisdn|gr303eoc|gr303tmc
dp=international|national|local|isdn|private|unknown
l1=alaw|ulaw <-- this recently is set based on the trunk type, but old
openzap in E1 mode needs to explicitly set it to alaw
debug=q921_raw,q921_dump,q921_state,config,q931_dump,q931_state,q931_anomaly,apdu,acc
none or all
opts=suggest_channel,omit_display,omit_redirecting_number

The following values are commonly known.
context=
dialplan=

On Sat, Oct 10, 2009 at 12:39 AM, lakshmanan ganapathy <lakindia89 at gmail.com
> wrote:

> Where can I found entire configurations for a libpri spans. Right now I've
> only 4 things in my libpri configuration. They are Switchtype, Node,
> Contect, Dialplan, which is given as example in wiki.
>
> But I think libprispan has more configuration. Where can I find those
> things?
>
>
>
> On Fri, Oct 9, 2009 at 11:27 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Fri, Oct 9, 2009 at 5:57 AM, <Russell.Mosemann at cune.org> wrote:
>>
>>> lakshmanan ganapathy <lakindia89 at gmail.com> said:
>>> > What difference it will make, if I use pri_span configuration and
>>> lib_pri
>>> > span configuration in openzap.conf.xml.
>>> > I'm preety much confused on the difference between this.
>>>
>>> Openzap can handle a T1/E1 itself, or openzap can use libpri to do that.
>>> libpri is more mature than openzap. In my case openzap was not able to
>>> handle everything invovling a T1 here, and using libpri fixed it.
>>>
>>> A libpri_span configuration is handled by libpri. A pri_span
>>> configuration is handled natively by openzap. Use the one that works for
>>> you.
>>>
>>
>> Right now the libpri stack is more mature than the ozmod_isdn stuff. See
>> the OpenZAP page on the wiki for instructions on how to set up ozmod_libpri.
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From shaheryarkh at googlemail.com  Sat Oct 10 12:01:04 2009
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Sun, 11 Oct 2009 01:01:04 +0600
Subject: [Freeswitch-users] Questions regarding to mod_nibble
In-Reply-To: <ebbc2d550910100730o6abae5b5p254b72cf1c3869c4@mail.gmail.com>
References: <a60fed5e0910082039yc8b7ef8x24260041a7c11d31@mail.gmail.com>
	<ebbc2d550910100730o6abae5b5p254b72cf1c3869c4@mail.gmail.com>
Message-ID: <b16156850910101201v23b30ffcxec887a7681066079@mail.gmail.com>

I haven't don't any nibble bill project so i am not aware of its
configuration details but here are some of the feature of mod_nibble
(reference to wiki url http://wiki.freeswitch.org/wiki/Mod_nibblebill),

   - Debit credit/cash from accounts real-time
   - Allow for billing at different rates during a single call
   - Allow for warning callers when their balance is low (via audio,
   in-channel)
   - Allow for disconnecting or re-routing calls when balance is depleted
   - Allow billing functions listed above to operate with multiple
   concurrent calls

This means you can have packages where you offer users say X amount of free
minutes and when they are all used up, you can start charging cost Y WITHOUT
dropping user's call. All you need to do is (reference to JM's reply below),

1. Set X in cash column, and deduct 1 per 60 seconds. If you are billing on
different time scale then per minute charges then cash value would be,

Cash = X * 1/T

Where,
X = number of free minutes in package
T = billing interval in minutes

So, if you have 30 free minutes and 30 second billing interval then cash
would be 30 x 1/0.5 = 60.

2. After X minutes have passed during a call (i.e. when cash value becomes
zero), you can change billing rate to whatever you want to charge user on
per minute basis (or whatever your billing interval may be) without dropping
the call (but it would be good to warn user about it before you start
charging him money).

Lastly using different tables schema for mod_nibble may not be a good idea.
I recommend using a standard billing table like mod_nibble already has. You
can of course extend this schema to store addition information of your
choice OR you can use SQL Views to combine different tables and behave as a
single table for any pre-existing reports etc. But do remember that SQL
Views are normally Read Only in nearly all modren DBMS.

Thank you.


2009/10/10 Jo?o Mesquita <jmesquita at freeswitch.org>

> I am testing the latest version of nibblebill so let me see if I can help
> you with your questions.
>
> jm
>
> On Fri, Oct 9, 2009 at 12:39 AM, Ahmed Munir <ahmedmunir007 at gmail.com>wrote:
>
>> I want to ask three questions related to mod_nibble bills, as I'm listing
>> down below;
>>
>> 1- Can we select/use dynamic tables for billing using nibble bill?
>>
> What do you mean for dynamic tables? Like LCR does that you can specify
> your own SQL statement to be executed? If that's what you are asking, no,
> but it would be a nice todo.
>
>
>> 2- Can we define more than two tables and attributes in
>> nibblebill.conf.xml?
>>
> What else do you want to define and how do you imagine it to behave?
>
>
>> 3- As Nibble bill is use to deduct amount of user account, Can we deduct
>> minutes instead of cash? Because my case is, if a user buy a package and I
>> only want to deducts his/her minutes. How we can resolve it by nibble bill?
>> / What other way we can resolve it?
>>
> When we say cash on the column, we are really saying just a number that is
> being deducted, that's it. If you deduct 1 every 60 seconds, you will have
> your "cash" converted to minutes, won't you?
>
>
>>
>> Kindly advise soon.
>>
>> --
>> Regards,
>>
>> Ahmed Munir
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
________________________________________________________
|
                     |
| FATAL ERROR                                                           ---
O X |
|_______________________________________________________|
|                        You have moved the mouse.
 |
| Windows must be restarted for the changes to take effect.   |
|                                        <OK>
               |
####################################/


Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From lon at kickasspixels.com  Sat Oct 10 12:31:58 2009
From: lon at kickasspixels.com (Lon Baker)
Date: Sat, 10 Oct 2009 12:31:58 -0700
Subject: [Freeswitch-users] Music on hold volume?
Message-ID: <5d3e0dc60910101231x5dfb02a0w1e56068dc5ccdd56@mail.gmail.com>

Is there a way to adjust the volume of music on hold?

Lon


From brian at freeswitch.org  Sat Oct 10 12:47:28 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 10 Oct 2009 14:47:28 -0500
Subject: [Freeswitch-users] Music on hold volume?
In-Reply-To: <5d3e0dc60910101231x5dfb02a0w1e56068dc5ccdd56@mail.gmail.com>
References: <5d3e0dc60910101231x5dfb02a0w1e56068dc5ccdd56@mail.gmail.com>
Message-ID: <A1B37588-4AE5-43C6-9BCA-56A5F4F3D91C@freeswitch.org>

adjust the files themselves.

/b

On Oct 10, 2009, at 2:31 PM, Lon Baker wrote:

> Is there a way to adjust the volume of music on hold?
>
> Lon



From eric at rf.com  Sat Oct 10 13:33:28 2009
From: eric at rf.com (Eric Chamberlain)
Date: Sat, 10 Oct 2009 13:33:28 -0700
Subject: [Freeswitch-users] Gateways,
	configuration or directory with mod_xml_curl
Message-ID: <5232A6B3-8104-4AA5-918F-E951DF023E22@rf.com>

Hello,

We are looking at Freeswitch to solve a problem of ours, we have  
thousands of users with individual gateway information.  The user's  
will not register with us, but we need to register with their gateways  
on their behalf.

Since users will be constantly adding/changing/deleting gateways, we  
figure mox_xml_curl is the best way to maintain the configuration.

What we are not sure of is if we should be putting the gateway  
information in the directory entry for each user (gateways are not  
shared between users) or if the gateway should go in the configuration.

If the gateway configuration is in the directory, how does sofia know  
when the gateway configuration changes?  The reloadxml documentation  
doesn't talk about gateways in the directory, does it work with the  
directory.  If so, is it one curl call or one call per user?  Is there  
a way to tell sofia that only one user's gateway configuration has  
changed?

Likewise with gateways in the configuration.  Does reloadxml generate  
one curl call or many?

In both cases, is there a way to minimize parsing the data that has  
not changed?

--
Eric Chamberlain






From juanbackson at gmail.com  Sat Oct 10 16:02:38 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Sun, 11 Oct 2009 07:02:38 +0800
Subject: [Freeswitch-users] directory variables not set correct with the
	latest trunk
In-Reply-To: <27c25bc40910100305q5ae87ddbjc9999852724470d8@mail.gmail.com>
References: <27c25bc40910080320x7b3b380eu933797197e334cb8@mail.gmail.com>
	<191c3a030910080821w691aa78aw272d71dee7675ff1@mail.gmail.com>
	<27c25bc40910100305q5ae87ddbjc9999852724470d8@mail.gmail.com>
Message-ID: <27c25bc40910101602p523f472fp390d2cc8dbc23a47@mail.gmail.com>

Hi,

I am still stuck in trying to get curl directory variables to show up in
channel even after the user has registered.

Is this something that gets changed in the latest trunk or is it just my
mis-configuration?

please help.

jb

On Sat, Oct 10, 2009 at 6:05 PM, Juan Backson <juanbackson at gmail.com> wrote:

> Hi,
>
> My sip phone did authentication ( sending in SIP REGISTER etc ), but it is
> still not showing up.  How come?
>
> It was working before.
>
> thanks,
> juan
>
>
> On Thu, Oct 8, 2009 at 11:21 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> they only set if you authenticate on sip or run the set_user app
>>
>>
>>   On Thu, Oct 8, 2009 at 5:20 AM, Juan Backson <juanbackson at gmail.com>wrote:
>>
>>>  Hi,
>>>
>>> My application fails to set the appropriate variables using directory xml
>>> after using the latest trunk as of yesterday.
>>>
>>> My curl looks like:
>>>
>>>
>>>
>>> <document type="freeswitch/xml">
>>>   <section name="directory">
>>>     <domain name="192.168.1.102">
>>>       <user id="200002">
>>>         <params>
>>>           <param name="password" value="123456"></param>
>>>           <param name="dial-string"
>>> value="sofia/internal/sip:200002 at 192.168.1.11:29440
>>> ;rinstance=0245b2a59ddff837"></param>
>>>         </params>
>>>         <variables>
>>>           <variable name="account_id" value="20"></variable>
>>>           <variable name="vm-code" value="1000"></variable>
>>>         </variables>
>>>       </user>
>>>     </domain>
>>>   </section>
>>> </document>
>>>
>>>
>>>
>>> In the info app, I am not seeing account-id and vm-code anymore.
>>>
>>> How to fix that?
>>>
>>> juan
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From brian at freeswitch.org  Sat Oct 10 17:02:57 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 10 Oct 2009 19:02:57 -0500
Subject: [Freeswitch-users] directory variables not set correct with the
	latest trunk
In-Reply-To: <27c25bc40910101602p523f472fp390d2cc8dbc23a47@mail.gmail.com>
References: <27c25bc40910080320x7b3b380eu933797197e334cb8@mail.gmail.com>
	<191c3a030910080821w691aa78aw272d71dee7675ff1@mail.gmail.com>
	<27c25bc40910100305q5ae87ddbjc9999852724470d8@mail.gmail.com>
	<27c25bc40910101602p523f472fp390d2cc8dbc23a47@mail.gmail.com>
Message-ID: <675B4EDC-AC25-40C0-A88D-A83AAE4B4BCF@freeswitch.org>

It happens on invite during a call not on REGISTER... I can 100%  
confirm it works so I do not know what the heck you're doing.

/b

On Oct 10, 2009, at 6:02 PM, Juan Backson wrote:

> Hi,
>
> I am still stuck in trying to get curl directory variables to show  
> up in channel even after the user has registered.
>
> Is this something that gets changed in the latest trunk or is it  
> just my mis-configuration?
>
> please help.
>
> jb



From brian at freeswitch.org  Sat Oct 10 17:07:52 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 10 Oct 2009 19:07:52 -0500
Subject: [Freeswitch-users] directory variables not set correct with the
	latest trunk
In-Reply-To: <27c25bc40910101602p523f472fp390d2cc8dbc23a47@mail.gmail.com>
References: <27c25bc40910080320x7b3b380eu933797197e334cb8@mail.gmail.com>
	<191c3a030910080821w691aa78aw272d71dee7675ff1@mail.gmail.com>
	<27c25bc40910100305q5ae87ddbjc9999852724470d8@mail.gmail.com>
	<27c25bc40910101602p523f472fp390d2cc8dbc23a47@mail.gmail.com>
Message-ID: <137DBF84-F0E5-41C6-B327-4C81A20FB4F6@freeswitch.org>

Also check to make sure you're returning the correct format as per the  
XML curl page.

/b

On Oct 10, 2009, at 6:02 PM, Juan Backson wrote:

> Hi,
>
> I am still stuck in trying to get curl directory variables to show  
> up in channel even after the user has registered.
>
> Is this something that gets changed in the latest trunk or is it  
> just my mis-configuration?
>
> please help.
>
> jb



From wangdq.no1 at gmail.com  Sat Oct 10 19:45:17 2009
From: wangdq.no1 at gmail.com (daqiang wang)
Date: Sun, 11 Oct 2009 10:45:17 +0800
Subject: [Freeswitch-users] how to match '#' in XML dialplan ?
In-Reply-To: <87f2f3b90910101030y7d1142b5kbdac5ffdbc12334b@mail.gmail.com>
References: <ecd05ce30910100253q1424cd4an9be93ab434912b3f@mail.gmail.com>
	<c29b930b0910100624n7d5dd257h526d2815705703d3@mail.gmail.com>
	<87f2f3b90910101030y7d1142b5kbdac5ffdbc12334b@mail.gmail.com>
Message-ID: <ecd05ce30910101945k4399256cp9bfb800424eb1654@mail.gmail.com>

it's work . Thank you very much .

2009/10/11 Michael Collins <msc at freeswitch.org>

> Some characters need a backslash to match in a regular expression. However,
> # is not one of them. I think your regex is wrong:
> <condition field="destination_number" expression="^1#(d+)#(d+)$"/>
>
> It should probably be:
> <condition field="destination_number" expression="^1#(\d+)#(\d+)$"/>
>
> Note the backslashes in front of the d+ entries. \d means "match a digit"
> whereas a bare d means "make a lowercase d character".
>
> Hope that helps.
> -MC
>
> P.S. - The * character does need to be escaped in regexes. See the
> default.xml dialplan file for some obvious examples.
>
>
> On Sat, Oct 10, 2009 at 6:24 AM, Milena <testeador01 at gmail.com> wrote:
>
>> escape character is '\'try
>>   <condition field="destination_number" expression="^1\#(d+)\#(d+)$"/>
>> 2009/10/10 daqiang wang <wangdq.no1 at gmail.com>
>>
>>> hello every one :
>>>    I want to match the # in XML dialplan , how to do  ?
>>>    example :
>>>       1#5555#6666 . how to do ?
>>>      I do this :
>>>      <condition field="destination_number" expression="^1#(d+)#(d+)$"/>
>>>     but it's not work
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mayamatakeshi at gmail.com  Sat Oct 10 20:18:39 2009
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Sun, 11 Oct 2009 12:18:39 +0900
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Update: Valet
	Parking
In-Reply-To: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>
References: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>
Message-ID: <15b9404e0910102018oda8834ah8cc87787ee00c22b@mail.gmail.com>

On Fri, Oct 9, 2009 at 10:42 AM, Michael Collins <msc at freeswitch.org> wrote:

> FYI,
>
> The FreeSWITCH devs have added valet parking! Check it out:
> http://www.freeswitch.org/node/207
>
> Let us know what you think.
>

Very nice.

But I think a valet_unpark app is missing.
If the intention of the person sent to the valet lot is to retrieve a call
there, the person can assume the call was already retrieved by someone else
or that the caller hung up if he/she hears MOH. But it would be nicer to
have a valet_unpark app to fail and let the dialplan play a message.

regards,
takeshi
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From mayamatakeshi at gmail.com  Sat Oct 10 21:26:43 2009
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Sun, 11 Oct 2009 13:26:43 +0900
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Update: Valet
	Parking
In-Reply-To: <15b9404e0910102018oda8834ah8cc87787ee00c22b@mail.gmail.com>
References: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>
	<15b9404e0910102018oda8834ah8cc87787ee00c22b@mail.gmail.com>
Message-ID: <15b9404e0910102126g67697544ie6091ed3d326de4f@mail.gmail.com>

On Sun, Oct 11, 2009 at 12:18 PM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:

>
>
> On Fri, Oct 9, 2009 at 10:42 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> FYI,
>>
>> The FreeSWITCH devs have added valet parking! Check it out:
>> http://www.freeswitch.org/node/207
>>
>> Let us know what you think.
>>
>
> Very nice.
>
> But I think a valet_unpark app is missing.
> If the intention of the person sent to the valet lot
>

I meant "parking stall"


> is to retrieve a call there, the person can assume the call was already
> retrieved by someone else or that the caller hung up if he/she hears MOH.
> But it would be nicer to have a valet_unpark app to fail and let the
> dialplan play a message.
>
> regards,
> takeshi
>
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From red.rain.seven at gmail.com  Sun Oct 11 03:10:07 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Sun, 11 Oct 2009 18:10:07 +0800
Subject: [Freeswitch-users] how to match '#' in XML dialplan ?
In-Reply-To: <ecd05ce30910101945k4399256cp9bfb800424eb1654@mail.gmail.com>
References: <ecd05ce30910100253q1424cd4an9be93ab434912b3f@mail.gmail.com>
	<c29b930b0910100624n7d5dd257h526d2815705703d3@mail.gmail.com>
	<87f2f3b90910101030y7d1142b5kbdac5ffdbc12334b@mail.gmail.com>
	<ecd05ce30910101945k4399256cp9bfb800424eb1654@mail.gmail.com>
Message-ID: <59ad9ca10910110310hc271115ja11626bc119d3e25@mail.gmail.com>

Daqiang:


How do you make your IP phone not dial right after you press "#"? Usually
the IP phone will dial the number already once you pushed "#"




On Sun, Oct 11, 2009 at 10:45 AM, daqiang wang <wangdq.no1 at gmail.com> wrote:

> it's work . Thank you very much .
>
> 2009/10/11 Michael Collins <msc at freeswitch.org>
>
>> Some characters need a backslash to match in a regular expression.
>> However, # is not one of them. I think your regex is wrong:
>> <condition field="destination_number" expression="^1#(d+)#(d+)$"/>
>>
>> It should probably be:
>> <condition field="destination_number" expression="^1#(\d+)#(\d+)$"/>
>>
>> Note the backslashes in front of the d+ entries. \d means "match a digit"
>> whereas a bare d means "make a lowercase d character".
>>
>> Hope that helps.
>> -MC
>>
>> P.S. - The * character does need to be escaped in regexes. See the
>> default.xml dialplan file for some obvious examples.
>>
>>
>> On Sat, Oct 10, 2009 at 6:24 AM, Milena <testeador01 at gmail.com> wrote:
>>
>>> escape character is '\'try
>>>   <condition field="destination_number" expression="^1\#(d+)\#(d+)$"/>
>>> 2009/10/10 daqiang wang <wangdq.no1 at gmail.com>
>>>
>>>> hello every one :
>>>>    I want to match the # in XML dialplan , how to do  ?
>>>>    example :
>>>>       1#5555#6666 . how to do ?
>>>>      I do this :
>>>>      <condition field="destination_number" expression="^1#(d+)#(d+)$"/>
>>>>     but it's not work
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From ivdreg at gmail.com  Sun Oct 11 05:05:45 2009
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Sun, 11 Oct 2009 15:05:45 +0300
Subject: [Freeswitch-users] More complex hunting question
Message-ID: <b437a0e0910110505s20a176b0vf78603aa9d7e9d78@mail.gmail.com>

Dear All,

I'm trying to use FS with xml_curl as routing server for otgoing calls to my
provides. Here is simplified setup:

[REGISTRARs] ----> [APP SERVERs] ----> [ROUTING SERVER/FreeSwitch] ---->
[SBC/no Transcoder]----> [Terminating GWs/ITSPs]

Because of simplicity I do not use hunting functionality in SBC but only in
FS.
I try to avoid if possible any trascoding but there are cases that it is
necessary. I want ROUTING SERVER to be
configured in proxy_media mode or bypass_media but on some errors such as
"415 Unsupported Media" that supposed are codec
negotiation issue to hunt again via trascoder before SBC.

Can someone help to find some elegant way functionality described below to
be achieved via dialplan_xml ?
Call to +XXXXXXXXXXXXXXXX for example:
available routes by priority are via: ITSP1,ITSP5,ITSP2,ITSP7
outgoing leg to ITSP1 - (returns unallocated number)
outgoing leg to ITSP5 - (returns "415 Unsupported Media")
    new outgoing leg to ITSP5 again but via transcoder - (If call is
connected OK, If other error hunt again to next GW - ITSP2)
outgoing leg to ITSP2 - (returns "415 Unsupported Media")
    new outgoing leg to ITSP2 again but via transcoder - (If call is
connected OK, If other error hunt again to next GW - ITSP7)
outgoing leg to ITSP7
Remarks: Codec errors may be more that one (415 Unsupported Media/488 Not
Acceptable Here ...) and number it ITSP/GW in not fixed.


Thanks to all.
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From tculjaga at gmail.com  Sun Oct 11 09:50:00 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Sun, 11 Oct 2009 18:50:00 +0200
Subject: [Freeswitch-users] how to match '#' in XML dialplan ?
In-Reply-To: <59ad9ca10910110310hc271115ja11626bc119d3e25@mail.gmail.com>
References: <ecd05ce30910100253q1424cd4an9be93ab434912b3f@mail.gmail.com>
	<c29b930b0910100624n7d5dd257h526d2815705703d3@mail.gmail.com>
	<87f2f3b90910101030y7d1142b5kbdac5ffdbc12334b@mail.gmail.com>
	<ecd05ce30910101945k4399256cp9bfb800424eb1654@mail.gmail.com>
	<59ad9ca10910110310hc271115ja11626bc119d3e25@mail.gmail.com>
Message-ID: <65d96fc80910110950n61738c06jb59ca4b91d98c4bb@mail.gmail.com>

this is up to your phone.... # means address complete and you phone sends
the number you dialed into an INVITE message.

if you want to support FAC with # you should modify the phone's dialplan and
make it expect more digits... for certain prefixes.

T.


On Sun, Oct 11, 2009 at 12:10 PM, Henry Huang <red.rain.seven at gmail.com>wrote:

> Daqiang:
>
>
> How do you make your IP phone not dial right after you press "#"? Usually
> the IP phone will dial the number already once you pushed "#"
>
>
>
>
>
> On Sun, Oct 11, 2009 at 10:45 AM, daqiang wang <wangdq.no1 at gmail.com>wrote:
>
>> it's work . Thank you very much .
>>
>> 2009/10/11 Michael Collins <msc at freeswitch.org>
>>
>>> Some characters need a backslash to match in a regular expression.
>>> However, # is not one of them. I think your regex is wrong:
>>> <condition field="destination_number" expression="^1#(d+)#(d+)$"/>
>>>
>>> It should probably be:
>>> <condition field="destination_number" expression="^1#(\d+)#(\d+)$"/>
>>>
>>> Note the backslashes in front of the d+ entries. \d means "match a digit"
>>> whereas a bare d means "make a lowercase d character".
>>>
>>> Hope that helps.
>>> -MC
>>>
>>> P.S. - The * character does need to be escaped in regexes. See the
>>> default.xml dialplan file for some obvious examples.
>>>
>>>
>>> On Sat, Oct 10, 2009 at 6:24 AM, Milena <testeador01 at gmail.com> wrote:
>>>
>>>> escape character is '\'try
>>>>   <condition field="destination_number" expression="^1\#(d+)\#(d+)$"/>
>>>> 2009/10/10 daqiang wang <wangdq.no1 at gmail.com>
>>>>
>>>>> hello every one :
>>>>>    I want to match the # in XML dialplan , how to do  ?
>>>>>    example :
>>>>>       1#5555#6666 . how to do ?
>>>>>      I do this :
>>>>>      <condition field="destination_number" expression="^1#(d+)#(d+)$"/>
>>>>>     but it's not work
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Henry Huang
> UniC Solution - Communication Unified
> VoIP & Open Source software Consultant
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From shiyanov at gmail.com  Sun Oct 11 09:59:09 2009
From: shiyanov at gmail.com (Artem Shiyanov)
Date: Sun, 11 Oct 2009 20:59:09 +0400
Subject: [Freeswitch-users] mod_socket: custom timeout for HEARTBEAT event
Message-ID: <d211f54a0910110959y3d3deed7s43201a3e39756585@mail.gmail.com>

Hi all!
As it stays in wiki:
...
HEARTBEAT
Status information for freeswitch trigerred by freeswitch's heartbeat every
20 seconds.
...

Is there any way to customize timeout of HEARTBEAT events?


Thanks in advance,
Artem
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From edpimentl at gmail.com  Sun Oct 11 10:48:47 2009
From: edpimentl at gmail.com (EdPimentl)
Date: Sun, 11 Oct 2009 13:48:47 -0400
Subject: [Freeswitch-users] Is assistivetech.net confusing the market place
	with their use of FreeSwitch product name?
Message-ID: <9dc4a1670910111048y2802b155oc7086b975193c9d7@mail.gmail.com>

http://www.assistivetech.net/search/productDisplay.php?product_id=18854
-E
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From herman.griffin at gmail.com  Sun Oct 11 12:12:08 2009
From: herman.griffin at gmail.com (frek818)
Date: Sun, 11 Oct 2009 12:12:08 -0700 (PDT)
Subject: [Freeswitch-users] problem compiling esl for use with freepbx v3
In-Reply-To: <20090823185152.D17845FE@sinclaire.sibble.net>
References: <20090823185152.D17845FE@sinclaire.sibble.net>
Message-ID: <25846572.post@talk.nabble.com>


Did anyone find a solution to this problem? I too would like to install the
esl module for PHP.

Herman

Harondel J. Sibble wrote:
> 
> Okay, just finished installing 1.0.4 from tarball on Ubuntu 9.0.4 server, 
> then went to install FreePBX v3, I've gotten all the prerequisities in the 
> wizard fixed except for ESL
> 
> As per
> 
> http://wiki.freeswitch.org/wiki/Event_Socket_Library
> http://wiki.freeswitch.org/wiki/Event_Socket
> 
> I go into my FS source dir
> 
> /home/sibbleh/freeswitch-1.0.4/libs/esl
> 
> Run make and then "sudo make phpmod-install" 
> 
> and I get 
> 
> 
> $ sudo make phpmod-install
> make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-
> I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g
> -ggdb 
> -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-
> variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes"
> CXXFLAGS="-
> I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g
> -ggdb 
> -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" 
> CXX_CFLAGS="" -C php
> make[1]: Entering directory `/home/sibbleh/freeswitch-1.0.4/libs/esl/php'
> g++  -I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE
> -g 
> -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable -
> I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM -
> I/usr/include/php5/Zend -I/usr/include/php5/ext -
> I/usr/include/php5/ext/date/lib -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64
> -
> Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o
> cc1plus: warnings being treated as errors
> esl_wrap.cpp: In function 'void _wrap_ESLevent_event_set(int, zval*,
> zval**, 
> zval*, int)':
> esl_wrap.cpp:1047: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_event_get(int, zval*,
> zval**, 
> zval*, int)':
> esl_wrap.cpp:1073: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_serialized_string_set(int, 
> zval*, zval**, zval*, int)':
> esl_wrap.cpp:1111: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_serialized_string_get(int, 
> zval*, zval**, zval*, int)':
> esl_wrap.cpp:1141: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_mine_set(int, zval*,
> zval**, 
> zval*, int)':
> esl_wrap.cpp:1172: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_mine_get(int, zval*,
> zval**, 
> zval*, int)':
> esl_wrap.cpp:1198: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_0(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1234: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_1(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1269: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_2(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1294: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_new_ESLevent(int, zval*, zval**,
> zval*, 
> int)':
> esl_wrap.cpp:1346: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_serialize(int, zval*,
> zval**, 
> zval*, int)':
> esl_wrap.cpp:1403: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_setPriority(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1441: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_getHeader(int, zval*,
> zval**, 
> zval*, int)':
> esl_wrap.cpp:1478: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_getBody(int, zval*, zval**, 
> zval*, int)':
> esl_wrap.cpp:1508: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_getType(int, zval*, zval**, 
> zval*, int)':
> esl_wrap.cpp:1538: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_addBody(int, zval*, zval**, 
> zval*, int)':
> esl_wrap.cpp:1571: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_addHeader(int, zval*,
> zval**, 
> zval*, int)':
> esl_wrap.cpp:1611: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_delHeader(int, zval*,
> zval**, 
> zval*, int)':
> esl_wrap.cpp:1644: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_firstHeader(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1674: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_nextHeader(int, zval*,
> zval**, 
> zval*, int)':
> esl_wrap.cpp:1704: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_new_ESLconnection__SWIG_0(int,
> zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1744: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_new_ESLconnection__SWIG_1(int,
> zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1770: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_new_ESLconnection(int, zval*,
> zval**, 
> zval*, int)':
> esl_wrap.cpp:1803: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_socketDescriptor(int, 
> zval*, zval**, zval*, int)':
> esl_wrap.cpp:1846: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_connected(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1872: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_getInfo(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1898: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_send(int, zval*,
> zval**, 
> zval*, int)':
> esl_wrap.cpp:1931: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_sendRecv(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1964: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_api(int, zval*,
> zval**, 
> zval*, int)':
> esl_wrap.cpp:2007: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_bgapi(int, zval*,
> zval**, 
> zval*, int)':
> esl_wrap.cpp:2050: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_sendEvent(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:2082: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_recvEvent(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:2108: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_recvEventTimed(int, 
> zval*, zval**, zval*, int)':
> esl_wrap.cpp:2141: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_filter(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:2181: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_events(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:2221: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_execute(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:2272: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_executeAsync(int,
> zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:2323: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_setAsyncExecute(int, 
> zval*, zval**, zval*, int)':
> esl_wrap.cpp:2356: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_setEventLock(int,
> zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:2389: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_disconnect(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:2415: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_eslSetLogLevel(int, zval*, zval**, 
> zval*, int)':
> esl_wrap.cpp:2438: error: format not a string literal and no format
> arguments
> make[1]: *** [esl_wrap.o] Error 1
> make[1]: Leaving directory `/home/sibbleh/freeswitch-1.0.4/libs/esl/php'
> make: *** [phpmod] Error 2
> 
> Same thing happens if I try sudo make everymod
> 
> Checking the list archives I found this thread
> 
> http://www.nabble.com/ESL-Wrapper-td22209991.html#a22222338
> 
> I've made sure that the php-dev packages are installed. Any suggestions on 
> what to do next?
> -- 
> Harondel J. Sibble 
> Sibble Computer Consulting
> Creating Solutions for the small and medium business computer user.
> help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
> (604) 739-3709 (voice)
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/problem-compiling-esl-for-use-with-freepbx-v3-tp25106337p25846572.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From dave at 3c.co.uk  Sun Oct 11 12:33:54 2009
From: dave at 3c.co.uk (David Knell)
Date: Sun, 11 Oct 2009 12:33:54 -0700
Subject: [Freeswitch-users] Gateways,
	configuration or directory with	mod_xml_curl
In-Reply-To: <5232A6B3-8104-4AA5-918F-E951DF023E22@rf.com>
References: <5232A6B3-8104-4AA5-918F-E951DF023E22@rf.com>
Message-ID: <1255289634.5451.19.camel@localhost.localdomain>

Hi Eric -

The way that we do it is to keep each gateway in its own Sofia profile.
Issuing 
api sofia profile <profile> start reloadxml
does one call to the web server for that profile's XML, which can be
pretty compact if it just contains one gateway.

--Dave

> Hello,
> 
> We are looking at Freeswitch to solve a problem of ours, we have  
> thousands of users with individual gateway information.  The user's  
> will not register with us, but we need to register with their gateways  
> on their behalf.
> 
> Since users will be constantly adding/changing/deleting gateways, we  
> figure mox_xml_curl is the best way to maintain the configuration.
> 
> What we are not sure of is if we should be putting the gateway  
> information in the directory entry for each user (gateways are not  
> shared between users) or if the gateway should go in the configuration.
> 
> If the gateway configuration is in the directory, how does sofia know  
> when the gateway configuration changes?  The reloadxml documentation  
> doesn't talk about gateways in the directory, does it work with the  
> directory.  If so, is it one curl call or one call per user?  Is there  
> a way to tell sofia that only one user's gateway configuration has  
> changed?
> 
> Likewise with gateways in the configuration.  Does reloadxml generate  
> one curl call or many?
> 
> In both cases, is there a way to minimize parsing the data that has  
> not changed?
> 
> --
> Eric Chamberlain
> 
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From mike at jerris.com  Sun Oct 11 14:34:30 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Oct 2009 17:34:30 -0400
Subject: [Freeswitch-users] mod_sofia.c registered calls how to know
In-Reply-To: <f8af5740910070748g1efee71ft5841b6fa4d2412c8@mail.gmail.com>
References: <f8af5740910070748g1efee71ft5841b6fa4d2412c8@mail.gmail.com>
Message-ID: <14A1216A-61C3-4C03-B710-57D2E7F4CD7A@jerris.com>


On Oct 7, 2009, at 10:48 AM, srinivasula reddy wrote:

>
> Hi
>
> can any please tell me where registered calls are stored, so when  
> incoming call came to mod_sofia.c how it will check it is registered  
> or not?\\
>


Calls are not registered and calls have nothing to do with  
registration.  Users are registered so that you may send calls to  
them.  Registration data is stored either in a sqlite database, or  
optionally if you setup odbc, in another database of your choice.  If  
you try to send a call to an unregistered user in the dialplan using  
the proper syntax to send calls to registered users (see the wiki for  
more details), and that user is not registered, the bridge app will  
fail, optionally letting you continue on in the dialplan based on  
variables such as continue_on_fail and hangup_after_bridge.  You can  
use the sofia_contact function to see if there is anyone registered to  
a specific user.

Mike

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From herman.griffin at gmail.com  Sun Oct 11 14:36:21 2009
From: herman.griffin at gmail.com (Herman Griffin)
Date: Sun, 11 Oct 2009 14:36:21 -0700
Subject: [Freeswitch-users] problem compiling esl for use with freepbx v3
In-Reply-To: <25846572.post@talk.nabble.com>
References: <20090823185152.D17845FE@sinclaire.sibble.net>
	<25846572.post@talk.nabble.com>
Message-ID: <4d6f26b0910111436h416121b7s2a4b5c79ffc01db0@mail.gmail.com>

Although probably not the best solution, I figured out a way to make it
compile and install:

I removed all of the -Werror instances in
PATH_TO_FREESWITCH_SOURCE/libs/esl/Makefile

If I was a hardcore c/c++ programmer, I'd figure out the real problem.

Herman aka frek818

On Sun, Oct 11, 2009 at 12:12 PM, frek818 <herman.griffin at gmail.com> wrote:

>
> Did anyone find a solution to this problem? I too would like to install the
> esl module for PHP.
>
> Herman
>
> Harondel J. Sibble wrote:
> >
> > Okay, just finished installing 1.0.4 from tarball on Ubuntu 9.0.4 server,
> > then went to install FreePBX v3, I've gotten all the prerequisities in
> the
> > wizard fixed except for ESL
> >
> > As per
> >
> > http://wiki.freeswitch.org/wiki/Event_Socket_Library
> > http://wiki.freeswitch.org/wiki/Event_Socket
> >
> > I go into my FS source dir
> >
> > /home/sibbleh/freeswitch-1.0.4/libs/esl
> >
> > Run make and then "sudo make phpmod-install"
> >
> > and I get
> >
> >
> > $ sudo make phpmod-install
> > make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-
> > I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g
> > -ggdb
> > -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-
> > variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes"
> > CXXFLAGS="-
> > I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g
> > -ggdb
> > -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable"
> > CXX_CFLAGS="" -C php
> > make[1]: Entering directory `/home/sibbleh/freeswitch-1.0.4/libs/esl/php'
> > g++  -I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include
> -DHAVE_EDITLINE
> > -g
> > -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable
> -
> > I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM -
> > I/usr/include/php5/Zend -I/usr/include/php5/ext -
> > I/usr/include/php5/ext/date/lib -D_LARGEFILE_SOURCE
> -D_FILE_OFFSET_BITS=64
> > -
> > Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o
> > cc1plus: warnings being treated as errors
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_event_set(int, zval*,
> > zval**,
> > zval*, int)':
> > esl_wrap.cpp:1047: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_event_get(int, zval*,
> > zval**,
> > zval*, int)':
> > esl_wrap.cpp:1073: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_serialized_string_set(int,
> > zval*, zval**, zval*, int)':
> > esl_wrap.cpp:1111: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_serialized_string_get(int,
> > zval*, zval**, zval*, int)':
> > esl_wrap.cpp:1141: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_mine_set(int, zval*,
> > zval**,
> > zval*, int)':
> > esl_wrap.cpp:1172: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_mine_get(int, zval*,
> > zval**,
> > zval*, int)':
> > esl_wrap.cpp:1198: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_0(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1234: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_1(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1269: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_2(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1294: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_new_ESLevent(int, zval*, zval**,
> > zval*,
> > int)':
> > esl_wrap.cpp:1346: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_serialize(int, zval*,
> > zval**,
> > zval*, int)':
> > esl_wrap.cpp:1403: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_setPriority(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1441: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_getHeader(int, zval*,
> > zval**,
> > zval*, int)':
> > esl_wrap.cpp:1478: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_getBody(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1508: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_getType(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1538: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_addBody(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1571: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_addHeader(int, zval*,
> > zval**,
> > zval*, int)':
> > esl_wrap.cpp:1611: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_delHeader(int, zval*,
> > zval**,
> > zval*, int)':
> > esl_wrap.cpp:1644: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_firstHeader(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1674: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_nextHeader(int, zval*,
> > zval**,
> > zval*, int)':
> > esl_wrap.cpp:1704: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_new_ESLconnection__SWIG_0(int,
> > zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1744: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_new_ESLconnection__SWIG_1(int,
> > zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1770: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_new_ESLconnection(int, zval*,
> > zval**,
> > zval*, int)':
> > esl_wrap.cpp:1803: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_socketDescriptor(int,
> > zval*, zval**, zval*, int)':
> > esl_wrap.cpp:1846: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_connected(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1872: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_getInfo(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1898: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_send(int, zval*,
> > zval**,
> > zval*, int)':
> > esl_wrap.cpp:1931: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_sendRecv(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1964: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_api(int, zval*,
> > zval**,
> > zval*, int)':
> > esl_wrap.cpp:2007: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_bgapi(int, zval*,
> > zval**,
> > zval*, int)':
> > esl_wrap.cpp:2050: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_sendEvent(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2082: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_recvEvent(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2108: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_recvEventTimed(int,
> > zval*, zval**, zval*, int)':
> > esl_wrap.cpp:2141: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_filter(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2181: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_events(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2221: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_execute(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2272: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_executeAsync(int,
> > zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2323: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_setAsyncExecute(int,
> > zval*, zval**, zval*, int)':
> > esl_wrap.cpp:2356: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_setEventLock(int,
> > zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2389: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_disconnect(int,
> zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2415: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_eslSetLogLevel(int, zval*, zval**,
> > zval*, int)':
> > esl_wrap.cpp:2438: error: format not a string literal and no format
> > arguments
> > make[1]: *** [esl_wrap.o] Error 1
> > make[1]: Leaving directory `/home/sibbleh/freeswitch-1.0.4/libs/esl/php'
> > make: *** [phpmod] Error 2
> >
> > Same thing happens if I try sudo make everymod
> >
> > Checking the list archives I found this thread
> >
> > http://www.nabble.com/ESL-Wrapper-td22209991.html#a22222338
> >
> > I've made sure that the php-dev packages are installed. Any suggestions
> on
> > what to do next?
> > --
> > Harondel J. Sibble
> > Sibble Computer Consulting
> > Creating Solutions for the small and medium business computer user.
> > help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
> > (604) 739-3709 (voice)
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> --
> View this message in context:
> http://www.nabble.com/problem-compiling-esl-for-use-with-freepbx-v3-tp25106337p25846572.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From diego.viola at gmail.com  Sun Oct 11 14:40:15 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 11 Oct 2009 21:40:15 +0000
Subject: [Freeswitch-users] REMINDER: Weekly FreeSWITCH Conference
	Scheduled for October 9, 11AM CST (GMT -6)
In-Reply-To: <87f2f3b90910081202m5dd3f567v240541dd2009c9a8@mail.gmail.com>
References: <87f2f3b90910081202m5dd3f567v240541dd2009c9a8@mail.gmail.com>
Message-ID: <86a32abc0910111440n184b882fwee29736bb15850ef@mail.gmail.com>

I'd like to add this for the next weekly conference.

I have added a few events to the event list, as you can see here:

http://wiki.freeswitch.org/wiki/Event_list

But I need more help from the community to complete that and add content to
the events, etc. So if you can add that for the next weekly conference it
would be really nice :).

Thanks,

Diego

On Thu, Oct 8, 2009 at 7:02 PM, Michael Collins <msc at freeswitch.org> wrote:

> Just a friendly reminder that we will be having the weekly FreeSWITCH
> conference call tomorrow, October 9th, at 11AM Central. The agenda is
> updated:
> http://bit.ly/lzEYy
>
> The conference lasts for six hours, so feel free to dial in at any time. We
> usually start going over the agenda about 15 minutes after the conference
> starts. The agenda itself only takes about an hour or two, after which we
> spend most of the time discussing whatever is on the minds of the community
> members. Remember, the core FreeSWITCH development team of Anthony Minessale
> (primary author, lead architect), Mike Jerris (build master), and Brian West
> (all around VoIP expert, FS configuration guru) are present for the whole
> conference. This is a great time to ask questions about the project. Please
> join us!
>
> Call in options include SIP, Skype, PSTN, and Jabber. Please see the agenda
> for more information.
>
> Looking forward to speaking with you all tomorrow,
> Michael
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From diego.viola at gmail.com  Sun Oct 11 14:44:59 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 11 Oct 2009 21:44:59 +0000
Subject: [Freeswitch-users] mod_socket: custom timeout for HEARTBEAT
	event
In-Reply-To: <d211f54a0910110959y3d3deed7s43201a3e39756585@mail.gmail.com>
References: <d211f54a0910110959y3d3deed7s43201a3e39756585@mail.gmail.com>
Message-ID: <86a32abc0910111444j1ab2e446p3cb94b0ae04126e3@mail.gmail.com>

Maybe you want enable_heartbeat or sched_heartbeat from mod_dptools?

You can pass your parameters in second to these two.

Example:

<action application="enable_heartbeat" data="1"/>
<action application="sched_heartbeat" data="1"/>

Where 1 in this case is the number of heartbeats per seconds.

You can use that example on the Dialplan XML but you can also use it on
mod_event_socket outbound, etc.

Best regards,

Diego



On Sun, Oct 11, 2009 at 4:59 PM, Artem Shiyanov <shiyanov at gmail.com> wrote:

> Hi all!
> As it stays in wiki:
> ...
> HEARTBEAT
> Status information for freeswitch trigerred by freeswitch's heartbeat every
> 20 seconds.
> ...
>
> Is there any way to customize timeout of HEARTBEAT events?
>
>
> Thanks in advance,
> Artem
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Sun Oct 11 14:52:08 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Oct 2009 17:52:08 -0400
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <903da5680910082341r485092vb0cd10c43c4a2534@mail.gmail.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<87f2f3b90910080958i1f5cfccax621fd12100a05e3e@mail.gmail.com>
	<903da5680910081201p427a8358n768edb61a96674a4@mail.gmail.com>
	<87f2f3b90910081538l6e4e4b7dv3aedb2c4d1af58bc@mail.gmail.com>
	<903da5680910082238x27bff476s8bea0e74ab516a8d@mail.gmail.com>
	<87f2f3b90910082308i7f6144d9kc887596be54dd609@mail.gmail.com>
	<903da5680910082341r485092vb0cd10c43c4a2534@mail.gmail.com>
Message-ID: <219716B3-C355-4128-A333-1459AC63C364@jerris.com>


On Oct 9, 2009, at 2:41 AM, Gabriel Gunderson wrote:

> On Fri, Oct 9, 2009 at 12:08 AM, Michael Collins  
> <msc at freeswitch.org> wrote:
>> Thanks for reporting back. Please let all the Asterisk users know  
>> that they
>> are welcome to join us in #freeswitch on irc.freenode.net and that  
>> they will
>> not be abused like people do in other less friendly IRC channels.
>
> Funny you mention this.  Many people report that the way the FS
> community refers to Asterisk in docs/wikis/irc/whatever makes the FS
> camp seem *less* welcoming to them.  After all, they identify as
> Asterisk Users and take the criticism as being kinda harsh.  Most of
> them acknowledge the shortcomings of Asterisk but are put off when
> someone else points them out.  It's crazy, I know.  The thing is, I
> remember thinking that too.  After getting to know FS better, I didn't
> notice it as much.  Nobody likes to hear their baby is ugly --even if
> they know it is.
>
> At our session, and in general, I've noticed people are more
> interested in hearing about FS when you don't make direct comparisons
> to Asterisk.  Besides, FS stands on it's own merit.
>
> Just what I've observed *and* my 2 additional cents.


I have certainly seen this on irc in the past and we should do our  
best to avoid this, I have not seen this in the docs or wiki, do you  
know of any specifics you can point me to so we can correct this issue.

Mike



From mike at jerris.com  Sun Oct 11 14:55:02 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Oct 2009 17:55:02 -0400
Subject: [Freeswitch-users] gateway FS informs it's client FS about
	users hanged up with a long delay
In-Reply-To: <1255099232783-3795011.post@n2.nabble.com>
References: <1A16501E57CA4727A593226A8C810308@procent>
	<98F72CCB-88B9-47ED-AED3-B6BA6DE648C0@jerris.com>
	<1255007884165-3787956.post@n2.nabble.com>
	<87f2f3b90910081000v5b505f4ao53104211b620ea77@mail.gmail.com>
	<1255099232783-3795011.post@n2.nabble.com>
Message-ID: <4EB2CA6F-F7A6-4206-9E68-0457704F5418@jerris.com>


On Oct 9, 2009, at 10:40 AM, Maciej Aniserowicz wrote:

> Hello,
> The issue is resolved. I feel stupid, because Michael Jerris was  
> right the first time. Setting external_rtp_ip and external_sip_ip to  
> $${local_ip_v4} made it work.
> But the strange thing is: it SOMETIMES worked before without any  
> delay, which 'should not be possible', because the original IP was  
> my external ip and the BYE message was sent straight to it. And  
> there is no way it could reach the target 'internal' FS, because it  
> runs on virtual machine, and no ports are forwarded on my router.
> Any thoughts? Why this could (rarely) work even with the previous  
> config?
>
> Thanks to both of you for your answers.
>
> MA
>
>
It could work at least in regards to rtp if you had something else on  
the other side that adjusted to incorrect rtp ip like freeswitch does,  
the bye probably never really worked.

Mike

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From mike at jerris.com  Sun Oct 11 14:59:38 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Oct 2009 17:59:38 -0400
Subject: [Freeswitch-users] On the handling of SIP headers
In-Reply-To: <2d9149cd0910091210s33fb2bb9oc0c94a64aa87e0be@mail.gmail.com>
References: <2d9149cd0910091210s33fb2bb9oc0c94a64aa87e0be@mail.gmail.com>
Message-ID: <85095FC9-4DC9-414D-A43F-8A7583D66DFF@jerris.com>

There is this endless push and pull on this topic, those who want them  
assume it should be default, those who don't assume that should be  
default.  This probably needs a configuration option defaulting to  
pass them (those who don't want to pass them are usually a bit more  
educated and would find the option better than the other way around).

Mike

On Oct 9, 2009, at 3:10 PM, Kristian Kielhofner wrote:

> Hello everyone,
>
>  In using FS for various scenarios I've noticed some behavior that
> I'm not sure is completely "proper".  Given that this probably lives
> in mod_sofia who knows what's really "proper".  It is SIP after all...
>
>  So the issue comes up when using FreeSWITCH as a B2BUA and bridging
> between endpoints (very common).  Should FreeSWITCH copy the X-
> headers (possibly others) as it does now?  I'd like to think it
> shouldn't by default and the behavior should be one of:
>
> 1)  Don't pass X-* (or anything else, really) from one leg to another.
> If you want to pass specific X- headers (or anything else), set them
> explicitly on the outbound leg.
> 2)  Make the behavior configurable with a channel variable and/or
> sofia config option:
>
> {sip_pass_headers=all|none|X-MyCustomHeaderByName}
>
>  Thoughts?



From mike at jerris.com  Sun Oct 11 15:04:41 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Oct 2009 18:04:41 -0400
Subject: [Freeswitch-users] Sending an Event to a Session for onInput
In-Reply-To: <4256bf830910091234t11f4f4d0w28283f341bd1fb06@mail.gmail.com>
References: <4256bf830910091234t11f4f4d0w28283f341bd1fb06@mail.gmail.com>
Message-ID: <ACA9A602-0925-4D6F-8B11-0B214A462500@jerris.com>

We don't have session messages directly exposed, except for things  
like display, respond, and deflect.  What specifically are you trying  
to send ?

Mike

On Oct 9, 2009, at 3:34 PM, Matthew Fong wrote:

> I'm used to using the onInput callbacks inside lua and javascript to  
> listen for dtmf and other events and perform a task accordingly. I'm  
> wondering if there is a way to send an event to a session or channel  
> that can be caught using the setInputCallback inside lua from  
> outside the session program. Maybe an API command that can generate  
> an event for a specific UUID. Does a mechanism exist to do this that  
> I'm over looking? Thanks.



From diego.viola at gmail.com  Sun Oct 11 15:06:25 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 11 Oct 2009 22:06:25 +0000
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <219716B3-C355-4128-A333-1459AC63C364@jerris.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<87f2f3b90910080958i1f5cfccax621fd12100a05e3e@mail.gmail.com>
	<903da5680910081201p427a8358n768edb61a96674a4@mail.gmail.com>
	<87f2f3b90910081538l6e4e4b7dv3aedb2c4d1af58bc@mail.gmail.com>
	<903da5680910082238x27bff476s8bea0e74ab516a8d@mail.gmail.com>
	<87f2f3b90910082308i7f6144d9kc887596be54dd609@mail.gmail.com>
	<903da5680910082341r485092vb0cd10c43c4a2534@mail.gmail.com>
	<219716B3-C355-4128-A333-1459AC63C364@jerris.com>
Message-ID: <86a32abc0910111506m4e6d2740tbd3ac5fe0f59178@mail.gmail.com>

I have seen this on the wiki too, for example:

Q: Does it require hardware, kernel modules, ztdumshit, etc?
>
> Nope! :D You must be thinking of [http://sofaswitch.org/eg/aac.jpg|Something
> Else]
>


I know that's just a joke and we might make one or two jokes, but we don't
really hate Asterisk and we welcome them, there are some people who might
get offended by the jokes (specially Asterisk users) but I have already
corrected these from the wiki.

Thanks for the feedback Gabriel.

Diego

On Sun, Oct 11, 2009 at 9:52 PM, Michael Jerris <mike at jerris.com> wrote:

>
> On Oct 9, 2009, at 2:41 AM, Gabriel Gunderson wrote:
>
> > On Fri, Oct 9, 2009 at 12:08 AM, Michael Collins
> > <msc at freeswitch.org> wrote:
> >> Thanks for reporting back. Please let all the Asterisk users know
> >> that they
> >> are welcome to join us in #freeswitch on irc.freenode.net and that
> >> they will
> >> not be abused like people do in other less friendly IRC channels.
> >
> > Funny you mention this.  Many people report that the way the FS
> > community refers to Asterisk in docs/wikis/irc/whatever makes the FS
> > camp seem *less* welcoming to them.  After all, they identify as
> > Asterisk Users and take the criticism as being kinda harsh.  Most of
> > them acknowledge the shortcomings of Asterisk but are put off when
> > someone else points them out.  It's crazy, I know.  The thing is, I
> > remember thinking that too.  After getting to know FS better, I didn't
> > notice it as much.  Nobody likes to hear their baby is ugly --even if
> > they know it is.
> >
> > At our session, and in general, I've noticed people are more
> > interested in hearing about FS when you don't make direct comparisons
> > to Asterisk.  Besides, FS stands on it's own merit.
> >
> > Just what I've observed *and* my 2 additional cents.
>
>
> I have certainly seen this on irc in the past and we should do our
> best to avoid this, I have not seen this in the docs or wiki, do you
> know of any specifics you can point me to so we can correct this issue.
>
> Mike
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Sun Oct 11 15:14:50 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Oct 2009 18:14:50 -0400
Subject: [Freeswitch-users] Bad sound quality while eavesdropping
In-Reply-To: <1255169044209-3799274.post@n2.nabble.com>
References: <41A44DD027064988A914974405788C2E@procent>
	<191c3a030910050731m2d74979ep4598e5a1945d58ae@mail.gmail.com>
	<1254901192035-3780245.post@n2.nabble.com>
	<8437F5BC-7AFF-4A74-B8CD-C5B8219021F6@jerris.com>
	<1255008427639-3788019.post@n2.nabble.com>
	<191c3a030910080823g79c7c596x1cd887e1538ce2e1@mail.gmail.com>
	<1255169044209-3799274.post@n2.nabble.com>
Message-ID: <CF8C1791-24CD-4132-83B5-7F83B5BDCCC3@jerris.com>

can you confirm from an rtp packet trace that they are all really  
sending 20ms?

Mike

On Oct 10, 2009, at 6:04 AM, Maciej Aniserowicz wrote:

>
> Hi,
> Here are the messages with a:ptime parameter. All the calls are  
> started by
> commands sent through socket.
> I'm not sure if this is all information you need, please let me know  
> if
> something is missing here and I'll post that.
>
> 1) starting connection with x-lite (number 2003, the eavesdropper):
>
>   INVITE sip:2003 at 192.168.3.100:60188;rinstance=80b8f8d92af87cd2 SIP/ 
> 2.0
>   Via: SIP/2.0/UDP  
> 192.168.3.159:15060;rport;branch=z9hG4bKD9meHQN5XQ88K
>   Max-Forwards: 69
>   From: "MyApp" <sip:0000000000 at 192.168.3.159>;tag=jpQ6D7D2jUXvF
>   To: <sip:2003 at 192.168.3.100:60188;rinstance=80b8f8d92af87cd2>
>   Call-ID: 663c4500-3024-122d-57a2-9be8d85a4cff
>   CSeq: 121465610 INVITE
>   Contact: <sip:mod_sofia at 192.168.3.159:15060>
>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>   Supported: timer, precondition, path, replaces
>   Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>   Content-Type: application/sdp
>   Content-Disposition: session
>   Content-Length: 447
>   Remote-Party-ID: "MyApp"
> <sip:0000000000 at 192.168.3.159>;party=calling;screen=yes;privacy=off
>
>   v=0
>   o=FreeSWITCH 2223565947735016740 3096553520713245589 IN IP4  
> 192.168.3.159
>   s=FreeSWITCH
>   c=IN IP4 192.168.3.159
>   t=0 0
>   m=audio 29966 RTP/AVP 0 115 107 9 8 3 101 13
>   a=rtpmap:0 PCMU/8000
>   a=rtpmap:115 G7221/32000
>   a=fmtp:115 bitrate=48000
>   a=rtpmap:107 G7221/16000
>   a=fmtp:107 bitrate=32000
>   a=rtpmap:9 G722/8000
>   a=rtpmap:8 PCMA/8000
>   a=rtpmap:3 GSM/8000
>   a=rtpmap:101 telephone-event/8000
>   a=fmtp:101 0-16
>   a=rtpmap:13 CN/8000
>   a=ptime:20
>
>
> 2) starting connection with cisco ip phone (number 2006, first leg of
> eavesdropped session):
>
>   INVITE sip:2006 at 192.168.2.106:5060;user=phone SIP/2.0
>   Via: SIP/2.0/UDP  
> 192.168.3.159:15060;rport;branch=z9hG4bKg40rp87FNjB1p
>   Max-Forwards: 69
>   From: "MyApp" <sip:0000000000 at 192.168.3.159>;tag=Q3N2pe2K47ctS
>   To: <sip:2006 at 192.168.2.106:5060;user=phone>
>   Call-ID: 6e6fd8a0-3024-122d-57a2-9be8d85a4cff
>   CSeq: 121465616 INVITE
>   Contact: <sip:mod_sofia at 192.168.3.159:15060>
>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>   Supported: timer, precondition, path, replaces
>   Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>   Content-Type: application/sdp
>   Content-Disposition: session
>   Content-Length: 447
>   Remote-Party-ID: "MyApp"
> <sip:0000000000 at 192.168.3.159>;party=calling;screen=yes;privacy=off
>
>   v=0
>   o=FreeSWITCH 8453497903781974949 2294419114567885490 IN IP4  
> 192.168.3.159
>   s=FreeSWITCH
>   c=IN IP4 192.168.3.159
>   t=0 0
>   m=audio 17670 RTP/AVP 0 115 107 9 8 3 101 13
>   a=rtpmap:0 PCMU/8000
>   a=rtpmap:115 G7221/32000
>   a=fmtp:115 bitrate=48000
>   a=rtpmap:107 G7221/16000
>   a=fmtp:107 bitrate=32000
>   a=rtpmap:9 G722/8000
>   a=rtpmap:8 PCMA/8000
>   a=rtpmap:3 GSM/8000
>   a=rtpmap:101 telephone-event/8000
>   a=fmtp:101 0-16
>   a=rtpmap:13 CN/8000
>   a=ptime:20
>
>
> 3) starting connection with extension playing a file (number 9999,  
> second
> leg of eavesdropped session):
>
>   SIP/2.0 200 OK
>   Via: SIP/2.0/UDP
> 192.168.3.159:5080;rport=5080;branch=z9hG4bKv6Dg7myj4tvBS
>   From: "FreeSWITCH"  
> <sip:myuser at mydomain;transport=udp>;tag=091j2Q0Fre8vp
>   To: <sip:9999 at 192.168.3.159:15060>;tag=U7t5Xt51rB64Q
>   Call-ID: 7551b7d0-3024-122d-b3a6-156c4939c4f8
>   CSeq: 121465623 INVITE
>   Contact: <sip:mod_sofia at 192.168.3.159:15060;transport=udp>
>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN
>   Accept: application/sdp
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>   Supported: timer, precondition, path, replaces
>   Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>   Content-Type: application/sdp
>   Content-Disposition: session
>   Content-Length: 263
>
>   v=0
>   o=FreeSWITCH 5614997529598779838 636912243381649698 IN IP4  
> 192.168.3.159
>   s=FreeSWITCH
>   c=IN IP4 192.168.3.159
>   t=0 0
>   m=audio 30086 RTP/AVP 0 101 13
>   a=rtpmap:0 PCMU/8000
>   a=rtpmap:101 telephone-event/8000
>   a=fmtp:101 0-16
>   a=rtpmap:13 CN/8000
>   a=ptime:20
>
>
>
>
> Anthony Minessale wrote:
>>
>> you probably have some device lying about ptime everywhere
>> look at a sip trace an pay especially close attention to ptime:x  
>> param in
>> sdp
>> if you don't understand this just attach it here
>>
>> execute the following at the cli
>> sofia profile internal siptrace on
>> sofila loglevel debug
>>
>>
>>
>> On Thu, Oct 8, 2009 at 8:27 AM, Maciej Aniserowicz <
>> maciej.aniserowicz at gmail.com> wrote:
>>
>>>
>>> It's the same on the trunk (the last rev I used was not so old  
>>> anyway).
>>>
>>> Codecs are the same on both legs:
>>> read codec/read rate: PCMU      8000
>>> write codec/write rate: PCMU    8000
>>>
>>> MA
>>>
>>>
>>>
>>>
>>> Michael Jerris wrote:
>>>>
>>>> What codecs are all the call legs using, also, please try current  
>>>> svn
>>>> trunk.
>>>>
>>>> Mike
>>>>
>>>> On Oct 7, 2009, at 3:39 AM, Maciej Aniserowicz wrote:
>>>>
>>>>>
>>>>> Sorry about posting several questions at once, I wasn't aware it's
>>>>> "rude".
>>>>> Let's concentrate on this issue then.
>>>>>
>>>>> I use FS rev 14994. Phones on extensions:
>>>>> 1) x-lite
>>>>> 2) cisco sip phone
>>>>> 3) audio played by fs to the extension being eavesdropped
>>>>>
>>>>> I did not change any codec configuration, I just use the standard
>>>>> one that
>>>>> comes with both FS and the phones.
>>>>> Some time ago someone on FS irc channel told me that this is just
>>>>> how FS
>>>>> eavesdropping works... from your response I understand that this  
>>>>> is
>>>>> not
>>>>> entirely true?
>>>>>
>>>>> Maciej Aniserowicz
>>>>>
>>>>>
>>>>>
>>>>> Anthony Minessale wrote:
>>>>>>
>>>>>> That's is a somewhat vague position.
>>>>>>
>>>>>> You did not mention which version of FreeSWITCH you are  
>>>>>> running, the
>>>>>> phones
>>>>>> being used in your example, your configuration, the codecs in use
>>>>>> etc.
>>>>>>
>>>>>> BTW,
>>>>>> I think you should only ask one question at a time on this list.
>>>>>> The list
>>>>>> is run by volunteers and it's sort of rude to expect 3 or 4  
>>>>>> threads
>>>>>> to be
>>>>>> tended to concerning the same one individual.
>>>>>>
>>>>>>
>>>>>> 2009/10/5 Maciej Aniserowicz <maciej.aniserowicz at gmail.com>
>>>>>>
>>>>>>> Hello,
>>>>>>> When I use eavesdropping in FreeSWITCH, the sound quality is
>>>>>>> really bad.
>>>>>>> Is
>>>>>>> there any way to improve it? Is this a known problem?
>>>>>>> Br/
>>>>>>> Maciej Aniserowicz
>>>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>>> users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> --
>>> View this message in context:
>>> http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3788019.html
>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>>> users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> -- 
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN 
>> %3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL 
>> %3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip 
>> %3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf 
>> %2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>>
>
> -- 
> View this message in context: http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3799274.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From mike at jerris.com  Sun Oct 11 15:52:32 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Oct 2009 18:52:32 -0400
Subject: [Freeswitch-users] mod_socket: custom timeout for HEARTBEAT
	event
In-Reply-To: <86a32abc0910111444j1ab2e446p3cb94b0ae04126e3@mail.gmail.com>
References: <d211f54a0910110959y3d3deed7s43201a3e39756585@mail.gmail.com>
	<86a32abc0910111444j1ab2e446p3cb94b0ae04126e3@mail.gmail.com>
Message-ID: <97B4BDC9-5B11-4A98-B1BF-CC00A82ECBD9@jerris.com>


On Oct 11, 2009, at 5:44 PM, Diego Viola wrote:

> Maybe you want enable_heartbeat or sched_heartbeat from mod_dptools?
>
> You can pass your parameters in second to these two.
>
> Example:
>
> <action application="enable_heartbeat" data="1"/>
> <action application="sched_heartbeat" data="1"/>
>
> Where 1 in this case is the number of heartbeats per seconds.
>

Number of seconds between hearbeats, not hearbeats per second.


> You can use that example on the Dialplan XML but you can also use it  
> on mod_event_socket outbound, etc.



From mike at jerris.com  Sun Oct 11 16:01:02 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Oct 2009 19:01:02 -0400
Subject: [Freeswitch-users] problem compiling esl for use with freepbx v3
In-Reply-To: <4d6f26b0910111436h416121b7s2a4b5c79ffc01db0@mail.gmail.com>
References: <20090823185152.D17845FE@sinclaire.sibble.net>
	<25846572.post@talk.nabble.com>
	<4d6f26b0910111436h416121b7s2a4b5c79ffc01db0@mail.gmail.com>
Message-ID: <0E66A94D-2F5E-421A-B997-A8705715B0AD@jerris.com>

I am still working on the new build system for esl, stay tuned for  
more info soon, it should be in 1.0.5.

Mike

On Oct 11, 2009, at 5:36 PM, Herman Griffin wrote:

> Although probably not the best solution, I figured out a way to make  
> it compile and install:
>
> I removed all of the -Werror instances in PATH_TO_FREESWITCH_SOURCE/ 
> libs/esl/Makefile
>
> If I was a hardcore c/c++ programmer, I'd figure out the real problem.
>
> Herman aka frek818
>
> On Sun, Oct 11, 2009 at 12:12 PM, frek818 <herman.griffin at gmail.com>  
> wrote:
>
> Did anyone find a solution to this problem? I too would like to  
> install the
> esl module for PHP.
>

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From kristian.kielhofner at gmail.com  Sun Oct 11 16:12:41 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Sun, 11 Oct 2009 19:12:41 -0400
Subject: [Freeswitch-users] On the handling of SIP headers
In-Reply-To: <85095FC9-4DC9-414D-A43F-8A7583D66DFF@jerris.com>
References: <2d9149cd0910091210s33fb2bb9oc0c94a64aa87e0be@mail.gmail.com>
	<85095FC9-4DC9-414D-A43F-8A7583D66DFF@jerris.com>
Message-ID: <2d9149cd0910111612w668abce4m7562a8fbda3ba13f@mail.gmail.com>

Mike,

  Thanks for getting back to me.  I agree.

  I'm willing to throw down on a bounty for this.  Any idea how much
work we're talking about here?

On Sun, Oct 11, 2009 at 5:59 PM, Michael Jerris <mike at jerris.com> wrote:
> There is this endless push and pull on this topic, those who want them
> assume it should be default, those who don't assume that should be
> default. ?This probably needs a configuration option defaulting to
> pass them (those who don't want to pass them are usually a bit more
> educated and would find the option better than the other way around).
>
> Mike
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From brian at freeswitch.org  Sun Oct 11 16:39:27 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 11 Oct 2009 18:39:27 -0500
Subject: [Freeswitch-users] SLAs and BLAs
In-Reply-To: <5CC2E91025314615ACBB65E029D5CB12@greyhawk.tonecommander.com>
References: <5B5F50E0B1D34BC0BCBCC05E1FE05C8A@greyhawk.tonecommander.com>
	<3F7AC380-CACB-432F-8B94-80FBA18628C0@freeswitch.org>
	<5CC2E91025314615ACBB65E029D5CB12@greyhawk.tonecommander.com>
Message-ID: <FCD5284F-8F0D-4D69-91E1-B25145F35F13@freeswitch.org>

No we do not issue a subscribe OUTBOUND.  We work with Polycom, Snom  
and a few other that outbound subscribe is something we should do if  
we want to know you took the phone off hook.

/b

On Oct 9, 2009, at 10:23 AM, Jerry Richards wrote:

>
> I gather from the mailing archive that BLAs are implemented using the
> draft-anil-sipping-bla-04.txt document.  According to the draft, the
> Appearance Agent is supposed to initiate a SUBSCRIBE request, but I  
> don't
> see FS doing this.
>
> What phone types/models are known to work with the FS BLA  
> implementation?
>
> Best Regards,
> Jerry



From brian at freeswitch.org  Sun Oct 11 16:43:37 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 11 Oct 2009 18:43:37 -0500
Subject: [Freeswitch-users] SLAs and BLAs
In-Reply-To: <5CC2E91025314615ACBB65E029D5CB12@greyhawk.tonecommander.com>
References: <5B5F50E0B1D34BC0BCBCC05E1FE05C8A@greyhawk.tonecommander.com>
	<3F7AC380-CACB-432F-8B94-80FBA18628C0@freeswitch.org>
	<5CC2E91025314615ACBB65E029D5CB12@greyhawk.tonecommander.com>
Message-ID: <422F970E-7708-4124-AD41-A51D22A3EDEF@freeswitch.org>

btw the polycom it will do the sub like you expect but the rest will  
only know when the phone is actually on a call or receiving a call...  
we won't know if the handset is taken off hook...I would like to  
rework the functionality similar to how sofia_sla.c handles it.    
Again if you want to work on it and patch it then I'm sure we can  
provide you guidance.


/b

On Oct 9, 2009, at 10:23 AM, Jerry Richards wrote:

>
> I gather from the mailing archive that BLAs are implemented using the
> draft-anil-sipping-bla-04.txt document.  According to the draft, the
> Appearance Agent is supposed to initiate a SUBSCRIBE request, but I  
> don't
> see FS doing this.
>
> What phone types/models are known to work with the FS BLA  
> implementation?
>
> Best Regards,
> Jerry



From brian at freeswitch.org  Sun Oct 11 16:44:35 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 11 Oct 2009 18:44:35 -0500
Subject: [Freeswitch-users] On the handling of SIP headers
In-Reply-To: <2d9149cd0910111612w668abce4m7562a8fbda3ba13f@mail.gmail.com>
References: <2d9149cd0910091210s33fb2bb9oc0c94a64aa87e0be@mail.gmail.com>
	<85095FC9-4DC9-414D-A43F-8A7583D66DFF@jerris.com>
	<2d9149cd0910111612w668abce4m7562a8fbda3ba13f@mail.gmail.com>
Message-ID: <62E38F6E-C929-423D-AF45-6FEBB58F6CD3@freeswitch.org>

Well since we aren't a proxy you shouldn't default to passing them  
right?

/b

On Oct 11, 2009, at 6:12 PM, Kristian Kielhofner wrote:

> Mike,
>
>  Thanks for getting back to me.  I agree.
>
>  I'm willing to throw down on a bounty for this.  Any idea how much
> work we're talking about here?



From brian at freeswitch.org  Sun Oct 11 16:49:58 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 11 Oct 2009 18:49:58 -0500
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <86a32abc0910111506m4e6d2740tbd3ac5fe0f59178@mail.gmail.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<87f2f3b90910080958i1f5cfccax621fd12100a05e3e@mail.gmail.com>
	<903da5680910081201p427a8358n768edb61a96674a4@mail.gmail.com>
	<87f2f3b90910081538l6e4e4b7dv3aedb2c4d1af58bc@mail.gmail.com>
	<903da5680910082238x27bff476s8bea0e74ab516a8d@mail.gmail.com>
	<87f2f3b90910082308i7f6144d9kc887596be54dd609@mail.gmail.com>
	<903da5680910082341r485092vb0cd10c43c4a2534@mail.gmail.com>
	<219716B3-C355-4128-A333-1459AC63C364@jerris.com>
	<86a32abc0910111506m4e6d2740tbd3ac5fe0f59178@mail.gmail.com>
Message-ID: <5162D0B7-357C-4D0E-9913-46564A8C0B1D@freeswitch.org>

I have tried to police the wiki when things like this appear.. its one  
thing to crack a joke in fun from time to time... but to put stuff  
like that on the wiki isn't acceptable.

/b

On Oct 11, 2009, at 5:06 PM, Diego Viola wrote:

>
> I know that's just a joke and we might make one or two jokes, but we  
> don't really hate Asterisk and we welcome them, there are some  
> people who might get offended by the jokes (specially Asterisk  
> users) but I have already corrected these from the wiki.
>
> Thanks for the feedback Gabriel.
>
> Diego



From brian at freeswitch.org  Sun Oct 11 16:53:03 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 11 Oct 2009 18:53:03 -0500
Subject: [Freeswitch-users] Re corded file as voicemail.
In-Reply-To: <23f91030910092240u53b0cc51jd5b1670901f20e21@mail.gmail.com>
References: <25830785.post@talk.nabble.com>
	<20091010040539.GB23188@jdc.jasonjgw.net>
	<23f91030910092240u53b0cc51jd5b1670901f20e21@mail.gmail.com>
Message-ID: <89F518C6-B470-482D-8C27-2812BDC20C39@freeswitch.org>

FreeSWITCH can play back stereo files it'll just mux them down to mono  
before playing... can you elaborate on the error you're getting?

/b

On Oct 10, 2009, at 12:40 AM, Seven Du wrote:

> Yes, it's discussed before.
>
> http://wiki.freeswitch.org/wiki/Channel_Variables#RECORD_STEREO
>
> set that var to false before you record.

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From dujinfang at gmail.com  Sun Oct 11 18:45:46 2009
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 12 Oct 2009 09:45:46 +0800
Subject: [Freeswitch-users] Re corded file as voicemail.
In-Reply-To: <89F518C6-B470-482D-8C27-2812BDC20C39@freeswitch.org>
References: <25830785.post@talk.nabble.com>
	<20091010040539.GB23188@jdc.jasonjgw.net>
	<23f91030910092240u53b0cc51jd5b1670901f20e21@mail.gmail.com>
	<89F518C6-B470-482D-8C27-2812BDC20C39@freeswitch.org>
Message-ID: <23f91030910111845yacd54ees40f246ef9152e306@mail.gmail.com>

got

2009-10-12 01:41:30.349961 [WARNING] switch_core_file.c:133 File has 2
channels, muxing to mono will occur.
2009-10-12 01:41:30.349961 [ERR] switch_core_codec.c:431 Stereo is currently
unsupported. please downsample audio source to mono.

freeswitch at internal>
freeswitch at internal> version
FreeSWITCH Version 1.0.trunk (14696)


originate {ignore_early_media=true,RECORD_STEREO=true}sofia/gateway/xx/xx
&bridge(sofia/gateway/yy/yy)

uuid_record uuid start /tmp/a.wav



play by sox:

Input File     : 'a.wav'
Sample Size    : 16-bit (2 bytes)
Sample Encoding: signed (2's complement)
Channels       : 2
Sample Rate    : 8000


full logs here:

freeswitch at internal> originate
{ignore_early_media=true}sofia/gateway/skype/seven1240 &playback(/tmp/a.wav)
+OK dcbd971e-c570-481d-843a-0c5582669c69

2009-10-12 01:41:26.89775 [DEBUG] switch_ivr_originate.c:1043 variable
string 0 = [ignore_early_media=true]
freeswitch at internal> 2009-10-12 01:41:26.89775 [NOTICE] switch_channel.c:602
New Channel sofia/eqenglish/seven1240 [dcbd971e-c570-481d-843a-0c5582669c69]
2009-10-12 01:41:26.89775 [DEBUG] mod_sofia.c:2867
(sofia/eqenglish/seven1240) State Change CS_NEW -> CS_INIT
2009-10-12 01:41:26.89775 [DEBUG] switch_core_session.c:932 Send signal
sofia/eqenglish/seven1240 [BREAK]
2009-10-12 01:41:26.89775 [DEBUG] switch_core_state_machine.c:398
(sofia/eqenglish/seven1240) Running State Change CS_INIT
2009-10-12 01:41:26.89775 [DEBUG] switch_core_state_machine.c:481
(sofia/eqenglish/seven1240) State INIT
2009-10-12 01:41:26.89775 [DEBUG] mod_sofia.c:83 sofia/eqenglish/seven1240
SOFIA INIT
2009-10-12 01:41:26.89775 [DEBUG] mod_sofia.c:111
(sofia/eqenglish/seven1240) State Change CS_INIT -> CS_ROUTING
2009-10-12 01:41:26.89775 [DEBUG] switch_core_session.c:932 Send signal
sofia/eqenglish/seven1240 [BREAK]
2009-10-12 01:41:26.89775 [DEBUG] switch_core_state_machine.c:481
(sofia/eqenglish/seven1240) State INIT going to sleep
2009-10-12 01:41:26.89775 [DEBUG] switch_core_state_machine.c:398
(sofia/eqenglish/seven1240) Running State Change CS_ROUTING
2009-10-12 01:41:26.89775 [DEBUG] switch_core_state_machine.c:484
(sofia/eqenglish/seven1240) State ROUTING
2009-10-12 01:41:26.89775 [DEBUG] mod_sofia.c:130 sofia/eqenglish/seven1240
SOFIA ROUTING
2009-10-12 01:41:26.89775 [DEBUG] switch_ivr_originate.c:63
(sofia/eqenglish/seven1240) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2009-10-12 01:41:26.89775 [DEBUG] switch_core_session.c:932 Send signal
sofia/eqenglish/seven1240 [BREAK]
2009-10-12 01:41:26.89775 [DEBUG] switch_core_state_machine.c:484
(sofia/eqenglish/seven1240) State ROUTING going to sleep
2009-10-12 01:41:26.89775 [DEBUG] switch_core_state_machine.c:398
(sofia/eqenglish/seven1240) Running State Change CS_CONSUME_MEDIA
2009-10-12 01:41:26.89775 [DEBUG] switch_core_state_machine.c:503
(sofia/eqenglish/seven1240) State CONSUME_MEDIA
2009-10-12 01:41:26.89775 [DEBUG] sofia.c:3302 Channel
sofia/eqenglish/seven1240 entering state [calling][0]
2009-10-12 01:41:26.169828 [DEBUG] sofia.c:3302 Channel
sofia/eqenglish/seven1240 entering state [calling][0]
2009-10-12 01:41:28.700782 [DEBUG] sofia.c:3302 Channel
sofia/eqenglish/seven1240 entering state [proceeding][180]
2009-10-12 01:41:28.700782 [NOTICE] sofia.c:3366 Ring-Ready
sofia/eqenglish/seven1240!
2009-10-12 01:41:30.330095 [DEBUG] sofia.c:3302 Channel
sofia/eqenglish/seven1240 entering state [ready][200]
2009-10-12 01:41:30.330095 [DEBUG] sofia.c:3309 Remote SDP:
v=0
o=FreeSWITCH 1255290734 1255290735 IN IP4 67.228.224.149
s=FreeSWITCH
c=IN IP4 127.0.0.1
t=0 0
m=audio 20956 RTP/AVP 0 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20

2009-10-12 01:41:30.330095 [DEBUG] sofia_glue.c:3132 Audio Codec Compare
[PCMU:0:8000:20]/[PCMU:0:8000:20]
2009-10-12 01:41:30.330095 [DEBUG] sofia_glue.c:2090 Set Codec
sofia/eqenglish/seven1240 PCMU/8000 20 ms 160 samples
2009-10-12 01:41:30.330095 [DEBUG] sofia_glue.c:3092 Set 2833 dtmf payload
to 101
2009-10-12 01:41:30.330095 [DEBUG] sofia_glue.c:2324 AUDIO RTP
[sofia/eqenglish/seven1240] 67.228.224.149 port 24862 -> 127.0.0.1 port
20956 codec: 0 ms: 20
2009-10-12 01:41:30.330095 [DEBUG] switch_rtp.c:1139 Starting timer [soft]
160 bytes per 20ms
2009-10-12 01:41:30.339941 [DEBUG] sofia_glue.c:2503 Set comfort noise
payload to 13
2009-10-12 01:41:30.339941 [NOTICE] sofia.c:3807 Channel
[sofia/eqenglish/seven1240] has been answered
2009-10-12 01:41:30.339941 [DEBUG] switch_channel.c:182
sofia/eqenglish/seven1240 receive message [AUDIO_SYNC]
2009-10-12 01:41:30.339941 [NOTICE] switch_cpp.cpp:1130 ==DebugVar==
gateway_name: skype
2009-10-12 01:41:30.349961 [DEBUG] switch_ivr_originate.c:2089 Originate
Resulted in Success: [sofia/eqenglish/seven1240]
2009-10-12 01:41:30.349961 [DEBUG] switch_channel.c:182
sofia/eqenglish/seven1240 receive message [AUDIO_SYNC]
2009-10-12 01:41:30.349961 [DEBUG] mod_commands.c:2334
(sofia/eqenglish/seven1240) State Change CS_CONSUME_MEDIA -> CS_EXECUTE
2009-10-12 01:41:30.349961 [DEBUG] switch_core_session.c:932 Send signal
sofia/eqenglish/seven1240 [BREAK]
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:503
(sofia/eqenglish/seven1240) State CONSUME_MEDIA going to sleep
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:398
(sofia/eqenglish/seven1240) Running State Change CS_EXECUTE
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:491
(sofia/eqenglish/seven1240) State EXECUTE
2009-10-12 01:41:30.349961 [DEBUG] mod_sofia.c:173 sofia/eqenglish/seven1240
SOFIA EXECUTE
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:151
sofia/eqenglish/seven1240 Standard EXECUTE
EXECUTE sofia/eqenglish/seven1240 playback(/tmp/a.wav)
2009-10-12 01:41:30.349961 [WARNING] switch_core_file.c:133 File has 2
channels, muxing to mono will occur.
2009-10-12 01:41:30.349961 [ERR] switch_core_codec.c:431 Stereo is currently
unsupported. please downsample audio source to mono.
2009-10-12 01:41:30.349961 [DEBUG] switch_ivr_play_say.c:1113 Raw Codec
Activation Failed L16 at 8000hz 2 channels 20ms
2009-10-12 01:41:30.349961 [NOTICE] switch_core_state_machine.c:179 Hangup
sofia/eqenglish/seven1240 [CS_EXECUTE] [NORMAL_CLEARING]
2009-10-12 01:41:30.349961 [DEBUG] switch_channel.c:1715 Send signal
sofia/eqenglish/seven1240 [KILL]
2009-10-12 01:41:30.349961 [DEBUG] switch_core_session.c:932 Send signal
sofia/eqenglish/seven1240 [BREAK]
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:491
(sofia/eqenglish/seven1240) State EXECUTE going to sleep
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:398
(sofia/eqenglish/seven1240) Running State Change CS_HANGUP
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:434
(sofia/eqenglish/seven1240) State HANGUP
2009-10-12 01:41:30.349961 [DEBUG] mod_sofia.c:338 Channel
sofia/eqenglish/seven1240 hanging up, cause: NORMAL_CLEARING
2009-10-12 01:41:30.349961 [DEBUG] mod_sofia.c:376 Sending BYE to
sofia/eqenglish/seven1240
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:46
sofia/eqenglish/seven1240 Standard HANGUP, cause: NORMAL_CLEARING
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:434
(sofia/eqenglish/seven1240) State HANGUP going to sleep
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:476
(sofia/eqenglish/seven1240) State Change CS_HANGUP -> CS_REPORTING
2009-10-12 01:41:30.349961 [DEBUG] switch_core_session.c:932 Send signal
sofia/eqenglish/seven1240 [BREAK]
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:398
(sofia/eqenglish/seven1240) Running State Change CS_REPORTING
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:613
(sofia/eqenglish/seven1240) State REPORTING
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:53
sofia/eqenglish/seven1240 Standard REPORTING, cause: NORMAL_CLEARING
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:613
(sofia/eqenglish/seven1240) State REPORTING going to sleep
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:411
(sofia/eqenglish/seven1240) State Change CS_REPORTING -> CS_DESTROY
2009-10-12 01:41:30.349961 [DEBUG] switch_core_session.c:932 Send signal
sofia/eqenglish/seven1240 [BREAK]
2009-10-12 01:41:30.349961 [DEBUG] switch_core_session.c:1068 Session 2178
(sofia/eqenglish/seven1240) Locked, Waiting on external entities
2009-10-12 01:41:30.349961 [NOTICE] switch_core_session.c:1086 Session 2178
(sofia/eqenglish/seven1240) Ended
2009-10-12 01:41:30.349961 [NOTICE] switch_core_session.c:1088 Close Channel
sofia/eqenglish/seven1240 [CS_DESTROY]
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:556
(sofia/eqenglish/seven1240) Running State Change CS_DESTROY
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:565
(sofia/eqenglish/seven1240) State DESTROY
2009-10-12 01:41:30.349961 [DEBUG] mod_sofia.c:255 sofia/eqenglish/seven1240
SOFIA DESTROY
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:60
sofia/eqenglish/seven1240 Standard DESTROY
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:565
(sofia/eqenglish/seven1240) State DESTROY going to sleep
2009-10-12 01:41:30.349961 [NOTICE] switch_cpp.cpp:1130 ==DebugVar==
gateway_name: skype


2009/10/12 Brian West <brian at freeswitch.org>

> FreeSWITCH can play back stereo files it'll just mux them down to mono
> before playing... can you elaborate on the error you're getting?
> /b
>
> On Oct 10, 2009, at 12:40 AM, Seven Du wrote:
>
> Yes, it's discussed before.
>
> http://wiki.freeswitch.org/wiki/Channel_Variables#RECORD_STEREO
>
> set that var to false before you record.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jason at jasonjgw.net  Sun Oct 11 19:07:16 2009
From: jason at jasonjgw.net (Jason White)
Date: Mon, 12 Oct 2009 13:07:16 +1100
Subject: [Freeswitch-users] Re corded file as voicemail.
In-Reply-To: <23f91030910111845yacd54ees40f246ef9152e306@mail.gmail.com>
References: <25830785.post@talk.nabble.com>
	<20091010040539.GB23188@jdc.jasonjgw.net>
	<23f91030910092240u53b0cc51jd5b1670901f20e21@mail.gmail.com>
	<89F518C6-B470-482D-8C27-2812BDC20C39@freeswitch.org>
	<23f91030910111845yacd54ees40f246ef9152e306@mail.gmail.com>
Message-ID: <20091012020716.GA8518@jdc.jasonjgw.net>

Seven Du <dujinfang at gmail.com> wrote:
> originate {ignore_early_media=true,RECORD_STEREO=true}sofia/gateway/xx/xx
> &bridge(sofia/gateway/yy/yy)

Shouldn't that be record_stereo=false for mono recording?



From kristian.kielhofner at gmail.com  Sun Oct 11 19:17:08 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Sun, 11 Oct 2009 22:17:08 -0400
Subject: [Freeswitch-users] On the handling of SIP headers
In-Reply-To: <62E38F6E-C929-423D-AF45-6FEBB58F6CD3@freeswitch.org>
References: <2d9149cd0910091210s33fb2bb9oc0c94a64aa87e0be@mail.gmail.com>
	<85095FC9-4DC9-414D-A43F-8A7583D66DFF@jerris.com>
	<2d9149cd0910111612w668abce4m7562a8fbda3ba13f@mail.gmail.com>
	<62E38F6E-C929-423D-AF45-6FEBB58F6CD3@freeswitch.org>
Message-ID: <2d9149cd0910111917n53b59be5m1ab839fc959c7dbd@mail.gmail.com>

Brian,

  You are correct, they *probably* shouldn't be passed by default.

On Sun, Oct 11, 2009 at 7:44 PM, Brian West <brian at freeswitch.org> wrote:
> Well since we aren't a proxy you shouldn't default to passing them
> right?
>
> /b
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From dujinfang at gmail.com  Sun Oct 11 19:27:11 2009
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 12 Oct 2009 10:27:11 +0800
Subject: [Freeswitch-users] Re corded file as voicemail.
In-Reply-To: <20091012020716.GA8518@jdc.jasonjgw.net>
References: <25830785.post@talk.nabble.com>
	<20091010040539.GB23188@jdc.jasonjgw.net>
	<23f91030910092240u53b0cc51jd5b1670901f20e21@mail.gmail.com>
	<89F518C6-B470-482D-8C27-2812BDC20C39@freeswitch.org>
	<23f91030910111845yacd54ees40f246ef9152e306@mail.gmail.com>
	<20091012020716.GA8518@jdc.jasonjgw.net>
Message-ID: <23f91030910111927v704b2eb9tfd9558068b063153@mail.gmail.com>

I set to true because brian said it can play stereo files but no lucky for
me.

2009/10/12 Jason White <jason at jasonjgw.net>

> Seven Du <dujinfang at gmail.com> wrote:
> > originate {ignore_early_media=true,RECORD_STEREO=true}sofia/gateway/xx/xx
> > &bridge(sofia/gateway/yy/yy)
>
> Shouldn't that be record_stereo=false for mono recording?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sun Oct 11 19:52:59 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 11 Oct 2009 21:52:59 -0500
Subject: [Freeswitch-users] Re corded file as voicemail.
In-Reply-To: <23f91030910111927v704b2eb9tfd9558068b063153@mail.gmail.com>
References: <25830785.post@talk.nabble.com>
	<20091010040539.GB23188@jdc.jasonjgw.net>
	<23f91030910092240u53b0cc51jd5b1670901f20e21@mail.gmail.com>
	<89F518C6-B470-482D-8C27-2812BDC20C39@freeswitch.org>
	<23f91030910111845yacd54ees40f246ef9152e306@mail.gmail.com>
	<20091012020716.GA8518@jdc.jasonjgw.net>
	<23f91030910111927v704b2eb9tfd9558068b063153@mail.gmail.com>
Message-ID: <5F170ABD-E654-4511-889B-5893FC33C03E@freeswitch.org>

Where are you playing the files?

/b

On Oct 11, 2009, at 9:27 PM, Seven Du wrote:

> I set to true because brian said it can play stereo files but no  
> lucky for me.



From brian at freeswitch.org  Sun Oct 11 20:01:23 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 11 Oct 2009 22:01:23 -0500
Subject: [Freeswitch-users] Re corded file as voicemail.
In-Reply-To: <23f91030910111927v704b2eb9tfd9558068b063153@mail.gmail.com>
References: <25830785.post@talk.nabble.com>
	<20091010040539.GB23188@jdc.jasonjgw.net>
	<23f91030910092240u53b0cc51jd5b1670901f20e21@mail.gmail.com>
	<89F518C6-B470-482D-8C27-2812BDC20C39@freeswitch.org>
	<23f91030910111845yacd54ees40f246ef9152e306@mail.gmail.com>
	<20091012020716.GA8518@jdc.jasonjgw.net>
	<23f91030910111927v704b2eb9tfd9558068b063153@mail.gmail.com>
Message-ID: <32DFC62F-D799-4C23-8F99-580CA6DBF6DD@freeswitch.org>

Please open a jira please this did work but a recent change in  
switch_core_codec caused this to appear I usually test this regularly  
but haven't run thru a full run of tests lately.

/b

On Oct 11, 2009, at 9:27 PM, Seven Du wrote:

> I set to true because brian said it can play stereo files but no  
> lucky for me.



From brian at freeswitch.org  Sun Oct 11 20:37:40 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 11 Oct 2009 22:37:40 -0500
Subject: [Freeswitch-users] Re corded file as voicemail.
In-Reply-To: <23f91030910111927v704b2eb9tfd9558068b063153@mail.gmail.com>
References: <25830785.post@talk.nabble.com>
	<20091010040539.GB23188@jdc.jasonjgw.net>
	<23f91030910092240u53b0cc51jd5b1670901f20e21@mail.gmail.com>
	<89F518C6-B470-482D-8C27-2812BDC20C39@freeswitch.org>
	<23f91030910111845yacd54ees40f246ef9152e306@mail.gmail.com>
	<20091012020716.GA8518@jdc.jasonjgw.net>
	<23f91030910111927v704b2eb9tfd9558068b063153@mail.gmail.com>
Message-ID: <D5F3C41A-36B9-4172-B719-8BA824EBFC95@freeswitch.org>

not sure I said this but open a jira... its a bug thats recent...  
usually it will mux them to mono and the codec engine is trying to  
open L16 with two channels so something has changed to cause this  
regression.

Expect a fix sometime tomorrow.

/b

On Oct 11, 2009, at 9:27 PM, Seven Du wrote:

> I set to true because brian said it can play stereo files but no  
> lucky for me.
>
> 2009/10/12 Jason White <jason at jasonjgw.net>
> Seven Du <dujinfang at gmail.com> wrote:
> > originate {ignore_early_media=true,RECORD_STEREO=true}sofia/ 
> gateway/xx/xx
> > &bridge(sofia/gateway/yy/yy)
>
> Shouldn't that be record_stereo=false for mono recording?
>

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From nagalenoj at gmail.com  Sun Oct 11 20:44:02 2009
From: nagalenoj at gmail.com (Nagalenoj)
Date: Sun, 11 Oct 2009 20:44:02 -0700 (PDT)
Subject: [Freeswitch-users] Re  corded file as voicemail.
In-Reply-To: <32DFC62F-D799-4C23-8F99-580CA6DBF6DD@freeswitch.org>
References: <25751158.post@talk.nabble.com>
	<BC536E8C-6813-4A69-9B80-EA11E41A79FC@jerris.com>
	<25830785.post@talk.nabble.com>
	<20091010040539.GB23188@jdc.jasonjgw.net>
	<23f91030910092240u53b0cc51jd5b1670901f20e21@mail.gmail.com>
	<89F518C6-B470-482D-8C27-2812BDC20C39@freeswitch.org>
	<23f91030910111845yacd54ees40f246ef9152e306@mail.gmail.com>
	<20091012020716.GA8518@jdc.jasonjgw.net>
	<23f91030910111927v704b2eb9tfd9558068b063153@mail.gmail.com>
	<32DFC62F-D799-4C23-8F99-580CA6DBF6DD@freeswitch.org>
Message-ID: <25850181.post@talk.nabble.com>


Is it possible to play a stereo file ?!!! whats the conclusion.?!


Brian West-3 wrote:
> 
> Please open a jira please this did work but a recent change in  
> switch_core_codec caused this to appear I usually test this regularly  
> but haven't run thru a full run of tests lately.
> /b
> 
> On Oct 11, 2009, at 9:27 PM, Seven Du wrote:
> 
>> I set to true because brian said it can play stereo files but no  
>> lucky for me.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/Recorded-file-as-voicemail.-tp25751158p25850181.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From nagalenoj at gmail.com  Sun Oct 11 20:46:02 2009
From: nagalenoj at gmail.com (Nagalenoj)
Date: Sun, 11 Oct 2009 20:46:02 -0700 (PDT)
Subject: [Freeswitch-users] Re  corded file as voicemail.
Message-ID: <25850181.post@talk.nabble.com>


Whats the conclusion.?! 


Brian West-3 wrote:
> 
> Please open a jira please this did work but a recent change in  
> switch_core_codec caused this to appear I usually test this regularly  
> but haven't run thru a full run of tests lately.
> /b
> 
> On Oct 11, 2009, at 9:27 PM, Seven Du wrote:
> 
>> I set to true because brian said it can play stereo files but no  
>> lucky for me.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/Recorded-file-as-voicemail.-tp25751158p25850181.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From brian at freeswitch.org  Sun Oct 11 21:11:43 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 11 Oct 2009 23:11:43 -0500
Subject: [Freeswitch-users] Re  corded file as voicemail.
In-Reply-To: <25850181.post@talk.nabble.com>
References: <25850181.post@talk.nabble.com>
Message-ID: <BEDBBF51-2D48-4CDE-A467-11D5BBB458E1@freeswitch.org>

It was possible but we have a regression in the code that isn't  
letting that happen right now... hence the reason i said Open a jira  
so we could fix it.

IS THAT not clear?

/b

On Oct 11, 2009, at 10:46 PM, Nagalenoj wrote:

>
> Whats the conclusion.?!



From srinivas.ksvreddy at gmail.com  Sun Oct 11 21:14:36 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Mon, 12 Oct 2009 09:44:36 +0530
Subject: [Freeswitch-users] mod_sofia.c registered calls how to know
In-Reply-To: <14A1216A-61C3-4C03-B710-57D2E7F4CD7A@jerris.com>
References: <f8af5740910070748g1efee71ft5841b6fa4d2412c8@mail.gmail.com>
	<14A1216A-61C3-4C03-B710-57D2E7F4CD7A@jerris.com>
Message-ID: <f8af5740910112114k3206c3cbq5757e2ff7de81162@mail.gmail.com>

Hi Mike,

Thanks for your valuable reply,
when i install freeswitch1.0.2 in my machine(Windows xp operation system) i
dont have any databasae installed in my system, then from sqllite will come
into picture, and how can i see the registered users data from sqllite.


Thanks
Srinivas

On Mon, Oct 12, 2009 at 3:04 AM, Michael Jerris <mike at jerris.com> wrote:

>
> On Oct 7, 2009, at 10:48 AM, srinivasula reddy wrote:
>
>
> Hi
>
> can any please tell me where registered calls are stored, so when incoming
> call came to mod_sofia.c how it will check it is registered or not?\\
>
>
> Calls are not registered and calls have nothing to do with registration.
>  Users are registered so that you may send calls to them.  Registration data
> is stored either in a sqlite database, or optionally if you setup odbc, in
> another database of your choice.  If you try to send a call to an
> unregistered user in the dialplan using the proper syntax to send calls to
> registered users (see the wiki for more details), and that user is not
> registered, the bridge app will fail, optionally letting you continue on in
> the dialplan based on variables such as continue_on_fail and
> hangup_after_bridge.  You can use the sofia_contact function to see if there
> is anyone registered to a specific user.
>
> Mike
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Srinivasula Reddy K
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From shiyanov at gmail.com  Sun Oct 11 22:38:08 2009
From: shiyanov at gmail.com (Artem Shiyanov)
Date: Mon, 12 Oct 2009 09:38:08 +0400
Subject: [Freeswitch-users] mod_socket: custom timeout for HEARTBEAT
	event
In-Reply-To: <97B4BDC9-5B11-4A98-B1BF-CC00A82ECBD9@jerris.com>
References: <d211f54a0910110959y3d3deed7s43201a3e39756585@mail.gmail.com>
	<86a32abc0910111444j1ab2e446p3cb94b0ae04126e3@mail.gmail.com>
	<97B4BDC9-5B11-4A98-B1BF-CC00A82ECBD9@jerris.com>
Message-ID: <d211f54a0910112238i5f69275ey7cad437e47e6b567@mail.gmail.com>

Michael, Diego,
thanks for the rapid answers!

As far as I understand, "enable_heartbeat" app is launching
SESSION_HEARTBEAT events that will stop when the call will be cleared. Also
I "heard" that "enable_heartbeat" works only for calls with proxied media.

What I want is to monitor FreeSwitch status: is it alive and what is the
system status message. This info is provided in HEARTBEAT event gracefully
but in constant time period = 20 sec. So the main question is- how to
customize this period?


Artem



On Mon, Oct 12, 2009 at 2:52 AM, Michael Jerris <mike at jerris.com> wrote:

>
> On Oct 11, 2009, at 5:44 PM, Diego Viola wrote:
>
> > Maybe you want enable_heartbeat or sched_heartbeat from mod_dptools?
> >
> > You can pass your parameters in second to these two.
> >
> > Example:
> >
> > <action application="enable_heartbeat" data="1"/>
> > <action application="sched_heartbeat" data="1"/>
> >
> > Where 1 in this case is the number of heartbeats per seconds.
> >
>
> Number of seconds between hearbeats, not hearbeats per second.
>
>
> > You can use that example on the Dialplan XML but you can also use it
> > on mod_event_socket outbound, etc.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mattdfong at gmail.com  Sun Oct 11 23:11:32 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Mon, 12 Oct 2009 13:11:32 +0700
Subject: [Freeswitch-users] Sending an Event to a Session for onInput
In-Reply-To: <ACA9A602-0925-4D6F-8B11-0B214A462500@jerris.com>
References: <4256bf830910091234t11f4f4d0w28283f341bd1fb06@mail.gmail.com>
	<ACA9A602-0925-4D6F-8B11-0B214A462500@jerris.com>
Message-ID: <4256bf830910112311t409c45a3q2f26cef3464e49e7@mail.gmail.com>

Hi Mike,
I'm just trying to send it an event with some custom event headers, just so
an external program can communicate with a session without having to
transfer the session to a different program.  I'm curious what uuid_display
does...the wiki only gives a brief description and my Google'ing could not
find any examples. Thanks for the help.

--matt
http://www.hellohunter.com

On Mon, Oct 12, 2009 at 5:04 AM, Michael Jerris <mike at jerris.com> wrote:

> We don't have session messages directly exposed, except for things
> like display, respond, and deflect.  What specifically are you trying
> to send ?
>
> Mike
>
> On Oct 9, 2009, at 3:34 PM, Matthew Fong wrote:
>
> > I'm used to using the onInput callbacks inside lua and javascript to
> > listen for dtmf and other events and perform a task accordingly. I'm
> > wondering if there is a way to send an event to a session or channel
> > that can be caught using the setInputCallback inside lua from
> > outside the session program. Maybe an API command that can generate
> > an event for a specific UUID. Does a mechanism exist to do this that
> > I'm over looking? Thanks.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dujinfang at gmail.com  Sun Oct 11 23:15:30 2009
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 12 Oct 2009 14:15:30 +0800
Subject: [Freeswitch-users] Re corded file as voicemail.
In-Reply-To: <BEDBBF51-2D48-4CDE-A467-11D5BBB458E1@freeswitch.org>
References: <25850181.post@talk.nabble.com>
	<BEDBBF51-2D48-4CDE-A467-11D5BBB458E1@freeswitch.org>
Message-ID: <23f91030910112315h5ea79b81ue55381a3caff0173@mail.gmail.com>

http://jira.freeswitch.org/browse/MODCODEC-15

Is it ok I assigned to you ?

Thanks.

2009/10/12 Brian West <brian at freeswitch.org>

> It was possible but we have a regression in the code that isn't
> letting that happen right now... hence the reason i said Open a jira
> so we could fix it.
>
> IS THAT not clear?
>
> /b
>
> On Oct 11, 2009, at 10:46 PM, Nagalenoj wrote:
>
> >
> > Whats the conclusion.?!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From diego.viola at gmail.com  Sun Oct 11 23:52:07 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Mon, 12 Oct 2009 06:52:07 +0000
Subject: [Freeswitch-users] mod_sofia.c registered calls how to know
In-Reply-To: <f8af5740910112114k3206c3cbq5757e2ff7de81162@mail.gmail.com>
References: <f8af5740910070748g1efee71ft5841b6fa4d2412c8@mail.gmail.com>
	<14A1216A-61C3-4C03-B710-57D2E7F4CD7A@jerris.com>
	<f8af5740910112114k3206c3cbq5757e2ff7de81162@mail.gmail.com>
Message-ID: <86a32abc0910112352q2e5cfac2u23a87f9f92a5431b@mail.gmail.com>

FreeSWITCH 1.0.2?

That's more than a year old I think, you should really update to 1.0.4 or
latest SVN trunk.

Diego

On Mon, Oct 12, 2009 at 4:14 AM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

> Hi Mike,
>
> Thanks for your valuable reply,
> when i install freeswitch1.0.2 in my machine(Windows xp operation system) i
> dont have any databasae installed in my system, then from sqllite will come
> into picture, and how can i see the registered users data from sqllite.
>
>
> Thanks
> Srinivas
>
> On Mon, Oct 12, 2009 at 3:04 AM, Michael Jerris <mike at jerris.com> wrote:
>
>>
>> On Oct 7, 2009, at 10:48 AM, srinivasula reddy wrote:
>>
>>
>> Hi
>>
>> can any please tell me where registered calls are stored, so when incoming
>> call came to mod_sofia.c how it will check it is registered or not?\\
>>
>>
>> Calls are not registered and calls have nothing to do with registration.
>>  Users are registered so that you may send calls to them.  Registration data
>> is stored either in a sqlite database, or optionally if you setup odbc, in
>> another database of your choice.  If you try to send a call to an
>> unregistered user in the dialplan using the proper syntax to send calls to
>> registered users (see the wiki for more details), and that user is not
>> registered, the bridge app will fail, optionally letting you continue on in
>> the dialplan based on variables such as continue_on_fail and
>> hangup_after_bridge.  You can use the sofia_contact function to see if there
>> is anyone registered to a specific user.
>>
>> Mike
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Srinivasula Reddy K
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From velu.technical at gmail.com  Mon Oct 12 00:36:06 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Mon, 12 Oct 2009 13:06:06 +0530
Subject: [Freeswitch-users] Play music on hold after parking the call
In-Reply-To: <1452e2980910100237u317d6138w1d13bd3cfb250353@mail.gmail.com>
References: <1452e2980910100237u317d6138w1d13bd3cfb250353@mail.gmail.com>
Message-ID: <1452e2980910120036k5ab2fb36vbec031c0b0280116@mail.gmail.com>

 Any one please help me to solve the mentioned problem........

On Sat, Oct 10, 2009 at 3:07 PM, velusamy velu <velu.technical at gmail.com>wrote:

> Dear All,
>       I am using ESL.pm module to control the dial plan application. I want
> to play some music while executing the some external scripts. I executed
> park after then I executed the playback the music didn't play.
>
>     Could any one please explain how can I solve this problem without using
> async mode in socket application?
>
> Thanks,
> Velusamy.
>
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From maciej.aniserowicz at gmail.com  Mon Oct 12 01:47:36 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Mon, 12 Oct 2009 01:47:36 -0700 (PDT)
Subject: [Freeswitch-users] Bad sound quality while eavesdropping
In-Reply-To: <CF8C1791-24CD-4132-83B5-7F83B5BDCCC3@jerris.com>
References: <41A44DD027064988A914974405788C2E@procent>
	<191c3a030910050731m2d74979ep4598e5a1945d58ae@mail.gmail.com>
	<1254901192035-3780245.post@n2.nabble.com>
	<8437F5BC-7AFF-4A74-B8CD-C5B8219021F6@jerris.com>
	<1255008427639-3788019.post@n2.nabble.com>
	<191c3a030910080823g79c7c596x1cd887e1538ce2e1@mail.gmail.com>
	<1255169044209-3799274.post@n2.nabble.com>
	<CF8C1791-24CD-4132-83B5-7F83B5BDCCC3@jerris.com>
Message-ID: <1255337256919-3806786.post@n2.nabble.com>


Yes, I confirmed that with Wireshark (filter "rtp and ip.src == <device ip>). RTP packets are sent every 20ms.

MAniserowicz

  ----- Original Message ----- 
  From: Michael Jerris (via Nabble) 
  To: Maciej Aniserowicz 
  Sent: Monday, October 12, 2009 12:21 AM
  Subject: Re: [Freeswitch-users] Bad sound quality while eavesdropping


  can you confirm from an rtp packet trace that they are all really   
  sending 20ms? 

  Mike 

  On Oct 10, 2009, at 6:04 AM, Maciej Aniserowicz wrote: 


  > 
  > Hi, 
  > Here are the messages with a:ptime parameter. All the calls are   
  > started by 
  > commands sent through socket. 
  > I'm not sure if this is all information you need, please let me know   
  > if 
  > something is missing here and I'll post that. 
  > 
  > 1) starting connection with x-lite (number 2003, the eavesdropper): 
  > 
  >   INVITE sip:[hidden email]:60188;rinstance=80b8f8d92af87cd2 SIP/ 
  > 2.0 
  >   Via: SIP/2.0/UDP   
  > 192.168.3.159:15060;rport;branch=z9hG4bKD9meHQN5XQ88K 
  >   Max-Forwards: 69 
  >   From: "MyApp" <sip:[hidden email]>;tag=jpQ6D7D2jUXvF 
  >   To: <sip:[hidden email]:60188;rinstance=80b8f8d92af87cd2> 
  >   Call-ID: 663c4500-3024-122d-57a2-9be8d85a4cff 
  >   CSeq: 121465610 INVITE 
  >   Contact: <sip:[hidden email]:15060> 
  >   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN 
  >   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, 
  > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH 
  >   Supported: timer, precondition, path, replaces 
  >   Allow-Events: talk, presence, dialog, call-info, sla, 
  > include-session-description, presence.winfo, message-summary, refer 
  >   Content-Type: application/sdp 
  >   Content-Disposition: session 
  >   Content-Length: 447 
  >   Remote-Party-ID: "MyApp" 
  > <sip:[hidden email]>;party=calling;screen=yes;privacy=off 
  > 
  >   v=0 
  >   o=FreeSWITCH 2223565947735016740 3096553520713245589 IN IP4   
  > 192.168.3.159 
  >   s=FreeSWITCH 
  >   c=IN IP4 192.168.3.159 
  >   t=0 0 
  >   m=audio 29966 RTP/AVP 0 115 107 9 8 3 101 13 
  >   a=rtpmap:0 PCMU/8000 
  >   a=rtpmap:115 G7221/32000 
  >   a=fmtp:115 bitrate=48000 
  >   a=rtpmap:107 G7221/16000 
  >   a=fmtp:107 bitrate=32000 
  >   a=rtpmap:9 G722/8000 
  >   a=rtpmap:8 PCMA/8000 
  >   a=rtpmap:3 GSM/8000 
  >   a=rtpmap:101 telephone-event/8000 
  >   a=fmtp:101 0-16 
  >   a=rtpmap:13 CN/8000 
  >   a=ptime:20 
  > 
  > 
  > 2) starting connection with cisco ip phone (number 2006, first leg of 
  > eavesdropped session): 
  > 
  >   INVITE sip:[hidden email]:5060;user=phone SIP/2.0 
  >   Via: SIP/2.0/UDP   
  > 192.168.3.159:15060;rport;branch=z9hG4bKg40rp87FNjB1p 
  >   Max-Forwards: 69 
  >   From: "MyApp" <sip:[hidden email]>;tag=Q3N2pe2K47ctS 
  >   To: <sip:[hidden email]:5060;user=phone> 
  >   Call-ID: 6e6fd8a0-3024-122d-57a2-9be8d85a4cff 
  >   CSeq: 121465616 INVITE 
  >   Contact: <sip:[hidden email]:15060> 
  >   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN 
  >   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, 
  > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH 
  >   Supported: timer, precondition, path, replaces 
  >   Allow-Events: talk, presence, dialog, call-info, sla, 
  > include-session-description, presence.winfo, message-summary, refer 
  >   Content-Type: application/sdp 
  >   Content-Disposition: session 
  >   Content-Length: 447 
  >   Remote-Party-ID: "MyApp" 
  > <sip:[hidden email]>;party=calling;screen=yes;privacy=off 
  > 
  >   v=0 
  >   o=FreeSWITCH 8453497903781974949 2294419114567885490 IN IP4   
  > 192.168.3.159 
  >   s=FreeSWITCH 
  >   c=IN IP4 192.168.3.159 
  >   t=0 0 
  >   m=audio 17670 RTP/AVP 0 115 107 9 8 3 101 13 
  >   a=rtpmap:0 PCMU/8000 
  >   a=rtpmap:115 G7221/32000 
  >   a=fmtp:115 bitrate=48000 
  >   a=rtpmap:107 G7221/16000 
  >   a=fmtp:107 bitrate=32000 
  >   a=rtpmap:9 G722/8000 
  >   a=rtpmap:8 PCMA/8000 
  >   a=rtpmap:3 GSM/8000 
  >   a=rtpmap:101 telephone-event/8000 
  >   a=fmtp:101 0-16 
  >   a=rtpmap:13 CN/8000 
  >   a=ptime:20 
  > 
  > 
  > 3) starting connection with extension playing a file (number 9999,   
  > second 
  > leg of eavesdropped session): 
  > 
  >   SIP/2.0 200 OK 
  >   Via: SIP/2.0/UDP 
  > 192.168.3.159:5080;rport=5080;branch=z9hG4bKv6Dg7myj4tvBS 
  >   From: "FreeSWITCH"   
  > <sip:myuser at mydomain;transport=udp>;tag=091j2Q0Fre8vp 
  >   To: <sip:[hidden email]:15060>;tag=U7t5Xt51rB64Q 
  >   Call-ID: 7551b7d0-3024-122d-b3a6-156c4939c4f8 
  >   CSeq: 121465623 INVITE 
  >   Contact: <sip:[hidden email]:15060;transport=udp> 
  >   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN 
  >   Accept: application/sdp 
  >   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, 
  > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH 
  >   Supported: timer, precondition, path, replaces 
  >   Allow-Events: talk, presence, dialog, call-info, sla, 
  > include-session-description, presence.winfo, message-summary, refer 
  >   Content-Type: application/sdp 
  >   Content-Disposition: session 
  >   Content-Length: 263 
  > 
  >   v=0 
  >   o=FreeSWITCH 5614997529598779838 636912243381649698 IN IP4   
  > 192.168.3.159 
  >   s=FreeSWITCH 
  >   c=IN IP4 192.168.3.159 
  >   t=0 0 
  >   m=audio 30086 RTP/AVP 0 101 13 
  >   a=rtpmap:0 PCMU/8000 
  >   a=rtpmap:101 telephone-event/8000 
  >   a=fmtp:101 0-16 
  >   a=rtpmap:13 CN/8000 
  >   a=ptime:20 
  > 
  > 
  > 
  > 
  > Anthony Minessale wrote: 
  >> 
  >> you probably have some device lying about ptime everywhere 
  >> look at a sip trace an pay especially close attention to ptime:x   
  >> param in 
  >> sdp 
  >> if you don't understand this just attach it here 
  >> 
  >> execute the following at the cli 
  >> sofia profile internal siptrace on 
  >> sofila loglevel debug 
  >> 
  >> 
  >> 
  >> On Thu, Oct 8, 2009 at 8:27 AM, Maciej Aniserowicz < 
  >> [hidden email]> wrote: 
  >> 
  >>> 
  >>> It's the same on the trunk (the last rev I used was not so old   
  >>> anyway). 
  >>> 
  >>> Codecs are the same on both legs: 
  >>> read codec/read rate: PCMU      8000 
  >>> write codec/write rate: PCMU    8000 
  >>> 
  >>> MA 
  >>> 
  >>> 
  >>> 
  >>> 
  >>> Michael Jerris wrote: 
  >>>> 
  >>>> What codecs are all the call legs using, also, please try current   
  >>>> svn 
  >>>> trunk. 
  >>>> 
  >>>> Mike 
  >>>> 
  >>>> On Oct 7, 2009, at 3:39 AM, Maciej Aniserowicz wrote: 
  >>>> 
  >>>>> 
  >>>>> Sorry about posting several questions at once, I wasn't aware it's 
  >>>>> "rude". 
  >>>>> Let's concentrate on this issue then. 
  >>>>> 
  >>>>> I use FS rev 14994. Phones on extensions: 
  >>>>> 1) x-lite 
  >>>>> 2) cisco sip phone 
  >>>>> 3) audio played by fs to the extension being eavesdropped 
  >>>>> 
  >>>>> I did not change any codec configuration, I just use the standard 
  >>>>> one that 
  >>>>> comes with both FS and the phones. 
  >>>>> Some time ago someone on FS irc channel told me that this is just 
  >>>>> how FS 
  >>>>> eavesdropping works... from your response I understand that this   
  >>>>> is 
  >>>>> not 
  >>>>> entirely true? 
  >>>>> 
  >>>>> Maciej Aniserowicz 
  >>>>> 
  >>>>> 
  >>>>> 
  >>>>> Anthony Minessale wrote: 
  >>>>>> 
  >>>>>> That's is a somewhat vague position. 
  >>>>>> 
  >>>>>> You did not mention which version of FreeSWITCH you are   
  >>>>>> running, the 
  >>>>>> phones 
  >>>>>> being used in your example, your configuration, the codecs in use 
  >>>>>> etc. 
  >>>>>> 
  >>>>>> BTW, 
  >>>>>> I think you should only ask one question at a time on this list. 
  >>>>>> The list 
  >>>>>> is run by volunteers and it's sort of rude to expect 3 or 4   
  >>>>>> threads 
  >>>>>> to be 
  >>>>>> tended to concerning the same one individual. 
  >>>>>> 
  >>>>>> 
  >>>>>> 2009/10/5 Maciej Aniserowicz <[hidden email]> 
  >>>>>> 
  >>>>>>> Hello, 
  >>>>>>> When I use eavesdropping in FreeSWITCH, the sound quality is 
  >>>>>>> really bad. 
  >>>>>>> Is 
  >>>>>>> there any way to improve it? Is this a known problem? 
  >>>>>>> Br/ 
  >>>>>>> Maciej Aniserowicz 
  >>>>>>> 
  >>>> 
  >>>> 
  >>>> _______________________________________________ 
  >>>> FreeSWITCH-users mailing list 
  >>>> [hidden email] 
  >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  >>>> 
  >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-  
  >>> users 
  >>>> http://www.freeswitch.org
  >>>> 
  >>>> 
  >>> 
  >>> -- 
  >>> View this message in context: 
  >>> http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3788019.html
  >>> Sent from the freeswitch-users mailing list archive at Nabble.com. 
  >>> 
  >>> _______________________________________________ 
  >>> FreeSWITCH-users mailing list 
  >>> [hidden email] 
  >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-  
  >>> users 
  >>> http://www.freeswitch.org
  >>> 
  >> 
  >> 
  >> 
  >> -- 
  >> Anthony Minessale II 
  >> 
  >> FreeSWITCH http://www.freeswitch.org/
  >> ClueCon http://www.cluecon.com/
  >> Twitter: http://twitter.com/FreeSWITCH_wire
  >> 
  >> AIM: anthm 
  >> MSN:[hidden email] <MSN 
  >> %[hidden email]> 
  >> GTALK/JABBER/PAYPAL:[hidden email]<PAYPAL 
  >> %[hidden email]> 
  >> IRC: irc.freenode.net #freeswitch 
  >> 
  >> FreeSWITCH Developer Conference 
  >> sip:[hidden email] <sip 
  >> %[hidden email]> 
  >> iax:[hidden email]/888 
  >> googletalk:[hidden email]<googletalk%3Aconf 
  >> %[hidden email]> 
  >> pstn:213-799-1400 
  >> 
  >> _______________________________________________ 
  >> FreeSWITCH-users mailing list 
  >> [hidden email] 
  >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-  
  >> users 
  >> http://www.freeswitch.org
  >> 
  >> 
  > 
  > -- 
  > View this message in context: http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3799274.html
  > Sent from the freeswitch-users mailing list archive at Nabble.com. 
  > 
  > _______________________________________________ 
  > FreeSWITCH-users mailing list 
  > [hidden email] 
  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-  
  > users 
  > http://www.freeswitch.org


  _______________________________________________ 
  FreeSWITCH-users mailing list 
  [hidden email] 
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org



------------------------------------------------------------------------------

  View message @ http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3805109.html 
  To unsubscribe from Re: Bad sound quality while eavesdropping, click here. 


-- 
View this message in context: http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3806786.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
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From dujinfang at gmail.com  Mon Oct 12 04:19:53 2009
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 12 Oct 2009 19:19:53 +0800
Subject: [Freeswitch-users] mod_sofia.c registered calls how to know
In-Reply-To: <f8af5740910112114k3206c3cbq5757e2ff7de81162@mail.gmail.com>
References: <f8af5740910070748g1efee71ft5841b6fa4d2412c8@mail.gmail.com>
	<14A1216A-61C3-4C03-B710-57D2E7F4CD7A@jerris.com>
	<f8af5740910112114k3206c3cbq5757e2ff7de81162@mail.gmail.com>
Message-ID: <23f91030910120419o74f53279v5f061988ff585d56@mail.gmail.com>

try open YOUR_FreeSWITCH_INSTALL_DIR/db/*.db, you need sqlite3 to open them.
not sure how to do that on windows, but on linux:


# sqlite3 xx.db
sqlite> select * from sip_registration;

2009/10/12 srinivasula reddy <srinivas.ksvreddy at gmail.com>

> Hi Mike,
>
> Thanks for your valuable reply,
> when i install freeswitch1.0.2 in my machine(Windows xp operation system) i
> dont have any databasae installed in my system, then from sqllite will come
> into picture, and how can i see the registered users data from sqllite.
>
>
> Thanks
> Srinivas
>
> On Mon, Oct 12, 2009 at 3:04 AM, Michael Jerris <mike at jerris.com> wrote:
>
>>
>> On Oct 7, 2009, at 10:48 AM, srinivasula reddy wrote:
>>
>>
>> Hi
>>
>> can any please tell me where registered calls are stored, so when incoming
>> call came to mod_sofia.c how it will check it is registered or not?\\
>>
>>
>> Calls are not registered and calls have nothing to do with registration.
>>  Users are registered so that you may send calls to them.  Registration data
>> is stored either in a sqlite database, or optionally if you setup odbc, in
>> another database of your choice.  If you try to send a call to an
>> unregistered user in the dialplan using the proper syntax to send calls to
>> registered users (see the wiki for more details), and that user is not
>> registered, the bridge app will fail, optionally letting you continue on in
>> the dialplan based on variables such as continue_on_fail and
>> hangup_after_bridge.  You can use the sofia_contact function to see if there
>> is anyone registered to a specific user.
>>
>> Mike
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Srinivasula Reddy K
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From elihay at savion.huji.ac.il  Mon Oct 12 04:37:12 2009
From: elihay at savion.huji.ac.il (Eli Hayun)
Date: Mon, 12 Oct 2009 13:37:12 +0200
Subject: [Freeswitch-users] mod_sofia.c registered calls how to know
In-Reply-To: <23f91030910120419o74f53279v5f061988ff585d56@mail.gmail.com>
References: <f8af5740910070748g1efee71ft5841b6fa4d2412c8@mail.gmail.com>
	<14A1216A-61C3-4C03-B710-57D2E7F4CD7A@jerris.com>
	<f8af5740910112114k3206c3cbq5757e2ff7de81162@mail.gmail.com>
	<23f91030910120419o74f53279v5f061988ff585d56@mail.gmail.com>
Message-ID: <1255347432.3724.4.camel@eli-desktop>

Is it possible to keep a list of registered phone, and when FS will
start it will register them all automatically?


On Mon, 2009-10-12 at 13:19 +0200, Seven Du wrote:

> try open YOUR_FreeSWITCH_INSTALL_DIR/db/*.db, you need sqlite3 to open
> them. not sure how to do that on windows, but on linux:
> 
> 
> # sqlite3 xx.db
> sqlite> select * from sip_registration;
> 
> 
> 2009/10/12 srinivasula reddy <srinivas.ksvreddy at gmail.com>
> 
>         Hi Mike,
>         
>         Thanks for your valuable reply,
>         when i install freeswitch1.0.2 in my machine(Windows xp
>         operation system) i dont have any databasae installed in my
>         system, then from sqllite will come into picture, and how can
>         i see the registered users data from sqllite.
>         
>         
>         Thanks
>         Srinivas
>         
>         
>         
>         On Mon, Oct 12, 2009 at 3:04 AM, Michael Jerris
>         <mike at jerris.com> wrote:
>         
>                 
>                 
>                 
>                 On Oct 7, 2009, at 10:48 AM, srinivasula reddy wrote:
>                 
>                 
>                 
>                 > 
>                 > Hi
>                 > 
>                 > can any please tell me where registered calls are
>                 > stored, so when incoming call came to mod_sofia.c
>                 > how it will check it is registered or not?\\
>                 > 
>                 
>                 
>                 Calls are not registered and calls have nothing to do
>                 with registration.  Users are registered so that you
>                 may send calls to them.  Registration data is stored
>                 either in a sqlite database, or optionally if you
>                 setup odbc, in another database of your choice.  If
>                 you try to send a call to an unregistered user in the
>                 dialplan using the proper syntax to send calls to
>                 registered users (see the wiki for more details), and
>                 that user is not registered, the bridge app will fail,
>                 optionally letting you continue on in the dialplan
>                 based on variables such as continue_on_fail and
>                 hangup_after_bridge.  You can use the sofia_contact
>                 function to see if there is anyone registered to a
>                 specific user.
>                 
>                 
>                 Mike
>                 
>                 
>                 
>                 
>                 
>                 _______________________________________________
>                 FreeSWITCH-users mailing list
>                 FreeSWITCH-users at lists.freeswitch.org
>                 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>                 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>                 http://www.freeswitch.org
>                 
>                 
>         
>         
>         
>         
>         -- 
>         Srinivasula Reddy K
>         
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>         
> 
> 
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From lakindia89 at gmail.com  Mon Oct 12 04:54:18 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Mon, 12 Oct 2009 17:24:18 +0530
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <87f2f3b90910101009j1295d812jd98cae91e30b1eff@mail.gmail.com>
References: <7d79b3930910090407m5dc81d12pf2b0a47f6017804@mail.gmail.com>
	<20091009123615.BC50A3FA6E9@mail.cune.org>
	<7d79b3930910100647u7fd11413ucec0a535d68b3e3a@mail.gmail.com>
	<87f2f3b90910101009j1295d812jd98cae91e30b1eff@mail.gmail.com>
Message-ID: <7d79b3930910120454m1ffc6dd4u42b91a4349597166@mail.gmail.com>

Ya ok. Here is the required stuff
Configuration: (PRI span)
<configuration name="openzap.conf" description="OpenZAP Configuration">
<settings>
<param name="debug" value="7"/>
</settings>
<pri_spans>
<span id="1" name="PRI_1">
<param name="q921loglevel" value="alert"/>
<param name="q931loglevel" value="alert"/>
<param name="mode" value="user"/>
<param name="dialect" value="q931"/>
<param name="dialplan" value="XML"/>
<param name="context" value="default"/>
</span>
</pri_spans>
</configuration>

Log while starting freeswitch:
http://pastebin.freeswitch.org/10646

Log when making a call:
http://pastebin.freeswitch.org/10647


Configuration: (LIBPRI SPAN)
<configuration name="openzap.conf" description="OpenZAP Configuration">
<settings>
<param name="debug" value="7"/>
</settings>
<libpri_spans>
<span name="PRI_1">
<param name="node" value="cpe"/>
<param name="switch" value="euroisdn"/>
<param name="dialplan" value="XML"/>
<param name="context" value="default"/>
</span>
</libpri_spans>
</configuration>

Log while starting freeswitch:
http://pastebin.freeswitch.org/10648

Log while making a call:
http://pastebin.freeswitch.org/10649



On Sat, Oct 10, 2009 at 10:39 PM, Michael Collins <msc at freeswitch.org>wrote:

> Okay, please go to pastebin.freeswitch.org and paste your openzap.conf.xml
> file. Also, paste the ENTIRE debug log from a call from start to finish.
> Telling us that you see INVALID_IE_CONTENTS doesn't help if we don't know
> what the information element contains. Finally, turn on PRI debugging and
> make another test call and pastebin that debug log as well. The debug will
> show details about the communications between your machine and the carrier.
> Instructions for turning on debugging for libpri are found in the OpenZAP
> wiki page in the same place where the libpri instructions are located.
>
> Put the pastebin number in this email thread and then we'll go have a look.
>
> Thanks,
> MC
>
>
> On Sat, Oct 10, 2009 at 6:47 AM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> Hi,
>> I've tried with the following dialplan(After making the changes that are
>> recommended). But still I got INVALID_IE_CONTENTS error.
>>
>> Another thing is, if I use prispan configuration, I'm able to make
>> outgoing calls.
>> But if I use the libpri span configuration I'm not able to make outgoing
>> calls, which says INVALID_IE_CONTENTS.
>>
>> I've been struck with this problem for the past 1 week.
>> Any solution to this???
>>
>>
>>
>> On Fri, Oct 9, 2009 at 6:06 PM, <Russell.Mosemann at cune.org> wrote:
>>
>>> lakshmanan ganapathy <lakindia89 at gmail.com> said:
>>>
>>> > But still I'm facing problem with the outgoing call. It says
>>> > INVALID_IE_CONTENTS.
>>> > What might be the issue? Even I tried the following dialplan to call by
>>> > using bridge.
>>> > <extension name="outgoing-fxo">
>>> >                   <condition field="destination_number"
>>> > expression="^(\d{10})$">
>>> >                           <action application="answer"/>
>>> >                           <action application="bridge"
>>> > data="openzap/1/1/${dialed_ext}"/>
>>>
>>> Does "answer" need to be called here? I haven't used an fxo. So, I don't
>>> know. What value does $dialed_ext have? If you want to use the number
>>> matched in the condition, then it should be
>>>
>>> openzap/1/1/$1
>>>
>>> --
>>> Russell Mosemann
>>>
>>>
>>>
>>> ________________________________________________________
>>> Concordia University, Nebraska
>>> See http://www.cune.edu/ for the latest news and events!
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Mon Oct 12 06:16:27 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Oct 2009 08:16:27 -0500
Subject: [Freeswitch-users] Re corded file as voicemail.
In-Reply-To: <23f91030910112315h5ea79b81ue55381a3caff0173@mail.gmail.com>
References: <25850181.post@talk.nabble.com>
	<BEDBBF51-2D48-4CDE-A467-11D5BBB458E1@freeswitch.org>
	<23f91030910112315h5ea79b81ue55381a3caff0173@mail.gmail.com>
Message-ID: <5CCE4643-063D-4F2D-B98F-35058F465C10@freeswitch.org>

Perfect! Thanks.

/b

On Oct 12, 2009, at 1:15 AM, Seven Du wrote:

> http://jira.freeswitch.org/browse/MODCODEC-15
>
> Is it ok I assigned to you ?
>
> Thanks.

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From brian at freeswitch.org  Mon Oct 12 06:23:20 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Oct 2009 08:23:20 -0500
Subject: [Freeswitch-users] Bad sound quality while eavesdropping
In-Reply-To: <1255337256919-3806786.post@n2.nabble.com>
References: <41A44DD027064988A914974405788C2E@procent>
	<191c3a030910050731m2d74979ep4598e5a1945d58ae@mail.gmail.com>
	<1254901192035-3780245.post@n2.nabble.com>
	<8437F5BC-7AFF-4A74-B8CD-C5B8219021F6@jerris.com>
	<1255008427639-3788019.post@n2.nabble.com>
	<191c3a030910080823g79c7c596x1cd887e1538ce2e1@mail.gmail.com>
	<1255169044209-3799274.post@n2.nabble.com>
	<CF8C1791-24CD-4132-83B5-7F83B5BDCCC3@jerris.com>
	<1255337256919-3806786.post@n2.nabble.com>
Message-ID: <59F3CD44-5FEA-403C-98BE-EEE49EC3815B@freeswitch.org>

Did you open a jira and attach all the info?

/b

On Oct 12, 2009, at 3:47 AM, Maciej Aniserowicz wrote:

> Yes, I confirmed that with Wireshark (filter "rtp and ip.src ==  
> <device ip>). RTP packets are sent every 20ms.
>
> MAniserowicz
>

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From bottleman at icf.org.ru  Mon Oct 12 06:31:03 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Mon, 12 Oct 2009 17:31:03 +0400 (MSD)
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>

On 2009-10-08 20:32 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>Hi Yuriy,
TC>
TC>can you share what you have so far, I'm sure we can help with RTP part...

ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/ alfa code, but seems it work, but should be buggy, 
to build need libpt 2.6.5 and h323plus cvs version, i test it now on fs 1.0.4.

TC>
TC>T.
TC>
TC>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
TC>
TC>> On 2009-10-08 13:25 -0400, Tuyan ?zipek wrote
TC>> freeswitch-users at lists.freesw...:
TC>>
TC>> Tz>Hi,
TC>> Tz>
TC>> Tz>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>:
TC>> Tz>> On 2009-10-08 10:43 -0500, Anthony Minessale wrote
TC>> freeswitch-users at lists.f...:
TC>> Tz>>
TC>> Tz>> AM>If you are going to make that alternate module are you going to
TC>> host it in
TC>> Tz>> AM>the FS tree along side mod_opal?
TC>> Tz>>
TC>> Tz>> Yes, but then it be useful, now i have working only signaling part and
TC>> some
TC>> Tz>> kinde of not working rtp part :)
TC>> Tz>
TC>> Tz>If you dont use fs rtp stack, its unlikely that it will be accepted
TC>> Tz>into the tree.
TC>> Tz>
TC>> Tz>>
TC>> Tz>> AM>also if were working on mod_opal why did you not try to involve us
TC>> and the
TC>> Tz>> AM>opal team?
TC>> Tz>>
TC>> Tz>> Because i made patches for libopal, one is a bugfix in rtp part, there
TC>> is a race condition
TC>> Tz>> in inicialisation in jitter buffer, another patch implements method to
TC>> send progress indicator,
TC>> Tz>> and i don't wont spent my time to incorporate this changes into
TC>> libopal.
TC>> Tz>
TC>> Tz>Thats bad.
TC>> Tz>Any bugfixes from fs, goes to upstream on any of the used libraries.
TC>> Tz>You should be doing the same.
TC>> Tz>And Opal developers, will either include or refuse your patches. If
TC>> Tz>they refuse it, they will give you the reason.
TC>>
TC>> i make this fix only to freeze my current mod_opal work on working state,
TC>> while it now work for me i work on
TC>> my new implementation of h323 proto for fs, i think opal developers will
TC>> fix this rtp bug himself becouse
TC>> it crashes and make library unuseful.
TC>>
TC>> Tz>
TC>> Tz>> without this changes
TC>> Tz>> my work on mod_opal in freeswitch don't useful at all, i provide link
TC>> to my work with all
TC>> Tz>> patches, if somebody wont incorporate it in libopal tree and fs - go
TC>> on, but i think
TC>> Tz>> better and more elegant make new module based on h323plus.
TC>> Tz>
TC>> Tz>If you dont publish your changes, all those you are trying to achieve,
TC>> Tz>wont happen.
TC>> Tz>
TC>> Tz>>
TC>> Tz>> AM>How far away from what is in tree are these patches you have?
TC>> Tz>> AM>
TC>> Tz>> AM>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
TC>> Tz>> AM>
TC>> Tz>> AM>> On 2009-10-07 15:09 -0500, Brian West wrote
TC>> Tz>> AM>> freeswitch-users at lists.freeswit...:
TC>> Tz>> AM>>
TC>> Tz>> AM>> opal have addition abstraction layer called opalmgr, and it
TC>> implementation
TC>> Tz>> AM>> is not so good in
TC>> Tz>> AM>> this case, for example to implemet pre_answer in mod_opal i need
TC>> patch
TC>> Tz>> AM>> libopal, because
TC>> Tz>
TC>> Tz>The patch you have in there, adds a method to the OpalCall,  it does
TC>> Tz>not touch any parts of OpalManager
TC>> Tz>so, i dont understand why opalmanager would be the cause of your pain?
TC>> Tz>
TC>> Tz>> AM>> there is no way to send progress inicator throuch opalmgr. and
TC>> there is
TC>> Tz>> AM>> many another issues like
TC>> Tz>> AM>> this in that layer.
TC>> Tz>
TC>> Tz>Please point me to the issues you have in opal, their bug reports ,
TC>> traces etc.
TC>> Tz>I dont think any of the opal people has psychic abilities to detect
TC>> Tz>-your- problems
TC>> Tz>and solve them.
TC>> Tz>
TC>> Tz>>ftp://srv.icf.org.ru/pub/soft/f/freeswitch/ - there is
TC>> Tz>> AM>> my work on mod_opal before
TC>> Tz>> AM>> i start moving to h323plus, may be this help somebody there.
TC>> Tz>> AM>>
TC>> Tz>> AM>> BW> From what I have been told h323plus is a based/fork of
TC>> OpenH323 which
TC>> Tz>> AM>> BW>OPAL is just a continuation of OpenH323.  So why not support
TC>> the
TC>> Tz>> AM>> BW>developers of OPAL/OpenH323 ?
TC>> Tz>> AM>> BW>
TC>> Tz>> AM>> BW>/b
TC>> Tz>> AM>> BW>
TC>> Tz>> AM>> BW>On Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote:
TC>> Tz>> AM>> BW>
TC>> Tz>> AM>> BW>> We are developing module to handle h323 proto now, we try to
TC>> use
TC>> Tz>> AM>> BW>> mod_opal and try improve it, but no luck,
TC>> Tz>> AM>> BW>> there is many issues in libopal, and finaly we now move to
TC>> h323plus
TC>> Tz>> AM>> BW>> library.
TC>> Tz>
TC>> Tz>Did any of you try to report those issues?
TC>> Tz>
TC>> Tz>Regards
TC>> Tz>
TC>> Tz>/tyn
TC>> Tz>
TC>> Tz>> AM>> BW>
TC>> Tz>> AM>> BW>
TC>> Tz>> AM>> BW>_______________________________________________
TC>> Tz>> AM>> BW>FreeSWITCH-users mailing list
TC>> Tz>> AM>> BW>FreeSWITCH-users at lists.freeswitch.org
TC>> Tz>> AM>> BW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> Tz>> AM>> BW>UNSUBSCRIBE:
TC>> Tz>> AM>> http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> Tz>> AM>> BW>http://www.freeswitch.org
TC>> Tz>> AM>> BW>
TC>> Tz>> AM>>
TC>> Tz>> AM>> C ?????????                       With Best Regards
TC>> Tz>> AM>> ???????????? ????.                Georgiewskiy Yuriy
TC>> Tz>> AM>> +7 4872 711666                    +7 4872 711666
TC>> Tz>> AM>> ???? +7 4872 711143               fax +7 4872 711143
TC>> Tz>> AM>> ???????? ??? "?? ?? ??????"       IT Service Ltd
TC>> Tz>> AM>> http://nkoort.ru                  http://nkoort.ru
TC>> Tz>> AM>> JID: GHhost at jabber.tula-ix.net.ru JID:
TC>> GHhost at jabber.tula-ix.net.ru
TC>> Tz>> AM>> YG129-RIPE                        YG129-RIPE
TC>> Tz>> AM>>
TC>> Tz>> AM>> _______________________________________________
TC>> Tz>> AM>> FreeSWITCH-users mailing list
TC>> Tz>> AM>> FreeSWITCH-users at lists.freeswitch.org
TC>> Tz>> AM>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> Tz>> AM>> UNSUBSCRIBE:
TC>> http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> Tz>> AM>> http://www.freeswitch.org
TC>> Tz>> AM>>
TC>> Tz>> AM>>
TC>> Tz>> AM>
TC>> Tz>> AM>
TC>> Tz>> AM>
TC>> Tz>>
TC>> Tz>> C ?????????                       With Best Regards
TC>> Tz>> ???????????? ????.                Georgiewskiy Yuriy
TC>> Tz>> +7 4872 711666                    +7 4872 711666
TC>> Tz>> ???? +7 4872 711143               fax +7 4872 711143
TC>> Tz>> ???????? ??? "?? ?? ??????"       IT Service Ltd
TC>> Tz>> http://nkoort.ru                  http://nkoort.ru
TC>> Tz>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
TC>> Tz>> YG129-RIPE                        YG129-RIPE
TC>> Tz>> _______________________________________________
TC>> Tz>> FreeSWITCH-users mailing list
TC>> Tz>> FreeSWITCH-users at lists.freeswitch.org
TC>> Tz>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> Tz>> UNSUBSCRIBE:
TC>> http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> Tz>> http://www.freeswitch.org
TC>> Tz>>
TC>> Tz>>
TC>> Tz>
TC>> Tz>_______________________________________________
TC>> Tz>FreeSWITCH-users mailing list
TC>> Tz>FreeSWITCH-users at lists.freeswitch.org
TC>> Tz>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> Tz>UNSUBSCRIBE:
TC>> http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> Tz>http://www.freeswitch.org
TC>> Tz>
TC>>
TC>> C ?????????                       With Best Regards
TC>> ???????????? ????.                Georgiewskiy Yuriy
TC>> +7 4872 711666                    +7 4872 711666
TC>> ???? +7 4872 711143               fax +7 4872 711143
TC>> ???????? ??? "?? ?? ??????"       IT Service Ltd
TC>> http://nkoort.ru                  http://nkoort.ru
TC>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
TC>> YG129-RIPE                        YG129-RIPE
TC>>
TC>> _______________________________________________
TC>> FreeSWITCH-users mailing list
TC>> FreeSWITCH-users at lists.freeswitch.org
TC>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> http://www.freeswitch.org
TC>>
TC>>
TC>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From anthony.minessale at gmail.com  Mon Oct 12 07:26:18 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 12 Oct 2009 09:26:18 -0500
Subject: [Freeswitch-users] mod_socket: custom timeout for HEARTBEAT
	event
In-Reply-To: <d211f54a0910112238i5f69275ey7cad437e47e6b567@mail.gmail.com>
References: <d211f54a0910110959y3d3deed7s43201a3e39756585@mail.gmail.com>
	<86a32abc0910111444j1ab2e446p3cb94b0ae04126e3@mail.gmail.com>
	<97B4BDC9-5B11-4A98-B1BF-CC00A82ECBD9@jerris.com>
	<d211f54a0910112238i5f69275ey7cad437e47e6b567@mail.gmail.com>
Message-ID: <191c3a030910120726p46d6410ejd94b61612686fbbc@mail.gmail.com>

it works in either case with or without media
the syntax for setting the frequency was answered above.


On Mon, Oct 12, 2009 at 12:38 AM, Artem Shiyanov <shiyanov at gmail.com> wrote:

> Michael, Diego,
> thanks for the rapid answers!
>
> As far as I understand, "enable_heartbeat" app is launching
> SESSION_HEARTBEAT events that will stop when the call will be cleared. Also
> I "heard" that "enable_heartbeat" works only for calls with proxied media.
>
> What I want is to monitor FreeSwitch status: is it alive and what is the
> system status message. This info is provided in HEARTBEAT event gracefully
> but in constant time period = 20 sec. So the main question is- how to
> customize this period?
>
>
> Artem
>
>
>
>
> On Mon, Oct 12, 2009 at 2:52 AM, Michael Jerris <mike at jerris.com> wrote:
>
>>
>> On Oct 11, 2009, at 5:44 PM, Diego Viola wrote:
>>
>> > Maybe you want enable_heartbeat or sched_heartbeat from mod_dptools?
>> >
>> > You can pass your parameters in second to these two.
>> >
>> > Example:
>> >
>> > <action application="enable_heartbeat" data="1"/>
>> > <action application="sched_heartbeat" data="1"/>
>> >
>> > Where 1 in this case is the number of heartbeats per seconds.
>> >
>>
>> Number of seconds between hearbeats, not hearbeats per second.
>>
>>
>> > You can use that example on the Dialplan XML but you can also use it
>> > on mod_event_socket outbound, etc.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From tculjaga at gmail.com  Mon Oct 12 07:27:25 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 12 Oct 2009 16:27:25 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
Message-ID: <65d96fc80910120727q6d35b51ard78005312079e021@mail.gmail.com>

2009/10/12 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-08 20:32 +0200, Tihomir Culjaga wrote
> freeswitch-users at lists.fre...:
>
> TC>Hi Yuriy,
> TC>
> TC>can you share what you have so far, I'm sure we can help with RTP
> part...
>
> ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/ alfa code, but seems
> it work, but should be buggy,
> to build need libpt 2.6.5 and h323plus cvs version, i test it now on fs
> 1.0.4.
>
> TC>
> TC>T.
> TC>
> TC>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
>

got it and building it right now...

T.
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From ngay01042005 at gmail.com  Mon Oct 12 04:01:10 2009
From: ngay01042005 at gmail.com (homqua)
Date: Mon, 12 Oct 2009 04:01:10 -0700 (PDT)
Subject: [Freeswitch-users]  Question about fax tone detection
Message-ID: <1255345270255-3807298.post@n2.nabble.com>


Hi,
I have implemented the solution for tone detection in wiki, and also answer
the channel before detecting the tone:
<condition field="destination_number" expression="^(055138419992)$">
    <action application="answer"/>
    <action application="tone_detect" data="fax 1100 r +15000 transfer fax
XML default"/>


<extension name="fax">
      <condition field="destination_number" expression="^fax$">
        <action application="answer" />
        <action application="sleep" data="1000"/>
        <action application="rxfax"
data="/usr/local/freeswitch/storage/fax/${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}.tiff"/>
    <action application="set" data="fax_mode=recv"/>
        <action application="hangup"/>
      </condition>
    </extension>


But FS cannot recognize the tone, and therefore cannot move to fax
extension.  Below are the error in FS:

2009-10-12 10:57:16.702287 [NOTICE] switch_channel.c:602 New Channel
sofia/external/anonymous at anonymous.invalid
[c431f0a3-9231-4724-ba39-9e4ef7edfca2]
2009-10-12 10:57:16.703413 [INFO] mod_dialplan_xml.c:315 Processing
Anonymous->055138419992 in context public
2009-10-12 10:57:16.719288 [NOTICE] switch_ivr.c:1349 Transfer
sofia/external/anonymous at anonymous.invalid to XML[055138419992 at default]
2009-10-12 10:57:16.719288 [INFO] mod_dialplan_xml.c:315 Processing
Anonymous->055138419992 in context default
2009-10-12 10:57:16.722289 [NOTICE] mod_dptools.c:649 Channel
[sofia/external/anonymous at anonymous.invalid] has been answered
2009-10-12 10:57:16.722289 [NOTICE] mod_dptools.c:1324 Enabling tone
detection 'fax' '1100'
2009-10-12 10:57:16.723302 [NOTICE] switch_core_state_machine.c:179 Hangup
sofia/external/anonymous at anonymous.invalid [CS_EXECUTE] [NORMAL_CLEARING]
2009-10-12 10:57:16.740285 [NOTICE] switch_core_session.c:1086 Session 1
(sofia/external/anonymous at anonymous.invalid) Ended
2009-10-12 10:57:16.740285 [NOTICE] switch_core_session.c:1088 Close Channel
sofia/external/anonymous at anonymous.invalid [CS_DESTROY]

And the trace for SIP messages:  http://pastebin.com/m4e47e7d9 

If anyone has any idea, tell me please.
Thanks.
-- 
View this message in context: http://n2.nabble.com/Question-about-fax-tone-detection-tp3807298p3807298.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From mattdfong at gmail.com  Mon Oct 12 07:41:30 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Mon, 12 Oct 2009 21:41:30 +0700
Subject: [Freeswitch-users] hangup_after_bridge=false not continueing after
	failed bridge
Message-ID: <4256bf830910120741g5facaa6cr7ed92303dfafe74b@mail.gmail.com>

I think think this might be a bug, but wanted to post here instead of Jira
in-case I'm overlooking a configuration variable
Dialplan

    <extension name="1920"><!--init agent for manual and power dial mode-->
      <condition field="destination_number" expression="^1920$">
        <action application="set" data="hangup_after_bridge=false"/>
        <action application="bridge" data="sofia/gateway/
debug.com/14159927717"/>
        <action application="transfer" data="1999"/><!-- send to unable to
reach any contacts-->
      </condition>
    </extension>

API Command
originate sofia/internal/sip_1%192.168.1.10 1920

When the bridge to 14159927717 fails (NO_ANSWER) both calls are terminated
instead of continuing on in the dial plan to exten 1999 (which in my
dialplan parks the call). hangup_after_bridge however seems to work OK if
someone picks up in the bridge. Is this the correct behavior? How else can I
prevent the call from hanging up if a bridge fails? Thanks.

I'm using 15135M

--matt
http://www.hellohunter.com - Predictive Dialer
http://www.hellohunter.com/voice_broadcast.php - Voice Broadcasting
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From brian at freeswitch.org  Mon Oct 12 07:43:46 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Oct 2009 09:43:46 -0500
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
Message-ID: <39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>

We can host this in our SVN if you wish?

/b

On Oct 12, 2009, at 8:31 AM, Georgiewskiy Yuriy wrote:

> ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/ alfa code, but  
> seems it work, but should be buggy,
> to build need libpt 2.6.5 and h323plus cvs version, i test it now on  
> fs 1.0.4.

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From bottleman at icf.org.ru  Mon Oct 12 07:53:18 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Mon, 12 Oct 2009 18:53:18 +0400 (MSD)
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
Message-ID: <Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>

On 2009-10-12 09:43 -0500, Brian West wrote freeswitch-users at lists.freeswit...:

BW>We can host this in our SVN if you wish?

If in fs svn i think yes. But i think may be little time later? 
i don't known is it builds on trunk because i develop it on 1.0.4.

BW>/b
BW>
BW>On Oct 12, 2009, at 8:31 AM, Georgiewskiy Yuriy wrote:
BW>
BW>> ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/ alfa code, but seems
BW>> it work, but should be buggy,
BW>> to build need libpt 2.6.5 and h323plus cvs version, i test it now on fs
BW>> 1.0.4.
BW>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From anthony.minessale at gmail.com  Mon Oct 12 08:06:54 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 12 Oct 2009 10:06:54 -0500
Subject: [Freeswitch-users] hangup_after_bridge=false not continueing
	after failed bridge
In-Reply-To: <4256bf830910120741g5facaa6cr7ed92303dfafe74b@mail.gmail.com>
References: <4256bf830910120741g5facaa6cr7ed92303dfafe74b@mail.gmail.com>
Message-ID: <191c3a030910120806o30911131o743aa58420b41ae4@mail.gmail.com>

which line is hanging up your A (inbound) leg?

look for a blue line that says "Hangup xyz...." that matches it so i can
see.

I think what is happening is you are getting early media so the bridge is
actually working then when nobody answers it dies but technically the bridge
worked.

On Mon, Oct 12, 2009 at 9:41 AM, Matthew Fong <mattdfong at gmail.com> wrote:

> I think think this might be a bug, but wanted to post here instead of Jira
> in-case I'm overlooking a configuration variable
> Dialplan
>
>     <extension name="1920"><!--init agent for manual and power dial mode-->
>       <condition field="destination_number" expression="^1920$">
>         <action application="set" data="hangup_after_bridge=false"/>
>         <action application="bridge" data="sofia/gateway/
> debug.com/14159927717"/>
>         <action application="transfer" data="1999"/><!-- send to unable to
> reach any contacts-->
>       </condition>
>     </extension>
>
> API Command
> originate sofia/internal/sip_1%192.168.1.10 1920
>
> When the bridge to 14159927717 fails (NO_ANSWER) both calls are terminated
> instead of continuing on in the dial plan to exten 1999 (which in my
> dialplan parks the call). hangup_after_bridge however seems to work OK if
> someone picks up in the bridge. Is this the correct behavior? How else can I
> prevent the call from hanging up if a bridge fails? Thanks.
>
> I'm using 15135M
>
> --matt
> http://www.hellohunter.com - Predictive Dialer
> http://www.hellohunter.com/voice_broadcast.php - Voice Broadcasting
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mattdfong at gmail.com  Mon Oct 12 08:23:54 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Mon, 12 Oct 2009 22:23:54 +0700
Subject: [Freeswitch-users] hangup_after_bridge=false not continueing
	after failed bridge
In-Reply-To: <191c3a030910120806o30911131o743aa58420b41ae4@mail.gmail.com>
References: <4256bf830910120741g5facaa6cr7ed92303dfafe74b@mail.gmail.com>
	<191c3a030910120806o30911131o743aa58420b41ae4@mail.gmail.com>
Message-ID: <4256bf830910120823w5fcbc3eja4afb76b60b68fda@mail.gmail.com>

2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179 Hangup
sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]

might be the line..or the entire output is below....

freeswitch at matthew-laptop> originate sofia/internal/sip_1%192.168.1.10 1920
2009-10-12 15:21:44.029517 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/sip_1 [1e722934-7e94-46aa-9d62-e6ec7e7449cf]
2009-10-12 15:21:44.121484 [NOTICE] sofia.c:3552 Ring-Ready
sofia/internal/sip_1!
2009-10-12 15:21:47.285531 [NOTICE] sofia.c:3998 Channel
[sofia/internal/sip_1] has been answered
2009-10-12 15:21:47.290996 [INFO] mod_dialplan_xml.c:391 Processing
FreeSWITCH->1920 in context default
2009-10-12 15:21:47.293452 [NOTICE] switch_channel.c:613 New Channel
sofia/external/14159927717 [6b6cc440-e1d6-415a-b84b-494117e7361d]
2009-10-12 15:21:47.293452 [NOTICE] switch_ivr.c:1367 Transfer
sofia/internal/sip_1 to XML[1920 at default]
API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
+OK 1e722934-7e94-46aa-9d62-e6ec7e7449cf

freeswitch at matthew-laptop> 2009-10-12 15:21:47.369855 [NOTICE] sofia.c:3552
Ring-Ready sofia/external/14159927717!
2009-10-12 15:22:47.009474 [NOTICE] switch_ivr_originate.c:2336 Hangup
sofia/external/14159927717 [CS_CONSUME_MEDIA] [NO_ANSWER]
2009-10-12 15:22:47.009474 [INFO] switch_cpp.cpp:1116 PCHangup
gw:debug.comhc:NO_ANSWER du:0 cn:sofia/external/14159927717
2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1087 Session 47
(sofia/external/14159927717) Ended
2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1089 Close Channel
sofia/external/14159927717 [CS_DESTROY]
2009-10-12 15:22:47.009474 [INFO] mod_dptools.c:2133 Originate Failed.
 Cause: NO_ANSWER
2009-10-12 15:22:47.009474 [NOTICE] switch_ivr.c:1367 Transfer
sofia/internal/sip_1 to XML[1999 at default]
2009-10-12 15:22:47.009474 [INFO] mod_dialplan_xml.c:391 Processing
FreeSWITCH->1999 in context default
2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179 Hangup
sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1087 Session 46
(sofia/internal/sip_1) Ended
2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1089 Close Channel
sofia/internal/sip_1 [CS_DESTROY]


thanks for looking at this.

On Mon, Oct 12, 2009 at 10:06 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> which line is hanging up your A (inbound) leg?
>
> look for a blue line that says "Hangup xyz...." that matches it so i can
> see.
>
> I think what is happening is you are getting early media so the bridge is
> actually working then when nobody answers it dies but technically the bridge
> worked.
>
> On Mon, Oct 12, 2009 at 9:41 AM, Matthew Fong <mattdfong at gmail.com> wrote:
>
>> I think think this might be a bug, but wanted to post here instead of Jira
>> in-case I'm overlooking a configuration variable
>> Dialplan
>>
>>     <extension name="1920"><!--init agent for manual and power dial
>> mode-->
>>       <condition field="destination_number" expression="^1920$">
>>         <action application="set" data="hangup_after_bridge=false"/>
>>         <action application="bridge" data="sofia/gateway/
>> debug.com/14159927717"/>
>>         <action application="transfer" data="1999"/><!-- send to unable to
>> reach any contacts-->
>>       </condition>
>>     </extension>
>>
>> API Command
>> originate sofia/internal/sip_1%192.168.1.10 1920
>>
>> When the bridge to 14159927717 fails (NO_ANSWER) both calls are terminated
>> instead of continuing on in the dial plan to exten 1999 (which in my
>> dialplan parks the call). hangup_after_bridge however seems to work OK if
>> someone picks up in the bridge. Is this the correct behavior? How else can I
>> prevent the call from hanging up if a bridge fails? Thanks.
>>
>> I'm using 15135M
>>
>> --matt
>> http://www.hellohunter.com - Predictive Dialer
>> http://www.hellohunter.com/voice_broadcast.php - Voice Broadcasting
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mattdfong at gmail.com  Mon Oct 12 08:25:53 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Mon, 12 Oct 2009 22:25:53 +0700
Subject: [Freeswitch-users] hangup_after_bridge=false not continueing
	after failed bridge
In-Reply-To: <4256bf830910120823w5fcbc3eja4afb76b60b68fda@mail.gmail.com>
References: <4256bf830910120741g5facaa6cr7ed92303dfafe74b@mail.gmail.com>
	<191c3a030910120806o30911131o743aa58420b41ae4@mail.gmail.com>
	<4256bf830910120823w5fcbc3eja4afb76b60b68fda@mail.gmail.com>
Message-ID: <4256bf830910120825u70e988p4d7a041c520048a@mail.gmail.com>

    <extension name="1999"><!--DIRECT POWER-->
      <condition field="destination_number" expression="^1997$">
        <action application="playback"
data="hh/hh-unable_to_connect_contact.wav"/>
        <action application="park"/>
      </condition>
    </extension>

my extn 1999... since it looks from the output like it's transferring, just
don't know why it's disconnecting the call instead of playing the .wav and
parking.

On Mon, Oct 12, 2009 at 10:23 PM, Matthew Fong <mattdfong at gmail.com> wrote:

> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179 Hangup
> sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
>
> might be the line..or the entire output is below....
>
> freeswitch at matthew-laptop> originate sofia/internal/sip_1%192.168.1.10
> 1920
> 2009-10-12 15:21:44.029517 [NOTICE] switch_channel.c:613 New Channel
> sofia/internal/sip_1 [1e722934-7e94-46aa-9d62-e6ec7e7449cf]
> 2009-10-12 15:21:44.121484 [NOTICE] sofia.c:3552 Ring-Ready
> sofia/internal/sip_1!
> 2009-10-12 15:21:47.285531 [NOTICE] sofia.c:3998 Channel
> [sofia/internal/sip_1] has been answered
> 2009-10-12 15:21:47.290996 [INFO] mod_dialplan_xml.c:391 Processing
> FreeSWITCH->1920 in context default
> 2009-10-12 15:21:47.293452 [NOTICE] switch_channel.c:613 New Channel
> sofia/external/14159927717 [6b6cc440-e1d6-415a-b84b-494117e7361d]
> 2009-10-12 15:21:47.293452 [NOTICE] switch_ivr.c:1367 Transfer
> sofia/internal/sip_1 to XML[1920 at default]
> API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
> +OK 1e722934-7e94-46aa-9d62-e6ec7e7449cf
>
> freeswitch at matthew-laptop> 2009-10-12 15:21:47.369855 [NOTICE]
> sofia.c:3552 Ring-Ready sofia/external/14159927717!
> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr_originate.c:2336 Hangup
> sofia/external/14159927717 [CS_CONSUME_MEDIA] [NO_ANSWER]
> 2009-10-12 15:22:47.009474 [INFO] switch_cpp.cpp:1116 PCHangup gw:
> debug.com hc:NO_ANSWER du:0 cn:sofia/external/14159927717
> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1087 Session 47
> (sofia/external/14159927717) Ended
> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1089 Close
> Channel sofia/external/14159927717 [CS_DESTROY]
> 2009-10-12 15:22:47.009474 [INFO] mod_dptools.c:2133 Originate Failed.
>  Cause: NO_ANSWER
> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr.c:1367 Transfer
> sofia/internal/sip_1 to XML[1999 at default]
> 2009-10-12 15:22:47.009474 [INFO] mod_dialplan_xml.c:391 Processing
> FreeSWITCH->1999 in context default
> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179 Hangup
> sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1087 Session 46
> (sofia/internal/sip_1) Ended
> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1089 Close
> Channel sofia/internal/sip_1 [CS_DESTROY]
>
>
> thanks for looking at this.
>
> On Mon, Oct 12, 2009 at 10:06 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> which line is hanging up your A (inbound) leg?
>>
>> look for a blue line that says "Hangup xyz...." that matches it so i can
>> see.
>>
>> I think what is happening is you are getting early media so the bridge is
>> actually working then when nobody answers it dies but technically the bridge
>> worked.
>>
>> On Mon, Oct 12, 2009 at 9:41 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>>
>>> I think think this might be a bug, but wanted to post here instead of
>>> Jira in-case I'm overlooking a configuration variable
>>> Dialplan
>>>
>>>     <extension name="1920"><!--init agent for manual and power dial
>>> mode-->
>>>       <condition field="destination_number" expression="^1920$">
>>>         <action application="set" data="hangup_after_bridge=false"/>
>>>         <action application="bridge" data="sofia/gateway/
>>> debug.com/14159927717"/>
>>>         <action application="transfer" data="1999"/><!-- send to unable
>>> to reach any contacts-->
>>>       </condition>
>>>     </extension>
>>>
>>> API Command
>>> originate sofia/internal/sip_1%192.168.1.10 1920
>>>
>>> When the bridge to 14159927717 fails (NO_ANSWER) both calls are
>>> terminated instead of continuing on in the dial plan to exten 1999 (which in
>>> my dialplan parks the call). hangup_after_bridge however seems to work OK if
>>> someone picks up in the bridge. Is this the correct behavior? How else can I
>>> prevent the call from hanging up if a bridge fails? Thanks.
>>>
>>> I'm using 15135M
>>>
>>> --matt
>>> http://www.hellohunter.com - Predictive Dialer
>>> http://www.hellohunter.com/voice_broadcast.php - Voice Broadcasting
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From anthony.minessale at gmail.com  Mon Oct 12 08:33:01 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 12 Oct 2009 10:33:01 -0500
Subject: [Freeswitch-users] hangup_after_bridge=false not continueing
	after failed bridge
In-Reply-To: <4256bf830910120825u70e988p4d7a041c520048a@mail.gmail.com>
References: <4256bf830910120741g5facaa6cr7ed92303dfafe74b@mail.gmail.com>
	<191c3a030910120806o30911131o743aa58420b41ae4@mail.gmail.com>
	<4256bf830910120823w5fcbc3eja4afb76b60b68fda@mail.gmail.com>
	<4256bf830910120825u70e988p4d7a041c520048a@mail.gmail.com>
Message-ID: <191c3a030910120833i691b6fbfk3eecbf40e5496479@mail.gmail.com>

because the regex is on 1997 not 1999


On Mon, Oct 12, 2009 at 10:25 AM, Matthew Fong <mattdfong at gmail.com> wrote:

>     <extension name="1999"><!--DIRECT POWER-->
>       <condition field="destination_number" expression="^1997$">
>         <action application="playback"
> data="hh/hh-unable_to_connect_contact.wav"/>
>         <action application="park"/>
>       </condition>
>     </extension>
>
> my extn 1999... since it looks from the output like it's transferring, just
> don't know why it's disconnecting the call instead of playing the .wav and
> parking.
>
> On Mon, Oct 12, 2009 at 10:23 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>
>> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179 Hangup
>> sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
>>
>> might be the line..or the entire output is below....
>>
>> freeswitch at matthew-laptop> originate sofia/internal/sip_1%192.168.1.10
>> 1920
>> 2009-10-12 15:21:44.029517 [NOTICE] switch_channel.c:613 New Channel
>> sofia/internal/sip_1 [1e722934-7e94-46aa-9d62-e6ec7e7449cf]
>> 2009-10-12 15:21:44.121484 [NOTICE] sofia.c:3552 Ring-Ready
>> sofia/internal/sip_1!
>> 2009-10-12 15:21:47.285531 [NOTICE] sofia.c:3998 Channel
>> [sofia/internal/sip_1] has been answered
>> 2009-10-12 15:21:47.290996 [INFO] mod_dialplan_xml.c:391 Processing
>> FreeSWITCH->1920 in context default
>> 2009-10-12 15:21:47.293452 [NOTICE] switch_channel.c:613 New Channel
>> sofia/external/14159927717 [6b6cc440-e1d6-415a-b84b-494117e7361d]
>> 2009-10-12 15:21:47.293452 [NOTICE] switch_ivr.c:1367 Transfer
>> sofia/internal/sip_1 to XML[1920 at default]
>> API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
>> +OK 1e722934-7e94-46aa-9d62-e6ec7e7449cf
>>
>> freeswitch at matthew-laptop> 2009-10-12 15:21:47.369855 [NOTICE]
>> sofia.c:3552 Ring-Ready sofia/external/14159927717!
>> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr_originate.c:2336 Hangup
>> sofia/external/14159927717 [CS_CONSUME_MEDIA] [NO_ANSWER]
>> 2009-10-12 15:22:47.009474 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>> debug.com hc:NO_ANSWER du:0 cn:sofia/external/14159927717
>> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1087 Session 47
>> (sofia/external/14159927717) Ended
>> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1089 Close
>> Channel sofia/external/14159927717 [CS_DESTROY]
>> 2009-10-12 15:22:47.009474 [INFO] mod_dptools.c:2133 Originate Failed.
>>  Cause: NO_ANSWER
>> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr.c:1367 Transfer
>> sofia/internal/sip_1 to XML[1999 at default]
>> 2009-10-12 15:22:47.009474 [INFO] mod_dialplan_xml.c:391 Processing
>> FreeSWITCH->1999 in context default
>> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179 Hangup
>> sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
>> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1087 Session 46
>> (sofia/internal/sip_1) Ended
>> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1089 Close
>> Channel sofia/internal/sip_1 [CS_DESTROY]
>>
>>
>> thanks for looking at this.
>>
>> On Mon, Oct 12, 2009 at 10:06 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> which line is hanging up your A (inbound) leg?
>>>
>>> look for a blue line that says "Hangup xyz...." that matches it so i can
>>> see.
>>>
>>> I think what is happening is you are getting early media so the bridge is
>>> actually working then when nobody answers it dies but technically the bridge
>>> worked.
>>>
>>> On Mon, Oct 12, 2009 at 9:41 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>
>>>> I think think this might be a bug, but wanted to post here instead of
>>>> Jira in-case I'm overlooking a configuration variable
>>>> Dialplan
>>>>
>>>>     <extension name="1920"><!--init agent for manual and power dial
>>>> mode-->
>>>>       <condition field="destination_number" expression="^1920$">
>>>>         <action application="set" data="hangup_after_bridge=false"/>
>>>>         <action application="bridge" data="sofia/gateway/
>>>> debug.com/14159927717"/>
>>>>         <action application="transfer" data="1999"/><!-- send to unable
>>>> to reach any contacts-->
>>>>       </condition>
>>>>     </extension>
>>>>
>>>> API Command
>>>> originate sofia/internal/sip_1%192.168.1.10 1920
>>>>
>>>> When the bridge to 14159927717 fails (NO_ANSWER) both calls are
>>>> terminated instead of continuing on in the dial plan to exten 1999 (which in
>>>> my dialplan parks the call). hangup_after_bridge however seems to work OK if
>>>> someone picks up in the bridge. Is this the correct behavior? How else can I
>>>> prevent the call from hanging up if a bridge fails? Thanks.
>>>>
>>>> I'm using 15135M
>>>>
>>>> --matt
>>>> http://www.hellohunter.com - Predictive Dialer
>>>> http://www.hellohunter.com/voice_broadcast.php - Voice Broadcasting
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:213-799-1400
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Mon Oct 12 08:38:11 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 12 Oct 2009 10:38:11 -0500
Subject: [Freeswitch-users] Play music on hold after parking the call
In-Reply-To: <1452e2980910120036k5ab2fb36vbec031c0b0280116@mail.gmail.com>
References: <1452e2980910100237u317d6138w1d13bd3cfb250353@mail.gmail.com>
	<1452e2980910120036k5ab2fb36vbec031c0b0280116@mail.gmail.com>
Message-ID: <191c3a030910120838w2efa7168o19dc070af231b524@mail.gmail.com>

You are asking how you can do asynchronous actions park then play a sound
before park exits without enabling async mode. Think about that.

you could use new valet_parking that parks with music i guess
or tell the channel execute playback instead of park since playback an park
are almost the same thing only one is silent and one is not.


On Mon, Oct 12, 2009 at 2:36 AM, velusamy velu <velu.technical at gmail.com>wrote:

>  Any one please help me to solve the mentioned problem........
>
>
> On Sat, Oct 10, 2009 at 3:07 PM, velusamy velu <velu.technical at gmail.com>wrote:
>
>> Dear All,
>>       I am using ESL.pm module to control the dial plan application. I
>> want to play some music while executing the some external scripts. I
>> executed park after then I executed the playback the music didn't play.
>>
>>     Could any one please explain how can I solve this problem without
>> using async mode in socket application?
>>
>> Thanks,
>> Velusamy.
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mattdfong at gmail.com  Mon Oct 12 08:45:10 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Mon, 12 Oct 2009 22:45:10 +0700
Subject: [Freeswitch-users] hangup_after_bridge=false not continueing
	after failed bridge
In-Reply-To: <191c3a030910120833i691b6fbfk3eecbf40e5496479@mail.gmail.com>
References: <4256bf830910120741g5facaa6cr7ed92303dfafe74b@mail.gmail.com>
	<191c3a030910120806o30911131o743aa58420b41ae4@mail.gmail.com>
	<4256bf830910120823w5fcbc3eja4afb76b60b68fda@mail.gmail.com>
	<4256bf830910120825u70e988p4d7a041c520048a@mail.gmail.com>
	<191c3a030910120833i691b6fbfk3eecbf40e5496479@mail.gmail.com>
Message-ID: <4256bf830910120845g4f991a55m8753c772dcc8cc31@mail.gmail.com>

doh! thanks!

On Mon, Oct 12, 2009 at 10:33 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> because the regex is on 1997 not 1999
>
>
>
> On Mon, Oct 12, 2009 at 10:25 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>
>>     <extension name="1999"><!--DIRECT POWER-->
>>       <condition field="destination_number" expression="^1997$">
>>         <action application="playback"
>> data="hh/hh-unable_to_connect_contact.wav"/>
>>         <action application="park"/>
>>       </condition>
>>     </extension>
>>
>> my extn 1999... since it looks from the output like it's transferring,
>> just don't know why it's disconnecting the call instead of playing the .wav
>> and parking.
>>
>> On Mon, Oct 12, 2009 at 10:23 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>>
>>> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179
>>> Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
>>>
>>> might be the line..or the entire output is below....
>>>
>>> freeswitch at matthew-laptop> originate sofia/internal/sip_1%192.168.1.10
>>> 1920
>>> 2009-10-12 15:21:44.029517 [NOTICE] switch_channel.c:613 New Channel
>>> sofia/internal/sip_1 [1e722934-7e94-46aa-9d62-e6ec7e7449cf]
>>> 2009-10-12 15:21:44.121484 [NOTICE] sofia.c:3552 Ring-Ready
>>> sofia/internal/sip_1!
>>> 2009-10-12 15:21:47.285531 [NOTICE] sofia.c:3998 Channel
>>> [sofia/internal/sip_1] has been answered
>>> 2009-10-12 15:21:47.290996 [INFO] mod_dialplan_xml.c:391 Processing
>>> FreeSWITCH->1920 in context default
>>> 2009-10-12 15:21:47.293452 [NOTICE] switch_channel.c:613 New Channel
>>> sofia/external/14159927717 [6b6cc440-e1d6-415a-b84b-494117e7361d]
>>> 2009-10-12 15:21:47.293452 [NOTICE] switch_ivr.c:1367 Transfer
>>> sofia/internal/sip_1 to XML[1920 at default]
>>> API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
>>> +OK 1e722934-7e94-46aa-9d62-e6ec7e7449cf
>>>
>>> freeswitch at matthew-laptop> 2009-10-12 15:21:47.369855 [NOTICE]
>>> sofia.c:3552 Ring-Ready sofia/external/14159927717!
>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr_originate.c:2336 Hangup
>>> sofia/external/14159927717 [CS_CONSUME_MEDIA] [NO_ANSWER]
>>> 2009-10-12 15:22:47.009474 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>>> debug.com hc:NO_ANSWER du:0 cn:sofia/external/14159927717
>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1087 Session 47
>>> (sofia/external/14159927717) Ended
>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1089 Close
>>> Channel sofia/external/14159927717 [CS_DESTROY]
>>> 2009-10-12 15:22:47.009474 [INFO] mod_dptools.c:2133 Originate Failed.
>>>  Cause: NO_ANSWER
>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr.c:1367 Transfer
>>> sofia/internal/sip_1 to XML[1999 at default]
>>> 2009-10-12 15:22:47.009474 [INFO] mod_dialplan_xml.c:391 Processing
>>> FreeSWITCH->1999 in context default
>>> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179
>>> Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
>>> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1087 Session 46
>>> (sofia/internal/sip_1) Ended
>>> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1089 Close
>>> Channel sofia/internal/sip_1 [CS_DESTROY]
>>>
>>>
>>> thanks for looking at this.
>>>
>>> On Mon, Oct 12, 2009 at 10:06 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> which line is hanging up your A (inbound) leg?
>>>>
>>>> look for a blue line that says "Hangup xyz...." that matches it so i can
>>>> see.
>>>>
>>>> I think what is happening is you are getting early media so the bridge
>>>> is actually working then when nobody answers it dies but technically the
>>>> bridge worked.
>>>>
>>>> On Mon, Oct 12, 2009 at 9:41 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>>
>>>>> I think think this might be a bug, but wanted to post here instead of
>>>>> Jira in-case I'm overlooking a configuration variable
>>>>> Dialplan
>>>>>
>>>>>     <extension name="1920"><!--init agent for manual and power dial
>>>>> mode-->
>>>>>       <condition field="destination_number" expression="^1920$">
>>>>>         <action application="set" data="hangup_after_bridge=false"/>
>>>>>         <action application="bridge" data="sofia/gateway/
>>>>> debug.com/14159927717"/>
>>>>>         <action application="transfer" data="1999"/><!-- send to unable
>>>>> to reach any contacts-->
>>>>>       </condition>
>>>>>     </extension>
>>>>>
>>>>> API Command
>>>>> originate sofia/internal/sip_1%192.168.1.10 1920
>>>>>
>>>>> When the bridge to 14159927717 fails (NO_ANSWER) both calls are
>>>>> terminated instead of continuing on in the dial plan to exten 1999 (which in
>>>>> my dialplan parks the call). hangup_after_bridge however seems to work OK if
>>>>> someone picks up in the bridge. Is this the correct behavior? How else can I
>>>>> prevent the call from hanging up if a bridge fails? Thanks.
>>>>>
>>>>> I'm using 15135M
>>>>>
>>>>> --matt
>>>>> http://www.hellohunter.com - Predictive Dialer
>>>>> http://www.hellohunter.com/voice_broadcast.php - Voice Broadcasting
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:213-799-1400
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Mon Oct 12 09:03:50 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 12 Oct 2009 18:03:50 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
Message-ID: <65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>

hi,

can't make it...

subZero:~/freeswitch-trunk$ make mod_h323

making all mod_h323
Compiling mod_h323.cpp...
quiet_libtool: compile:  g++ -g -ggdb -I/usr/local/include/ptlib
-I/usr/local/include/openh323 -I. -DPTRACING=1 -D_REENTRANT -fno-exceptions
-I/home/tculjaga/freeswitch-trunk/src/include
-I/home/tculjaga/freeswitch-trunk/libs/libteletone/src -fPIC
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2
-D_GNU_SOURCE -DHAVE_CONFIG_H -c mod_h323.cpp  -fPIC -DPIC -o
.libs/mod_h323.o
In file included from /usr/local/include/openh323/h323.h:493,
                 from mod_h323.h:8,
                 from mod_h323.cpp:3:
/usr/local/include/openh323/h323ep.h: In member function ???virtual void
NATFactoryStartup::OnShutdown()???:
/usr/local/include/openh323/h323ep.h:2731: error: ???NatFactory??? has not
been declared
make[4]: *** [mod_h323.lo] Error 1
make[3]: *** [all] Error 1
make[2]: *** [mod_h323-all] Error 1
make[1]: *** [mod_h323] Error 2
make: *** [mod_h323] Error 2



what exact ptlib and h323plus versions did you use? .. can you send us a
link so we can use the exact ones.


T.

2009/10/12 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-12 09:43 -0500, Brian West wrote
> freeswitch-users at lists.freeswit...:
>
> BW>We can host this in our SVN if you wish?
>
> If in fs svn i think yes. But i think may be little time later?
> i don't known is it builds on trunk because i develop it on 1.0.4.
>
> BW>/b
> BW>
> BW>On Oct 12, 2009, at 8:31 AM, Georgiewskiy Yuriy wrote:
> BW>
> BW>> ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/ alfa code, but
> seems
> BW>> it work, but should be buggy,
> BW>> to build need libpt 2.6.5 and h323plus cvs version, i test it now on
> fs
> BW>> 1.0.4.
> BW>
>
> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129-RIPE
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From maciej.aniserowicz at gmail.com  Mon Oct 12 09:04:52 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Mon, 12 Oct 2009 09:04:52 -0700 (PDT)
Subject: [Freeswitch-users] Bad sound quality while eavesdropping
In-Reply-To: <59F3CD44-5FEA-403C-98BE-EEE49EC3815B@freeswitch.org>
References: <41A44DD027064988A914974405788C2E@procent>
	<191c3a030910050731m2d74979ep4598e5a1945d58ae@mail.gmail.com>
	<1254901192035-3780245.post@n2.nabble.com>
	<8437F5BC-7AFF-4A74-B8CD-C5B8219021F6@jerris.com>
	<1255008427639-3788019.post@n2.nabble.com>
	<191c3a030910080823g79c7c596x1cd887e1538ce2e1@mail.gmail.com>
	<1255169044209-3799274.post@n2.nabble.com>
	<CF8C1791-24CD-4132-83B5-7F83B5BDCCC3@jerris.com>
	<1255337256919-3806786.post@n2.nabble.com>
	<59F3CD44-5FEA-403C-98BE-EEE49EC3815B@freeswitch.org>
Message-ID: <1255363492193-3808860.post@n2.nabble.com>


Nope, I wanted to make sure that this is indeed a bug. I opened an issue in
JIRA before regarding some other matter and it turned out to be my mistake,
so I decided to try mailing list first this time.
MA



Brian West wrote:
> 
> Did you open a jira and attach all the info?
> 
> /b
> 
> On Oct 12, 2009, at 3:47 AM, Maciej Aniserowicz wrote:
> 
>> Yes, I confirmed that with Wireshark (filter "rtp and ip.src ==  
>> <device ip>). RTP packets are sent every 20ms.
>>
>> MAniserowicz
>>
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3808860.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From brian at freeswitch.org  Mon Oct 12 09:44:21 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Oct 2009 11:44:21 -0500
Subject: [Freeswitch-users] Re corded file as voicemail.
In-Reply-To: <23f91030910112315h5ea79b81ue55381a3caff0173@mail.gmail.com>
References: <25850181.post@talk.nabble.com>
	<BEDBBF51-2D48-4CDE-A467-11D5BBB458E1@freeswitch.org>
	<23f91030910112315h5ea79b81ue55381a3caff0173@mail.gmail.com>
Message-ID: <BA9A12F7-EB74-485D-8612-12E448621257@freeswitch.org>

Fixed... svn up.

/b

On Oct 12, 2009, at 1:15 AM, Seven Du wrote:

> http://jira.freeswitch.org/browse/MODCODEC-15
>
> Is it ok I assigned to you ?
>
> Thanks.

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From ryannyl at gmail.com  Mon Oct 12 10:18:51 2009
From: ryannyl at gmail.com (Ryanny Lin)
Date: Tue, 13 Oct 2009 01:18:51 +0800
Subject: [Freeswitch-users] Can freeswitch forward an un-implemented SIP
	method?(like a SIP proxy server)
Message-ID: <4bfcac7e0910121018m39e1f820qb39b2eb13348da9c@mail.gmail.com>

Hi, all

I used sipp to test to forward an un-implemented SIP method, but I got 501
Not Implemented.
Can freeswitch be a SIP proxy server?

SIP/2.0 501 Not Implemented
Via: SIP/2.0/UDP 10.10.59.161:5061
From: <sip:1000 at 10.10.59.69:5060>
To: <sip:1001 at 10.10.59.69:5060>;tag=gBpay50SD2SUD
Call-ID: 10-27119 at 10.10.59.161
CSeq: 3 TESTA
User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Content-Length: 0


It looks like freeswitch always processes SIP messages.
-- 
Sincerely regards,
Ryanny
Oct. 13
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From msc at freeswitch.org  Mon Oct 12 10:21:44 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 10:21:44 -0700
Subject: [Freeswitch-users] REMINDER: Weekly FreeSWITCH Conference
	Scheduled for October 9, 11AM CST (GMT -6)
In-Reply-To: <86a32abc0910111440n184b882fwee29736bb15850ef@mail.gmail.com>
References: <87f2f3b90910081202m5dd3f567v240541dd2009c9a8@mail.gmail.com>
	<86a32abc0910111440n184b882fwee29736bb15850ef@mail.gmail.com>
Message-ID: <87f2f3b90910121021q59a99c5cgafa3f718436e83e@mail.gmail.com>

On Sun, Oct 11, 2009 at 2:40 PM, Diego Viola <diego.viola at gmail.com> wrote:

> I'd like to add this for the next weekly conference.
>
> I have added a few events to the event list, as you can see here:
>
> http://wiki.freeswitch.org/wiki/Event_list
>
> But I need more help from the community to complete that and add content to
> the events, etc. So if you can add that for the next weekly conference it
> would be really nice :).
>
> I added this to the Oct 16th agenda. I'll be sending out the reminder email
today.
-MC
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From bottleman at icf.org.ru  Mon Oct 12 10:29:46 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Mon, 12 Oct 2009 21:29:46 +0400 (MSD)
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>

On 2009-10-12 18:03 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

i already write this, ptlib 2.6.5, tou can find link to it on oplalvoip.org, h32plus latest CVS version,
you can find it on www.h323plus.org.

TC>hi,
TC>
TC>can't make it...
TC>
TC>subZero:~/freeswitch-trunk$ make mod_h323
TC>
TC>making all mod_h323
TC>Compiling mod_h323.cpp...
TC>quiet_libtool: compile:  g++ -g -ggdb -I/usr/local/include/ptlib
TC>-I/usr/local/include/openh323 -I. -DPTRACING=1 -D_REENTRANT -fno-exceptions
TC>-I/home/tculjaga/freeswitch-trunk/src/include
TC>-I/home/tculjaga/freeswitch-trunk/libs/libteletone/src -fPIC
TC>-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2
TC>-D_GNU_SOURCE -DHAVE_CONFIG_H -c mod_h323.cpp  -fPIC -DPIC -o
TC>.libs/mod_h323.o
TC>In file included from /usr/local/include/openh323/h323.h:493,
TC>                 from mod_h323.h:8,
TC>                 from mod_h323.cpp:3:
TC>/usr/local/include/openh323/h323ep.h: In member function ??????virtual void
TC>NATFactoryStartup::OnShutdown()??????:
TC>/usr/local/include/openh323/h323ep.h:2731: error: ??????NatFactory?????? has not
TC>been declared
TC>make[4]: *** [mod_h323.lo] Error 1
TC>make[3]: *** [all] Error 1
TC>make[2]: *** [mod_h323-all] Error 1
TC>make[1]: *** [mod_h323] Error 2
TC>make: *** [mod_h323] Error 2
TC>
TC>
TC>
TC>what exact ptlib and h323plus versions did you use? .. can you send us a
TC>link so we can use the exact ones.
TC>
TC>
TC>T.
TC>
TC>2009/10/12 Georgiewskiy Yuriy <bottleman at icf.org.ru>
TC>
TC>> On 2009-10-12 09:43 -0500, Brian West wrote
TC>> freeswitch-users at lists.freeswit...:
TC>>
TC>> BW>We can host this in our SVN if you wish?
TC>>
TC>> If in fs svn i think yes. But i think may be little time later?
TC>> i don't known is it builds on trunk because i develop it on 1.0.4.
TC>>
TC>> BW>/b
TC>> BW>
TC>> BW>On Oct 12, 2009, at 8:31 AM, Georgiewskiy Yuriy wrote:
TC>> BW>
TC>> BW>> ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/ alfa code, but
TC>> seems
TC>> BW>> it work, but should be buggy,
TC>> BW>> to build need libpt 2.6.5 and h323plus cvs version, i test it now on
TC>> fs
TC>> BW>> 1.0.4.
TC>> BW>
TC>>
TC>> C ?????????                       With Best Regards
TC>> ???????????? ????.                Georgiewskiy Yuriy
TC>> +7 4872 711666                    +7 4872 711666
TC>> ???? +7 4872 711143               fax +7 4872 711143
TC>> ???????? ??? "?? ?? ??????"       IT Service Ltd
TC>> http://nkoort.ru                  http://nkoort.ru
TC>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
TC>> YG129-RIPE                        YG129-RIPE
TC>>
TC>> _______________________________________________
TC>> FreeSWITCH-users mailing list
TC>> FreeSWITCH-users at lists.freeswitch.org
TC>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> http://www.freeswitch.org
TC>>
TC>>
TC>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From sicfslist at gmail.com  Mon Oct 12 10:31:24 2009
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Mon, 12 Oct 2009 12:31:24 -0500
Subject: [Freeswitch-users] Can freeswitch forward an un-implemented SIP
	method?(like a SIP proxy server)
In-Reply-To: <4bfcac7e0910121018m39e1f820qb39b2eb13348da9c@mail.gmail.com>
References: <4bfcac7e0910121018m39e1f820qb39b2eb13348da9c@mail.gmail.com>
Message-ID: <35b355e90910121031k2a8aa6ah2ea5bc063e6a59de@mail.gmail.com>

This is because FS is a B2BUA ... not a proxy.  You should consider
OpenSER/SIPS/Kaemillio for this type of application.

SDR
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From msc at freeswitch.org  Mon Oct 12 10:35:08 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 10:35:08 -0700
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <5162D0B7-357C-4D0E-9913-46564A8C0B1D@freeswitch.org>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<87f2f3b90910080958i1f5cfccax621fd12100a05e3e@mail.gmail.com>
	<903da5680910081201p427a8358n768edb61a96674a4@mail.gmail.com>
	<87f2f3b90910081538l6e4e4b7dv3aedb2c4d1af58bc@mail.gmail.com>
	<903da5680910082238x27bff476s8bea0e74ab516a8d@mail.gmail.com>
	<87f2f3b90910082308i7f6144d9kc887596be54dd609@mail.gmail.com>
	<903da5680910082341r485092vb0cd10c43c4a2534@mail.gmail.com>
	<219716B3-C355-4128-A333-1459AC63C364@jerris.com>
	<86a32abc0910111506m4e6d2740tbd3ac5fe0f59178@mail.gmail.com>
	<5162D0B7-357C-4D0E-9913-46564A8C0B1D@freeswitch.org>
Message-ID: <87f2f3b90910121035v5cfce2bel7d32da1671e325b6@mail.gmail.com>

On Sun, Oct 11, 2009 at 4:49 PM, Brian West <brian at freeswitch.org> wrote:

> I have tried to police the wiki when things like this appear.. its one
> thing to crack a joke in fun from time to time... but to put stuff
> like that on the wiki isn't acceptable.
>
> /b
>
>
It seemed appropriate to do so, therefore I added a small snippet on the
documentation guidelines:
http://wiki.freeswitch.org/wiki/Documentation_guidelines#Keep_It_Professional

-MC
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From mattdfong at gmail.com  Mon Oct 12 10:42:15 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Tue, 13 Oct 2009 00:42:15 +0700
Subject: [Freeswitch-users] hangup_after_bridge=false not continueing
	after failed bridge
In-Reply-To: <4256bf830910120845g4f991a55m8753c772dcc8cc31@mail.gmail.com>
References: <4256bf830910120741g5facaa6cr7ed92303dfafe74b@mail.gmail.com>
	<191c3a030910120806o30911131o743aa58420b41ae4@mail.gmail.com>
	<4256bf830910120823w5fcbc3eja4afb76b60b68fda@mail.gmail.com>
	<4256bf830910120825u70e988p4d7a041c520048a@mail.gmail.com>
	<191c3a030910120833i691b6fbfk3eecbf40e5496479@mail.gmail.com>
	<4256bf830910120845g4f991a55m8753c772dcc8cc31@mail.gmail.com>
Message-ID: <4256bf830910121042o50ee79edife2243682bc9d3e2@mail.gmail.com>

when I add a leg_timeout, I get an ALLOTTED_TIMEOUT from my failed bridge...
when an ALLOTTED_TIMEOUT is received, the hangup_after_bridge=false is not
recognized (I think). Is there anyway to get an alloted_timeout to continue
after bridge (failure)?

revised dialplan & cmd output

    <extension name="1920"><!--DEBUG-->
      <condition field="destination_number" expression="^1920$">
        <action application="set" data="hangup_after_bridge=false"/>
        <action application="bridge" data="{leg_timeout=10}sofia/gateway/
debug.com/14159927717"/>
        <action application="transfer" data="1999"/><!-- send to unable to
reach any contacts-->
      </condition>
    </extension>

freeswitch at matthew-laptop> originate sofia/internal/sip_1%192.168.1.10 1920
2009-10-12 17:39:22.237622 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/sip_1 [1d58fb59-c7f9-4908-b612-5bd1c12083cc]
2009-10-12 17:39:22.313524 [NOTICE] sofia.c:3552 Ring-Ready
sofia/internal/sip_1!
2009-10-12 17:39:25.142400 [NOTICE] sofia.c:3998 Channel
[sofia/internal/sip_1] has been answered
2009-10-12 17:39:25.146259 [INFO] mod_dialplan_xml.c:391 Processing
FreeSWITCH->1920 in context default
2009-10-12 17:39:25.146259 [NOTICE] switch_channel.c:613 New Channel
sofia/external/14159927717 [38cb7046-0c0d-47ef-94b5-bf8ccf35d185]
2009-10-12 17:39:25.152026 [NOTICE] switch_ivr.c:1367 Transfer
sofia/internal/sip_1 to XML[1920 at default]
API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
+OK 1d58fb59-c7f9-4908-b612-5bd1c12083cc

freeswitch at matthew-laptop> 2009-10-12 17:39:25.217629 [NOTICE] sofia.c:3552
Ring-Ready sofia/external/14159927717!
2009-10-12 17:39:36.017513 [NOTICE] switch_ivr_originate.c:297 Hangup
sofia/external/14159927717 [CS_CONSUME_MEDIA] [ALLOTTED_TIMEOUT]
2009-10-12 17:39:36.017513 [INFO] switch_cpp.cpp:1116 PCHangup
gw:debug.comhc:ALLOTTED_TIMEOUT du:0 cn:sofia/external/14159927717
2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1087 Session 12
(sofia/external/14159927717) Ended
2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1089 Close Channel
sofia/external/14159927717 [CS_DESTROY]
2009-10-12 17:39:36.037520 [INFO] mod_dptools.c:2133 Originate Failed.
 Cause: ALLOTTED_TIMEOUT
2009-10-12 17:39:36.037520 [NOTICE] mod_dptools.c:2166 Hangup
sofia/internal/sip_1 [CS_EXECUTE] [ALLOTTED_TIMEOUT]
2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1087 Session 11
(sofia/internal/sip_1) Ended
2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1089 Close Channel
sofia/internal/sip_1 [CS_DESTROY]

thanks.

--matt


On Mon, Oct 12, 2009 at 10:45 PM, Matthew Fong <mattdfong at gmail.com> wrote:

> doh! thanks!
>
>
> On Mon, Oct 12, 2009 at 10:33 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> because the regex is on 1997 not 1999
>>
>>
>>
>> On Mon, Oct 12, 2009 at 10:25 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>>
>>>     <extension name="1999"><!--DIRECT POWER-->
>>>       <condition field="destination_number" expression="^1997$">
>>>         <action application="playback"
>>> data="hh/hh-unable_to_connect_contact.wav"/>
>>>         <action application="park"/>
>>>       </condition>
>>>     </extension>
>>>
>>> my extn 1999... since it looks from the output like it's transferring,
>>> just don't know why it's disconnecting the call instead of playing the .wav
>>> and parking.
>>>
>>> On Mon, Oct 12, 2009 at 10:23 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>
>>>> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179
>>>> Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
>>>>
>>>> might be the line..or the entire output is below....
>>>>
>>>> freeswitch at matthew-laptop> originate sofia/internal/sip_1%192.168.1.10
>>>> 1920
>>>> 2009-10-12 15:21:44.029517 [NOTICE] switch_channel.c:613 New Channel
>>>> sofia/internal/sip_1 [1e722934-7e94-46aa-9d62-e6ec7e7449cf]
>>>> 2009-10-12 15:21:44.121484 [NOTICE] sofia.c:3552 Ring-Ready
>>>> sofia/internal/sip_1!
>>>> 2009-10-12 15:21:47.285531 [NOTICE] sofia.c:3998 Channel
>>>> [sofia/internal/sip_1] has been answered
>>>> 2009-10-12 15:21:47.290996 [INFO] mod_dialplan_xml.c:391 Processing
>>>> FreeSWITCH->1920 in context default
>>>> 2009-10-12 15:21:47.293452 [NOTICE] switch_channel.c:613 New Channel
>>>> sofia/external/14159927717 [6b6cc440-e1d6-415a-b84b-494117e7361d]
>>>> 2009-10-12 15:21:47.293452 [NOTICE] switch_ivr.c:1367 Transfer
>>>> sofia/internal/sip_1 to XML[1920 at default]
>>>> API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
>>>> +OK 1e722934-7e94-46aa-9d62-e6ec7e7449cf
>>>>
>>>> freeswitch at matthew-laptop> 2009-10-12 15:21:47.369855 [NOTICE]
>>>> sofia.c:3552 Ring-Ready sofia/external/14159927717!
>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr_originate.c:2336 Hangup
>>>> sofia/external/14159927717 [CS_CONSUME_MEDIA] [NO_ANSWER]
>>>> 2009-10-12 15:22:47.009474 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>>>> debug.com hc:NO_ANSWER du:0 cn:sofia/external/14159927717
>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1087 Session
>>>> 47 (sofia/external/14159927717) Ended
>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1089 Close
>>>> Channel sofia/external/14159927717 [CS_DESTROY]
>>>> 2009-10-12 15:22:47.009474 [INFO] mod_dptools.c:2133 Originate Failed.
>>>>  Cause: NO_ANSWER
>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr.c:1367 Transfer
>>>> sofia/internal/sip_1 to XML[1999 at default]
>>>> 2009-10-12 15:22:47.009474 [INFO] mod_dialplan_xml.c:391 Processing
>>>> FreeSWITCH->1999 in context default
>>>> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179
>>>> Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
>>>> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1087 Session
>>>> 46 (sofia/internal/sip_1) Ended
>>>> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1089 Close
>>>> Channel sofia/internal/sip_1 [CS_DESTROY]
>>>>
>>>>
>>>> thanks for looking at this.
>>>>
>>>> On Mon, Oct 12, 2009 at 10:06 PM, Anthony Minessale <
>>>> anthony.minessale at gmail.com> wrote:
>>>>
>>>>> which line is hanging up your A (inbound) leg?
>>>>>
>>>>> look for a blue line that says "Hangup xyz...." that matches it so i
>>>>> can see.
>>>>>
>>>>> I think what is happening is you are getting early media so the bridge
>>>>> is actually working then when nobody answers it dies but technically the
>>>>> bridge worked.
>>>>>
>>>>> On Mon, Oct 12, 2009 at 9:41 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>>>
>>>>>> I think think this might be a bug, but wanted to post here instead of
>>>>>> Jira in-case I'm overlooking a configuration variable
>>>>>> Dialplan
>>>>>>
>>>>>>     <extension name="1920"><!--init agent for manual and power dial
>>>>>> mode-->
>>>>>>       <condition field="destination_number" expression="^1920$">
>>>>>>         <action application="set" data="hangup_after_bridge=false"/>
>>>>>>         <action application="bridge" data="sofia/gateway/
>>>>>> debug.com/14159927717"/>
>>>>>>         <action application="transfer" data="1999"/><!-- send to
>>>>>> unable to reach any contacts-->
>>>>>>       </condition>
>>>>>>     </extension>
>>>>>>
>>>>>> API Command
>>>>>> originate sofia/internal/sip_1%192.168.1.10 1920
>>>>>>
>>>>>> When the bridge to 14159927717 fails (NO_ANSWER) both calls are
>>>>>> terminated instead of continuing on in the dial plan to exten 1999 (which in
>>>>>> my dialplan parks the call). hangup_after_bridge however seems to work OK if
>>>>>> someone picks up in the bridge. Is this the correct behavior? How else can I
>>>>>> prevent the call from hanging up if a bridge fails? Thanks.
>>>>>>
>>>>>> I'm using 15135M
>>>>>>
>>>>>> --matt
>>>>>> http://www.hellohunter.com - Predictive Dialer
>>>>>> http://www.hellohunter.com/voice_broadcast.php - Voice Broadcasting
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>> iax:guest at conference.freeswitch.org/888
>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>> pstn:213-799-1400
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From msc at freeswitch.org  Mon Oct 12 11:00:18 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 11:00:18 -0700
Subject: [Freeswitch-users] Question about fax tone detection
In-Reply-To: <1255345270255-3807298.post@n2.nabble.com>
References: <1255345270255-3807298.post@n2.nabble.com>
Message-ID: <87f2f3b90910121100l40a1b22hff33df032411cf31@mail.gmail.com>

On Mon, Oct 12, 2009 at 4:01 AM, homqua <ngay01042005 at gmail.com> wrote:

>
> Hi,
> I have implemented the solution for tone detection in wiki, and also answer
> the channel before detecting the tone:
> <condition field="destination_number" expression="^(055138419992)$">
>    <action application="answer"/>
>    <action application="tone_detect" data="fax 1100 r +15000 transfer fax
> XML default"/>
>
>
> <extension name="fax">
>      <condition field="destination_number" expression="^fax$">
>        <action application="answer" />
>        <action application="sleep" data="1000"/>
>        <action application="rxfax"
>
> data="/usr/local/freeswitch/storage/fax/${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}.tiff"/>
>    <action application="set" data="fax_mode=recv"/>
>        <action application="hangup"/>
>      </condition>
>    </extension>
>
>
> But FS cannot recognize the tone, and therefore cannot move to fax
> extension.  Below are the error in FS:
>
> 2009-10-12 10:57:16.702287 [NOTICE] switch_channel.c:602 New Channel
> sofia/external/anonymous at anonymous.invalid
> [c431f0a3-9231-4724-ba39-9e4ef7edfca2]
> 2009-10-12 10:57:16.703413 [INFO] mod_dialplan_xml.c:315 Processing
> Anonymous->055138419992 in context public
> 2009-10-12 10:57:16.719288 [NOTICE] switch_ivr.c:1349 Transfer
> sofia/external/anonymous at anonymous.invalid to XML[055138419992 at default]
> 2009-10-12 10:57:16.719288 [INFO] mod_dialplan_xml.c:315 Processing
> Anonymous->055138419992 in context default
> 2009-10-12 10:57:16.722289 [NOTICE] mod_dptools.c:649 Channel
> [sofia/external/anonymous at anonymous.invalid] has been answered
> 2009-10-12 10:57:16.722289 [NOTICE] mod_dptools.c:1324 Enabling tone
> detection 'fax' '1100'
> 2009-10-12 10:57:16.723302 [NOTICE] switch_core_state_machine.c:179 Hangup
> sofia/external/anonymous at anonymous.invalid [CS_EXECUTE] [NORMAL_CLEARING]
> 2009-10-12 10:57:16.740285 [NOTICE] switch_core_session.c:1086 Session 1
> (sofia/external/anonymous at anonymous.invalid) Ended
> 2009-10-12 10:57:16.740285 [NOTICE] switch_core_session.c:1088 Close
> Channel
> sofia/external/anonymous at anonymous.invalid [CS_DESTROY]
>
> And the trace for SIP messages:  http://pastebin.com/m4e47e7d9
>
> If anyone has any idea, tell me please.
> Thanks.
>

I think the trouble here is that you don't have anything else in the
dialplan after the tone_detect. The tone_detect app is non-block, which
means that it doesn't sit there and wait for a tone. If you want the
dialplan to sit and wait then do a sleep app after your tone_detect. The
other question I would have is this: what happens if the incoming call is
not a fax? What do you want to do then?

-MC
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From msc at freeswitch.org  Mon Oct 12 11:11:24 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 11:11:24 -0700
Subject: [Freeswitch-users] hangup_after_bridge=false not continueing
	after failed bridge
In-Reply-To: <4256bf830910121042o50ee79edife2243682bc9d3e2@mail.gmail.com>
References: <4256bf830910120741g5facaa6cr7ed92303dfafe74b@mail.gmail.com>
	<191c3a030910120806o30911131o743aa58420b41ae4@mail.gmail.com>
	<4256bf830910120823w5fcbc3eja4afb76b60b68fda@mail.gmail.com>
	<4256bf830910120825u70e988p4d7a041c520048a@mail.gmail.com>
	<191c3a030910120833i691b6fbfk3eecbf40e5496479@mail.gmail.com>
	<4256bf830910120845g4f991a55m8753c772dcc8cc31@mail.gmail.com>
	<4256bf830910121042o50ee79edife2243682bc9d3e2@mail.gmail.com>
Message-ID: <87f2f3b90910121111m54cf2586pfca0d867307b1e86@mail.gmail.com>

On Mon, Oct 12, 2009 at 10:42 AM, Matthew Fong <mattdfong at gmail.com> wrote:

> when I add a leg_timeout, I get an ALLOTTED_TIMEOUT from my failed
> bridge...
> when an ALLOTTED_TIMEOUT is received, the hangup_after_bridge=false is not
> recognized (I think). Is there anyway to get an alloted_timeout to continue
> after bridge (failure)?
>

Try it with ignore_early_media=true and see if it's the early media that's
tripping you up.
-MC


>
> revised dialplan & cmd output
>
>     <extension name="1920"><!--DEBUG-->
>       <condition field="destination_number" expression="^1920$">
>         <action application="set" data="hangup_after_bridge=false"/>
>         <action application="bridge" data="{leg_timeout=10}sofia/gateway/
> debug.com/14159927717"/>
>         <action application="transfer" data="1999"/><!-- send to unable to
> reach any contacts-->
>       </condition>
>     </extension>
>
> freeswitch at matthew-laptop> originate sofia/internal/sip_1%192.168.1.10
> 1920
> 2009-10-12 17:39:22.237622 [NOTICE] switch_channel.c:613 New Channel
> sofia/internal/sip_1 [1d58fb59-c7f9-4908-b612-5bd1c12083cc]
> 2009-10-12 17:39:22.313524 [NOTICE] sofia.c:3552 Ring-Ready
> sofia/internal/sip_1!
> 2009-10-12 17:39:25.142400 [NOTICE] sofia.c:3998 Channel
> [sofia/internal/sip_1] has been answered
> 2009-10-12 17:39:25.146259 [INFO] mod_dialplan_xml.c:391 Processing
> FreeSWITCH->1920 in context default
> 2009-10-12 17:39:25.146259 [NOTICE] switch_channel.c:613 New Channel
> sofia/external/14159927717 [38cb7046-0c0d-47ef-94b5-bf8ccf35d185]
> 2009-10-12 17:39:25.152026 [NOTICE] switch_ivr.c:1367 Transfer
> sofia/internal/sip_1 to XML[1920 at default]
> API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
> +OK 1d58fb59-c7f9-4908-b612-5bd1c12083cc
>
> freeswitch at matthew-laptop> 2009-10-12 17:39:25.217629 [NOTICE]
> sofia.c:3552 Ring-Ready sofia/external/14159927717!
> 2009-10-12 17:39:36.017513 [NOTICE] switch_ivr_originate.c:297 Hangup
> sofia/external/14159927717 [CS_CONSUME_MEDIA] [ALLOTTED_TIMEOUT]
> 2009-10-12 17:39:36.017513 [INFO] switch_cpp.cpp:1116 PCHangup gw:
> debug.com hc:ALLOTTED_TIMEOUT du:0 cn:sofia/external/14159927717
> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1087 Session 12
> (sofia/external/14159927717) Ended
> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1089 Close
> Channel sofia/external/14159927717 [CS_DESTROY]
> 2009-10-12 17:39:36.037520 [INFO] mod_dptools.c:2133 Originate Failed.
>  Cause: ALLOTTED_TIMEOUT
> 2009-10-12 17:39:36.037520 [NOTICE] mod_dptools.c:2166 Hangup
> sofia/internal/sip_1 [CS_EXECUTE] [ALLOTTED_TIMEOUT]
> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1087 Session 11
> (sofia/internal/sip_1) Ended
> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1089 Close
> Channel sofia/internal/sip_1 [CS_DESTROY]
>
> thanks.
>
> --matt
>
>
> On Mon, Oct 12, 2009 at 10:45 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>
>> doh! thanks!
>>
>>
>> On Mon, Oct 12, 2009 at 10:33 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> because the regex is on 1997 not 1999
>>>
>>>
>>>
>>> On Mon, Oct 12, 2009 at 10:25 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>
>>>>     <extension name="1999"><!--DIRECT POWER-->
>>>>       <condition field="destination_number" expression="^1997$">
>>>>         <action application="playback"
>>>> data="hh/hh-unable_to_connect_contact.wav"/>
>>>>         <action application="park"/>
>>>>       </condition>
>>>>     </extension>
>>>>
>>>> my extn 1999... since it looks from the output like it's transferring,
>>>> just don't know why it's disconnecting the call instead of playing the .wav
>>>> and parking.
>>>>
>>>> On Mon, Oct 12, 2009 at 10:23 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>>
>>>>> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179
>>>>> Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
>>>>>
>>>>> might be the line..or the entire output is below....
>>>>>
>>>>> freeswitch at matthew-laptop> originate sofia/internal/sip_1%192.168.1.10
>>>>> 1920
>>>>> 2009-10-12 15:21:44.029517 [NOTICE] switch_channel.c:613 New Channel
>>>>> sofia/internal/sip_1 [1e722934-7e94-46aa-9d62-e6ec7e7449cf]
>>>>> 2009-10-12 15:21:44.121484 [NOTICE] sofia.c:3552 Ring-Ready
>>>>> sofia/internal/sip_1!
>>>>> 2009-10-12 15:21:47.285531 [NOTICE] sofia.c:3998 Channel
>>>>> [sofia/internal/sip_1] has been answered
>>>>> 2009-10-12 15:21:47.290996 [INFO] mod_dialplan_xml.c:391 Processing
>>>>> FreeSWITCH->1920 in context default
>>>>> 2009-10-12 15:21:47.293452 [NOTICE] switch_channel.c:613 New Channel
>>>>> sofia/external/14159927717 [6b6cc440-e1d6-415a-b84b-494117e7361d]
>>>>> 2009-10-12 15:21:47.293452 [NOTICE] switch_ivr.c:1367 Transfer
>>>>> sofia/internal/sip_1 to XML[1920 at default]
>>>>> API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
>>>>> +OK 1e722934-7e94-46aa-9d62-e6ec7e7449cf
>>>>>
>>>>> freeswitch at matthew-laptop> 2009-10-12 15:21:47.369855 [NOTICE]
>>>>> sofia.c:3552 Ring-Ready sofia/external/14159927717!
>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr_originate.c:2336 Hangup
>>>>> sofia/external/14159927717 [CS_CONSUME_MEDIA] [NO_ANSWER]
>>>>> 2009-10-12 15:22:47.009474 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>>>>> debug.com hc:NO_ANSWER du:0 cn:sofia/external/14159927717
>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1087 Session
>>>>> 47 (sofia/external/14159927717) Ended
>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1089 Close
>>>>> Channel sofia/external/14159927717 [CS_DESTROY]
>>>>> 2009-10-12 15:22:47.009474 [INFO] mod_dptools.c:2133 Originate Failed.
>>>>>  Cause: NO_ANSWER
>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr.c:1367 Transfer
>>>>> sofia/internal/sip_1 to XML[1999 at default]
>>>>> 2009-10-12 15:22:47.009474 [INFO] mod_dialplan_xml.c:391 Processing
>>>>> FreeSWITCH->1999 in context default
>>>>> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179
>>>>> Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
>>>>> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1087 Session
>>>>> 46 (sofia/internal/sip_1) Ended
>>>>> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1089 Close
>>>>> Channel sofia/internal/sip_1 [CS_DESTROY]
>>>>>
>>>>>
>>>>> thanks for looking at this.
>>>>>
>>>>> On Mon, Oct 12, 2009 at 10:06 PM, Anthony Minessale <
>>>>> anthony.minessale at gmail.com> wrote:
>>>>>
>>>>>> which line is hanging up your A (inbound) leg?
>>>>>>
>>>>>> look for a blue line that says "Hangup xyz...." that matches it so i
>>>>>> can see.
>>>>>>
>>>>>> I think what is happening is you are getting early media so the bridge
>>>>>> is actually working then when nobody answers it dies but technically the
>>>>>> bridge worked.
>>>>>>
>>>>>> On Mon, Oct 12, 2009 at 9:41 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>>>>
>>>>>>> I think think this might be a bug, but wanted to post here instead of
>>>>>>> Jira in-case I'm overlooking a configuration variable
>>>>>>> Dialplan
>>>>>>>
>>>>>>>     <extension name="1920"><!--init agent for manual and power dial
>>>>>>> mode-->
>>>>>>>       <condition field="destination_number" expression="^1920$">
>>>>>>>         <action application="set" data="hangup_after_bridge=false"/>
>>>>>>>         <action application="bridge" data="sofia/gateway/
>>>>>>> debug.com/14159927717"/>
>>>>>>>         <action application="transfer" data="1999"/><!-- send to
>>>>>>> unable to reach any contacts-->
>>>>>>>       </condition>
>>>>>>>     </extension>
>>>>>>>
>>>>>>> API Command
>>>>>>> originate sofia/internal/sip_1%192.168.1.10 1920
>>>>>>>
>>>>>>> When the bridge to 14159927717 fails (NO_ANSWER) both calls are
>>>>>>> terminated instead of continuing on in the dial plan to exten 1999 (which in
>>>>>>> my dialplan parks the call). hangup_after_bridge however seems to work OK if
>>>>>>> someone picks up in the bridge. Is this the correct behavior? How else can I
>>>>>>> prevent the call from hanging up if a bridge fails? Thanks.
>>>>>>>
>>>>>>> I'm using 15135M
>>>>>>>
>>>>>>> --matt
>>>>>>> http://www.hellohunter.com - Predictive Dialer
>>>>>>> http://www.hellohunter.com/voice_broadcast.php - Voice Broadcasting
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Anthony Minessale II
>>>>>>
>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>> ClueCon http://www.cluecon.com/
>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>
>>>>>> AIM: anthm
>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>
>>>>>> FreeSWITCH Developer Conference
>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>> pstn:213-799-1400
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:213-799-1400
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mattdfong at gmail.com  Mon Oct 12 11:26:32 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Tue, 13 Oct 2009 01:26:32 +0700
Subject: [Freeswitch-users] hangup_after_bridge=false not continueing
	after failed bridge
In-Reply-To: <87f2f3b90910121111m54cf2586pfca0d867307b1e86@mail.gmail.com>
References: <4256bf830910120741g5facaa6cr7ed92303dfafe74b@mail.gmail.com>
	<191c3a030910120806o30911131o743aa58420b41ae4@mail.gmail.com>
	<4256bf830910120823w5fcbc3eja4afb76b60b68fda@mail.gmail.com>
	<4256bf830910120825u70e988p4d7a041c520048a@mail.gmail.com>
	<191c3a030910120833i691b6fbfk3eecbf40e5496479@mail.gmail.com>
	<4256bf830910120845g4f991a55m8753c772dcc8cc31@mail.gmail.com>
	<4256bf830910121042o50ee79edife2243682bc9d3e2@mail.gmail.com>
	<87f2f3b90910121111m54cf2586pfca0d867307b1e86@mail.gmail.com>
Message-ID: <4256bf830910121126l6f4326fco9ee3a83f51820742@mail.gmail.com>

still no luck...

    <extension name="1920"><!--DEBUG-->
      <condition field="destination_number" expression="^1920$">
        <action application="set" data="hangup_after_bridge=false"/>
        <action application="bridge"
data="{leg_timeout=10,ignore_early_media=true}sofia/gateway/
debug.com/14159927717"/>
        <action application="transfer" data="1999"/><!-- send to unable to
reach any contacts-->
      </condition>
    </extension>


freeswitch at matthew-laptop> originate sofia/internal/sip_1%192.168.1.10 1920
2009-10-12 18:25:44.345480 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/sip_1 [3fc6efb2-e4fa-454a-abb7-ebe39da748f5]
2009-10-12 18:25:44.489480 [NOTICE] sofia.c:3552 Ring-Ready
sofia/internal/sip_1!
2009-10-12 18:25:46.601509 [NOTICE] sofia.c:3998 Channel
[sofia/internal/sip_1] has been answered
2009-10-12 18:25:46.601509 [INFO] mod_dialplan_xml.c:391 Processing
FreeSWITCH->1920 in context default
2009-10-12 18:25:46.601509 [NOTICE] switch_channel.c:613 New Channel
sofia/external/14159927717 [1976e3c2-187c-4f05-98f5-36742ab8248f]
2009-10-12 18:25:46.601509 [NOTICE] switch_ivr.c:1367 Transfer
sofia/internal/sip_1 to XML[1920 at default]
API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
+OK 3fc6efb2-e4fa-454a-abb7-ebe39da748f5

freeswitch at matthew-laptop> 2009-10-12 18:25:46.677650 [NOTICE] sofia.c:3552
Ring-Ready sofia/external/14159927717!
2009-10-12 18:25:57.017477 [NOTICE] switch_ivr_originate.c:297 Hangup
sofia/external/14159927717 [CS_CONSUME_MEDIA] [ALLOTTED_TIMEOUT]
2009-10-12 18:25:57.017477 [INFO] switch_cpp.cpp:1116 PCHangup
gw:debug.comhc:ALLOTTED_TIMEOUT du:0 cn:sofia/external/14159927717
2009-10-12 18:25:57.037695 [NOTICE] switch_core_session.c:1087 Session 4
(sofia/external/14159927717) Ended
2009-10-12 18:25:57.037695 [NOTICE] switch_core_session.c:1089 Close Channel
sofia/external/14159927717 [CS_DESTROY]
2009-10-12 18:25:57.037695 [INFO] mod_dptools.c:2133 Originate Failed.
 Cause: ALLOTTED_TIMEOUT
2009-10-12 18:25:57.037695 [NOTICE] mod_dptools.c:2166 Hangup
sofia/internal/sip_1 [CS_EXECUTE] [ALLOTTED_TIMEOUT]
2009-10-12 18:25:57.037695 [NOTICE] switch_core_session.c:1087 Session 3
(sofia/internal/sip_1) Ended
2009-10-12 18:25:57.037695 [NOTICE] switch_core_session.c:1089 Close Channel
sofia/internal/sip_1 [CS_DESTROY]

--matt

On Tue, Oct 13, 2009 at 1:11 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Oct 12, 2009 at 10:42 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>
>> when I add a leg_timeout, I get an ALLOTTED_TIMEOUT from my failed
>> bridge...
>> when an ALLOTTED_TIMEOUT is received, the hangup_after_bridge=false is not
>> recognized (I think). Is there anyway to get an alloted_timeout to continue
>> after bridge (failure)?
>>
>
> Try it with ignore_early_media=true and see if it's the early media that's
> tripping you up.
> -MC
>
>
>>
>> revised dialplan & cmd output
>>
>>     <extension name="1920"><!--DEBUG-->
>>       <condition field="destination_number" expression="^1920$">
>>         <action application="set" data="hangup_after_bridge=false"/>
>>         <action application="bridge" data="{leg_timeout=10}sofia/gateway/
>> debug.com/14159927717"/>
>>          <action application="transfer" data="1999"/><!-- send to unable
>> to reach any contacts-->
>>       </condition>
>>     </extension>
>>
>> freeswitch at matthew-laptop> originate sofia/internal/sip_1%192.168.1.10
>> 1920
>> 2009-10-12 17:39:22.237622 [NOTICE] switch_channel.c:613 New Channel
>> sofia/internal/sip_1 [1d58fb59-c7f9-4908-b612-5bd1c12083cc]
>> 2009-10-12 17:39:22.313524 [NOTICE] sofia.c:3552 Ring-Ready
>> sofia/internal/sip_1!
>> 2009-10-12 17:39:25.142400 [NOTICE] sofia.c:3998 Channel
>> [sofia/internal/sip_1] has been answered
>> 2009-10-12 17:39:25.146259 [INFO] mod_dialplan_xml.c:391 Processing
>> FreeSWITCH->1920 in context default
>> 2009-10-12 17:39:25.146259 [NOTICE] switch_channel.c:613 New Channel
>> sofia/external/14159927717 [38cb7046-0c0d-47ef-94b5-bf8ccf35d185]
>> 2009-10-12 17:39:25.152026 [NOTICE] switch_ivr.c:1367 Transfer
>> sofia/internal/sip_1 to XML[1920 at default]
>>  API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
>> +OK 1d58fb59-c7f9-4908-b612-5bd1c12083cc
>>
>> freeswitch at matthew-laptop> 2009-10-12 17:39:25.217629 [NOTICE]
>> sofia.c:3552 Ring-Ready sofia/external/14159927717!
>> 2009-10-12 17:39:36.017513 [NOTICE] switch_ivr_originate.c:297 Hangup
>> sofia/external/14159927717 [CS_CONSUME_MEDIA] [ALLOTTED_TIMEOUT]
>> 2009-10-12 17:39:36.017513 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>> debug.com hc:ALLOTTED_TIMEOUT du:0 cn:sofia/external/14159927717
>> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1087 Session 12
>> (sofia/external/14159927717) Ended
>> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1089 Close
>> Channel sofia/external/14159927717 [CS_DESTROY]
>> 2009-10-12 17:39:36.037520 [INFO] mod_dptools.c:2133 Originate Failed.
>>  Cause: ALLOTTED_TIMEOUT
>> 2009-10-12 17:39:36.037520 [NOTICE] mod_dptools.c:2166 Hangup
>> sofia/internal/sip_1 [CS_EXECUTE] [ALLOTTED_TIMEOUT]
>> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1087 Session 11
>> (sofia/internal/sip_1) Ended
>> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1089 Close
>> Channel sofia/internal/sip_1 [CS_DESTROY]
>>
>> thanks.
>>
>> --matt
>>
>>
>> On Mon, Oct 12, 2009 at 10:45 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>>
>>> doh! thanks!
>>>
>>>
>>> On Mon, Oct 12, 2009 at 10:33 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> because the regex is on 1997 not 1999
>>>>
>>>>
>>>>
>>>> On Mon, Oct 12, 2009 at 10:25 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>>
>>>>>     <extension name="1999"><!--DIRECT POWER-->
>>>>>       <condition field="destination_number" expression="^1997$">
>>>>>         <action application="playback"
>>>>> data="hh/hh-unable_to_connect_contact.wav"/>
>>>>>         <action application="park"/>
>>>>>       </condition>
>>>>>     </extension>
>>>>>
>>>>> my extn 1999... since it looks from the output like it's transferring,
>>>>> just don't know why it's disconnecting the call instead of playing the .wav
>>>>> and parking.
>>>>>
>>>>> On Mon, Oct 12, 2009 at 10:23 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>>>
>>>>>> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179
>>>>>> Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
>>>>>>
>>>>>> might be the line..or the entire output is below....
>>>>>>
>>>>>> freeswitch at matthew-laptop> originate
>>>>>> sofia/internal/sip_1%192.168.1.10 1920
>>>>>> 2009-10-12 15:21:44.029517 [NOTICE] switch_channel.c:613 New Channel
>>>>>> sofia/internal/sip_1 [1e722934-7e94-46aa-9d62-e6ec7e7449cf]
>>>>>> 2009-10-12 15:21:44.121484 [NOTICE] sofia.c:3552 Ring-Ready
>>>>>> sofia/internal/sip_1!
>>>>>> 2009-10-12 15:21:47.285531 [NOTICE] sofia.c:3998 Channel
>>>>>> [sofia/internal/sip_1] has been answered
>>>>>> 2009-10-12 15:21:47.290996 [INFO] mod_dialplan_xml.c:391 Processing
>>>>>> FreeSWITCH->1920 in context default
>>>>>> 2009-10-12 15:21:47.293452 [NOTICE] switch_channel.c:613 New Channel
>>>>>> sofia/external/14159927717 [6b6cc440-e1d6-415a-b84b-494117e7361d]
>>>>>> 2009-10-12 15:21:47.293452 [NOTICE] switch_ivr.c:1367 Transfer
>>>>>> sofia/internal/sip_1 to XML[1920 at default]
>>>>>> API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
>>>>>> +OK 1e722934-7e94-46aa-9d62-e6ec7e7449cf
>>>>>>
>>>>>> freeswitch at matthew-laptop> 2009-10-12 15:21:47.369855 [NOTICE]
>>>>>> sofia.c:3552 Ring-Ready sofia/external/14159927717!
>>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr_originate.c:2336 Hangup
>>>>>> sofia/external/14159927717 [CS_CONSUME_MEDIA] [NO_ANSWER]
>>>>>> 2009-10-12 15:22:47.009474 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>>>>>> debug.com hc:NO_ANSWER du:0 cn:sofia/external/14159927717
>>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1087 Session
>>>>>> 47 (sofia/external/14159927717) Ended
>>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1089 Close
>>>>>> Channel sofia/external/14159927717 [CS_DESTROY]
>>>>>> 2009-10-12 15:22:47.009474 [INFO] mod_dptools.c:2133 Originate Failed.
>>>>>>  Cause: NO_ANSWER
>>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr.c:1367 Transfer
>>>>>> sofia/internal/sip_1 to XML[1999 at default]
>>>>>> 2009-10-12 15:22:47.009474 [INFO] mod_dialplan_xml.c:391 Processing
>>>>>> FreeSWITCH->1999 in context default
>>>>>> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179
>>>>>> Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
>>>>>> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1087 Session
>>>>>> 46 (sofia/internal/sip_1) Ended
>>>>>> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1089 Close
>>>>>> Channel sofia/internal/sip_1 [CS_DESTROY]
>>>>>>
>>>>>>
>>>>>> thanks for looking at this.
>>>>>>
>>>>>> On Mon, Oct 12, 2009 at 10:06 PM, Anthony Minessale <
>>>>>> anthony.minessale at gmail.com> wrote:
>>>>>>
>>>>>>> which line is hanging up your A (inbound) leg?
>>>>>>>
>>>>>>> look for a blue line that says "Hangup xyz...." that matches it so i
>>>>>>> can see.
>>>>>>>
>>>>>>> I think what is happening is you are getting early media so the
>>>>>>> bridge is actually working then when nobody answers it dies but technically
>>>>>>> the bridge worked.
>>>>>>>
>>>>>>> On Mon, Oct 12, 2009 at 9:41 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>>>>>
>>>>>>>> I think think this might be a bug, but wanted to post here instead
>>>>>>>> of Jira in-case I'm overlooking a configuration variable
>>>>>>>> Dialplan
>>>>>>>>
>>>>>>>>     <extension name="1920"><!--init agent for manual and power dial
>>>>>>>> mode-->
>>>>>>>>       <condition field="destination_number" expression="^1920$">
>>>>>>>>         <action application="set" data="hangup_after_bridge=false"/>
>>>>>>>>         <action application="bridge" data="sofia/gateway/
>>>>>>>> debug.com/14159927717"/>
>>>>>>>>         <action application="transfer" data="1999"/><!-- send to
>>>>>>>> unable to reach any contacts-->
>>>>>>>>       </condition>
>>>>>>>>     </extension>
>>>>>>>>
>>>>>>>> API Command
>>>>>>>> originate sofia/internal/sip_1%192.168.1.10 1920
>>>>>>>>
>>>>>>>> When the bridge to 14159927717 fails (NO_ANSWER) both calls are
>>>>>>>> terminated instead of continuing on in the dial plan to exten 1999 (which in
>>>>>>>> my dialplan parks the call). hangup_after_bridge however seems to work OK if
>>>>>>>> someone picks up in the bridge. Is this the correct behavior? How else can I
>>>>>>>> prevent the call from hanging up if a bridge fails? Thanks.
>>>>>>>>
>>>>>>>> I'm using 15135M
>>>>>>>>
>>>>>>>> --matt
>>>>>>>> http://www.hellohunter.com - Predictive Dialer
>>>>>>>> http://www.hellohunter.com/voice_broadcast.php - Voice Broadcasting
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Anthony Minessale II
>>>>>>>
>>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>>> ClueCon http://www.cluecon.com/
>>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>>
>>>>>>> AIM: anthm
>>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>>
>>>>>>> FreeSWITCH Developer Conference
>>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>>> pstn:213-799-1400
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:213-799-1400
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Oct 12 11:34:27 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 11:34:27 -0700
Subject: [Freeswitch-users] hangup_after_bridge=false not continueing
	after failed bridge
In-Reply-To: <87f2f3b90910121111m54cf2586pfca0d867307b1e86@mail.gmail.com>
References: <4256bf830910120741g5facaa6cr7ed92303dfafe74b@mail.gmail.com>
	<191c3a030910120806o30911131o743aa58420b41ae4@mail.gmail.com>
	<4256bf830910120823w5fcbc3eja4afb76b60b68fda@mail.gmail.com>
	<4256bf830910120825u70e988p4d7a041c520048a@mail.gmail.com>
	<191c3a030910120833i691b6fbfk3eecbf40e5496479@mail.gmail.com>
	<4256bf830910120845g4f991a55m8753c772dcc8cc31@mail.gmail.com>
	<4256bf830910121042o50ee79edife2243682bc9d3e2@mail.gmail.com>
	<87f2f3b90910121111m54cf2586pfca0d867307b1e86@mail.gmail.com>
Message-ID: <87f2f3b90910121134j25c728f5tb831cde418a14de7@mail.gmail.com>

Turn on debug, make another test call, and pastebin the output.
-MC

On Mon, Oct 12, 2009 at 11:11 AM, Michael Collins <msc at freeswitch.org>wrote:

>
>
> On Mon, Oct 12, 2009 at 10:42 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>
>> when I add a leg_timeout, I get an ALLOTTED_TIMEOUT from my failed
>> bridge...
>> when an ALLOTTED_TIMEOUT is received, the hangup_after_bridge=false is not
>> recognized (I think). Is there anyway to get an alloted_timeout to continue
>> after bridge (failure)?
>>
>
> Try it with ignore_early_media=true and see if it's the early media that's
> tripping you up.
> -MC
>
>
>>
>> revised dialplan & cmd output
>>
>>     <extension name="1920"><!--DEBUG-->
>>       <condition field="destination_number" expression="^1920$">
>>         <action application="set" data="hangup_after_bridge=false"/>
>>         <action application="bridge" data="{leg_timeout=10}sofia/gateway/
>> debug.com/14159927717"/>
>>          <action application="transfer" data="1999"/><!-- send to unable
>> to reach any contacts-->
>>       </condition>
>>     </extension>
>>
>> freeswitch at matthew-laptop> originate sofia/internal/sip_1%192.168.1.10
>> 1920
>> 2009-10-12 17:39:22.237622 [NOTICE] switch_channel.c:613 New Channel
>> sofia/internal/sip_1 [1d58fb59-c7f9-4908-b612-5bd1c12083cc]
>> 2009-10-12 17:39:22.313524 [NOTICE] sofia.c:3552 Ring-Ready
>> sofia/internal/sip_1!
>> 2009-10-12 17:39:25.142400 [NOTICE] sofia.c:3998 Channel
>> [sofia/internal/sip_1] has been answered
>> 2009-10-12 17:39:25.146259 [INFO] mod_dialplan_xml.c:391 Processing
>> FreeSWITCH->1920 in context default
>> 2009-10-12 17:39:25.146259 [NOTICE] switch_channel.c:613 New Channel
>> sofia/external/14159927717 [38cb7046-0c0d-47ef-94b5-bf8ccf35d185]
>> 2009-10-12 17:39:25.152026 [NOTICE] switch_ivr.c:1367 Transfer
>> sofia/internal/sip_1 to XML[1920 at default]
>>  API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
>> +OK 1d58fb59-c7f9-4908-b612-5bd1c12083cc
>>
>> freeswitch at matthew-laptop> 2009-10-12 17:39:25.217629 [NOTICE]
>> sofia.c:3552 Ring-Ready sofia/external/14159927717!
>> 2009-10-12 17:39:36.017513 [NOTICE] switch_ivr_originate.c:297 Hangup
>> sofia/external/14159927717 [CS_CONSUME_MEDIA] [ALLOTTED_TIMEOUT]
>> 2009-10-12 17:39:36.017513 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>> debug.com hc:ALLOTTED_TIMEOUT du:0 cn:sofia/external/14159927717
>> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1087 Session 12
>> (sofia/external/14159927717) Ended
>> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1089 Close
>> Channel sofia/external/14159927717 [CS_DESTROY]
>> 2009-10-12 17:39:36.037520 [INFO] mod_dptools.c:2133 Originate Failed.
>>  Cause: ALLOTTED_TIMEOUT
>> 2009-10-12 17:39:36.037520 [NOTICE] mod_dptools.c:2166 Hangup
>> sofia/internal/sip_1 [CS_EXECUTE] [ALLOTTED_TIMEOUT]
>> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1087 Session 11
>> (sofia/internal/sip_1) Ended
>> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1089 Close
>> Channel sofia/internal/sip_1 [CS_DESTROY]
>>
>> thanks.
>>
>> --matt
>>
>>
>> On Mon, Oct 12, 2009 at 10:45 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>>
>>> doh! thanks!
>>>
>>>
>>> On Mon, Oct 12, 2009 at 10:33 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> because the regex is on 1997 not 1999
>>>>
>>>>
>>>>
>>>> On Mon, Oct 12, 2009 at 10:25 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>>
>>>>>     <extension name="1999"><!--DIRECT POWER-->
>>>>>       <condition field="destination_number" expression="^1997$">
>>>>>         <action application="playback"
>>>>> data="hh/hh-unable_to_connect_contact.wav"/>
>>>>>         <action application="park"/>
>>>>>       </condition>
>>>>>     </extension>
>>>>>
>>>>> my extn 1999... since it looks from the output like it's transferring,
>>>>> just don't know why it's disconnecting the call instead of playing the .wav
>>>>> and parking.
>>>>>
>>>>> On Mon, Oct 12, 2009 at 10:23 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>>>
>>>>>> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179
>>>>>> Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
>>>>>>
>>>>>> might be the line..or the entire output is below....
>>>>>>
>>>>>> freeswitch at matthew-laptop> originate
>>>>>> sofia/internal/sip_1%192.168.1.10 1920
>>>>>> 2009-10-12 15:21:44.029517 [NOTICE] switch_channel.c:613 New Channel
>>>>>> sofia/internal/sip_1 [1e722934-7e94-46aa-9d62-e6ec7e7449cf]
>>>>>> 2009-10-12 15:21:44.121484 [NOTICE] sofia.c:3552 Ring-Ready
>>>>>> sofia/internal/sip_1!
>>>>>> 2009-10-12 15:21:47.285531 [NOTICE] sofia.c:3998 Channel
>>>>>> [sofia/internal/sip_1] has been answered
>>>>>> 2009-10-12 15:21:47.290996 [INFO] mod_dialplan_xml.c:391 Processing
>>>>>> FreeSWITCH->1920 in context default
>>>>>> 2009-10-12 15:21:47.293452 [NOTICE] switch_channel.c:613 New Channel
>>>>>> sofia/external/14159927717 [6b6cc440-e1d6-415a-b84b-494117e7361d]
>>>>>> 2009-10-12 15:21:47.293452 [NOTICE] switch_ivr.c:1367 Transfer
>>>>>> sofia/internal/sip_1 to XML[1920 at default]
>>>>>> API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
>>>>>> +OK 1e722934-7e94-46aa-9d62-e6ec7e7449cf
>>>>>>
>>>>>> freeswitch at matthew-laptop> 2009-10-12 15:21:47.369855 [NOTICE]
>>>>>> sofia.c:3552 Ring-Ready sofia/external/14159927717!
>>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr_originate.c:2336 Hangup
>>>>>> sofia/external/14159927717 [CS_CONSUME_MEDIA] [NO_ANSWER]
>>>>>> 2009-10-12 15:22:47.009474 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>>>>>> debug.com hc:NO_ANSWER du:0 cn:sofia/external/14159927717
>>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1087 Session
>>>>>> 47 (sofia/external/14159927717) Ended
>>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1089 Close
>>>>>> Channel sofia/external/14159927717 [CS_DESTROY]
>>>>>> 2009-10-12 15:22:47.009474 [INFO] mod_dptools.c:2133 Originate Failed.
>>>>>>  Cause: NO_ANSWER
>>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr.c:1367 Transfer
>>>>>> sofia/internal/sip_1 to XML[1999 at default]
>>>>>> 2009-10-12 15:22:47.009474 [INFO] mod_dialplan_xml.c:391 Processing
>>>>>> FreeSWITCH->1999 in context default
>>>>>> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179
>>>>>> Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
>>>>>> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1087 Session
>>>>>> 46 (sofia/internal/sip_1) Ended
>>>>>> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1089 Close
>>>>>> Channel sofia/internal/sip_1 [CS_DESTROY]
>>>>>>
>>>>>>
>>>>>> thanks for looking at this.
>>>>>>
>>>>>> On Mon, Oct 12, 2009 at 10:06 PM, Anthony Minessale <
>>>>>> anthony.minessale at gmail.com> wrote:
>>>>>>
>>>>>>> which line is hanging up your A (inbound) leg?
>>>>>>>
>>>>>>> look for a blue line that says "Hangup xyz...." that matches it so i
>>>>>>> can see.
>>>>>>>
>>>>>>> I think what is happening is you are getting early media so the
>>>>>>> bridge is actually working then when nobody answers it dies but technically
>>>>>>> the bridge worked.
>>>>>>>
>>>>>>> On Mon, Oct 12, 2009 at 9:41 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>>>>>
>>>>>>>> I think think this might be a bug, but wanted to post here instead
>>>>>>>> of Jira in-case I'm overlooking a configuration variable
>>>>>>>> Dialplan
>>>>>>>>
>>>>>>>>     <extension name="1920"><!--init agent for manual and power dial
>>>>>>>> mode-->
>>>>>>>>       <condition field="destination_number" expression="^1920$">
>>>>>>>>         <action application="set" data="hangup_after_bridge=false"/>
>>>>>>>>         <action application="bridge" data="sofia/gateway/
>>>>>>>> debug.com/14159927717"/>
>>>>>>>>         <action application="transfer" data="1999"/><!-- send to
>>>>>>>> unable to reach any contacts-->
>>>>>>>>       </condition>
>>>>>>>>     </extension>
>>>>>>>>
>>>>>>>> API Command
>>>>>>>> originate sofia/internal/sip_1%192.168.1.10 1920
>>>>>>>>
>>>>>>>> When the bridge to 14159927717 fails (NO_ANSWER) both calls are
>>>>>>>> terminated instead of continuing on in the dial plan to exten 1999 (which in
>>>>>>>> my dialplan parks the call). hangup_after_bridge however seems to work OK if
>>>>>>>> someone picks up in the bridge. Is this the correct behavior? How else can I
>>>>>>>> prevent the call from hanging up if a bridge fails? Thanks.
>>>>>>>>
>>>>>>>> I'm using 15135M
>>>>>>>>
>>>>>>>> --matt
>>>>>>>> http://www.hellohunter.com - Predictive Dialer
>>>>>>>> http://www.hellohunter.com/voice_broadcast.php - Voice Broadcasting
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Anthony Minessale II
>>>>>>>
>>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>>> ClueCon http://www.cluecon.com/
>>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>>
>>>>>>> AIM: anthm
>>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>>
>>>>>>> FreeSWITCH Developer Conference
>>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>>> pstn:213-799-1400
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:213-799-1400
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From msc at freeswitch.org  Mon Oct 12 11:35:29 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 11:35:29 -0700
Subject: [Freeswitch-users] Is assistivetech.net confusing the market
	place with their use of FreeSwitch product name?
In-Reply-To: <9dc4a1670910111048y2802b155oc7086b975193c9d7@mail.gmail.com>
References: <9dc4a1670910111048y2802b155oc7086b975193c9d7@mail.gmail.com>
Message-ID: <87f2f3b90910121135v89c348eoc033cfea692fa041@mail.gmail.com>

On Sun, Oct 11, 2009 at 10:48 AM, EdPimentl <edpimentl at gmail.com> wrote:

> http://www.assistivetech.net/search/productDisplay.php?product_id=18854
> -E
>
> Let's keep an eye on this but I doubt that an assistive technology
application is going to create any real confusion.
-MC

>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Oct 12 11:49:54 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 11:49:54 -0700
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <7d79b3930910120454m1ffc6dd4u42b91a4349597166@mail.gmail.com>
References: <7d79b3930910090407m5dc81d12pf2b0a47f6017804@mail.gmail.com>
	<20091009123615.BC50A3FA6E9@mail.cune.org>
	<7d79b3930910100647u7fd11413ucec0a535d68b3e3a@mail.gmail.com>
	<87f2f3b90910101009j1295d812jd98cae91e30b1eff@mail.gmail.com>
	<7d79b3930910120454m1ffc6dd4u42b91a4349597166@mail.gmail.com>
Message-ID: <87f2f3b90910121149x6e2114f0q996a262ad272a1f6@mail.gmail.com>

Lak,

Okay I will need a little bit of time to dig into the IE's and what they
contain. In the meantime can you tell me who the carrier is? I'd like to
find out if they have some specific requirements. The fact that it doesn't
work with libpri surprises me because that would mean that Asterisk systems
would probably not work with this carrier as well.

BTW, thanks for the very complete pastebin entries. :)
-MC
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From msc at freeswitch.org  Mon Oct 12 11:51:56 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 11:51:56 -0700
Subject: [Freeswitch-users] Upgrading causes no answer
In-Reply-To: <00ea01ca469b$7929faf0$6b7df0d0$@com>
References: <00ea01ca469b$7929faf0$6b7df0d0$@com>
Message-ID: <87f2f3b90910121151v229fb65fy1fd60ae1db2604c9@mail.gmail.com>

On Tue, Oct 6, 2009 at 8:41 AM, Lars Zeb <larclap at yahoo.com> wrote:

>  http://pastebin.freeswitch.org/10612
>
>
>
> I having been running v14996 OK for a while. I have upgraded a couple of
> times after, but every time, an inbound call is hung up on. The only thing
> that has changed is the upgrade. This morning I upgraded to v15098 and the
> problem persists.
>
>
>
> I believe it has to do with a lua script I use for inbound calls. Reading
> from the log, just after the script is launched, the following two lines
> appear:
>
>
>
> switch_cpp.cpp:1116  session not ready
>
> switch_cpp.cpp:925 destroy/unlink session from object
>
>
>
> Has something changed recently with lua processing? Is there something in
> the lua script which is causing the problem?
>
> Just following up. Have you upgraded to the most recent SVN and tried
again? If so, is the problem still here? Please let us know.
-MC
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From mattdfong at gmail.com  Mon Oct 12 12:00:29 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Tue, 13 Oct 2009 02:00:29 +0700
Subject: [Freeswitch-users] hangup_after_bridge=false not continueing
	after failed bridge
In-Reply-To: <87f2f3b90910121134j25c728f5tb831cde418a14de7@mail.gmail.com>
References: <4256bf830910120741g5facaa6cr7ed92303dfafe74b@mail.gmail.com>
	<191c3a030910120806o30911131o743aa58420b41ae4@mail.gmail.com>
	<4256bf830910120823w5fcbc3eja4afb76b60b68fda@mail.gmail.com>
	<4256bf830910120825u70e988p4d7a041c520048a@mail.gmail.com>
	<191c3a030910120833i691b6fbfk3eecbf40e5496479@mail.gmail.com>
	<4256bf830910120845g4f991a55m8753c772dcc8cc31@mail.gmail.com>
	<4256bf830910121042o50ee79edife2243682bc9d3e2@mail.gmail.com>
	<87f2f3b90910121111m54cf2586pfca0d867307b1e86@mail.gmail.com>
	<87f2f3b90910121134j25c728f5tb831cde418a14de7@mail.gmail.com>
Message-ID: <4256bf830910121200x7c0c0f30lfdab7ea5fbb4abc5@mail.gmail.com>

http://pastebin.freeswitch.org/10656

On Tue, Oct 13, 2009 at 1:34 AM, Michael Collins <msc at freeswitch.org> wrote:

> Turn on debug, make another test call, and pastebin the output.
> -MC
>
>
> On Mon, Oct 12, 2009 at 11:11 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Mon, Oct 12, 2009 at 10:42 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>>
>>> when I add a leg_timeout, I get an ALLOTTED_TIMEOUT from my failed
>>> bridge...
>>> when an ALLOTTED_TIMEOUT is received, the hangup_after_bridge=false is
>>> not recognized (I think). Is there anyway to get an alloted_timeout to
>>> continue after bridge (failure)?
>>>
>>
>> Try it with ignore_early_media=true and see if it's the early media that's
>> tripping you up.
>> -MC
>>
>>
>>>
>>> revised dialplan & cmd output
>>>
>>>     <extension name="1920"><!--DEBUG-->
>>>       <condition field="destination_number" expression="^1920$">
>>>         <action application="set" data="hangup_after_bridge=false"/>
>>>         <action application="bridge" data="{leg_timeout=10}sofia/gateway/
>>> debug.com/14159927717"/>
>>>          <action application="transfer" data="1999"/><!-- send to unable
>>> to reach any contacts-->
>>>       </condition>
>>>     </extension>
>>>
>>> freeswitch at matthew-laptop> originate sofia/internal/sip_1%192.168.1.10
>>> 1920
>>> 2009-10-12 17:39:22.237622 [NOTICE] switch_channel.c:613 New Channel
>>> sofia/internal/sip_1 [1d58fb59-c7f9-4908-b612-5bd1c12083cc]
>>> 2009-10-12 17:39:22.313524 [NOTICE] sofia.c:3552 Ring-Ready
>>> sofia/internal/sip_1!
>>> 2009-10-12 17:39:25.142400 [NOTICE] sofia.c:3998 Channel
>>> [sofia/internal/sip_1] has been answered
>>> 2009-10-12 17:39:25.146259 [INFO] mod_dialplan_xml.c:391 Processing
>>> FreeSWITCH->1920 in context default
>>> 2009-10-12 17:39:25.146259 [NOTICE] switch_channel.c:613 New Channel
>>> sofia/external/14159927717 [38cb7046-0c0d-47ef-94b5-bf8ccf35d185]
>>> 2009-10-12 17:39:25.152026 [NOTICE] switch_ivr.c:1367 Transfer
>>> sofia/internal/sip_1 to XML[1920 at default]
>>>  API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
>>> +OK 1d58fb59-c7f9-4908-b612-5bd1c12083cc
>>>
>>> freeswitch at matthew-laptop> 2009-10-12 17:39:25.217629 [NOTICE]
>>> sofia.c:3552 Ring-Ready sofia/external/14159927717!
>>> 2009-10-12 17:39:36.017513 [NOTICE] switch_ivr_originate.c:297 Hangup
>>> sofia/external/14159927717 [CS_CONSUME_MEDIA] [ALLOTTED_TIMEOUT]
>>> 2009-10-12 17:39:36.017513 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>>> debug.com hc:ALLOTTED_TIMEOUT du:0 cn:sofia/external/14159927717
>>> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1087 Session 12
>>> (sofia/external/14159927717) Ended
>>> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1089 Close
>>> Channel sofia/external/14159927717 [CS_DESTROY]
>>> 2009-10-12 17:39:36.037520 [INFO] mod_dptools.c:2133 Originate Failed.
>>>  Cause: ALLOTTED_TIMEOUT
>>> 2009-10-12 17:39:36.037520 [NOTICE] mod_dptools.c:2166 Hangup
>>> sofia/internal/sip_1 [CS_EXECUTE] [ALLOTTED_TIMEOUT]
>>> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1087 Session 11
>>> (sofia/internal/sip_1) Ended
>>> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1089 Close
>>> Channel sofia/internal/sip_1 [CS_DESTROY]
>>>
>>> thanks.
>>>
>>> --matt
>>>
>>>
>>> On Mon, Oct 12, 2009 at 10:45 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>
>>>> doh! thanks!
>>>>
>>>>
>>>> On Mon, Oct 12, 2009 at 10:33 PM, Anthony Minessale <
>>>> anthony.minessale at gmail.com> wrote:
>>>>
>>>>> because the regex is on 1997 not 1999
>>>>>
>>>>>
>>>>>
>>>>> On Mon, Oct 12, 2009 at 10:25 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>>>
>>>>>>     <extension name="1999"><!--DIRECT POWER-->
>>>>>>       <condition field="destination_number" expression="^1997$">
>>>>>>         <action application="playback"
>>>>>> data="hh/hh-unable_to_connect_contact.wav"/>
>>>>>>         <action application="park"/>
>>>>>>       </condition>
>>>>>>     </extension>
>>>>>>
>>>>>> my extn 1999... since it looks from the output like it's transferring,
>>>>>> just don't know why it's disconnecting the call instead of playing the .wav
>>>>>> and parking.
>>>>>>
>>>>>> On Mon, Oct 12, 2009 at 10:23 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>>>>
>>>>>>> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179
>>>>>>> Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
>>>>>>>
>>>>>>> might be the line..or the entire output is below....
>>>>>>>
>>>>>>> freeswitch at matthew-laptop> originate
>>>>>>> sofia/internal/sip_1%192.168.1.10 1920
>>>>>>> 2009-10-12 15:21:44.029517 [NOTICE] switch_channel.c:613 New Channel
>>>>>>> sofia/internal/sip_1 [1e722934-7e94-46aa-9d62-e6ec7e7449cf]
>>>>>>> 2009-10-12 15:21:44.121484 [NOTICE] sofia.c:3552 Ring-Ready
>>>>>>> sofia/internal/sip_1!
>>>>>>> 2009-10-12 15:21:47.285531 [NOTICE] sofia.c:3998 Channel
>>>>>>> [sofia/internal/sip_1] has been answered
>>>>>>> 2009-10-12 15:21:47.290996 [INFO] mod_dialplan_xml.c:391 Processing
>>>>>>> FreeSWITCH->1920 in context default
>>>>>>> 2009-10-12 15:21:47.293452 [NOTICE] switch_channel.c:613 New Channel
>>>>>>> sofia/external/14159927717 [6b6cc440-e1d6-415a-b84b-494117e7361d]
>>>>>>> 2009-10-12 15:21:47.293452 [NOTICE] switch_ivr.c:1367 Transfer
>>>>>>> sofia/internal/sip_1 to XML[1920 at default]
>>>>>>> API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
>>>>>>> +OK 1e722934-7e94-46aa-9d62-e6ec7e7449cf
>>>>>>>
>>>>>>> freeswitch at matthew-laptop> 2009-10-12 15:21:47.369855 [NOTICE]
>>>>>>> sofia.c:3552 Ring-Ready sofia/external/14159927717!
>>>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr_originate.c:2336
>>>>>>> Hangup sofia/external/14159927717 [CS_CONSUME_MEDIA] [NO_ANSWER]
>>>>>>> 2009-10-12 15:22:47.009474 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>>>>>>> debug.com hc:NO_ANSWER du:0 cn:sofia/external/14159927717
>>>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1087
>>>>>>> Session 47 (sofia/external/14159927717) Ended
>>>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1089 Close
>>>>>>> Channel sofia/external/14159927717 [CS_DESTROY]
>>>>>>> 2009-10-12 15:22:47.009474 [INFO] mod_dptools.c:2133 Originate
>>>>>>> Failed.  Cause: NO_ANSWER
>>>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr.c:1367 Transfer
>>>>>>> sofia/internal/sip_1 to XML[1999 at default]
>>>>>>> 2009-10-12 15:22:47.009474 [INFO] mod_dialplan_xml.c:391 Processing
>>>>>>> FreeSWITCH->1999 in context default
>>>>>>> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179
>>>>>>> Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
>>>>>>> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1087
>>>>>>> Session 46 (sofia/internal/sip_1) Ended
>>>>>>> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1089 Close
>>>>>>> Channel sofia/internal/sip_1 [CS_DESTROY]
>>>>>>>
>>>>>>>
>>>>>>> thanks for looking at this.
>>>>>>>
>>>>>>> On Mon, Oct 12, 2009 at 10:06 PM, Anthony Minessale <
>>>>>>> anthony.minessale at gmail.com> wrote:
>>>>>>>
>>>>>>>> which line is hanging up your A (inbound) leg?
>>>>>>>>
>>>>>>>> look for a blue line that says "Hangup xyz...." that matches it so i
>>>>>>>> can see.
>>>>>>>>
>>>>>>>> I think what is happening is you are getting early media so the
>>>>>>>> bridge is actually working then when nobody answers it dies but technically
>>>>>>>> the bridge worked.
>>>>>>>>
>>>>>>>> On Mon, Oct 12, 2009 at 9:41 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>>>>>>
>>>>>>>>> I think think this might be a bug, but wanted to post here instead
>>>>>>>>> of Jira in-case I'm overlooking a configuration variable
>>>>>>>>> Dialplan
>>>>>>>>>
>>>>>>>>>     <extension name="1920"><!--init agent for manual and power dial
>>>>>>>>> mode-->
>>>>>>>>>       <condition field="destination_number" expression="^1920$">
>>>>>>>>>         <action application="set"
>>>>>>>>> data="hangup_after_bridge=false"/>
>>>>>>>>>         <action application="bridge" data="sofia/gateway/
>>>>>>>>> debug.com/14159927717"/>
>>>>>>>>>         <action application="transfer" data="1999"/><!-- send to
>>>>>>>>> unable to reach any contacts-->
>>>>>>>>>       </condition>
>>>>>>>>>     </extension>
>>>>>>>>>
>>>>>>>>> API Command
>>>>>>>>> originate sofia/internal/sip_1%192.168.1.10 1920
>>>>>>>>>
>>>>>>>>> When the bridge to 14159927717 fails (NO_ANSWER) both calls are
>>>>>>>>> terminated instead of continuing on in the dial plan to exten 1999 (which in
>>>>>>>>> my dialplan parks the call). hangup_after_bridge however seems to work OK if
>>>>>>>>> someone picks up in the bridge. Is this the correct behavior? How else can I
>>>>>>>>> prevent the call from hanging up if a bridge fails? Thanks.
>>>>>>>>>
>>>>>>>>> I'm using 15135M
>>>>>>>>>
>>>>>>>>> --matt
>>>>>>>>> http://www.hellohunter.com - Predictive Dialer
>>>>>>>>> http://www.hellohunter.com/voice_broadcast.php - Voice
>>>>>>>>> Broadcasting
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> Anthony Minessale II
>>>>>>>>
>>>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>>>> ClueCon http://www.cluecon.com/
>>>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>>>
>>>>>>>> AIM: anthm
>>>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>>>
>>>>>>>> FreeSWITCH Developer Conference
>>>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>>>> pstn:213-799-1400
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>> iax:guest at conference.freeswitch.org/888
>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>> pstn:213-799-1400
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Oct 12 12:01:05 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 12:01:05 -0700
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Update: Valet
	Parking
In-Reply-To: <15b9404e0910102018oda8834ah8cc87787ee00c22b@mail.gmail.com>
References: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>
	<15b9404e0910102018oda8834ah8cc87787ee00c22b@mail.gmail.com>
Message-ID: <87f2f3b90910121201o422c7891vd30ee3761aef0eb7@mail.gmail.com>

On Sat, Oct 10, 2009 at 8:18 PM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:

>
>
> On Fri, Oct 9, 2009 at 10:42 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> FYI,
>>
>> The FreeSWITCH devs have added valet parking! Check it out:
>> http://www.freeswitch.org/node/207
>>
>> Let us know what you think.
>>
>
> Very nice.
>
> But I think a valet_unpark app is missing.
> If the intention of the person sent to the valet lot is to retrieve a call
> there, the person can assume the call was already retrieved by someone else
> or that the caller hung up if he/she hears MOH. But it would be nicer to
> have a valet_unpark app to fail and let the dialplan play a message.
>
> I understand what you are saying. I'm not sure I agree, but we'll kick the
idea around when we have a few minutes and let you know what we decide.
-MC
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From Russell.Mosemann at cune.org  Mon Oct 12 12:20:07 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Mon, 12 Oct 2009 19:20:07 -0000
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Update:
	Valet	Parking
In-Reply-To: <87f2f3b90910121201o422c7891vd30ee3761aef0eb7@mail.gmail.com>
Message-ID: <20091012192008.05FE3301FB6@mail.cune.org>

Michael Collins <msc at freeswitch.org> said:

> On Sat, Oct 10, 2009 at 8:18 PM, mayamatakeshi
<mayamatakeshi at gmail.com>wrote:
> 
> >
> >
> > On Fri, Oct 9, 2009 at 10:42 AM, Michael Collins
<msc at freeswitch.org>wrote:
> >
> >> FYI,
> >>
> >> The FreeSWITCH devs have added valet parking! Check it out:
> >> http://www.freeswitch.org/node/207
> >>
> >> Let us know what you think.
> >>
> >
> > Very nice.
> >
> > But I think a valet_unpark app is missing.
> > If the intention of the person sent to the valet lot is to retrieve a
call
> > there, the person can assume the call was already retrieved by
someone else
> > or that the caller hung up if he/she hears MOH. But it would be nicer to
> > have a valet_unpark app to fail and let the dialplan play a message.
> >
> > I understand what you are saying. I'm not sure I agree, but we'll
kick the
> idea around when we have a few minutes and let you know what we decide.
> -MC

If you do decide to implement something, I would encourage that it be
flexible so that when the parking meter runs out :-), it could either
play a message or forward the call to an extension (default to the
extension that parked it).

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From jmesquita at freeswitch.org  Mon Oct 12 12:37:26 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 12 Oct 2009 16:37:26 -0300
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Update: Valet
	Parking
In-Reply-To: <20091012192008.05FE3301FB6@mail.cune.org>
References: <87f2f3b90910121201o422c7891vd30ee3761aef0eb7@mail.gmail.com>
	<20091012192008.05FE3301FB6@mail.cune.org>
Message-ID: <ebbc2d550910121237rbca507fpa76880b0b99d03f4@mail.gmail.com>

I would say that the parking meter is a good idea and it is the default
behavior of parking on legacy PBXs. Since we always do _more_, what do you
think about having the option to transfer to any extension instead of just
the one that transfered the call?

Regards,

jm

On Mon, Oct 12, 2009 at 4:20 PM, <Russell.Mosemann at cune.org> wrote:

> Michael Collins <msc at freeswitch.org> said:
>
> > On Sat, Oct 10, 2009 at 8:18 PM, mayamatakeshi
> <mayamatakeshi at gmail.com>wrote:
> >
> > >
> > >
> > > On Fri, Oct 9, 2009 at 10:42 AM, Michael Collins
> <msc at freeswitch.org>wrote:
> > >
> > >> FYI,
> > >>
> > >> The FreeSWITCH devs have added valet parking! Check it out:
> > >> http://www.freeswitch.org/node/207
> > >>
> > >> Let us know what you think.
> > >>
> > >
> > > Very nice.
> > >
> > > But I think a valet_unpark app is missing.
> > > If the intention of the person sent to the valet lot is to retrieve a
> call
> > > there, the person can assume the call was already retrieved by
> someone else
> > > or that the caller hung up if he/she hears MOH. But it would be nicer
> to
> > > have a valet_unpark app to fail and let the dialplan play a message.
> > >
> > > I understand what you are saying. I'm not sure I agree, but we'll
> kick the
> > idea around when we have a few minutes and let you know what we decide.
> > -MC
>
> If you do decide to implement something, I would encourage that it be
> flexible so that when the parking meter runs out :-), it could either
> play a message or forward the call to an extension (default to the
> extension that parked it).
>
> --
> Russell Mosemann
>
>
>
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From quentusrex at gmail.com  Mon Oct 12 12:55:20 2009
From: quentusrex at gmail.com (William King)
Date: Mon, 12 Oct 2009 12:55:20 -0700
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Update: Valet
 Parking
In-Reply-To: <87f2f3b90910121201o422c7891vd30ee3761aef0eb7@mail.gmail.com>
References: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>	<15b9404e0910102018oda8834ah8cc87787ee00c22b@mail.gmail.com>
	<87f2f3b90910121201o422c7891vd30ee3761aef0eb7@mail.gmail.com>
Message-ID: <4AD389A8.1080603@gmail.com>

I don't know if this was mentioned yet. It would be useful to have a way
to have the parking lot automatically find the next available spot and
tts it to the person parking the call.

Then the auto unpark would pop off the lowest numbered lot, or return
fail if there is nobody in the parking lots etc.

-William King

Michael Collins wrote:
>
>
> On Sat, Oct 10, 2009 at 8:18 PM, mayamatakeshi
> <mayamatakeshi at gmail.com <mailto:mayamatakeshi at gmail.com>> wrote:
>
>
>
>     On Fri, Oct 9, 2009 at 10:42 AM, Michael Collins
>     <msc at freeswitch.org <mailto:msc at freeswitch.org>> wrote:
>
>         FYI,
>
>         The FreeSWITCH devs have added valet parking! Check it out:
>         http://www.freeswitch.org/node/207
>
>         Let us know what you think.
>
>
>     Very nice.
>
>     But I think a valet_unpark app is missing.
>     If the intention of the person sent to the valet lot is to
>     retrieve a call there, the person can assume the call was already
>     retrieved by someone else or that the caller hung up if he/she
>     hears MOH. But it would be nicer to have a valet_unpark app to
>     fail and let the dialplan play a message.
>
> I understand what you are saying. I'm not sure I agree, but we'll kick
> the idea around when we have a few minutes and let you know what we
> decide.
> -MC
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From vinuth.madinur at gmail.com  Mon Oct 12 12:58:32 2009
From: vinuth.madinur at gmail.com (Vinuth Madinur)
Date: Tue, 13 Oct 2009 01:28:32 +0530
Subject: [Freeswitch-users] SIT tones and SIP Trunk provider.
Message-ID: <910309030910121258r4afe6b96g9db4756e9f0e090b@mail.gmail.com>

Hi,
Does Freeswitch detect all of these hangup cases mentioned here [
http://wiki.freeswitch.org/wiki/Hangup_causes] when using it through a SIP
Trunk provider?

If not, should I put in tone_detect application in the dialplan for
detecting the SITs?

Won't freeswitch have to depend on the SIP status sent from SIP trunk to
know the hangup status? So, I'm wondering if tone_detect will work at all?

Please provide your advice.

Thanks,
Vinuth.
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From msc at freeswitch.org  Mon Oct 12 13:31:19 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 13:31:19 -0700
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Update: Valet
	Parking
In-Reply-To: <4AD389A8.1080603@gmail.com>
References: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>
	<15b9404e0910102018oda8834ah8cc87787ee00c22b@mail.gmail.com>
	<87f2f3b90910121201o422c7891vd30ee3761aef0eb7@mail.gmail.com>
	<4AD389A8.1080603@gmail.com>
Message-ID: <87f2f3b90910121331y23a11983vd40f75f861249623@mail.gmail.com>

On Mon, Oct 12, 2009 at 12:55 PM, William King <quentusrex at gmail.com> wrote:

> I don't know if this was mentioned yet. It would be useful to have a way
> to have the parking lot automatically find the next available spot and
> tts it to the person parking the call.
>
> Then the auto unpark would pop off the lowest numbered lot, or return
> fail if there is nobody in the parking lots etc.
>

Of course, if you're gonna do this then you might as well use FIFO queues.
;)
-MC
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From msc at freeswitch.org  Mon Oct 12 13:39:09 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 13:39:09 -0700
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Update: Valet
	Parking
In-Reply-To: <ebbc2d550910121237rbca507fpa76880b0b99d03f4@mail.gmail.com>
References: <87f2f3b90910121201o422c7891vd30ee3761aef0eb7@mail.gmail.com>
	<20091012192008.05FE3301FB6@mail.cune.org>
	<ebbc2d550910121237rbca507fpa76880b0b99d03f4@mail.gmail.com>
Message-ID: <87f2f3b90910121339m3e08484enbc2d9eab746bf03d@mail.gmail.com>

2009/10/12 Jo?o Mesquita <jmesquita at freeswitch.org>

> I would say that the parking meter is a good idea and it is the default
> behavior of parking on legacy PBXs. Since we always do _more_, what do you
> think about having the option to transfer to any extension instead of just
> the one that transfered the call?
>
>
Gents,

Thanks for all the feedback. I think the proper thing to do is to make sure
that we have, as Brian would put it, all the Lego(tm) bricks necessary to
build the functionality that you wish to implement. A lot of what I'm
hearing that you guys want can be handled with FIFO. Before we do any more
programming with valet_park let's make sure that the existing Legos don't
already give you the tools that you need.

-MC
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From msc at freeswitch.org  Mon Oct 12 13:45:13 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 13:45:13 -0700
Subject: [Freeswitch-users] SIT tones and SIP Trunk provider.
In-Reply-To: <910309030910121258r4afe6b96g9db4756e9f0e090b@mail.gmail.com>
References: <910309030910121258r4afe6b96g9db4756e9f0e090b@mail.gmail.com>
Message-ID: <87f2f3b90910121345wef2a3f6u760fde22df0aba81@mail.gmail.com>

On Mon, Oct 12, 2009 at 12:58 PM, Vinuth Madinur
<vinuth.madinur at gmail.com>wrote:

> Hi,
> Does Freeswitch detect all of these hangup cases mentioned here [
> http://wiki.freeswitch.org/wiki/Hangup_causes] when using it through a SIP
> Trunk provider?
>
> If not, should I put in tone_detect application in the dialplan for
> detecting the SITs?
>
> Won't freeswitch have to depend on the SIP status sent from SIP trunk to
> know the hangup status? So, I'm wondering if tone_detect will work at all?
>
>
Vinuth,

As usual, "it depends." Your provider is the key to this whole operation. If
the SIP provider sends the information inband then you will definitely need
to use tone_detect to look for the SIT tones. However, if the information
comes back with the normal SIP messages then you're good to go. I've seen
more than a few SIP providers do both, which means that you have to prepare
for both cases.

My advice to you is to get pcaps of failed calls and analyze them with
Wireshark. If you need help analyzing them then put your pcaps on a web
server and post a link so that others can download them. The wiki has some
information on grabbing pcaps:

http://wiki.freeswitch.org/wiki/Packet_Capture

If you haven't already done so, go to cluecon.com and download the torrent
file that has the ClueCon speaker presentations. The last presentation on
Day 3 is Jason Garland and he walks you through using Wireshark for
analyzing a SIP call, including both the signaling (SIP) and the media (RTP)
parts of the call. BTW, if you have a copy of "VoIP Deployment For Dummies"
it has a small section on using Wireshark for call analysis.

-MC
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From vinuth.madinur at gmail.com  Mon Oct 12 14:19:22 2009
From: vinuth.madinur at gmail.com (Vinuth Madinur)
Date: Tue, 13 Oct 2009 02:49:22 +0530
Subject: [Freeswitch-users] SIT tones and SIP Trunk provider.
In-Reply-To: <87f2f3b90910121345wef2a3f6u760fde22df0aba81@mail.gmail.com>
References: <910309030910121258r4afe6b96g9db4756e9f0e090b@mail.gmail.com>
	<87f2f3b90910121345wef2a3f6u760fde22df0aba81@mail.gmail.com>
Message-ID: <910309030910121419o56a547bft83e5218946a9a0f4@mail.gmail.com>

Thanks Michael. I'll go through the resources you mentioned.
Thanks,
Vinuth.


On Tue, Oct 13, 2009 at 2:15 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Oct 12, 2009 at 12:58 PM, Vinuth Madinur <vinuth.madinur at gmail.com
> > wrote:
>
>> Hi,
>> Does Freeswitch detect all of these hangup cases mentioned here [
>> http://wiki.freeswitch.org/wiki/Hangup_causes] when using it through a
>> SIP Trunk provider?
>>
>> If not, should I put in tone_detect application in the dialplan for
>> detecting the SITs?
>>
>> Won't freeswitch have to depend on the SIP status sent from SIP trunk to
>> know the hangup status? So, I'm wondering if tone_detect will work at all?
>>
>>
> Vinuth,
>
> As usual, "it depends." Your provider is the key to this whole operation.
> If the SIP provider sends the information inband then you will definitely
> need to use tone_detect to look for the SIT tones. However, if the
> information comes back with the normal SIP messages then you're good to go.
> I've seen more than a few SIP providers do both, which means that you have
> to prepare for both cases.
>
> My advice to you is to get pcaps of failed calls and analyze them with
> Wireshark. If you need help analyzing them then put your pcaps on a web
> server and post a link so that others can download them. The wiki has some
> information on grabbing pcaps:
>
> http://wiki.freeswitch.org/wiki/Packet_Capture
>
> If you haven't already done so, go to cluecon.com and download the torrent
> file that has the ClueCon speaker presentations. The last presentation on
> Day 3 is Jason Garland and he walks you through using Wireshark for
> analyzing a SIP call, including both the signaling (SIP) and the media (RTP)
> parts of the call. BTW, if you have a copy of "VoIP Deployment For Dummies"
> it has a small section on using Wireshark for call analysis.
>
> -MC
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Mon Oct 12 14:43:47 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 12 Oct 2009 23:43:47 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
Message-ID: <65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>

hi,

finally i compiled it right ... had a stupid issue with ekiga and wrong
ptlib in place...

anyhow, i loaded the module and will continue the tests tomorrow ...first
thing i arrive in my office :P



freeswitch at subZero>
freeswitch at subZero>
API CALL [console(loglevel 7)] output:
+OK console log level set to DEBUG

freeswitch at subZero> load mod_h323
2009-10-12 23:38:09.509279 [CONSOLE] mod_h323.cpp:93 Starting loading
mod_h323
2009-10-12 23:38:09.511413 [DEBUG] mod_h323.cpp:461 Created Listener
'default'
2009-10-12 23:38:09.511413 [DEBUG] mod_h323.cpp:305 Config capabilliti
PCMA,GSM,G729,G726
2009-10-12 23:38:09.511413 [DEBUG] mod_h323.cpp:309 Find capabilliti PCMU to
PCMA,GSM,G729,G726
2009-10-12 23:38:09.511413 [DEBUG] mod_h323.cpp:309 Find capabilliti PCMA to
PCMA,GSM,G729,G726
2009-10-12 23:38:09.511413 [DEBUG] h323caps.cxx:3264 FindCapability:
"G.711-uLaw-64k{sw}"
2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:3176 Added capability:
G.711-uLaw-64k <1>
2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:3264 FindCapability:
"G.711-uLaw-64k{sw}"
2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:3264 FindCapability:
"G.711-uLaw-64k{sw}"
2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:3264 FindCapability:
"G.711-uLaw-64k{sw}"
2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:3264 FindCapability:
"G.711-uLaw-64k{sw}"
2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:1989 No Extended
Capabilities found to load
2009-10-12 23:38:09.512480 [DEBUG] mod_h323.cpp:325 H.323 added 1
capabilities 'G.711-uLaw-64k*{sw}'
2009-10-12 23:38:09.512480 [DEBUG] mod_h323.cpp:309 Find capabilliti gsm to
PCMA,GSM,G729,G726
2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti msgsm
to PCMA,GSM,G729,G726
2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti speex
to PCMA,GSM,G729,G726
2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti lpc10
to PCMA,GSM,G729,G726
2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti ilbc20
to PCMA,GSM,G729,G726
2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti ilbc30
to PCMA,GSM,G729,G726
2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g723 to
PCMA,GSM,G729,G726
2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g726 to
PCMA,GSM,G729,G726
2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g728 to
PCMA,GSM,G729,G726
2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g729b
to PCMA,GSM,G729,G726
2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g729 to
PCMA,GSM,G729,G726
2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti slin to
PCMA,GSM,G729,G726
2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g729a
to PCMA,GSM,G729,G726
2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g729ab
to PCMA,GSM,G729,G726
2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti g723.1
to PCMA,GSM,G729,G726
2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
g723.1-5k3 to PCMA,GSM,G729,G726
2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
g723.1a-5k3 to PCMA,GSM,G729,G726
2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
g723.1a-6k3 to PCMA,GSM,G729,G726
2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
g723.1a-6k3-cisco to PCMA,GSM,G729,G726
2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti g726-16
to PCMA,GSM,G729,G726
2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti g726-24
to PCMA,GSM,G729,G726
2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti g726-32
to PCMA,GSM,G729,G726
2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti g726-40
to PCMA,GSM,G729,G726
2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti ilbc to
PCMA,GSM,G729,G726
2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
speex-18k2 to PCMA,GSM,G729,G726
2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
speex-15k to PCMA,GSM,G729,G726
2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
speex-11k to PCMA,GSM,G729,G726
2009-10-12 23:38:09.515708 [DEBUG] mod_h323.cpp:309 Find capabilliti
speex-8k to PCMA,GSM,G729,G726
2009-10-12 23:38:09.515708 [DEBUG] mod_h323.cpp:309 Find capabilliti
speex-5k95 to PCMA,GSM,G729,G726
2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
UserInput/hookflash <2>
2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
UserInput/basicString <3>
2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
UserInput/dtmf <4>
2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
UserInput/RFC2833 <5>
2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
UserInput/Navigation <6>
2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
UserInput/Softkey <7>
2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
UserInput/PointDevice <8>
2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
UserInput/Modal <9>
2009-10-12 23:38:09.516760 [DEBUG] osutil.cxx:188 File handle high water
mark set: 44 PTCPSocket
2009-10-12 23:38:09.516760 [DEBUG] h323ep.cxx:1995 Started listener
Listener[ip$10.4.62.7:1720]
2009-10-12 23:38:09.516760 [DEBUG] tlibthrd.cxx:547 Thread high water mark
set: 3
2009-10-12 23:38:09.516760 [CONSOLE] mod_h323.cpp:108 Opal manager
initialized and running
2009-10-12 23:38:09.516760 [DEBUG] tlibthrd.cxx:454 Started thread
0xb75e0f18 H323 Listener:b5c98b90
2009-10-12 23:38:09.516760 [CONSOLE] switch_loadable_module.c:889
Successfully Loaded [mod_h323]
2009-10-12 23:38:09.516760 [NOTICE] switch_loadable_module.c:142 Adding
Endpoint 'h323'
API CALL [load(mod_h323)] output:
+OK

2009-10-12 23:38:09.516760 [DEBUG] transports.cxx:1521 Awaiting TCP
connections on port 1720
2009-10-12 23:38:09.517832 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.4.62.7:1720
2009-10-12 23:38:09.517832 [DEBUG] tlibthrd.cxx:1023
PThread::PXBlockOnIO(44,2)
freeswitch at subZero>


BTW: to compile everytihng,  I used FS trunk, ptlib 2.7.1 and h323plus
1.22.0

Will let you know the results asap.


T.



2009/10/12 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-12 18:03 +0200, Tihomir Culjaga wrote
> freeswitch-users at lists.fre...:
>
> i already write this, ptlib 2.6.5, tou can find link to it on
> oplalvoip.org, h32plus latest CVS version,
> you can find it on www.h323plus.org.
>
> TC>hi,
> TC>
> TC>can't make it...
> TC>
> TC>subZero:~/freeswitch-trunk$ make mod_h323
> TC>
> TC>making all mod_h323
> TC>Compiling mod_h323.cpp...
> TC>quiet_libtool: compile:  g++ -g -ggdb -I/usr/local/include/ptlib
> TC>-I/usr/local/include/openh323 -I. -DPTRACING=1 -D_REENTRANT
> -fno-exceptions
> TC>-I/home/tculjaga/freeswitch-trunk/src/include
> TC>-I/home/tculjaga/freeswitch-trunk/libs/libteletone/src -fPIC
> TC>-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2
> TC>-D_GNU_SOURCE -DHAVE_CONFIG_H -c mod_h323.cpp  -fPIC -DPIC -o
> TC>.libs/mod_h323.o
> TC>In file included from /usr/local/include/openh323/h323.h:493,
> TC>                 from mod_h323.h:8,
> TC>                 from mod_h323.cpp:3:
> TC>/usr/local/include/openh323/h323ep.h: In member function ??????virtual
> void
> TC>NATFactoryStartup::OnShutdown()??????:
> TC>/usr/local/include/openh323/h323ep.h:2731: error: ??????NatFactory??????
> has not
> TC>been declared
> TC>make[4]: *** [mod_h323.lo] Error 1
> TC>make[3]: *** [all] Error 1
> TC>make[2]: *** [mod_h323-all] Error 1
> TC>make[1]: *** [mod_h323] Error 2
> TC>make: *** [mod_h323] Error 2
> TC>
> TC>
> TC>
> TC>what exact ptlib and h323plus versions did you use? .. can you send us a
> TC>link so we can use the exact ones.
> TC>
> TC>
> TC>T.
> TC>
> TC>2009/10/12 Georgiewskiy Yuriy <bottleman at icf.org.ru>
> TC>
> TC>> On 2009-10-12 09:43 -0500, Brian West wrote
> TC>> freeswitch-users at lists.freeswit...:
> TC>>
> TC>> BW>We can host this in our SVN if you wish?
> TC>>
> TC>> If in fs svn i think yes. But i think may be little time later?
> TC>> i don't known is it builds on trunk because i develop it on 1.0.4.
> TC>>
> TC>> BW>/b
> TC>> BW>
> TC>> BW>On Oct 12, 2009, at 8:31 AM, Georgiewskiy Yuriy wrote:
> TC>> BW>
> TC>> BW>> ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/ alfa code,
> but
> TC>> seems
> TC>> BW>> it work, but should be buggy,
> TC>> BW>> to build need libpt 2.6.5 and h323plus cvs version, i test it now
> on
> TC>> fs
> TC>> BW>> 1.0.4.
> TC>> BW>
> TC>>
> TC>> C ?????????                       With Best Regards
> TC>> ???????????? ????.                Georgiewskiy Yuriy
> TC>> +7 4872 711666                    +7 4872 711666
> TC>> ???? +7 4872 711143               fax +7 4872 711143
> TC>> ???????? ??? "?? ?? ??????"       IT Service Ltd
> TC>> http://nkoort.ru                  http://nkoort.ru
> TC>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> TC>> YG129-RIPE                        YG129-RIPE
> TC>>
> TC>> _______________________________________________
> TC>> FreeSWITCH-users mailing list
> TC>> FreeSWITCH-users at lists.freeswitch.org
> TC>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> TC>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> TC>> http://www.freeswitch.org
> TC>>
> TC>>
> TC>
>
> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129-RIPE
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Oct 12 14:48:57 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 14:48:57 -0700
Subject: [Freeswitch-users] hangup_after_bridge=false not continueing
	after failed bridge
In-Reply-To: <4256bf830910121200x7c0c0f30lfdab7ea5fbb4abc5@mail.gmail.com>
References: <4256bf830910120741g5facaa6cr7ed92303dfafe74b@mail.gmail.com>
	<191c3a030910120806o30911131o743aa58420b41ae4@mail.gmail.com>
	<4256bf830910120823w5fcbc3eja4afb76b60b68fda@mail.gmail.com>
	<4256bf830910120825u70e988p4d7a041c520048a@mail.gmail.com>
	<191c3a030910120833i691b6fbfk3eecbf40e5496479@mail.gmail.com>
	<4256bf830910120845g4f991a55m8753c772dcc8cc31@mail.gmail.com>
	<4256bf830910121042o50ee79edife2243682bc9d3e2@mail.gmail.com>
	<87f2f3b90910121111m54cf2586pfca0d867307b1e86@mail.gmail.com>
	<87f2f3b90910121134j25c728f5tb831cde418a14de7@mail.gmail.com>
	<4256bf830910121200x7c0c0f30lfdab7ea5fbb4abc5@mail.gmail.com>
Message-ID: <87f2f3b90910121448w581854d7tf878671430cce133@mail.gmail.com>

On Mon, Oct 12, 2009 at 12:00 PM, Matthew Fong <mattdfong at gmail.com> wrote:

> http://pastebin.freeswitch.org/10656
>
>
> Matthew,

Try continue_on_fail=true instead of hangup_after_bridge=false.
http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail

I think it will do what you want.
-MC
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From diego.viola at gmail.com  Mon Oct 12 15:11:51 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Mon, 12 Oct 2009 22:11:51 +0000
Subject: [Freeswitch-users] Some documentation thoughts
Message-ID: <86a32abc0910121511m4241faefg4b8d1caf3b0f7a87@mail.gmail.com>

Hello,

I have been doing some work recently on the FreeSWITCH wiki, to improve
things.

You can see some of my work here:

http://wiki.freeswitch.org/index.php?title=Special:Contributions&limit=500&target=Diego.viola

I am trying to polish the wiki and give it a more professional and clean
look, this means that we need to enforce some guidelines and strive to make
the wiki even better.

How can we do this?

Well, the first thing is to review everything we have, right now we have too
many separated and little pages here and there that no one cares or read, we
should avoid doing little or small pages. We need to correct typos, I
suggest that people click on "Random page" link and start correcting typos,
install an English (US) dictionary in your browser and enable Spell Checking
and start correcting these typos if you want to help, also, make acronyms
and initialisms all-uppercase when you make one.

When you are adding documentation make sure you don't do a separate page for
an existing page, etc. and we also need to define a size for titles, body,
etc, so different pages don't look different from each other, I will make
sure I add all this into our guidelines and that all pages follow this.

My goal is to have great documentation, similar to the Apache documentation:
http://httpd.apache.org/docs/2.2/

Or even better, I hope you like this idea.

Diego
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From diego.viola at gmail.com  Mon Oct 12 15:21:59 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Mon, 12 Oct 2009 22:21:59 +0000
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <86a32abc0910121511m4241faefg4b8d1caf3b0f7a87@mail.gmail.com>
References: <86a32abc0910121511m4241faefg4b8d1caf3b0f7a87@mail.gmail.com>
Message-ID: <86a32abc0910121521h46c6e0a8r19718c020b6ded5e@mail.gmail.com>

Is this page still necessary:

http://wiki.freeswitch.org/wiki/Old_mod_python

I'd like to have the mod_python page only:

http://wiki.freeswitch.org/wiki/Mod_python

If there is something in the old one please let me know so we can move to
the new one and then get rid of the older one.

Thanks,

Diego

On Mon, Oct 12, 2009 at 10:11 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Hello,
>
> I have been doing some work recently on the FreeSWITCH wiki, to improve
> things.
>
> You can see some of my work here:
>
>
> http://wiki.freeswitch.org/index.php?title=Special:Contributions&limit=500&target=Diego.viola
>
> I am trying to polish the wiki and give it a more professional and clean
> look, this means that we need to enforce some guidelines and strive to make
> the wiki even better.
>
> How can we do this?
>
> Well, the first thing is to review everything we have, right now we have
> too many separated and little pages here and there that no one cares or
> read, we should avoid doing little or small pages. We need to correct typos,
> I suggest that people click on "Random page" link and start correcting
> typos, install an English (US) dictionary in your browser and enable Spell
> Checking and start correcting these typos if you want to help, also, make
> acronyms and initialisms all-uppercase when you make one.
>
> When you are adding documentation make sure you don't do a separate page
> for an existing page, etc. and we also need to define a size for titles,
> body, etc, so different pages don't look different from each other, I will
> make sure I add all this into our guidelines and that all pages follow this.
>
> My goal is to have great documentation, similar to the Apache
> documentation:
> http://httpd.apache.org/docs/2.2/
>
> Or even better, I hope you like this idea.
>
> Diego
>
>
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From anthony.minessale at gmail.com  Mon Oct 12 15:28:11 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 12 Oct 2009 17:28:11 -0500
Subject: [Freeswitch-users] Upgrading causes no answer
In-Reply-To: <00ea01ca469b$7929faf0$6b7df0d0$@com>
References: <00ea01ca469b$7929faf0$6b7df0d0$@com>
Message-ID: <191c3a030910121528w1adb6779xb9923d2a8cdd5943@mail.gmail.com>

if you are using ringback variable you now must also use
ignore_early_media=true either exported from A leg or in {} on the b leg
dial string to get the original behavior.



On Tue, Oct 6, 2009 at 10:41 AM, Lars Zeb <larclap at yahoo.com> wrote:

>  http://pastebin.freeswitch.org/10612
>
>
>
> I having been running v14996 OK for a while. I have upgraded a couple of
> times after, but every time, an inbound call is hung up on. The only thing
> that has changed is the upgrade. This morning I upgraded to v15098 and the
> problem persists.
>
>
>
> I believe it has to do with a lua script I use for inbound calls. Reading
> from the log, just after the script is launched, the following two lines
> appear:
>
>
>
> switch_cpp.cpp:1116  session not ready
>
> switch_cpp.cpp:925 destroy/unlink session from object
>
>
>
> Has something changed recently with lua processing? Is there something in
> the lua script which is causing the problem?
>
>
>
> I would appreciate any help.
>
>
>
> Thanks, Lars
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From gabe at gundy.org  Mon Oct 12 15:33:33 2009
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 12 Oct 2009 16:33:33 -0600
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <87f2f3b90910121035v5cfce2bel7d32da1671e325b6@mail.gmail.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<903da5680910081201p427a8358n768edb61a96674a4@mail.gmail.com>
	<87f2f3b90910081538l6e4e4b7dv3aedb2c4d1af58bc@mail.gmail.com>
	<903da5680910082238x27bff476s8bea0e74ab516a8d@mail.gmail.com>
	<87f2f3b90910082308i7f6144d9kc887596be54dd609@mail.gmail.com>
	<903da5680910082341r485092vb0cd10c43c4a2534@mail.gmail.com>
	<219716B3-C355-4128-A333-1459AC63C364@jerris.com>
	<86a32abc0910111506m4e6d2740tbd3ac5fe0f59178@mail.gmail.com>
	<5162D0B7-357C-4D0E-9913-46564A8C0B1D@freeswitch.org>
	<87f2f3b90910121035v5cfce2bel7d32da1671e325b6@mail.gmail.com>
Message-ID: <903da5680910121533wed3544ek6929ee9191a90848@mail.gmail.com>

On Mon, Oct 12, 2009 at 11:35 AM, Michael Collins <msc at freeswitch.org> wrote:
> It seemed appropriate to do so, therefore I added a small snippet on the
> documentation guidelines:
> http://wiki.freeswitch.org/wiki/Documentation_guidelines#Keep_It_Professional


Good guidelines...


Just in time to help the author of this lovely bit:
'I would love some brief descriptions of what you guys are doing with
FreeSWITCH so I can write an article to shove down the throats of
idiots who keep saying that "FreeSWITCH is an Asterisk alternative
that offers basic PBX functionality."'


Link:
http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_09


Best,
Gabe


From msc at freeswitch.org  Mon Oct 12 15:38:27 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 15:38:27 -0700
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <86a32abc0910121511m4241faefg4b8d1caf3b0f7a87@mail.gmail.com>
References: <86a32abc0910121511m4241faefg4b8d1caf3b0f7a87@mail.gmail.com>
Message-ID: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>

On Mon, Oct 12, 2009 at 3:11 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Hello,
>
> I have been doing some work recently on the FreeSWITCH wiki, to improve
> things.
>
> You can see some of my work here:
>
>
> http://wiki.freeswitch.org/index.php?title=Special:Contributions&limit=500&target=Diego.viola
>
> I am trying to polish the wiki and give it a more professional and clean
> look, this means that we need to enforce some guidelines and strive to make
> the wiki even better.
>
> How can we do this?
>
> Well, the first thing is to review everything we have, right now we have
> too many separated and little pages here and there that no one cares or
> read, we should avoid doing little or small pages. We need to correct typos,
> I suggest that people click on "Random page" link and start correcting
> typos, install an English (US) dictionary in your browser and enable Spell
> Checking and start correcting these typos if you want to help, also, make
> acronyms and initialisms all-uppercase when you make one.
>
> When you are adding documentation make sure you don't do a separate page
> for an existing page, etc. and we also need to define a size for titles,
> body, etc, so different pages don't look different from each other, I will
> make sure I add all this into our guidelines and that all pages follow this.
>
> My goal is to have great documentation, similar to the Apache
> documentation:
> http://httpd.apache.org/docs/2.2/
>
> Or even better, I hope you like this idea.
>

Diego,

Thank you so much for all of your help on this. Many in the community have
seen your work - it has not gone unnoticed and it most truly is appreciated.
I like the idea of improving the documentation. One thing we need to do is
re-think the organization. In many cases the issue with the docs isn't that
they aren't complete, but rather that they are hard to find. It's all about
organization. I'm definitely open to ideas.

For those who are interested in helping out with the wiki please let me know
of your availability and skill set. I am maintaining a list of volunteers.

Lastly, if you want to talk about documentation in real time please join us
in #freeswitch-docs.

Thanks,
MC
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From msc at freeswitch.org  Mon Oct 12 15:41:48 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 15:41:48 -0700
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <903da5680910121533wed3544ek6929ee9191a90848@mail.gmail.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<87f2f3b90910081538l6e4e4b7dv3aedb2c4d1af58bc@mail.gmail.com>
	<903da5680910082238x27bff476s8bea0e74ab516a8d@mail.gmail.com>
	<87f2f3b90910082308i7f6144d9kc887596be54dd609@mail.gmail.com>
	<903da5680910082341r485092vb0cd10c43c4a2534@mail.gmail.com>
	<219716B3-C355-4128-A333-1459AC63C364@jerris.com>
	<86a32abc0910111506m4e6d2740tbd3ac5fe0f59178@mail.gmail.com>
	<5162D0B7-357C-4D0E-9913-46564A8C0B1D@freeswitch.org>
	<87f2f3b90910121035v5cfce2bel7d32da1671e325b6@mail.gmail.com>
	<903da5680910121533wed3544ek6929ee9191a90848@mail.gmail.com>
Message-ID: <87f2f3b90910121541u180f10fdj18f97f0d8981b097@mail.gmail.com>

On Mon, Oct 12, 2009 at 3:33 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Mon, Oct 12, 2009 at 11:35 AM, Michael Collins <msc at freeswitch.org>
> wrote:
> > It seemed appropriate to do so, therefore I added a small snippet on the
> > documentation guidelines:
> >
> http://wiki.freeswitch.org/wiki/Documentation_guidelines#Keep_It_Professional
>
>
> Good guidelines...
>
>
> Just in time to help the author of this lovely bit:
> 'I would love some brief descriptions of what you guys are doing with
> FreeSWITCH so I can write an article to shove down the throats of
> idiots who keep saying that "FreeSWITCH is an Asterisk alternative
> that offers basic PBX functionality."'
>

Touch?, my good man, touch?. I'll fix that right away. :)
-MC
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From frank at carmickle.com  Mon Oct 12 16:02:47 2009
From: frank at carmickle.com (Frank Carmickle)
Date: Mon, 12 Oct 2009 19:02:47 -0400
Subject: [Freeswitch-users] torrents
Message-ID: <20091012230246.GF25698@base.carmickle.com>

Hello 
I would like to watch these files.  Can someone seed please.  Thank you.

--FC


From quentusrex at gmail.com  Mon Oct 12 16:15:44 2009
From: quentusrex at gmail.com (William King)
Date: Mon, 12 Oct 2009 16:15:44 -0700
Subject: [Freeswitch-users] torrents
In-Reply-To: <20091012230246.GF25698@base.carmickle.com>
References: <20091012230246.GF25698@base.carmickle.com>
Message-ID: <4AD3B8A0.5030501@gmail.com>

Opps. didn't realize it was closed. Here you go. Moving it to my seedbox
for permanent seeding.

-William King

Frank Carmickle wrote:
> Hello 
> I would like to watch these files.  Can someone seed please.  Thank you.
>
> --FC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From djbinter at yahoo.com  Mon Oct 12 16:45:27 2009
From: djbinter at yahoo.com (DJB)
Date: Mon, 12 Oct 2009 16:45:27 -0700 (PDT)
Subject: [Freeswitch-users] Problem with Luasql on x86_64
Message-ID: <626589.60602.qm@web37505.mail.mud.yahoo.com>

Hello,

I am trying to use luasql with freeswitch but having a hard time compile it with x86_64 Server running CentOS 5.3

Linux  2.6.18-128.el5 #1 SMP Wed Jan 21 10:41:14 EST 2009 x86_64 x86_64 x86_64 GNU/Linux

[root at fsx1 luasql-2.1.1]# make
export MACOSX_DEPLOYMENT_TARGET="10.3"; gcc -O2 -Wall -Wmissing-prototypes -Wmissing-declarations -ansi -pedantic -I../compat/src -I/opt/local/include -I/usr/local/include  -o src/sqlite3.so -shared  src/luasql.o src/ls_sqlite3.o  -L/opt/local/lib -lsqlite3
/usr/bin/ld: src/luasql.o: relocation R_X86_64_32 against `a local symbol' can not be used when making a shared object; recompile with -fPIC
src/luasql.o: could not read symbols: Bad value
collect2: ld returned 1 exit status
make: *** [src/sqlite3.so] Error 1


Any suggestions.

Thank you.


      


From stkn at freeswitch.org  Mon Oct 12 16:56:01 2009
From: stkn at freeswitch.org (Stefan Knoblich)
Date: Tue, 13 Oct 2009 01:56:01 +0200
Subject: [Freeswitch-users] Problem with Luasql on x86_64
In-Reply-To: <626589.60602.qm@web37505.mail.mud.yahoo.com>
References: <626589.60602.qm@web37505.mail.mud.yahoo.com>
Message-ID: <4AD3C211.5070305@freeswitch.org>

DJB wrote:
> Hello,
> 
> I am trying to use luasql with freeswitch but having a hard time compile it with x86_64 Server running CentOS 5.3
> 
> Linux  2.6.18-128.el5 #1 SMP Wed Jan 21 10:41:14 EST 2009 x86_64 x86_64 x86_64 GNU/Linux
> 
> [root at fsx1 luasql-2.1.1]# make
> export MACOSX_DEPLOYMENT_TARGET="10.3"; gcc -O2 -Wall -Wmissing-prototypes -Wmissing-declarations -ansi -pedantic -I../compat/src -I/opt/local/include -I/usr/local/include  -o src/sqlite3.so -shared  src/luasql.o src/ls_sqlite3.o  -L/opt/local/lib -lsqlite3
> /usr/bin/ld: src/luasql.o: relocation R_X86_64_32 against `a local symbol' can not be used when making a shared object; recompile with -fPIC
> src/luasql.o: could not read symbols: Bad value
> collect2: ld returned 1 exit status
> make: *** [src/sqlite3.so] Error 1
> 
> 
> Any suggestions.
> 
> Thank you.


Well, i'm sure people could've helped you on IRC if you stay for more than two minutes,
after asking your question...


From Mailings at kh-dev.de  Mon Oct 12 17:01:30 2009
From: Mailings at kh-dev.de (Klaus Hochlehnert)
Date: Tue, 13 Oct 2009 02:01:30 +0200
Subject: [Freeswitch-users] Problem with transfer and Snom phones...
Message-ID: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330251@srv02.khdev.corp>

Hi,

I'm facing an issue and I don't know why this happens or what I can do to solve this.

Here's the scenario:
- FS call timeout set to 30 sec
- Setting continue_on_fail=true and hangup_after_bridge=false
- 1 Snom phone and an external SIP/ISDN gateway (Lancom) connected to FS established a call and talk to each other (dir. from GW to Phone)
- The called phone transfers the call to another internal Snom using the transfer button on the phone
- The third phone doesn't answer and after 30 sec the call gets hung up with status NORMAL_CLEARING

Actually I want the call back to the initiator of the transfer if the 3rd phone doesn't answer.

Can anyone tell me why this happens and how I can set it up to do what I want?

Thanks, Klaus

P.S.: I'm using FS trunk from last week and newest Snom Firmware

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From gabe at gundy.org  Mon Oct 12 17:26:42 2009
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 12 Oct 2009 18:26:42 -0600
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <87f2f3b90910121541u180f10fdj18f97f0d8981b097@mail.gmail.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<903da5680910082238x27bff476s8bea0e74ab516a8d@mail.gmail.com>
	<87f2f3b90910082308i7f6144d9kc887596be54dd609@mail.gmail.com>
	<903da5680910082341r485092vb0cd10c43c4a2534@mail.gmail.com>
	<219716B3-C355-4128-A333-1459AC63C364@jerris.com>
	<86a32abc0910111506m4e6d2740tbd3ac5fe0f59178@mail.gmail.com>
	<5162D0B7-357C-4D0E-9913-46564A8C0B1D@freeswitch.org>
	<87f2f3b90910121035v5cfce2bel7d32da1671e325b6@mail.gmail.com>
	<903da5680910121533wed3544ek6929ee9191a90848@mail.gmail.com>
	<87f2f3b90910121541u180f10fdj18f97f0d8981b097@mail.gmail.com>
Message-ID: <903da5680910121726s17e4dd39u306c2255cd7479ac@mail.gmail.com>

On Mon, Oct 12, 2009 at 4:41 PM, Michael Collins <msc at freeswitch.org> wrote:
> On Mon, Oct 12, 2009 at 3:33 PM, Gabriel Gunderson <gabe at gundy.org> wrote:
>> On Mon, Oct 12, 2009 at 11:35 AM, Michael Collins <msc at freeswitch.org>
>> wrote:
>> > It seemed appropriate to do so, therefore I added a small snippet on the
>> > documentation guidelines:
>> >
>> > http://wiki.freeswitch.org/wiki/Documentation_guidelines#Keep_It_Professional
>>
>> Good guidelines...
>>
>> Just in time to help the author of this lovely bit:
>> 'I would love some brief descriptions of what you guys are doing with
>> FreeSWITCH so I can write an article to shove down the throats of
>> idiots who keep saying that "FreeSWITCH is an Asterisk alternative
>> that offers basic PBX functionality."'
>
> Touch?, my good man, touch?. I'll fix that right away. :)

Never let it be said that you're not a good sport.

:)

Gabe


From djbinter at yahoo.com  Mon Oct 12 17:28:13 2009
From: djbinter at yahoo.com (DJB)
Date: Mon, 12 Oct 2009 17:28:13 -0700 (PDT)
Subject: [Freeswitch-users] Problem with Luasql on x86_64
In-Reply-To: <626589.60602.qm@web37505.mail.mud.yahoo.com>
References: <626589.60602.qm@web37505.mail.mud.yahoo.com>
Message-ID: <536709.63053.qm@web37502.mail.mud.yahoo.com>

If anyone ran into the same problem, here is the fix that I got from [stkn] on IRC:

make CFLAGS="-O2 -fPIC" 

I hope it might help someone in the future.

Regards.



----- Original Message ----
From: DJB <djbinter at yahoo.com>
To: FREESWITCH-USERS MAILING LIST <freeswitch-users at lists.freeswitch.org>
Sent: Mon, October 12, 2009 4:45:27 PM
Subject: [Freeswitch-users] Problem with Luasql on x86_64

Hello,

I am trying to use luasql with freeswitch but having a hard time compile it with x86_64 Server running CentOS 5.3

Linux  2.6.18-128.el5 #1 SMP Wed Jan 21 10:41:14 EST 2009 x86_64 x86_64 x86_64 GNU/Linux

[root at fsx1 luasql-2.1.1]# make
export MACOSX_DEPLOYMENT_TARGET="10.3"; gcc -O2 -Wall -Wmissing-prototypes -Wmissing-declarations -ansi -pedantic -I../compat/src -I/opt/local/include -I/usr/local/include  -o src/sqlite3.so -shared  src/luasql.o src/ls_sqlite3.o  -L/opt/local/lib -lsqlite3
/usr/bin/ld: src/luasql.o: relocation R_X86_64_32 against `a local symbol' can not be used when making a shared object; recompile with -fPIC
src/luasql.o: could not read symbols: Bad value
collect2: ld returned 1 exit status
make: *** [src/sqlite3.so] Error 1


Any suggestions.

Thank you.


      

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



      


From mcampbellsmith at gmail.com  Mon Oct 12 19:31:14 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 13 Oct 2009 13:31:14 +1100
Subject: [Freeswitch-users] Question about fax tone detection
In-Reply-To: <87f2f3b90910121100l40a1b22hff33df032411cf31@mail.gmail.com>
References: <1255345270255-3807298.post@n2.nabble.com>
	<87f2f3b90910121100l40a1b22hff33df032411cf31@mail.gmail.com>
Message-ID: <33c87fa30910121931y48c57a94sb8fce42cc946c552@mail.gmail.com>

This is what I have in my dialplan and the fax is detected
beautifully.  Note that in my case, extension 1000 will ring for a
second or two before the fax is detected.  So in your example, the fax
does not have time to be detected, the dialplan exists and the call is
hungup.

When the fax is detected, the call is transferred to the receivefax
extension in context features.  The extension 1000 does not have to be
answered for the transfer to occur.

    <extension name="Local_1000s">
      <condition field="destination_number" expression="^(10[01][0-9])$">
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
        <action application="set" data="ringback=${au-ring}"/>
        <action application="tone_detect" data="fax 1100 r +5000
transfer 'receivefax XML features' 1" />
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="bridge" data="user/${dialed_extension}@${domain}"/>
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="voicemail" data="default ${domain_name}
${dialed_extension}"/>
      </condition>
   </extension>

and in the features context I have
    <extension name="receivefax">
      <condition field="destination_number" expression="^receivefax$">
        <action application="answer" />
        <action application="playback" data="silence_stream://2000"/>
        <action application="rxfax"
data="//usr//local//freeswitch//storage//${caller_id_number}-${uuid}.rxfax.tiff"/>
        <action application="system"
data="/usr/local/freeswitch/scripts/emailfax.sh
/usr/local/freeswitch/storage/${caller_id_number}-${uuid}.rxfax.tiff"/>
        <action application="hangup"/>
      </condition>
    </extension>
On Tue, Oct 13, 2009 at 5:00 AM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Mon, Oct 12, 2009 at 4:01 AM, homqua <ngay01042005 at gmail.com> wrote:
>>
>> Hi,
>> I have implemented the solution for tone detection in wiki, and also
>> answer
>> the channel before detecting the tone:
>> <condition field="destination_number" expression="^(055138419992)$">
>> ? ?<action application="answer"/>
>> ? ?<action application="tone_detect" data="fax 1100 r +15000 transfer fax
>> XML default"/>
>>
>>
>> <extension name="fax">
>> ? ? ?<condition field="destination_number" expression="^fax$">
>> ? ? ? ?<action application="answer" />
>> ? ? ? ?<action application="sleep" data="1000"/>
>> ? ? ? ?<action application="rxfax"
>>
>> data="/usr/local/freeswitch/storage/fax/${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}.tiff"/>
>> ? ?<action application="set" data="fax_mode=recv"/>
>> ? ? ? ?<action application="hangup"/>
>> ? ? ?</condition>
>> ? ?</extension>
>>
>>
>> But FS cannot recognize the tone, and therefore cannot move to fax
>> extension. ?Below are the error in FS:
>>
>> 2009-10-12 10:57:16.702287 [NOTICE] switch_channel.c:602 New Channel
>> sofia/external/anonymous at anonymous.invalid
>> [c431f0a3-9231-4724-ba39-9e4ef7edfca2]
>> 2009-10-12 10:57:16.703413 [INFO] mod_dialplan_xml.c:315 Processing
>> Anonymous->055138419992 in context public
>> 2009-10-12 10:57:16.719288 [NOTICE] switch_ivr.c:1349 Transfer
>> sofia/external/anonymous at anonymous.invalid to XML[055138419992 at default]
>> 2009-10-12 10:57:16.719288 [INFO] mod_dialplan_xml.c:315 Processing
>> Anonymous->055138419992 in context default
>> 2009-10-12 10:57:16.722289 [NOTICE] mod_dptools.c:649 Channel
>> [sofia/external/anonymous at anonymous.invalid] has been answered
>> 2009-10-12 10:57:16.722289 [NOTICE] mod_dptools.c:1324 Enabling tone
>> detection 'fax' '1100'
>> 2009-10-12 10:57:16.723302 [NOTICE] switch_core_state_machine.c:179 Hangup
>> sofia/external/anonymous at anonymous.invalid [CS_EXECUTE] [NORMAL_CLEARING]
>> 2009-10-12 10:57:16.740285 [NOTICE] switch_core_session.c:1086 Session 1
>> (sofia/external/anonymous at anonymous.invalid) Ended
>> 2009-10-12 10:57:16.740285 [NOTICE] switch_core_session.c:1088 Close
>> Channel
>> sofia/external/anonymous at anonymous.invalid [CS_DESTROY]
>>
>> And the trace for SIP messages: ?http://pastebin.com/m4e47e7d9
>>
>> If anyone has any idea, tell me please.
>> Thanks.
>
> I think the trouble here is that you don't have anything else in the
> dialplan after the tone_detect. The tone_detect app is non-block, which
> means that it doesn't sit there and wait for a tone. If you want the
> dialplan to sit and wait then do a sleep app after your tone_detect. The
> other question I would have is this: what happens if the incoming call is
> not a fax? What do you want to do then?
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From diego.viola at gmail.com  Mon Oct 12 19:37:45 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 13 Oct 2009 02:37:45 +0000
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
References: <86a32abc0910121511m4241faefg4b8d1caf3b0f7a87@mail.gmail.com>
	<87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
Message-ID: <86a32abc0910121937l5fe537fcu48089d1c5e1bfc1@mail.gmail.com>

Sure, I'm happy to put my little two cents to help the project :).

Diego

On Mon, Oct 12, 2009 at 10:38 PM, Michael Collins <msc at freeswitch.org>wrote:

>
> On Mon, Oct 12, 2009 at 3:11 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> Hello,
>>
>> I have been doing some work recently on the FreeSWITCH wiki, to improve
>> things.
>>
>> You can see some of my work here:
>>
>>
>> http://wiki.freeswitch.org/index.php?title=Special:Contributions&limit=500&target=Diego.viola
>>
>> I am trying to polish the wiki and give it a more professional and clean
>> look, this means that we need to enforce some guidelines and strive to make
>> the wiki even better.
>>
>> How can we do this?
>>
>> Well, the first thing is to review everything we have, right now we have
>> too many separated and little pages here and there that no one cares or
>> read, we should avoid doing little or small pages. We need to correct typos,
>> I suggest that people click on "Random page" link and start correcting
>> typos, install an English (US) dictionary in your browser and enable Spell
>> Checking and start correcting these typos if you want to help, also, make
>> acronyms and initialisms all-uppercase when you make one.
>>
>> When you are adding documentation make sure you don't do a separate page
>> for an existing page, etc. and we also need to define a size for titles,
>> body, etc, so different pages don't look different from each other, I will
>> make sure I add all this into our guidelines and that all pages follow this.
>>
>> My goal is to have great documentation, similar to the Apache
>> documentation:
>> http://httpd.apache.org/docs/2.2/
>>
>> Or even better, I hope you like this idea.
>>
>
> Diego,
>
> Thank you so much for all of your help on this. Many in the community have
> seen your work - it has not gone unnoticed and it most truly is appreciated.
> I like the idea of improving the documentation. One thing we need to do is
> re-think the organization. In many cases the issue with the docs isn't that
> they aren't complete, but rather that they are hard to find. It's all about
> organization. I'm definitely open to ideas.
>
> For those who are interested in helping out with the wiki please let me
> know of your availability and skill set. I am maintaining a list of
> volunteers.
>
> Lastly, if you want to talk about documentation in real time please join us
> in #freeswitch-docs.
>
> Thanks,
> MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Mon Oct 12 20:16:38 2009
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 13 Oct 2009 11:16:38 +0800
Subject: [Freeswitch-users] Re corded file as voicemail.
In-Reply-To: <25850181.post@talk.nabble.com>
References: <25850181.post@talk.nabble.com>
Message-ID: <23f91030910122016q33012754n791ab9629e59c7a0@mail.gmail.com>

It was a problem and has been fixed in the last trunk. Just update to the
latest code should be ok.

btw, the developers using jira to track bugs, so feel free to report one (as
you see http://jira.freeswitch.org/browse/FSCORE-463) if you think it's a
bug next time.

2009/10/12 Nagalenoj <nagalenoj at gmail.com>

>
> Whats the conclusion.?!
>
>
> Brian West-3 wrote:
> >
> > Please open a jira please this did work but a recent change in
> > switch_core_codec caused this to appear I usually test this regularly
> > but haven't run thru a full run of tests lately.
> > /b
> >
>
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From dujinfang at gmail.com  Mon Oct 12 20:17:37 2009
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 13 Oct 2009 11:17:37 +0800
Subject: [Freeswitch-users] Re corded file as voicemail.
In-Reply-To: <BA9A12F7-EB74-485D-8612-12E448621257@freeswitch.org>
References: <25850181.post@talk.nabble.com>
	<BEDBBF51-2D48-4CDE-A467-11D5BBB458E1@freeswitch.org>
	<23f91030910112315h5ea79b81ue55381a3caff0173@mail.gmail.com>
	<BA9A12F7-EB74-485D-8612-12E448621257@freeswitch.org>
Message-ID: <23f91030910122017j6efdfa26l958704ce8d67e53e@mail.gmail.com>

I won't try until I need that, but I believe it works. Thanks Brian.

2009/10/13 Brian West <brian at freeswitch.org>

> Fixed... svn up.
> /b
>
> On Oct 12, 2009, at 1:15 AM, Seven Du wrote:
>
> http://jira.freeswitch.org/browse/MODCODEC-15
>
> Is it ok I assigned to you ?
>
> Thanks.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From srinivas.ksvreddy at gmail.com  Mon Oct 12 21:44:01 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 13 Oct 2009 10:14:01 +0530
Subject: [Freeswitch-users] mod_sofia.c registered calls how to know
In-Reply-To: <1255347432.3724.4.camel@eli-desktop>
References: <f8af5740910070748g1efee71ft5841b6fa4d2412c8@mail.gmail.com>
	<14A1216A-61C3-4C03-B710-57D2E7F4CD7A@jerris.com>
	<f8af5740910112114k3206c3cbq5757e2ff7de81162@mail.gmail.com>
	<23f91030910120419o74f53279v5f061988ff585d56@mail.gmail.com>
	<1255347432.3724.4.camel@eli-desktop>
Message-ID: <f8af5740910122144g6a46eabbxd498ba2a6044a0f7@mail.gmail.com>

exactly that is what i am trying to do now, if any help that would be great.

On Mon, Oct 12, 2009 at 5:07 PM, Eli Hayun <elihay at savion.huji.ac.il> wrote:

>  Is it possible to keep a list of registered phone, and when FS will start
> it will register them all automatically?
>
>
>
> On Mon, 2009-10-12 at 13:19 +0200, Seven Du wrote:
>
> try open YOUR_FreeSWITCH_INSTALL_DIR/db/*.db, you need sqlite3 to open
> them. not sure how to do that on windows, but on linux:
>
>
> # sqlite3 xx.db
> sqlite> select * from sip_registration;
>
>  2009/10/12 srinivasula reddy <srinivas.ksvreddy at gmail.com>
>
> Hi Mike,
>
> Thanks for your valuable reply,
> when i install freeswitch1.0.2 in my machine(Windows xp operation system) i
> dont have any databasae installed in my system, then from sqllite will come
> into picture, and how can i see the registered users data from sqllite.
>
>
> Thanks
> Srinivas
>
>
>   On Mon, Oct 12, 2009 at 3:04 AM, Michael Jerris <mike at jerris.com> wrote:
>
>
>
>    On Oct 7, 2009, at 10:48 AM, srinivasula reddy wrote:
>
>
>
> Hi
>
> can any please tell me where registered calls are stored, so when incoming
> call came to mod_sofia.c how it will check it is registered or not?\\
>
>
>
>    Calls are not registered and calls have nothing to do with
> registration.  Users are registered so that you may send calls to them.
>  Registration data is stored either in a sqlite database, or optionally if
> you setup odbc, in another database of your choice.  If you try to send a
> call to an unregistered user in the dialplan using the proper syntax to send
> calls to registered users (see the wiki for more details), and that user is
> not registered, the bridge app will fail, optionally letting you continue on
> in the dialplan based on variables such as continue_on_fail and
> hangup_after_bridge.  You can use the sofia_contact function to see if there
> is anyone registered to a specific user.
>
>
>
>    Mike
>
>
>
>
>
>    _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
> --
> Srinivasula Reddy K
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Srinivasula Reddy K
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From msc at freeswitch.org  Mon Oct 12 21:51:48 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 21:51:48 -0700
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <903da5680910121726s17e4dd39u306c2255cd7479ac@mail.gmail.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<87f2f3b90910082308i7f6144d9kc887596be54dd609@mail.gmail.com>
	<903da5680910082341r485092vb0cd10c43c4a2534@mail.gmail.com>
	<219716B3-C355-4128-A333-1459AC63C364@jerris.com>
	<86a32abc0910111506m4e6d2740tbd3ac5fe0f59178@mail.gmail.com>
	<5162D0B7-357C-4D0E-9913-46564A8C0B1D@freeswitch.org>
	<87f2f3b90910121035v5cfce2bel7d32da1671e325b6@mail.gmail.com>
	<903da5680910121533wed3544ek6929ee9191a90848@mail.gmail.com>
	<87f2f3b90910121541u180f10fdj18f97f0d8981b097@mail.gmail.com>
	<903da5680910121726s17e4dd39u306c2255cd7479ac@mail.gmail.com>
Message-ID: <87f2f3b90910122151s324d5dbfpa62901ac653b5c84@mail.gmail.com>

On Mon, Oct 12, 2009 at 5:26 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Mon, Oct 12, 2009 at 4:41 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> > On Mon, Oct 12, 2009 at 3:33 PM, Gabriel Gunderson <gabe at gundy.org>
> wrote:
> >> On Mon, Oct 12, 2009 at 11:35 AM, Michael Collins <msc at freeswitch.org>
> >> wrote:
> >> > It seemed appropriate to do so, therefore I added a small snippet on
> the
> >> > documentation guidelines:
> >> >
> >> >
> http://wiki.freeswitch.org/wiki/Documentation_guidelines#Keep_It_Professional
> >>
> >> Good guidelines...
> >>
> >> Just in time to help the author of this lovely bit:
> >> 'I would love some brief descriptions of what you guys are doing with
> >> FreeSWITCH so I can write an article to shove down the throats of
> >> idiots who keep saying that "FreeSWITCH is an Asterisk alternative
> >> that offers basic PBX functionality."'
> >
> > Touch?, my good man, touch?. I'll fix that right away. :)
>
> Never let it be said that you're not a good sport.
>
> :)
>
> Gabe
>
> Well, I'm occasionally a good sport. ;)
-MC
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From mctch at yahoo.com  Mon Oct 12 22:52:03 2009
From: mctch at yahoo.com (Mark Crane)
Date: Mon, 12 Oct 2009 22:52:03 -0700 (PDT)
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
Message-ID: <891080.87141.qm@web56406.mail.re3.yahoo.com>

"In many cases the issue with the docs isn't that they aren't complete but rather that they are hard to find."

Agreed! The biggest problem with the wiki is that it is hard to find things. 

How do books solve this problem they use an index. It is quite surprising that the wiki software doesn't come with ability to do this basic task automatically and instead relies solely upon a manually created index. It would be extremely beneficial to have a index of all pages and topics that are available on the wiki. The average person may only see the main links in the documentation and not realize there are actually hundreds of pages.

There used to be a pdf that was auto generated from the wiki and it gave a much easier view of all the pages on the wiki.

Mark J Crane



--- On Mon, 10/12/09, Michael Collins <msc at freeswitch.org> wrote:

From: Michael Collins <msc at freeswitch.org>
Subject: Re: [Freeswitch-users] Some documentation thoughts
To: freeswitch-users at lists.freeswitch.org
Date: Monday, October 12, 2009, 4:38 PM


On Mon, Oct 12, 2009 at 3:11 PM, Diego Viola <diego.viola at gmail.com> wrote:

Hello,

I have been doing some work recently on the FreeSWITCH wiki, to improve things.

You can see some of my work here:

http://wiki.freeswitch.org/index.php?title=Special:Contributions&limit=500&target=Diego.viola



I am trying to polish the wiki and give it a more professional and clean look, this means that we need to enforce some guidelines and strive to make the wiki even better.

How can we do this?

Well, the first thing is to review everything we have, right now we have too many separated and little pages here and there that no one cares or read, we should avoid doing little or small pages. We need to correct typos, I suggest that people click on "Random page" link and start correcting typos, install an English (US) dictionary in your browser and enable Spell Checking and start correcting these typos if you want to help, also, make acronyms and initialisms all-uppercase when you make one.



When you are adding documentation make sure you don't do a separate page for an existing page, etc. and we also need to define a size for titles, body, etc, so different pages don't look different from each other, I will make sure I add all this into our guidelines and that all pages follow this.



My goal is to have great documentation, similar to the Apache documentation:
http://httpd.apache.org/docs/2.2/

Or even better, I hope you like this idea.


Diego,

Thank you so much for all of your help on this. Many in the community have seen your work - it has not gone unnoticed and it most truly is appreciated. I like the idea of improving the documentation. One thing we need to do is re-think the organization. In many cases the issue with the docs isn't that they aren't complete, but rather that they are hard to find. It's all about organization. I'm definitely open to ideas.


For those who are interested in helping out with the wiki please let me know of your availability and skill set. I am maintaining a list of volunteers.

Lastly, if you want to talk about documentation in real time please join us in #freeswitch-docs.


Thanks,
MC



-----Inline Attachment Follows-----

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



      
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From velu.technical at gmail.com  Mon Oct 12 22:59:49 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Tue, 13 Oct 2009 11:29:49 +0530
Subject: [Freeswitch-users] Asynchronous execution in ESL.pm
Message-ID: <1452e2980910122259w3e007870m4bda8ce673d584b2@mail.gmail.com>

Dear All,
         I have set my socket mode as full. I have used ESL.pm to develop an
IVR. I encounter the situation that I need play some music file while
executing some external application. So, I used executeAsync method  to play
the music file. But the executeAsync application didn't work.

       What is the problem? Please help me....

Thanks,
Velusamy.
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From brian at freeswitch.org  Mon Oct 12 23:00:11 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Oct 2009 23:00:11 -0700
Subject: [Freeswitch-users] Problem with transfer and Snom phones...
In-Reply-To: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330251@srv02.khdev.corp>
References: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330251@srv02.khdev.corp>
Message-ID: <BD3BCAC6-5CA0-4DBD-AF75-5DC5B106F3C1@freeswitch.org>

Update to latest trunk and try again... then if it persists then  
collect all the sip traces and debug logs as per the wiki on reporting  
bugs.

Thanks,
Brian

On Oct 12, 2009, at 5:01 PM, Klaus Hochlehnert wrote:

> P.S.: I?m using FS trunk from last week and newest Snom Firmware

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From srinivas.ksvreddy at gmail.com  Mon Oct 12 23:01:55 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 13 Oct 2009 11:31:55 +0530
Subject: [Freeswitch-users] Groups information in sqllite
Message-ID: <f8af5740910122301u338ca128macfc0b769ea0e961@mail.gmail.com>

Hi,

can any know where group information is exactly stored in sqllite database,
i have seen sip_registration here i can find the registered users,
in the same way how i can i find the group information, and which user
belongs to which user?
any help would be great.

thanks
-- 
Srinivasula Reddy K
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From brian at freeswitch.org  Mon Oct 12 23:02:14 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Oct 2009 23:02:14 -0700
Subject: [Freeswitch-users] mod_sofia.c registered calls how to know
In-Reply-To: <1255347432.3724.4.camel@eli-desktop>
References: <f8af5740910070748g1efee71ft5841b6fa4d2412c8@mail.gmail.com>
	<14A1216A-61C3-4C03-B710-57D2E7F4CD7A@jerris.com>
	<f8af5740910112114k3206c3cbq5757e2ff7de81162@mail.gmail.com>
	<23f91030910120419o74f53279v5f061988ff585d56@mail.gmail.com>
	<1255347432.3724.4.camel@eli-desktop>
Message-ID: <FE9BBD84-64D4-4219-AA90-A6A5FF842446@freeswitch.org>

Eli,
	Well FreeSWITCH already keeps a list... but its the phone's job to  
register to FreeSWITCH not the other way around.  Their are various  
ways to accomplish your goals but not sure how well each will work.   
Check out the wiki for sip-force-contact.

/b

On Oct 12, 2009, at 4:37 AM, Eli Hayun wrote:

> Is it possible to keep a list of registered phone, and when FS will  
> start it will register them all automatically?
>



From srinivas.ksvreddy at gmail.com  Mon Oct 12 23:02:28 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 13 Oct 2009 11:32:28 +0530
Subject: [Freeswitch-users] Fwd: Groups information in sqllite
In-Reply-To: <f8af5740910122301u338ca128macfc0b769ea0e961@mail.gmail.com>
References: <f8af5740910122301u338ca128macfc0b769ea0e961@mail.gmail.com>
Message-ID: <f8af5740910122302i47ccde5bo9df2bae965408394@mail.gmail.com>

Hi,

can any know where group information is exactly stored in sqllite database,
i have seen sip_registration here i can find the registered users,
in the same way how i can i find the group information, and which user
belongs to which user?
any help would be great.

thanks
-- 
Srinivasula Reddy K
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From brian at freeswitch.org  Mon Oct 12 23:04:35 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Oct 2009 23:04:35 -0700
Subject: [Freeswitch-users] Asynchronous execution in ESL.pm
In-Reply-To: <1452e2980910122259w3e007870m4bda8ce673d584b2@mail.gmail.com>
References: <1452e2980910122259w3e007870m4bda8ce673d584b2@mail.gmail.com>
Message-ID: <994D92AC-0760-453C-AD7E-4D6B706347AE@freeswitch.org>

You need "async full"

/b

On Oct 12, 2009, at 10:59 PM, velusamy velu wrote:

> What is the problem? Please help me....



From brian at freeswitch.org  Mon Oct 12 23:05:26 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Oct 2009 23:05:26 -0700
Subject: [Freeswitch-users] Groups information in sqllite
In-Reply-To: <f8af5740910122301u338ca128macfc0b769ea0e961@mail.gmail.com>
References: <f8af5740910122301u338ca128macfc0b769ea0e961@mail.gmail.com>
Message-ID: <FD1AC150-0636-4861-8341-58553D3F43A0@freeswitch.org>

The sip_registration table contains the contacts for each registered  
endpoint.  Its not for the directory from a database per se... If you  
wish to serve up your users and groups from a database check out the  
XML Curl wiki page.

/b

On Oct 12, 2009, at 11:01 PM, srinivasula reddy wrote:

> Hi,
>
> can any know where group information is exactly stored in sqllite  
> database, i have seen sip_registration here i can find the  
> registered users,
> in the same way how i can i find the group information, and which  
> user belongs to which user?
> any help would be great.
>
> thanks
> -- 
> Srinivasula Reddy K



From brian at freeswitch.org  Mon Oct 12 23:05:47 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Oct 2009 23:05:47 -0700
Subject: [Freeswitch-users] Fwd: Groups information in sqllite
In-Reply-To: <f8af5740910122302i47ccde5bo9df2bae965408394@mail.gmail.com>
References: <f8af5740910122301u338ca128macfc0b769ea0e961@mail.gmail.com>
	<f8af5740910122302i47ccde5bo9df2bae965408394@mail.gmail.com>
Message-ID: <3AEBF9B1-E366-44FA-8EE0-1265CED486ED@freeswitch.org>

Please DO NOT cross post.

/b

On Oct 12, 2009, at 11:02 PM, srinivasula reddy wrote:

>
>
>
>
> Hi,
>
> can any know where group information is exactly stored in sqllite  
> database, i have seen sip_registration here i can find the  
> registered users,
> in the same way how i can i find the group information, and which  
> user belongs to which user?
> any help would be great.
>
> thanks
> -- 
> Srinivasula Reddy K
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Mon Oct 12 23:08:13 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Oct 2009 23:08:13 -0700
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
Message-ID: <FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>

Does anyone see a problem with hosting mod_h323 in our SVN?  I would  
like to centralize everything we can to reuse our issue tracking  
resources and not fragment the community if possible.

/b

On Oct 12, 2009, at 2:43 PM, Tihomir Culjaga wrote:

> hi,
>
> finally i compiled it right ... had a stupid issue with ekiga and  
> wrong ptlib in place...
>
> anyhow, i loaded the module and will continue the tests  
> tomorrow ...first thing i arrive in my office :P
>
>



From brian at freeswitch.org  Mon Oct 12 23:09:16 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Oct 2009 23:09:16 -0700
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Update: Valet
	Parking
In-Reply-To: <4AD389A8.1080603@gmail.com>
References: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>	<15b9404e0910102018oda8834ah8cc87787ee00c22b@mail.gmail.com>
	<87f2f3b90910121201o422c7891vd30ee3761aef0eb7@mail.gmail.com>
	<4AD389A8.1080603@gmail.com>
Message-ID: <719ED4DA-A7C6-4256-8349-09C874505930@freeswitch.org>

Thats called mod_fifo.

/b

On Oct 12, 2009, at 12:55 PM, William King wrote:

> I don't know if this was mentioned yet. It would be useful to have a  
> way
> to have the parking lot automatically find the next available spot and
> tts it to the person parking the call.
>
> Then the auto unpark would pop off the lowest numbered lot, or return
> fail if there is nobody in the parking lots etc.
>
> -William King



From srinivas.ksvreddy at gmail.com  Mon Oct 12 23:16:55 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 13 Oct 2009 11:46:55 +0530
Subject: [Freeswitch-users] Groups information in sqllite
In-Reply-To: <FD1AC150-0636-4861-8341-58553D3F43A0@freeswitch.org>
References: <f8af5740910122301u338ca128macfc0b769ea0e961@mail.gmail.com>
	<FD1AC150-0636-4861-8341-58553D3F43A0@freeswitch.org>
Message-ID: <f8af5740910122316g7db0ae7ci1f79e3aede07d112@mail.gmail.com>

HI Brain,

thank u very much for valuable reply,

you are right, sip_Registration table contains the registered endpoint
details, in the same way there is any table for groups, how many groups are
there? and information about groups(directory/default.xml this file having
the group configuration).

Thanks
Srinivas

On Tue, Oct 13, 2009 at 11:35 AM, Brian West <brian at freeswitch.org> wrote:

> The sip_registration table contains the contacts for each registered
> endpoint.  Its not for the directory from a database per se... If you
> wish to serve up your users and groups from a database check out the
> XML Curl wiki page.
>
> /b
>
> On Oct 12, 2009, at 11:01 PM, srinivasula reddy wrote:
>
> > Hi,
> >
> > can any know where group information is exactly stored in sqllite
> > database, i have seen sip_registration here i can find the
> > registered users,
> > in the same way how i can i find the group information, and which
> > user belongs to which user?
> > any help would be great.
> >
> > thanks
> > --
> > Srinivasula Reddy K
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Srinivasula Reddy K
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From brian at freeswitch.org  Mon Oct 12 23:25:05 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Oct 2009 23:25:05 -0700
Subject: [Freeswitch-users] Groups information in sqllite
In-Reply-To: <f8af5740910122316g7db0ae7ci1f79e3aede07d112@mail.gmail.com>
References: <f8af5740910122301u338ca128macfc0b769ea0e961@mail.gmail.com>
	<FD1AC150-0636-4861-8341-58553D3F43A0@freeswitch.org>
	<f8af5740910122316g7db0ae7ci1f79e3aede07d112@mail.gmail.com>
Message-ID: <25EFC4E3-88E6-4286-8432-8118CC9E899C@freeswitch.org>

NO this is in the XML... not in the db table.

/b

On Oct 12, 2009, at 11:16 PM, srinivasula reddy wrote:

>  in the same way there is any table for groups, how many groups are   
> there? and information about groups(directory/default.xml this file  
> having the group configuration).



From tculjaga at gmail.com  Mon Oct 12 23:27:39 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 13 Oct 2009 08:27:39 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>
Message-ID: <65d96fc80910122327u54505b44r4ff53e89293bcf9a@mail.gmail.com>

this will be perfect ... but it is up to Yuriy if he is willing to donate
his work...

T.


On Tue, Oct 13, 2009 at 8:08 AM, Brian West <brian at freeswitch.org> wrote:

> Does anyone see a problem with hosting mod_h323 in our SVN?  I would
> like to centralize everything we can to reuse our issue tracking
> resources and not fragment the community if possible.
>
> /b
>
> On Oct 12, 2009, at 2:43 PM, Tihomir Culjaga wrote:
>
> > hi,
> >
> > finally i compiled it right ... had a stupid issue with ekiga and
> > wrong ptlib in place...
> >
> > anyhow, i loaded the module and will continue the tests
> > tomorrow ...first thing i arrive in my office :P
> >
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Mon Oct 12 23:31:39 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Oct 2009 23:31:39 -0700
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910122327u54505b44r4ff53e89293bcf9a@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>
	<65d96fc80910122327u54505b44r4ff53e89293bcf9a@mail.gmail.com>
Message-ID: <5F65174C-2CE6-4007-B288-C81227776281@freeswitch.org>

I wouldn't call it donating per se... Its just giving it a place to  
live with easy access for end users without having to do anything  
extra go get it!  ;)

/b

On Oct 12, 2009, at 11:27 PM, Tihomir Culjaga wrote:

> this will be perfect ... but it is up to Yuriy if he is willing to  
> donate his work...
>
> T.



From srinivas.ksvreddy at gmail.com  Mon Oct 12 23:32:19 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 13 Oct 2009 12:02:19 +0530
Subject: [Freeswitch-users] Groups information in sqllite
In-Reply-To: <25EFC4E3-88E6-4286-8432-8118CC9E899C@freeswitch.org>
References: <f8af5740910122301u338ca128macfc0b769ea0e961@mail.gmail.com>
	<FD1AC150-0636-4861-8341-58553D3F43A0@freeswitch.org>
	<f8af5740910122316g7db0ae7ci1f79e3aede07d112@mail.gmail.com>
	<25EFC4E3-88E6-4286-8432-8118CC9E899C@freeswitch.org>
Message-ID: <f8af5740910122332i6c266250q7a550fb05439ae53@mail.gmail.com>

OK, then when i call to group number(911) how it will call to all the
registered members in group?


On Tue, Oct 13, 2009 at 11:55 AM, Brian West <brian at freeswitch.org> wrote:

> NO this is in the XML... not in the db table.
>
> /b
>
> On Oct 12, 2009, at 11:16 PM, srinivasula reddy wrote:
>
> >  in the same way there is any table for groups, how many groups are
> > there? and information about groups(directory/default.xml this file
> > having the group configuration).
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Srinivasula Reddy K
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From brian at freeswitch.org  Mon Oct 12 23:48:01 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Oct 2009 23:48:01 -0700
Subject: [Freeswitch-users] Groups information in sqllite
In-Reply-To: <f8af5740910122332i6c266250q7a550fb05439ae53@mail.gmail.com>
References: <f8af5740910122301u338ca128macfc0b769ea0e961@mail.gmail.com>
	<FD1AC150-0636-4861-8341-58553D3F43A0@freeswitch.org>
	<f8af5740910122316g7db0ae7ci1f79e3aede07d112@mail.gmail.com>
	<25EFC4E3-88E6-4286-8432-8118CC9E899C@freeswitch.org>
	<f8af5740910122332i6c266250q7a550fb05439ae53@mail.gmail.com>
Message-ID: <5FBD4715-DB4A-481A-AAAF-09D7541EC3F4@freeswitch.org>

Its based on the directory and who is in the group... check out the  
defaults it does exactly this on the 2000 range if I recall correctly.

/b

On Oct 12, 2009, at 11:32 PM, srinivasula reddy wrote:

> OK, then when i call to group number(911) how it will call to all  
> the registered members in group?



From nandy1925 at gmail.com  Tue Oct 13 00:00:43 2009
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Tue, 13 Oct 2009 15:00:43 +0800
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Update: Valet
	Parking
In-Reply-To: <4AD389A8.1080603@gmail.com>
References: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>
	<15b9404e0910102018oda8834ah8cc87787ee00c22b@mail.gmail.com>
	<87f2f3b90910121201o422c7891vd30ee3761aef0eb7@mail.gmail.com>
	<4AD389A8.1080603@gmail.com>
Message-ID: <7d0bfd8c0910130000i25012d75ldc7f873214d30e12@mail.gmail.com>

i come across the valet_park application when i just finished an
improved-version of the call parking (using mod_fifo)  such that it is
parked to different ext'n numbers when the caller is att_xfer'd to ext 777.
i used  strftime(%s) to generate 700~759 parking numbers. i also added
feature that if the parked call is not picked up after xx secs timeout, it's
returned back to the transferer. the only glitch here is - when another
parking is done exactly 1 minute later (unless we'll limit the timeout to 59
secs). i hv to use att_xfer so that the transferer can hear the parking
number.

i'll submit the dialplan if it's worth seeing it.

On Tue, Oct 13, 2009 at 3:55 AM, William King <quentusrex at gmail.com> wrote:

> I don't know if this was mentioned yet. It would be useful to have a way
> to have the parking lot automatically find the next available spot and
> tts it to the person parking the call.
>
> Then the auto unpark would pop off the lowest numbered lot, or return
> fail if there is nobody in the parking lots etc.
>
> -William King
>
> Michael Collins wrote:
> >
> >
> > On Sat, Oct 10, 2009 at 8:18 PM, mayamatakeshi
> > <mayamatakeshi at gmail.com <mailto:mayamatakeshi at gmail.com>> wrote:
> >
> >
> >
> >     On Fri, Oct 9, 2009 at 10:42 AM, Michael Collins
> >     <msc at freeswitch.org <mailto:msc at freeswitch.org>> wrote:
> >
> >         FYI,
> >
> >         The FreeSWITCH devs have added valet parking! Check it out:
> >         http://www.freeswitch.org/node/207
> >
> >         Let us know what you think.
> >
> >
> >     Very nice.
> >
> >     But I think a valet_unpark app is missing.
> >     If the intention of the person sent to the valet lot is to
> >     retrieve a call there, the person can assume the call was already
> >     retrieved by someone else or that the caller hung up if he/she
> >     hears MOH. But it would be nicer to have a valet_unpark app to
> >     fail and let the dialplan play a message.
> >
> > I understand what you are saying. I'm not sure I agree, but we'll kick
> > the idea around when we have a few minutes and let you know what we
> > decide.
> > -MC
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dujinfang at gmail.com  Tue Oct 13 00:05:24 2009
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 13 Oct 2009 15:05:24 +0800
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>
Message-ID: <23f91030910130005m1dc40f90k8f5b047baf1df144@mail.gmail.com>

that will make life easier.

2009/10/13 Brian West <brian at freeswitch.org>

> Does anyone see a problem with hosting mod_h323 in our SVN?  I would
> like to centralize everything we can to reuse our issue tracking
> resources and not fragment the community if possible.
>
> /b
>
> On Oct 12, 2009, at 2:43 PM, Tihomir Culjaga wrote:
>
> > hi,
> >
> > finally i compiled it right ... had a stupid issue with ekiga and
> > wrong ptlib in place...
> >
> > anyhow, i loaded the module and will continue the tests
> > tomorrow ...first thing i arrive in my office :P
> >
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tculjaga at gmail.com  Tue Oct 13 00:17:54 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 13 Oct 2009 09:17:54 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <5F65174C-2CE6-4007-B288-C81227776281@freeswitch.org>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>
	<65d96fc80910122327u54505b44r4ff53e89293bcf9a@mail.gmail.com>
	<5F65174C-2CE6-4007-B288-C81227776281@freeswitch.org>
Message-ID: <65d96fc80910130017m5cafef68m89efa99493b0b16d@mail.gmail.com>

On Tue, Oct 13, 2009 at 8:31 AM, Brian West <brian at freeswitch.org> wrote:

> I wouldn't call it donating per se... Its just giving it a place to
> live with easy access for end users without having to do anything
> extra go get it!  ;)
>
> /b
>
>
I agree with you Brian.


>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tayeb.meftah at gmail.com  Tue Oct 13 01:33:49 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Tue, 13 Oct 2009 08:33:49 +0000
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>	<881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>
Message-ID: <4AD43B6D.7000204@gmail.com>

hello,
yes, host please to let users test it and report bug
YATE/Asterisk fully support but freeswitch no fully support it
Brian West a ?crit :
> Does anyone see a problem with hosting mod_h323 in our SVN?  I would  
> like to centralize everything we can to reuse our issue tracking  
> resources and not fragment the community if possible.
>
> /b
>
> On Oct 12, 2009, at 2:43 PM, Tihomir Culjaga wrote:
>
>   
>> hi,
>>
>> finally i compiled it right ... had a stupid issue with ekiga and  
>> wrong ptlib in place...
>>
>> anyhow, i loaded the module and will continue the tests  
>> tomorrow ...first thing i arrive in my office :P
>>
>>
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4501 (20091012) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
>   




__________ Information from ESET NOD32 Antivirus, version of virus signature database 4501 (20091012) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From srinivas.ksvreddy at gmail.com  Tue Oct 13 01:30:24 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 13 Oct 2009 14:00:24 +0530
Subject: [Freeswitch-users] Groups information in sqllite
In-Reply-To: <5FBD4715-DB4A-481A-AAAF-09D7541EC3F4@freeswitch.org>
References: <f8af5740910122301u338ca128macfc0b769ea0e961@mail.gmail.com>
	<FD1AC150-0636-4861-8341-58553D3F43A0@freeswitch.org>
	<f8af5740910122316g7db0ae7ci1f79e3aede07d112@mail.gmail.com>
	<25EFC4E3-88E6-4286-8432-8118CC9E899C@freeswitch.org>
	<f8af5740910122332i6c266250q7a550fb05439ae53@mail.gmail.com>
	<5FBD4715-DB4A-481A-AAAF-09D7541EC3F4@freeswitch.org>
Message-ID: <f8af5740910130130u6f16745h965cefdfb32c6441@mail.gmail.com>

hi brain,

thank u very much for your valuable time,
can u please tell me where the groups data will maintain thought the
session,

Thanks
Srinivas

On Tue, Oct 13, 2009 at 12:18 PM, Brian West <brian at freeswitch.org> wrote:

> Its based on the directory and who is in the group... check out the
> defaults it does exactly this on the 2000 range if I recall correctly.
>
> /b
>
> On Oct 12, 2009, at 11:32 PM, srinivasula reddy wrote:
>
> > OK, then when i call to group number(911) how it will call to all
> > the registered members in group?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Srinivasula Reddy K
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From srinivas.ksvreddy at gmail.com  Tue Oct 13 01:56:19 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 13 Oct 2009 14:26:19 +0530
Subject: [Freeswitch-users] 606 error
Message-ID: <f8af5740910130156r1145eabcl3c59f83779cb864f@mail.gmail.com>

Hi,

two users are registered in freeswitch, when i making call to another user i
am getting 606 error,
any help

-- 
Srinivasula Reddy K
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From tculjaga at gmail.com  Tue Oct 13 02:05:18 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 13 Oct 2009 11:05:18 +0200
Subject: [Freeswitch-users] 606 error
In-Reply-To: <f8af5740910130156r1145eabcl3c59f83779cb864f@mail.gmail.com>
References: <f8af5740910130156r1145eabcl3c59f83779cb864f@mail.gmail.com>
Message-ID: <65d96fc80910130205t5ce571ecu7746e4886b9362a3@mail.gmail.com>

what about some console logs & sip traces ?

T.

On Tue, Oct 13, 2009 at 10:56 AM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

> Hi,
>
> two users are registered in freeswitch, when i making call to another user
> i am getting 606 error,
> any help
>
> --
> Srinivasula Reddy K
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From srinivas.ksvreddy at gmail.com  Tue Oct 13 02:13:06 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 13 Oct 2009 14:43:06 +0530
Subject: [Freeswitch-users] 606 error
In-Reply-To: <65d96fc80910130205t5ce571ecu7746e4886b9362a3@mail.gmail.com>
References: <f8af5740910130156r1145eabcl3c59f83779cb864f@mail.gmail.com>
	<65d96fc80910130205t5ce571ecu7746e4886b9362a3@mail.gmail.com>
Message-ID: <f8af5740910130213p73d15225qbddce09854440b84@mail.gmail.com>

Hi,



Console user1181 attempted to call console user1171 resulted in failure. Sip
server returned "Temporarily unavailable" with reason header cause=606;

text="user-not-registered".  This also happened with other consoles.

Thanks
SRINIVAS


On Tue, Oct 13, 2009 at 2:35 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> what about some console logs & sip traces ?
>
> T.
>
> On Tue, Oct 13, 2009 at 10:56 AM, srinivasula reddy <
> srinivas.ksvreddy at gmail.com> wrote:
>
>> Hi,
>>
>> two users are registered in freeswitch, when i making call to another user
>> i am getting 606 error,
>> any help
>>
>> --
>> Srinivasula Reddy K
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Srinivasula Reddy K
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From vhatz at kinetix.gr  Tue Oct 13 02:40:58 2009
From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI))
Date: Tue, 13 Oct 2009 12:40:58 +0300
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <4AD43B6D.7000204@gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>	<881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>	<FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>
	<4AD43B6D.7000204@gmail.com>
Message-ID: <4AD44B2A.1040109@kinetix.gr>

Meftah Tayeb wrote:
> hello,
> yes, host please to let users test it and report bug
> YATE/Asterisk fully support but freeswitch no fully support it

I would disagree about YATE & Asterisk fully supporting H323.

:)

They both have some support for H323 for years now, but only for voice 
calls. None of the two platforms can do fax or video calls in H323 for 
example.

YATE especially, cannot even pass DTMF between H323 and SIP unless it is 
in RTP (RFC 2833).

I think it is a good opportunity for FS to make a difference if proper 
H323 support is built for it.


Best regards,
Vlasis Hatzistavrou.


From mcampbellsmith at gmail.com  Tue Oct 13 02:56:16 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 13 Oct 2009 20:56:16 +1100
Subject: [Freeswitch-users] dingaling: Destination out of order
Message-ID: <33c87fa30910130256j285c5f52l7a50d8ce19e11f80@mail.gmail.com>

Hi!

I am trying to call from FS to gtalk.  This used to work, so not sure
if there is a problem with my build (FreeSWITCH Version 1.0.trunk
(15126))

freeswitch at internal> dingaling status
--DingaLing status--
login   |       connected
mygmailid at gmail.com/gtalk    |       AUTHORIZED

It looks okay and I also see FS registered and online in the GTALK client.

When I dial 9999 (which is to call my gtalk user), I get the following
in the console:

2009-10-13 20:49:13.458712 [INFO] mod_dialplan_xml.c:391 Processing
1000->9999 in context default
2009-10-13 20:49:13.490719 [NOTICE] mod_dingaling.c:712 Close Channel
N/A [CS_NEW]
2009-10-13 20:49:13.498706 [ERR] switch_ivr_originate.c:1667 Cannot
create outgoing channel of type [dingaling] cause:
[DESTINATION_OUT_OF_ORDER

In dingaling.conf.xml, I only have the PCMU codec specified, and the
ATA is requesting PCMU/8000.

Any ideas why I am seeing this?

Thanks!


From fdelawarde at wirelessmundi.com  Tue Oct 13 03:09:45 2009
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Tue, 13 Oct 2009 12:09:45 +0200
Subject: [Freeswitch-users] sofia gateways and linux multipath routing
Message-ID: <1255428585.11786.90.camel@francois.tc.commsmundi.com>

Hello all,

I'm interested in using mod_sofia with multiple Internet connections
(configured as a unique load-balancing route using multipath).

One solution would be to define a different profile for each connection,
but it would be more practical having a unique external profile that
would automatically handle everything (detecting multiple public IPs and
selecting the right one for a call, being able to select the router for
gateway registration...).

Routing table:
192.168.10.0/24 dev eth0  proto kernel  scope link  src 192.168.10.1
192.168.1.0/24 dev eth1  proto kernel  scope link  src 192.168.1.2
192.168.2.0/24 dev eth2  proto kernel  scope link  src 192.168.2.2
default  proto static
	nexthop via 192.168.1.1  dev eth1 weight 1
	nexthop via 192.168.2.1  dev eth2 weight 1

Both default routers (192.168.1.1 and 192.168.2.1) would have a distinct
public IP.

Several questions cross my mind:
- Can a unique sofia profile be bound to multiple IPs (not 0.0.0.0)?

- How would FS behave with a unique external profile in that situation?
   * Would FS reply to an incoming call using the same router it came
from forcing packet source address?
   * Would FS stick to a unique router for all flows of an outgoing call
(SIP, RTP, UDPTL)?
   * Can I force a gateway to use a given router (for calls,
registration, ...)?
   * Would the NAT system (using stun or auto-nat) work in that
situation, or does it assume only one default router (and a unique
public IP) exists per profile?

- Knowing the above, would it be necessary to use a different profile
for each router/interface, and define the same gateway in each of these?

- Tricky question: What if multiple routers are on the same
network/interface (192.168.1.1, 192.168.1.2, ...)?


Thanks in advance,
Fran?ois.



From b_ball_henry at hotmail.com  Tue Oct 13 03:14:22 2009
From: b_ball_henry at hotmail.com (Henry Huang)
Date: Tue, 13 Oct 2009 18:14:22 +0800
Subject: [Freeswitch-users] sched_api doesn't get launched
Message-ID: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>

Hi:
I am using mod_java. And in my script I was able to achieve using:

execute("sched_hangup", "+300 alloted_timeout");

However, when I try to run sched_api in the same way, system log returns
that it's an invalid application. I have also tried to trigger it with many
conditional channel variable api calls , but non of them seemed to execute
the api command (because I turned on the highest level of debugging and see
no where the sched_api is being called.

The closest thing I got was by using "api_after_bridge" like the following,
but it only launches when the bridge is teared down(which is not what I
want). I originally thought after bridge means right after the 2 party is
connected. All I want is to be able to play some message to leg A at certain
time.

setVariable("api_after_bridge", "sched_api +10 none uuid_displace ${uuid}
start /path/to/some.wav 20 mux");

I have been struggling with different combination for a week now.. Please
shed some light if you know something.

Thanks,

-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From jason at jasonjgw.net  Tue Oct 13 03:16:02 2009
From: jason at jasonjgw.net (Jason White)
Date: Tue, 13 Oct 2009 21:16:02 +1100
Subject: [Freeswitch-users] dingaling: Destination out of order
In-Reply-To: <33c87fa30910130256j285c5f52l7a50d8ce19e11f80@mail.gmail.com>
References: <33c87fa30910130256j285c5f52l7a50d8ce19e11f80@mail.gmail.com>
Message-ID: <20091013101602.GA19589@jdc.jasonjgw.net>

Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
> When I dial 9999 (which is to call my gtalk user), I get the following
> in the console:

[snip]

Could you turn on debug logging in the console and post the output?



From shaheryarkh at googlemail.com  Tue Oct 13 03:15:45 2009
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Tue, 13 Oct 2009 16:15:45 +0600
Subject: [Freeswitch-users] Registering a large number of SIP users
Message-ID: <b16156850910130315k4f4fdfebw96955d1b5d8e79fa@mail.gmail.com>

Hi,

I am creating a load test setup for FreeSWITCH using Sofia SIP. I have two
machines both configured with mod_xml_curl, one machine (lets call it SIP
Server) has 100 SIP accounts. Now i want to register second machine (lets
call it SIP Client) to all these 100 SIP accounts on first machine. How can
i do that?

One approach that i can think of is that to create a new profile (or use
existing external profile) on SIP Client and add all accounts to it as
gateways having,

*<param name="register" value="true">*

define in their configuration.

Is my approach correct? Are there any better ways to do this?

Thank you.


-- 
________________________________________________________
|
                     |
| FATAL ERROR                                                           ---
O X |
|_______________________________________________________|
|                        You have moved the mouse.
 |
| Windows must be restarted for the changes to take effect.   |
|                                        <OK>
               |
####################################/


Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From Prometheus001 at gmx.net  Tue Oct 13 03:17:04 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Tue, 13 Oct 2009 12:17:04 +0200
Subject: [Freeswitch-users] Mod_fifo posision in queue
In-Reply-To: <87f2f3b90909101322v68e49f74jbc4cf2052e1811c3@mail.gmail.com>
References: <8ccbff060909010737v4a8f17ep4d6175c2a2a2f866@mail.gmail.com>	<E09986E3-1177-4260-AFFA-9018F92D81A2@jerris.com>	<8ccbff060909092332q1d06e74m8d5ca3a39ca4f572@mail.gmail.com>	<87f2f3b90909100037m45b93a9ei7ec62b4d1f84e986@mail.gmail.com>	<191c3a030909101123k6ccbf949q74fa22197f44517e@mail.gmail.com>	<8ccbff060909101203m5625a991y32138e32913959c5@mail.gmail.com>	<86a32abc0909101232k656be844qfa19bb7a679a2e02@mail.gmail.com>
	<87f2f3b90909101322v68e49f74jbc4cf2052e1811c3@mail.gmail.com>
Message-ID: <4AD453A0.4070207@gmx.net>

Has anybody managed to get this to work already?
How do you play the announcements dependent on the variable in the dialplan?

Best regards
Peter


Michael Collins schrieb:
>
>
> On Thu, Sep 10, 2009 at 12:32 PM, Diego Viola <diego.viola at gmail.com
> <mailto:diego.viola at gmail.com>> wrote:
>
>     Lets make sure we add it on the wiki too =D.
>
> Yep, as soon as we verify its functionality we'll wikify it. :)
> -MC
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From tculjaga at gmail.com  Tue Oct 13 03:32:34 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 13 Oct 2009 12:32:34 +0200
Subject: [Freeswitch-users] 606 error
In-Reply-To: <f8af5740910130213p73d15225qbddce09854440b84@mail.gmail.com>
References: <f8af5740910130156r1145eabcl3c59f83779cb864f@mail.gmail.com>
	<65d96fc80910130205t5ce571ecu7746e4886b9362a3@mail.gmail.com>
	<f8af5740910130213p73d15225qbddce09854440b84@mail.gmail.com>
Message-ID: <65d96fc80910130332hc94075bq182887b5f6760f49@mail.gmail.com>

and you are sure both users are registered to the same context and your
dialplan is correct ?

T.

On Tue, Oct 13, 2009 at 11:13 AM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

>
> Hi,
>
>
>
> Console user1181 attempted to call console user1171 resulted in failure.
> Sip server returned "Temporarily unavailable" with reason header cause=606;
>
> text="user-not-registered".  This also happened with other consoles.
>
> Thanks
> SRINIVAS
>
>
>
> On Tue, Oct 13, 2009 at 2:35 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> what about some console logs & sip traces ?
>>
>> T.
>>
>> On Tue, Oct 13, 2009 at 10:56 AM, srinivasula reddy <
>> srinivas.ksvreddy at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> two users are registered in freeswitch, when i making call to another
>>> user i am getting 606 error,
>>> any help
>>>
>>> --
>>> Srinivasula Reddy K
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Srinivasula Reddy K
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mcampbellsmith at gmail.com  Tue Oct 13 03:37:07 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 13 Oct 2009 21:37:07 +1100
Subject: [Freeswitch-users] dingaling: Destination out of order
In-Reply-To: <20091013101602.GA19589@jdc.jasonjgw.net>
References: <33c87fa30910130256j285c5f52l7a50d8ce19e11f80@mail.gmail.com>
	<20091013101602.GA19589@jdc.jasonjgw.net>
Message-ID: <33c87fa30910130337y631ffc41l8064bfc0d9125d1e@mail.gmail.com>

This is all I see:

console loglevel 9
+OK console log level set to DEBUG

freeswitch at internal> 2009-10-13 21:33:05.578863 [NOTICE]
switch_channel.c:613 New Channel sofia/internal_nat/1000 at 192.168.1.120
[cad049fe-b7e3-11de-94a7-1dd4d003eac8]
2009-10-13 21:33:05.634924 [INFO] mod_dialplan_xml.c:391 Processing
10000->9999 in context default
2009-10-13 21:33:05.666835 [NOTICE] mod_dingaling.c:712 Close Channel
N/A [CS_NEW]
2009-10-13 21:33:05.674929 [ERR] switch_ivr_originate.c:1667 Cannot
create outgoing channel of type [dingaling] cause:
[DESTINATION_OUT_OF_ORDER]
2009-10-13 21:33:05.674929 [INFO] mod_dptools.c:2133 Originate Failed.
 Cause: DESTINATION_OUT_OF_ORDER
2009-10-13 21:33:05.674929 [NOTICE] mod_dptools.c:2166 Hangup
sofia/internal_nat/1000 at 192.168.1.120 [CS_EXECUTE]
[DESTINATION_OUT_OF_ORDER]
2009-10-13 21:33:05.802716 [NOTICE] switch_core_session.c:1087 Session
16 (sofia/internal_nat/1000 at 192.168.1.120) Ended
2009-10-13 21:33:05.807529 [NOTICE] switch_core_session.c:1089 Close
Channel sofia/internal_nat/1000 at 192.168.1.120 [CS_DESTROY]


On Tue, Oct 13, 2009 at 9:16 PM, Jason White <jason at jasonjgw.net> wrote:
> Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
>> When I dial 9999 (which is to call my gtalk user), I get the following
>> in the console:
>
> [snip]
>
> Could you turn on debug logging in the console and post the output?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From bottleman at icf.org.ru  Tue Oct 13 03:44:59 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Tue, 13 Oct 2009 14:44:59 +0400 (MSD)
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <23f91030910130005m1dc40f90k8f5b047baf1df144@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>
	<23f91030910130005m1dc40f90k8f5b047baf1df144@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910131435550.8815@srv.icf.org.ru>

On 2009-10-13 15:05 +0800, Seven Du wrote freeswitch-users at lists.freeswitch.org:

hm, host it if you wont, i has nothing against it.

SD>that will make life easier.
SD>
SD>2009/10/13 Brian West <brian at freeswitch.org>
SD>
SD>> Does anyone see a problem with hosting mod_h323 in our SVN?  I would
SD>> like to centralize everything we can to reuse our issue tracking
SD>> resources and not fragment the community if possible.
SD>>
SD>> /b
SD>>
SD>> On Oct 12, 2009, at 2:43 PM, Tihomir Culjaga wrote:
SD>>
SD>> > hi,
SD>> >
SD>> > finally i compiled it right ... had a stupid issue with ekiga and
SD>> > wrong ptlib in place...
SD>> >
SD>> > anyhow, i loaded the module and will continue the tests
SD>> > tomorrow ...first thing i arrive in my office :P
SD>> >
SD>> >
SD>>
SD>>
SD>> _______________________________________________
SD>> FreeSWITCH-users mailing list
SD>> FreeSWITCH-users at lists.freeswitch.org
SD>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
SD>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
SD>> http://www.freeswitch.org
SD>>
SD>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From mcampbellsmith at gmail.com  Tue Oct 13 03:52:36 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 13 Oct 2009 21:52:36 +1100
Subject: [Freeswitch-users] dingaling: Destination out of order
In-Reply-To: <33c87fa30910130256j285c5f52l7a50d8ce19e11f80@mail.gmail.com>
References: <33c87fa30910130256j285c5f52l7a50d8ce19e11f80@mail.gmail.com>
Message-ID: <33c87fa30910130352i1f3991d9x5ab1bf9eedd5b6df@mail.gmail.com>

I've fixed the problem.

My dialplan for outbound calling had a typo:

    <action application="bridge"
data="dingaling/gtalk/mygmailid at gmail.com"/>

The gtalk was gtallk somehow .....

On Tue, Oct 13, 2009 at 8:56 PM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> Hi!
>
> I am trying to call from FS to gtalk. ?This used to work, so not sure
> if there is a problem with my build (FreeSWITCH Version 1.0.trunk
> (15126))
>
> freeswitch at internal> dingaling status
> --DingaLing status--
> login ? | ? ? ? connected
> mygmailid at gmail.com/gtalk ? ?| ? ? ? AUTHORIZED
>
> It looks okay and I also see FS registered and online in the GTALK client.
>
> When I dial 9999 (which is to call my gtalk user), I get the following
> in the console:
>
> 2009-10-13 20:49:13.458712 [INFO] mod_dialplan_xml.c:391 Processing
> 1000->9999 in context default
> 2009-10-13 20:49:13.490719 [NOTICE] mod_dingaling.c:712 Close Channel
> N/A [CS_NEW]
> 2009-10-13 20:49:13.498706 [ERR] switch_ivr_originate.c:1667 Cannot
> create outgoing channel of type [dingaling] cause:
> [DESTINATION_OUT_OF_ORDER
>
> In dingaling.conf.xml, I only have the PCMU codec specified, and the
> ATA is requesting PCMU/8000.
>
> Any ideas why I am seeing this?
>
> Thanks!
>


From srinivas.ksvreddy at gmail.com  Tue Oct 13 04:03:28 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 13 Oct 2009 16:33:28 +0530
Subject: [Freeswitch-users] 606 error
In-Reply-To: <65d96fc80910130332hc94075bq182887b5f6760f49@mail.gmail.com>
References: <f8af5740910130156r1145eabcl3c59f83779cb864f@mail.gmail.com>
	<65d96fc80910130205t5ce571ecu7746e4886b9362a3@mail.gmail.com>
	<f8af5740910130213p73d15225qbddce09854440b84@mail.gmail.com>
	<65d96fc80910130332hc94075bq182887b5f6760f49@mail.gmail.com>
Message-ID: <f8af5740910130403v714651a8x31eff11b53097ee9@mail.gmail.com>

Hi,

thank u very much for your valuable time,
s am sure they are both in same it is not occur continuously, i dont know
the reason,
i am having the wireshark file, any help?

thanks
srinivas

On Tue, Oct 13, 2009 at 4:02 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> and you are sure both users are registered to the same context and your
> dialplan is correct ?
>
> T.
>
>
> On Tue, Oct 13, 2009 at 11:13 AM, srinivasula reddy <
> srinivas.ksvreddy at gmail.com> wrote:
>
>>
>> Hi,
>>
>>
>>
>> Console user1181 attempted to call console user1171 resulted in failure.
>> Sip server returned "Temporarily unavailable" with reason header cause=606;
>>
>> text="user-not-registered".  This also happened with other consoles.
>>
>> Thanks
>> SRINIVAS
>>
>>
>>
>> On Tue, Oct 13, 2009 at 2:35 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>
>>> what about some console logs & sip traces ?
>>>
>>> T.
>>>
>>> On Tue, Oct 13, 2009 at 10:56 AM, srinivasula reddy <
>>> srinivas.ksvreddy at gmail.com> wrote:
>>>
>>>> Hi,
>>>>
>>>> two users are registered in freeswitch, when i making call to another
>>>> user i am getting 606 error,
>>>> any help
>>>>
>>>> --
>>>> Srinivasula Reddy K
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Srinivasula Reddy K
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Srinivasula Reddy K
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From srinivas.ksvreddy at gmail.com  Tue Oct 13 04:23:36 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 13 Oct 2009 16:53:36 +0530
Subject: [Freeswitch-users] 606 error
In-Reply-To: <f8af5740910130403v714651a8x31eff11b53097ee9@mail.gmail.com>
References: <f8af5740910130156r1145eabcl3c59f83779cb864f@mail.gmail.com>
	<65d96fc80910130205t5ce571ecu7746e4886b9362a3@mail.gmail.com>
	<f8af5740910130213p73d15225qbddce09854440b84@mail.gmail.com>
	<65d96fc80910130332hc94075bq182887b5f6760f49@mail.gmail.com>
	<f8af5740910130403v714651a8x31eff11b53097ee9@mail.gmail.com>
Message-ID: <f8af5740910130423o6ac1ae61g7babc49a9c4eddfa@mail.gmail.com>

wireshark image

thanks
srinivas

On Tue, Oct 13, 2009 at 4:33 PM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

> Hi,
>
> thank u very much for your valuable time,
> s am sure they are both in same it is not occur continuously, i dont know
> the reason,
> i am having the wireshark file, any help?
>
> thanks
> srinivas
>
>
> On Tue, Oct 13, 2009 at 4:02 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> and you are sure both users are registered to the same context and your
>> dialplan is correct ?
>>
>> T.
>>
>>
>> On Tue, Oct 13, 2009 at 11:13 AM, srinivasula reddy <
>> srinivas.ksvreddy at gmail.com> wrote:
>>
>>>
>>> Hi,
>>>
>>>
>>>
>>> Console user1181 attempted to call console user1171 resulted in failure.
>>> Sip server returned "Temporarily unavailable" with reason header cause=606;
>>>
>>> text="user-not-registered".  This also happened with other consoles.
>>>
>>> Thanks
>>> SRINIVAS
>>>
>>>
>>>
>>> On Tue, Oct 13, 2009 at 2:35 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>>
>>>> what about some console logs & sip traces ?
>>>>
>>>> T.
>>>>
>>>> On Tue, Oct 13, 2009 at 10:56 AM, srinivasula reddy <
>>>> srinivas.ksvreddy at gmail.com> wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> two users are registered in freeswitch, when i making call to another
>>>>> user i am getting 606 error,
>>>>> any help
>>>>>
>>>>> --
>>>>> Srinivasula Reddy K
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Srinivasula Reddy K
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Srinivasula Reddy K
>



-- 
Srinivasula Reddy K
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From tculjaga at gmail.com  Tue Oct 13 04:35:45 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 13 Oct 2009 13:35:45 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910131435550.8815@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>
	<23f91030910130005m1dc40f90k8f5b047baf1df144@mail.gmail.com>
	<Pine.LNX.4.64.0910131435550.8815@srv.icf.org.ru>
Message-ID: <65d96fc80910130435x67405db8s4d0927ad68be05d6@mail.gmail.com>

static const char* h323_formats[] = {
    "G.711-*A*Law-64k", "PCM*U*",
    "G.711-*u*Law-64k", "PCM*A*",
    "GSM-06.10", "gsm",
    "MS-GSM", "msgsm",



I've changed this to meed desired caps ... need more tests ...


2009/10/13 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-13 15:05 +0800, Seven Du wrote
> freeswitch-users at lists.freeswitch.org:
>
> hm, host it if you wont, i has nothing against it.
>
> SD>that will make life easier.
> SD>
> SD>2009/10/13 Brian West <brian at freeswitch.org>
> SD>
> SD>> Does anyone see a problem with hosting mod_h323 in our SVN?  I would
> SD>> like to centralize everything we can to reuse our issue tracking
> SD>> resources and not fragment the community if possible.
> SD>>
> SD>> /b
> SD>>
> SD>> On Oct 12, 2009, at 2:43 PM, Tihomir Culjaga wrote:
> SD>>
> SD>> > hi,
> SD>> >
> SD>> > finally i compiled it right ... had a stupid issue with ekiga and
> SD>> > wrong ptlib in place...
> SD>> >
> SD>> > anyhow, i loaded the module and will continue the tests
> SD>> > tomorrow ...first thing i arrive in my office :P
> SD>> >
> SD>> >
> SD>>
> SD>>
> SD>> _______________________________________________
> SD>> FreeSWITCH-users mailing list
> SD>> FreeSWITCH-users at lists.freeswitch.org
> SD>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> SD>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> SD>> http://www.freeswitch.org
> SD>>
> SD>
>
> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129-RIPE
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bottleman at icf.org.ru  Tue Oct 13 04:50:13 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Tue, 13 Oct 2009 15:50:13 +0400 (MSD)
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910130435x67405db8s4d0927ad68be05d6@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>
	<23f91030910130005m1dc40f90k8f5b047baf1df144@mail.gmail.com>
	<Pine.LNX.4.64.0910131435550.8815@srv.icf.org.ru>
	<65d96fc80910130435x67405db8s4d0927ad68be05d6@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910131542350.28673@srv.icf.org.ru>

On 2009-10-13 13:35 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

this morning me bring in hospital, and now i cannot make much work, 
i think return to the ranks in 1-2 week.

TC>static const char* h323_formats[] = {
TC>    "G.711-*A*Law-64k", "PCM*U*",
TC>    "G.711-*u*Law-64k", "PCM*A*",
TC>    "GSM-06.10", "gsm",
TC>    "MS-GSM", "msgsm",
TC>
TC>
TC>
TC>I've changed this to meed desired caps ... need more tests ...
TC>
TC>
TC>2009/10/13 Georgiewskiy Yuriy <bottleman at icf.org.ru>
TC>
TC>> On 2009-10-13 15:05 +0800, Seven Du wrote
TC>> freeswitch-users at lists.freeswitch.org:
TC>>
TC>> hm, host it if you wont, i has nothing against it.
TC>>
TC>> SD>that will make life easier.
TC>> SD>
TC>> SD>2009/10/13 Brian West <brian at freeswitch.org>
TC>> SD>
TC>> SD>> Does anyone see a problem with hosting mod_h323 in our SVN?  I would
TC>> SD>> like to centralize everything we can to reuse our issue tracking
TC>> SD>> resources and not fragment the community if possible.
TC>> SD>>
TC>> SD>> /b
TC>> SD>>
TC>> SD>> On Oct 12, 2009, at 2:43 PM, Tihomir Culjaga wrote:
TC>> SD>>
TC>> SD>> > hi,
TC>> SD>> >
TC>> SD>> > finally i compiled it right ... had a stupid issue with ekiga and
TC>> SD>> > wrong ptlib in place...
TC>> SD>> >
TC>> SD>> > anyhow, i loaded the module and will continue the tests
TC>> SD>> > tomorrow ...first thing i arrive in my office :P
TC>> SD>> >
TC>> SD>> >
TC>> SD>>
TC>> SD>>
TC>> SD>> _______________________________________________
TC>> SD>> FreeSWITCH-users mailing list
TC>> SD>> FreeSWITCH-users at lists.freeswitch.org
TC>> SD>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> SD>> UNSUBSCRIBE:
TC>> http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> SD>> http://www.freeswitch.org
TC>> SD>>
TC>> SD>
TC>>
TC>> C ?????????                       With Best Regards
TC>> ???????????? ????.                Georgiewskiy Yuriy
TC>> +7 4872 711666                    +7 4872 711666
TC>> ???? +7 4872 711143               fax +7 4872 711143
TC>> ???????? ??? "?? ?? ??????"       IT Service Ltd
TC>> http://nkoort.ru                  http://nkoort.ru
TC>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
TC>> YG129-RIPE                        YG129-RIPE
TC>>
TC>> _______________________________________________
TC>> FreeSWITCH-users mailing list
TC>> FreeSWITCH-users at lists.freeswitch.org
TC>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> http://www.freeswitch.org
TC>>
TC>>
TC>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From tculjaga at gmail.com  Tue Oct 13 04:54:34 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 13 Oct 2009 13:54:34 +0200
Subject: [Freeswitch-users] 606 error
In-Reply-To: <f8af5740910130403v714651a8x31eff11b53097ee9@mail.gmail.com>
References: <f8af5740910130156r1145eabcl3c59f83779cb864f@mail.gmail.com>
	<65d96fc80910130205t5ce571ecu7746e4886b9362a3@mail.gmail.com>
	<f8af5740910130213p73d15225qbddce09854440b84@mail.gmail.com>
	<65d96fc80910130332hc94075bq182887b5f6760f49@mail.gmail.com>
	<f8af5740910130403v714651a8x31eff11b53097ee9@mail.gmail.com>
Message-ID: <65d96fc80910130454n1860dc09l968454d32b4cd361@mail.gmail.com>

of course, if you can send it thi will be great...

T.

On Tue, Oct 13, 2009 at 1:03 PM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

> Hi,
>
> thank u very much for your valuable time,
> s am sure they are both in same it is not occur continuously, i dont know
> the reason,
> i am having the wireshark file, any help?
>
> thanks
> srinivas
>
>
> On Tue, Oct 13, 2009 at 4:02 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> and you are sure both users are registered to the same context and your
>> dialplan is correct ?
>>
>> T.
>>
>>
>> On Tue, Oct 13, 2009 at 11:13 AM, srinivasula reddy <
>> srinivas.ksvreddy at gmail.com> wrote:
>>
>>>
>>> Hi,
>>>
>>>
>>>
>>> Console user1181 attempted to call console user1171 resulted in failure.
>>> Sip server returned "Temporarily unavailable" with reason header cause=606;
>>>
>>> text="user-not-registered".  This also happened with other consoles.
>>>
>>> Thanks
>>> SRINIVAS
>>>
>>>
>>>
>>> On Tue, Oct 13, 2009 at 2:35 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>>
>>>> what about some console logs & sip traces ?
>>>>
>>>> T.
>>>>
>>>> On Tue, Oct 13, 2009 at 10:56 AM, srinivasula reddy <
>>>> srinivas.ksvreddy at gmail.com> wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> two users are registered in freeswitch, when i making call to another
>>>>> user i am getting 606 error,
>>>>> any help
>>>>>
>>>>> --
>>>>> Srinivasula Reddy K
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Srinivasula Reddy K
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Srinivasula Reddy K
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Tue Oct 13 05:02:56 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 13 Oct 2009 14:02:56 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910131542350.28673@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>
	<23f91030910130005m1dc40f90k8f5b047baf1df144@mail.gmail.com>
	<Pine.LNX.4.64.0910131435550.8815@srv.icf.org.ru>
	<65d96fc80910130435x67405db8s4d0927ad68be05d6@mail.gmail.com>
	<Pine.LNX.4.64.0910131542350.28673@srv.icf.org.ru>
Message-ID: <65d96fc80910130502k7a3fedf5y1c4d6fb5ad8f7bd7@mail.gmail.com>

2009/10/13 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-13 13:35 +0200, Tihomir Culjaga wrote
> freeswitch-users at lists.fre...:
>
> this morning me bring in hospital, and now i cannot make much work,
> i think return to the ranks in 1-2 week.
>
>
damn, hope you will recover soon... take it easy.

T.
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From vhatz at kinetix.gr  Tue Oct 13 05:13:48 2009
From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI))
Date: Tue, 13 Oct 2009 15:13:48 +0300
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910131542350.28673@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>	<FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>	<23f91030910130005m1dc40f90k8f5b047baf1df144@mail.gmail.com>	<Pine.LNX.4.64.0910131435550.8815@srv.icf.org.ru>	<65d96fc80910130435x67405db8s4d0927ad68be05d6@mail.gmail.com>
	<Pine.LNX.4.64.0910131542350.28673@srv.icf.org.ru>
Message-ID: <4AD46EFC.7060306@kinetix.gr>


Georgiewskiy Yuriy wrote:
> On 2009-10-13 13:35 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:
> 
> this morning me bring in hospital, and now i cannot make much work, 
> i think return to the ranks in 1-2 week.

I wish you a speedy recovery, Yuriy.

Regards,
Vlasis.


From ryannyl at gmail.com  Tue Oct 13 05:21:06 2009
From: ryannyl at gmail.com (Ryanny Lin)
Date: Tue, 13 Oct 2009 20:21:06 +0800
Subject: [Freeswitch-users] Registering a large number of SIP users
In-Reply-To: <b16156850910130315k4f4fdfebw96955d1b5d8e79fa@mail.gmail.com>
References: <b16156850910130315k4f4fdfebw96955d1b5d8e79fa@mail.gmail.com>
Message-ID: <4bfcac7e0910130521s44a042c0m25eb5c62032d4764@mail.gmail.com>

You may try this tool, "sipp", to execute a load test.
http://sipp.sourceforge.net/

2009/10/13 Muhammad Shahzad <shaheryarkh at googlemail.com>

> Hi,
>
> I am creating a load test setup for FreeSWITCH using Sofia SIP. I have two
> machines both configured with mod_xml_curl, one machine (lets call it SIP
> Server) has 100 SIP accounts. Now i want to register second machine (lets
> call it SIP Client) to all these 100 SIP accounts on first machine. How can
> i do that?
>
> One approach that i can think of is that to create a new profile (or use
> existing external profile) on SIP Client and add all accounts to it as
> gateways having,
>
> *<param name="register" value="true">*
>
> define in their configuration.
>
> Is my approach correct? Are there any better ways to do this?
>
> Thank you.
>
>
> --
> ________________________________________________________
> |
>                      |
> | FATAL ERROR                                                           ---
> O X |
> |_______________________________________________________|
> |                        You have moved the mouse.
>    |
> | Windows must be restarted for the changes to take effect.   |
> |                                        <OK>
>                |
> ####################################/
>
>
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sincerely regards,
Wen-Jen
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From shaheryarkh at googlemail.com  Tue Oct 13 05:31:00 2009
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Tue, 13 Oct 2009 18:31:00 +0600
Subject: [Freeswitch-users] Registering a large number of SIP users
In-Reply-To: <4bfcac7e0910130521s44a042c0m25eb5c62032d4764@mail.gmail.com>
References: <b16156850910130315k4f4fdfebw96955d1b5d8e79fa@mail.gmail.com>
	<4bfcac7e0910130521s44a042c0m25eb5c62032d4764@mail.gmail.com>
Message-ID: <b16156850910130531i27fa8695j3f83806b27d9ab45@mail.gmail.com>

Yes i have used this tool before, but sip registration is just first part of
my load test, i will be make SIP call load testing too. Also, i want to use
FreeSWITCH against FreeSWITCH to test its capability both as SIP Server and
SIP Client.

Thank you.


On Tue, Oct 13, 2009 at 6:21 PM, Ryanny Lin <ryannyl at gmail.com> wrote:

> You may try this tool, "sipp", to execute a load test.
> http://sipp.sourceforge.net/
>
> 2009/10/13 Muhammad Shahzad <shaheryarkh at googlemail.com>
>
>> Hi,
>>
>> I am creating a load test setup for FreeSWITCH using Sofia SIP. I have two
>> machines both configured with mod_xml_curl, one machine (lets call it SIP
>> Server) has 100 SIP accounts. Now i want to register second machine (lets
>> call it SIP Client) to all these 100 SIP accounts on first machine. How can
>> i do that?
>>
>> One approach that i can think of is that to create a new profile (or use
>> existing external profile) on SIP Client and add all accounts to it as
>> gateways having,
>>
>> *<param name="register" value="true">*
>>
>> define in their configuration.
>>
>> Is my approach correct? Are there any better ways to do this?
>>
>> Thank you.
>>
>>
>> --
>> ________________________________________________________
>> |
>>                        |
>> | FATAL ERROR
>> --- O X |
>> |_______________________________________________________|
>> |                        You have moved the mouse.
>>    |
>> | Windows must be restarted for the changes to take effect.   |
>> |                                        <OK>
>>                  |
>> ####################################/
>>
>>
>> Muhammad Shahzad
>> -----------------------------------
>> CISCO Rich Media Communication Specialist (CRMCS)
>> CISCO Certified Network Associate (CCNA)
>> Cell: +92 334 422 40 88
>> MSN: shari_786pk at hotmail.com
>> Email: shaheryarkh at googlemail.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Sincerely regards,
> Wen-Jen
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
________________________________________________________
|
                     |
| FATAL ERROR                                                           ---
O X |
|_______________________________________________________|
|                        You have moved the mouse.
 |
| Windows must be restarted for the changes to take effect.   |
|                                        <OK>
               |
####################################/


Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From lakindia89 at gmail.com  Tue Oct 13 06:45:04 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Tue, 13 Oct 2009 19:15:04 +0530
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <87f2f3b90910121149x6e2114f0q996a262ad272a1f6@mail.gmail.com>
References: <7d79b3930910090407m5dc81d12pf2b0a47f6017804@mail.gmail.com>
	<20091009123615.BC50A3FA6E9@mail.cune.org>
	<7d79b3930910100647u7fd11413ucec0a535d68b3e3a@mail.gmail.com>
	<87f2f3b90910101009j1295d812jd98cae91e30b1eff@mail.gmail.com>
	<7d79b3930910120454m1ffc6dd4u42b91a4349597166@mail.gmail.com>
	<87f2f3b90910121149x6e2114f0q996a262ad272a1f6@mail.gmail.com>
Message-ID: <7d79b3930910130645y658ae041v8219c4c881fa5698@mail.gmail.com>

We are using Reliance as the Carrier.
I think, with this same Reliance carrier, in my office, they are able to
make outgoing calls through asterisk+libpri.

On Tue, Oct 13, 2009 at 12:19 AM, Michael Collins <msc at freeswitch.org>wrote:

> Lak,
>
> Okay I will need a little bit of time to dig into the IE's and what they
> contain. In the meantime can you tell me who the carrier is? I'd like to
> find out if they have some specific requirements. The fact that it doesn't
> work with libpri surprises me because that would mean that Asterisk systems
> would probably not work with this carrier as well.
>
> BTW, thanks for the very complete pastebin entries. :)
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From odermann at googlemail.com  Tue Oct 13 06:51:11 2009
From: odermann at googlemail.com (Dennis)
Date: Tue, 13 Oct 2009 15:51:11 +0200
Subject: [Freeswitch-users] SIP Overlap support?
Message-ID: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>

hi there,

i would like to ask, if fs has support for something like "SIP Overlap"?

instead of receiving the phonenumber from our carrier in a block, we
want to receive the phonenumber digit-by-digit and we want to tell fs
when the number is complete. our carrier could send us the phonenumber
digit-by-digit, but what about the fs-side?


thanks and kind regards
dennis


From jerry.richards at teotech.com  Tue Oct 13 07:54:30 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 13 Oct 2009 07:54:30 -0700
Subject: [Freeswitch-users] FS Extension Groups Documentation
Message-ID: <E8FDE7D1E87C4DDC98BD76DAF7DC50A5@greyhawk.tonecommander.com>


Is there a link to the documentation of FS groups and what can be done with
a group (i.e. capabilities)?

Thanks And Best Regards,
Jerry



From anthony.minessale at gmail.com  Tue Oct 13 08:03:38 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 13 Oct 2009 10:03:38 -0500
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
Message-ID: <191c3a030910130803k321f39e3n265e64ab3c84a5e6@mail.gmail.com>

have you tried it?
I *think* either we did support it or we would with a small patch to sofia
lib that I cannot recall if we ever got committed.


On Tue, Oct 13, 2009 at 8:51 AM, Dennis <odermann at googlemail.com> wrote:

> hi there,
>
> i would like to ask, if fs has support for something like "SIP Overlap"?
>
> instead of receiving the phonenumber from our carrier in a block, we
> want to receive the phonenumber digit-by-digit and we want to tell fs
> when the number is complete. our carrier could send us the phonenumber
> digit-by-digit, but what about the fs-side?
>
>
> thanks and kind regards
> dennis
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From odermann at googlemail.com  Tue Oct 13 08:26:56 2009
From: odermann at googlemail.com (Dennis)
Date: Tue, 13 Oct 2009 17:26:56 +0200
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <191c3a030910130803k321f39e3n265e64ab3c84a5e6@mail.gmail.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
	<191c3a030910130803k321f39e3n265e64ab3c84a5e6@mail.gmail.com>
Message-ID: <5e414ed0910130826j11d45a66q86d785f900e4196e@mail.gmail.com>

how could we try? we played arround with a snom phone (snom seems to
support something in this direction, but are not shure, how we can
test it and how we can see if it is supported or not.

any hint?


2009/10/13 Anthony Minessale <anthony.minessale at gmail.com>:
> have you tried it?
> I *think* either we did support it or we would with a small patch to sofia
> lib that I cannot recall if we ever got committed.
>
>
> On Tue, Oct 13, 2009 at 8:51 AM, Dennis <odermann at googlemail.com> wrote:
>>
>> hi there,
>>
>> i would like to ask, if fs has support for something like "SIP Overlap"?
>>
>> instead of receiving the phonenumber from our carrier in a block, we
>> want to receive the phonenumber digit-by-digit and we want to tell fs
>> when the number is complete. our carrier could send us the phonenumber
>> digit-by-digit, but what about the fs-side?
>>
>>
>> thanks and kind regards
>> dennis
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From shiyanov at gmail.com  Tue Oct 13 08:32:12 2009
From: shiyanov at gmail.com (Artem Shiyanov)
Date: Tue, 13 Oct 2009 19:32:12 +0400
Subject: [Freeswitch-users] mod_socket: custom timeout for HEARTBEAT
	event
In-Reply-To: <191c3a030910120726p46d6410ejd94b61612686fbbc@mail.gmail.com>
References: <d211f54a0910110959y3d3deed7s43201a3e39756585@mail.gmail.com>
	<86a32abc0910111444j1ab2e446p3cb94b0ae04126e3@mail.gmail.com>
	<97B4BDC9-5B11-4A98-B1BF-CC00A82ECBD9@jerris.com>
	<d211f54a0910112238i5f69275ey7cad437e47e6b567@mail.gmail.com>
	<191c3a030910120726p46d6410ejd94b61612686fbbc@mail.gmail.com>
Message-ID: <d211f54a0910130832m25aaadd3t4d6e938d1c71d701@mail.gmail.com>

Sorry for my foolishness but I stil can't grasp it.
I'm developing app based on inbound mod_event_socket and I don't know how to
run "enable_heartbeat" or " sched_heartbeat" without specifying any alive
session uuid. I tried to use "create_uuid" and send mentioned  commands to
the created uuid  but this approach doesn't work. I do need to monitor the
FS state itself rather then any particular call.

Please, enlighten me!
Artem



On Mon, Oct 12, 2009 at 6:26 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> it works in either case with or without media
> the syntax for setting the frequency was answered above.
>
>
>
> On Mon, Oct 12, 2009 at 12:38 AM, Artem Shiyanov <shiyanov at gmail.com>wrote:
>
>> Michael, Diego,
>> thanks for the rapid answers!
>>
>> As far as I understand, "enable_heartbeat" app is launching
>> SESSION_HEARTBEAT events that will stop when the call will be cleared. Also
>> I "heard" that "enable_heartbeat" works only for calls with proxied media.
>>
>> What I want is to monitor FreeSwitch status: is it alive and what is the
>> system status message. This info is provided in HEARTBEAT event gracefully
>> but in constant time period = 20 sec. So the main question is- how to
>> customize this period?
>>
>>
>> Artem
>>
>>
>>
>>
>> On Mon, Oct 12, 2009 at 2:52 AM, Michael Jerris <mike at jerris.com> wrote:
>>
>>>
>>> On Oct 11, 2009, at 5:44 PM, Diego Viola wrote:
>>>
>>> > Maybe you want enable_heartbeat or sched_heartbeat from mod_dptools?
>>> >
>>> > You can pass your parameters in second to these two.
>>> >
>>> > Example:
>>> >
>>> > <action application="enable_heartbeat" data="1"/>
>>> > <action application="sched_heartbeat" data="1"/>
>>> >
>>> > Where 1 in this case is the number of heartbeats per seconds.
>>> >
>>>
>>> Number of seconds between hearbeats, not hearbeats per second.
>>>
>>>
>>> > You can use that example on the Dialplan XML but you can also use it
>>> > on mod_event_socket outbound, etc.
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Tue Oct 13 08:42:46 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 13 Oct 2009 11:42:46 -0400
Subject: [Freeswitch-users] Fwd: Groups information in sqllite
In-Reply-To: <f8af5740910122302i47ccde5bo9df2bae965408394@mail.gmail.com>
References: <f8af5740910122301u338ca128macfc0b769ea0e961@mail.gmail.com>
	<f8af5740910122302i47ccde5bo9df2bae965408394@mail.gmail.com>
Message-ID: <B3C83D64-BC34-4F67-B72E-9EAE6F3A362E@jerris.com>

Group information is not stored in sqlite, it is pulled from the xml  
registry (switch_xml_locate_group function can find them) .  Also,  
please do not cross post between lists.

http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups
http://wiki.freeswitch.org/wiki/Mod_commands#in_group
http://wiki.freeswitch.org/wiki/Mod_commands#group_call

Mike


On Oct 13, 2009, at 2:02 AM, srinivasula reddy wrote:
> can any know where group information is exactly stored in sqllite  
> database, i have seen sip_registration here i can find the  
> registered users,
> in the same way how i can i find the group information, and which  
> user belongs to which user?
> any help would be great.



From pippyduck1127 at hotmail.com  Tue Oct 13 06:58:56 2009
From: pippyduck1127 at hotmail.com (Pajongjit Buntaokit)
Date: Tue, 13 Oct 2009 13:58:56 +0000
Subject: [Freeswitch-users] Database for Audio Data
Message-ID: <BAY119-W52BD912544FAA1DE624B88B3C70@phx.gbl>


Hi,

 

Does anyone know whether FreeSWITCH has a function to automatically record every call as an audio file in a server 
or forward them to be stored in a database with additional parameters such as caller ID, date, starting time and ending time?

 

So that these recorded audio data can be queried and retrieved with the caller ID, date and time.

 

Any suggestion or guidance, please advise.

 

Thank you very much!
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
http://clk.atdmt.com/GBL/go/177141664/direct/01/
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From msc at freeswitch.org  Tue Oct 13 08:46:10 2009
From: msc at freeswitch.org (Michael S Collins)
Date: Tue, 13 Oct 2009 08:46:10 -0700
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <7d79b3930910130645y658ae041v8219c4c881fa5698@mail.gmail.com>
References: <7d79b3930910090407m5dc81d12pf2b0a47f6017804@mail.gmail.com>
	<20091009123615.BC50A3FA6E9@mail.cune.org>
	<7d79b3930910100647u7fd11413ucec0a535d68b3e3a@mail.gmail.com>
	<87f2f3b90910101009j1295d812jd98cae91e30b1eff@mail.gmail.com>
	<7d79b3930910120454m1ffc6dd4u42b91a4349597166@mail.gmail.com>
	<87f2f3b90910121149x6e2114f0q996a262ad272a1f6@mail.gmail.com>
	<7d79b3930910130645y658ae041v8219c4c881fa5698@mail.gmail.com>
Message-ID: <5E7A8451-6264-4822-8152-1B94240BDD48@freeswitch.org>


On Oct 13, 2009, at 6:45 AM, lakshmanan ganapathy  
<lakindia89 at gmail.com> wrote:

> We are using Reliance as the Carrier.
> I think, with this same Reliance carrier, in my office, they are  
> able to make outgoing calls through asterisk+libpri.

If that's the case I would be very interested in seeing a pri debug  
from a working call on an asterisk box. It might give us a clue as to  
what is not working.
-MC
>
>
> On Tue, Oct 13, 2009 at 12:19 AM, Michael Collins  
> <msc at freeswitch.org> wrote:
> Lak,
>
> Okay I will need a little bit of time to dig into the IE's and what  
> they contain. In the meantime can you tell me who the carrier is?  
> I'd like to find out if they have some specific requirements. The  
> fact that it doesn't work with libpri surprises me because that  
> would mean that Asterisk systems would probably not work with this  
> carrier as well.
>
> BTW, thanks for the very complete pastebin entries. :)
> -MC
>
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From rupa at rupa.com  Tue Oct 13 08:57:05 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 13 Oct 2009 09:57:05 -0600
Subject: [Freeswitch-users] Database for Audio Data
In-Reply-To: <BAY119-W52BD912544FAA1DE624B88B3C70@phx.gbl>
References: <BAY119-W52BD912544FAA1DE624B88B3C70@phx.gbl>
Message-ID: <b39273530910130857jb7dec2eidd71e950d3902b25@mail.gmail.com>

What I do is record all calls and store the call with the UUID as the
filename.  Then when the call is hung up a CDR entry is sent to my web
server.  This CDR contains callerid and other info I might want to
query by.  The service on the web server inserts appropriate record(s)
into the database.  The recordings are available to the webserver.
When one clicks on the "listen" link, the web server serves up the
recording by UUID in the recording directory.  I have a process that
periodically removes old recordings from that dir.  I don't purge the
CDRs, though that is certainly possible.

On Tue, Oct 13, 2009 at 7:58 AM, Pajongjit Buntaokit
<pippyduck1127 at hotmail.com> wrote:
> Hi,
>
> Does anyone know whether FreeSWITCH has a function to automatically record
> every call as an audio file in a server
> or forward them to be stored in a database with additional parameters such
> as caller ID, date, starting time and ending time?
>
> So that these recorded audio data can be queried and retrieved with the
> caller ID, date and time.
>
> Any suggestion or guidance, please advise.
>
> Thank you very much!
>
> ________________________________
> Hotmail: Trusted email with Microsoft?s powerful SPAM protection. Sign up
> now.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa


From msc at freeswitch.org  Tue Oct 13 09:52:35 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Oct 2009 09:52:35 -0700
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <891080.87141.qm@web56406.mail.re3.yahoo.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
	<891080.87141.qm@web56406.mail.re3.yahoo.com>
Message-ID: <87f2f3b90910130952r5657c8e2g6e11c22140da919c@mail.gmail.com>

I will add this thought to the weekly discussion. Perhaps we can crowdsource
this one.
-MC

On Mon, Oct 12, 2009 at 10:52 PM, Mark Crane <mctch at yahoo.com> wrote:

> "In many cases the issue with the docs isn't that they aren't complete but
> rather that they are hard to find."
>
> Agreed! The biggest problem with the wiki is that it is hard to find
> things.
>
> How do books solve this problem they use an index. It is quite surprising
> that the wiki software doesn't come with ability to do this basic task
> automatically and instead relies solely upon a manually created index. It
> would be extremely beneficial to have a index of all pages and topics that
> are available on the wiki. The average person may only see the main links in
> the documentation and not realize there are actually hundreds of pages.
>
> There used to be a pdf that was auto generated from the wiki and it gave a
> much easier view of all the pages on the wiki.
>
> Mark J Crane
>
>
>
> --- On *Mon, 10/12/09, Michael Collins <msc at freeswitch.org>* wrote:
>
>
> From: Michael Collins <msc at freeswitch.org>
> Subject: Re: [Freeswitch-users] Some documentation thoughts
> To: freeswitch-users at lists.freeswitch.org
> Date: Monday, October 12, 2009, 4:38 PM
>
>
> On Mon, Oct 12, 2009 at 3:11 PM, Diego Viola <diego.viola at gmail.com<http://mc/compose?to=diego.viola at gmail.com>
> > wrote:
>
>> Hello,
>>
>> I have been doing some work recently on the FreeSWITCH wiki, to improve
>> things.
>>
>> You can see some of my work here:
>>
>>
>> http://wiki.freeswitch.org/index.php?title=Special:Contributions&limit=500&target=Diego.viola
>>
>> I am trying to polish the wiki and give it a more professional and clean
>> look, this means that we need to enforce some guidelines and strive to make
>> the wiki even better.
>>
>> How can we do this?
>>
>> Well, the first thing is to review everything we have, right now we have
>> too many separated and little pages here and there that no one cares or
>> read, we should avoid doing little or small pages. We need to correct typos,
>> I suggest that people click on "Random page" link and start correcting
>> typos, install an English (US) dictionary in your browser and enable Spell
>> Checking and start correcting these typos if you want to help, also, make
>> acronyms and initialisms all-uppercase when you make one.
>>
>> When you are adding documentation make sure you don't do a separate page
>> for an existing page, etc. and we also need to define a size for titles,
>> body, etc, so different pages don't look different from each other, I will
>> make sure I add all this into our guidelines and that all pages follow this.
>>
>> My goal is to have great documentation, similar to the Apache
>> documentation:
>> http://httpd.apache.org/docs/2.2/
>>
>> Or even better, I hope you like this idea.
>>
>
> Diego,
>
> Thank you so much for all of your help on this. Many in the community have
> seen your work - it has not gone unnoticed and it most truly is appreciated.
> I like the idea of improving the documentation. One thing we need to do is
> re-think the organization. In many cases the issue with the docs isn't that
> they aren't complete, but rather that they are hard to find. It's all about
> organization. I'm definitely open to ideas.
>
> For those who are interested in helping out with the wiki please let me
> know of your availability and skill set. I am maintaining a list of
> volunteers.
>
> Lastly, if you want to talk about documentation in real time please join us
> in #freeswitch-docs.
>
> Thanks,
> MC
>
>
> -----Inline Attachment Follows-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From william.suffill at gmail.com  Tue Oct 13 10:09:11 2009
From: william.suffill at gmail.com (William Suffill)
Date: Tue, 13 Oct 2009 13:09:11 -0400
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <87f2f3b90910130952r5657c8e2g6e11c22140da919c@mail.gmail.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com> 
	<891080.87141.qm@web56406.mail.re3.yahoo.com>
	<87f2f3b90910130952r5657c8e2g6e11c22140da919c@mail.gmail.com>
Message-ID: <6b65470d0910131009y1cf136c3q8c0932375c19f272@mail.gmail.com>

Under special pages there is ways to get a list of all the wiki pages. (
http://wiki.freeswitch.org/wiki/Special:SpecialPages)

http://wiki.freeswitch.org/wiki/Special:AllPages

Due to the number of pages it's broken into sub pages based in alphabetical
order of the page names.

-- W
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From tculjaga at gmail.com  Tue Oct 13 10:12:11 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 13 Oct 2009 19:12:11 +0200
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <5e414ed0910130826j11d45a66q86d785f900e4196e@mail.gmail.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
	<191c3a030910130803k321f39e3n265e64ab3c84a5e6@mail.gmail.com>
	<5e414ed0910130826j11d45a66q86d785f900e4196e@mail.gmail.com>
Message-ID: <65d96fc80910131012v1f72ac49o6bfe9f2267e541c5@mail.gmail.com>

you need a softswitch.... i'm afraid a SIP phone is not designed for
overlap...

T.

On Tue, Oct 13, 2009 at 5:26 PM, Dennis <odermann at googlemail.com> wrote:

> how could we try? we played arround with a snom phone (snom seems to
> support something in this direction, but are not shure, how we can
> test it and how we can see if it is supported or not.
>
> any hint?
>
>
> 2009/10/13 Anthony Minessale <anthony.minessale at gmail.com>:
> > have you tried it?
> > I *think* either we did support it or we would with a small patch to
> sofia
> > lib that I cannot recall if we ever got committed.
> >
> >
> > On Tue, Oct 13, 2009 at 8:51 AM, Dennis <odermann at googlemail.com> wrote:
> >>
> >> hi there,
> >>
> >> i would like to ask, if fs has support for something like "SIP Overlap"?
> >>
> >> instead of receiving the phonenumber from our carrier in a block, we
> >> want to receive the phonenumber digit-by-digit and we want to tell fs
> >> when the number is complete. our carrier could send us the phonenumber
> >> digit-by-digit, but what about the fs-side?
> >>
> >>
> >> thanks and kind regards
> >> dennis
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:213-799-1400
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Tue Oct 13 11:01:29 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 13 Oct 2009 13:01:29 -0500
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <65d96fc80910131012v1f72ac49o6bfe9f2267e541c5@mail.gmail.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
	<191c3a030910130803k321f39e3n265e64ab3c84a5e6@mail.gmail.com>
	<5e414ed0910130826j11d45a66q86d785f900e4196e@mail.gmail.com>
	<65d96fc80910131012v1f72ac49o6bfe9f2267e541c5@mail.gmail.com>
Message-ID: <191c3a030910131101p418e0961t373db68285248ff6@mail.gmail.com>

i do think some softphone can do it but i forgot which one it was either
snom or grandstream


On Tue, Oct 13, 2009 at 12:12 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:

> you need a softswitch.... i'm afraid a SIP phone is not designed for
> overlap...
>
> T.
>
>
> On Tue, Oct 13, 2009 at 5:26 PM, Dennis <odermann at googlemail.com> wrote:
>
>> how could we try? we played arround with a snom phone (snom seems to
>> support something in this direction, but are not shure, how we can
>> test it and how we can see if it is supported or not.
>>
>> any hint?
>>
>>
>> 2009/10/13 Anthony Minessale <anthony.minessale at gmail.com>:
>> > have you tried it?
>> > I *think* either we did support it or we would with a small patch to
>> sofia
>> > lib that I cannot recall if we ever got committed.
>> >
>> >
>> > On Tue, Oct 13, 2009 at 8:51 AM, Dennis <odermann at googlemail.com>
>> wrote:
>> >>
>> >> hi there,
>> >>
>> >> i would like to ask, if fs has support for something like "SIP
>> Overlap"?
>> >>
>> >> instead of receiving the phonenumber from our carrier in a block, we
>> >> want to receive the phonenumber digit-by-digit and we want to tell fs
>> >> when the number is complete. our carrier could send us the phonenumber
>> >> digit-by-digit, but what about the fs-side?
>> >>
>> >>
>> >> thanks and kind regards
>> >> dennis
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> > pstn:213-799-1400
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From diego.viola at gmail.com  Tue Oct 13 11:45:38 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 13 Oct 2009 18:45:38 +0000
Subject: [Freeswitch-users] sched_api doesn't get launched
In-Reply-To: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
References: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
Message-ID: <86a32abc0910131145t730d0bc0m842ed7ed0891d425@mail.gmail.com>

You need to pass the UUID to sched_hangup.

Usage: sched_hangup [+]<time> <uuid> [<cause>]

http://wiki.freeswitch.org/wiki/Mod_commands#sched_hangup


On Tue, Oct 13, 2009 at 10:14 AM, Henry Huang <b_ball_henry at hotmail.com>wrote:

> Hi:
> I am using mod_java. And in my script I was able to achieve using:
>
> execute("sched_hangup", "+300 alloted_timeout");
>
> However, when I try to run sched_api in the same way, system log returns
> that it's an invalid application. I have also tried to trigger it with many
> conditional channel variable api calls , but non of them seemed to execute
> the api command (because I turned on the highest level of debugging and see
> no where the sched_api is being called.
>
> The closest thing I got was by using "api_after_bridge" like the following,
> but it only launches when the bridge is teared down(which is not what I
> want). I originally thought after bridge means right after the 2 party is
> connected. All I want is to be able to play some message to leg A at certain
> time.
>
> setVariable("api_after_bridge", "sched_api +10 none uuid_displace ${uuid}
> start /path/to/some.wav 20 mux");
>
> I have been struggling with different combination for a week now.. Please
> shed some light if you know something.
>
> Thanks,
>
> --
> Henry Huang
> UniC Solution - Communication Unified
> VoIP & Open Source software Consultant
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Oct 13 11:48:21 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Oct 2009 11:48:21 -0700
Subject: [Freeswitch-users] mod_socket: custom timeout for HEARTBEAT
	event
In-Reply-To: <d211f54a0910130832m25aaadd3t4d6e938d1c71d701@mail.gmail.com>
References: <d211f54a0910110959y3d3deed7s43201a3e39756585@mail.gmail.com>
	<86a32abc0910111444j1ab2e446p3cb94b0ae04126e3@mail.gmail.com>
	<97B4BDC9-5B11-4A98-B1BF-CC00A82ECBD9@jerris.com>
	<d211f54a0910112238i5f69275ey7cad437e47e6b567@mail.gmail.com>
	<191c3a030910120726p46d6410ejd94b61612686fbbc@mail.gmail.com>
	<d211f54a0910130832m25aaadd3t4d6e938d1c71d701@mail.gmail.com>
Message-ID: <87f2f3b90910131148m1853a219md8788f55e40ef4c3@mail.gmail.com>

On Tue, Oct 13, 2009 at 8:32 AM, Artem Shiyanov <shiyanov at gmail.com> wrote:

> Sorry for my foolishness but I stil can't grasp it.
> I'm developing app based on inbound mod_event_socket and I don't know how
> to run "enable_heartbeat" or " sched_heartbeat" without specifying any alive
> session uuid. I tried to use "create_uuid" and send mentioned  commands to
> the created uuid  but this approach doesn't work. I do need to monitor the
> FS state itself rather then any particular call.
>
> Please, enlighten me!
> Artem
>

So if I understand correctly, you want to see the system heartbeat more
frequently than every 20 seconds? What, exactly, is in the heartbeat that
you need? You can get the same information by sending the API "status" and
reading the results.

If I read the source code correctly it looks like the system heartbeat is
hard-coded at 20 seconds. See switch_core.c, line 155. However, before you
go messing with a core file I would make sure that you can't already use the
tools that are available, like sending a "status" API every 5 seconds (or
whatever interval you need).

-MC
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From freeswitch-users-list at metik.com  Tue Oct 13 12:03:50 2009
From: freeswitch-users-list at metik.com (Metik)
Date: Tue, 13 Oct 2009 15:03:50 -0400
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <191c3a030910131101p418e0961t373db68285248ff6@mail.gmail.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com><191c3a030910130803k321f39e3n265e64ab3c84a5e6@mail.gmail.com><5e414ed0910130826j11d45a66q86d785f900e4196e@mail.gmail.com><65d96fc80910131012v1f72ac49o6bfe9f2267e541c5@mail.gmail.com>
	<191c3a030910131101p418e0961t373db68285248ff6@mail.gmail.com>
Message-ID: <538F4CF43B294D0EACABA281E1913B4C@chaos>

Both support it.  In the Grandstream, I believe it is called Early Dial (vs. SNOM's Overlap Dialing).  It can be problematic if you have a device somewhere in the middle that doesn't support 484s.

-metik
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, October 13, 2009 2:01 PM
  Subject: Re: [Freeswitch-users] SIP Overlap support?


  i do think some softphone can do it but i forgot which one it was either snom or grandstream



  On Tue, Oct 13, 2009 at 12:12 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

    you need a softswitch.... i'm afraid a SIP phone is not designed for overlap...

    T.



    On Tue, Oct 13, 2009 at 5:26 PM, Dennis <odermann at googlemail.com> wrote:

      how could we try? we played arround with a snom phone (snom seems to
      support something in this direction, but are not shure, how we can
      test it and how we can see if it is supported or not.

      any hint?


      2009/10/13 Anthony Minessale <anthony.minessale at gmail.com>:

      > have you tried it?
      > I *think* either we did support it or we would with a small patch to sofia
      > lib that I cannot recall if we ever got committed.
      >
      >
      > On Tue, Oct 13, 2009 at 8:51 AM, Dennis <odermann at googlemail.com> wrote:
      >>
      >> hi there,
      >>
      >> i would like to ask, if fs has support for something like "SIP Overlap"?
      >>
      >> instead of receiving the phonenumber from our carrier in a block, we
      >> want to receive the phonenumber digit-by-digit and we want to tell fs
      >> when the number is complete. our carrier could send us the phonenumber
      >> digit-by-digit, but what about the fs-side?
      >>
      >>
      >> thanks and kind regards
      >> dennis
      >>
      >> _______________________________________________
      >> FreeSWITCH-users mailing list
      >> FreeSWITCH-users at lists.freeswitch.org
      >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      >> http://www.freeswitch.org
      >
      >
      >
      > --
      > Anthony Minessale II
      >
      > FreeSWITCH http://www.freeswitch.org/
      > ClueCon http://www.cluecon.com/
      > Twitter: http://twitter.com/FreeSWITCH_wire
      >
      > AIM: anthm
      > MSN:anthony_minessale at hotmail.com
      > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
      > IRC: irc.freenode.net #freeswitch
      >
      > FreeSWITCH Developer Conference
      > sip:888 at conference.freeswitch.org
      > iax:guest at conference.freeswitch.org/888
      > googletalk:conf+888 at conference.freeswitch.org
      > pstn:213-799-1400
      >
      > _______________________________________________
      > FreeSWITCH-users mailing list
      > FreeSWITCH-users at lists.freeswitch.org
      > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      > http://www.freeswitch.org
      >
      >

      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org




    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





  -- 
  Anthony Minessale II

  FreeSWITCH http://www.freeswitch.org/
  ClueCon http://www.cluecon.com/
  Twitter: http://twitter.com/FreeSWITCH_wire

  AIM: anthm
  MSN:anthony_minessale at hotmail.com
  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
  IRC: irc.freenode.net #freeswitch

  FreeSWITCH Developer Conference
  sip:888 at conference.freeswitch.org
  iax:guest at conference.freeswitch.org/888
  googletalk:conf+888 at conference.freeswitch.org
  pstn:213-799-1400



------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From tculjaga at gmail.com  Tue Oct 13 12:24:55 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 13 Oct 2009 21:24:55 +0200
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <538F4CF43B294D0EACABA281E1913B4C@chaos>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
	<191c3a030910130803k321f39e3n265e64ab3c84a5e6@mail.gmail.com>
	<5e414ed0910130826j11d45a66q86d785f900e4196e@mail.gmail.com>
	<65d96fc80910131012v1f72ac49o6bfe9f2267e541c5@mail.gmail.com>
	<191c3a030910131101p418e0961t373db68285248ff6@mail.gmail.com>
	<538F4CF43B294D0EACABA281E1913B4C@chaos>
Message-ID: <65d96fc80910131224m73f35b87g355be818ad3c0caf@mail.gmail.com>

i never found it working properly... i always had some interoperability
issues and i finished having a "dialplan" on my phones being delivered
through a config file via tftp or http .. depending of the phone capability.

BTW: using overlap can lead to a greater system load... be careful when
setting the minimum number of digits you will send in 1st message. I wish
you luck...

T.

On Tue, Oct 13, 2009 at 9:03 PM, Metik <freeswitch-users-list at metik.com>wrote:

>  Both support it.  In the Grandstream, I believe it is called Early Dial
> (vs. SNOM's Overlap Dialing).  It can be problematic if you have a device
> somewhere in the middle that doesn't support 484s.
>
> -metik
>
> ----- Original Message -----
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, October 13, 2009 2:01 PM
> *Subject:* Re: [Freeswitch-users] SIP Overlap support?
>
> i do think some softphone can do it but i forgot which one it was either
> snom or grandstream
>
>
> On Tue, Oct 13, 2009 at 12:12 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> you need a softswitch.... i'm afraid a SIP phone is not designed for
>> overlap...
>>
>> T.
>>
>>
>> On Tue, Oct 13, 2009 at 5:26 PM, Dennis <odermann at googlemail.com> wrote:
>>
>>> how could we try? we played arround with a snom phone (snom seems to
>>> support something in this direction, but are not shure, how we can
>>> test it and how we can see if it is supported or not.
>>>
>>> any hint?
>>>
>>>
>>> 2009/10/13 Anthony Minessale <anthony.minessale at gmail.com>:
>>>  > have you tried it?
>>> > I *think* either we did support it or we would with a small patch to
>>> sofia
>>> > lib that I cannot recall if we ever got committed.
>>> >
>>> >
>>> > On Tue, Oct 13, 2009 at 8:51 AM, Dennis <odermann at googlemail.com>
>>> wrote:
>>> >>
>>> >> hi there,
>>> >>
>>> >> i would like to ask, if fs has support for something like "SIP
>>> Overlap"?
>>> >>
>>> >> instead of receiving the phonenumber from our carrier in a block, we
>>> >> want to receive the phonenumber digit-by-digit and we want to tell fs
>>> >> when the number is complete. our carrier could send us the phonenumber
>>> >> digit-by-digit, but what about the fs-side?
>>> >>
>>> >>
>>> >> thanks and kind regards
>>> >> dennis
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> >
>>> > --
>>> > Anthony Minessale II
>>> >
>>> > FreeSWITCH http://www.freeswitch.org/
>>> > ClueCon http://www.cluecon.com/
>>> > Twitter: http://twitter.com/FreeSWITCH_wire
>>> >
>>> > AIM: anthm
>>> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> > IRC: irc.freenode.net #freeswitch
>>> >
>>> > FreeSWITCH Developer Conference
>>> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>> > iax:guest at conference.freeswitch.org/888
>>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> > pstn:213-799-1400
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From larclap at yahoo.com  Tue Oct 13 12:38:35 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Tue, 13 Oct 2009 12:38:35 -0700
Subject: [Freeswitch-users] lua script causing FreeSwitch to crash?
Message-ID: <005a01ca4c3c$c183ceb0$448b6c10$@com>

http://pastebin.freeswitch.org/10686

 

I am trying to use a lua script for inbound calls. The caller hears a busy
signal. After this call fails to go through, FreeSwitch is no longer in
memory - it does not appear in  'ps -ef' output.

 

I don't have a clue what I might be doing incorrectly in the script in order
to cause this to happen. There is no indication in the log that FreeSwitch
has terminated. But it is not resident in memory.

 

I am using v15134. I had been using this same script without incident as
late as v14929. Until I can resolve the issue, I have removed the call to
lua from the dialplan and done all I could in it instead.

 

I have included the dialplan, the script and the cli log in the pastebin. I
would appreciate any help or suggestions.

 

Thanks, Lars

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From msc at freeswitch.org  Tue Oct 13 13:12:21 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Oct 2009 13:12:21 -0700
Subject: [Freeswitch-users] lua script causing FreeSwitch to crash?
In-Reply-To: <005a01ca4c3c$c183ceb0$448b6c10$@com>
References: <005a01ca4c3c$c183ceb0$448b6c10$@com>
Message-ID: <87f2f3b90910131312k7dac8980h70ce8be9c07aa7b2@mail.gmail.com>

On Tue, Oct 13, 2009 at 12:38 PM, Lars Zeb <larclap at yahoo.com> wrote:

>  http://pastebin.freeswitch.org/10686
>
>
>
> I am trying to use a lua script for inbound calls. The caller hears a busy
> signal. After this call fails to go through, FreeSwitch is no longer in
> memory ? it does not appear in  ?ps ?ef? output.
>
>
>
> I don?t have a clue what I might be doing incorrectly in the script in
> order to cause this to happen. There is no indication in the log that
> FreeSwitch has terminated. But it is not resident in memory.
>
>
>
> I am using v15134. I had been using this same script without incident as
> late as v14929. Until I can resolve the issue, I have removed the call to
> lua from the dialplan and done all I could in it instead.
>
We sent out a notice about a bug in event socket between revs 15119 and
15134. Update to latest trunk. (FYI, the fix was in r15135, so you just
barely missed it.)

After you update, let us know if the problem persists.
-MC
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From freeswitch-users-list at metik.com  Tue Oct 13 13:22:00 2009
From: freeswitch-users-list at metik.com (Metik)
Date: Tue, 13 Oct 2009 16:22:00 -0400
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <65d96fc80910131224m73f35b87g355be818ad3c0caf@mail.gmail.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com><191c3a030910130803k321f39e3n265e64ab3c84a5e6@mail.gmail.com><5e414ed0910130826j11d45a66q86d785f900e4196e@mail.gmail.com><65d96fc80910131012v1f72ac49o6bfe9f2267e541c5@mail.gmail.com><191c3a030910131101p418e0961t373db68285248ff6@mail.gmail.com><538F4CF43B294D0EACABA281E1913B4C@chaos>
	<65d96fc80910131224m73f35b87g355be818ad3c0caf@mail.gmail.com>
Message-ID: <48B637E0506F4A809E4C0CAF39BD08BB@chaos>

As evidenced by various DTMF interop issues (with RFC2833, inband, etc) over the years, I would avoid it if at all possible.  

What does it particularly do that can not accomplished by using RFC 2833 or (less ideal) inband DTMF?  Or are you attempting to use it as a band-aid to address some sort of interop issue with the carrier involved that is wrecking havoc with your particular application?

-metik
  ----- Original Message ----- 
  From: Tihomir Culjaga 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, October 13, 2009 3:24 PM
  Subject: Re: [Freeswitch-users] SIP Overlap support?


  i never found it working properly... i always had some interoperability issues and i finished having a "dialplan" on my phones being delivered through a config file via tftp or http .. depending of the phone capability.

  BTW: using overlap can lead to a greater system load... be careful when setting the minimum number of digits you will send in 1st message. I wish you luck...

  T.


  On Tue, Oct 13, 2009 at 9:03 PM, Metik <freeswitch-users-list at metik.com> wrote:

    Both support it.  In the Grandstream, I believe it is called Early Dial (vs. SNOM's Overlap Dialing).  It can be problematic if you have a device somewhere in the middle that doesn't support 484s.

    -metik
      ----- Original Message ----- 
      From: Anthony Minessale 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Tuesday, October 13, 2009 2:01 PM
      Subject: Re: [Freeswitch-users] SIP Overlap support?


      i do think some softphone can do it but i forgot which one it was either snom or grandstream



      On Tue, Oct 13, 2009 at 12:12 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

        you need a softswitch.... i'm afraid a SIP phone is not designed for overlap...

        T. 



        On Tue, Oct 13, 2009 at 5:26 PM, Dennis <odermann at googlemail.com> wrote:

          how could we try? we played arround with a snom phone (snom seems to
          support something in this direction, but are not shure, how we can
          test it and how we can see if it is supported or not.

          any hint?


          2009/10/13 Anthony Minessale <anthony.minessale at gmail.com>:

          > have you tried it?
          > I *think* either we did support it or we would with a small patch to sofia
          > lib that I cannot recall if we ever got committed.
          >
          >
          > On Tue, Oct 13, 2009 at 8:51 AM, Dennis <odermann at googlemail.com> wrote:
          >>
          >> hi there,
          >>
          >> i would like to ask, if fs has support for something like "SIP Overlap"?
          >>
          >> instead of receiving the phonenumber from our carrier in a block, we
          >> want to receive the phonenumber digit-by-digit and we want to tell fs
          >> when the number is complete. our carrier could send us the phonenumber
          >> digit-by-digit, but what about the fs-side?
          >>
          >>
          >> thanks and kind regards
          >> dennis
          >>
          >> _______________________________________________
          >> FreeSWITCH-users mailing list
          >> FreeSWITCH-users at lists.freeswitch.org
          >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
          >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
          >> http://www.freeswitch.org
          >
          >
          >
          > --
          > Anthony Minessale II
          >
          > FreeSWITCH http://www.freeswitch.org/
          > ClueCon http://www.cluecon.com/
          > Twitter: http://twitter.com/FreeSWITCH_wire
          >
          > AIM: anthm
          > MSN:anthony_minessale at hotmail.com
          > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
          > IRC: irc.freenode.net #freeswitch
          >
          > FreeSWITCH Developer Conference
          > sip:888 at conference.freeswitch.org
          > iax:guest at conference.freeswitch.org/888
          > googletalk:conf+888 at conference.freeswitch.org
          > pstn:213-799-1400
          >
          > _______________________________________________
          > FreeSWITCH-users mailing list
          > FreeSWITCH-users at lists.freeswitch.org
          > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
          > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
          > http://www.freeswitch.org
          >
          >

          _______________________________________________
          FreeSWITCH-users mailing list
          FreeSWITCH-users at lists.freeswitch.org
          http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
          UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
          http://www.freeswitch.org




        _______________________________________________
        FreeSWITCH-users mailing list
        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org





      -- 
      Anthony Minessale II

      FreeSWITCH http://www.freeswitch.org/
      ClueCon http://www.cluecon.com/
      Twitter: http://twitter.com/FreeSWITCH_wire

      AIM: anthm
      MSN:anthony_minessale at hotmail.com
      GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
      IRC: irc.freenode.net #freeswitch

      FreeSWITCH Developer Conference
      sip:888 at conference.freeswitch.org
      iax:guest at conference.freeswitch.org/888
      googletalk:conf+888 at conference.freeswitch.org
      pstn:213-799-1400



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From msc at freeswitch.org  Tue Oct 13 13:38:05 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Oct 2009 13:38:05 -0700
Subject: [Freeswitch-users] FS Extension Groups Documentation
In-Reply-To: <E8FDE7D1E87C4DDC98BD76DAF7DC50A5@greyhawk.tonecommander.com>
References: <E8FDE7D1E87C4DDC98BD76DAF7DC50A5@greyhawk.tonecommander.com>
Message-ID: <87f2f3b90910131338o2d161d1w397295efefd4138f@mail.gmail.com>

On Tue, Oct 13, 2009 at 7:54 AM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>
> Is there a link to the documentation of FS groups and what can be done with
> a group (i.e. capabilities)?
>
> Thanks And Best Regards,
> Jerry
>

http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups
http://wiki.freeswitch.org/wiki/Mod_commands#group_call

And this intercept example from the default.xml dialplan file:
    <extension name="group-intercept">
      <condition field="destination_number" expression="^\*8$">
    <action application="answer"/>
    <action application="intercept"
data="${hash(select/${domain_name}-last_dial/${callgroup})}"/>
    <action application="sleep" data="2000"/>
      </condition>
    </extension>

Note: groups in the dialplan and directory are not related to the mod_limit
group insert/delete/call stuff.
-MC
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From msc at freeswitch.org  Tue Oct 13 13:47:13 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Oct 2009 13:47:13 -0700
Subject: [Freeswitch-users] FreeSWITCH Conf Call Agenda For October 16th
Message-ID: <87f2f3b90910131347g23bd5f11p8e17a91374ac2a76@mail.gmail.com>

FYI,

The agenda for this week's call is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_16

Please feel free to add agenda items and updates as you see fit.

-Michael
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From diego.viola at gmail.com  Tue Oct 13 14:53:02 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 13 Oct 2009 21:53:02 +0000
Subject: [Freeswitch-users] Some help with my post-paid billing project
Message-ID: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>

Hello,

I'm trying to write a post-paid billing script, I have the CDR on my
database and also a "rates" table, the CDR contains fields like
caller_destination_number, variable_duration, etc. and the rates table
contains: destination, prefix, rate (cost).

The problem is that I can't just strip the destination number to take the
prefix from it because I have to deal with destination numbers from
different countries and they all have different prefix lengths... so I need
to find another way to take the prefix from the destination number.

Any ideas how to do this?

Thanks,

Diego
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From diego.viola at gmail.com  Tue Oct 13 14:58:47 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 13 Oct 2009 21:58:47 +0000
Subject: [Freeswitch-users] Some help with my post-paid billing project
In-Reply-To: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
References: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
Message-ID: <86a32abc0910131458i207c59d1x1c4f0dce4dfa4ccb@mail.gmail.com>

Should I try to pass the whole DNIS and compare with teh rates list/table
and when a prefix from the DNIS matches with the rates list I get hte rate
that way?


On Tue, Oct 13, 2009 at 9:53 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Hello,
>
> I'm trying to write a post-paid billing script, I have the CDR on my
> database and also a "rates" table, the CDR contains fields like
> caller_destination_number, variable_duration, etc. and the rates table
> contains: destination, prefix, rate (cost).
>
> The problem is that I can't just strip the destination number to take the
> prefix from it because I have to deal with destination numbers from
> different countries and they all have different prefix lengths... so I need
> to find another way to take the prefix from the destination number.
>
> Any ideas how to do this?
>
> Thanks,
>
> Diego
>
>
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From mcampbellsmith at gmail.com  Tue Oct 13 15:02:50 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Wed, 14 Oct 2009 09:02:50 +1100
Subject: [Freeswitch-users] Dingaling: using a hostname instead of stun for
	rtp
Message-ID: <33c87fa30910131502t2af24801i1c3319518b5db50b@mail.gmail.com>

Hi!

I have a hostname set in vars.conf.xml for the parameters
external_rtp_ip and the external_sip_ip instead of the usual stun.  I
found that stun was timing out and was causing some problems.  And as
I have a hostname, it makes sense to use that instead of relying on
stun.

However, when I use Dingaling, I see the following error message in
the dl_debug tracing:

<iq to="<FS_EMAIL>@gmail.com/gtalk77BBCB94" id="344" type="error"
from="<email1>@gmail.com/Talk.v10488E8134A">
  <ses:session type="candidates" id="3697758817"
initiator=""<email1>@gmail.com/Talk.v10488D9134A"
xmlns:ses="http://www.google.com/session">
    <ses:candidate name="rtp" address="host:<hostname>" port="30710"
username="nVv7FqwNyWbiMuTt" password="nVv7BqwQyVbiMuTt"
preference="1.0" protocol="udp" type="local" network="0"
generation="0"></ses:candidate>
  </ses:session>
  <error type="modify">
    <sta:bad-request
xmlns:sta="urn:ietf:params:xml:ns:xmpp-stanzas"></sta:bad-request>
    <sta:text xml:lang="en"
xmlns:sta="urn:ietf:params:xml:ns:xmpp-stanzas">candidate has address
of zero</sta:text>
  </error>
</iq>

Is Gtalk expecting an address in this field '<ses:candidate name="rtp"
address="host:<hostname>"' instead of a hostname?  Or should FS be
sending the hostname only and not host:hostname?

Thanks!


From grevenx at me.com  Tue Oct 13 15:45:11 2009
From: grevenx at me.com (=?iso-8859-1?Q?Even_Andr=E9_Fiskvik?=)
Date: Wed, 14 Oct 2009 00:45:11 +0200
Subject: [Freeswitch-users] Some help with my post-paid billing project
In-Reply-To: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
References: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
Message-ID: <74730C11-75D0-4BC0-9F63-8B6C9F0F499F@me.com>

What database are you using?
You could do this with regular SQL, but it would by a costly operation,
for PostgreSQL we're using the prefix module: http://pgfoundry.org/projects/prefix/

You can then match the closest prefix by using something like
"WHERE myprefix_col @> caller_destination_number ORDER BY LENGTH 
(myprefix_col::text) LIMIT 1;"


Best regards,
Even Andr?


On 13. okt. 2009, at 23.53, Diego Viola wrote:

> Hello,
>
> I'm trying to write a post-paid billing script, I have the CDR on my  
> database and also a "rates" table, the CDR contains fields like  
> caller_destination_number, variable_duration, etc. and the rates  
> table contains: destination, prefix, rate (cost).
>
> The problem is that I can't just strip the destination number to  
> take the prefix from it because I have to deal with destination  
> numbers from different countries and they all have different prefix  
> lengths... so I need to find another way to take the prefix from the  
> destination number.
>
> Any ideas how to do this?
>
> Thanks,
>
> Diego
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From kjv at ken-ton.com  Tue Oct 13 15:59:36 2009
From: kjv at ken-ton.com (Karl Vesterling)
Date: Tue, 13 Oct 2009 18:59:36 -0400
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <87f2f3b90910090005i51dbfc60t20dd77789872bc9e@mail.gmail.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<87f2f3b90910080958i1f5cfccax621fd12100a05e3e@mail.gmail.com>
	<903da5680910081201p427a8358n768edb61a96674a4@mail.gmail.com>
	<87f2f3b90910081538l6e4e4b7dv3aedb2c4d1af58bc@mail.gmail.com>
	<903da5680910082238x27bff476s8bea0e74ab516a8d@mail.gmail.com>
	<87f2f3b90910082308i7f6144d9kc887596be54dd609@mail.gmail.com>
	<903da5680910082341r485092vb0cd10c43c4a2534@mail.gmail.com>
	<87f2f3b90910090005i51dbfc60t20dd77789872bc9e@mail.gmail.com>
Message-ID: <BC5F7F95-767B-4A1D-95F5-FB393E31E6D3@ken-ton.com>


If that's the case, does this also apply to VoIP gateway services?

If so, I'll consider phrasing this differently:

http://wiki.freeswitch.org/wiki/Voicepulse.xml
--begin--
Voicepulse Warning

WARNING!!! Recent changes (Sep 2008) in VoicePulse have limited  
connect03 to only IAX termination. Please remove any entries that you  
may have reflecting the gateway below from your dialplan as it will  
most likely answer the call and play a message saying something to the  
effect of "Your call could not be completed as dialed because we  
changed our production environment in an effort to demonstrate what  
bad engineering is all about. BAD is an acronymn for Broken As  
Designed. Please hang up and make changes on all your switches. Even  
though we shouldn't have answered the call and simply ignored it  
entirely so that another gateway could facilitate your call, we FELT  
it wise to seriously inconvenience you by answering the call  
immediately giving no other gateway a chance to place the call. Thanks  
for using our service and we look forward to abusing you again in the  
near future.
UPDATE - Problem getting worse, not better 3/30/2009. I switched DID  
providers because of problems w/ Voicepulse, such as echo, dropped  
calls, and consistent registration problems hence calls don't get  
placed and inbound don't get routed to me. Incidentally, they will not  
port DID's, they'll tell you they will, but they don't.
--end--

Perhaps just saying, "stay away, far far away. Recommend this service  
only to people and/or organizations you don't like or are in direct  
competition with" would suffice eh?

Suggestions? Recommendations?

Best Regards,
Karl J. Vesterling
kjv at ken-ton.com
202-461-3231 x0

On Oct 9, 2009, at 3:05 AM, Michael Collins wrote:

>
>
> On Thu, Oct 8, 2009 at 11:41 PM, Gabriel Gunderson <gabe at gundy.org>  
> wrote:
> On Fri, Oct 9, 2009 at 12:08 AM, Michael Collins  
> <msc at freeswitch.org> wrote:
> > Thanks for reporting back. Please let all the Asterisk users know  
> that they
> > are welcome to join us in #freeswitch on irc.freenode.net and that  
> they will
> > not be abused like people do in other less friendly IRC channels.
>
> Funny you mention this.  Many people report that the way the FS
> community refers to Asterisk in docs/wikis/irc/whatever makes the FS
> camp seem *less* welcoming to them.  After all, they identify as
> Asterisk Users and take the criticism as being kinda harsh.  Most of
> them acknowledge the shortcomings of Asterisk but are put off when
> someone else points them out.  It's crazy, I know.  The thing is, I
> remember thinking that too.  After getting to know FS better, I didn't
> notice it as much.  Nobody likes to hear their baby is ugly --even if
> they know it is.
>
> Interesting. I'd like to know what specifically turned them off. It  
> could be node 117, but that was written by Anthony in response to  
> dozens of questions about the subject. In any case, if the Asterisk  
> guys look at the first edition of the Starfish/TFOT book they'll see  
> glowing praise of Anthony Minessale and Brian West, so maybe they'll  
> be willing to listen to those guys since they've got so much  
> Asterisk knowledge. The other thing is that many people in the  
> FreeSWITCH community are Asterisk refugees. They left Asterisk for  
> FreeSWITCH because they had some sort of trouble with Asterisk. If  
> that comes across in the docs then we'll have to see about cleaning  
> that up.
>
> At our session, and in general, I've noticed people are more
> interested in hearing about FS when you don't make direct comparisons
> to Asterisk.  Besides, FS stands on it's own merit.
>
> Agreed. Direct comparisons can always be slanted in any direction.  
> Besides, FreeSWITCH stacks up very well against a lot of modern  
> (i.e. expensive) equipment put out by the big boys (Cisco, Avaya,  
> NEC, ShoreTel, etc.).
>
> Just what I've observed *and* my 2 additional cents.
>
> Gabe
> Thanks,
> MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From rupa at rupa.com  Tue Oct 13 16:07:48 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 13 Oct 2009 18:07:48 -0500
Subject: [Freeswitch-users] Some help with my post-paid billing project
In-Reply-To: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
References: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
Message-ID: <b39273530910131607s2ad3306fp63e715e33a266c9c@mail.gmail.com>

Why not use mod_lcr to determine your rates?  It already does all this
work for you.  I even added a feature recently that allows one to
calculate/lookup end user rates (in addition to the carrier rate you
are paying for a given route).

On Tue, Oct 13, 2009 at 4:53 PM, Diego Viola <diego.viola at gmail.com> wrote:
> Hello,
>
> I'm trying to write a post-paid billing script, I have the CDR on my
> database and also a "rates" table, the CDR contains fields like
> caller_destination_number, variable_duration, etc. and the rates table
> contains: destination, prefix, rate (cost).
>
> The problem is that I can't just strip the destination number to take the
> prefix from it because I have to deal with destination numbers from
> different countries and they all have different prefix lengths... so I need
> to find another way to take the prefix from the destination number.
>
> Any ideas how to do this?
>
> Thanks,
>



-- 
-Rupa


From sjmudd at pobox.com  Tue Oct 13 13:51:14 2009
From: sjmudd at pobox.com (Simon J Mudd)
Date: Tue, 13 Oct 2009 22:51:14 +0200
Subject: [Freeswitch-users] Preannouncing a message before making a call
Message-ID: <20091013205114.GA4927@mad06.wl0.org>

I'm looking to migrate to Freeswitch from Asterisk (SOHO usage).
One thing I have configured at the moment which I'm unsure how to do in
FreeSwitch is the following.

When the call's dial pattern is recognised I've been getting Asterisk
to say: "Dialing <destination> via <gateway>" where <destination> might
be a country, or area and <gateway> may be the name of one of several
SIP gateways I have configured.  Each phrase has been a prerecorded wav
file played in sequence.

[macro-XXXXX]
exten => s,1,Playback(/etc/asterisk/wav/dialing)
exten => s,2,Playback(/etc/asterisk/wav/${ARG2})    # Name of destination
exten => s,3,Playback(/etc/asterisk/wav/via)
exten => s,4,Playback(/etc/asterisk/wav/XXXXX)      # Gateway name
exten => s,5,Dial(SIP/XXXXX/${ARG1},60,tT)          # ARG1 = Destination number


# actual usage
exten => _0044.,1,Macro(XXXXX,${EXTEN},touk)

How would I approach doing something like this in FreeSwtich?

Thanks for any pointers.

Simon


From sjmudd at pobox.com  Tue Oct 13 13:59:07 2009
From: sjmudd at pobox.com (Simon J Mudd)
Date: Tue, 13 Oct 2009 22:59:07 +0200
Subject: [Freeswitch-users] Building Freeswitch with VPATH (src and obj
	directories are different)?
Message-ID: <20091013205907.GA7545@mad06.wl0.org>

I was hoping to be able to build Freeswitch using VPATH, that is have the src and object directories in different locations.
This ensures that the src tree is unspoilt and I don't have to reclean it if I do updates.

However this does not seem to work:

[sjmudd at mad07 src]$ pwd
/Users/sjmudd/src
[sjmudd at mad07 src]$ ls -l | grep freeswitch
drwxr-xr-x   3 sjmudd  staff       102 13 Oct 07:52 freeswitch-obj
drwxr-xr-x  56 sjmudd  staff      1904 13 Oct 14:29 freeswitch-svn
[sjmudd at mad07 src]$ cd freeswitch-obj
[sjmudd at mad07 freeswitch-obj]$ ../freeswitch-svn/configure
checking for a BSD-compatible install... /usr/bin/install -c
checking whether build environment is sane... yes
checking for a thread-safe mkdir -p... ../freeswitch-svn/build/config/install-sh -c -d
checking for gawk... no
checking for mawk... no
checking for nawk... no
checking for awk... awk
checking whether make sets $(MAKE)... yes
configure: error: source directory already configured; run "make distclean" there first
[sjmudd at mad07 freeswitch-obj]$ cd ../freeswitch-svn
[sjmudd at mad07 freeswitch-svn]$ make distclean
Making clean in build
Making clean in src
Making clean in mod

making clean mod_amr

making clean mod_cdr_csv

making clean mod_cluechoo

making clean mod_commands

making clean mod_conference

making clean mod_console

making clean mod_dialplan_asterisk

making clean mod_dialplan_xml

making clean mod_dptools

making clean mod_enum

making clean mod_esf

making clean mod_event_socket

making clean mod_expr

making clean mod_fifo

making clean mod_file_string

making clean mod_fsv

making clean mod_g723_1

making clean mod_g729

making clean mod_h26x

making clean mod_iax

making clean mod_ilbc

making clean mod_limit

making clean mod_local_stream

making clean mod_logfile

making clean mod_loopback

making clean mod_lua

making clean mod_native_file

making clean mod_say_en

making clean mod_siren

making clean mod_sndfile

making clean mod_sofia
rm -f "./so_locations"

making clean mod_speex

making clean mod_spidermonkey

making clean mod_spidermonkey_core_db

making clean mod_spidermonkey_socket

making clean mod_spidermonkey_teletone

making clean mod_syslog

making clean mod_tone_stream

making clean mod_voicemail

making clean mod_voipcodecs

making clean mod_xml_cdr

making clean mod_xml_rpc

making clean mod_alsa

making clean mod_amrwb

making clean mod_celt

making clean mod_cepstral

making clean mod_dahdi_codec

making clean mod_dialplan_directory

making clean mod_dingaling

making clean mod_easyroute

making clean mod_erlang_event

making clean mod_event_multicast

making clean mod_fax

making clean mod_flite

WARNING mod_http is not a valid FreeSWITCH module dir, skipping it.....

making clean mod_lcr

making clean mod_ldap

making clean mod_memcache

making clean mod_nibblebill

making clean mod_opal
Package opal was not found in the pkg-config search path.
Perhaps you should add the directory containing `opal.pc'
to the PKG_CONFIG_PATH environment variable
No package 'opal' found
Package opal was not found in the pkg-config search path.
Perhaps you should add the directory containing `opal.pc'
to the PKG_CONFIG_PATH environment variable
No package 'opal' found

making clean mod_openzap
rm -f "./so_locations"
rm -f "./so_locations"
rm -f "./so_locations"
rm -f "./so_locations"
rm -f "./so_locations"
rm -f "./so_locations"
rm -f "./so_locations"
 rm -f testtones testtones
 rm -f detect_tones detect_tones
 rm -f detect_dtmf detect_dtmf
 rm -f testisdn testisdn
 rm -f testpri testpri
 rm -f testr2 testr2
 rm -f testboost testboost
 rm -f testanalog testanalog
 rm -f testapp testapp
 rm -f testcid testcid

making clean mod_perl

making clean mod_pocketsphinx

making clean mod_portaudio

making clean mod_python

making clean mod_radius_cdr

making clean mod_rss

making clean mod_say_de

making clean mod_say_es

making clean mod_say_fr

making clean mod_say_it

making clean mod_say_nl

making clean mod_say_ru

making clean mod_say_zh

making clean mod_shout

making clean mod_skypiax

making clean mod_snom

making clean mod_soundtouch

making clean mod_spidermonkey_odbc

making clean mod_spy

making clean mod_vmd

making clean mod_xml_curl

making clean mod_xml_ldap

making clean mod_yaml
Making clean in .
Making clean in .
 rm -f freeswitch freeswitch
 rm -f fs_cli fs_cli
 rm -f fs_ivrd fs_ivrd
rm -f "./so_locations"
[sjmudd at mad07 freeswitch-svn]$ cd ../freeswitch-obj
[sjmudd at mad07 freeswitch-obj]$ ../freeswitch-svn/configure
checking for a BSD-compatible install... /usr/bin/install -c
checking whether build environment is sane... yes
checking for a thread-safe mkdir -p... ../freeswitch-svn/build/config/install-sh -c -d
checking for gawk... no
checking for mawk... no
checking for nawk... no
checking for awk... awk
checking whether make sets $(MAKE)... yes
configure: error: source directory already configured; run "make distclean" there first
[sjmudd at mad07 freeswitch-obj]$ 

This is with 

[sjmudd at mad07 freeswitch-svn]$ svn info
Path: .
URL: http://svn.freeswitch.org/svn/freeswitch/trunk
Repository Root: http://svn.freeswitch.org/svn
Repository UUID: d0543943-73ff-0310-b7d9-9358b9ac24b2
Revision: 15138
Node Kind: directory
Schedule: normal
Last Changed Author: anthm
Last Changed Rev: 15138
Last Changed Date: 2009-10-12 16:58:09 +0200 (Mon, 12 Oct 2009)

Simon



From frank at carmickle.com  Tue Oct 13 16:32:20 2009
From: frank at carmickle.com (Frank Carmickle)
Date: Tue, 13 Oct 2009 19:32:20 -0400
Subject: [Freeswitch-users] Building Freeswitch with VPATH (src and
	obj	directories are different)?
In-Reply-To: <20091013205907.GA7545@mad06.wl0.org>
References: <20091013205907.GA7545@mad06.wl0.org>
Message-ID: <20091013233220.GM25698@base.carmickle.com>

On Tue, Oct 13, Simon J Mudd wrote:
> I was hoping to be able to build Freeswitch using VPATH, that is have the src and object directories in different locations.
> This ensures that the src tree is unspoilt and I don't have to reclean it if I do updates.

You can use svn export.  Not sure if this is what you want.

--FC


From msc at freeswitch.org  Tue Oct 13 17:09:46 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Oct 2009 17:09:46 -0700
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <BC5F7F95-767B-4A1D-95F5-FB393E31E6D3@ken-ton.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<87f2f3b90910080958i1f5cfccax621fd12100a05e3e@mail.gmail.com>
	<903da5680910081201p427a8358n768edb61a96674a4@mail.gmail.com>
	<87f2f3b90910081538l6e4e4b7dv3aedb2c4d1af58bc@mail.gmail.com>
	<903da5680910082238x27bff476s8bea0e74ab516a8d@mail.gmail.com>
	<87f2f3b90910082308i7f6144d9kc887596be54dd609@mail.gmail.com>
	<903da5680910082341r485092vb0cd10c43c4a2534@mail.gmail.com>
	<87f2f3b90910090005i51dbfc60t20dd77789872bc9e@mail.gmail.com>
	<BC5F7F95-767B-4A1D-95F5-FB393E31E6D3@ken-ton.com>
Message-ID: <87f2f3b90910131709v368ccee9o81cc2892a332f1a4@mail.gmail.com>

Karl,

Yeah, this post is wrong. You've got "acronym" misspelled. Once you fix that
then I think it's all good! ;)

Seriously, it's okay to put critical information into the wiki, but perhaps
you could remove the tongue-in-cheek comments.

-MC

On Tue, Oct 13, 2009 at 3:59 PM, Karl Vesterling <kjv at ken-ton.com> wrote:

>
> If that's the case, does this also apply to VoIP gateway services?
>
> If so, I'll consider phrasing this differently:
>
> http://wiki.freeswitch.org/wiki/Voicepulse.xml
> --begin--
> Voicepulse Warning*WARNING!!!* Recent changes (Sep 2008) in VoicePulse
> have limited connect03 to only IAX termination. Please remove any entries
> that you may have reflecting the gateway below from your dialplan as it will
> most likely answer the call and play a message saying something to the
> effect of "Your call could not be completed as dialed because we changed our
> production environment in an effort to demonstrate what bad engineering is
> all about. BAD is an acronymn for Broken As Designed. Please hang up and
> make changes on all your switches. Even though we shouldn't have answered
> the call and simply ignored it entirely so that another gateway could
> facilitate your call, we FELT it wise to seriously inconvenience you by
> answering the call immediately giving no other gateway a chance to place the
> call. Thanks for using our service and we look forward to abusing you again
> in the near future.
> *UPDATE* - Problem getting worse, not better *3/30/2009*. I switched DID
> providers because of problems w/ Voicepulse, such as echo, dropped calls,
> and consistent registration problems hence calls don't get placed and
> inbound don't get routed to me. Incidentally, they will not port DID's,
> they'll tell you they will, but they don't.
> --end--
>
> Perhaps just saying, "stay away, far far away. Recommend this service only
> to people and/or organizations you don't like or are in direct competition
> with" would suffice eh?
>
> Suggestions? Recommendations?
>
> Best Regards,
> Karl J. Vesterling
> kjv at ken-ton.com
> 202-461-3231 x0
>
> On Oct 9, 2009, at 3:05 AM, Michael Collins wrote:
>
>
>
> On Thu, Oct 8, 2009 at 11:41 PM, Gabriel Gunderson <gabe at gundy.org> wrote:
>
>> On Fri, Oct 9, 2009 at 12:08 AM, Michael Collins <msc at freeswitch.org>
>> wrote:
>> > Thanks for reporting back. Please let all the Asterisk users know that
>> they
>> > are welcome to join us in #freeswitch on irc.freenode.net and that they
>> will
>> > not be abused like people do in other less friendly IRC channels.
>>
>> Funny you mention this.  Many people report that the way the FS
>> community refers to Asterisk in docs/wikis/irc/whatever makes the FS
>> camp seem *less* welcoming to them.  After all, they identify as
>> Asterisk Users and take the criticism as being kinda harsh.  Most of
>> them acknowledge the shortcomings of Asterisk but are put off when
>> someone else points them out.  It's crazy, I know.  The thing is, I
>> remember thinking that too.  After getting to know FS better, I didn't
>> notice it as much.  Nobody likes to hear their baby is ugly --even if
>> they know it is.
>>
>
> Interesting. I'd like to know what specifically turned them off. It could
> be node 117, but that was written by Anthony in response to dozens of
> questions about the subject. In any case, if the Asterisk guys look at the
> first edition of the Starfish/TFOT book they'll see glowing praise of
> Anthony Minessale and Brian West, so maybe they'll be willing to listen to
> those guys since they've got so much Asterisk knowledge. The other thing is
> that many people in the FreeSWITCH community are Asterisk refugees. They
> left Asterisk for FreeSWITCH because they had some sort of trouble with
> Asterisk. If that comes across in the docs then we'll have to see about
> cleaning that up.
>
>>
>> At our session, and in general, I've noticed people are more
>> interested in hearing about FS when you don't make direct comparisons
>> to Asterisk.  Besides, FS stands on it's own merit.
>>
>
> Agreed. Direct comparisons can always be slanted in any direction. Besides,
> FreeSWITCH stacks up very well against a lot of modern (i.e. expensive)
> equipment put out by the big boys (Cisco, Avaya, NEC, ShoreTel, etc.).
>
>>
>> Just what I've observed *and* my 2 additional cents.
>>
>> Gabe
>>
> Thanks,
> MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Tue Oct 13 17:33:14 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 13 Oct 2009 17:33:14 -0700
Subject: [Freeswitch-users] Dingaling: using a hostname instead of stun
	for rtp
In-Reply-To: <33c87fa30910131502t2af24801i1c3319518b5db50b@mail.gmail.com>
References: <33c87fa30910131502t2af24801i1c3319518b5db50b@mail.gmail.com>
Message-ID: <C874E25D-6D73-49AA-8DD3-0A0022617FD2@freeswitch.org>

I don't think mod_dingaling will do a lookup for host: like sofia will  
as it doesn't have the code for that last I checked... I could be  
wrong but I don't recall it doing that.

/b

On Oct 13, 2009, at 3:02 PM, Mark Campbell-Smith wrote:

> I have a hostname set in vars.conf.xml for the parameters
> external_rtp_ip and the external_sip_ip instead of the usual stun.  I
> found that stun was timing out and was causing some problems.  And as
> I have a hostname, it makes sense to use that instead of relying on
> stun.



From brian at freeswitch.org  Tue Oct 13 17:36:16 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 13 Oct 2009 17:36:16 -0700
Subject: [Freeswitch-users] Building Freeswitch with VPATH (src and obj
	directories are different)?
In-Reply-To: <20091013205907.GA7545@mad06.wl0.org>
References: <20091013205907.GA7545@mad06.wl0.org>
Message-ID: <D8EB1A6C-38A6-4BC2-AD8C-46CF11993366@freeswitch.org>

You shouldn't have to make clean usually ... doing so might break your  
tree... You can usually get by with "make current" that will ensure  
the critical things are cleaned and built correctly... every now and  
they you'll hit a snafu but we'll usually tell you about it.

/b

On Oct 13, 2009, at 1:59 PM, Simon J Mudd wrote:

> I was hoping to be able to build Freeswitch using VPATH, that is  
> have the src and object directories in different locations.
> This ensures that the src tree is unspoilt and I don't have to  
> reclean it if I do updates.



From diego.viola at gmail.com  Tue Oct 13 18:36:04 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Wed, 14 Oct 2009 01:36:04 +0000
Subject: [Freeswitch-users] Some help with my post-paid billing project
In-Reply-To: <74730C11-75D0-4BC0-9F63-8B6C9F0F499F@me.com>
References: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
	<74730C11-75D0-4BC0-9F63-8B6C9F0F499F@me.com>
Message-ID: <86a32abc0910131836w5f113a16s247caf5eeb0f8fe4@mail.gmail.com>

I'm using MySQL now but I will try PostgreSQL with the prefix module, is
there a way to do that without the prefix module and with regular SQL?

Any examples?

Diego

On Tue, Oct 13, 2009 at 10:45 PM, Even Andr? Fiskvik <grevenx at me.com> wrote:

> What database are you using?
> You could do this with regular SQL, but it would by a costly operation,
> for PostgreSQL we're using the prefix module:
> http://pgfoundry.org/projects/prefix/
>
> You can then match the closest prefix by using something like
> "WHERE myprefix_col @> caller_destination_number ORDER BY LENGTH
> (myprefix_col::text) LIMIT 1;"
>
>
> Best regards,
> Even Andr?
>
>
> On 13. okt. 2009, at 23.53, Diego Viola wrote:
>
> > Hello,
> >
> > I'm trying to write a post-paid billing script, I have the CDR on my
> > database and also a "rates" table, the CDR contains fields like
> > caller_destination_number, variable_duration, etc. and the rates
> > table contains: destination, prefix, rate (cost).
> >
> > The problem is that I can't just strip the destination number to
> > take the prefix from it because I have to deal with destination
> > numbers from different countries and they all have different prefix
> > lengths... so I need to find another way to take the prefix from the
> > destination number.
> >
> > Any ideas how to do this?
> >
> > Thanks,
> >
> > Diego
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kjv at ken-ton.com  Tue Oct 13 18:38:47 2009
From: kjv at ken-ton.com (Karl Vesterling)
Date: Tue, 13 Oct 2009 21:38:47 -0400
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <87f2f3b90910131709v368ccee9o81cc2892a332f1a4@mail.gmail.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<87f2f3b90910080958i1f5cfccax621fd12100a05e3e@mail.gmail.com>
	<903da5680910081201p427a8358n768edb61a96674a4@mail.gmail.com>
	<87f2f3b90910081538l6e4e4b7dv3aedb2c4d1af58bc@mail.gmail.com>
	<903da5680910082238x27bff476s8bea0e74ab516a8d@mail.gmail.com>
	<87f2f3b90910082308i7f6144d9kc887596be54dd609@mail.gmail.com>
	<903da5680910082341r485092vb0cd10c43c4a2534@mail.gmail.com>
	<87f2f3b90910090005i51dbfc60t20dd77789872bc9e@mail.gmail.com>
	<BC5F7F95-767B-4A1D-95F5-FB393E31E6D3@ken-ton.com>
	<87f2f3b90910131709v368ccee9o81cc2892a332f1a4@mail.gmail.com>
Message-ID: <302A8ECD-A1C8-41FD-87D0-0DB7F61E5E76@ken-ton.com>

Michael;

Roger that.  I'll check my schedule to see when I might have some time  
available to make the change.
It seems as if I have some time available the tuesday after (bleep)  
freezes over.

Acronym corrected. ;-)

Best Regards,
Karl J. Vesterling
kjv at ken-ton.com
202-461-3231 x0

On Oct 13, 2009, at 8:09 PM, Michael Collins wrote:

> Karl,
>
> Yeah, this post is wrong. You've got "acronym" misspelled. Once you  
> fix that then I think it's all good! ;)
>
> Seriously, it's okay to put critical information into the wiki, but  
> perhaps you could remove the tongue-in-cheek comments.
>
> -MC
>
> On Tue, Oct 13, 2009 at 3:59 PM, Karl Vesterling <kjv at ken-ton.com>  
> wrote:
>
> If that's the case, does this also apply to VoIP gateway services?
>
> If so, I'll consider phrasing this differently:
>
> http://wiki.freeswitch.org/wiki/Voicepulse.xml
> --begin--
> Voicepulse Warning
>
> WARNING!!! Recent changes (Sep 2008) in VoicePulse have limited  
> connect03 to only IAX termination. Please remove any entries that  
> you may have reflecting the gateway below from your dialplan as it  
> will most likely answer the call and play a message saying something  
> to the effect of "Your call could not be completed as dialed because  
> we changed our production environment in an effort to demonstrate  
> what bad engineering is all about. BAD is an acronymn for Broken As  
> Designed. Please hang up and make changes on all your switches. Even  
> though we shouldn't have answered the call and simply ignored it  
> entirely so that another gateway could facilitate your call, we FELT  
> it wise to seriously inconvenience you by answering the call  
> immediately giving no other gateway a chance to place the call.  
> Thanks for using our service and we look forward to abusing you  
> again in the near future.
> UPDATE - Problem getting worse, not better 3/30/2009. I switched DID  
> providers because of problems w/ Voicepulse, such as echo, dropped  
> calls, and consistent registration problems hence calls don't get  
> placed and inbound don't get routed to me. Incidentally, they will  
> not port DID's, they'll tell you they will, but they don't.
> --end--
>
> Perhaps just saying, "stay away, far far away. Recommend this  
> service only to people and/or organizations you don't like or are in  
> direct competition with" would suffice eh?
>
> Suggestions? Recommendations?
>
> Best Regards,
> Karl J. Vesterling
> kjv at ken-ton.com
> 202-461-3231 x0
>
> On Oct 9, 2009, at 3:05 AM, Michael Collins wrote:
>
>>
>>
>> On Thu, Oct 8, 2009 at 11:41 PM, Gabriel Gunderson <gabe at gundy.org>  
>> wrote:
>> On Fri, Oct 9, 2009 at 12:08 AM, Michael Collins  
>> <msc at freeswitch.org> wrote:
>> > Thanks for reporting back. Please let all the Asterisk users know  
>> that they
>> > are welcome to join us in #freeswitch on irc.freenode.net and  
>> that they will
>> > not be abused like people do in other less friendly IRC channels.
>>
>> Funny you mention this.  Many people report that the way the FS
>> community refers to Asterisk in docs/wikis/irc/whatever makes the FS
>> camp seem *less* welcoming to them.  After all, they identify as
>> Asterisk Users and take the criticism as being kinda harsh.  Most of
>> them acknowledge the shortcomings of Asterisk but are put off when
>> someone else points them out.  It's crazy, I know.  The thing is, I
>> remember thinking that too.  After getting to know FS better, I  
>> didn't
>> notice it as much.  Nobody likes to hear their baby is ugly --even if
>> they know it is.
>>
>> Interesting. I'd like to know what specifically turned them off. It  
>> could be node 117, but that was written by Anthony in response to  
>> dozens of questions about the subject. In any case, if the Asterisk  
>> guys look at the first edition of the Starfish/TFOT book they'll  
>> see glowing praise of Anthony Minessale and Brian West, so maybe  
>> they'll be willing to listen to those guys since they've got so  
>> much Asterisk knowledge. The other thing is that many people in the  
>> FreeSWITCH community are Asterisk refugees. They left Asterisk for  
>> FreeSWITCH because they had some sort of trouble with Asterisk. If  
>> that comes across in the docs then we'll have to see about cleaning  
>> that up.
>>
>> At our session, and in general, I've noticed people are more
>> interested in hearing about FS when you don't make direct comparisons
>> to Asterisk.  Besides, FS stands on it's own merit.
>>
>> Agreed. Direct comparisons can always be slanted in any direction.  
>> Besides, FreeSWITCH stacks up very well against a lot of modern  
>> (i.e. expensive) equipment put out by the big boys (Cisco, Avaya,  
>> NEC, ShoreTel, etc.).
>>
>> Just what I've observed *and* my 2 additional cents.
>>
>> Gabe
>> Thanks,
>> MC
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From diego.viola at gmail.com  Tue Oct 13 18:53:54 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Wed, 14 Oct 2009 01:53:54 +0000
Subject: [Freeswitch-users] Some help with my post-paid billing project
In-Reply-To: <86a32abc0910131836w5f113a16s247caf5eeb0f8fe4@mail.gmail.com>
References: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
	<74730C11-75D0-4BC0-9F63-8B6C9F0F499F@me.com>
	<86a32abc0910131836w5f113a16s247caf5eeb0f8fe4@mail.gmail.com>
Message-ID: <86a32abc0910131853v244ed61bof9c24f345a0944b2@mail.gmail.com>

Wrong question.

Is there a way to compare numbers with prefixes without using the prefix
module?

Diego

On Wed, Oct 14, 2009 at 1:36 AM, Diego Viola <diego.viola at gmail.com> wrote:

> I'm using MySQL now but I will try PostgreSQL with the prefix module, is
> there a way to do that without the prefix module and with regular SQL?
>
> Any examples?
>
> Diego
>
>
> On Tue, Oct 13, 2009 at 10:45 PM, Even Andr? Fiskvik <grevenx at me.com>wrote:
>
>> What database are you using?
>> You could do this with regular SQL, but it would by a costly operation,
>> for PostgreSQL we're using the prefix module:
>> http://pgfoundry.org/projects/prefix/
>>
>> You can then match the closest prefix by using something like
>> "WHERE myprefix_col @> caller_destination_number ORDER BY LENGTH
>> (myprefix_col::text) LIMIT 1;"
>>
>>
>> Best regards,
>> Even Andr?
>>
>>
>> On 13. okt. 2009, at 23.53, Diego Viola wrote:
>>
>> > Hello,
>> >
>> > I'm trying to write a post-paid billing script, I have the CDR on my
>> > database and also a "rates" table, the CDR contains fields like
>> > caller_destination_number, variable_duration, etc. and the rates
>> > table contains: destination, prefix, rate (cost).
>> >
>> > The problem is that I can't just strip the destination number to
>> > take the prefix from it because I have to deal with destination
>> > numbers from different countries and they all have different prefix
>> > lengths... so I need to find another way to take the prefix from the
>> > destination number.
>> >
>> > Any ideas how to do this?
>> >
>> > Thanks,
>> >
>> > Diego
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> > users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From technical at ttnc.co.uk  Tue Oct 13 19:07:11 2009
From: technical at ttnc.co.uk (TTNC - Adnan Barakat)
Date: Wed, 14 Oct 2009 03:07:11 +0100
Subject: [Freeswitch-users] Some help with my post-paid billing project
In-Reply-To: <86a32abc0910131836w5f113a16s247caf5eeb0f8fe4@mail.gmail.com>
References: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>	<74730C11-75D0-4BC0-9F63-8B6C9F0F499F@me.com>
	<86a32abc0910131836w5f113a16s247caf5eeb0f8fe4@mail.gmail.com>
Message-ID: <4AD5324F.2030000@ttnc.co.uk>

Diego Viola wrote:
> I'm using MySQL now but I will try PostgreSQL with the prefix module, is
> there a way to do that without the prefix module and with regular SQL?
> 
> Any examples?
SELECT * FROM rates WHERE prefix = SUBSTRING('$NUMBER$', 1,
LENGTH(prefix)) LIMIT 1

Adnan

> Diego
> 
> On Tue, Oct 13, 2009 at 10:45 PM, Even Andr? Fiskvik <grevenx at me.com
> <mailto:grevenx at me.com>> wrote:
> 
>     What database are you using?
>     You could do this with regular SQL, but it would by a costly operation,
>     for PostgreSQL we're using the prefix module:
>     http://pgfoundry.org/projects/prefix/
> 
>     You can then match the closest prefix by using something like
>     "WHERE myprefix_col @> caller_destination_number ORDER BY LENGTH
>     (myprefix_col::text) LIMIT 1;"
> 
> 
>     Best regards,
>     Even Andr?
> 
> 
>     On 13. okt. 2009, at 23.53, Diego Viola wrote:
> 
>     > Hello,
>     >
>     > I'm trying to write a post-paid billing script, I have the CDR on my
>     > database and also a "rates" table, the CDR contains fields like
>     > caller_destination_number, variable_duration, etc. and the rates
>     > table contains: destination, prefix, rate (cost).
>     >
>     > The problem is that I can't just strip the destination number to
>     > take the prefix from it because I have to deal with destination
>     > numbers from different countries and they all have different prefix
>     > lengths... so I need to find another way to take the prefix from the
>     > destination number.
>     >
>     > Any ideas how to do this?
>     >
>     > Thanks,
>     >
>     > Diego
>     >
>     > _______________________________________________
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>     > users
>     > http://www.freeswitch.org
> 
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From dome at tel.co.th  Tue Oct 13 19:39:13 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Wed, 14 Oct 2009 09:39:13 +0700
Subject: [Freeswitch-users] T.38 in sip header
Message-ID: <8ccbff060910131939g75b12346x2e9156fe17b61aa6@mail.gmail.com>

Dear all,

             Is posible to check T.38 in sip header? i want to
disconnect call when T.38 not support in gateway and try next gateway

BG

Dome C.


From red.rain.seven at gmail.com  Tue Oct 13 20:30:27 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Wed, 14 Oct 2009 11:30:27 +0800
Subject: [Freeswitch-users] sched_api doesn't get launched
In-Reply-To: <86a32abc0910131145t730d0bc0m842ed7ed0891d425@mail.gmail.com>
References: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
	<86a32abc0910131145t730d0bc0m842ed7ed0891d425@mail.gmail.com>
Message-ID: <59ad9ca10910132030l1fb55edbvb9fc4107dd15f61c@mail.gmail.com>

Diego:
You probably miss understood me. I said I was able to make "sched_hangup"
work, but not the "sched_api" in the same way I script for "sched_hangup"

The problem was on the second paragraph.

thanks,

On Wed, Oct 14, 2009 at 2:45 AM, Diego Viola <diego.viola at gmail.com> wrote:

> You need to pass the UUID to sched_hangup.
>
> Usage: sched_hangup [+]<time> <uuid> [<cause>]
>
> http://wiki.freeswitch.org/wiki/Mod_commands#sched_hangup
>
>
> On Tue, Oct 13, 2009 at 10:14 AM, Henry Huang <b_ball_henry at hotmail.com>wrote:
>
>> Hi:
>> I am using mod_java. And in my script I was able to achieve using:
>>
>> execute("sched_hangup", "+300 alloted_timeout");
>>
>> However, when I try to run sched_api in the same way, system log returns
>> that it's an invalid application. I have also tried to trigger it with many
>> conditional channel variable api calls , but non of them seemed to execute
>> the api command (because I turned on the highest level of debugging and see
>> no where the sched_api is being called.
>>
>> The closest thing I got was by using "api_after_bridge" like the
>> following, but it only launches when the bridge is teared down(which is not
>> what I want). I originally thought after bridge means right after the 2
>> party is connected. All I want is to be able to play some message to leg A
>> at certain time.
>>
>> setVariable("api_after_bridge", "sched_api +10 none uuid_displace ${uuid}
>> start /path/to/some.wav 20 mux");
>>
>> I have been struggling with different combination for a week now.. Please
>> shed some light if you know something.
>>
>> Thanks,
>>
>> --
>> Henry Huang
>> UniC Solution - Communication Unified
>> VoIP & Open Source software Consultant
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From red.rain.seven at gmail.com  Tue Oct 13 20:33:23 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Wed, 14 Oct 2009 11:33:23 +0800
Subject: [Freeswitch-users] Some help with my post-paid billing project
In-Reply-To: <4AD5324F.2030000@ttnc.co.uk>
References: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
	<74730C11-75D0-4BC0-9F63-8B6C9F0F499F@me.com>
	<86a32abc0910131836w5f113a16s247caf5eeb0f8fe4@mail.gmail.com>
	<4AD5324F.2030000@ttnc.co.uk>
Message-ID: <59ad9ca10910132033k28fb7f8ckdd5ba1f772ba2a7e@mail.gmail.com>

Diego:
Didn't wrote a prepaid system already? how did you compare the prefix in
that system then?

On Wed, Oct 14, 2009 at 10:07 AM, TTNC - Adnan Barakat <technical at ttnc.co.uk
> wrote:

> Diego Viola wrote:
> > I'm using MySQL now but I will try PostgreSQL with the prefix module, is
> > there a way to do that without the prefix module and with regular SQL?
> >
> > Any examples?
> SELECT * FROM rates WHERE prefix = SUBSTRING('$NUMBER$', 1,
> LENGTH(prefix)) LIMIT 1
>
> Adnan
>
> > Diego
> >
> > On Tue, Oct 13, 2009 at 10:45 PM, Even Andr? Fiskvik <grevenx at me.com
> > <mailto:grevenx at me.com>> wrote:
> >
> >     What database are you using?
> >     You could do this with regular SQL, but it would by a costly
> operation,
> >     for PostgreSQL we're using the prefix module:
> >     http://pgfoundry.org/projects/prefix/
> >
> >     You can then match the closest prefix by using something like
> >     "WHERE myprefix_col @> caller_destination_number ORDER BY LENGTH
> >     (myprefix_col::text) LIMIT 1;"
> >
> >
> >     Best regards,
> >     Even Andr?
> >
> >
> >     On 13. okt. 2009, at 23.53, Diego Viola wrote:
> >
> >     > Hello,
> >     >
> >     > I'm trying to write a post-paid billing script, I have the CDR on
> my
> >     > database and also a "rates" table, the CDR contains fields like
> >     > caller_destination_number, variable_duration, etc. and the rates
> >     > table contains: destination, prefix, rate (cost).
> >     >
> >     > The problem is that I can't just strip the destination number to
> >     > take the prefix from it because I have to deal with destination
> >     > numbers from different countries and they all have different prefix
> >     > lengths... so I need to find another way to take the prefix from
> the
> >     > destination number.
> >     >
> >     > Any ideas how to do this?
> >     >
> >     > Thanks,
> >     >
> >     > Diego
> >     >
> >     > _______________________________________________
> >     > FreeSWITCH-users mailing list
> >     > FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-
> >     > users
> >     > http://www.freeswitch.org
> >
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From msc at freeswitch.org  Tue Oct 13 21:16:07 2009
From: msc at freeswitch.org (Michael S Collins)
Date: Tue, 13 Oct 2009 21:16:07 -0700
Subject: [Freeswitch-users] sched_api doesn't get launched
In-Reply-To: <59ad9ca10910132030l1fb55edbvb9fc4107dd15f61c@mail.gmail.com>
References: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
	<86a32abc0910131145t730d0bc0m842ed7ed0891d425@mail.gmail.com>
	<59ad9ca10910132030l1fb55edbvb9fc4107dd15f61c@mail.gmail.com>
Message-ID: <31EC1285-3EF7-4B9F-A39D-CA3E8DB34A46@freeswitch.org>


On Oct 13, 2009, at 8:30 PM, Henry Huang <red.rain.seven at gmail.com>  
wrote:

> Diego:
>
> You probably miss understood me. I said I was able to make  
> "sched_hangup" work, but not the "sched_api" in the same way I  
> script for "sched_hangup"
>
> The problem was on the second paragraph.

Henry, can you capture the debug output of a call and put it in  
pastebin?That should yield a clue as to what is going on.

-MC


From brian at freeswitch.org  Tue Oct 13 21:15:25 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 13 Oct 2009 21:15:25 -0700
Subject: [Freeswitch-users] T.38 in sip header
In-Reply-To: <8ccbff060910131939g75b12346x2e9156fe17b61aa6@mail.gmail.com>
References: <8ccbff060910131939g75b12346x2e9156fe17b61aa6@mail.gmail.com>
Message-ID: <75A104CA-BE97-4912-BA7D-F091A6759EFC@freeswitch.org>

Well if you look at the SDP you can look for "image" usually but thats  
not something I think you can see PRE call setup.

/b

On Oct 13, 2009, at 7:39 PM, Dome Charoenyost wrote:

> Dear all,
>
>             Is posible to check T.38 in sip header? i want to
> disconnect call when T.38 not support in gateway and try next gateway
>
> BG
>
> Dome C.



From mgg at giagnocavo.net  Tue Oct 13 21:15:22 2009
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Wed, 14 Oct 2009 00:15:22 -0400
Subject: [Freeswitch-users] Some help with my post-paid billing project
In-Reply-To: <86a32abc0910131853v244ed61bof9c24f345a0944b2@mail.gmail.com>
References: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
	<74730C11-75D0-4BC0-9F63-8B6C9F0F499F@me.com>
	<86a32abc0910131836w5f113a16s247caf5eeb0f8fe4@mail.gmail.com>
	<86a32abc0910131853v244ed61bof9c24f345a0944b2@mail.gmail.com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C6703241D1E52@mse17be1.mse17.exchange.ms>

In our testing with SQL Server, we found that executing several queries for direct matches yielded far better performance than one query trying to check prefixes. (The column was also part of the clustered index, but AFAIK MySQL doesn't support defining your own clustered indexes; you get the PK always.)

-Michael

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Diego Viola
Sent: Tuesday, October 13, 2009 7:54 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Some help with my post-paid billing project

Wrong question.

Is there a way to compare numbers with prefixes without using the prefix module?

Diego
On Wed, Oct 14, 2009 at 1:36 AM, Diego Viola <diego.viola at gmail.com<mailto:diego.viola at gmail.com>> wrote:
I'm using MySQL now but I will try PostgreSQL with the prefix module, is there a way to do that without the prefix module and with regular SQL?

Any examples?

Diego

On Tue, Oct 13, 2009 at 10:45 PM, Even Andr? Fiskvik <grevenx at me.com<mailto:grevenx at me.com>> wrote:
What database are you using?
You could do this with regular SQL, but it would by a costly operation,
for PostgreSQL we're using the prefix module: http://pgfoundry.org/projects/prefix/

You can then match the closest prefix by using something like
"WHERE myprefix_col @> caller_destination_number ORDER BY LENGTH
(myprefix_col::text) LIMIT 1;"


Best regards,
Even Andr?


On 13. okt. 2009, at 23.53, Diego Viola wrote:

> Hello,
>
> I'm trying to write a post-paid billing script, I have the CDR on my
> database and also a "rates" table, the CDR contains fields like
> caller_destination_number, variable_duration, etc. and the rates
> table contains: destination, prefix, rate (cost).
>
> The problem is that I can't just strip the destination number to
> take the prefix from it because I have to deal with destination
> numbers from different countries and they all have different prefix
> lengths... so I need to find another way to take the prefix from the
> destination number.
>
> Any ideas how to do this?
>
> Thanks,
>
> Diego
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> users
> http://www.freeswitch.org


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


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From lakindia89 at gmail.com  Tue Oct 13 21:29:30 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Wed, 14 Oct 2009 09:59:30 +0530
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <5E7A8451-6264-4822-8152-1B94240BDD48@freeswitch.org>
References: <7d79b3930910090407m5dc81d12pf2b0a47f6017804@mail.gmail.com>
	<20091009123615.BC50A3FA6E9@mail.cune.org>
	<7d79b3930910100647u7fd11413ucec0a535d68b3e3a@mail.gmail.com>
	<87f2f3b90910101009j1295d812jd98cae91e30b1eff@mail.gmail.com>
	<7d79b3930910120454m1ffc6dd4u42b91a4349597166@mail.gmail.com>
	<87f2f3b90910121149x6e2114f0q996a262ad272a1f6@mail.gmail.com>
	<7d79b3930910130645y658ae041v8219c4c881fa5698@mail.gmail.com>
	<5E7A8451-6264-4822-8152-1B94240BDD48@freeswitch.org>
Message-ID: <7d79b3930910132129t30e25479j12cb596b2ce0a57d@mail.gmail.com>

But that setup is not there right now. It was an year back they used that.
I'll try to make those setup and send you the log. I think it may take some
2 days time, and I'm not very familiar with asterisk.

In the mean time, I request you to kindly go-through freeswitch debug log.
You may still get some clue.


On Tue, Oct 13, 2009 at 9:16 PM, Michael S Collins <msc at freeswitch.org>wrote:

>
> On Oct 13, 2009, at 6:45 AM, lakshmanan ganapathy <lakindia89 at gmail.com>
> wrote:
>
> We are using Reliance as the Carrier.
> I think, with this same Reliance carrier, in my office, they are able to
> make outgoing calls through asterisk+libpri.
>
>
> If that's the case I would be very interested in seeing a pri debug from a
> working call on an asterisk box. It might give us a clue as to what is not
> working. -MC
>
>
>
> On Tue, Oct 13, 2009 at 12:19 AM, Michael Collins < <msc at freeswitch.org>
> msc at freeswitch.org> wrote:
>
>> Lak,
>>
>> Okay I will need a little bit of time to dig into the IE's and what they
>> contain. In the meantime can you tell me who the carrier is? I'd like to
>> find out if they have some specific requirements. The fact that it doesn't
>> work with libpri surprises me because that would mean that Asterisk systems
>> would probably not work with this carrier as well.
>>
>> BTW, thanks for the very complete pastebin entries. :)
>> -MC
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
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From lakindia89 at gmail.com  Tue Oct 13 22:34:49 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Wed, 14 Oct 2009 11:04:49 +0530
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <5E7A8451-6264-4822-8152-1B94240BDD48@freeswitch.org>
References: <7d79b3930910090407m5dc81d12pf2b0a47f6017804@mail.gmail.com>
	<20091009123615.BC50A3FA6E9@mail.cune.org>
	<7d79b3930910100647u7fd11413ucec0a535d68b3e3a@mail.gmail.com>
	<87f2f3b90910101009j1295d812jd98cae91e30b1eff@mail.gmail.com>
	<7d79b3930910120454m1ffc6dd4u42b91a4349597166@mail.gmail.com>
	<87f2f3b90910121149x6e2114f0q996a262ad272a1f6@mail.gmail.com>
	<7d79b3930910130645y658ae041v8219c4c881fa5698@mail.gmail.com>
	<5E7A8451-6264-4822-8152-1B94240BDD48@freeswitch.org>
Message-ID: <7d79b3930910132234k4b99c794q8854ce20d14ee74e@mail.gmail.com>

Hi. I made the setup quickly with one of my colleague, and here is the debug
log given by asterisk

Steps done by me :
	+ Call Made from 04443902743 to 04439114600
	+ When call lands to 04439114600, I just route it to 09176454982.

*CLI>
*CLI>
*CLI> pri debug span 1
Enabled debugging on span 1
*CLI>
*CLI>
*CLI>
*CLI> < Protocol Discriminator: Q.931 (8)  len=41
< Call Ref: len= 2 (reference 21/0x15) (Originator)
< Message type: SETUP (5)
< [a1]
< Sending Complete (len= 1)
< [04 03 80 90 a3]
< Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer
capability: Speech (0)
<                              Ext: 1  Trans mode/rate: 64kbps,
circuit-mode (16)
<                                User information layer 1: A-Law (35)
< [18 03 a1 83 9f]
< Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0
Preferred  Dchan: 0
<                        ChanSel: As indicated in following octets
<                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
<                       Ext: 1  Channel: 31 ]
< [6c 0c 01 81 34 34 34 33 39 30 32 37 34 33]
< Calling Number (len=14) [ Ext: 0  TON: Unknown Number Type (0)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
<                           Presentation: Presentation permitted, user
number passed network screening (1)  '4443902743' ]
< [70 09 c1 33 39 31 31 34 36 30 30]
< Called Number (len=11) [ Ext: 1  TON: Subscriber Number (4)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)  '39114600' ]
-- Making new call for cr 21
-- Processing Q.931 Call Setup
-- Processing IE 161 (cs0, Sending Complete)
-- Processing IE 4 (cs0, Bearer Capability)
-- Processing IE 24 (cs0, Channel Identification)
-- Processing IE 108 (cs0, Calling Party Number)
-- Processing IE 112 (cs0, Called Party Number)
q931.c:3504 q931_receive: call 21 on channel 31 enters state 6 (Call Present)
q931.c:2769 q931_call_proceeding: call 21 on channel 31 enters state 9
(Incoming Call Proceeding)

> Protocol Discriminator: Q.931 (8)  len=10
> Call Ref: len= 2 (reference 21/0x15) (Terminator)
> Message type: CALL PROCEEDING (2)
> [18 03 a9 83 9f]
> Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  Dchan: 0
>                        ChanSel: As indicated in following octets
>                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
>                       Ext: 1  Channel: 31 ]

    -- Accepting call from '4443902743' to '39114600' on channel 0/31, span 1
    -- Executing [39114600 at incoming-from-pri:1] NoOp("Zap/31-1", "FMS
Call Extn:39114600:| CID:: <cid::>| Channel:Zap/31-1:| ") in new stack
    -- Executing [39114600 at incoming-from-pri:2] Dial("Zap/31-1",
"Zap/g1/w09176454982||rj") in new stack
-- Making new call for cr 32770
    -- Requested transfer capability: 0x00 - SPEECH

> Protocol Discriminator: Q.931 (8)  len=43
> Call Ref: len= 2 (reference 2/0x2) (Originator)
> Message type: SETUP (5)
> [04 03 80 90 a3]
> Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer capability: Speech (0)
>                              Ext: 1  Trans mode/rate: 64kbps, circuit-mode (16)
>                                User information layer 1: A-Law (35)
> [18 03 a9 83 81]
> Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  Dchan: 0
>                        ChanSel: As indicated in following octets
>                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
>                       Ext: 1  Channel: 1 ]
> [6c 0c 00 81 34 34 34 33 39 30 32 37 34 33]
> Calling Number (len=14) [ Ext: 0  TON: Unknown Number Type (0)  NPI: Unknown Number Plan (0)
>                           Presentation: Presentation permitted, user number passed network screening (1)  '4443902743' ]
> [70 0c 80 30 39 31 37 36 34 35 34 39 38 32]
> Called Number (len=14) [ Ext: 1  TON: Unknown Number Type (0)  NPI: Unknown Number Plan (0)  '09176454982' ]

q931.c:3087 q931_setup: call 32770 on channel 1 enters state 1 (Call Initiated)
    -- Called g1/w09176454982
q931.c:2797 q931_alerting: call 21 on channel 31 enters state 7 (Call Received)

> Protocol Discriminator: Q.931 (8)  len=9
> Call Ref: len= 2 (reference 21/0x15) (Terminator)
> Message type: ALERTING (1)
> [1e 02 81 88]
> Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Private network serving the local user (1)
>                               Ext: 1  Progress Description: Inband information or appropriate pattern now available. (8) ]

< Protocol Discriminator: Q.931 (8)  len=10
< Call Ref: len= 2 (reference 2/0x2) (Terminator)
< Message type: CALL PROCEEDING (2)
< [18 03 a9 83 81]
< Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0
Exclusive  Dchan: 0
<                        ChanSel: As indicated in following octets
<                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
<                       Ext: 1  Channel: 1 ]
-- Processing IE 24 (cs0, Channel Identification)
q931.c:3636 q931_receive: call 32770 on channel 1 enters state 3
(Outgoing call  Proceeding)
    -- Zap/1-1 is proceeding passing it to Zap/31-1
< Protocol Discriminator: Q.931 (8)  len=13
< Call Ref: len= 2 (reference 2/0x2) (Terminator)
< Message type: ALERTING (1)
< [1e 02 84 88]
< Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard
(0)  0: 0  Location: Public network serving the remote user (4)
<                               Ext: 1  Progress Description: Inband
information or appropriate pattern now available. (8) ]
< [1e 02 84 82]
< Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard
(0)  0: 0  Location: Public network serving the remote user (4)
<                               Ext: 1  Progress Description: Called
equipment is non-ISDN. (2) ]
-- Processing IE 30 (cs0, Progress Indicator)
-- Processing IE 30 (cs0, Progress Indicator)
q931.c:3549 q931_receive: call 32770 on channel 1 enters state 4 (Call
Delivered)
    -- Zap/1-1 is ringing
< Protocol Discriminator: Q.931 (8)  len=9
< Call Ref: len= 2 (reference 2/0x2) (Terminator)
< Message type: PROGRESS (3)
< [1e 02 84 84]
< Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard
(0)  0: 0  Location: Public network serving the remote user (4)
<                               Ext: 1  Progress Description: Unknown (4) ]
-- Processing IE 30 (cs0, Progress Indicator)
    -- Zap/1-1 is making progress passing it to Zap/31-1
< Protocol Discriminator: Q.931 (8)  len=9
< Call Ref: len= 2 (reference 2/0x2) (Terminator)
< Message type: PROGRESS (3)
< [1e 02 84 88]
< Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard
(0)  0: 0  Location: Public network serving the remote user (4)
<                               Ext: 1  Progress Description: Inband
information or appropriate pattern now available. (8) ]
-- Processing IE 30 (cs0, Progress Indicator)
    -- Zap/1-1 is making progress passing it to Zap/31-1
< Protocol Discriminator: Q.931 (8)  len=19
< Call Ref: len= 2 (reference 2/0x2) (Terminator)
< Message type: CONNECT (7)
< [4c 0c 01 83 39 31 37 36 34 35 34 39 38 32]
< Connected Number (len=14) [ Ext: 0  TON: Unknown Number Type (0)
NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
<                             Ext: 1  Presentation: Presentation
allowed of network provided number (3)  '9176454982' ]
-- Processing IE 76 (cs0, Connected Number)
q931.c:3579 q931_receive: call 32770 on channel 1 enters state 10 (Active)

> Protocol Discriminator: Q.931 (8)  len=5
> Call Ref: len= 2 (reference 2/0x2) (Originator)
> Message type: CONNECT ACKNOWLEDGE (15)

    -- Zap/1-1 answered Zap/31-1
q931.c:2904 q931_connect: call 21 on channel 31 enters state 8 (Connect Request)

> Protocol Discriminator: Q.931 (8)  len=14
> Call Ref: len= 2 (reference 21/0x15) (Terminator)
> Message type: CONNECT (7)
> [18 03 a9 83 9f]
> Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  Dchan: 0
>                        ChanSel: As indicated in following octets
>                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
>                       Ext: 1  Channel: 31 ]
> [1e 02 81 82]
> Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Private network serving the local user (1)
>                               Ext: 1  Progress Description: Called equipment is non-ISDN. (2) ]

    -- Native bridging Zap/31-1 and Zap/1-1
< Protocol Discriminator: Q.931 (8)  len=5
< Call Ref: len= 2 (reference 21/0x15) (Originator)
< Message type: CONNECT ACKNOWLEDGE (15)
q931.c:3664 q931_receive: call 21 on channel 31 enters state 10 (Active)
< Protocol Discriminator: Q.931 (8)  len=9
< Call Ref: len= 2 (reference 2/0x2) (Terminator)
< Message type: DISCONNECT (69)
< [08 02 80 90]
< Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0
Location: User (0)
<                  Ext: 1  Cause: Normal Clearing (16), class = Normal
Event (1) ]
-- Processing IE 8 (cs0, Cause)
q931.c:3779 q931_receive: call 32770 on channel 1 enters state 12
(Disconnect Indication)
    -- Channel 0/1, span 1 got hangup request, cause 16
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication,
peerstate Disconnect Request
q931.c:2920 q931_release: call 32770 on channel 1 enters state 19
(Release Request)

> Protocol Discriminator: Q.931 (8)  len=9
> Call Ref: len= 2 (reference 2/0x2) (Originator)
> Message type: RELEASE (77)
> [08 02 81 90]
> Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  Location: Private network serving the local user (1)
>                  Ext: 1  Cause: Normal Clearing (16), class = Normal Event (1) ]

    -- Hungup 'Zap/1-1'
  == Spawn extension (incoming-from-pri, 39114600, 2) exited non-zero
on 'Zap/31-1'
    -- Executing [h at incoming-from-pri:1] Hangup("Zap/31-1", "") in new stack
  == Spawn extension (incoming-from-pri, h, 1) exited non-zero on 'Zap/31-1'
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Active
q931.c:2968 q931_disconnect: call 21 on channel 31 enters state 11
(Disconnect Request)

> Protocol Discriminator: Q.931 (8)  len=9
> Call Ref: len= 2 (reference 21/0x15) (Terminator)
> Message type: DISCONNECT (69)
> [08 02 81 90]
> Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  Location: Private network serving the local user (1)
>                  Ext: 1  Cause: Normal Clearing (16), class = Normal Event (1) ]

    -- Hungup 'Zap/31-1'
< Protocol Discriminator: Q.931 (8)  len=5
< Call Ref: len= 2 (reference 2/0x2) (Terminator)
< Message type: RELEASE COMPLETE (90)
q931.c:3719 q931_receive: call 32770 on channel 1 enters state 0 (Null)
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
< Protocol Discriminator: Q.931 (8)  len=5
< Call Ref: len= 2 (reference 21/0x15) (Originator)
< Message type: RELEASE (77)
q931.c:3754 q931_receive: call 21 on channel 31 enters state 0 (Null)
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request

> Protocol Discriminator: Q.931 (8)  len=9
> Call Ref: len= 2 (reference 21/0x15) (Terminator)
> Message type: RELEASE COMPLETE (90)
> [08 02 81 90]
> Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  Location: Private network serving the local user (1)
>                  Ext: 1  Cause: Normal Clearing (16), class = Normal Event (1) ]

NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null


On Tue, Oct 13, 2009 at 9:16 PM, Michael S Collins <msc at freeswitch.org>wrote:

>
> On Oct 13, 2009, at 6:45 AM, lakshmanan ganapathy <lakindia89 at gmail.com>
> wrote:
>
> We are using Reliance as the Carrier.
> I think, with this same Reliance carrier, in my office, they are able to
> make outgoing calls through asterisk+libpri.
>
>
> If that's the case I would be very interested in seeing a pri debug from a
> working call on an asterisk box. It might give us a clue as to what is not
> working. -MC
>
>
>
> On Tue, Oct 13, 2009 at 12:19 AM, Michael Collins < <msc at freeswitch.org>
> msc at freeswitch.org> wrote:
>
>> Lak,
>>
>> Okay I will need a little bit of time to dig into the IE's and what they
>> contain. In the meantime can you tell me who the carrier is? I'd like to
>> find out if they have some specific requirements. The fact that it doesn't
>> work with libpri surprises me because that would mean that Asterisk systems
>> would probably not work with this carrier as well.
>>
>> BTW, thanks for the very complete pastebin entries. :)
>> -MC
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From velu.technical at gmail.com  Tue Oct 13 22:50:39 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Wed, 14 Oct 2009 11:20:39 +0530
Subject: [Freeswitch-users] Perl event socket library problem
Message-ID: <1452e2980910132250w293596d6jce5515d787b4fa2a@mail.gmail.com>

Dear All,
       I am implementing an IVR framework using Perl event socket libraries.
At first I set socket mode async full. I have faced problem that the Perl
statements executing before dailplan execution. So, I couldn't control my
process.

       Next I have tried without async mode, it solved that race condition
problem. I had another requirement that while executing external application
I need to play some voice file some time limit. When I played that voice
file in playback, the playback application blocked. So, I can't break that
playback application after some time limit.

       I am hanging on this problem for past one week. Please any one help
me to solve above problems...

Thanks,
Velusamy.
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From msc at freeswitch.org  Tue Oct 13 23:03:06 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Oct 2009 23:03:06 -0700
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <7d79b3930910132234k4b99c794q8854ce20d14ee74e@mail.gmail.com>
References: <7d79b3930910090407m5dc81d12pf2b0a47f6017804@mail.gmail.com>
	<20091009123615.BC50A3FA6E9@mail.cune.org>
	<7d79b3930910100647u7fd11413ucec0a535d68b3e3a@mail.gmail.com>
	<87f2f3b90910101009j1295d812jd98cae91e30b1eff@mail.gmail.com>
	<7d79b3930910120454m1ffc6dd4u42b91a4349597166@mail.gmail.com>
	<87f2f3b90910121149x6e2114f0q996a262ad272a1f6@mail.gmail.com>
	<7d79b3930910130645y658ae041v8219c4c881fa5698@mail.gmail.com>
	<5E7A8451-6264-4822-8152-1B94240BDD48@freeswitch.org>
	<7d79b3930910132234k4b99c794q8854ce20d14ee74e@mail.gmail.com>
Message-ID: <87f2f3b90910132303i5966a477ue82667bef975cc93@mail.gmail.com>

Lak,

Okay, it stood out right away: FS is trying u-law but Asterisk is trying
A-law. I'm not sure where the codec for openzap gets selected but you can
try modifying this line in vars.xml:

<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>

Try putting PCMA first. Also, I've never tried this but perhaps in your
dialplan you can set absolute_codec_string to PCMA. I'm very interesting in
hearing about the results. Please try them as soon as you can and report
back.

Thanks,
MC

On Tue, Oct 13, 2009 at 10:34 PM, lakshmanan ganapathy <lakindia89 at gmail.com
> wrote:

> Hi. I made the setup quickly with one of my colleague, and here is the
> debug log given by asterisk
>
> Steps done by me :
> 	+ Call Made from 04443902743 to 04439114600
> 	+ When call lands to 04439114600, I just route it to 09176454982.
>
> *CLI>
> *CLI>
> *CLI> pri debug span 1
> Enabled debugging on span 1
> *CLI>
> *CLI>
> *CLI>
> *CLI> < Protocol Discriminator: Q.931 (8)  len=41
> < Call Ref: len= 2 (reference 21/0x15) (Originator)
>
> < Message type: SETUP (5)
> < [a1]
> < Sending Complete (len= 1)
> < [04 03 80 90 a3]
> < Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer capability: Speech (0)
> <                              Ext: 1  Trans mode/rate: 64kbps, circuit-mode (16)
>
> <                                User information layer 1: A-Law (35)
> < [18 03 a1 83 9f]
> < Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Preferred  Dchan: 0
> <                        ChanSel: As indicated in following octets
>
> <                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
> <                       Ext: 1  Channel: 31 ]
> < [6c 0c 01 81 34 34 34 33 39 30 32 37 34 33]
> < Calling Number (len=14) [ Ext: 0  TON: Unknown Number Type (0)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
>
> <                           Presentation: Presentation permitted, user number passed network screening (1)  '4443902743' ]
> < [70 09 c1 33 39 31 31 34 36 30 30]
> < Called Number (len=11) [ Ext: 1  TON: Subscriber Number (4)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)  '39114600' ]
>
> -- Making new call for cr 21
> -- Processing Q.931 Call Setup
> -- Processing IE 161 (cs0, Sending Complete)
> -- Processing IE 4 (cs0, Bearer Capability)
> -- Processing IE 24 (cs0, Channel Identification)
> -- Processing IE 108 (cs0, Calling Party Number)
>
> -- Processing IE 112 (cs0, Called Party Number)
> q931.c:3504 q931_receive: call 21 on channel 31 enters state 6 (Call Present)
> q931.c:2769 q931_call_proceeding: call 21 on channel 31 enters state 9 (Incoming Call Proceeding)
>
> > Protocol Discriminator: Q.931 (8)  len=10
> > Call Ref: len= 2 (reference 21/0x15) (Terminator)
> > Message type: CALL PROCEEDING (2)
> > [18 03 a9 83 9f]
> > Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  Dchan: 0
> >                        ChanSel: As indicated in following octets
> >                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
> >                       Ext: 1  Channel: 31 ]
>
>     -- Accepting call from '4443902743' to '39114600' on channel 0/31, span 1
>     -- Executing [39114600 at incoming-from-pri:1] NoOp("Zap/31-1", "FMS Call Extn:39114600:| CID::| Channel:Zap/31-1:| ") in new stack
>
>     -- Executing [39114600 at incoming-from-pri:2] Dial("Zap/31-1", "Zap/g1/w09176454982||rj") in new stack
> -- Making new call for cr 32770
>     -- Requested transfer capability: 0x00 - SPEECH
>
> > Protocol Discriminator: Q.931 (8)  len=43
> > Call Ref: len= 2 (reference 2/0x2) (Originator)
> > Message type: SETUP (5)
> > [04 03 80 90 a3]
> > Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer capability: Speech (0)
> >                              Ext: 1  Trans mode/rate: 64kbps, circuit-mode (16)
> >                                User information layer 1: A-Law (35)
> > [18 03 a9 83 81]
> > Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  Dchan: 0
> >                        ChanSel: As indicated in following octets
> >                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
> >                       Ext: 1  Channel: 1 ]
> > [6c 0c 00 81 34 34 34 33 39 30 32 37 34 33]
> > Calling Number (len=14) [ Ext: 0  TON: Unknown Number Type (0)  NPI: Unknown Number Plan (0)
> >                           Presentation: Presentation permitted, user number passed network screening (1)  '4443902743' ]
> > [70 0c 80 30 39 31 37 36 34 35 34 39 38 32]
> > Called Number (len=14) [ Ext: 1  TON: Unknown Number Type (0)  NPI: Unknown Number Plan (0)  '09176454982' ]
>
> q931.c:3087 q931_setup: call 32770 on channel 1 enters state 1 (Call Initiated)
>
>     -- Called g1/w09176454982
> q931.c:2797 q931_alerting: call 21 on channel 31 enters state 7 (Call Received)
>
> > Protocol Discriminator: Q.931 (8)  len=9
> > Call Ref: len= 2 (reference 21/0x15) (Terminator)
> > Message type: ALERTING (1)
> > [1e 02 81 88]
> > Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Private network serving the local user (1)
> >                               Ext: 1  Progress Description: Inband information or appropriate pattern now available. (8) ]
>
> < Protocol Discriminator: Q.931 (8)  len=10
> < Call Ref: len= 2 (reference 2/0x2) (Terminator)
> < Message type: CALL PROCEEDING (2)
> < [18 03 a9 83 81]
> < Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  Dchan: 0
>
> <                        ChanSel: As indicated in following octets
> <                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
> <                       Ext: 1  Channel: 1 ]
> -- Processing IE 24 (cs0, Channel Identification)
>
> q931.c:3636 q931_receive: call 32770 on channel 1 enters state 3 (Outgoing call  Proceeding)
>     -- Zap/1-1 is proceeding passing it to Zap/31-1
> < Protocol Discriminator: Q.931 (8)  len=13
> < Call Ref: len= 2 (reference 2/0x2) (Terminator)
>
> < Message type: ALERTING (1)
> < [1e 02 84 88]
> < Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Public network serving the remote user (4)
> <                               Ext: 1  Progress Description: Inband information or appropriate pattern now available. (8) ]
>
> < [1e 02 84 82]
> < Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Public network serving the remote user (4)
> <                               Ext: 1  Progress Description: Called equipment is non-ISDN. (2) ]
>
> -- Processing IE 30 (cs0, Progress Indicator)
> -- Processing IE 30 (cs0, Progress Indicator)
> q931.c:3549 q931_receive: call 32770 on channel 1 enters state 4 (Call Delivered)
>     -- Zap/1-1 is ringing
> < Protocol Discriminator: Q.931 (8)  len=9
>
> < Call Ref: len= 2 (reference 2/0x2) (Terminator)
> < Message type: PROGRESS (3)
> < [1e 02 84 84]
> < Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Public network serving the remote user (4)
>
> <                               Ext: 1  Progress Description: Unknown (4) ]
> -- Processing IE 30 (cs0, Progress Indicator)
>     -- Zap/1-1 is making progress passing it to Zap/31-1
> < Protocol Discriminator: Q.931 (8)  len=9
>
> < Call Ref: len= 2 (reference 2/0x2) (Terminator)
> < Message type: PROGRESS (3)
> < [1e 02 84 88]
> < Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Public network serving the remote user (4)
>
> <                               Ext: 1  Progress Description: Inband information or appropriate pattern now available. (8) ]
> -- Processing IE 30 (cs0, Progress Indicator)
>     -- Zap/1-1 is making progress passing it to Zap/31-1
>
> < Protocol Discriminator: Q.931 (8)  len=19
> < Call Ref: len= 2 (reference 2/0x2) (Terminator)
> < Message type: CONNECT (7)
> < [4c 0c 01 83 39 31 37 36 34 35 34 39 38 32]
> < Connected Number (len=14) [ Ext: 0  TON: Unknown Number Type (0)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
>
> <                             Ext: 1  Presentation: Presentation allowed of network provided number (3)  '9176454982' ]
> -- Processing IE 76 (cs0, Connected Number)
> q931.c:3579 q931_receive: call 32770 on channel 1 enters state 10 (Active)
>
> > Protocol Discriminator: Q.931 (8)  len=5
> > Call Ref: len= 2 (reference 2/0x2) (Originator)
> > Message type: CONNECT ACKNOWLEDGE (15)
>
>     -- Zap/1-1 answered Zap/31-1
> q931.c:2904 q931_connect: call 21 on channel 31 enters state 8 (Connect Request)
>
> > Protocol Discriminator: Q.931 (8)  len=14
> > Call Ref: len= 2 (reference 21/0x15) (Terminator)
> > Message type: CONNECT (7)
> > [18 03 a9 83 9f]
> > Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  Dchan: 0
> >                        ChanSel: As indicated in following octets
> >                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
> >                       Ext: 1  Channel: 31 ]
> > [1e 02 81 82]
> > Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Private network serving the local user (1)
> >                               Ext: 1  Progress Description: Called equipment is non-ISDN. (2) ]
>
>     -- Native bridging Zap/31-1 and Zap/1-1
> < Protocol Discriminator: Q.931 (8)  len=5
> < Call Ref: len= 2 (reference 21/0x15) (Originator)
> < Message type: CONNECT ACKNOWLEDGE (15)
>
> q931.c:3664 q931_receive: call 21 on channel 31 enters state 10 (Active)
> < Protocol Discriminator: Q.931 (8)  len=9
> < Call Ref: len= 2 (reference 2/0x2) (Terminator)
> < Message type: DISCONNECT (69)
> < [08 02 80 90]
>
> < Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  Location: User (0)
> <                  Ext: 1  Cause: Normal Clearing (16), class = Normal Event (1) ]
> -- Processing IE 8 (cs0, Cause)
> q931.c:3779 q931_receive: call 32770 on channel 1 enters state 12 (Disconnect Indication)
>
>     -- Channel 0/1, span 1 got hangup request, cause 16
> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request
> q931.c:2920 q931_release: call 32770 on channel 1 enters state 19 (Release Request)
>
> > Protocol Discriminator: Q.931 (8)  len=9
> > Call Ref: len= 2 (reference 2/0x2) (Originator)
> > Message type: RELEASE (77)
> > [08 02 81 90]
> > Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  Location: Private network serving the local user (1)
> >                  Ext: 1  Cause: Normal Clearing (16), class = Normal Event (1) ]
>
>     -- Hungup 'Zap/1-1'
>   == Spawn extension (incoming-from-pri, 39114600, 2) exited non-zero on 'Zap/31-1'
>
>     -- Executing [h at incoming-from-pri:1] Hangup("Zap/31-1", "") in new stack
>   == Spawn extension (incoming-from-pri, h, 1) exited non-zero on 'Zap/31-1'
> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Active
>
> q931.c:2968 q931_disconnect: call 21 on channel 31 enters state 11 (Disconnect Request)
>
> > Protocol Discriminator: Q.931 (8)  len=9
> > Call Ref: len= 2 (reference 21/0x15) (Terminator)
> > Message type: DISCONNECT (69)
> > [08 02 81 90]
> > Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  Location: Private network serving the local user (1)
> >                  Ext: 1  Cause: Normal Clearing (16), class = Normal Event (1) ]
>
>     -- Hungup 'Zap/31-1'
> < Protocol Discriminator: Q.931 (8)  len=5
>
> < Call Ref: len= 2 (reference 2/0x2) (Terminator)
> < Message type: RELEASE COMPLETE (90)
> q931.c:3719 q931_receive: call 32770 on channel 1 enters state 0 (Null)
> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
>
> NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
> < Protocol Discriminator: Q.931 (8)  len=5
> < Call Ref: len= 2 (reference 21/0x15) (Originator)
> < Message type: RELEASE (77)
> q931.c:3754 q931_receive: call 21 on channel 31 enters state 0 (Null)
>
> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request
>
> > Protocol Discriminator: Q.931 (8)  len=9
> > Call Ref: len= 2 (reference 21/0x15) (Terminator)
> > Message type: RELEASE COMPLETE (90)
> > [08 02 81 90]
> > Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  Location: Private network serving the local user (1)
> >                  Ext: 1  Cause: Normal Clearing (16), class = Normal Event (1) ]
>
> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
>
> NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
>
>
> On Tue, Oct 13, 2009 at 9:16 PM, Michael S Collins <msc at freeswitch.org>wrote:
>
>>
>> On Oct 13, 2009, at 6:45 AM, lakshmanan ganapathy <lakindia89 at gmail.com>
>> wrote:
>>
>> We are using Reliance as the Carrier.
>> I think, with this same Reliance carrier, in my office, they are able to
>> make outgoing calls through asterisk+libpri.
>>
>>
>> If that's the case I would be very interested in seeing a pri debug from a
>> working call on an asterisk box. It might give us a clue as to what is not
>> working. -MC
>>
>>
>>
>> On Tue, Oct 13, 2009 at 12:19 AM, Michael Collins < <msc at freeswitch.org>
>> msc at freeswitch.org> wrote:
>>
>>> Lak,
>>>
>>> Okay I will need a little bit of time to dig into the IE's and what they
>>> contain. In the meantime can you tell me who the carrier is? I'd like to
>>> find out if they have some specific requirements. The fact that it doesn't
>>> work with libpri surprises me because that would mean that Asterisk systems
>>> would probably not work with this carrier as well.
>>>
>>> BTW, thanks for the very complete pastebin entries. :)
>>> -MC
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From shiyanov at gmail.com  Tue Oct 13 23:31:05 2009
From: shiyanov at gmail.com (Artem Shiyanov)
Date: Wed, 14 Oct 2009 10:31:05 +0400
Subject: [Freeswitch-users] mod_socket: custom timeout for HEARTBEAT
	event
In-Reply-To: <87f2f3b90910131148m1853a219md8788f55e40ef4c3@mail.gmail.com>
References: <d211f54a0910110959y3d3deed7s43201a3e39756585@mail.gmail.com>
	<86a32abc0910111444j1ab2e446p3cb94b0ae04126e3@mail.gmail.com>
	<97B4BDC9-5B11-4A98-B1BF-CC00A82ECBD9@jerris.com>
	<d211f54a0910112238i5f69275ey7cad437e47e6b567@mail.gmail.com>
	<191c3a030910120726p46d6410ejd94b61612686fbbc@mail.gmail.com>
	<d211f54a0910130832m25aaadd3t4d6e938d1c71d701@mail.gmail.com>
	<87f2f3b90910131148m1853a219md8788f55e40ef4c3@mail.gmail.com>
Message-ID: <d211f54a0910132331j14ccf510xd87bcb092a43f0b3@mail.gmail.com>

Finally!!
Thank you Michael, I didn't know about "status" app. It satisfies all my
desires.

And again,
thanks for all the community for the strong support!

Artem




On Tue, Oct 13, 2009 at 10:48 PM, Michael Collins <msc at freeswitch.org>wrote:

>
>
> On Tue, Oct 13, 2009 at 8:32 AM, Artem Shiyanov <shiyanov at gmail.com>wrote:
>
>> Sorry for my foolishness but I stil can't grasp it.
>> I'm developing app based on inbound mod_event_socket and I don't know how
>> to run "enable_heartbeat" or " sched_heartbeat" without specifying any alive
>> session uuid. I tried to use "create_uuid" and send mentioned  commands to
>> the created uuid  but this approach doesn't work. I do need to monitor the
>> FS state itself rather then any particular call.
>>
>> Please, enlighten me!
>> Artem
>>
>
> So if I understand correctly, you want to see the system heartbeat more
> frequently than every 20 seconds? What, exactly, is in the heartbeat that
> you need? You can get the same information by sending the API "status" and
> reading the results.
>
> If I read the source code correctly it looks like the system heartbeat is
> hard-coded at 20 seconds. See switch_core.c, line 155. However, before you
> go messing with a core file I would make sure that you can't already use the
> tools that are available, like sending a "status" API every 5 seconds (or
> whatever interval you need).
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sjmudd at pobox.com  Tue Oct 13 23:33:43 2009
From: sjmudd at pobox.com (Simon J Mudd)
Date: 14 Oct 2009 08:33:43 +0200
Subject: [Freeswitch-users] Building Freeswitch with VPATH (src and obj
	directories are different)?
In-Reply-To: <D8EB1A6C-38A6-4BC2-AD8C-46CF11993366@freeswitch.org>
References: <20091013205907.GA7545@mad06.wl0.org>
	<D8EB1A6C-38A6-4BC2-AD8C-46CF11993366@freeswitch.org>
Message-ID: <m3eip65vso.fsf@mad06.wl0.org>

brian at freeswitch.org (Brian West) writes:

> You shouldn't have to make clean usually ... doing so might break your  
> tree...

Why?

In any case on my mac (Leopard) neither make clean or distclean fully
cleans up afterwards.  I would prefer to get a completely "untouched"
source tree after doing this.  If that's supposed to work and I should
post a bug or report the problem in more detail then I can do that.

> You can usually get by with "make current" that will ensure  
> the critical things are cleaned and built correctly... every now and  
> they you'll hit a snafu but we'll usually tell you about it.

ok. That's one of the reasons for liking to use VPATH builds. It's also 
why many BSD system builds use this so that you can mount the source
tree readonly (perhaps from another server) and build the binaries
in a separate directory independently.

If that's not supported then that's fine.

I was trying to build the latest svn version of freeswitch (Revision:
15138) and this didn't build. As I'm not too familiar with compiling
at least on the Mac I thought I'd first try to see if a VPATH build
would work better. I'll have to try and investigate further.

Simon


From tculjaga at gmail.com  Tue Oct 13 23:48:26 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 14 Oct 2009 08:48:26 +0200
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <48B637E0506F4A809E4C0CAF39BD08BB@chaos>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
	<191c3a030910130803k321f39e3n265e64ab3c84a5e6@mail.gmail.com>
	<5e414ed0910130826j11d45a66q86d785f900e4196e@mail.gmail.com>
	<65d96fc80910131012v1f72ac49o6bfe9f2267e541c5@mail.gmail.com>
	<191c3a030910131101p418e0961t373db68285248ff6@mail.gmail.com>
	<538F4CF43B294D0EACABA281E1913B4C@chaos>
	<65d96fc80910131224m73f35b87g355be818ad3c0caf@mail.gmail.com>
	<48B637E0506F4A809E4C0CAF39BD08BB@chaos>
Message-ID: <65d96fc80910132348t202905fbub57cc4c814eb4e21@mail.gmail.com>

I suppose he want to have a central dialplan and a dummy phone instead...
something as a MGCP phone behavior.

T.

On Tue, Oct 13, 2009 at 10:22 PM, Metik <freeswitch-users-list at metik.com>wrote:

>  As evidenced by various DTMF interop issues (with RFC2833, inband,
> etc) over the years, I would avoid it if at all possible.
>
> What does it particularly do that can not accomplished by using RFC 2833 or
> (less ideal) inband DTMF?  Or are you attempting to use it as a band-aid to
> address some sort of interop issue with the carrier involved that is
> wrecking havoc with your particular application?
>
> -metik
>
> ----- Original Message -----
> *From:* Tihomir Culjaga <tculjaga at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, October 13, 2009 3:24 PM
> *Subject:* Re: [Freeswitch-users] SIP Overlap support?
>
> i never found it working properly... i always had some interoperability
> issues and i finished having a "dialplan" on my phones being delivered
> through a config file via tftp or http .. depending of the phone capability.
>
> BTW: using overlap can lead to a greater system load... be careful when
> setting the minimum number of digits you will send in 1st message. I wish
> you luck...
>
> T.
>
> On Tue, Oct 13, 2009 at 9:03 PM, Metik <freeswitch-users-list at metik.com>wrote:
>
>>  Both support it.  In the Grandstream, I believe it is called Early Dial
>> (vs. SNOM's Overlap Dialing).  It can be problematic if you have a device
>> somewhere in the middle that doesn't support 484s.
>>
>> -metik
>>
>>   ----- Original Message -----
>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Sent:* Tuesday, October 13, 2009 2:01 PM
>> *Subject:* Re: [Freeswitch-users] SIP Overlap support?
>>
>> i do think some softphone can do it but i forgot which one it was either
>> snom or grandstream
>>
>>
>> On Tue, Oct 13, 2009 at 12:12 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>
>>> you need a softswitch.... i'm afraid a SIP phone is not designed for
>>> overlap...
>>>
>>> T.
>>>
>>>
>>> On Tue, Oct 13, 2009 at 5:26 PM, Dennis <odermann at googlemail.com> wrote:
>>>
>>>> how could we try? we played arround with a snom phone (snom seems to
>>>> support something in this direction, but are not shure, how we can
>>>> test it and how we can see if it is supported or not.
>>>>
>>>> any hint?
>>>>
>>>>
>>>> 2009/10/13 Anthony Minessale <anthony.minessale at gmail.com>:
>>>>  > have you tried it?
>>>> > I *think* either we did support it or we would with a small patch to
>>>> sofia
>>>> > lib that I cannot recall if we ever got committed.
>>>> >
>>>> >
>>>> > On Tue, Oct 13, 2009 at 8:51 AM, Dennis <odermann at googlemail.com>
>>>> wrote:
>>>> >>
>>>> >> hi there,
>>>> >>
>>>> >> i would like to ask, if fs has support for something like "SIP
>>>> Overlap"?
>>>> >>
>>>> >> instead of receiving the phonenumber from our carrier in a block, we
>>>> >> want to receive the phonenumber digit-by-digit and we want to tell fs
>>>> >> when the number is complete. our carrier could send us the
>>>> phonenumber
>>>> >> digit-by-digit, but what about the fs-side?
>>>> >>
>>>> >>
>>>> >> thanks and kind regards
>>>> >> dennis
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >
>>>> >
>>>> >
>>>> > --
>>>> > Anthony Minessale II
>>>> >
>>>> > FreeSWITCH http://www.freeswitch.org/
>>>> > ClueCon http://www.cluecon.com/
>>>> > Twitter: http://twitter.com/FreeSWITCH_wire
>>>> >
>>>> > AIM: anthm
>>>> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> > IRC: irc.freenode.net #freeswitch
>>>> >
>>>> > FreeSWITCH Developer Conference
>>>> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>> > iax:guest at conference.freeswitch.org/888
>>>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> > pstn:213-799-1400
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Tue Oct 13 23:59:00 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 14 Oct 2009 08:59:00 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910130014hc5f5fcet677f87b7fb57559e@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<65d96fc80910130014hc5f5fcet677f87b7fb57559e@mail.gmail.com>
Message-ID: <65d96fc80910132359i6d53d8ban10f943795481bf7c@mail.gmail.com>

the e-mail bounced back .. don't know whether you received this...






here are the results!


Outbound calls:

SIP_user(1001) => FS => H323_SoftSwitch:
http://pastebin.freeswitch.org/10672

i get coredumps but this might be because i'm missing the bridge format....
what i'm using is: <action application="bridge" data="h323/0$1@${NCX_IP}"/>
that can be wrong :) Please advice here.




Inbound Calls:

openPhone => FS => SIP_user(1001): http://pastebin.freeswitch.org/10671


the call went in, hit a registered sip user (1001) and the phone started
ringing. The phone answered the call but no audio can be heared (i see RTP
from the SIP uset towards FS). Also, the openPhone never receives H.225
connect nor any OLC before connect. After some time, the H323 leg constates
the user never answered and releases the call with cause 19.




this is the h323.config.xml

tculjaga at subZero:~$ cat
/usr/local/freeswitch/conf/autoload_configs/h323.conf.xml
<configuration name="h323.conf" description="H323 Endpoints">
  <settings>
    <param name="trace-level" value="10"/>
    <param name="context" value="default"/>
    <param name="dialplan" value="XML"/>
    <param name="codec-prefs" value="PCMA,GSM,G729,G726"/>
    <param name="gk-address" value=""/>    <!-- empty to disable, "*" to
search LAN -->
    <param name="gk-identifer" value=""/>  <!-- optional name of gk -->
    <param name="gk-interface" value=""/>  <!-- optional listener interface
name -->
  </settings>
  <listeners>
    <listener name="default">
      <param name="h323-ip" value="10.4.62.7"/>
      <param name="h323-port" value="1720"/>
    </listener>
  </listeners>
</configuration>



and this is a dialplan for outgoing that i'm using


<include>

  <extension name="EMERGENCY">
    <condition field="destination_number" expression="^0(112|9[23456])$">
      <action application="set"
data="effective_caller_id_number=1001282122"/>
      <action application="set" data="NCX_IP=10.4.4.254"/>
      <action application="set" data="call_timeout=30"/>
      <action application="set" data="hangup_after_bridge=true"/>

      <!--action application="bridge" data="opal/h323:0$1@${NCX_IP}"/-->
      <action application="bridge" data="h323/0$1@${NCX_IP}"/>
    </condition>
  </extension>

  <extension name="SPECIAL_SERVICES">
    <condition field="destination_number" expression="^0(9[01789]\d{3,4})$">
      <action application="set"
data="effective_caller_id_number=1001282122"/>
      <action application="set" data="NCX_IP=10.4.4.254"/>
      <action application="set" data="call_timeout=30"/>
      <action application="set" data="hangup_after_bridge=true"/>

      <!--action application="bridge" data="opal/h323:0$1@${NCX_IP}"/-->
      <action application="bridge" data="h323/0$1@${NCX_IP}"/>
    </condition>
  </extension>

  <extension name="ENYTHING_ELSE">
    <condition field="destination_number"
expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
      <action application="set"
data="effective_caller_id_number=1001282122"/>
      <action application="set" data="NCX_IP=10.4.4.254"/>
      <action application="set" data="call_timeout=30"/>
      <action application="set" data="hangup_after_bridge=true"/>

      <!--action application="bridge" data="opal/h323:0$1@${NCX_IP}"/-->
      <action application="bridge" data="h323/0$1@${NCX_IP}"/>
    </condition>
  </extension>

</include>



P.S.: sorry for the attached rar, but don't know how to upload it to
pastebin :P


T.




2009/10/12 Tihomir Culjaga <tculjaga at gmail.com>

hi,
>
> finally i compiled it right ... had a stupid issue with ekiga and wrong
> ptlib in place...
>
> anyhow, i loaded the module and will continue the tests tomorrow ...first
> thing i arrive in my office :P
>
>
>
> freeswitch at subZero>
> freeswitch at subZero>
> API CALL [console(loglevel 7)] output:
> +OK console log level set to DEBUG
>
> freeswitch at subZero> load mod_h323
> 2009-10-12 23:38:09.509279 [CONSOLE] mod_h323.cpp:93 Starting loading
> mod_h323
> 2009-10-12 23:38:09.511413 [DEBUG] mod_h323.cpp:461 Created Listener
> 'default'
> 2009-10-12 23:38:09.511413 [DEBUG] mod_h323.cpp:305 Config capabilliti
> PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.511413 [DEBUG] mod_h323.cpp:309 Find capabilliti PCMU
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.511413 [DEBUG] mod_h323.cpp:309 Find capabilliti PCMA
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.511413 [DEBUG] h323caps.cxx:3264 FindCapability:
> "G.711-uLaw-64k{sw}"
> 2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:3176 Added capability:
> G.711-uLaw-64k <1>
> 2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:3264 FindCapability:
> "G.711-uLaw-64k{sw}"
> 2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:3264 FindCapability:
> "G.711-uLaw-64k{sw}"
> 2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:3264 FindCapability:
> "G.711-uLaw-64k{sw}"
> 2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:3264 FindCapability:
> "G.711-uLaw-64k{sw}"
> 2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:1989 No Extended
> Capabilities found to load
> 2009-10-12 23:38:09.512480 [DEBUG] mod_h323.cpp:325 H.323 added 1
> capabilities 'G.711-uLaw-64k*{sw}'
> 2009-10-12 23:38:09.512480 [DEBUG] mod_h323.cpp:309 Find capabilliti gsm to
> PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti msgsm
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti speex
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti lpc10
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti ilbc20
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti ilbc30
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g723
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g726
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g728
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g729b
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g729
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti slin
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g729a
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g729ab
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti g723.1
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
> g723.1-5k3 to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
> g723.1a-5k3 to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
> g723.1a-6k3 to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
> g723.1a-6k3-cisco to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
> g726-16 to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
> g726-24 to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
> g726-32 to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
> g726-40 to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti ilbc
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
> speex-18k2 to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
> speex-15k to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
> speex-11k to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.515708 [DEBUG] mod_h323.cpp:309 Find capabilliti
> speex-8k to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.515708 [DEBUG] mod_h323.cpp:309 Find capabilliti
> speex-5k95 to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
> UserInput/hookflash <2>
> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
> UserInput/basicString <3>
> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
> UserInput/dtmf <4>
> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
> UserInput/RFC2833 <5>
> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
> UserInput/Navigation <6>
> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
> UserInput/Softkey <7>
> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
> UserInput/PointDevice <8>
> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
> UserInput/Modal <9>
> 2009-10-12 23:38:09.516760 [DEBUG] osutil.cxx:188 File handle high water
> mark set: 44 PTCPSocket
> 2009-10-12 23:38:09.516760 [DEBUG] h323ep.cxx:1995 Started listener
> Listener[ip$10.4.62.7:1720]
> 2009-10-12 23:38:09.516760 [DEBUG] tlibthrd.cxx:547 Thread high water mark
> set: 3
> 2009-10-12 23:38:09.516760 [CONSOLE] mod_h323.cpp:108 Opal manager
> initialized and running
> 2009-10-12 23:38:09.516760 [DEBUG] tlibthrd.cxx:454 Started thread
> 0xb75e0f18 H323 Listener:b5c98b90
> 2009-10-12 23:38:09.516760 [CONSOLE] switch_loadable_module.c:889
> Successfully Loaded [mod_h323]
> 2009-10-12 23:38:09.516760 [NOTICE] switch_loadable_module.c:142 Adding
> Endpoint 'h323'
> API CALL [load(mod_h323)] output:
> +OK
>
> 2009-10-12 23:38:09.516760 [DEBUG] transports.cxx:1521 Awaiting TCP
> connections on port 1720
> 2009-10-12 23:38:09.517832 [DEBUG] transports.cxx:1475 Waiting on socket
> accept on ip$10.4.62.7:1720
> 2009-10-12 23:38:09.517832 [DEBUG] tlibthrd.cxx:1023
> PThread::PXBlockOnIO(44,2)
> freeswitch at subZero>
>
>
> BTW: to compile everytihng,  I used FS trunk, ptlib 2.7.1 and h323plus
> 1.22.0
>
> Will let you know the results asap.
>
>
> T.
>
>
>
> 2009/10/12 Georgiewskiy Yuriy <bottleman at icf.org.ru>
>
>> On 2009-10-12 18:03 +0200, Tihomir Culjaga wrote
>> freeswitch-users at lists.fre...:
>>
>> i already write this, ptlib 2.6.5, tou can find link to it on
>> oplalvoip.org, h32plus latest CVS version,
>> you can find it on www.h323plus.org.
>>
>> TC>hi,
>> TC>
>> TC>can't make it...
>> TC>
>> TC>subZero:~/freeswitch-trunk$ make mod_h323
>> TC>
>> TC>making all mod_h323
>> TC>Compiling mod_h323.cpp...
>> TC>quiet_libtool: compile:  g++ -g -ggdb -I/usr/local/include/ptlib
>> TC>-I/usr/local/include/openh323 -I. -DPTRACING=1 -D_REENTRANT
>> -fno-exceptions
>> TC>-I/home/tculjaga/freeswitch-trunk/src/include
>> TC>-I/home/tculjaga/freeswitch-trunk/libs/libteletone/src -fPIC
>> TC>-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g
>> -O2
>> TC>-D_GNU_SOURCE -DHAVE_CONFIG_H -c mod_h323.cpp  -fPIC -DPIC -o
>> TC>.libs/mod_h323.o
>> TC>In file included from /usr/local/include/openh323/h323.h:493,
>> TC>                 from mod_h323.h:8,
>> TC>                 from mod_h323.cpp:3:
>> TC>/usr/local/include/openh323/h323ep.h: In member function ??????virtual
>> void
>> TC>NATFactoryStartup::OnShutdown()??????:
>> TC>/usr/local/include/openh323/h323ep.h:2731: error:
>> ??????NatFactory?????? has not
>> TC>been declared
>> TC>make[4]: *** [mod_h323.lo] Error 1
>> TC>make[3]: *** [all] Error 1
>> TC>make[2]: *** [mod_h323-all] Error 1
>> TC>make[1]: *** [mod_h323] Error 2
>> TC>make: *** [mod_h323] Error 2
>> TC>
>> TC>
>> TC>
>> TC>what exact ptlib and h323plus versions did you use? .. can you send us
>> a
>> TC>link so we can use the exact ones.
>> TC>
>> TC>
>> TC>T.
>> TC>
>> TC>2009/10/12 Georgiewskiy Yuriy <bottleman at icf.org.ru>
>> TC>
>> TC>> On 2009-10-12 09:43 -0500, Brian West wrote
>> TC>> freeswitch-users at lists.freeswit...:
>> TC>>
>> TC>> BW>We can host this in our SVN if you wish?
>> TC>>
>> TC>> If in fs svn i think yes. But i think may be little time later?
>> TC>> i don't known is it builds on trunk because i develop it on 1.0.4.
>> TC>>
>> TC>> BW>/b
>> TC>> BW>
>> TC>> BW>On Oct 12, 2009, at 8:31 AM, Georgiewskiy Yuriy wrote:
>> TC>> BW>
>> TC>> BW>> ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/ alfa code,
>> but
>> TC>> seems
>> TC>> BW>> it work, but should be buggy,
>> TC>> BW>> to build need libpt 2.6.5 and h323plus cvs version, i test it
>> now on
>> TC>> fs
>> TC>> BW>> 1.0.4.
>> TC>> BW>
>> TC>>
>> TC>> C ?????????                       With Best Regards
>> TC>> ???????????? ????.                Georgiewskiy Yuriy
>> TC>> +7 4872 711666                    +7 4872 711666
>> TC>> ???? +7 4872 711143               fax +7 4872 711143
>> TC>> ???????? ??? "?? ?? ??????"       IT Service Ltd
>> TC>> http://nkoort.ru                  http://nkoort.ru
>> TC>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
>> TC>> YG129-RIPE                        YG129-RIPE
>> TC>>
>> TC>> _______________________________________________
>> TC>> FreeSWITCH-users mailing list
>> TC>> FreeSWITCH-users at lists.freeswitch.org
>> TC>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> TC>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> TC>> http://www.freeswitch.org
>> TC>>
>> TC>>
>> TC>
>>
>> C ?????????                       With Best Regards
>> ???????????? ????.                Georgiewskiy Yuriy
>> +7 4872 711666                    +7 4872 711666
>> ???? +7 4872 711143               fax +7 4872 711143
>> ???????? ??? "?? ?? ??????"       IT Service Ltd
>> http://nkoort.ru                  http://nkoort.ru
>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
>> YG129-RIPE                        YG129-RIPE
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From bottleman at icf.org.ru  Wed Oct 14 00:36:13 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Wed, 14 Oct 2009 11:36:13 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910132359i6d53d8ban10f943795481bf7c@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<65d96fc80910130014hc5f5fcet677f87b7fb57559e@mail.gmail.com>
	<65d96fc80910132359i6d53d8ban10f943795481bf7c@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910141134270.11620@srv.icf.org.ru>

On 2009-10-14 08:59 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

try sow start on h323 channel, there is a bug in faststart, i will fix it later.

TC>the e-mail bounced back .. don't know whether you received this...
TC>
TC>
TC>
TC>
TC>
TC>
TC>here are the results!
TC>
TC>
TC>Outbound calls:
TC>
TC>SIP_user(1001) => FS => H323_SoftSwitch:
TC>http://pastebin.freeswitch.org/10672
TC>
TC>i get coredumps but this might be because i'm missing the bridge format....
TC>what i'm using is: <action application="bridge" data="h323/0$1@${NCX_IP}"/>
TC>that can be wrong :) Please advice here.
TC>
TC>
TC>
TC>
TC>Inbound Calls:
TC>
TC>openPhone => FS => SIP_user(1001): http://pastebin.freeswitch.org/10671
TC>
TC>
TC>the call went in, hit a registered sip user (1001) and the phone started
TC>ringing. The phone answered the call but no audio can be heared (i see RTP
TC>from the SIP uset towards FS). Also, the openPhone never receives H.225
TC>connect nor any OLC before connect. After some time, the H323 leg constates
TC>the user never answered and releases the call with cause 19.
TC>
TC>
TC>
TC>
TC>this is the h323.config.xml
TC>
TC>tculjaga at subZero:~$ cat
TC>/usr/local/freeswitch/conf/autoload_configs/h323.conf.xml
TC><configuration name="h323.conf" description="H323 Endpoints">
TC>  <settings>
TC>    <param name="trace-level" value="10"/>
TC>    <param name="context" value="default"/>
TC>    <param name="dialplan" value="XML"/>
TC>    <param name="codec-prefs" value="PCMA,GSM,G729,G726"/>
TC>    <param name="gk-address" value=""/>    <!-- empty to disable, "*" to
TC>search LAN -->
TC>    <param name="gk-identifer" value=""/>  <!-- optional name of gk -->
TC>    <param name="gk-interface" value=""/>  <!-- optional listener interface
TC>name -->
TC>  </settings>
TC>  <listeners>
TC>    <listener name="default">
TC>      <param name="h323-ip" value="10.4.62.7"/>
TC>      <param name="h323-port" value="1720"/>
TC>    </listener>
TC>  </listeners>
TC></configuration>
TC>
TC>
TC>
TC>and this is a dialplan for outgoing that i'm using
TC>
TC>
TC><include>
TC>
TC>  <extension name="EMERGENCY">
TC>    <condition field="destination_number" expression="^0(112|9[23456])$">
TC>      <action application="set"
TC>data="effective_caller_id_number=1001282122"/>
TC>      <action application="set" data="NCX_IP=10.4.4.254"/>
TC>      <action application="set" data="call_timeout=30"/>
TC>      <action application="set" data="hangup_after_bridge=true"/>
TC>
TC>      <!--action application="bridge" data="opal/h323:0$1@${NCX_IP}"/-->
TC>      <action application="bridge" data="h323/0$1@${NCX_IP}"/>
TC>    </condition>
TC>  </extension>
TC>
TC>  <extension name="SPECIAL_SERVICES">
TC>    <condition field="destination_number" expression="^0(9[01789]\d{3,4})$">
TC>      <action application="set"
TC>data="effective_caller_id_number=1001282122"/>
TC>      <action application="set" data="NCX_IP=10.4.4.254"/>
TC>      <action application="set" data="call_timeout=30"/>
TC>      <action application="set" data="hangup_after_bridge=true"/>
TC>
TC>      <!--action application="bridge" data="opal/h323:0$1@${NCX_IP}"/-->
TC>      <action application="bridge" data="h323/0$1@${NCX_IP}"/>
TC>    </condition>
TC>  </extension>
TC>
TC>  <extension name="ENYTHING_ELSE">
TC>    <condition field="destination_number"
TC>expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
TC>      <action application="set"
TC>data="effective_caller_id_number=1001282122"/>
TC>      <action application="set" data="NCX_IP=10.4.4.254"/>
TC>      <action application="set" data="call_timeout=30"/>
TC>      <action application="set" data="hangup_after_bridge=true"/>
TC>
TC>      <!--action application="bridge" data="opal/h323:0$1@${NCX_IP}"/-->
TC>      <action application="bridge" data="h323/0$1@${NCX_IP}"/>
TC>    </condition>
TC>  </extension>
TC>
TC></include>
TC>
TC>
TC>
TC>P.S.: sorry for the attached rar, but don't know how to upload it to
TC>pastebin :P
TC>
TC>
TC>T.
TC>
TC>
TC>
TC>
TC>2009/10/12 Tihomir Culjaga <tculjaga at gmail.com>
TC>
TC>hi,
TC>>
TC>> finally i compiled it right ... had a stupid issue with ekiga and wrong
TC>> ptlib in place...
TC>>
TC>> anyhow, i loaded the module and will continue the tests tomorrow ...first
TC>> thing i arrive in my office :P
TC>>
TC>>
TC>>
TC>> freeswitch at subZero>
TC>> freeswitch at subZero>
TC>> API CALL [console(loglevel 7)] output:
TC>> +OK console log level set to DEBUG
TC>>
TC>> freeswitch at subZero> load mod_h323
TC>> 2009-10-12 23:38:09.509279 [CONSOLE] mod_h323.cpp:93 Starting loading
TC>> mod_h323
TC>> 2009-10-12 23:38:09.511413 [DEBUG] mod_h323.cpp:461 Created Listener
TC>> 'default'
TC>> 2009-10-12 23:38:09.511413 [DEBUG] mod_h323.cpp:305 Config capabilliti
TC>> PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.511413 [DEBUG] mod_h323.cpp:309 Find capabilliti PCMU
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.511413 [DEBUG] mod_h323.cpp:309 Find capabilliti PCMA
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.511413 [DEBUG] h323caps.cxx:3264 FindCapability:
TC>> "G.711-uLaw-64k{sw}"
TC>> 2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:3176 Added capability:
TC>> G.711-uLaw-64k <1>
TC>> 2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:3264 FindCapability:
TC>> "G.711-uLaw-64k{sw}"
TC>> 2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:3264 FindCapability:
TC>> "G.711-uLaw-64k{sw}"
TC>> 2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:3264 FindCapability:
TC>> "G.711-uLaw-64k{sw}"
TC>> 2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:3264 FindCapability:
TC>> "G.711-uLaw-64k{sw}"
TC>> 2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:1989 No Extended
TC>> Capabilities found to load
TC>> 2009-10-12 23:38:09.512480 [DEBUG] mod_h323.cpp:325 H.323 added 1
TC>> capabilities 'G.711-uLaw-64k*{sw}'
TC>> 2009-10-12 23:38:09.512480 [DEBUG] mod_h323.cpp:309 Find capabilliti gsm to
TC>> PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti msgsm
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti speex
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti lpc10
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti ilbc20
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti ilbc30
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g723
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g726
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g728
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g729b
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g729
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti slin
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g729a
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g729ab
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti g723.1
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
TC>> g723.1-5k3 to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
TC>> g723.1a-5k3 to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
TC>> g723.1a-6k3 to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
TC>> g723.1a-6k3-cisco to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
TC>> g726-16 to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
TC>> g726-24 to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
TC>> g726-32 to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
TC>> g726-40 to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti ilbc
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
TC>> speex-18k2 to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
TC>> speex-15k to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
TC>> speex-11k to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.515708 [DEBUG] mod_h323.cpp:309 Find capabilliti
TC>> speex-8k to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.515708 [DEBUG] mod_h323.cpp:309 Find capabilliti
TC>> speex-5k95 to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
TC>> UserInput/hookflash <2>
TC>> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
TC>> UserInput/basicString <3>
TC>> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
TC>> UserInput/dtmf <4>
TC>> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
TC>> UserInput/RFC2833 <5>
TC>> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
TC>> UserInput/Navigation <6>
TC>> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
TC>> UserInput/Softkey <7>
TC>> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
TC>> UserInput/PointDevice <8>
TC>> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
TC>> UserInput/Modal <9>
TC>> 2009-10-12 23:38:09.516760 [DEBUG] osutil.cxx:188 File handle high water
TC>> mark set: 44 PTCPSocket
TC>> 2009-10-12 23:38:09.516760 [DEBUG] h323ep.cxx:1995 Started listener
TC>> Listener[ip$10.4.62.7:1720]
TC>> 2009-10-12 23:38:09.516760 [DEBUG] tlibthrd.cxx:547 Thread high water mark
TC>> set: 3
TC>> 2009-10-12 23:38:09.516760 [CONSOLE] mod_h323.cpp:108 Opal manager
TC>> initialized and running
TC>> 2009-10-12 23:38:09.516760 [DEBUG] tlibthrd.cxx:454 Started thread
TC>> 0xb75e0f18 H323 Listener:b5c98b90
TC>> 2009-10-12 23:38:09.516760 [CONSOLE] switch_loadable_module.c:889
TC>> Successfully Loaded [mod_h323]
TC>> 2009-10-12 23:38:09.516760 [NOTICE] switch_loadable_module.c:142 Adding
TC>> Endpoint 'h323'
TC>> API CALL [load(mod_h323)] output:
TC>> +OK
TC>>
TC>> 2009-10-12 23:38:09.516760 [DEBUG] transports.cxx:1521 Awaiting TCP
TC>> connections on port 1720
TC>> 2009-10-12 23:38:09.517832 [DEBUG] transports.cxx:1475 Waiting on socket
TC>> accept on ip$10.4.62.7:1720
TC>> 2009-10-12 23:38:09.517832 [DEBUG] tlibthrd.cxx:1023
TC>> PThread::PXBlockOnIO(44,2)
TC>> freeswitch at subZero>
TC>>
TC>>
TC>> BTW: to compile everytihng,  I used FS trunk, ptlib 2.7.1 and h323plus
TC>> 1.22.0
TC>>
TC>> Will let you know the results asap.
TC>>
TC>>
TC>> T.
TC>>
TC>>
TC>>
TC>> 2009/10/12 Georgiewskiy Yuriy <bottleman at icf.org.ru>
TC>>
TC>>> On 2009-10-12 18:03 +0200, Tihomir Culjaga wrote
TC>>> freeswitch-users at lists.fre...:
TC>>>
TC>>> i already write this, ptlib 2.6.5, tou can find link to it on
TC>>> oplalvoip.org, h32plus latest CVS version,
TC>>> you can find it on www.h323plus.org.
TC>>>
TC>>> TC>hi,
TC>>> TC>
TC>>> TC>can't make it...
TC>>> TC>
TC>>> TC>subZero:~/freeswitch-trunk$ make mod_h323
TC>>> TC>
TC>>> TC>making all mod_h323
TC>>> TC>Compiling mod_h323.cpp...
TC>>> TC>quiet_libtool: compile:  g++ -g -ggdb -I/usr/local/include/ptlib
TC>>> TC>-I/usr/local/include/openh323 -I. -DPTRACING=1 -D_REENTRANT
TC>>> -fno-exceptions
TC>>> TC>-I/home/tculjaga/freeswitch-trunk/src/include
TC>>> TC>-I/home/tculjaga/freeswitch-trunk/libs/libteletone/src -fPIC
TC>>> TC>-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g
TC>>> -O2
TC>>> TC>-D_GNU_SOURCE -DHAVE_CONFIG_H -c mod_h323.cpp  -fPIC -DPIC -o
TC>>> TC>.libs/mod_h323.o
TC>>> TC>In file included from /usr/local/include/openh323/h323.h:493,
TC>>> TC>                 from mod_h323.h:8,
TC>>> TC>                 from mod_h323.cpp:3:
TC>>> TC>/usr/local/include/openh323/h323ep.h: In member function ??????virtual
TC>>> void
TC>>> TC>NATFactoryStartup::OnShutdown()??????:
TC>>> TC>/usr/local/include/openh323/h323ep.h:2731: error:
TC>>> ??????NatFactory?????? has not
TC>>> TC>been declared
TC>>> TC>make[4]: *** [mod_h323.lo] Error 1
TC>>> TC>make[3]: *** [all] Error 1
TC>>> TC>make[2]: *** [mod_h323-all] Error 1
TC>>> TC>make[1]: *** [mod_h323] Error 2
TC>>> TC>make: *** [mod_h323] Error 2
TC>>> TC>
TC>>> TC>
TC>>> TC>
TC>>> TC>what exact ptlib and h323plus versions did you use? .. can you send us
TC>>> a
TC>>> TC>link so we can use the exact ones.
TC>>> TC>
TC>>> TC>
TC>>> TC>T.
TC>>> TC>
TC>>> TC>2009/10/12 Georgiewskiy Yuriy <bottleman at icf.org.ru>
TC>>> TC>
TC>>> TC>> On 2009-10-12 09:43 -0500, Brian West wrote
TC>>> TC>> freeswitch-users at lists.freeswit...:
TC>>> TC>>
TC>>> TC>> BW>We can host this in our SVN if you wish?
TC>>> TC>>
TC>>> TC>> If in fs svn i think yes. But i think may be little time later?
TC>>> TC>> i don't known is it builds on trunk because i develop it on 1.0.4.
TC>>> TC>>
TC>>> TC>> BW>/b
TC>>> TC>> BW>
TC>>> TC>> BW>On Oct 12, 2009, at 8:31 AM, Georgiewskiy Yuriy wrote:
TC>>> TC>> BW>
TC>>> TC>> BW>> ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/ alfa code,
TC>>> but
TC>>> TC>> seems
TC>>> TC>> BW>> it work, but should be buggy,
TC>>> TC>> BW>> to build need libpt 2.6.5 and h323plus cvs version, i test it
TC>>> now on
TC>>> TC>> fs
TC>>> TC>> BW>> 1.0.4.
TC>>> TC>> BW>
TC>>> TC>>
TC>>> TC>> C ?????????                       With Best Regards
TC>>> TC>> ???????????? ????.                Georgiewskiy Yuriy
TC>>> TC>> +7 4872 711666                    +7 4872 711666
TC>>> TC>> ???? +7 4872 711143               fax +7 4872 711143
TC>>> TC>> ???????? ??? "?? ?? ??????"       IT Service Ltd
TC>>> TC>> http://nkoort.ru                  http://nkoort.ru
TC>>> TC>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
TC>>> TC>> YG129-RIPE                        YG129-RIPE
TC>>> TC>>
TC>>> TC>> _______________________________________________
TC>>> TC>> FreeSWITCH-users mailing list
TC>>> TC>> FreeSWITCH-users at lists.freeswitch.org
TC>>> TC>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>>> TC>> UNSUBSCRIBE:
TC>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>>> TC>> http://www.freeswitch.org
TC>>> TC>>
TC>>> TC>>
TC>>> TC>
TC>>>
TC>>> C ?????????                       With Best Regards
TC>>> ???????????? ????.                Georgiewskiy Yuriy
TC>>> +7 4872 711666                    +7 4872 711666
TC>>> ???? +7 4872 711143               fax +7 4872 711143
TC>>> ???????? ??? "?? ?? ??????"       IT Service Ltd
TC>>> http://nkoort.ru                  http://nkoort.ru
TC>>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
TC>>> YG129-RIPE                        YG129-RIPE
TC>>>
TC>>> _______________________________________________
TC>>> FreeSWITCH-users mailing list
TC>>> FreeSWITCH-users at lists.freeswitch.org
TC>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>>> http://www.freeswitch.org
TC>>>
TC>>>
TC>>
TC>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From shaheryarkh at googlemail.com  Wed Oct 14 00:55:22 2009
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Wed, 14 Oct 2009 13:55:22 +0600
Subject: [Freeswitch-users] Some help with my post-paid billing project
In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6703241D1E52@mse17be1.mse17.exchange.ms>
References: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
	<74730C11-75D0-4BC0-9F63-8B6C9F0F499F@me.com>
	<86a32abc0910131836w5f113a16s247caf5eeb0f8fe4@mail.gmail.com>
	<86a32abc0910131853v244ed61bof9c24f345a0944b2@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C6703241D1E52@mse17be1.mse17.exchange.ms>
Message-ID: <b16156850910140055r37581014iddf52d915aa7eaa4@mail.gmail.com>

I fully agree that direct matching is much faster then pattern matching in
SQL.

One of my clients had same problem, he had around 12 million number prefixes
in a table and during each call an AGI script use to query that table to
find longest prefix match, but this use to take like 3-5 seconds even with
indexed columns. So after a lots of R & D we come up with following logic,

1. We break the destination number length-wise, e.g. suppose number is
923344224088 then length chunks would be,

923344224088, 92334422408, 9233442240, 923344224, 92334422, 9233442, 923344,
92334, 9233, 923, 92, 9

2. Then use SQL LIKE function in WHERE clause (you can also use SQL OR
function if your DBMS doesn't support SQL LIKE function), and pass all these
chunks to it, e.g.

WHERE prefix LIKE (923344224088, 92334422408, 9233442240, 923344224,
92334422, 9233442, 923344, 92334, 9233, 923, 92, 9)

3. Lastly we ORDER the result by prefix length, e.g.

ORDER BY LENGTH(prefix) DESC LIMIT 1

4. The complete query will be,

SELECT * FROM prefixes
WHERE prefix LIKE (923344224088, 92334422408, 9233442240, 923344224,
92334422, 9233442, 923344, 92334, 9233, 923, 92, 9)
ORDER BY LENGTH(prefix) DESC LIMIT 1

Now the query takes less then 150 ms to execute. :-)

Here is an STL method that can generate this query, i am sure you can
convert it to any programming language of your choice easily.

=========================================================
*std::string GetQuery(std::string destination) {
        std::string query = "SELECT * FROM prefixes WHERE prefix LIKE ('" +
destination;

        for(int i=1; i<destination.length(); i++) {
                query += "','" + destination.substr(0, (i * -1));

        }

        query += "') ORDER BY LENGTH(prefix) DESC LIMIT 1";
        return query;
}
*=========================================================

I am pretty sure this query is 100% ANSI SQL compatible (
http://en.wikipedia.org/wiki/SQL).

Thank you.


On Wed, Oct 14, 2009 at 10:15 AM, Michael Giagnocavo <mgg at giagnocavo.net>wrote:

>  In our testing with SQL Server, we found that executing several queries
> for direct matches yielded far better performance than one query trying to
> check prefixes. (The column was also part of the clustered index, but AFAIK
> MySQL doesn?t support defining your own clustered indexes; you get the PK
> always.)
>
>
>
> -Michael
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Diego Viola
> *Sent:* Tuesday, October 13, 2009 7:54 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Some help with my post-paid billing
> project
>
>
>
> Wrong question.
>
> Is there a way to compare numbers with prefixes without using the prefix
> module?
>
> Diego
>
> On Wed, Oct 14, 2009 at 1:36 AM, Diego Viola <diego.viola at gmail.com>
> wrote:
>
> I'm using MySQL now but I will try PostgreSQL with the prefix module, is
> there a way to do that without the prefix module and with regular SQL?
>
> Any examples?
>
> Diego
>
>
>
> On Tue, Oct 13, 2009 at 10:45 PM, Even Andr? Fiskvik <grevenx at me.com>
> wrote:
>
> What database are you using?
> You could do this with regular SQL, but it would by a costly operation,
> for PostgreSQL we're using the prefix module:
> http://pgfoundry.org/projects/prefix/
>
> You can then match the closest prefix by using something like
> "WHERE myprefix_col @> caller_destination_number ORDER BY LENGTH
> (myprefix_col::text) LIMIT 1;"
>
>
> Best regards,
> Even Andr?
>
>
>
> On 13. okt. 2009, at 23.53, Diego Viola wrote:
>
> > Hello,
> >
> > I'm trying to write a post-paid billing script, I have the CDR on my
> > database and also a "rates" table, the CDR contains fields like
> > caller_destination_number, variable_duration, etc. and the rates
> > table contains: destination, prefix, rate (cost).
> >
> > The problem is that I can't just strip the destination number to
> > take the prefix from it because I have to deal with destination
> > numbers from different countries and they all have different prefix
> > lengths... so I need to find another way to take the prefix from the
> > destination number.
> >
> > Any ideas how to do this?
> >
> > Thanks,
> >
> > Diego
> >
>
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
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Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From tculjaga at gmail.com  Wed Oct 14 00:58:47 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 14 Oct 2009 09:58:47 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910141134270.11620@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<65d96fc80910130014hc5f5fcet677f87b7fb57559e@mail.gmail.com>
	<65d96fc80910132359i6d53d8ban10f943795481bf7c@mail.gmail.com>
	<Pine.LNX.4.64.0910141134270.11620@srv.icf.org.ru>
Message-ID: <65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>

2009/10/14 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-14 08:59 +0200, Tihomir Culjaga wrote
> freeswitch-users at lists.fre...:
>
> try sow start on h323 channel, there is a bug in faststart, i will fix it
> later.
>
>
there are few things,

1. capability PCMU/PCMA needs to be inverted
2. when you place outgoing calls SIP_user => FS => H323_endpoint FS
corediumps: what is the data format for bridge? Is that correct =>
data="h323/1001 at 10.1.1.1" ?
3. when you place incoming calls H323_endpoint => FS => SIPUser, the call
goes through but there is no audio. After H.225 connect and TCS/MSD, FS
stops sending RTP back to te originator.
4. when you place incoming calls H323_endpoint => FS => SIPUser and you
hangup from SIPUser side, the call is not released on H323 side...
switch_core_state_machine
complains about a wrong thread.


i will send you logs for:

1. slow start
2. slow start early h245
3. fast start tunneling true
4. faststart tunneling false
5. faststart with early h245 tunneling true
6. faststart with early h245 tunneling false


do you need a tcpdump for every scenario as well ?

T.
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From bottleman at icf.org.ru  Wed Oct 14 01:05:37 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Wed, 14 Oct 2009 12:05:37 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<65d96fc80910130014hc5f5fcet677f87b7fb57559e@mail.gmail.com>
	<65d96fc80910132359i6d53d8ban10f943795481bf7c@mail.gmail.com>
	<Pine.LNX.4.64.0910141134270.11620@srv.icf.org.ru>
	<65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910141202500.21535@srv.icf.org.ru>

On 2009-10-14 09:58 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

i need trace level 4 from mod_h323 and debug log of entire call, tcpdump may be needed later, i have no way 
to test it on this time, i do it later.


TC>2009/10/14 Georgiewskiy Yuriy <bottleman at icf.org.ru>
TC>
TC>> On 2009-10-14 08:59 +0200, Tihomir Culjaga wrote
TC>> freeswitch-users at lists.fre...:
TC>>
TC>> try sow start on h323 channel, there is a bug in faststart, i will fix it
TC>> later.
TC>>
TC>>
TC>there are few things,
TC>
TC>1. capability PCMU/PCMA needs to be inverted
TC>2. when you place outgoing calls SIP_user => FS => H323_endpoint FS
TC>corediumps: what is the data format for bridge? Is that correct =>
TC>data="h323/1001 at 10.1.1.1" ?
TC>3. when you place incoming calls H323_endpoint => FS => SIPUser, the call
TC>goes through but there is no audio. After H.225 connect and TCS/MSD, FS
TC>stops sending RTP back to te originator.
TC>4. when you place incoming calls H323_endpoint => FS => SIPUser and you
TC>hangup from SIPUser side, the call is not released on H323 side...
TC>switch_core_state_machine
TC>complains about a wrong thread.
TC>
TC>
TC>i will send you logs for:
TC>
TC>1. slow start
TC>2. slow start early h245
TC>3. fast start tunneling true
TC>4. faststart tunneling false
TC>5. faststart with early h245 tunneling true
TC>6. faststart with early h245 tunneling false
TC>
TC>
TC>do you need a tcpdump for every scenario as well ?
TC>
TC>T.
TC>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From tculjaga at gmail.com  Wed Oct 14 01:16:42 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 14 Oct 2009 10:16:42 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910141202500.21535@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<65d96fc80910130014hc5f5fcet677f87b7fb57559e@mail.gmail.com>
	<65d96fc80910132359i6d53d8ban10f943795481bf7c@mail.gmail.com>
	<Pine.LNX.4.64.0910141134270.11620@srv.icf.org.ru>
	<65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>
	<Pine.LNX.4.64.0910141202500.21535@srv.icf.org.ru>
Message-ID: <65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>

2009/10/14 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-14 09:58 +0200, Tihomir Culjaga wrote
> freeswitch-users at lists.fre...:
>
> i need trace level 4 from mod_h323 and debug log of entire call, tcpdump
> may be needed later, i have no way
> to test it on this time, i do it later.
>
>
> Ok, will generate this logs...

hope you will recover soon.

T.
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From diego.viola at gmail.com  Wed Oct 14 01:28:11 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Wed, 14 Oct 2009 08:28:11 +0000
Subject: [Freeswitch-users] Some help with my post-paid billing project
In-Reply-To: <4AD5324F.2030000@ttnc.co.uk>
References: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
	<74730C11-75D0-4BC0-9F63-8B6C9F0F499F@me.com>
	<86a32abc0910131836w5f113a16s247caf5eeb0f8fe4@mail.gmail.com>
	<4AD5324F.2030000@ttnc.co.uk>
Message-ID: <86a32abc0910140128y553786ean13d5699a799f116d@mail.gmail.com>

Nice, I just converted this to Ruby/Sequel.

DB[:rates].first{{prefix=>substring('number', 1, length(prefix))}}

Thanks for the help :).

On Wed, Oct 14, 2009 at 2:07 AM, TTNC - Adnan Barakat
<technical at ttnc.co.uk>wrote:

> Diego Viola wrote:
> > I'm using MySQL now but I will try PostgreSQL with the prefix module, is
> > there a way to do that without the prefix module and with regular SQL?
> >
> > Any examples?
> SELECT * FROM rates WHERE prefix = SUBSTRING('$NUMBER$', 1,
> LENGTH(prefix)) LIMIT 1
>
> Adnan
>
> > Diego
> >
> > On Tue, Oct 13, 2009 at 10:45 PM, Even Andr? Fiskvik <grevenx at me.com
> > <mailto:grevenx at me.com>> wrote:
> >
> >     What database are you using?
> >     You could do this with regular SQL, but it would by a costly
> operation,
> >     for PostgreSQL we're using the prefix module:
> >     http://pgfoundry.org/projects/prefix/
> >
> >     You can then match the closest prefix by using something like
> >     "WHERE myprefix_col @> caller_destination_number ORDER BY LENGTH
> >     (myprefix_col::text) LIMIT 1;"
> >
> >
> >     Best regards,
> >     Even Andr?
> >
> >
> >     On 13. okt. 2009, at 23.53, Diego Viola wrote:
> >
> >     > Hello,
> >     >
> >     > I'm trying to write a post-paid billing script, I have the CDR on
> my
> >     > database and also a "rates" table, the CDR contains fields like
> >     > caller_destination_number, variable_duration, etc. and the rates
> >     > table contains: destination, prefix, rate (cost).
> >     >
> >     > The problem is that I can't just strip the destination number to
> >     > take the prefix from it because I have to deal with destination
> >     > numbers from different countries and they all have different prefix
> >     > lengths... so I need to find another way to take the prefix from
> the
> >     > destination number.
> >     >
> >     > Any ideas how to do this?
> >     >
> >     > Thanks,
> >     >
> >     > Diego
> >     >
> >     > _______________________________________________
> >     > FreeSWITCH-users mailing list
> >     > FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-
> >     > users
> >     > http://www.freeswitch.org
> >
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From diego.viola at gmail.com  Wed Oct 14 01:42:13 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Wed, 14 Oct 2009 08:42:13 +0000
Subject: [Freeswitch-users] Some help with my post-paid billing project
In-Reply-To: <86a32abc0910140128y553786ean13d5699a799f116d@mail.gmail.com>
References: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
	<74730C11-75D0-4BC0-9F63-8B6C9F0F499F@me.com>
	<86a32abc0910131836w5f113a16s247caf5eeb0f8fe4@mail.gmail.com>
	<4AD5324F.2030000@ttnc.co.uk>
	<86a32abc0910140128y553786ean13d5699a799f116d@mail.gmail.com>
Message-ID: <86a32abc0910140142g3f3ee930p7afc6d39baf5b7ec@mail.gmail.com>

DB[:rates].where(:prefix => substring('number', 1, length(prefix)).first

Rather.

On Wed, Oct 14, 2009 at 8:28 AM, Diego Viola <diego.viola at gmail.com> wrote:

> Nice, I just converted this to Ruby/Sequel.
>
> DB[:rates].first{{prefix=>substring('number', 1, length(prefix))}}
>
> Thanks for the help :).
>
>
> On Wed, Oct 14, 2009 at 2:07 AM, TTNC - Adnan Barakat <
> technical at ttnc.co.uk> wrote:
>
>> Diego Viola wrote:
>> > I'm using MySQL now but I will try PostgreSQL with the prefix module, is
>> > there a way to do that without the prefix module and with regular SQL?
>> >
>> > Any examples?
>> SELECT * FROM rates WHERE prefix = SUBSTRING('$NUMBER$', 1,
>> LENGTH(prefix)) LIMIT 1
>>
>> Adnan
>>
>> > Diego
>> >
>> > On Tue, Oct 13, 2009 at 10:45 PM, Even Andr? Fiskvik <grevenx at me.com
>> > <mailto:grevenx at me.com>> wrote:
>> >
>> >     What database are you using?
>> >     You could do this with regular SQL, but it would by a costly
>> operation,
>> >     for PostgreSQL we're using the prefix module:
>> >     http://pgfoundry.org/projects/prefix/
>> >
>> >     You can then match the closest prefix by using something like
>> >     "WHERE myprefix_col @> caller_destination_number ORDER BY LENGTH
>> >     (myprefix_col::text) LIMIT 1;"
>> >
>> >
>> >     Best regards,
>> >     Even Andr?
>> >
>> >
>> >     On 13. okt. 2009, at 23.53, Diego Viola wrote:
>> >
>> >     > Hello,
>> >     >
>> >     > I'm trying to write a post-paid billing script, I have the CDR on
>> my
>> >     > database and also a "rates" table, the CDR contains fields like
>> >     > caller_destination_number, variable_duration, etc. and the rates
>> >     > table contains: destination, prefix, rate (cost).
>> >     >
>> >     > The problem is that I can't just strip the destination number to
>> >     > take the prefix from it because I have to deal with destination
>> >     > numbers from different countries and they all have different
>> prefix
>> >     > lengths... so I need to find another way to take the prefix from
>> the
>> >     > destination number.
>> >     >
>> >     > Any ideas how to do this?
>> >     >
>> >     > Thanks,
>> >     >
>> >     > Diego
>> >     >
>> >     > _______________________________________________
>> >     > FreeSWITCH-users mailing list
>> >     > FreeSWITCH-users at lists.freeswitch.org
>> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >     > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-
>> >     > users
>> >     > http://www.freeswitch.org
>> >
>> >
>> >     _______________________________________________
>> >     FreeSWITCH-users mailing list
>> >     FreeSWITCH-users at lists.freeswitch.org
>> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >     UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >     http://www.freeswitch.org
>> >
>> >
>> >
>> > ------------------------------------------------------------------------
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From god.nirvana at gmail.com  Wed Oct 14 01:45:55 2009
From: god.nirvana at gmail.com (qian ma)
Date: Wed, 14 Oct 2009 16:45:55 +0800
Subject: [Freeswitch-users] how can i get the caller-number when the caller
	in the conference
Message-ID: <49161ef40910140145r7b8d28dbu3d9e44b4c17547de@mail.gmail.com>

hi all:
      i can get the member_id in the conference,but how can can i get the
number??
      thx!
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From tculjaga at gmail.com  Wed Oct 14 02:15:30 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 14 Oct 2009 11:15:30 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<65d96fc80910130014hc5f5fcet677f87b7fb57559e@mail.gmail.com>
	<65d96fc80910132359i6d53d8ban10f943795481bf7c@mail.gmail.com>
	<Pine.LNX.4.64.0910141134270.11620@srv.icf.org.ru>
	<65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>
	<Pine.LNX.4.64.0910141202500.21535@srv.icf.org.ru>
	<65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>
Message-ID: <65d96fc80910140215y357fede1l37f6cba9725ddd89@mail.gmail.com>

On Wed, Oct 14, 2009 at 10:16 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:

>
>
> 2009/10/14 Georgiewskiy Yuriy <bottleman at icf.org.ru>
>
>> On 2009-10-14 09:58 +0200, Tihomir Culjaga wrote
>> freeswitch-users at lists.fre...:
>>
>> i need trace level 4 from mod_h323 and debug log of entire call, tcpdump
>> may be needed later, i have no way
>> to test it on this time, i do it later.
>>
>>

Here are the logs:

call flow is H323_endpoint => FS => SIP_user

slow start                => http://pastebin.freeswitch.org/10693
slow start w eH245            => http://pastebin.freeswitch.org/10694
fastStart w tunneling true         => http://pastebin.freeswitch.org/10701
fastStart w tunneling false        => http://pastebin.freeswitch.org/10702
fastStart w tunneling true eH245    => http://pastebin.freeswitch.org/10699
fastStart w tunneling false eH245    => http://pastebin.freeswitch.org/10700


In all cases, there is no audio on SIP User side... i see the SIP Phone
(terminator) sending RTP to FS but FS is not forwarding it back to the
originator. Also, i see originator (H323) sending RTP to FS but FS doesn't
forward it to the terminator (SIP).



call flow is SIP_user => FS => H323_endpoint is failing .. coredumped....
http://pastebin.freeswitch.org/10703



T.




>
>
> hope you will recover soon.
>
> T.
>
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From srinivas.ksvreddy at gmail.com  Wed Oct 14 02:49:40 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Wed, 14 Oct 2009 15:19:40 +0530
Subject: [Freeswitch-users] Fwd: Groups information in sqllite
In-Reply-To: <B3C83D64-BC34-4F67-B72E-9EAE6F3A362E@jerris.com>
References: <f8af5740910122301u338ca128macfc0b769ea0e961@mail.gmail.com>
	<f8af5740910122302i47ccde5bo9df2bae965408394@mail.gmail.com>
	<B3C83D64-BC34-4F67-B72E-9EAE6F3A362E@jerris.com>
Message-ID: <f8af5740910140249t67df3364k1bb81a9c544f185e@mail.gmail.com>

Hi michael,

Thank you very much for your valuable time,
you are right, i have seen freeswitch.xml.fsxml,
groups data is taking from these xml file. instead of taking from these xml
file can i take from sqllite?


Thanks
Srinivasula Reddy

On Tue, Oct 13, 2009 at 9:12 PM, Michael Jerris <mike at jerris.com> wrote:

> Group information is not stored in sqlite, it is pulled from the xml
> registry (switch_xml_locate_group function can find them) .  Also,
> please do not cross post between lists.
>
> http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups
> http://wiki.freeswitch.org/wiki/Mod_commands#in_group
> http://wiki.freeswitch.org/wiki/Mod_commands#group_call
>
> Mike
>
>
> On Oct 13, 2009, at 2:02 AM, srinivasula reddy wrote:
> > can any know where group information is exactly stored in sqllite
> > database, i have seen sip_registration here i can find the
> > registered users,
> > in the same way how i can i find the group information, and which
> > user belongs to which user?
> > any help would be great.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>



-- 
Srinivasula Reddy K
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From lakindia89 at gmail.com  Wed Oct 14 03:03:33 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Wed, 14 Oct 2009 15:33:33 +0530
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <87f2f3b90910132303i5966a477ue82667bef975cc93@mail.gmail.com>
References: <7d79b3930910090407m5dc81d12pf2b0a47f6017804@mail.gmail.com>
	<20091009123615.BC50A3FA6E9@mail.cune.org>
	<7d79b3930910100647u7fd11413ucec0a535d68b3e3a@mail.gmail.com>
	<87f2f3b90910101009j1295d812jd98cae91e30b1eff@mail.gmail.com>
	<7d79b3930910120454m1ffc6dd4u42b91a4349597166@mail.gmail.com>
	<87f2f3b90910121149x6e2114f0q996a262ad272a1f6@mail.gmail.com>
	<7d79b3930910130645y658ae041v8219c4c881fa5698@mail.gmail.com>
	<5E7A8451-6264-4822-8152-1B94240BDD48@freeswitch.org>
	<7d79b3930910132234k4b99c794q8854ce20d14ee74e@mail.gmail.com>
	<87f2f3b90910132303i5966a477ue82667bef975cc93@mail.gmail.com>
Message-ID: <7d79b3930910140303k4bff6d5fgbc55a173194e22c9@mail.gmail.com>

Hi,
Tried setting PCMA as first parameter in vars.xml, no use. Still I got the
same error.

Tried setting the absolute_codec_string to PCMA, but got the same error.

But I tried with a another parameter in openzap.conf.xml, which I got from
the following thread
http://www.nabble.com/Problem-dialing-out-via-E1-to22479047.html#a22626383

<param name="l1" value="alaw"/>

After that I was able to make outgoing call, with the dialplan that I've.

But I don't know why setting PCMA has not worked. Have any guess???

I planned to document this in the wiki.

I was struggling to originate a call from the CLI.
originate openzap/1/1/9952248266 openzap/1/1/9952248266 says "No route.
Aborting".

Have to find a way for this.

I sincerely thank you for your valuable time and effort on this matter.



On Wed, Oct 14, 2009 at 11:33 AM, Michael Collins <msc at freeswitch.org>wrote:

> Lak,
>
> Okay, it stood out right away: FS is trying u-law but Asterisk is trying
> A-law. I'm not sure where the codec for openzap gets selected but you can
> try modifying this line in vars.xml:
>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>
> Try putting PCMA first. Also, I've never tried this but perhaps in your
> dialplan you can set absolute_codec_string to PCMA. I'm very interesting in
> hearing about the results. Please try them as soon as you can and report
> back.
>
> Thanks,
> MC
>
>
> On Tue, Oct 13, 2009 at 10:34 PM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> Hi. I made the setup quickly with one of my colleague, and here is the
>> debug log given by asterisk
>>
>> Steps done by me :
>> 	+ Call Made from 04443902743 to 04439114600
>> 	+ When call lands to 04439114600, I just route it to 09176454982.
>>
>>
>> *CLI>
>> *CLI>
>> *CLI> pri debug span 1
>> Enabled debugging on span 1
>> *CLI>
>> *CLI>
>> *CLI>
>> *CLI> < Protocol Discriminator: Q.931 (8)  len=41
>> < Call Ref: len= 2 (reference 21/0x15) (Originator)
>>
>>
>> < Message type: SETUP (5)
>> < [a1]
>> < Sending Complete (len= 1)
>> < [04 03 80 90 a3]
>> < Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer capability: Speech (0)
>> <                              Ext: 1  Trans mode/rate: 64kbps, circuit-mode (16)
>>
>>
>> <                                User information layer 1: A-Law (35)
>> < [18 03 a1 83 9f]
>> < Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Preferred  Dchan: 0
>> <                        ChanSel: As indicated in following octets
>>
>>
>> <                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
>> <                       Ext: 1  Channel: 31 ]
>> < [6c 0c 01 81 34 34 34 33 39 30 32 37 34 33]
>> < Calling Number (len=14) [ Ext: 0  TON: Unknown Number Type (0)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
>>
>>
>> <                           Presentation: Presentation permitted, user number passed network screening (1)  '4443902743' ]
>> < [70 09 c1 33 39 31 31 34 36 30 30]
>> < Called Number (len=11) [ Ext: 1  TON: Subscriber Number (4)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)  '39114600' ]
>>
>>
>> -- Making new call for cr 21
>> -- Processing Q.931 Call Setup
>> -- Processing IE 161 (cs0, Sending Complete)
>> -- Processing IE 4 (cs0, Bearer Capability)
>> -- Processing IE 24 (cs0, Channel Identification)
>> -- Processing IE 108 (cs0, Calling Party Number)
>>
>>
>> -- Processing IE 112 (cs0, Called Party Number)
>> q931.c:3504 q931_receive: call 21 on channel 31 enters state 6 (Call Present)
>> q931.c:2769 q931_call_proceeding: call 21 on channel 31 enters state 9 (Incoming Call Proceeding)
>>
>> > Protocol Discriminator: Q.931 (8)  len=10
>> > Call Ref: len= 2 (reference 21/0x15) (Terminator)
>> > Message type: CALL PROCEEDING (2)
>>
>> > [18 03 a9 83 9f]
>> > Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  Dchan: 0
>> >                        ChanSel: As indicated in following octets
>>
>> >                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
>> >                       Ext: 1  Channel: 31 ]
>>
>>     -- Accepting call from '4443902743' to '39114600' on channel 0/31, span 1
>>     -- Executing [39114600 at incoming-from-pri:1] NoOp("Zap/31-1", "FMS Call Extn:39114600:| CID::| Channel:Zap/31-1:| ") in new stack
>>
>>
>>     -- Executing [39114600 at incoming-from-pri:2] Dial("Zap/31-1", "Zap/g1/w09176454982||rj") in new stack
>> -- Making new call for cr 32770
>>     -- Requested transfer capability: 0x00 - SPEECH
>>
>> > Protocol Discriminator: Q.931 (8)  len=43
>> > Call Ref: len= 2 (reference 2/0x2) (Originator)
>> > Message type: SETUP (5)
>> > [04 03 80 90 a3]
>> > Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer capability: Speech (0)
>> >                              Ext: 1  Trans mode/rate: 64kbps, circuit-mode (16)
>>
>> >                                User information layer 1: A-Law (35)
>> > [18 03 a9 83 81]
>> > Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  Dchan: 0
>>
>> >                        ChanSel: As indicated in following octets
>> >                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
>> >                       Ext: 1  Channel: 1 ]
>> > [6c 0c 00 81 34 34 34 33 39 30 32 37 34 33]
>> > Calling Number (len=14) [ Ext: 0  TON: Unknown Number Type (0)  NPI: Unknown Number Plan (0)
>>
>> >                           Presentation: Presentation permitted, user number passed network screening (1)  '4443902743' ]
>> > [70 0c 80 30 39 31 37 36 34 35 34 39 38 32]
>> > Called Number (len=14) [ Ext: 1  TON: Unknown Number Type (0)  NPI: Unknown Number Plan (0)  '09176454982' ]
>>
>> q931.c:3087 q931_setup: call 32770 on channel 1 enters state 1 (Call Initiated)
>>
>>
>>     -- Called g1/w09176454982
>> q931.c:2797 q931_alerting: call 21 on channel 31 enters state 7 (Call Received)
>>
>> > Protocol Discriminator: Q.931 (8)  len=9
>> > Call Ref: len= 2 (reference 21/0x15) (Terminator)
>> > Message type: ALERTING (1)
>> > [1e 02 81 88]
>> > Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Private network serving the local user (1)
>> >                               Ext: 1  Progress Description: Inband information or appropriate pattern now available. (8) ]
>>
>> < Protocol Discriminator: Q.931 (8)  len=10
>> < Call Ref: len= 2 (reference 2/0x2) (Terminator)
>> < Message type: CALL PROCEEDING (2)
>> < [18 03 a9 83 81]
>> < Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  Dchan: 0
>>
>>
>> <                        ChanSel: As indicated in following octets
>> <                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
>> <                       Ext: 1  Channel: 1 ]
>> -- Processing IE 24 (cs0, Channel Identification)
>>
>>
>> q931.c:3636 q931_receive: call 32770 on channel 1 enters state 3 (Outgoing call  Proceeding)
>>     -- Zap/1-1 is proceeding passing it to Zap/31-1
>> < Protocol Discriminator: Q.931 (8)  len=13
>> < Call Ref: len= 2 (reference 2/0x2) (Terminator)
>>
>>
>> < Message type: ALERTING (1)
>> < [1e 02 84 88]
>> < Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Public network serving the remote user (4)
>> <                               Ext: 1  Progress Description: Inband information or appropriate pattern now available. (8) ]
>>
>>
>> < [1e 02 84 82]
>> < Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Public network serving the remote user (4)
>> <                               Ext: 1  Progress Description: Called equipment is non-ISDN. (2) ]
>>
>>
>> -- Processing IE 30 (cs0, Progress Indicator)
>> -- Processing IE 30 (cs0, Progress Indicator)
>> q931.c:3549 q931_receive: call 32770 on channel 1 enters state 4 (Call Delivered)
>>     -- Zap/1-1 is ringing
>> < Protocol Discriminator: Q.931 (8)  len=9
>>
>>
>> < Call Ref: len= 2 (reference 2/0x2) (Terminator)
>> < Message type: PROGRESS (3)
>> < [1e 02 84 84]
>> < Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Public network serving the remote user (4)
>>
>>
>> <                               Ext: 1  Progress Description: Unknown (4) ]
>> -- Processing IE 30 (cs0, Progress Indicator)
>>     -- Zap/1-1 is making progress passing it to Zap/31-1
>> < Protocol Discriminator: Q.931 (8)  len=9
>>
>>
>> < Call Ref: len= 2 (reference 2/0x2) (Terminator)
>> < Message type: PROGRESS (3)
>> < [1e 02 84 88]
>> < Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Public network serving the remote user (4)
>>
>>
>> <                               Ext: 1  Progress Description: Inband information or appropriate pattern now available. (8) ]
>> -- Processing IE 30 (cs0, Progress Indicator)
>>     -- Zap/1-1 is making progress passing it to Zap/31-1
>>
>>
>> < Protocol Discriminator: Q.931 (8)  len=19
>> < Call Ref: len= 2 (reference 2/0x2) (Terminator)
>> < Message type: CONNECT (7)
>> < [4c 0c 01 83 39 31 37 36 34 35 34 39 38 32]
>> < Connected Number (len=14) [ Ext: 0  TON: Unknown Number Type (0)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
>>
>>
>> <                             Ext: 1  Presentation: Presentation allowed of network provided number (3)  '9176454982' ]
>> -- Processing IE 76 (cs0, Connected Number)
>> q931.c:3579 q931_receive: call 32770 on channel 1 enters state 10 (Active)
>>
>> > Protocol Discriminator: Q.931 (8)  len=5
>> > Call Ref: len= 2 (reference 2/0x2) (Originator)
>> > Message type: CONNECT ACKNOWLEDGE (15)
>>
>>     -- Zap/1-1 answered Zap/31-1
>> q931.c:2904 q931_connect: call 21 on channel 31 enters state 8 (Connect Request)
>>
>> > Protocol Discriminator: Q.931 (8)  len=14
>>
>> > Call Ref: len= 2 (reference 21/0x15) (Terminator)
>> > Message type: CONNECT (7)
>> > [18 03 a9 83 9f]
>> > Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  Dchan: 0
>> >                        ChanSel: As indicated in following octets
>> >                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
>> >                       Ext: 1  Channel: 31 ]
>> > [1e 02 81 82]
>> > Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Private network serving the local user (1)
>> >                               Ext: 1  Progress Description: Called equipment is non-ISDN. (2) ]
>>
>>     -- Native bridging Zap/31-1 and Zap/1-1
>> < Protocol Discriminator: Q.931 (8)  len=5
>> < Call Ref: len= 2 (reference 21/0x15) (Originator)
>> < Message type: CONNECT ACKNOWLEDGE (15)
>>
>>
>> q931.c:3664 q931_receive: call 21 on channel 31 enters state 10 (Active)
>> < Protocol Discriminator: Q.931 (8)  len=9
>> < Call Ref: len= 2 (reference 2/0x2) (Terminator)
>> < Message type: DISCONNECT (69)
>> < [08 02 80 90]
>>
>>
>> < Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  Location: User (0)
>> <                  Ext: 1  Cause: Normal Clearing (16), class = Normal Event (1) ]
>> -- Processing IE 8 (cs0, Cause)
>>
>> q931.c:3779 q931_receive: call 32770 on channel 1 enters state 12 (Disconnect Indication)
>>
>>     -- Channel 0/1, span 1 got hangup request, cause 16
>> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request
>> q931.c:2920 q931_release: call 32770 on channel 1 enters state 19 (Release Request)
>>
>> > Protocol Discriminator: Q.931 (8)  len=9
>> > Call Ref: len= 2 (reference 2/0x2) (Originator)
>> > Message type: RELEASE (77)
>> > [08 02 81 90]
>> > Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  Location: Private network serving the local user (1)
>> >                  Ext: 1  Cause: Normal Clearing (16), class = Normal Event (1) ]
>>
>>     -- Hungup 'Zap/1-1'
>>   == Spawn extension (incoming-from-pri, 39114600, 2) exited non-zero on 'Zap/31-1'
>>
>>
>>     -- Executing [h at incoming-from-pri:1] Hangup("Zap/31-1", "") in new stack
>>   == Spawn extension (incoming-from-pri, h, 1) exited non-zero on 'Zap/31-1'
>> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Active
>>
>>
>> q931.c:2968 q931_disconnect: call 21 on channel 31 enters state 11 (Disconnect Request)
>>
>> > Protocol Discriminator: Q.931 (8)  len=9
>> > Call Ref: len= 2 (reference 21/0x15) (Terminator)
>>
>> > Message type: DISCONNECT (69)
>> > [08 02 81 90]
>> > Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  Location: Private network serving the local user (1)
>>
>> >                  Ext: 1  Cause: Normal Clearing (16), class = Normal Event (1) ]
>>
>>     -- Hungup 'Zap/31-1'
>> < Protocol Discriminator: Q.931 (8)  len=5
>>
>> < Call Ref: len= 2 (reference 2/0x2) (Terminator)
>> < Message type: RELEASE COMPLETE (90)
>> q931.c:3719 q931_receive: call 32770 on channel 1 enters state 0 (Null)
>> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
>>
>>
>> NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
>> < Protocol Discriminator: Q.931 (8)  len=5
>> < Call Ref: len= 2 (reference 21/0x15) (Originator)
>> < Message type: RELEASE (77)
>> q931.c:3754 q931_receive: call 21 on channel 31 enters state 0 (Null)
>>
>>
>> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request
>>
>> > Protocol Discriminator: Q.931 (8)  len=9
>> > Call Ref: len= 2 (reference 21/0x15) (Terminator)
>>
>> > Message type: RELEASE COMPLETE (90)
>> > [08 02 81 90]
>> > Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  Location: Private network serving the local user (1)
>>
>> >                  Ext: 1  Cause: Normal Clearing (16), class = Normal Event (1) ]
>>
>> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
>>
>> NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
>>
>>
>> On Tue, Oct 13, 2009 at 9:16 PM, Michael S Collins <msc at freeswitch.org>wrote:
>>
>>>
>>> On Oct 13, 2009, at 6:45 AM, lakshmanan ganapathy <lakindia89 at gmail.com>
>>> wrote:
>>>
>>> We are using Reliance as the Carrier.
>>> I think, with this same Reliance carrier, in my office, they are able to
>>> make outgoing calls through asterisk+libpri.
>>>
>>>
>>> If that's the case I would be very interested in seeing a pri debug from
>>> a working call on an asterisk box. It might give us a clue as to what is not
>>> working. -MC
>>>
>>>
>>>
>>> On Tue, Oct 13, 2009 at 12:19 AM, Michael Collins < <msc at freeswitch.org>
>>> msc at freeswitch.org> wrote:
>>>
>>>> Lak,
>>>>
>>>> Okay I will need a little bit of time to dig into the IE's and what they
>>>> contain. In the meantime can you tell me who the carrier is? I'd like to
>>>> find out if they have some specific requirements. The fact that it doesn't
>>>> work with libpri surprises me because that would mean that Asterisk systems
>>>> would probably not work with this carrier as well.
>>>>
>>>> BTW, thanks for the very complete pastebin entries. :)
>>>> -MC
>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From nandy1925 at gmail.com  Wed Oct 14 03:41:59 2009
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Wed, 14 Oct 2009 18:41:59 +0800
Subject: [Freeswitch-users] Preannouncing a message before making a call
In-Reply-To: <20091013205114.GA4927@mad06.wl0.org>
References: <20091013205114.GA4927@mad06.wl0.org>
Message-ID: <7d0bfd8c0910140341l1ee4502fiaaff6c5087666048@mail.gmail.com>

hi simon, implementation is almost the same. here's my dialplan:

<extension name="say_destination_info">
 <condition field="destination_number" expression="^0044(\d)$">
   <action application="answer"/>
   <action application="playback" data="misc/dialing.wav"/>
   <action application="say" data="en name_spelled iterated $1"/>
   ... add more files to play here ...
   <action application="playback" data="misc/gateway.wav"/>
   <action application="bridge" data="sofia/gateway/$1/>
 </condition>
</extension>

warning: i have not tested this. i'm just showing for illustration purposes.

-nandy

On Wed, Oct 14, 2009 at 4:51 AM, Simon J Mudd <sjmudd at pobox.com> wrote:

> I'm looking to migrate to Freeswitch from Asterisk (SOHO usage).
> One thing I have configured at the moment which I'm unsure how to do in
> FreeSwitch is the following.
>
> When the call's dial pattern is recognised I've been getting Asterisk
> to say: "Dialing <destination> via <gateway>" where <destination> might
> be a country, or area and <gateway> may be the name of one of several
> SIP gateways I have configured.  Each phrase has been a prerecorded wav
> file played in sequence.
>
> [macro-XXXXX]
> exten => s,1,Playback(/etc/asterisk/wav/dialing)
> exten => s,2,Playback(/etc/asterisk/wav/${ARG2})    # Name of destination
> exten => s,3,Playback(/etc/asterisk/wav/via)
> exten => s,4,Playback(/etc/asterisk/wav/XXXXX)      # Gateway name
> exten => s,5,Dial(SIP/XXXXX/${ARG1},60,tT)          # ARG1 = Destination
> number
>
>
> # actual usage
> exten => _0044.,1,Macro(XXXXX,${EXTEN},touk)
>
> How would I approach doing something like this in FreeSwtich?
>
> Thanks for any pointers.
>
> Simon
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From Prometheus001 at gmx.net  Wed Oct 14 03:55:58 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Wed, 14 Oct 2009 12:55:58 +0200
Subject: [Freeswitch-users] T.38 via UPDATE request
Message-ID: <4AD5AE3E.9070206@gmx.net>

Hello,

we have the following problem.
2 Fax machines are communicating via Freeswitch. One is externally
attached via a Telco who is able to handle T.38. The other one is
attached locally.

When 2 Fax machines start syncing each other, the Telco sends a SIP
UPDATE message with T.38 SDP, as it detects fax during the fax negociations.
Freeswitch answers with an SIP OK message back to the telco, and I can
see the T.38 SDP on the debug console of freeswitch.
Then nothing happens any more until one of fax machines detects timeout.

We have set proxy-media to true. However is was done during call setup
when both machines communicated with G711 SDP.
The UPDATE message was commited by FS to the telco, but was not sent to
the other fax, so I think in this case Freeswitch is supposed to
transcode between T.38 and G711 which it cannot do, as we know.

How can I overcome this scenario? Is this a defect, should freeswitch
send the UPDATE message to the other fax?  Or is there a workaround?

Best regards
Peter

Here's the UPDATE message:
UPDATE sip:mod_sofia at 82.115.xx.xxx:5080 SIP/2.0.
Call-ID: 5283fe4e-334f-122d-d1b9-001517956764.
Contact: <sip:212.91.xxx.xxx:5060>.
Content-Type: application/sdp.
CSeq: 16340063 UPDATE.
From: <sip:06912345678 at sip.telco.de>;tag=00-08135-017e041a-4d21f6037.
Max-Forwards: 31.
Route: <sip:82.115.96.165;lr;ftag=pH663F4S02erm>.
To: "030987654321" <sip:030987654321 at 82.115.xx.xxx>;tag=pH663F4S02erm.
User-Agent: Cirpack/v4.41e (gw_sip).
Via: SIP/2.0/UDP 212.91.xxx.xxx:5060;branch=z9hG4bK-178C-1923B31.
Content-Length: 300.
.
v=0.
o=cp10 125551618103 125551618105 IN IP4 212.91.xxx.xx.
s=SIP Call.
c=IN IP4 212.91.xxx.xx.
t=0 0.
m=image 6860 udptl t38.
a=sendrecv.
a=T38FaxVersion:0.
a=T38MaxBitRate:9600.
a=T38FaxRateManagement:transferredTCF.
a=T38FaxMaxBuffer:1000.
a=T38FaxMaxDatagram:200.
a=T38FaxUdpEC:t38UDPRedundancy.


From srinivas.ksvreddy at gmail.com  Wed Oct 14 04:58:49 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Wed, 14 Oct 2009 17:28:49 +0530
Subject: [Freeswitch-users] adding new extension
Message-ID: <f8af5740910140458w4666ee2fwa93d1ecb70cc015a@mail.gmail.com>

Hi,

can any please let me know, how to add new extension(eg 1000.xml)
dynamically while running freeswitch?
while running freeswitch i have created new xml(eg: 1500.xml) file and i
have changed freeswitch.xml.fsxml still  not working,

Thanks
Srinivasula Reddy K
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From mgg at giagnocavo.net  Wed Oct 14 05:43:32 2009
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Wed, 14 Oct 2009 08:43:32 -0400
Subject: [Freeswitch-users] Some help with my post-paid billing project
In-Reply-To: <b16156850910140055r37581014iddf52d915aa7eaa4@mail.gmail.com>
References: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
	<74730C11-75D0-4BC0-9F63-8B6C9F0F499F@me.com>
	<86a32abc0910131836w5f113a16s247caf5eeb0f8fe4@mail.gmail.com>
	<86a32abc0910131853v244ed61bof9c24f345a0944b2@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C6703241D1E52@mse17be1.mse17.exchange.ms>
	<b16156850910140055r37581014iddf52d915aa7eaa4@mail.gmail.com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C6703241D1E8A@mse17be1.mse17.exchange.ms>

Wouldn?t that be SQL ?IN? instead of LIKE?

First off, see if you can do this in memory. If it?s just a ?rate sheet? list of prefixes + billing, it?s probably not that much data. I?d guess maybe 64-128 bytes per record (in Ruby it?s probably more, maybe 256 bytes?) inside a hashtable. Even so, we?re only talking about 1.5 to 3GB of RAM. Stick it in a nice performing dictionary of some sort and you?re set.

Just as a quick test, on one core of a Core2, it takes about 150ms to do 100K lookups against a 1M item hashtable (F#) ? that?s just a 1 line loop I tried out with no optimization, doing about 650K/sec. For comparison on the same machine, looping on a simple ?SELECT @@VERSION? command only achieved 5000 executions/second on a single thread ? that?s just the SQL overhead (to a local SQL instance!).

If the memory usage is too excessive, consider caching only subsets of the prefixes. Keep track of which leading prefix (1 or 2 digits) are the most used, and keep those in memory. Have a reasonably fast DB to fall back to.

If you?re sure the storage requirements or the change frequency truly eliminates keeping it in your own memory, here are some suggestions for working on DB performance:

A little perf gain is to pre-calculate possible lengths, so you?re performing less lookups. Just have a table based on the first 2 or 3 digits, giving you the valid prefix lengths. That can easily cut down the number of lookups from 10-12 to, say, 3-4. Compare the perf of temp tables, CTEs, and dynamic SQL ?IN?.

But I would be hesitant to do this from a higher level ORM or via dynamic SQL queries:

-          You?ll have a much harder time doing DB-specific optimizations/hints

-          Dynamic SQL needs to be parsed each time (possibly minor issue)

-          If the ad hoc queries are not properly parameterized, query plans might not get reused very well

-          You can easily send many times more data to the server (your entire SQL, versus just the sproc name + parameters)

Really look into doing programmability on the server. There?s all sorts of things you can do to minimize what you need to send to the database, and let the DB engine figure out the most efficient way. You might want to use a product that?s a bit more advanced than MySQL.

And finally, test, test, test. You should have a nice benchmark with several million rows and be able to execute, say, 10-50K lookups (on multiple threads) after each change. Modify your query, execute it, _inspect the execution plan_. Check your indexes ? make sure they are covering the query in the right way. (For example, if the table is (ID, Prefix, Price), you want an index of (Prefix, Price).) Review possible hints (for example, SQL Server usually needs a hint to choose a indexed view). Make sure your transaction level is set appropriately (can you do the query without locking?). Can you batch up several numbers in one command, saving on the per-command overhead?

Also consider it in light of the rest of your application. If everything is in the same database, then you can probably do the entire rate call/save cdr/update balances in a single command and transaction.

For us, our initial draft of routing inside SQL Server performed well under 100 calls/sec ? not usable. After a few days of playing with things, it went over 1000 calls/sec, which is comfortable enough for a $500 server. This is matching 100 gateways and dialplans with several thousand entries per dialplan, across 4 million+ routes, doing QBR/LCR in the process.

-Michael


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Muhammad Shahzad
Sent: Wednesday, October 14, 2009 1:55 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Some help with my post-paid billing project

I fully agree that direct matching is much faster then pattern matching in SQL.

One of my clients had same problem, he had around 12 million number prefixes in a table and during each call an AGI script use to query that table to find longest prefix match, but this use to take like 3-5 seconds even with indexed columns. So after a lots of R & D we come up with following logic,

1. We break the destination number length-wise, e.g. suppose number is 923344224088 then length chunks would be,

923344224088, 92334422408, 9233442240, 923344224, 92334422, 9233442, 923344, 92334, 9233, 923, 92, 9

2. Then use SQL LIKE function in WHERE clause (you can also use SQL OR function if your DBMS doesn't support SQL LIKE function), and pass all these chunks to it, e.g.

WHERE prefix LIKE (923344224088, 92334422408, 9233442240, 923344224, 92334422, 9233442, 923344, 92334, 9233, 923, 92, 9)

3. Lastly we ORDER the result by prefix length, e.g.

ORDER BY LENGTH(prefix) DESC LIMIT 1

4. The complete query will be,

SELECT * FROM prefixes
WHERE prefix LIKE (923344224088, 92334422408, 9233442240, 923344224, 92334422, 9233442, 923344, 92334, 9233, 923, 92, 9)
ORDER BY LENGTH(prefix) DESC LIMIT 1

Now the query takes less then 150 ms to execute. :-)

Here is an STL method that can generate this query, i am sure you can convert it to any programming language of your choice easily.

=========================================================
std::string GetQuery(std::string destination) {
        std::string query = "SELECT * FROM prefixes WHERE prefix LIKE ('" + destination;

        for(int i=1; i<destination.length(); i++) {
                query += "','" + destination.substr(0, (i * -1));

        }

        query += "') ORDER BY LENGTH(prefix) DESC LIMIT 1";
        return query;
}
=========================================================

I am pretty sure this query is 100% ANSI SQL compatible (http://en.wikipedia.org/wiki/SQL).

Thank you.
On Wed, Oct 14, 2009 at 10:15 AM, Michael Giagnocavo <mgg at giagnocavo.net<mailto:mgg at giagnocavo.net>> wrote:
In our testing with SQL Server, we found that executing several queries for direct matches yielded far better performance than one query trying to check prefixes. (The column was also part of the clustered index, but AFAIK MySQL doesn?t support defining your own clustered indexes; you get the PK always.)

-Michael

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Diego Viola
Sent: Tuesday, October 13, 2009 7:54 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Some help with my post-paid billing project

Wrong question.

Is there a way to compare numbers with prefixes without using the prefix module?

Diego
On Wed, Oct 14, 2009 at 1:36 AM, Diego Viola <diego.viola at gmail.com<mailto:diego.viola at gmail.com>> wrote:
I'm using MySQL now but I will try PostgreSQL with the prefix module, is there a way to do that without the prefix module and with regular SQL?

Any examples?

Diego

On Tue, Oct 13, 2009 at 10:45 PM, Even Andr? Fiskvik <grevenx at me.com<mailto:grevenx at me.com>> wrote:
What database are you using?
You could do this with regular SQL, but it would by a costly operation,
for PostgreSQL we're using the prefix module: http://pgfoundry.org/projects/prefix/

You can then match the closest prefix by using something like
"WHERE myprefix_col @> caller_destination_number ORDER BY LENGTH
(myprefix_col::text) LIMIT 1;"


Best regards,
Even Andr?


On 13. okt. 2009, at 23.53, Diego Viola wrote:

> Hello,
>
> I'm trying to write a post-paid billing script, I have the CDR on my
> database and also a "rates" table, the CDR contains fields like
> caller_destination_number, variable_duration, etc. and the rates
> table contains: destination, prefix, rate (cost).
>
> The problem is that I can't just strip the destination number to
> take the prefix from it because I have to deal with destination
> numbers from different countries and they all have different prefix
> lengths... so I need to find another way to take the prefix from the
> destination number.
>
> Any ideas how to do this?
>
> Thanks,
>
> Diego
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> users
> http://www.freeswitch.org


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



--
________________________________________________________
|                                                                                                |
| FATAL ERROR                                                           --- O X |
|_______________________________________________________|
|                        You have moved the mouse.                            |
| Windows must be restarted for the changes to take effect.   |
|                                        <OK>                                              |
####################################/


Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com<mailto:shari_786pk at hotmail.com>
Email: shaheryarkh at googlemail.com<mailto:shaheryarkh at googlemail.com>
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From Russell.Mosemann at cune.org  Wed Oct 14 06:21:18 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Wed, 14 Oct 2009 13:21:18 -0000
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <7d79b3930910140303k4bff6d5fgbc55a173194e22c9@mail.gmail.com>
Message-ID: <20091014132118.3A34C403DBC@mail.cune.org>

lakshmanan ganapathy <lakindia89 at gmail.com> said:

> I was struggling to originate a call from the CLI.
> originate openzap/1/1/9952248266 openzap/1/1/9952248266 says "No route.
> Aborting".

The syntax is incorrect.

originate openzap/1/1/9952248266 <extension>

For example,

openzap/1/1/9952248266 1234

See the wiki for details about originate. I think I included a link in a
previous message.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From shaheryarkh at googlemail.com  Wed Oct 14 06:22:51 2009
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Wed, 14 Oct 2009 19:22:51 +0600
Subject: [Freeswitch-users] Some help with my post-paid billing project
In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6703241D1E8A@mse17be1.mse17.exchange.ms>
References: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
	<74730C11-75D0-4BC0-9F63-8B6C9F0F499F@me.com>
	<86a32abc0910131836w5f113a16s247caf5eeb0f8fe4@mail.gmail.com>
	<86a32abc0910131853v244ed61bof9c24f345a0944b2@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C6703241D1E52@mse17be1.mse17.exchange.ms>
	<b16156850910140055r37581014iddf52d915aa7eaa4@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C6703241D1E8A@mse17be1.mse17.exchange.ms>
Message-ID: <b16156850910140622w78b356b4hd7280bc504416685@mail.gmail.com>

Oops, you are right, this is what happens when you are doing more then one
thing at a time, i was writing a db function while replying in this email
thread and confused LIKE with IN. There is one more mistake in my email if
any one noticed, that is, string class in STL does not support negative
length in method substr.

Thanks for correction.


On Wed, Oct 14, 2009 at 6:43 PM, Michael Giagnocavo <mgg at giagnocavo.net>wrote:

>  Wouldn?t that be SQL ?IN? instead of LIKE?
>
>
>
> First off, see if you can do this in memory. If it?s just a ?rate sheet?
> list of prefixes + billing, it?s probably not that much data. I?d guess
> maybe 64-128 bytes per record (in Ruby it?s probably more, maybe 256 bytes?)
> inside a hashtable. Even so, we?re only talking about 1.5 to 3GB of RAM.
> Stick it in a nice performing dictionary of some sort and you?re set.
>
>
>
> Just as a quick test, on one core of a Core2, it takes about 150ms to do
> 100K lookups against a 1M item hashtable (F#) ? that?s just a 1 line loop I
> tried out with no optimization, doing about 650K/sec. For comparison on the
> same machine, looping on a simple ?SELECT @@VERSION? command only achieved
> 5000 executions/second on a single thread ? that?s just the SQL overhead (to
> a local SQL instance!).
>
>
>
> If the memory usage is too excessive, consider caching only subsets of the
> prefixes. Keep track of which leading prefix (1 or 2 digits) are the most
> used, and keep those in memory. Have a reasonably fast DB to fall back to.
>
>
>
> If you?re sure the storage requirements or the change frequency truly
> eliminates keeping it in your own memory, here are some suggestions for
> working on DB performance:
>
>
>
> A little perf gain is to pre-calculate possible lengths, so you?re
> performing less lookups. Just have a table based on the first 2 or 3 digits,
> giving you the valid prefix lengths. That can easily cut down the number of
> lookups from 10-12 to, say, 3-4. Compare the perf of temp tables, CTEs, and
> dynamic SQL ?IN?.
>
>
>
> But I would be hesitant to do this from a higher level ORM or via dynamic
> SQL queries:
>
> -          You?ll have a much harder time doing DB-specific
> optimizations/hints
>
> -          Dynamic SQL needs to be parsed each time (possibly minor issue)
>
> -          If the ad hoc queries are not properly parameterized, query
> plans might not get reused very well
>
> -          You can easily send many times more data to the server (your
> entire SQL, versus just the sproc name + parameters)
>
>
>
> Really look into doing programmability on the server. There?s all sorts of
> things you can do to minimize what you need to send to the database, and let
> the DB engine figure out the most efficient way. You might want to use a
> product that?s a bit more advanced than MySQL.
>
>
>
> And finally, test, test, test. You should have a nice benchmark with
> several million rows and be able to execute, say, 10-50K lookups (on
> multiple threads) after each change. Modify your query, execute it, _*inspect
> the execution plan*_. Check your indexes ? make sure they are covering the
> query in the right way. (For example, if the table is (ID, Prefix, Price),
> you want an index of (Prefix, Price).) Review possible hints (for example,
> SQL Server usually needs a hint to choose a indexed view). Make sure your
> transaction level is set appropriately (can you do the query without
> locking?). Can you batch up several numbers in one command, saving on the
> per-command overhead?
>
>
>
> Also consider it in light of the rest of your application. If everything is
> in the same database, then you can probably do the entire rate call/save
> cdr/update balances in a single command and transaction.
>
>
>
> For us, our initial draft of routing inside SQL Server performed well under
> 100 calls/sec ? not usable. After a few days of playing with things, it went
> over 1000 calls/sec, which is comfortable enough for a $500 server. This is
> matching 100 gateways and dialplans with several thousand entries per
> dialplan, across 4 million+ routes, doing QBR/LCR in the process.
>
>
>
> -Michael
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Muhammad
> Shahzad
> *Sent:* Wednesday, October 14, 2009 1:55 AM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Some help with my post-paid billing
> project
>
>
>
> I fully agree that direct matching is much faster then pattern matching in
> SQL.
>
> One of my clients had same problem, he had around 12 million number
> prefixes in a table and during each call an AGI script use to query that
> table to find longest prefix match, but this use to take like 3-5 seconds
> even with indexed columns. So after a lots of R & D we come up with
> following logic,
>
> 1. We break the destination number length-wise, e.g. suppose number is
> 923344224088 then length chunks would be,
>
> 923344224088, 92334422408, 9233442240, 923344224, 92334422, 9233442,
> 923344, 92334, 9233, 923, 92, 9
>
> 2. Then use SQL LIKE function in WHERE clause (you can also use SQL OR
> function if your DBMS doesn't support SQL LIKE function), and pass all these
> chunks to it, e.g.
>
> WHERE prefix LIKE (923344224088, 92334422408, 9233442240, 923344224,
> 92334422, 9233442, 923344, 92334, 9233, 923, 92, 9)
>
> 3. Lastly we ORDER the result by prefix length, e.g.
>
> ORDER BY LENGTH(prefix) DESC LIMIT 1
>
> 4. The complete query will be,
>
> SELECT * FROM prefixes
> WHERE prefix LIKE (923344224088, 92334422408, 9233442240, 923344224,
> 92334422, 9233442, 923344, 92334, 9233, 923, 92, 9)
> ORDER BY LENGTH(prefix) DESC LIMIT 1
>
> Now the query takes less then 150 ms to execute. :-)
>
> Here is an STL method that can generate this query, i am sure you can
> convert it to any programming language of your choice easily.
>
> =========================================================
> *std::string GetQuery(std::string destination) {
>         std::string query = "SELECT * FROM prefixes WHERE prefix LIKE ('" +
> destination;
>
>         for(int i=1; i<destination.length(); i++) {
>                 query += "','" + destination.substr(0, (i * -1));
>
>         }
>
>         query += "') ORDER BY LENGTH(prefix) DESC LIMIT 1";
>         return query;
> }
> *=========================================================
>
> I am pretty sure this query is 100% ANSI SQL compatible (
> http://en.wikipedia.org/wiki/SQL).
>
> Thank you.
>
> On Wed, Oct 14, 2009 at 10:15 AM, Michael Giagnocavo <mgg at giagnocavo.net>
> wrote:
>
> In our testing with SQL Server, we found that executing several queries for
> direct matches yielded far better performance than one query trying to check
> prefixes. (The column was also part of the clustered index, but AFAIK MySQL
> doesn?t support defining your own clustered indexes; you get the PK always.)
>
>
>
> -Michael
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Diego Viola
> *Sent:* Tuesday, October 13, 2009 7:54 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Some help with my post-paid billing
> project
>
>
>
> Wrong question.
>
> Is there a way to compare numbers with prefixes without using the prefix
> module?
>
> Diego
>
> On Wed, Oct 14, 2009 at 1:36 AM, Diego Viola <diego.viola at gmail.com>
> wrote:
>
> I'm using MySQL now but I will try PostgreSQL with the prefix module, is
> there a way to do that without the prefix module and with regular SQL?
>
> Any examples?
>
> Diego
>
>
>
> On Tue, Oct 13, 2009 at 10:45 PM, Even Andr? Fiskvik <grevenx at me.com>
> wrote:
>
> What database are you using?
> You could do this with regular SQL, but it would by a costly operation,
> for PostgreSQL we're using the prefix module:
> http://pgfoundry.org/projects/prefix/
>
> You can then match the closest prefix by using something like
> "WHERE myprefix_col @> caller_destination_number ORDER BY LENGTH
> (myprefix_col::text) LIMIT 1;"
>
>
> Best regards,
> Even Andr?
>
>
>
> On 13. okt. 2009, at 23.53, Diego Viola wrote:
>
> > Hello,
> >
> > I'm trying to write a post-paid billing script, I have the CDR on my
> > database and also a "rates" table, the CDR contains fields like
> > caller_destination_number, variable_duration, etc. and the rates
> > table contains: destination, prefix, rate (cost).
> >
> > The problem is that I can't just strip the destination number to
> > take the prefix from it because I have to deal with destination
> > numbers from different countries and they all have different prefix
> > lengths... so I need to find another way to take the prefix from the
> > destination number.
> >
> > Any ideas how to do this?
> >
> > Thanks,
> >
> > Diego
> >
>
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> ________________________________________________________
> |
>                      |
> | FATAL ERROR                                                           ---
> O X |
> |_______________________________________________________|
> |                        You have moved the mouse.
>    |
> | Windows must be restarted for the changes to take effect.   |
> |                                        <OK>
>                |
> ####################################/
>
>
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
________________________________________________________
|
                     |
| FATAL ERROR                                                           ---
O X |
|_______________________________________________________|
|                        You have moved the mouse.
 |
| Windows must be restarted for the changes to take effect.   |
|                                        <OK>
               |
####################################/


Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From rupa at rupa.com  Wed Oct 14 06:36:07 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 14 Oct 2009 08:36:07 -0500
Subject: [Freeswitch-users] Some help with my post-paid billing project
In-Reply-To: <b16156850910140622w78b356b4hd7280bc504416685@mail.gmail.com>
References: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
	<74730C11-75D0-4BC0-9F63-8B6C9F0F499F@me.com>
	<86a32abc0910131836w5f113a16s247caf5eeb0f8fe4@mail.gmail.com>
	<86a32abc0910131853v244ed61bof9c24f345a0944b2@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C6703241D1E52@mse17be1.mse17.exchange.ms>
	<b16156850910140055r37581014iddf52d915aa7eaa4@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C6703241D1E8A@mse17be1.mse17.exchange.ms>
	<b16156850910140622w78b356b4hd7280bc504416685@mail.gmail.com>
Message-ID: <b39273530910140636j72652456ga0ace41468b5c546@mail.gmail.com>

I would still suggest using mod_lcr for this...  If you have any real
volume, use postgresql with the prefix module.

It also supports IN lists, OR lists, optional quoting (since mysql is
retarded), and custom sql so you can interface with whatever stored
proc or deal with whatever database table you may need to support.

It also supports multiple profiles (so you can charge different
amounts based on some other criteria like account code) and...
interstate/intrastate/intralata rating.  probably other stuff I forgot
to mention...

Let me just recommend to reuse what already exists and improve on that.

On Wed, Oct 14, 2009 at 8:22 AM, Muhammad Shahzad
<shaheryarkh at googlemail.com> wrote:
> Oops, you are right, this is what happens when you are doing more then one
> thing at a time, i was writing a db function while replying in this email
> thread and confused LIKE with IN. There is one more mistake in my email if
> any one noticed, that is, string class in STL does not support negative
> length in method substr.
>
> Thanks for correction.
>
>
> On Wed, Oct 14, 2009 at 6:43 PM, Michael Giagnocavo <mgg at giagnocavo.net>
> wrote:
>>
>> Wouldn?t that be SQL ?IN? instead of LIKE?
>>
>>
>>
>> First off, see if you can do this in memory. If it?s just a ?rate sheet?
>> list of prefixes + billing, it?s probably not that much data. I?d guess
>> maybe 64-128 bytes per record (in Ruby it?s probably more, maybe 256 bytes?)
>> inside a hashtable. Even so, we?re only talking about 1.5 to 3GB of RAM.
>> Stick it in a nice performing dictionary of some sort and you?re set.
>>
>>
>>
>> Just as a quick test, on one core of a Core2, it takes about 150ms to do
>> 100K lookups against a 1M item hashtable (F#) ? that?s just a 1 line loop I
>> tried out with no optimization, doing about 650K/sec. For comparison on the
>> same machine, looping on a simple ?SELECT @@VERSION? command only achieved
>> 5000 executions/second on a single thread ? that?s just the SQL overhead (to
>> a local SQL instance!).
>>
>>
>>
>> If the memory usage is too excessive, consider caching only subsets of the
>> prefixes. Keep track of which leading prefix (1 or 2 digits) are the most
>> used, and keep those in memory. Have a reasonably fast DB to fall back to.
>>
>>
>>
>> If you?re sure the storage requirements or the change frequency truly
>> eliminates keeping it in your own memory, here are some suggestions for
>> working on DB performance:
>>
>>
>>
>> A little perf gain is to pre-calculate possible lengths, so you?re
>> performing less lookups. Just have a table based on the first 2 or 3 digits,
>> giving you the valid prefix lengths. That can easily cut down the number of
>> lookups from 10-12 to, say, 3-4. Compare the perf of temp tables, CTEs, and
>> dynamic SQL ?IN?.
>>
>>
>>
>> But I would be hesitant to do this from a higher level ORM or via dynamic
>> SQL queries:
>>
>> -????????? You?ll have a much harder time doing DB-specific
>> optimizations/hints
>>
>> -????????? Dynamic SQL needs to be parsed each time (possibly minor issue)
>>
>> -????????? If the ad hoc queries are not properly parameterized, query
>> plans might not get reused very well
>>
>> -????????? You can easily send many times more data to the server (your
>> entire SQL, versus just the sproc name + parameters)
>>
>>
>>
>> Really look into doing programmability on the server. There?s all sorts of
>> things you can do to minimize what you need to send to the database, and let
>> the DB engine figure out the most efficient way. You might want to use a
>> product that?s a bit more advanced than MySQL.
>>
>>
>>
>> And finally, test, test, test. You should have a nice benchmark with
>> several million rows and be able to execute, say, 10-50K lookups (on
>> multiple threads) after each change. Modify your query, execute it, _inspect
>> the execution plan_. Check your indexes ? make sure they are covering the
>> query in the right way. (For example, if the table is (ID, Prefix, Price),
>> you want an index of (Prefix, Price).) Review possible hints (for example,
>> SQL Server usually needs a hint to choose a indexed view). Make sure your
>> transaction level is set appropriately (can you do the query without
>> locking?). Can you batch up several numbers in one command, saving on the
>> per-command overhead?
>>
>>
>>
>> Also consider it in light of the rest of your application. If everything
>> is in the same database, then you can probably do the entire rate call/save
>> cdr/update balances in a single command and transaction.
>>
>>
>>
>> For us, our initial draft of routing inside SQL Server performed well
>> under 100 calls/sec ? not usable. After a few days of playing with things,
>> it went over 1000 calls/sec, which is comfortable enough for a $500 server.
>> This is matching 100 gateways and dialplans with several thousand entries
>> per dialplan, across 4 million+ routes, doing QBR/LCR in the process.
>>
>>
>>
>> -Michael
>>
>>
>>
>>
>>
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Muhammad
>> Shahzad
>> Sent: Wednesday, October 14, 2009 1:55 AM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Some help with my post-paid billing
>> project
>>
>>
>>
>> I fully agree that direct matching is much faster then pattern matching in
>> SQL.
>>
>> One of my clients had same problem, he had around 12 million number
>> prefixes in a table and during each call an AGI script use to query that
>> table to find longest prefix match, but this use to take like 3-5 seconds
>> even with indexed columns. So after a lots of R & D we come up with
>> following logic,
>>
>> 1. We break the destination number length-wise, e.g. suppose number is
>> 923344224088 then length chunks would be,
>>
>> 923344224088, 92334422408, 9233442240, 923344224, 92334422, 9233442,
>> 923344, 92334, 9233, 923, 92, 9
>>
>> 2. Then use SQL LIKE function in WHERE clause (you can also use SQL OR
>> function if your DBMS doesn't support SQL LIKE function), and pass all these
>> chunks to it, e.g.
>>
>> WHERE prefix LIKE (923344224088, 92334422408, 9233442240, 923344224,
>> 92334422, 9233442, 923344, 92334, 9233, 923, 92, 9)
>>
>> 3. Lastly we ORDER the result by prefix length, e.g.
>>
>> ORDER BY LENGTH(prefix) DESC LIMIT 1
>>
>> 4. The complete query will be,
>>
>> SELECT * FROM prefixes
>> WHERE prefix LIKE (923344224088, 92334422408, 9233442240, 923344224,
>> 92334422, 9233442, 923344, 92334, 9233, 923, 92, 9)
>> ORDER BY LENGTH(prefix) DESC LIMIT 1
>>
>> Now the query takes less then 150 ms to execute. :-)
>>
>> Here is an STL method that can generate this query, i am sure you can
>> convert it to any programming language of your choice easily.
>>
>> =========================================================
>> std::string GetQuery(std::string destination) {
>> ??????? std::string query = "SELECT * FROM prefixes WHERE prefix LIKE ('"
>> + destination;
>>
>> ??????? for(int i=1; i<destination.length(); i++) {
>> ??????????????? query += "','" + destination.substr(0, (i * -1));
>>
>> ??????? }
>>
>> ??????? query += "') ORDER BY LENGTH(prefix) DESC LIMIT 1";
>> ??????? return query;
>> }
>> =========================================================
>>
>> I am pretty sure this query is 100% ANSI SQL compatible
>> (http://en.wikipedia.org/wiki/SQL).
>>
>> Thank you.
>>
>> On Wed, Oct 14, 2009 at 10:15 AM, Michael Giagnocavo <mgg at giagnocavo.net>
>> wrote:
>>
>> In our testing with SQL Server, we found that executing several queries
>> for direct matches yielded far better performance than one query trying to
>> check prefixes. (The column was also part of the clustered index, but AFAIK
>> MySQL doesn?t support defining your own clustered indexes; you get the PK
>> always.)
>>
>>
>>
>> -Michael
>>
>>
>>
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Diego
>> Viola
>> Sent: Tuesday, October 13, 2009 7:54 PM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Some help with my post-paid billing
>> project
>>
>>
>>
>> Wrong question.
>>
>> Is there a way to compare numbers with prefixes without using the prefix
>> module?
>>
>> Diego
>>
>> On Wed, Oct 14, 2009 at 1:36 AM, Diego Viola <diego.viola at gmail.com>
>> wrote:
>>
>> I'm using MySQL now but I will try PostgreSQL with the prefix module, is
>> there a way to do that without the prefix module and with regular SQL?
>>
>> Any examples?
>>
>> Diego
>>
>>
>>
>> On Tue, Oct 13, 2009 at 10:45 PM, Even Andr? Fiskvik <grevenx at me.com>
>> wrote:
>>
>> What database are you using?
>> You could do this with regular SQL, but it would by a costly operation,
>> for PostgreSQL we're using the prefix module:
>> http://pgfoundry.org/projects/prefix/
>>
>> You can then match the closest prefix by using something like
>> "WHERE myprefix_col @> caller_destination_number ORDER BY LENGTH
>> (myprefix_col::text) LIMIT 1;"
>>
>>
>> Best regards,
>> Even Andr?
>>
>> On 13. okt. 2009, at 23.53, Diego Viola wrote:
>>
>> > Hello,
>> >
>> > I'm trying to write a post-paid billing script, I have the CDR on my
>> > database and also a "rates" table, the CDR contains fields like
>> > caller_destination_number, variable_duration, etc. and the rates
>> > table contains: destination, prefix, rate (cost).
>> >
>> > The problem is that I can't just strip the destination number to
>> > take the prefix from it because I have to deal with destination
>> > numbers from different countries and they all have different prefix
>> > lengths... so I need to find another way to take the prefix from the
>> > destination number.
>> >
>> > Any ideas how to do this?
>> >
>> > Thanks,
>> >
>> > Diego
>> >
>>
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> > users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> --
>> ________________________________________________________
>> |
>> ? ? ? ? ? ? ? ? ? ? ? ?|
>> | FATAL ERROR
>> --- O X |
>> |_______________________________________________________|
>> | ? ? ? ? ? ? ? ? ? ? ? ?You have moved the mouse.
>> ? ?|
>> | Windows must be restarted for the changes to take effect. ? |
>> | ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?<OK>
>> ? ? ? ? ? ? ? ? ?|
>> ####################################/
>>
>>
>> Muhammad Shahzad
>> -----------------------------------
>> CISCO Rich Media Communication Specialist (CRMCS)
>> CISCO Certified Network Associate (CCNA)
>> Cell: +92 334 422 40 88
>> MSN: shari_786pk at hotmail.com
>> Email: shaheryarkh at googlemail.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> ________________________________________________________
> |
> ? ? ? ? ? ? ? ? ? ? ?|
> | FATAL ERROR ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ---
> O X |
> |_______________________________________________________|
> | ? ? ? ? ? ? ? ? ? ? ? ?You have moved the mouse.
> ?|
> | Windows must be restarted for the changes to take effect. ? |
> | ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?<OK>
> ? ? ? ? ? ? ? ?|
> ####################################/
>
>
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa


From nagalenoj at gmail.com  Wed Oct 14 07:10:42 2009
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Wed, 14 Oct 2009 19:40:42 +0530
Subject: [Freeswitch-users] Bridge in event socket - async mode
Message-ID: <fe67a3340910140710g136e1aa2x7a23aaff77a7b6c0@mail.gmail.com>

Hi,

I'm using event outboud socket(perl) in async mode.

Scenario where I face problem:
When a call comes to an extension(1000), my program will play some message
to the user, and get some DTMF.
I'll get the DTMF event(as I'm in async mode) and store the digits in a
variable.

When he presses "#", I call $con->execute("bridge","user/$dtmf"), to bridge
him to the dialed number.

It is bridging sometimes, and sometimes it is not bridging.

Is it the problem with async mode??
I also set $con->setEventLock("true"), before executing the bridge. But
still the problem persists.

I've seen the logs too, I didn't find the bridge application getting
executed.
What makes the problem here..??

The freeswitch log, when bridge is happening:
    http://pastebin.freeswitch.org/10704

The freeswitch log, when bridging is not happening:
    http://pastebin.freeswitch.org/10705

Code:

 # some statements

if($name eq "CHANNEL_BRIDGE") { # checking for events
.
}
elsif ($name eq "DTMF") {
   my $digit = $e->getHeader("dtmf-digit");
	
   if($digit eq '#') {
	print "Going to bridge\n";
	$con->execute("bridge","user/$dtmf_digit");		
   }
   else {
	$dtmf_digit.=$digit;
   }
}


-- 
Regards,
Nagalenoj H.
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From anthony.minessale at gmail.com  Wed Oct 14 07:26:39 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 14 Oct 2009 09:26:39 -0500
Subject: [Freeswitch-users] Bridge in event socket - async mode
In-Reply-To: <fe67a3340910140710g136e1aa2x7a23aaff77a7b6c0@mail.gmail.com>
References: <fe67a3340910140710g136e1aa2x7a23aaff77a7b6c0@mail.gmail.com>
Message-ID: <191c3a030910140726y2198213fhc97f3c2d0c3aeb5d@mail.gmail.com>

try putting
ESL::eslSetLogLevel(7);
at the top so you can get a trace of the esl data on the stdout from your
script
also put some debug code to confirm you are getting the digit


On Wed, Oct 14, 2009 at 9:10 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> Hi,
>
> I'm using event outboud socket(perl) in async mode.
>
> Scenario where I face problem:
> When a call comes to an extension(1000), my program will play some message
> to the user, and get some DTMF.
> I'll get the DTMF event(as I'm in async mode) and store the digits in a
> variable.
>
> When he presses "#", I call $con->execute("bridge","user/$dtmf"), to bridge
> him to the dialed number.
>
> It is bridging sometimes, and sometimes it is not bridging.
>
> Is it the problem with async mode??
> I also set $con->setEventLock("true"), before executing the bridge. But
> still the problem persists.
>
> I've seen the logs too, I didn't find the bridge application getting
> executed.
> What makes the problem here..??
>
> The freeswitch log, when bridge is happening:
>     http://pastebin.freeswitch.org/10704
>
> The freeswitch log, when bridging is not happening:
>     http://pastebin.freeswitch.org/10705
>
> Code:
>
>  # some statements
>
> if($name eq "CHANNEL_BRIDGE") { # checking for events
> .
> }
> elsif ($name eq "DTMF") {
>    my $digit = $e->getHeader("dtmf-digit");
> 	
>    if($digit eq '#') {
>
> 	print "Going to bridge\n";
> 	$con->execute("bridge","user/$dtmf_digit");		
>    }
>    else {
> 	$dtmf_digit.=$digit;
>    }
> }
>
>
> --
> Regards,
> Nagalenoj H.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From odermann at googlemail.com  Wed Oct 14 07:31:31 2009
From: odermann at googlemail.com (Dennis)
Date: Wed, 14 Oct 2009 16:31:31 +0200
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <65d96fc80910132348t202905fbub57cc4c814eb4e21@mail.gmail.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
	<191c3a030910130803k321f39e3n265e64ab3c84a5e6@mail.gmail.com>
	<5e414ed0910130826j11d45a66q86d785f900e4196e@mail.gmail.com>
	<65d96fc80910131012v1f72ac49o6bfe9f2267e541c5@mail.gmail.com>
	<191c3a030910131101p418e0961t373db68285248ff6@mail.gmail.com>
	<538F4CF43B294D0EACABA281E1913B4C@chaos>
	<65d96fc80910131224m73f35b87g355be818ad3c0caf@mail.gmail.com>
	<48B637E0506F4A809E4C0CAF39BD08BB@chaos>
	<65d96fc80910132348t202905fbub57cc4c814eb4e21@mail.gmail.com>
Message-ID: <5e414ed0910140731w1c7ebedr150e69cda8073155@mail.gmail.com>

the thing we want to make working nicer is the following:

we want the main/basic phonenumber (123456) to be reachable, so that
the telephone rings. but we also want it to be expandable with
ddi-digits.

example: dial the 123456 to reach the company, dial the 123456 1 to
reach the support.

in the moment our carrier waits x (milli-)seconds to see, if the
123456 was dialed or if there were more digits attached. after this
time, the carrier sends us the whole number as a block (may it be
123456 or 1234561) and we answer.

we want to avoid the waiting time the carrier needs to wait for
(possible) more digits. we want to receive every single digit from our
carrier and we want to tell fs, when the number is complete and fs can
answer.


From anthony.minessale at gmail.com  Wed Oct 14 07:47:19 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 14 Oct 2009 09:47:19 -0500
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <5e414ed0910140731w1c7ebedr150e69cda8073155@mail.gmail.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
	<191c3a030910130803k321f39e3n265e64ab3c84a5e6@mail.gmail.com>
	<5e414ed0910130826j11d45a66q86d785f900e4196e@mail.gmail.com>
	<65d96fc80910131012v1f72ac49o6bfe9f2267e541c5@mail.gmail.com>
	<191c3a030910131101p418e0961t373db68285248ff6@mail.gmail.com>
	<538F4CF43B294D0EACABA281E1913B4C@chaos>
	<65d96fc80910131224m73f35b87g355be818ad3c0caf@mail.gmail.com>
	<48B637E0506F4A809E4C0CAF39BD08BB@chaos>
	<65d96fc80910132348t202905fbub57cc4c814eb4e21@mail.gmail.com>
	<5e414ed0910140731w1c7ebedr150e69cda8073155@mail.gmail.com>
Message-ID: <191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com>

So with overlap you will have to keep refusing the call until the right
amount of digits are dialed.
This mode would send 1 then 12 then 123 then 1234 then 12345 then 123456 as
they were being dialed.
once you have 123456 won't you still be unsure if he will type the next 1 or
not and be forced to refuse it and wait anyway?

This feature is mostly used to allow the server to accept the call the
instant it knows there is an exact match in the dialplan or a non-existent
number has been dialed.  For instance, if 1234 was a unique ext, it could
instantly return since it knows.


On Wed, Oct 14, 2009 at 9:31 AM, Dennis <odermann at googlemail.com> wrote:

> the thing we want to make working nicer is the following:
>
> we want the main/basic phonenumber (123456) to be reachable, so that
> the telephone rings. but we also want it to be expandable with
> ddi-digits.
>
> example: dial the 123456 to reach the company, dial the 123456 1 to
> reach the support.
>
> in the moment our carrier waits x (milli-)seconds to see, if the
> 123456 was dialed or if there were more digits attached. after this
> time, the carrier sends us the whole number as a block (may it be
> 123456 or 1234561) and we answer.
>
> we want to avoid the waiting time the carrier needs to wait for
> (possible) more digits. we want to receive every single digit from our
> carrier and we want to tell fs, when the number is complete and fs can
> answer.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Wed Oct 14 07:52:14 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 14 Oct 2009 09:52:14 -0500
Subject: [Freeswitch-users] mod_socket: custom timeout for HEARTBEAT
	event
In-Reply-To: <d211f54a0910132331j14ccf510xd87bcb092a43f0b3@mail.gmail.com>
References: <d211f54a0910110959y3d3deed7s43201a3e39756585@mail.gmail.com>
	<86a32abc0910111444j1ab2e446p3cb94b0ae04126e3@mail.gmail.com>
	<97B4BDC9-5B11-4A98-B1BF-CC00A82ECBD9@jerris.com>
	<d211f54a0910112238i5f69275ey7cad437e47e6b567@mail.gmail.com>
	<191c3a030910120726p46d6410ejd94b61612686fbbc@mail.gmail.com>
	<d211f54a0910130832m25aaadd3t4d6e938d1c71d701@mail.gmail.com>
	<87f2f3b90910131148m1853a219md8788f55e40ef4c3@mail.gmail.com>
	<d211f54a0910132331j14ccf510xd87bcb092a43f0b3@mail.gmail.com>
Message-ID: <191c3a030910140752v2360c4d1r536f7eefa97603d6@mail.gmail.com>

the heartbeat you were seeing that needs a uuid was the session heartbeat
which is a per-call heartbeat.
indeed, the system heartbeat is not customizable.


On Wed, Oct 14, 2009 at 1:31 AM, Artem Shiyanov <shiyanov at gmail.com> wrote:

> Finally!!
> Thank you Michael, I didn't know about "status" app. It satisfies all my
> desires.
>
> And again,
> thanks for all the community for the strong support!
>
> Artem
>
>
>
>
> On Tue, Oct 13, 2009 at 10:48 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Tue, Oct 13, 2009 at 8:32 AM, Artem Shiyanov <shiyanov at gmail.com>wrote:
>>
>>> Sorry for my foolishness but I stil can't grasp it.
>>> I'm developing app based on inbound mod_event_socket and I don't know how
>>> to run "enable_heartbeat" or " sched_heartbeat" without specifying any alive
>>> session uuid. I tried to use "create_uuid" and send mentioned  commands to
>>> the created uuid  but this approach doesn't work. I do need to monitor the
>>> FS state itself rather then any particular call.
>>>
>>> Please, enlighten me!
>>> Artem
>>>
>>
>> So if I understand correctly, you want to see the system heartbeat more
>> frequently than every 20 seconds? What, exactly, is in the heartbeat that
>> you need? You can get the same information by sending the API "status" and
>> reading the results.
>>
>> If I read the source code correctly it looks like the system heartbeat is
>> hard-coded at 20 seconds. See switch_core.c, line 155. However, before you
>> go messing with a core file I would make sure that you can't already use the
>> tools that are available, like sending a "status" API every 5 seconds (or
>> whatever interval you need).
>>
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From jmesquita at freeswitch.org  Wed Oct 14 08:08:46 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 14 Oct 2009 12:08:46 -0300
Subject: [Freeswitch-users] adding new extension
In-Reply-To: <f8af5740910140458w4666ee2fwa93d1ecb70cc015a@mail.gmail.com>
References: <f8af5740910140458w4666ee2fwa93d1ecb70cc015a@mail.gmail.com>
Message-ID: <ebbc2d550910140808h3bc09fb2s53224c76389efe08@mail.gmail.com>

Never _EVER_ change the freeswitch.xml.fsxml file directly. If you need
something dynamic you would have to implement directory using mod_xml_curl,
otherwise, you change the files on the conf/ tree and do a reloadxml on the
CLI.

JM

On Wed, Oct 14, 2009 at 8:58 AM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

> Hi,
>
> can any please let me know, how to add new extension(eg 1000.xml)
> dynamically while running freeswitch?
> while running freeswitch i have created new xml(eg: 1500.xml) file and i
> have changed freeswitch.xml.fsxml still  not working,
>
> Thanks
> Srinivasula Reddy K
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Wed Oct 14 08:11:59 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 14 Oct 2009 12:11:59 -0300
Subject: [Freeswitch-users] Some help with my post-paid billing project
In-Reply-To: <b39273530910140636j72652456ga0ace41468b5c546@mail.gmail.com>
References: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
	<74730C11-75D0-4BC0-9F63-8B6C9F0F499F@me.com>
	<86a32abc0910131836w5f113a16s247caf5eeb0f8fe4@mail.gmail.com>
	<86a32abc0910131853v244ed61bof9c24f345a0944b2@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C6703241D1E52@mse17be1.mse17.exchange.ms>
	<b16156850910140055r37581014iddf52d915aa7eaa4@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C6703241D1E8A@mse17be1.mse17.exchange.ms>
	<b16156850910140622w78b356b4hd7280bc504416685@mail.gmail.com>
	<b39273530910140636j72652456ga0ace41468b5c546@mail.gmail.com>
Message-ID: <ebbc2d550910140811q2e5d93cbx171d2cf358aa921f@mail.gmail.com>

Could not agree more with Rupa. mod_lcr is an awesome piece of software.
Thank you Rupa for that.

JM

On Wed, Oct 14, 2009 at 10:36 AM, Rupa Schomaker <rupa at rupa.com> wrote:

> I would still suggest using mod_lcr for this...  If you have any real
> volume, use postgresql with the prefix module.
>
> It also supports IN lists, OR lists, optional quoting (since mysql is
> retarded), and custom sql so you can interface with whatever stored
> proc or deal with whatever database table you may need to support.
>
> It also supports multiple profiles (so you can charge different
> amounts based on some other criteria like account code) and...
> interstate/intrastate/intralata rating.  probably other stuff I forgot
> to mention...
>
> Let me just recommend to reuse what already exists and improve on that.
>
> On Wed, Oct 14, 2009 at 8:22 AM, Muhammad Shahzad
> <shaheryarkh at googlemail.com> wrote:
> > Oops, you are right, this is what happens when you are doing more then
> one
> > thing at a time, i was writing a db function while replying in this email
> > thread and confused LIKE with IN. There is one more mistake in my email
> if
> > any one noticed, that is, string class in STL does not support negative
> > length in method substr.
> >
> > Thanks for correction.
> >
> >
> > On Wed, Oct 14, 2009 at 6:43 PM, Michael Giagnocavo <mgg at giagnocavo.net>
> > wrote:
> >>
> >> Wouldn?t that be SQL ?IN? instead of LIKE?
> >>
> >>
> >>
> >> First off, see if you can do this in memory. If it?s just a ?rate sheet?
> >> list of prefixes + billing, it?s probably not that much data. I?d guess
> >> maybe 64-128 bytes per record (in Ruby it?s probably more, maybe 256
> bytes?)
> >> inside a hashtable. Even so, we?re only talking about 1.5 to 3GB of RAM.
> >> Stick it in a nice performing dictionary of some sort and you?re set.
> >>
> >>
> >>
> >> Just as a quick test, on one core of a Core2, it takes about 150ms to do
> >> 100K lookups against a 1M item hashtable (F#) ? that?s just a 1 line
> loop I
> >> tried out with no optimization, doing about 650K/sec. For comparison on
> the
> >> same machine, looping on a simple ?SELECT @@VERSION? command only
> achieved
> >> 5000 executions/second on a single thread ? that?s just the SQL overhead
> (to
> >> a local SQL instance!).
> >>
> >>
> >>
> >> If the memory usage is too excessive, consider caching only subsets of
> the
> >> prefixes. Keep track of which leading prefix (1 or 2 digits) are the
> most
> >> used, and keep those in memory. Have a reasonably fast DB to fall back
> to.
> >>
> >>
> >>
> >> If you?re sure the storage requirements or the change frequency truly
> >> eliminates keeping it in your own memory, here are some suggestions for
> >> working on DB performance:
> >>
> >>
> >>
> >> A little perf gain is to pre-calculate possible lengths, so you?re
> >> performing less lookups. Just have a table based on the first 2 or 3
> digits,
> >> giving you the valid prefix lengths. That can easily cut down the number
> of
> >> lookups from 10-12 to, say, 3-4. Compare the perf of temp tables, CTEs,
> and
> >> dynamic SQL ?IN?.
> >>
> >>
> >>
> >> But I would be hesitant to do this from a higher level ORM or via
> dynamic
> >> SQL queries:
> >>
> >> -          You?ll have a much harder time doing DB-specific
> >> optimizations/hints
> >>
> >> -          Dynamic SQL needs to be parsed each time (possibly minor
> issue)
> >>
> >> -          If the ad hoc queries are not properly parameterized, query
> >> plans might not get reused very well
> >>
> >> -          You can easily send many times more data to the server (your
> >> entire SQL, versus just the sproc name + parameters)
> >>
> >>
> >>
> >> Really look into doing programmability on the server. There?s all sorts
> of
> >> things you can do to minimize what you need to send to the database, and
> let
> >> the DB engine figure out the most efficient way. You might want to use a
> >> product that?s a bit more advanced than MySQL.
> >>
> >>
> >>
> >> And finally, test, test, test. You should have a nice benchmark with
> >> several million rows and be able to execute, say, 10-50K lookups (on
> >> multiple threads) after each change. Modify your query, execute it,
> _inspect
> >> the execution plan_. Check your indexes ? make sure they are covering
> the
> >> query in the right way. (For example, if the table is (ID, Prefix,
> Price),
> >> you want an index of (Prefix, Price).) Review possible hints (for
> example,
> >> SQL Server usually needs a hint to choose a indexed view). Make sure
> your
> >> transaction level is set appropriately (can you do the query without
> >> locking?). Can you batch up several numbers in one command, saving on
> the
> >> per-command overhead?
> >>
> >>
> >>
> >> Also consider it in light of the rest of your application. If everything
> >> is in the same database, then you can probably do the entire rate
> call/save
> >> cdr/update balances in a single command and transaction.
> >>
> >>
> >>
> >> For us, our initial draft of routing inside SQL Server performed well
> >> under 100 calls/sec ? not usable. After a few days of playing with
> things,
> >> it went over 1000 calls/sec, which is comfortable enough for a $500
> server.
> >> This is matching 100 gateways and dialplans with several thousand
> entries
> >> per dialplan, across 4 million+ routes, doing QBR/LCR in the process.
> >>
> >>
> >>
> >> -Michael
> >>
> >>
> >>
> >>
> >>
> >> From: freeswitch-users-bounces at lists.freeswitch.org
> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Muhammad
> >> Shahzad
> >> Sent: Wednesday, October 14, 2009 1:55 AM
> >> To: freeswitch-users at lists.freeswitch.org
> >> Subject: Re: [Freeswitch-users] Some help with my post-paid billing
> >> project
> >>
> >>
> >>
> >> I fully agree that direct matching is much faster then pattern matching
> in
> >> SQL.
> >>
> >> One of my clients had same problem, he had around 12 million number
> >> prefixes in a table and during each call an AGI script use to query that
> >> table to find longest prefix match, but this use to take like 3-5
> seconds
> >> even with indexed columns. So after a lots of R & D we come up with
> >> following logic,
> >>
> >> 1. We break the destination number length-wise, e.g. suppose number is
> >> 923344224088 then length chunks would be,
> >>
> >> 923344224088, 92334422408, 9233442240, 923344224, 92334422, 9233442,
> >> 923344, 92334, 9233, 923, 92, 9
> >>
> >> 2. Then use SQL LIKE function in WHERE clause (you can also use SQL OR
> >> function if your DBMS doesn't support SQL LIKE function), and pass all
> these
> >> chunks to it, e.g.
> >>
> >> WHERE prefix LIKE (923344224088, 92334422408, 9233442240, 923344224,
> >> 92334422, 9233442, 923344, 92334, 9233, 923, 92, 9)
> >>
> >> 3. Lastly we ORDER the result by prefix length, e.g.
> >>
> >> ORDER BY LENGTH(prefix) DESC LIMIT 1
> >>
> >> 4. The complete query will be,
> >>
> >> SELECT * FROM prefixes
> >> WHERE prefix LIKE (923344224088, 92334422408, 9233442240, 923344224,
> >> 92334422, 9233442, 923344, 92334, 9233, 923, 92, 9)
> >> ORDER BY LENGTH(prefix) DESC LIMIT 1
> >>
> >> Now the query takes less then 150 ms to execute. :-)
> >>
> >> Here is an STL method that can generate this query, i am sure you can
> >> convert it to any programming language of your choice easily.
> >>
> >> =========================================================
> >> std::string GetQuery(std::string destination) {
> >>         std::string query = "SELECT * FROM prefixes WHERE prefix LIKE
> ('"
> >> + destination;
> >>
> >>         for(int i=1; i<destination.length(); i++) {
> >>                 query += "','" + destination.substr(0, (i * -1));
> >>
> >>         }
> >>
> >>         query += "') ORDER BY LENGTH(prefix) DESC LIMIT 1";
> >>         return query;
> >> }
> >> =========================================================
> >>
> >> I am pretty sure this query is 100% ANSI SQL compatible
> >> (http://en.wikipedia.org/wiki/SQL).
> >>
> >> Thank you.
> >>
> >> On Wed, Oct 14, 2009 at 10:15 AM, Michael Giagnocavo <
> mgg at giagnocavo.net>
> >> wrote:
> >>
> >> In our testing with SQL Server, we found that executing several queries
> >> for direct matches yielded far better performance than one query trying
> to
> >> check prefixes. (The column was also part of the clustered index, but
> AFAIK
> >> MySQL doesn?t support defining your own clustered indexes; you get the
> PK
> >> always.)
> >>
> >>
> >>
> >> -Michael
> >>
> >>
> >>
> >> From: freeswitch-users-bounces at lists.freeswitch.org
> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Diego
> >> Viola
> >> Sent: Tuesday, October 13, 2009 7:54 PM
> >> To: freeswitch-users at lists.freeswitch.org
> >> Subject: Re: [Freeswitch-users] Some help with my post-paid billing
> >> project
> >>
> >>
> >>
> >> Wrong question.
> >>
> >> Is there a way to compare numbers with prefixes without using the prefix
> >> module?
> >>
> >> Diego
> >>
> >> On Wed, Oct 14, 2009 at 1:36 AM, Diego Viola <diego.viola at gmail.com>
> >> wrote:
> >>
> >> I'm using MySQL now but I will try PostgreSQL with the prefix module, is
> >> there a way to do that without the prefix module and with regular SQL?
> >>
> >> Any examples?
> >>
> >> Diego
> >>
> >>
> >>
> >> On Tue, Oct 13, 2009 at 10:45 PM, Even Andr? Fiskvik <grevenx at me.com>
> >> wrote:
> >>
> >> What database are you using?
> >> You could do this with regular SQL, but it would by a costly operation,
> >> for PostgreSQL we're using the prefix module:
> >> http://pgfoundry.org/projects/prefix/
> >>
> >> You can then match the closest prefix by using something like
> >> "WHERE myprefix_col @> caller_destination_number ORDER BY LENGTH
> >> (myprefix_col::text) LIMIT 1;"
> >>
> >>
> >> Best regards,
> >> Even Andr?
> >>
> >> On 13. okt. 2009, at 23.53, Diego Viola wrote:
> >>
> >> > Hello,
> >> >
> >> > I'm trying to write a post-paid billing script, I have the CDR on my
> >> > database and also a "rates" table, the CDR contains fields like
> >> > caller_destination_number, variable_duration, etc. and the rates
> >> > table contains: destination, prefix, rate (cost).
> >> >
> >> > The problem is that I can't just strip the destination number to
> >> > take the prefix from it because I have to deal with destination
> >> > numbers from different countries and they all have different prefix
> >> > lengths... so I need to find another way to take the prefix from the
> >> > destination number.
> >> >
> >> > Any ideas how to do this?
> >> >
> >> > Thanks,
> >> >
> >> > Diego
> >> >
> >>
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> > users
> >> > http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >> --
> >> ________________________________________________________
> >> |
> >>                        |
> >> | FATAL ERROR
> >> --- O X |
> >> |_______________________________________________________|
> >> |                        You have moved the mouse.
> >>    |
> >> | Windows must be restarted for the changes to take effect.   |
> >> |                                        <OK>
> >>                  |
> >> ####################################/
> >>
> >>
> >> Muhammad Shahzad
> >> -----------------------------------
> >> CISCO Rich Media Communication Specialist (CRMCS)
> >> CISCO Certified Network Associate (CCNA)
> >> Cell: +92 334 422 40 88
> >> MSN: shari_786pk at hotmail.com
> >> Email: shaheryarkh at googlemail.com
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > --
> > ________________________________________________________
> > |
> >                      |
> > | FATAL ERROR
> ---
> > O X |
> > |_______________________________________________________|
> > |                        You have moved the mouse.
> >  |
> > | Windows must be restarted for the changes to take effect.   |
> > |                                        <OK>
> >                |
> > ####################################/
> >
> >
> > Muhammad Shahzad
> > -----------------------------------
> > CISCO Rich Media Communication Specialist (CRMCS)
> > CISCO Certified Network Associate (CCNA)
> > Cell: +92 334 422 40 88
> > MSN: shari_786pk at hotmail.com
> > Email: shaheryarkh at googlemail.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Wed Oct 14 08:11:33 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 14 Oct 2009 08:11:33 -0700
Subject: [Freeswitch-users] Fwd: Groups information in sqllite
In-Reply-To: <f8af5740910140249t67df3364k1bb81a9c544f185e@mail.gmail.com>
References: <f8af5740910122301u338ca128macfc0b769ea0e961@mail.gmail.com>
	<f8af5740910122302i47ccde5bo9df2bae965408394@mail.gmail.com>
	<B3C83D64-BC34-4F67-B72E-9EAE6F3A362E@jerris.com>
	<f8af5740910140249t67df3364k1bb81a9c544f185e@mail.gmail.com>
Message-ID: <42BA0F36-1BF7-45B3-A50A-321E2879C003@freeswitch.org>

I think we have said NO more than once in the thread... so to clear  
this up RIGHT now without coding anything else you can NOT load your  
groups from SQLITE.

/b

On Oct 14, 2009, at 2:49 AM, srinivasula reddy wrote:

> groups data is taking from these xml file. instead of taking from  
> these xml file can i take from sqllite?



From dipen at entvoice.com  Wed Oct 14 05:54:28 2009
From: dipen at entvoice.com (dipen at entvoice.com)
Date: Wed, 14 Oct 2009 08:54:28 -0400
Subject: [Freeswitch-users] Mod_java loading error
Message-ID: <34733.1255524868@entvoice.com>

Hi Baskar,

Can you tell us how did you solve your problem of Loading mod_java.
I mean which steps to follow to solve the problem.

Currently i am getting the error..

 [ERR] modjava.c:133 load_config() Error loading /opt/SDK/jdk/jre/lib/i386/client/libjvm.so
 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_java.so
**Module load routine returned an error**

Thanks & Regards
Dipen Velani.


On Wed 15/04/09  3:15 AM , Baskar <yudha2008 at gmail.com> wrote:

> Hi,
> 
>    Now i can able to load the mod_java in the freeswitch console.
> 
> After that i have followed these method to run the PhoneTest.java
> 1) VERIFIED MY CLASSPATH IN THE JAVA.CONF.XML:   
> 
> 2)MY PHONETEST.CLASS IS LOCATED IN /USR/LOCAL/FREESWITCH/SCRIPT -
> DIRECTORY, SAME AS WHERE FREESWITCH.JAR IS LOCATED 
> 
> 3) I CREATED PHONETEST.JAR FILE FROM PHONETEST.CLASS 
> 
> 4)CHANGED MY DIALPLAN TO:  
> Then i have complied and Run the PhoneTest.java  by this methods
> 
> Method1:
> [localhost scripts]# javac -cp freeswitch.jar PhoneTest.java 
> [localhost scripts]# java -cp freeswitch.jar PhoneTest
> Exception in thread "main" java.lang.NoClassDefFoundError: PhoneTest
> Caused by: java.lang.ClassNotFoundException: PhoneTest
>         at
> java.net.URLClassLoader$1.run(URLClassLoader.java:200)
>         at java.security.AccessController.doPrivileged(Native
> Method)
>         at
> java.net.URLClassLoader.findClass(URLClassLoader.java:188)
>         at
> java.lang.ClassLoader.loadClass(ClassLoader.java:306)
>         at
> sun.misc.Launcher$AppClassLoader.loadClass(Launcher.java:276)
>         at
> java.lang.ClassLoader.loadClass(ClassLoader.java:251)
>         at
> java.lang.ClassLoader.loadClassInternal(ClassLoader.java:319)
> 
> Method2:
> [localhost scripts]# javac -cp PhoneTest.jar:freeswitch.jar
> PhoneTest.java
> [localhost scripts]# java -cp PhoneTest.jar:freeswitch.jar PhoneTest
> Exception in thread "main" java.lang.NoSuchMethodError: main
> 
> But I get these above errors .So Please Correct me where i am wrong
> and assist me how to compile and run the PhoneTest.java.
> -- 
> Warm Regards,
> N.Baskar
> 
> 
> 
---- Msg sent via @Mail - http://atmail.com/


From nbelov at abisoft.spb.ru  Wed Oct 14 05:59:25 2009
From: nbelov at abisoft.spb.ru (Nikita Belov)
Date: Wed, 14 Oct 2009 16:59:25 +0400
Subject: [Freeswitch-users] conference call
Message-ID: <064A5201C10D4D20AC76EB93CA5EA4B5@abisoft.biz>

HI all,

 

I want to configure FS to make special conference call between three users
(A, B, C).  In this conference C will hear A and B, but A will hear only B.
Can I make it using FS API commands? Does anybody know which approach is
better to use?

 

___

 

Thanks, Nikita

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From szasz.szabolcs at gmail.com  Tue Oct 13 23:16:37 2009
From: szasz.szabolcs at gmail.com (Szasz Szabolcs)
Date: Wed, 14 Oct 2009 09:16:37 +0300
Subject: [Freeswitch-users] TLS Audiocodes
Message-ID: <4a4d27d30910132316g206580f0ya32287291af00c5b@mail.gmail.com>

Hi,

Did anybody set up TLS between Freeswitch and Audiodes MP11X ? I got to work
TLS between freeswitch  and a softphone (phonerlite), but I have problem
with Audiocodes during the TLS authentication. I've loaded the certification
but it still doesn't work. Can I debug the tls in freeswitch? Please help
me.

Thank you in advance.

Szasz Szabolcs
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From szasz.szabolcs at gmail.com  Wed Oct 14 03:02:20 2009
From: szasz.szabolcs at gmail.com (Szasz Szabolcs)
Date: Wed, 14 Oct 2009 13:02:20 +0300
Subject: [Freeswitch-users] TLS Audiocodes
In-Reply-To: <4a4d27d30910132316g206580f0ya32287291af00c5b@mail.gmail.com>
References: <4a4d27d30910132316g206580f0ya32287291af00c5b@mail.gmail.com>
Message-ID: <4a4d27d30910140302s7c4e109j3cee9a6fe0d755db@mail.gmail.com>

Hello,

Turning on debug I got the following messages:

tls_connect(0x942e8f0): events NEGOTIATING
tls_connect(0x942e8f0): events NEGOTIATING
tls_connect(0x942e8f0): TLS setup failed
(error:00000001:lib(0):func(0):reason(1))
tport_close(0x942e8f0): tls/82.77.201.227:64571/sips

Szasz Szabolcs

On Wed, Oct 14, 2009 at 9:16 AM, Szasz Szabolcs <szasz.szabolcs at gmail.com>wrote:

> Hi,
>
> Did anybody set up TLS between Freeswitch and Audiodes MP11X ? I got to
> work TLS between freeswitch  and a softphone (phonerlite), but I have
> problem with Audiocodes during the TLS authentication. I've loaded the
> certification but it still doesn't work. Can I debug the tls in freeswitch?
> Please help me.
>
> Thank you in advance.
>
> Szasz Szabolcs
>
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From larclap at yahoo.com  Wed Oct 14 08:28:05 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Wed, 14 Oct 2009 08:28:05 -0700
Subject: [Freeswitch-users] lua script causing FreeSwitch to crash?
In-Reply-To: <87f2f3b90910131312k7dac8980h70ce8be9c07aa7b2@mail.gmail.com>
References: <005a01ca4c3c$c183ceb0$448b6c10$@com>
	<87f2f3b90910131312k7dac8980h70ce8be9c07aa7b2@mail.gmail.com>
Message-ID: <000601ca4ce2$ed1de640$c759b2c0$@com>

Michael,

 

I upgraded to v15152. There is no apparent change in the behavior. When I
call into a number handled by the lua script, I may get a fast busy signal,
or a recording saying the connection cannot be made.

 

But FreeSwitch is removed from memory as a result. I cannot see from the log
that the FreeSwitch ended.

 

Is there anything I can try on my side?

 

http://pastebin.freeswitch.org/10707

 

Lars

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, October 13, 2009 1:12 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] lua script causing FreeSwitch to crash?

 

 

On Tue, Oct 13, 2009 at 12:38 PM, Lars Zeb <larclap at yahoo.com> wrote:

http://pastebin.freeswitch.org/10686

 

I am trying to use a lua script for inbound calls. The caller hears a busy
signal. After this call fails to go through, FreeSwitch is no longer in
memory - it does not appear in  'ps -ef' output.

 

I don't have a clue what I might be doing incorrectly in the script in order
to cause this to happen. There is no indication in the log that FreeSwitch
has terminated. But it is not resident in memory.

 

I am using v15134. I had been using this same script without incident as
late as v14929. Until I can resolve the issue, I have removed the call to
lua from the dialplan and done all I could in it instead.

We sent out a notice about a bug in event socket between revs 15119 and
15134. Update to latest trunk. (FYI, the fix was in r15135, so you just
barely missed it.)

After you update, let us know if the problem persists. 
-MC

 

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From jmesquita at freeswitch.org  Wed Oct 14 08:38:22 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 14 Oct 2009 12:38:22 -0300
Subject: [Freeswitch-users] conference call
In-Reply-To: <064A5201C10D4D20AC76EB93CA5EA4B5@abisoft.biz>
References: <064A5201C10D4D20AC76EB93CA5EA4B5@abisoft.biz>
Message-ID: <ebbc2d550910140838v3125579ev9fa16e3e83d25770@mail.gmail.com>

Look at eavesdrop on the wiki.

JM

2009/10/14 Nikita Belov <nbelov at abisoft.spb.ru>

>  HI all,
>
>
>
> I want to configure FS to make special conference call between three users
> (A, B, C).  In this conference C will hear A and B, but A will hear only B.
> Can I make it using FS API commands? Does anybody know which approach is
> better to use?
>
>
>
> ___
>
>
>
> Thanks, Nikita
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Wed Oct 14 09:02:09 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 14 Oct 2009 11:02:09 -0500
Subject: [Freeswitch-users] lua script causing FreeSwitch to crash?
In-Reply-To: <000601ca4ce2$ed1de640$c759b2c0$@com>
References: <005a01ca4c3c$c183ceb0$448b6c10$@com>
	<87f2f3b90910131312k7dac8980h70ce8be9c07aa7b2@mail.gmail.com>
	<000601ca4ce2$ed1de640$c759b2c0$@com>
Message-ID: <191c3a030910140902j401a9c6dg7704616d27bac051@mail.gmail.com>

By which process did you upgrade?
Did you use "make current"
if so, run freeswitch from the directory you built it from then reporduce
the problem and watch for it to crash and type

./support-d/fscore_pb

at your shell to upload a crash report to pastebin and tell us the url it
generated.


On Wed, Oct 14, 2009 at 10:28 AM, Lars Zeb <larclap at yahoo.com> wrote:

>  Michael,
>
>
>
> I upgraded to v15152. There is no apparent change in the behavior. When I
> call into a number handled by the lua script, I may get a fast busy signal,
> or a recording saying the connection cannot be made.
>
>
>
> But FreeSwitch is removed from memory as a result. I cannot see from the
> log that the FreeSwitch ended.
>
>
>
> Is there anything I can try on my side?
>
>
>
> http://pastebin.freeswitch.org/10707
>
>
>
> Lars
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Tuesday, October 13, 2009 1:12 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] lua script causing FreeSwitch to crash?
>
>
>
>
>
> On Tue, Oct 13, 2009 at 12:38 PM, Lars Zeb <larclap at yahoo.com> wrote:
>
> http://pastebin.freeswitch.org/10686
>
>
>
> I am trying to use a lua script for inbound calls. The caller hears a busy
> signal. After this call fails to go through, FreeSwitch is no longer in
> memory ? it does not appear in  ?ps ?ef? output.
>
>
>
> I don?t have a clue what I might be doing incorrectly in the script in
> order to cause this to happen. There is no indication in the log that
> FreeSwitch has terminated. But it is not resident in memory.
>
>
>
> I am using v15134. I had been using this same script without incident as
> late as v14929. Until I can resolve the issue, I have removed the call to
> lua from the dialplan and done all I could in it instead.
>
> We sent out a notice about a bug in event socket between revs 15119 and
> 15134. Update to latest trunk. (FYI, the fix was in r15135, so you just
> barely missed it.)
>
> After you update, let us know if the problem persists.
> -MC
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From bjbrashier at gmail.com  Wed Oct 14 09:11:51 2009
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Wed, 14 Oct 2009 09:11:51 -0700
Subject: [Freeswitch-users] how can i get the caller-number when the
	caller in the conference
In-Reply-To: <49161ef40910140145r7b8d28dbu3d9e44b4c17547de@mail.gmail.com>
References: <49161ef40910140145r7b8d28dbu3d9e44b4c17547de@mail.gmail.com>
Message-ID: <7bcfdd290910140911mfe30ccclea156e5dc397a6bb@mail.gmail.com>

What method are you trying to use?

If you're using the event socket, it's in several of the parameters.
For instance, with an add-member event, you could look at
Caller-Username, Caller-Caller-ID-Name, or Caller-Caller-ID-Number
depending on your specific needs. There's also a uuid available if
that's better.

BB

On Wed, Oct 14, 2009 at 1:45 AM, qian ma <god.nirvana at gmail.com> wrote:
> hi all:
> ??????i can get the member_id in the conference,but how can can i get the
> number??
> ????? thx!
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From larclap at yahoo.com  Wed Oct 14 10:09:17 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Wed, 14 Oct 2009 10:09:17 -0700
Subject: [Freeswitch-users] lua script causing FreeSwitch to crash?
In-Reply-To: <191c3a030910140902j401a9c6dg7704616d27bac051@mail.gmail.com>
References: <005a01ca4c3c$c183ceb0$448b6c10$@com>	<87f2f3b90910131312k7dac8980h70ce8be9c07aa7b2@mail.gmail.com>	<000601ca4ce2$ed1de640$c759b2c0$@com>
	<191c3a030910140902j401a9c6dg7704616d27bac051@mail.gmail.com>
Message-ID: <005b01ca4cf1$10a30970$31e91c50$@com>

I did svn up, ./configure, make and make install.

 

Do you want me to do make current before proceeding, or just try to make it
stop with the current build from the above commands?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Wednesday, October 14, 2009 9:02 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] lua script causing FreeSwitch to crash?

 

By which process did you upgrade?
Did you use "make current"
if so, run freeswitch from the directory you built it from then reporduce
the problem and watch for it to crash and type

./support-d/fscore_pb

at your shell to upload a crash report to pastebin and tell us the url it
generated.



On Wed, Oct 14, 2009 at 10:28 AM, Lars Zeb <larclap at yahoo.com> wrote:

Michael,

 

I upgraded to v15152. There is no apparent change in the behavior. When I
call into a number handled by the lua script, I may get a fast busy signal,
or a recording saying the connection cannot be made.

 

But FreeSwitch is removed from memory as a result. I cannot see from the log
that the FreeSwitch ended.

 

Is there anything I can try on my side?

 

http://pastebin.freeswitch.org/10707

 

Lars

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, October 13, 2009 1:12 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] lua script causing FreeSwitch to crash?

 

 

On Tue, Oct 13, 2009 at 12:38 PM, Lars Zeb <larclap at yahoo.com> wrote:

http://pastebin.freeswitch.org/10686

 

I am trying to use a lua script for inbound calls. The caller hears a busy
signal. After this call fails to go through, FreeSwitch is no longer in
memory - it does not appear in  'ps -ef' output.

 

I don't have a clue what I might be doing incorrectly in the script in order
to cause this to happen. There is no indication in the log that FreeSwitch
has terminated. But it is not resident in memory.

 

I am using v15134. I had been using this same script without incident as
late as v14929. Until I can resolve the issue, I have removed the call to
lua from the dialplan and done all I could in it instead.

We sent out a notice about a bug in event socket between revs 15119 and
15134. Update to latest trunk. (FYI, the fix was in r15135, so you just
barely missed it.)

After you update, let us know if the problem persists. 
-MC

 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:213-799-1400

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From ryannyl at gmail.com  Wed Oct 14 10:40:55 2009
From: ryannyl at gmail.com (Ryanny Lin)
Date: Thu, 15 Oct 2009 01:40:55 +0800
Subject: [Freeswitch-users] how to make more profiles listen on port 5060 ?
Message-ID: <4bfcac7e0910141040i6edb596ej5877176035084262@mail.gmail.com>

Hi, all,

I read the article "Performance testing and configurations" on the wiki.
Here is a Recommended from wiki:
"libsofia only handles 1 thread per profile, so if that is your bottle
neck use more profiles"
How can I use more profiles and listen on port 5060? Is it possible?
I used sipp to tested SIP REGISTER performance which is about 165 cps.
It is about 260 cps when turned lots of modules.
Does anyone test higher than this?

-- 
Sincerely regards,
Ryanny
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From msc at freeswitch.org  Wed Oct 14 11:05:49 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 14 Oct 2009 11:05:49 -0700
Subject: [Freeswitch-users] how to make more profiles listen on port
	5060 ?
In-Reply-To: <4bfcac7e0910141040i6edb596ej5877176035084262@mail.gmail.com>
References: <4bfcac7e0910141040i6edb596ej5877176035084262@mail.gmail.com>
Message-ID: <87f2f3b90910141105m2a31ca4btef4b06f9253a275a@mail.gmail.com>

On Wed, Oct 14, 2009 at 10:40 AM, Ryanny Lin <ryannyl at gmail.com> wrote:

> Hi, all,
>
> I read the article "Performance testing and configurations" on the wiki.
> Here is a Recommended from wiki:
> "libsofia only handles 1 thread per profile, so if that is your bottle neck use more profiles"
>
> How can I use more profiles and listen on port 5060? Is it possible?
> I used sipp to tested SIP REGISTER performance which is about 165 cps. It is about 260 cps when turned lots of modules.
> Does anyone test higher than this?
>
> You can't have more than one profile listening on a particular port/IP
since by definition a profile is a UA that services exactly one IP/port
combo. As for SIPp, unless you are an expert in load testing then tread very
carefully. Your best bet for testing is to put the machine into production
and see how it perform with real traffic.

-MC
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From msc at freeswitch.org  Wed Oct 14 11:36:35 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 14 Oct 2009 11:36:35 -0700
Subject: [Freeswitch-users] adding new extension
In-Reply-To: <f8af5740910140458w4666ee2fwa93d1ecb70cc015a@mail.gmail.com>
References: <f8af5740910140458w4666ee2fwa93d1ecb70cc015a@mail.gmail.com>
Message-ID: <87f2f3b90910141136s2a233cabg166bb91f8b4e795b@mail.gmail.com>

On Wed, Oct 14, 2009 at 4:58 AM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

> Hi,
>
> can any please let me know, how to add new extension(eg 1000.xml)
> dynamically while running freeswitch?
> while running freeswitch i have created new xml(eg: 1500.xml) file and i
> have changed freeswitch.xml.fsxml still  not working,
>
> Boy, it would be great if someone wrote an article that covers this and
published it in a magazine... oh wait, we have that!
http://bit.ly/EpVrv

I cover this exact scenario. In fact, my example actually uses "1500" as a
new extension to add. Definitely read this article - it will help ease your
transition into the world of FreeSWITCH.
-MC
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From msc at freeswitch.org  Wed Oct 14 11:45:10 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 14 Oct 2009 11:45:10 -0700
Subject: [Freeswitch-users] FreeSWITCH training poll: please vote soon
Message-ID: <87f2f3b90910141145u56a2a14eh5e1a5c61918d3af5@mail.gmail.com>

Everyone,

Please go vote here:
http://www.freeswitch.org/node/208

Be honest - we need real feedback. Don't click the first choice unless you
really have the money. :)

Thanks everyone! We appreciate having such a great community.
-MC
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From msc at freeswitch.org  Wed Oct 14 11:48:11 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 14 Oct 2009 11:48:11 -0700
Subject: [Freeswitch-users] lua script causing FreeSwitch to crash?
In-Reply-To: <005b01ca4cf1$10a30970$31e91c50$@com>
References: <005a01ca4c3c$c183ceb0$448b6c10$@com>
	<87f2f3b90910131312k7dac8980h70ce8be9c07aa7b2@mail.gmail.com>
	<000601ca4ce2$ed1de640$c759b2c0$@com>
	<191c3a030910140902j401a9c6dg7704616d27bac051@mail.gmail.com>
	<005b01ca4cf1$10a30970$31e91c50$@com>
Message-ID: <87f2f3b90910141148p7d2d33a3od7098142853b0964@mail.gmail.com>

On Wed, Oct 14, 2009 at 10:09 AM, Lars Zeb <larclap at yahoo.com> wrote:

>  I did svn up, ./configure, make and make install.
>
>
>
> Do you want me to do make current before proceeding, or just try to make it
> stop with the current build from the above commands?
>
> Lars,

The "make current" method was designed exactly for the scenario you are in:
to upgrade smoothly and cleanly. It help prevents build skew among other
things. Please go do "make current" and then run your test again and report
back.

Thanks!
-MC
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From sjmudd at pobox.com  Wed Oct 14 12:46:47 2009
From: sjmudd at pobox.com (Simon J Mudd)
Date: 14 Oct 2009 21:46:47 +0200
Subject: [Freeswitch-users] FreeSWITCH training poll: please vote soon
In-Reply-To: <87f2f3b90910141145u56a2a14eh5e1a5c61918d3af5@mail.gmail.com>
References: <87f2f3b90910141145u56a2a14eh5e1a5c61918d3af5@mail.gmail.com>
Message-ID: <m3zl7tzrko.fsf@mad06.wl0.org>

msc at freeswitch.org (Michael Collins) writes:

> Please go vote here:
> http://www.freeswitch.org/node/208
> 
> Be honest - we need real feedback. Don't click the first choice unless you
> really have the money. :)

There's a missing option IMO at least for SOHO people like
myself. That's better documentation, such as books on FreeSWITCH,
perhaps showing how to build a small SOHO system, perhaps using
"standard" example components like a few hard SIP phones and a SIP
gateway, and perhaps a local SIP -> PSTN gateway.  Then show how to
configure things like recording conversations, enable "out of hours"
voicemail for "non-special" outside callers.  Something like that.

I'd buy a book that covered stuff like that today. I bought several
books on VoIP, a couple of Asterisk books and have read some related
RFCs but I'm not a VoIP professioal.  So any (good) books on FreeSWITCH
would help me adapt my knowledge on Asterisk to FreeSWITCH, and help
me setup FreeSWITCH the way I want.

Simon


From msc at freeswitch.org  Wed Oct 14 12:56:31 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 14 Oct 2009 12:56:31 -0700
Subject: [Freeswitch-users] FreeSWITCH training poll: please vote soon
In-Reply-To: <m3zl7tzrko.fsf@mad06.wl0.org>
References: <87f2f3b90910141145u56a2a14eh5e1a5c61918d3af5@mail.gmail.com>
	<m3zl7tzrko.fsf@mad06.wl0.org>
Message-ID: <87f2f3b90910141256i565d9062te9235f49ccf8b668@mail.gmail.com>

On Wed, Oct 14, 2009 at 12:46 PM, Simon J Mudd <sjmudd at pobox.com> wrote:

> msc at freeswitch.org (Michael Collins) writes:
>
> > Please go vote here:
> > http://www.freeswitch.org/node/208
> >
> > Be honest - we need real feedback. Don't click the first choice unless
> you
> > really have the money. :)
>
> There's a missing option IMO at least for SOHO people like
> myself. That's better documentation, such as books on FreeSWITCH,
> perhaps showing how to build a small SOHO system, perhaps using
> "standard" example components like a few hard SIP phones and a SIP
> gateway, and perhaps a local SIP -> PSTN gateway.  Then show how to
> configure things like recording conversations, enable "out of hours"
> voicemail for "non-special" outside callers.  Something like that.
>
> I'd buy a book that covered stuff like that today. I bought several
> books on VoIP, a couple of Asterisk books and have read some related
> RFCs but I'm not a VoIP professioal.  So any (good) books on FreeSWITCH
> would help me adapt my knowledge on Asterisk to FreeSWITCH, and help
> me setup FreeSWITCH the way I want.
>

We are aware of this issue and are addressing it separately. Pretty much
everyone is on board with better written documentation. This poll is
specifically asking about on-site training or a boot-camp style training
program, both of which would require a serious financial commitment from the
participants. If you are not in a position for serious FS training then
option #3 would be for you.
-MC
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From anthony.minessale at gmail.com  Wed Oct 14 13:01:30 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 14 Oct 2009 15:01:30 -0500
Subject: [Freeswitch-users] Perl event socket library problem
In-Reply-To: <1452e2980910132250w293596d6jce5515d787b4fa2a@mail.gmail.com>
References: <1452e2980910132250w293596d6jce5515d787b4fa2a@mail.gmail.com>
Message-ID: <191c3a030910141301q21301c8v5f3ff7cd9ac79a0f@mail.gmail.com>

That's what asynchronous means "do not block"

try $con->setEventLock("true"); with the socket in async mode, then your
requests will stack in order.



On Wed, Oct 14, 2009 at 12:50 AM, velusamy velu <velu.technical at gmail.com>wrote:

> Dear All,
>        I am implementing an IVR framework using Perl event socket
> libraries. At first I set socket mode async full. I have faced problem that
> the Perl statements executing before dailplan execution. So, I couldn't
> control my process.
>
>        Next I have tried without async mode, it solved that race condition
> problem. I had another requirement that while executing external application
> I need to play some voice file some time limit. When I played that voice
> file in playback, the playback application blocked. So, I can't break that
> playback application after some time limit.
>
>        I am hanging on this problem for past one week. Please any one help
> me to solve above problems...
>
> Thanks,
> Velusamy.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From sjmudd at pobox.com  Wed Oct 14 15:01:35 2009
From: sjmudd at pobox.com (Simon J Mudd)
Date: Thu, 15 Oct 2009 00:01:35 +0200
Subject: [Freeswitch-users] macros for performing repeated dialplan actions?
Message-ID: <20091014220135.GA12614@mad06.wl0.org>

Hi again,

I'm trying to build up a soho dialplan to replicate an existing Asterisk
setup I want to replace.

I'm editing
/usr/local/freeswitch/conf/dialplan/default/00_mydomain.com.xml to add
some dialplan rules for various destination numbers I'm using. These
dialplans may use one of several gateways I have configured.

I currently have:

<include>
  <extension name="spanish_mobiles.mydomain.com">
    <condition field="destination_number" expression="^(6\d{8})$">
      <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
      <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>
      <action application="bridge" data="sofia/gateway/asterisk/$1"/>
    </condition>
  </extension>

  <extension name="spanish_fijos.mydomain.com">
    <condition field="destination_number" expression="^(9\d{8})$">
      <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
      <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>
      <action application="bridge" data="sofia/gateway/asterisk/$1"/>
    </condition>
  </extension>

  ... more stuff not shown ...

</include>

Which routes out to my existing asterisk installation.

I now want to record all calls. Looking at
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session
it seems I need to add a section like into each entry prior to the
application="bridge".

    <action application="set" data="gateway_name=XXXXX"/>

    <action application="set" data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/>
    <action application="set" data="RECORD_COPYRIGHT=(C) 2009 my data here."/>
    <action application="set" data="RECORD_SOFTWARE=FreeSwitch"/>
    <action application="set" data="RECORD_ARTIST=My name"/>
    <action application="set" data="RECORD_COMMENT=My comment"/>
    <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/>
    <action application="set" data="RECORD_STEREO=true"/>
    <action application="record_session" data="$${base_dir}/recordings/${gateway_name}_${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/>

My question is how do I avoid repeatedly using these same 8-9 lines and
"merge" this into some macro which given the right parameters allows me
to define various extensions using different gateways but all configured
"the same way"?

Thanks for any pointers to how this should be done.

Simon


From msc at freeswitch.org  Wed Oct 14 15:10:51 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 14 Oct 2009 15:10:51 -0700
Subject: [Freeswitch-users] macros for performing repeated dialplan
	actions?
In-Reply-To: <20091014220135.GA12614@mad06.wl0.org>
References: <20091014220135.GA12614@mad06.wl0.org>
Message-ID: <87f2f3b90910141510n45401995y11e3e380653e9790@mail.gmail.com>

On Wed, Oct 14, 2009 at 3:01 PM, Simon J Mudd <sjmudd at pobox.com> wrote:

> Hi again,
>
> I'm trying to build up a soho dialplan to replicate an existing Asterisk
> setup I want to replace.
>
> I'm editing
> /usr/local/freeswitch/conf/dialplan/default/00_mydomain.com.xml to add
> some dialplan rules for various destination numbers I'm using. These
> dialplans may use one of several gateways I have configured.
>
> I currently have:
>
> <include>
>  <extension name="spanish_mobiles.mydomain.com">
>    <condition field="destination_number" expression="^(6\d{8})$">
>      <action application="set"
> data="effective_caller_id_number=${outbound_caller_id_number}"/>
>      <action application="set"
> data="effective_caller_id_name=${outbound_caller_id_name}"/>
>      <action application="bridge" data="sofia/gateway/asterisk/$1"/>
>    </condition>
>  </extension>
>
>  <extension name="spanish_fijos.mydomain.com">
>    <condition field="destination_number" expression="^(9\d{8})$">
>      <action application="set"
> data="effective_caller_id_number=${outbound_caller_id_number}"/>
>      <action application="set"
> data="effective_caller_id_name=${outbound_caller_id_name}"/>
>      <action application="bridge" data="sofia/gateway/asterisk/$1"/>
>    </condition>
>  </extension>
>
>  ... more stuff not shown ...
>
> </include>
>
> Which routes out to my existing asterisk installation.
>
> I now want to record all calls. Looking at
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session
> it seems I need to add a section like into each entry prior to the
> application="bridge".
>
>    <action application="set" data="gateway_name=XXXXX"/>
>
>    <action application="set" data="RECORD_TITLE=Recording
> ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/>
>    <action application="set" data="RECORD_COPYRIGHT=(C) 2009 my data
> here."/>
>    <action application="set" data="RECORD_SOFTWARE=FreeSwitch"/>
>    <action application="set" data="RECORD_ARTIST=My name"/>
>    <action application="set" data="RECORD_COMMENT=My comment"/>
>    <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d
> %H:%M)}"/>
>    <action application="set" data="RECORD_STEREO=true"/>
>    <action application="record_session"
> data="$${base_dir}/recordings/${gateway_name}_${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/>
>
> My question is how do I avoid repeatedly using these same 8-9 lines and
> "merge" this into some macro which given the right parameters allows me
> to define various extensions using different gateways but all configured
> "the same way"?
>
> Thanks for any pointers to how this should be done.
>

Try using the execute_extension app. It's kinda like a gosub in BASIC: it'll
go and execute an extension and then come back and continue executing. Put
all those lines into a single extension and then call them from other
extensions via execute_extension.
-MC
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From larclap at yahoo.com  Wed Oct 14 16:26:11 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Wed, 14 Oct 2009 16:26:11 -0700
Subject: [Freeswitch-users] Outbound sip address call fails
Message-ID: <00d701ca4d25$b7d615f0$278241d0$@com>

I am trying to make a test call to the FreeSwitch conference address,
sip:888 at conference.freeswitch.org on a Bria softphone. However, it fails
with a 404 Not found message.

 

I have struggled with this for a while. Is there something special I must do
on the Bria to make this happen correctly?

 

Thanks, Lars

 

http://pastebin.freeswitch.org/10715

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From orien at tx.rr.com  Wed Oct 14 16:31:43 2009
From: orien at tx.rr.com (Orien Love)
Date: Wed, 14 Oct 2009 18:31:43 -0500
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 129
In-Reply-To: <mailman.1259.1255550525.2495.freeswitch-users@lists.freeswitch.org>
References: <mailman.1259.1255550525.2495.freeswitch-users@lists.freeswitch.org>
Message-ID: <4AD65F5F.9090306@tx.rr.com>

Hello, a few weeks ago I asked about a setup, and about using the 
freeswitch package on pfSense. Since then I have received most of my 
hardware and started setting up freeswitch.

First my setup
WWW --> 1st pfsense box <-- private (192.168.2.xxx) --> pfsense w 
freeswtich <-- Test (192.168.3.xxx) --> phones

pfsense w freeswtich Port Addresses
WAN 192.168.2.210
LAN 192.168.3.1

this is setup to simulate a pfsense box hooked up to an external network.


here are the things I have done.

1st attempt
#1 set my WAN IP port to fixed in pfsense (192.168.2.210)
#2 set my soft phone to register to 192.168.2.210 (The phones registered 
and showed up in the status section)
#3 Tried to make a call to extension 1004, the soft phone tells me that 
it is "Temporally Unavailable 480).

2nd attempt
#1 set my WAN IP port to fixed in pfsense (192.168.2.210)
#2 deleted the lan.xml profile
#3 rebooted freeswitch.
#4 set my soft phone to register to 192.168.2.210 (The phones registered 
and showed up in the status section)
#5 Tried to make a call to extension 1004, the soft phone tells me that 
it is "Temporally Unavailable 480).

3rd attempt
#1 set my WAN IP port to fixed in pfsense (192.168.2.210)
#2 edited the internal.xml file as described in the multi-home tutorial 
(192.168.3.1 is my phone lan address)
    <param name="rtp-ip" value="192.168.3.1"/>
    <param name="sip-ip" value="192.168.3.1"/>
    <param name="ext-rtp-ip" value="192.168.3.1"/>
    <param name="ext-sip-ip" value="192.168.3.1"/>
#3 rebooted freeswitch.
#4 set my soft phone to register to 192.168.2.210 (The phone would not 
register)
#5 set my soft phone to register to 192.168.3.1 (The phone registered 
and showed up in the status section)
#6 Tried to make a call to extension 1004, the soft phone tells me that 
it is "Temporally Unavailable 480).
#7 edited the file /usr/local/freeswitch/conf/directory/default.xml
  <domain name="192.168.3.1">
#8 rebooted freeswitch.
#9 set my soft phone to register to 192.168.2.210 (The phone would not 
register)
#10 set my soft phone to register to 192.168.3.1 (The phone would not 
register got a forbidden error)
#11 did some reading and found a place in the vars tab to change the domain
#12 edited the file in the vars tab
  <X-PRE-PROCESS cmd="set" data="domain=192.168.3.1"/>
#13 rebooted freeswitch.
#14 set my soft phone to register to 192.168.2.210 (The phone would not 
register)
#15 set my soft phone to register to 192.168.3.1 (The phone registered 
and showed up in the status section)
#16 Tried to make a call to extension 1004, the soft phone tells me that 
it is "Temporally Unavailable 480).

here is part of the log file:
2009-10-14 22:18:43.560147 [DEBUG] sofia.c:910 Write lock lan
2009-10-14 22:18:43.563938 [NOTICE] sofia.c:917 Waiting for worker thread
2009-10-14 22:18:43.563938 [DEBUG] sofia.c:975 Write unlock lan
2009-10-14 22:26:00.23234 [DEBUG] sofia.c:4549 IP 192.168.3.199 Rejected 
by acl "domains". Falling back to Digest auth.
2009-10-14 22:26:00.55225 [DEBUG] sofia.c:4549 IP 192.168.3.199 Rejected 
by acl "domains". Falling back to Digest auth.
2009-10-14 22:26:00.55225 [NOTICE] switch_channel.c:602 New Channel 
sofia/internal/1001 at 192.168.2.210 [8aa7cd8c-10b9-de11-a0e9-00a0cc5a11e3]
2009-10-14 22:26:00.55225 [DEBUG] sofia.c:5196 Setting NAT mode based on 
rfc1918.auto
2009-10-14 22:26:00.55225 [DEBUG] sofia.c:3210 Channel 
sofia/internal/1001 at 192.168.2.210 entering state [received][100]
2009-10-14 22:26:00.55225 [DEBUG] sofia.c:3217 Remote SDP:
v=0
o=- 78902911 78902911 IN IP4 192.168.189.1
s=http://www.portsip.com
c=IN IP4 192.168.189.1
t=0 0
m=audio 21328 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
m=video 40216 RTP/AVP 34 125
a=rtpmap:34 H263/90000
a=fmtp:34 CIF=2
a=rtpmap:125 H264/90000
a=fmtp:125 profile-level-id=42e015

2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3081 Audio Codec Compare 
[PCMU:0:8000:20]/[G7221:115:32000:20]
2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3081 Audio Codec Compare 
[PCMU:0:8000:20]/[G7221:107:16000:20]
2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3081 Audio Codec Compare 
[PCMU:0:8000:20]/[G722:9:8000:20]
2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3081 Audio Codec Compare 
[PCMU:0:8000:20]/[PCMU:0:8000:20]
2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:2039 Set Codec 
sofia/internal/1001 at 192.168.2.210 PCMU/8000 20 ms 160 samples
2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3041 Set 2833 dtmf 
payload to 101
2009-10-14 22:26:00.55225 [DEBUG] sofia.c:3376 
(sofia/internal/1001 at 192.168.2.210) State Change CS_NEW -> CS_INIT
2009-10-14 22:26:00.55225 [DEBUG] switch_core_session.c:933 Send signal 
sofia/internal/1001 at 192.168.2.210 [BREAK]
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397 
(sofia/internal/1001 at 192.168.2.210) Running State Change CS_INIT
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:480 
(sofia/internal/1001 at 192.168.2.210) State INIT
2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:83 
sofia/internal/1001 at 192.168.2.210 SOFIA INIT
2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:111 
(sofia/internal/1001 at 192.168.2.210) State Change CS_INIT -> CS_ROUTING
2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:933 Send signal 
sofia/internal/1001 at 192.168.2.210 [BREAK]
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:480 
(sofia/internal/1001 at 192.168.2.210) State INIT going to sleep
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397 
(sofia/internal/1001 at 192.168.2.210) Running State Change CS_ROUTING
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:483 
(sofia/internal/1001 at 192.168.2.210) State ROUTING
2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:130 
sofia/internal/1001 at 192.168.2.210 SOFIA ROUTING
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:78 
sofia/internal/1001 at 192.168.2.210 Standard ROUTING
2009-10-14 22:26:00.63551 [INFO] mod_dialplan_xml.c:252 Processing 
1001->1004 in context public
Dialplan: sofia/internal/1001 at 192.168.2.210 parsing [public->unloop] 
continue=false
Dialplan: sofia/internal/1001 at 192.168.2.210 Regex (PASS) [unloop] 
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1001 at 192.168.2.210 Regex (FAIL) [unloop] 
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/1001 at 192.168.2.210 parsing 
[public->outside_call] continue=true
Dialplan: sofia/internal/1001 at 192.168.2.210 Absolute Condition 
[outside_call]
Dialplan: sofia/internal/1001 at 192.168.2.210 Action set(outside_call=true)
Dialplan: sofia/internal/1001 at 192.168.2.210 parsing [public->call_debug] 
continue=true
Dialplan: sofia/internal/1001 at 192.168.2.210 Regex (FAIL) [call_debug] 
${call_debug}(false) =~ /^true$/ break=never
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:114 
(sofia/internal/1001 at 192.168.2.210) State Change CS_ROUTING -> CS_EXECUTE
2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:933 Send signal 
sofia/internal/1001 at 192.168.2.210 [BREAK]
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:483 
(sofia/internal/1001 at 192.168.2.210) State ROUTING going to sleep
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397 
(sofia/internal/1001 at 192.168.2.210) Running State Change CS_EXECUTE
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:490 
(sofia/internal/1001 at 192.168.2.210) State EXECUTE
2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:173 
sofia/internal/1001 at 192.168.2.210 SOFIA EXECUTE
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:151 
sofia/internal/1001 at 192.168.2.210 Standard EXECUTE
EXECUTE sofia/internal/1001 at 192.168.2.210 set(outside_call=true)
2009-10-14 22:26:00.63551 [DEBUG] mod_dptools.c:748 
sofia/internal/1001 at 192.168.2.210 SET [outside_call]=[true]
2009-10-14 22:26:00.63551 [NOTICE] switch_core_state_machine.c:179 
Hangup sofia/internal/1001 at 192.168.2.210 [CS_EXECUTE] [NORMAL_CLEARING]
2009-10-14 22:26:00.63551 [DEBUG] switch_channel.c:1683 Send signal 
sofia/internal/1001 at 192.168.2.210 [KILL]
2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:933 Send signal 
sofia/internal/1001 at 192.168.2.210 [BREAK]
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:490 
(sofia/internal/1001 at 192.168.2.210) State EXECUTE going to sleep
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397 
(sofia/internal/1001 at 192.168.2.210) Running State Change CS_HANGUP
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:433 
(sofia/internal/1001 at 192.168.2.210) State HANGUP
2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:338 Channel 
sofia/internal/1001 at 192.168.2.210 hanging up, cause: NORMAL_CLEARING
2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:414 Responding to INVITE 
with: 480
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:46 
sofia/internal/1001 at 192.168.2.210 Standard HANGUP, cause: NORMAL_CLEARING
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:433 
(sofia/internal/1001 at 192.168.2.210) State HANGUP going to sleep
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:475 
(sofia/internal/1001 at 192.168.2.210) State Change CS_HANGUP -> CS_REPORTING
2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:933 Send signal 
sofia/internal/1001 at 192.168.2.210 [BREAK]
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397 
(sofia/internal/1001 at 192.168.2.210) Running State Change CS_REPORTING
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:607 
(sofia/internal/1001 at 192.168.2.210) State REPORTING
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:53 
sofia/internal/1001 at 192.168.2.210 Standard REPORTING, cause: NORMAL_CLEARING
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:607 
(sofia/internal/1001 at 192.168.2.210) State REPORTING going to sleep
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:410 
(sofia/internal/1001 at 192.168.2.210) State Change CS_REPORTING -> CS_DESTROY
2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:1067 Session 2 
(sofia/internal/1001 at 192.168.2.210) Locked, Waiting on external entities
2009-10-14 22:26:00.63551 [NOTICE] switch_core_session.c:1085 Session 2 
(sofia/internal/1001 at 192.168.2.210) Ended
2009-10-14 22:26:00.63551 [NOTICE] switch_core_session.c:1087 Close 
Channel sofia/internal/1001 at 192.168.2.210 [CS_DESTROY]
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:559 
(sofia/internal/1001 at 192.168.2.210) State DESTROY
2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:255 
sofia/internal/1001 at 192.168.2.210 SOFIA DESTROY
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:60 
sofia/internal/1001 at 192.168.2.210 Standard DESTROY
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:559 
(sofia/internal/1001 at 192.168.2.210) State DESTROY going to sleep
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:559 
(sofia/internal/1001 at 192.168.2.210) State DESTROY going to sleep

I am at a loss, any help would be welcome.

Orien Love




From Mailings at kh-dev.de  Wed Oct 14 16:48:09 2009
From: Mailings at kh-dev.de (Klaus Hochlehnert)
Date: Thu, 15 Oct 2009 01:48:09 +0200
Subject: [Freeswitch-users] Strange problems after upgrade...
Message-ID: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330252@srv02.khdev.corp>

Hi,

I just upgraded FS from 15094 to 15161 and after the upgrade I have the following problem:
I use t38modem together with a SIP/ISDN gateway.

With the old release everything fax related works fine. I call the fax -> I hear a fax tone.
With the new release I call the fax -> I don't hear a fax tone. The modem answers the call but there's no media.

I use proxy_media=true, bypass_media=false and inbound-late-negotiation=true.

Is there anything else I need to do with the new release?

Thanks, Klaus
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From Mailings at kh-dev.de  Wed Oct 14 16:56:32 2009
From: Mailings at kh-dev.de (Klaus Hochlehnert)
Date: Thu, 15 Oct 2009 01:56:32 +0200
Subject: [Freeswitch-users] Media got stuck after attended transfer...
Message-ID: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330253@srv02.khdev.corp>

Hi,

sometimes I have the problem that after doing an attended transfer the media got stuck in FS.
Meaning the call goes through, but I don't hear anything and the caller still hears music.

Now I found out that setting the sofia parameter  media-option=resume-media-on-hold  helps here.
But after setting this parameter I always get the error "Codec PROXY PASS-THROUGH encoder error!" when using t38modem with proxy_media=true.

So I stuck here. I can either use fax or do attended transfers.

Does anyone have a solution for this?

Thanks, Klaus
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From msc at freeswitch.org  Wed Oct 14 17:03:18 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 14 Oct 2009 17:03:18 -0700
Subject: [Freeswitch-users] Strange problems after upgrade...
In-Reply-To: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330252@srv02.khdev.corp>
References: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330252@srv02.khdev.corp>
Message-ID: <87f2f3b90910141703hc8311bfg8c1d1280708d70f0@mail.gmail.com>

On Wed, Oct 14, 2009 at 4:48 PM, Klaus Hochlehnert <Mailings at kh-dev.de>wrote:

>  Hi,
>
>
>
> I just upgraded FS from 15094 to 15161 and after the upgrade I have the
> following problem:
>
> I use t38modem together with a SIP/ISDN gateway.
>
>
>
> With the old release everything fax related works fine. I call the fax -> I
> hear a fax tone.
>
> With the new release I call the fax -> I don?t hear a fax tone. The modem
> answers the call but there?s no media.
>
>
>
> I use proxy_media=true, bypass_media=false and
> inbound-late-negotiation=true.
>
>
>
> Is there anything else I need to do with the new release?
>
> How exactly did you update?
-MC
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From Mailings at kh-dev.de  Wed Oct 14 17:14:10 2009
From: Mailings at kh-dev.de (Klaus Hochlehnert)
Date: Thu, 15 Oct 2009 02:14:10 +0200
Subject: [Freeswitch-users] Strange problems after upgrade...
In-Reply-To: <87f2f3b90910141703hc8311bfg8c1d1280708d70f0@mail.gmail.com>
References: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330252@srv02.khdev.corp>
	<87f2f3b90910141703hc8311bfg8c1d1280708d70f0@mail.gmail.com>
Message-ID: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330254@srv02.khdev.corp>

Basically I did an normal make + make install

The conf and the scripts dir were untouched after that (which I wanted because I didn't want to reconfigure everything).

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Thursday, October 15, 2009 2:03 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Strange problems after upgrade...


On Wed, Oct 14, 2009 at 4:48 PM, Klaus Hochlehnert <Mailings at kh-dev.de<mailto:Mailings at kh-dev.de>> wrote:
Hi,

I just upgraded FS from 15094 to 15161 and after the upgrade I have the following problem:
I use t38modem together with a SIP/ISDN gateway.

With the old release everything fax related works fine. I call the fax -> I hear a fax tone.
With the new release I call the fax -> I don't hear a fax tone. The modem answers the call but there's no media.

I use proxy_media=true, bypass_media=false and inbound-late-negotiation=true.

Is there anything else I need to do with the new release?
How exactly did you update?
-MC

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From mgende at gendesign.com  Wed Oct 14 19:57:57 2009
From: mgende at gendesign.com (Michael Gende)
Date: Wed, 14 Oct 2009 21:57:57 -0500
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 129
In-Reply-To: <4AD65F5F.9090306@tx.rr.com>
References: <mailman.1259.1255550525.2495.freeswitch-users@lists.freeswitch.org>
	<4AD65F5F.9090306@tx.rr.com>
Message-ID: <a87d43fd0910141957t13523f73p580847d7e5875288@mail.gmail.com>

Hey Orien,

All of these guys are more experienced than I, but I'll try to throw you a
line.

I'll have to check a thing or two out at work, but here's an observation or
two (I'll post again tomorrow morning after I get in).

The mult-homed example is set to work in a very specifc way: A dual-homed
box where the WAN is used for the WWW connection, and then using the LAN
side for phones.

To work like the example, you need to have your phones on the LAN network.
Then, configure your internal.xml file on FS to use the LAN port, not the
WAN. Next, set your phones to register to that LAN address.

One thing I prefer to do on pfsense, is at the FS box console (or ssh to the
box), kill the "pfsense FS" running the background, and run FS in the
foreground. Then you can "reloadxml" instead of rebooting when XML changes
are made (if I understand your post).

As I said, let me check a thing or two out, and I'll ask to see some
configuration files. I'm sure we can get you going.

BTW: What kind of phones are you using?

Regards,

Mike G.


On Wed, Oct 14, 2009 at 6:31 PM, Orien Love <orien at tx.rr.com> wrote:

> Hello, a few weeks ago I asked about a setup, and about using the
> freeswitch package on pfSense. Since then I have received most of my
> hardware and started setting up freeswitch.
>
> First my setup
> WWW --> 1st pfsense box <-- private (192.168.2.xxx) --> pfsense w
> freeswtich <-- Test (192.168.3.xxx) --> phones
>
> pfsense w freeswtich Port Addresses
> WAN 192.168.2.210
> LAN 192.168.3.1
>
> this is setup to simulate a pfsense box hooked up to an external network.
>
>
> here are the things I have done.
>
> 1st attempt
> #1 set my WAN IP port to fixed in pfsense (192.168.2.210)
> #2 set my soft phone to register to 192.168.2.210 (The phones registered
> and showed up in the status section)
> #3 Tried to make a call to extension 1004, the soft phone tells me that
> it is "Temporally Unavailable 480).
>
> 2nd attempt
> #1 set my WAN IP port to fixed in pfsense (192.168.2.210)
> #2 deleted the lan.xml profile
> #3 rebooted freeswitch.
> #4 set my soft phone to register to 192.168.2.210 (The phones registered
> and showed up in the status section)
> #5 Tried to make a call to extension 1004, the soft phone tells me that
> it is "Temporally Unavailable 480).
>
> 3rd attempt
> #1 set my WAN IP port to fixed in pfsense (192.168.2.210)
> #2 edited the internal.xml file as described in the multi-home tutorial
> (192.168.3.1 is my phone lan address)
>    <param name="rtp-ip" value="192.168.3.1"/>
>    <param name="sip-ip" value="192.168.3.1"/>
>    <param name="ext-rtp-ip" value="192.168.3.1"/>
>    <param name="ext-sip-ip" value="192.168.3.1"/>
> #3 rebooted freeswitch.
> #4 set my soft phone to register to 192.168.2.210 (The phone would not
> register)
> #5 set my soft phone to register to 192.168.3.1 (The phone registered
> and showed up in the status section)
> #6 Tried to make a call to extension 1004, the soft phone tells me that
> it is "Temporally Unavailable 480).
> #7 edited the file /usr/local/freeswitch/conf/directory/default.xml
>  <domain name="192.168.3.1">
> #8 rebooted freeswitch.
> #9 set my soft phone to register to 192.168.2.210 (The phone would not
> register)
> #10 set my soft phone to register to 192.168.3.1 (The phone would not
> register got a forbidden error)
> #11 did some reading and found a place in the vars tab to change the domain
> #12 edited the file in the vars tab
>  <X-PRE-PROCESS cmd="set" data="domain=192.168.3.1"/>
> #13 rebooted freeswitch.
> #14 set my soft phone to register to 192.168.2.210 (The phone would not
> register)
> #15 set my soft phone to register to 192.168.3.1 (The phone registered
> and showed up in the status section)
> #16 Tried to make a call to extension 1004, the soft phone tells me that
> it is "Temporally Unavailable 480).
>
> here is part of the log file:
> 2009-10-14 22:18:43.560147 [DEBUG] sofia.c:910 Write lock lan
> 2009-10-14 22:18:43.563938 [NOTICE] sofia.c:917 Waiting for worker thread
> 2009-10-14 22:18:43.563938 [DEBUG] sofia.c:975 Write unlock lan
> 2009-10-14 22:26:00.23234 [DEBUG] sofia.c:4549 IP 192.168.3.199 Rejected
> by acl "domains". Falling back to Digest auth.
> 2009-10-14 22:26:00.55225 [DEBUG] sofia.c:4549 IP 192.168.3.199 Rejected
> by acl "domains". Falling back to Digest auth.
> 2009-10-14 22:26:00.55225 [NOTICE] switch_channel.c:602 New Channel
> sofia/internal/1001 at 192.168.2.210 [8aa7cd8c-10b9-de11-a0e9-00a0cc5a11e3]
> 2009-10-14 22:26:00.55225 [DEBUG] sofia.c:5196 Setting NAT mode based on
> rfc1918.auto
> 2009-10-14 22:26:00.55225 [DEBUG] sofia.c:3210 Channel
> sofia/internal/1001 at 192.168.2.210 entering state [received][100]
> 2009-10-14 22:26:00.55225 [DEBUG] sofia.c:3217 Remote SDP:
> v=0
> o=- 78902911 78902911 IN IP4 192.168.189.1
> s=http://www.portsip.com
> c=IN IP4 192.168.189.1
> t=0 0
> m=audio 21328 RTP/AVP 0 8 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
> m=video 40216 RTP/AVP 34 125
> a=rtpmap:34 H263/90000
> a=fmtp:34 CIF=2
> a=rtpmap:125 H264/90000
> a=fmtp:125 profile-level-id=42e015
>
> 2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3081 Audio Codec Compare
> [PCMU:0:8000:20]/[G7221:115:32000:20]
> 2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3081 Audio Codec Compare
> [PCMU:0:8000:20]/[G7221:107:16000:20]
> 2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3081 Audio Codec Compare
> [PCMU:0:8000:20]/[G722:9:8000:20]
> 2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3081 Audio Codec Compare
> [PCMU:0:8000:20]/[PCMU:0:8000:20]
> 2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:2039 Set Codec
> sofia/internal/1001 at 192.168.2.210 PCMU/8000 20 ms 160 samples
> 2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3041 Set 2833 dtmf
> payload to 101
> 2009-10-14 22:26:00.55225 [DEBUG] sofia.c:3376
> (sofia/internal/1001 at 192.168.2.210) State Change CS_NEW -> CS_INIT
> 2009-10-14 22:26:00.55225 [DEBUG] switch_core_session.c:933 Send signal
> sofia/internal/1001 at 192.168.2.210 [BREAK]
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397
> (sofia/internal/1001 at 192.168.2.210) Running State Change CS_INIT
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:480
> (sofia/internal/1001 at 192.168.2.210) State INIT
> 2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:83
> sofia/internal/1001 at 192.168.2.210 SOFIA INIT
> 2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:111
> (sofia/internal/1001 at 192.168.2.210) State Change CS_INIT -> CS_ROUTING
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:933 Send signal
> sofia/internal/1001 at 192.168.2.210 [BREAK]
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:480
> (sofia/internal/1001 at 192.168.2.210) State INIT going to sleep
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397
> (sofia/internal/1001 at 192.168.2.210) Running State Change CS_ROUTING
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:483
> (sofia/internal/1001 at 192.168.2.210) State ROUTING
> 2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:130
> sofia/internal/1001 at 192.168.2.210 SOFIA ROUTING
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:78
> sofia/internal/1001 at 192.168.2.210 Standard ROUTING
> 2009-10-14 22:26:00.63551 [INFO] mod_dialplan_xml.c:252 Processing
> 1001->1004 in context public
> Dialplan: sofia/internal/1001 at 192.168.2.210 parsing [public->unloop]
> continue=false
> Dialplan: sofia/internal/1001 at 192.168.2.210 Regex (PASS) [unloop]
> ${unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/internal/1001 at 192.168.2.210 Regex (FAIL) [unloop]
> ${sip_looped_call}() =~ /^true$/ break=on-false
> Dialplan: sofia/internal/1001 at 192.168.2.210 parsing
> [public->outside_call] continue=true
> Dialplan: sofia/internal/1001 at 192.168.2.210 Absolute Condition
> [outside_call]
> Dialplan: sofia/internal/1001 at 192.168.2.210 Action set(outside_call=true)
> Dialplan: sofia/internal/1001 at 192.168.2.210 parsing [public->call_debug]
> continue=true
> Dialplan: sofia/internal/1001 at 192.168.2.210 Regex (FAIL) [call_debug]
> ${call_debug}(false) =~ /^true$/ break=never
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:114
> (sofia/internal/1001 at 192.168.2.210) State Change CS_ROUTING -> CS_EXECUTE
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:933 Send signal
> sofia/internal/1001 at 192.168.2.210 [BREAK]
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:483
> (sofia/internal/1001 at 192.168.2.210) State ROUTING going to sleep
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397
> (sofia/internal/1001 at 192.168.2.210) Running State Change CS_EXECUTE
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:490
> (sofia/internal/1001 at 192.168.2.210) State EXECUTE
> 2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:173
> sofia/internal/1001 at 192.168.2.210 SOFIA EXECUTE
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:151
> sofia/internal/1001 at 192.168.2.210 Standard EXECUTE
> EXECUTE sofia/internal/1001 at 192.168.2.210 set(outside_call=true)
> 2009-10-14 22:26:00.63551 [DEBUG] mod_dptools.c:748
> sofia/internal/1001 at 192.168.2.210 SET [outside_call]=[true]
> 2009-10-14 22:26:00.63551 [NOTICE] switch_core_state_machine.c:179
> Hangup sofia/internal/1001 at 192.168.2.210 [CS_EXECUTE] [NORMAL_CLEARING]
> 2009-10-14 22:26:00.63551 [DEBUG] switch_channel.c:1683 Send signal
> sofia/internal/1001 at 192.168.2.210 [KILL]
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:933 Send signal
> sofia/internal/1001 at 192.168.2.210 [BREAK]
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:490
> (sofia/internal/1001 at 192.168.2.210) State EXECUTE going to sleep
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397
> (sofia/internal/1001 at 192.168.2.210) Running State Change CS_HANGUP
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:433
> (sofia/internal/1001 at 192.168.2.210) State HANGUP
> 2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:338 Channel
> sofia/internal/1001 at 192.168.2.210 hanging up, cause: NORMAL_CLEARING
> 2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:414 Responding to INVITE
> with: 480
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/1001 at 192.168.2.210 Standard HANGUP, cause: NORMAL_CLEARING
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:433
> (sofia/internal/1001 at 192.168.2.210) State HANGUP going to sleep
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:475
> (sofia/internal/1001 at 192.168.2.210) State Change CS_HANGUP -> CS_REPORTING
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:933 Send signal
> sofia/internal/1001 at 192.168.2.210 [BREAK]
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397
> (sofia/internal/1001 at 192.168.2.210) Running State Change CS_REPORTING
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:607
> (sofia/internal/1001 at 192.168.2.210) State REPORTING
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/1001 at 192.168.2.210 Standard REPORTING, cause:
> NORMAL_CLEARING
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:607
> (sofia/internal/1001 at 192.168.2.210) State REPORTING going to sleep
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:410
> (sofia/internal/1001 at 192.168.2.210) State Change CS_REPORTING ->
> CS_DESTROY
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:1067 Session 2
> (sofia/internal/1001 at 192.168.2.210) Locked, Waiting on external entities
> 2009-10-14 22:26:00.63551 [NOTICE] switch_core_session.c:1085 Session 2
> (sofia/internal/1001 at 192.168.2.210) Ended
> 2009-10-14 22:26:00.63551 [NOTICE] switch_core_session.c:1087 Close
> Channel sofia/internal/1001 at 192.168.2.210 [CS_DESTROY]
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:559
> (sofia/internal/1001 at 192.168.2.210) State DESTROY
> 2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:255
> sofia/internal/1001 at 192.168.2.210 SOFIA DESTROY
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:60
> sofia/internal/1001 at 192.168.2.210 Standard DESTROY
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:559
> (sofia/internal/1001 at 192.168.2.210) State DESTROY going to sleep
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:559
> (sofia/internal/1001 at 192.168.2.210) State DESTROY going to sleep
>
> I am at a loss, any help would be welcome.
>
> Orien Love
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From orien at tx.rr.com  Wed Oct 14 20:30:03 2009
From: orien at tx.rr.com (Orien Love)
Date: Wed, 14 Oct 2009 22:30:03 -0500
Subject: [Freeswitch-users] FreeSWITCH-users Digest,
	Mike G. answer to Orien L
In-Reply-To: <mailman.1269.1255575509.2495.freeswitch-users@lists.freeswitch.org>
References: <mailman.1269.1255575509.2495.freeswitch-users@lists.freeswitch.org>
Message-ID: <4AD6973B.70106@tx.rr.com>

Mike,
     Thank you for your quick response, this has been driving me nuts.

in my third attempt I tried to follow the mulit home example. I did set 
the phones to register to the local lan IP. actually had them 
registering, but it did help it in any way still the same problem.
After my post I reinstalled the package and set the firewall to allow 
phones on the wan to connect, as per 
http://doc.pfsense.org/index.php/FreeSWITCH#Firewall
put my computer with the soft phone on the wan network (192,168.2.xxx) 
had it registering and it had the same problem. so far all the setups I 
have tried are not working, the phones register, but cannot 
dial/connect  another  phone, also does not go to voice mail, maybe it 
is in my dial plan. which is default.

Shoot, I am not picky, I just want to get it to work. so any suggestions 
on how to accomplish this are greatly appreciated.

My phones are currently a couple of polycom 601's, and a Pangolin Softphone.

Thanks
   Orien L.

>
> ------------------------------------------------------------------------
>
> Subject:
> Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 129
> From:
> Michael Gende <mgende at gendesign.com>
> Date:
> Wed, 14 Oct 2009 21:57:57 -0500
> To:
> freeswitch-users at lists.freeswitch.org
>
> To:
> freeswitch-users at lists.freeswitch.org
>
>
> Hey Orien,
>
> All of these guys are more experienced than I, but I'll try to throw 
> you a line.
>
> I'll have to check a thing or two out at work, but here's an 
> observation or two (I'll post again tomorrow morning after I get in).
>
> The mult-homed example is set to work in a very specifc way: A 
> dual-homed box where the WAN is used for the WWW connection, and then 
> using the LAN side for phones.
>
> To work like the example, you need to have your phones on the LAN 
> network. Then, configure your internal.xml file on FS to use the LAN 
> port, not the WAN. Next, set your phones to register to that LAN 
> address. 
>
> One thing I prefer to do on pfsense, is at the FS box console (or ssh 
> to the box), kill the "pfsense FS" running the background, and run FS 
> in the foreground. Then you can "reloadxml" instead of rebooting when 
> XML changes are made (if I understand your post).
>
> As I said, let me check a thing or two out, and I'll ask to see some 
> configuration files. I'm sure we can get you going.
>
> BTW: What kind of phones are you using?
>
> Regards,
>
> Mike G.
>
>
> On Wed, Oct 14, 2009 at 6:31 PM, Orien Love <orien at tx.rr.com 
> <mailto:orien at tx.rr.com>> wrote:
>
>     Hello, a few weeks ago I asked about a setup, and about using the
>     freeswitch package on pfSense. Since then I have received most of my
>     hardware and started setting up freeswitch.
>
>     First my setup
>     WWW --> 1st pfsense box <-- private (192.168.2.xxx) --> pfsense w
>     freeswtich <-- Test (192.168.3.xxx) --> phones
>
>     pfsense w freeswtich Port Addresses
>     WAN 192.168.2.210
>     LAN 192.168.3.1
>
>     this is setup to simulate a pfsense box hooked up to an external
>     network.
>
>
>     here are the things I have done.
>
>     1st attempt
>     #1 set my WAN IP port to fixed in pfsense (192.168.2.210)
>     #2 set my soft phone to register to 192.168.2.210 (The phones
>     registered
>     and showed up in the status section)
>     #3 Tried to make a call to extension 1004, the soft phone tells me
>     that
>     it is "Temporally Unavailable 480).
>
>     2nd attempt
>     #1 set my WAN IP port to fixed in pfsense (192.168.2.210)
>     #2 deleted the lan.xml profile
>     #3 rebooted freeswitch.
>     #4 set my soft phone to register to 192.168.2.210 (The phones
>     registered
>     and showed up in the status section)
>     #5 Tried to make a call to extension 1004, the soft phone tells me
>     that
>     it is "Temporally Unavailable 480).
>
>     3rd attempt
>     #1 set my WAN IP port to fixed in pfsense (192.168.2.210)
>     #2 edited the internal.xml file as described in the multi-home
>     tutorial
>     (192.168.3.1 is my phone lan address)
>        <param name="rtp-ip" value="192.168.3.1"/>
>        <param name="sip-ip" value="192.168.3.1"/>
>        <param name="ext-rtp-ip" value="192.168.3.1"/>
>        <param name="ext-sip-ip" value="192.168.3.1"/>
>     #3 rebooted freeswitch.
>     #4 set my soft phone to register to 192.168.2.210 (The phone would not
>     register)
>     #5 set my soft phone to register to 192.168.3.1 (The phone registered
>     and showed up in the status section)
>     #6 Tried to make a call to extension 1004, the soft phone tells me
>     that
>     it is "Temporally Unavailable 480).
>     #7 edited the file /usr/local/freeswitch/conf/directory/default.xml
>      <domain name="192.168.3.1">
>     #8 rebooted freeswitch.
>     #9 set my soft phone to register to 192.168.2.210 (The phone would not
>     register)
>     #10 set my soft phone to register to 192.168.3.1 (The phone would not
>     register got a forbidden error)
>     #11 did some reading and found a place in the vars tab to change
>     the domain
>     #12 edited the file in the vars tab
>      <X-PRE-PROCESS cmd="set" data="domain=192.168.3.1"/>
>     #13 rebooted freeswitch.
>     #14 set my soft phone to register to 192.168.2.210 (The phone
>     would not
>     register)
>     #15 set my soft phone to register to 192.168.3.1 (The phone registered
>     and showed up in the status section)
>     #16 Tried to make a call to extension 1004, the soft phone tells
>     me that
>     it is "Temporally Unavailable 480).
>
>     here is part of the log file:
>     2009-10-14 22:18:43.560147 [DEBUG] sofia.c:910 Write lock lan
>     2009-10-14 22:18:43.563938 [NOTICE] sofia.c:917 Waiting for worker
>     thread
>     2009-10-14 22:18:43.563938 [DEBUG] sofia.c:975 Write unlock lan
>     2009-10-14 22:26:00.23234 [DEBUG] sofia.c:4549 IP 192.168.3.199
>     Rejected
>     by acl "domains". Falling back to Digest auth.
>     2009-10-14 22:26:00.55225 [DEBUG] sofia.c:4549 IP 192.168.3.199
>     Rejected
>     by acl "domains". Falling back to Digest auth.
>     2009-10-14 22:26:00.55225 [NOTICE] switch_channel.c:602 New Channel
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>
>     [8aa7cd8c-10b9-de11-a0e9-00a0cc5a11e3]
>     2009-10-14 22:26:00.55225 [DEBUG] sofia.c:5196 Setting NAT mode
>     based on
>     rfc1918.auto
>     2009-10-14 22:26:00.55225 [DEBUG] sofia.c:3210 Channel
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>
>     entering state [received][100]
>     2009-10-14 22:26:00.55225 [DEBUG] sofia.c:3217 Remote SDP:
>     v=0
>     o=- 78902911 78902911 IN IP4 192.168.189.1
>     s=http://www.portsip.com
>     c=IN IP4 192.168.189.1
>     t=0 0
>     m=audio 21328 RTP/AVP 0 8 18 101
>     a=rtpmap:0 PCMU/8000
>     a=rtpmap:8 PCMA/8000
>     a=rtpmap:18 G729/8000
>     a=fmtp:18 annexb=no
>     a=rtpmap:101 telephone-event/8000
>     a=fmtp:101 0-15
>     a=ptime:20
>     m=video 40216 RTP/AVP 34 125
>     a=rtpmap:34 H263/90000
>     a=fmtp:34 CIF=2
>     a=rtpmap:125 H264/90000
>     a=fmtp:125 profile-level-id=42e015
>
>     2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3081 Audio Codec
>     Compare
>     [PCMU:0:8000:20]/[G7221:115:32000:20]
>     2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3081 Audio Codec
>     Compare
>     [PCMU:0:8000:20]/[G7221:107:16000:20]
>     2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3081 Audio Codec
>     Compare
>     [PCMU:0:8000:20]/[G722:9:8000:20]
>     2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3081 Audio Codec
>     Compare
>     [PCMU:0:8000:20]/[PCMU:0:8000:20]
>     2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:2039 Set Codec
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>
>     PCMU/8000 20 ms 160 samples
>     2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3041 Set 2833 dtmf
>     payload to 101
>     2009-10-14 22:26:00.55225 [DEBUG] sofia.c:3376
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State Change CS_NEW -> CS_INIT
>     2009-10-14 22:26:00.55225 [DEBUG] switch_core_session.c:933 Send
>     signal
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210> [BREAK]
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     Running State Change CS_INIT
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:480
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State INIT
>     2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:83
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>
>     SOFIA INIT
>     2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:111
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State Change CS_INIT -> CS_ROUTING
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:933 Send
>     signal
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210> [BREAK]
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:480
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State INIT going to sleep
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     Running State Change CS_ROUTING
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:483
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State ROUTING
>     2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:130
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>
>     SOFIA ROUTING
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:78
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>
>     Standard ROUTING
>     2009-10-14 22:26:00.63551 [INFO] mod_dialplan_xml.c:252 Processing
>     1001->1004 in context public
>     Dialplan: sofia/internal/1001 at 192.168.2.210
>     <mailto:1001 at 192.168.2.210> parsing [public->unloop]
>     continue=false
>     Dialplan: sofia/internal/1001 at 192.168.2.210
>     <mailto:1001 at 192.168.2.210> Regex (PASS) [unloop]
>     ${unroll_loops}(true) =~ /^true$/ break=on-false
>     Dialplan: sofia/internal/1001 at 192.168.2.210
>     <mailto:1001 at 192.168.2.210> Regex (FAIL) [unloop]
>     ${sip_looped_call}() =~ /^true$/ break=on-false
>     Dialplan: sofia/internal/1001 at 192.168.2.210
>     <mailto:1001 at 192.168.2.210> parsing
>     [public->outside_call] continue=true
>     Dialplan: sofia/internal/1001 at 192.168.2.210
>     <mailto:1001 at 192.168.2.210> Absolute Condition
>     [outside_call]
>     Dialplan: sofia/internal/1001 at 192.168.2.210
>     <mailto:1001 at 192.168.2.210> Action set(outside_call=true)
>     Dialplan: sofia/internal/1001 at 192.168.2.210
>     <mailto:1001 at 192.168.2.210> parsing [public->call_debug]
>     continue=true
>     Dialplan: sofia/internal/1001 at 192.168.2.210
>     <mailto:1001 at 192.168.2.210> Regex (FAIL) [call_debug]
>     ${call_debug}(false) =~ /^true$/ break=never
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:114
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State Change CS_ROUTING -> CS_EXECUTE
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:933 Send
>     signal
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210> [BREAK]
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:483
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State ROUTING going to sleep
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     Running State Change CS_EXECUTE
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:490
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State EXECUTE
>     2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:173
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>
>     SOFIA EXECUTE
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:151
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>
>     Standard EXECUTE
>     EXECUTE sofia/internal/1001 at 192.168.2.210
>     <mailto:1001 at 192.168.2.210> set(outside_call=true)
>     2009-10-14 22:26:00.63551 [DEBUG] mod_dptools.c:748
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210> SET
>     [outside_call]=[true]
>     2009-10-14 22:26:00.63551 [NOTICE] switch_core_state_machine.c:179
>     Hangup sofia/internal/1001 at 192.168.2.210
>     <mailto:1001 at 192.168.2.210> [CS_EXECUTE] [NORMAL_CLEARING]
>     2009-10-14 22:26:00.63551 [DEBUG] switch_channel.c:1683 Send signal
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210> [KILL]
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:933 Send
>     signal
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210> [BREAK]
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:490
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State EXECUTE going to sleep
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     Running State Change CS_HANGUP
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:433
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State HANGUP
>     2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:338 Channel
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>
>     hanging up, cause: NORMAL_CLEARING
>     2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:414 Responding to INVITE
>     with: 480
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:46
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>
>     Standard HANGUP, cause: NORMAL_CLEARING
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:433
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State HANGUP going to sleep
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:475
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State Change CS_HANGUP -> CS_REPORTING
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:933 Send
>     signal
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210> [BREAK]
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     Running State Change CS_REPORTING
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:607
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State REPORTING
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:53
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>
>     Standard REPORTING, cause: NORMAL_CLEARING
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:607
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State REPORTING going to sleep
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:410
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State Change CS_REPORTING -> CS_DESTROY
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:1067 Session 2
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     Locked, Waiting on external entities
>     2009-10-14 22:26:00.63551 [NOTICE] switch_core_session.c:1085
>     Session 2
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>) Ended
>     2009-10-14 22:26:00.63551 [NOTICE] switch_core_session.c:1087 Close
>     Channel sofia/internal/1001 at 192.168.2.210
>     <mailto:1001 at 192.168.2.210> [CS_DESTROY]
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:559
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State DESTROY
>     2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:255
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>
>     SOFIA DESTROY
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:60
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>
>     Standard DESTROY
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:559
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State DESTROY going to sleep
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:559
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State DESTROY going to sleep
>
>     I am at a loss, any help would be welcome.
>
>     Orien Love
>
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From brian at freeswitch.org  Wed Oct 14 20:55:14 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 14 Oct 2009 20:55:14 -0700
Subject: [Freeswitch-users] Outbound sip address call fails
In-Reply-To: <00d701ca4d25$b7d615f0$278241d0$@com>
References: <00d701ca4d25$b7d615f0$278241d0$@com>
Message-ID: <8A5CCD06-A37C-423B-807B-01F24D537E49@freeswitch.org>

Sounds like bria is sending the invite via the proxy which is your  
local FS box and NOT direct.

/b

On Oct 14, 2009, at 4:26 PM, Lars Zeb wrote:

> I am trying to make a test call to the FreeSwitch conference  
> address, sip:888 at conference.freeswitch.org on a Bria softphone.  
> However, it fails with a 404 Not found message.
>
> I have struggled with this for a while. Is there something special I  
> must do on the Bria to make this happen correctly?
>
> Thanks, Lars
>
> http://pastebin.freeswitch.org/10715

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From brian at freeswitch.org  Wed Oct 14 20:55:48 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 14 Oct 2009 20:55:48 -0700
Subject: [Freeswitch-users] Media got stuck after attended transfer...
In-Reply-To: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330253@srv02.khdev.corp>
References: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330253@srv02.khdev.corp>
Message-ID: <EEB6A298-BCA7-45C5-9550-E0DB43162093@freeswitch.org>

Why exactly are you doing proxy media?

/b

On Oct 14, 2009, at 4:56 PM, Klaus Hochlehnert wrote:

> Does anyone have a solution for this?

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From brian at freeswitch.org  Wed Oct 14 20:56:15 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 14 Oct 2009 20:56:15 -0700
Subject: [Freeswitch-users] Strange problems after upgrade...
In-Reply-To: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330254@srv02.khdev.corp>
References: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330252@srv02.khdev.corp>
	<87f2f3b90910141703hc8311bfg8c1d1280708d70f0@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F04027F02330254@srv02.khdev.corp>
Message-ID: <1ADD58A4-E64B-4E16-9366-B205FBB33980@freeswitch.org>

Make current will NOT touch your configs and is designed to prevent  
build skew.

/b

On Oct 14, 2009, at 5:14 PM, Klaus Hochlehnert wrote:

> The conf and the scripts dir were untouched after that (which I  
> wanted because I didn?t want to reconfigure everything).

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From mgende at gendesign.com  Wed Oct 14 21:34:33 2009
From: mgende at gendesign.com (Michael Gende)
Date: Wed, 14 Oct 2009 23:34:33 -0500
Subject: [Freeswitch-users] FreeSWITCH-users Digest,
	Mike G. answer to 	Orien L
In-Reply-To: <4AD6973B.70106@tx.rr.com>
References: <mailman.1269.1255575509.2495.freeswitch-users@lists.freeswitch.org>
	<4AD6973B.70106@tx.rr.com>
Message-ID: <a87d43fd0910142134v366551a1p52f74e436f7f5709@mail.gmail.com>

I'll get to you in the morning (where do you live?) and we'll see what's
what.


On Wed, Oct 14, 2009 at 10:30 PM, Orien Love <orien at tx.rr.com> wrote:

> Mike,
>     Thank you for your quick response, this has been driving me nuts.
>
> in my third attempt I tried to follow the mulit home example. I did set
> the phones to register to the local lan IP. actually had them
> registering, but it did help it in any way still the same problem.
> After my post I reinstalled the package and set the firewall to allow
> phones on the wan to connect, as per
> http://doc.pfsense.org/index.php/FreeSWITCH#Firewall
> put my computer with the soft phone on the wan network (192,168.2.xxx)
> had it registering and it had the same problem. so far all the setups I
> have tried are not working, the phones register, but cannot
> dial/connect  another  phone, also does not go to voice mail, maybe it
> is in my dial plan. which is default.
>
> Shoot, I am not picky, I just want to get it to work. so any suggestions
> on how to accomplish this are greatly appreciated.
>
> My phones are currently a couple of polycom 601's, and a Pangolin
> Softphone.
>
> Thanks
>   Orien L.
>
>
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From msc at freeswitch.org  Wed Oct 14 21:38:24 2009
From: msc at freeswitch.org (Michael S Collins)
Date: Wed, 14 Oct 2009 21:38:24 -0700
Subject: [Freeswitch-users] Strange problems after upgrade...
In-Reply-To: <1ADD58A4-E64B-4E16-9366-B205FBB33980@freeswitch.org>
References: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330252@srv02.khdev.corp>
	<87f2f3b90910141703hc8311bfg8c1d1280708d70f0@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F04027F02330254@srv02.khdev.corp>
	<1ADD58A4-E64B-4E16-9366-B205FBB33980@freeswitch.org>
Message-ID: <3EFB01FB-6990-4D50-9A2A-0FE42CCE4C4E@freeswitch.org>

Everyone repeat after me:
make current is my friend
make current is my friend
make current is my friend

ALWAYS use make current unless you know more about FreeSWITCH than the  
project's authors do.

-MC

On Oct 14, 2009, at 8:56 PM, Brian West <brian at freeswitch.org> wrote:

> Make current will NOT touch your configs and is designed to prevent  
> build skew.
>
> /b
>
> On Oct 14, 2009, at 5:14 PM, Klaus Hochlehnert wrote:
>
>> The conf and the scripts dir were untouched after that (which I  
>> wanted because I didn?t want to reconfigure everything).
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From larclap at yahoo.com  Thu Oct 15 00:00:45 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Thu, 15 Oct 2009 00:00:45 -0700
Subject: [Freeswitch-users] Outbound sip address call fails
In-Reply-To: <8A5CCD06-A37C-423B-807B-01F24D537E49@freeswitch.org>
References: <00d701ca4d25$b7d615f0$278241d0$@com>
	<8A5CCD06-A37C-423B-807B-01F24D537E49@freeswitch.org>
Message-ID: <001e01ca4d65$3814bfc0$a83e3f40$@com>

Brian,

 

Does this mean that Bria should be sending the invite directly to
sip:888 at conference.freeswitch.org and the From address should be the address
of the Bria itself rather than FS?

 

Lars

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Wednesday, October 14, 2009 8:55 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Outbound sip address call fails

 

Sounds like bria is sending the invite via the proxy which is your local FS
box and NOT direct.

 

/b

 

On Oct 14, 2009, at 4:26 PM, Lars Zeb wrote:





I am trying to make a test call to the FreeSwitch conference address,
sip:888 at conference.freeswitch.org on a Bria softphone. However, it fails
with a 404 Not found message.

 

I have struggled with this for a while. Is there something special I must do
on the Bria to make this happen correctly?

 

Thanks, Lars

 

http://pastebin.freeswitch.org/10715

 

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From odermann at googlemail.com  Thu Oct 15 00:47:28 2009
From: odermann at googlemail.com (Dennis)
Date: Thu, 15 Oct 2009 09:47:28 +0200
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
	<5e414ed0910130826j11d45a66q86d785f900e4196e@mail.gmail.com>
	<65d96fc80910131012v1f72ac49o6bfe9f2267e541c5@mail.gmail.com>
	<191c3a030910131101p418e0961t373db68285248ff6@mail.gmail.com>
	<538F4CF43B294D0EACABA281E1913B4C@chaos>
	<65d96fc80910131224m73f35b87g355be818ad3c0caf@mail.gmail.com>
	<48B637E0506F4A809E4C0CAF39BD08BB@chaos>
	<65d96fc80910132348t202905fbub57cc4c814eb4e21@mail.gmail.com>
	<5e414ed0910140731w1c7ebedr150e69cda8073155@mail.gmail.com>
	<191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com>
Message-ID: <5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com>

> once you have 123456 won't you still be unsure if he will type the next 1 or
> not and be forced to refuse it and wait anyway?

basically you are right. BUT, we know, that a basic phone number has 6
digits - so, we do not have to check anything before. as soon as we
have 6 digits, we look in our database for this number. a flag in the
database tells us, if this number is allowed to use ddi (further
digits) or not. if the number is not allowed, we can answer the call
directly without any further waiting.
other numbers are maked to have 2 ddi's - here we know, that we have
to wait for 2 more digits, before we answer.
if a number can have 6 digits or 8 digits, you are right, we have to
wait anyway.

but without having overlap, there is ALWAYS a waiting time.

we would like to make the answer-times as short as possible. if no
waiting is needed, we do not want to wait.

we would like to tell fs to send a 484 response incomplete to the
cirpack of our carrier if we need/want more digits.
perhaps this works with fs, but not with "socket outbound", which we are using?

we know, that this is more work for our servers, but they are powerful
enough and have enough resources, that they should be able to handle
that.
of course we do not want to become to experimental, because fs works
so extremly smooth and reliable, that it is nearly unbelievable.


From nbelov at abisoft.spb.ru  Thu Oct 15 01:16:21 2009
From: nbelov at abisoft.spb.ru (Nikita Belov)
Date: Thu, 15 Oct 2009 12:16:21 +0400
Subject: [Freeswitch-users] conference call
In-Reply-To: <ebbc2d550910140838v3125579ev9fa16e3e83d25770@mail.gmail.com>
References: <064A5201C10D4D20AC76EB93CA5EA4B5@abisoft.biz>
	<ebbc2d550910140838v3125579ev9fa16e3e83d25770@mail.gmail.com>
Message-ID: <BE4EDDA84CB64AA98F18D412B51912FC@abisoft.biz>

Thanks for answer. 

 

As I've understood eavesdrop allows C to hear A and B, but A and B will not
hear C. But in my case B SHOULD hear C. In my case A - client, B - agent
worked with client and C - coach/supervisor for B. Can I make such
conference in FS using eavesdrop or other API command?

 

___

 

Thanks, Nikita

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Joao
Mesquita
Sent: Wednesday, October 14, 2009 7:38 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] conference call

 

Look at eavesdrop on the wiki.

JM

2009/10/14 Nikita Belov <nbelov at abisoft.spb.ru>

HI all,

 

I want to configure FS to make special conference call between three users
(A, B, C).  In this conference C will hear A and B, but A will hear only B.
Can I make it using FS API commands? Does anybody know which approach is
better to use?

 

___

 

Thanks, Nikita


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

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From god.nirvana at gmail.com  Thu Oct 15 04:44:49 2009
From: god.nirvana at gmail.com (god.nirvana)
Date: Thu, 15 Oct 2009 19:44:49 +0800
Subject: [Freeswitch-users] conference digits and conference control
Message-ID: <200910151944464068246@gmail.com>

hi all
      how can i get the digits when users in the conference??
      and,in conference.conf.xml
      <control action="mute" digits="0"/> the "action" will set another value?e.g:transfer?
      thanks

2009-10-15 



god.nirvana 
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From rupa at rupa.com  Thu Oct 15 05:04:17 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 15 Oct 2009 07:04:17 -0500
Subject: [Freeswitch-users] conference call
In-Reply-To: <BE4EDDA84CB64AA98F18D412B51912FC@abisoft.biz>
References: <064A5201C10D4D20AC76EB93CA5EA4B5@abisoft.biz>
	<ebbc2d550910140838v3125579ev9fa16e3e83d25770@mail.gmail.com>
	<BE4EDDA84CB64AA98F18D412B51912FC@abisoft.biz>
Message-ID: <b39273530910150504l5c7411b9g39b9ed33c592e85f@mail.gmail.com>

The wiki has a pretty good explanation of how eavesdrop works.
Enabling a talk path to A or B or both A and B requires dtmf.

So, if C hits the 1 button on the phone they can talk to the UUID you
bound the eavesdrop to.   2 talks to the other party.  3 talks to both
parties.

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop

Adding more api calls to control eavesdrop via event socket should be
possible, though I haven't looked at the code to see what it would
take...

On Thu, Oct 15, 2009 at 3:16 AM, Nikita Belov <nbelov at abisoft.spb.ru> wrote:
> Thanks for answer.
>
>
>
> As I?ve understood eavesdrop allows C to hear A and B, but A and B will not
> hear C. But in my case B SHOULD hear C. In my case A ? client, B ? agent
> worked with client and C ? coach/supervisor for B. Can I make such
> conference in FS using eavesdrop or other API command?
>
>
>
> ___
>
>
>
> Thanks, Nikita
>
>
>
> ________________________________
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Joao
> Mesquita
> Sent: Wednesday, October 14, 2009 7:38 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] conference call
>
>
>
> Look at eavesdrop on the wiki.
>
> JM
>
> 2009/10/14 Nikita Belov <nbelov at abisoft.spb.ru>
>
> HI all,
>
>
>
> I want to configure FS to make special conference call between three users
> (A, B, C).? In this conference C will hear A and B, but A will hear only B.
> Can I make it using FS API commands? Does anybody know which approach is
> better to use?
>
>
>
> ___
>
>
>
> Thanks, Nikita
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa


From anthony.minessale at gmail.com  Thu Oct 15 06:53:37 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 15 Oct 2009 08:53:37 -0500
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
	<65d96fc80910131012v1f72ac49o6bfe9f2267e541c5@mail.gmail.com>
	<191c3a030910131101p418e0961t373db68285248ff6@mail.gmail.com>
	<538F4CF43B294D0EACABA281E1913B4C@chaos>
	<65d96fc80910131224m73f35b87g355be818ad3c0caf@mail.gmail.com>
	<48B637E0506F4A809E4C0CAF39BD08BB@chaos>
	<65d96fc80910132348t202905fbub57cc4c814eb4e21@mail.gmail.com>
	<5e414ed0910140731w1c7ebedr150e69cda8073155@mail.gmail.com>
	<191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com>
	<5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com>
Message-ID: <191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com>

right you can reply 484 in your dp at any time
<action application="respond" data="484 Address Incomplete"/>

then it should try again.

The bit i can't remember is if we committed a certain 1 line patch that
makes sofia parse the next invite to the same call properly, the patch was
to the sofia lib itself so test it and see.  I may need to dig up the answer
again from the sofia dev.



On Thu, Oct 15, 2009 at 2:47 AM, Dennis <odermann at googlemail.com> wrote:

> > once you have 123456 won't you still be unsure if he will type the next 1
> or
> > not and be forced to refuse it and wait anyway?
>
> basically you are right. BUT, we know, that a basic phone number has 6
> digits - so, we do not have to check anything before. as soon as we
> have 6 digits, we look in our database for this number. a flag in the
> database tells us, if this number is allowed to use ddi (further
> digits) or not. if the number is not allowed, we can answer the call
> directly without any further waiting.
> other numbers are maked to have 2 ddi's - here we know, that we have
> to wait for 2 more digits, before we answer.
> if a number can have 6 digits or 8 digits, you are right, we have to
> wait anyway.
>
> but without having overlap, there is ALWAYS a waiting time.
>
> we would like to make the answer-times as short as possible. if no
> waiting is needed, we do not want to wait.
>
> we would like to tell fs to send a 484 response incomplete to the
> cirpack of our carrier if we need/want more digits.
> perhaps this works with fs, but not with "socket outbound", which we are
> using?
>
> we know, that this is more work for our servers, but they are powerful
> enough and have enough resources, that they should be able to handle
> that.
> of course we do not want to become to experimental, because fs works
> so extremly smooth and reliable, that it is nearly unbelievable.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From brian at freeswitch.org  Thu Oct 15 06:54:58 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 15 Oct 2009 06:54:58 -0700
Subject: [Freeswitch-users] Outbound sip address call fails
In-Reply-To: <001e01ca4d65$3814bfc0$a83e3f40$@com>
References: <00d701ca4d25$b7d615f0$278241d0$@com>
	<8A5CCD06-A37C-423B-807B-01F24D537E49@freeswitch.org>
	<001e01ca4d65$3814bfc0$a83e3f40$@com>
Message-ID: <D9915F47-D22D-4D6F-8B27-F76356A7CBBB@freeswitch.org>

no clue it depends on how you have it configured.  If you have an  
outbound proxy set it'll send all calls to the proxy.  If not then  
most likely it will go direct to the destination.

/b

On Oct 15, 2009, at 12:00 AM, Lars Zeb wrote:

> Brian,
>
> Does this mean that Bria should be sending the invite directly to sip:888 at conference.freeswitch.org 
>  and the From address should be the address of the Bria itself  
> rather than FS?
>
> Lars
>

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From anthony.minessale at gmail.com  Thu Oct 15 06:57:48 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 15 Oct 2009 08:57:48 -0500
Subject: [Freeswitch-users] Media got stuck after attended transfer...
In-Reply-To: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330253@srv02.khdev.corp>
References: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330253@srv02.khdev.corp>
Message-ID: <191c3a030910150657r668eb5a3q24c641e312d2b113@mail.gmail.com>

This is a known limitation until we add actual t38 support to the project.


On Wed, Oct 14, 2009 at 6:56 PM, Klaus Hochlehnert <Mailings at kh-dev.de>wrote:

>  Hi,
>
>
>
> sometimes I have the problem that after doing an attended transfer the
> media got stuck in FS.
>
> Meaning the call goes through, but I don?t hear anything and the caller
> still hears music.
>
>
>
> Now I found out that setting the sofia parameter
>  media-option=resume-media-on-hold  helps here.
>
> But after setting this parameter I always get the error ?Codec PROXY
> PASS-THROUGH encoder error!? when using t38modem with proxy_media=true.
>
>
>
> So I stuck here. I can either use fax or do attended transfers.
>
>
>
> Does anyone have a solution for this?
>
>
>
> Thanks, Klaus
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From jbarou at sqli.com  Thu Oct 15 07:10:53 2009
From: jbarou at sqli.com (Jonathan Barou)
Date: Thu, 15 Oct 2009 16:10:53 +0200
Subject: [Freeswitch-users] Alias user mapping
In-Reply-To: <4AC27BC1.2050201@ParcelPool.com>
References: <8048ff7f0909290633t24f15224v40fcfe5aa0c8fdf0@mail.gmail.com>
	<4AC27BC1.2050201@ParcelPool.com>
Message-ID: <8048ff7f0910150710g810c7e2kbb0f8b0f0a28ef6b@mail.gmail.com>

Thanks, now it's working, but in my user directory I have something like
this :

  <user id="john" number-alias="1368">

and in my dialplan.xml I have to put the two entry for working :

<condition field="destination_number" expression="^(toto|john|8400|1368)$">

If I only put john, when I try to call the "1368" it's not working. Is it a
normal way ?

Thanks



2009/9/29 Peter Hinman <Peter.Hinman at parcelpool.com>

>  I've noticed that when using an alphanumeric ID, case sensitivity causes
> issues.  It doesn't seem to matter what case the userID is in the XML
> directory.  What does seem to matter is how the user registered their
> phone.  If the user registered in all upper case (JOHN_SMITH), then dialing
> John_Smith will not work.
>
> My work around for this is to tell all users to register their phones in
> lower case, and then my dial plan uses LUA to force all dialed extensions to
> lowercase.
>
> Peter
>
>
> Jonathan Barou wrote:
>
> Hi everybody,
>  I have a problem with "Alphanumeric to numeric user mapping"
>
>  I have done like it's written here :
>
>
>  Alphanumeric to numeric user mapping
>
> Say you want a user's id to be alphanumeric (like an email username), such
> as johnsmith at pbx.example.com. These users have alphanumeric usernames in
> their sip phone config, but you want to map them from their sip username to
> a numeric extension, and vice versa.
>
> As of version 1.0.4, Freeswitch makes this trivial to accomplish. A user's
> ID can be any alphanumeric string, and this can be simply tied to an
> extension number using the 'number-alias' property. This property creates an
> aliased directory entry that points to the alphanumeric user entry.
>
> *NOTE: When using this attribute, you must be careful not to create a
> directory collision by having another user whose ID is the same as another
> user's alias*
>
> Here is an example from the user directory:
>
>   <user id="johnsmith" number-alias="1001">
>     <!-- Insert the usual user configuration variables and params here,
>          including user password, voicemail password, caller ID info, etc -->
>   </user>
>
>
> So when a user dials extension number 1001, your dialplan can use the
> 'user_data' function to look up the ID attribute associated with that number
> alias. In the default dialplan, the 'Local Extension' section can be made to
> work with a small change to the 'bridge' line:
>
>   <action application="bridge" data="user/${user_data(${dialed_extension}@${domain_name} attr id)}@${domain_name}"/>
>
>
> *NOTE: Using this user_data function in combination with mod_xml_curl will
> generate an additional request each time the user_data function is called.
> Note that it is already called once in the Local Extension section to
> determine the callgroup. Beware of performance implications of this with
> high-volume systems.*
>
>
>  But when I want to call my alias-number, FS says "No Route, Abording"
>
>  My version of FreeSWITCH is the 1.0.4pre9.
>
>  Do you have any ideas ? Did I forget something to do?
>
>  Thanks
>
>
>
> --
> Jonathan BAROU
> SQLI LYON - CRCI
> jbarou at sqli.com
> lyon.crci at sqli.com
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Jonathan BAROU
SQLI LYON - CRCI
jbarou at sqli.com
lyon.crci at sqli.com
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From odermann at googlemail.com  Thu Oct 15 07:18:56 2009
From: odermann at googlemail.com (Dennis)
Date: Thu, 15 Oct 2009 16:18:56 +0200
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
	<191c3a030910131101p418e0961t373db68285248ff6@mail.gmail.com>
	<538F4CF43B294D0EACABA281E1913B4C@chaos>
	<65d96fc80910131224m73f35b87g355be818ad3c0caf@mail.gmail.com>
	<48B637E0506F4A809E4C0CAF39BD08BB@chaos>
	<65d96fc80910132348t202905fbub57cc4c814eb4e21@mail.gmail.com>
	<5e414ed0910140731w1c7ebedr150e69cda8073155@mail.gmail.com>
	<191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com>
	<5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com>
	<191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com>
Message-ID: <5e414ed0910150718k41d747a6t6ca98daf1c2a01f4@mail.gmail.com>

ok, we will try this with the cirpack of our carrier. this will take
some days, till everything is set up.

after the tests i will come back to report.


2009/10/15 Anthony Minessale <anthony.minessale at gmail.com>:
> right you can reply 484 in your dp at any time
> <action application="respond" data="484 Address Incomplete"/>
>
> then it should try again.
>
> The bit i can't remember is if we committed a certain 1 line patch that
> makes sofia parse the next invite to the same call properly, the patch was
> to the sofia lib itself so test it and see.? I may need to dig up the answer
> again from the sofia dev.


From mgende at gendesign.com  Thu Oct 15 07:36:55 2009
From: mgende at gendesign.com (Michael Gende)
Date: Thu, 15 Oct 2009 09:36:55 -0500
Subject: [Freeswitch-users] FreeSWITCH-users Digest,
	Mike G. answer to 	Orien L
In-Reply-To: <a87d43fd0910142134v366551a1p52f74e436f7f5709@mail.gmail.com>
References: <mailman.1269.1255575509.2495.freeswitch-users@lists.freeswitch.org>
	<4AD6973B.70106@tx.rr.com>
	<a87d43fd0910142134v366551a1p52f74e436f7f5709@mail.gmail.com>
Message-ID: <a87d43fd0910150736k2690d130gcebf829f74d6950@mail.gmail.com>

Hey Orien,

OK, let's have a look here. How about going to the console of your FS and
posting the output of the following commands:

sofia status profile internal
sofia status profile external

We'll take it from there. Your problems are just like what I ran into
initially, so I'm hopeful we get round them.

Mike G.

On Wed, Oct 14, 2009 at 11:34 PM, Michael Gende <mgende at gendesign.com>wrote:

> I'll get to you in the morning (where do you live?) and we'll see what's
> what.
>
>
>
> On Wed, Oct 14, 2009 at 10:30 PM, Orien Love <orien at tx.rr.com> wrote:
>
>> Mike,
>>     Thank you for your quick response, this has been driving me nuts.
>>
>> in my third attempt I tried to follow the mulit home example. I did set
>> the phones to register to the local lan IP. actually had them
>> registering, but it did help it in any way still the same problem.
>> After my post I reinstalled the package and set the firewall to allow
>> phones on the wan to connect, as per
>> http://doc.pfsense.org/index.php/FreeSWITCH#Firewall
>> put my computer with the soft phone on the wan network (192,168.2.xxx)
>> had it registering and it had the same problem. so far all the setups I
>> have tried are not working, the phones register, but cannot
>> dial/connect  another  phone, also does not go to voice mail, maybe it
>> is in my dial plan. which is default.
>>
>> Shoot, I am not picky, I just want to get it to work. so any suggestions
>> on how to accomplish this are greatly appreciated.
>>
>> My phones are currently a couple of polycom 601's, and a Pangolin
>> Softphone.
>>
>> Thanks
>>   Orien L.
>>
>>
>
>
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From larclap at yahoo.com  Thu Oct 15 07:56:52 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Thu, 15 Oct 2009 07:56:52 -0700
Subject: [Freeswitch-users] lua script causing FreeSwitch to crash?
In-Reply-To: <87f2f3b90910131312k7dac8980h70ce8be9c07aa7b2@mail.gmail.com>
References: <005a01ca4c3c$c183ceb0$448b6c10$@com>
	<87f2f3b90910131312k7dac8980h70ce8be9c07aa7b2@mail.gmail.com>
Message-ID: <007901ca4da7$bafed320$30fc7960$@com>

Thanks, Michael, using "make current" resolved the issue. I will use only
that command in the future.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, October 13, 2009 1:12 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] lua script causing FreeSwitch to crash?

 

 

On Tue, Oct 13, 2009 at 12:38 PM, Lars Zeb <larclap at yahoo.com> wrote:

http://pastebin.freeswitch.org/10686

 

I am trying to use a lua script for inbound calls. The caller hears a busy
signal. After this call fails to go through, FreeSwitch is no longer in
memory - it does not appear in  'ps -ef' output.

 

I don't have a clue what I might be doing incorrectly in the script in order
to cause this to happen. There is no indication in the log that FreeSwitch
has terminated. But it is not resident in memory.

 

I am using v15134. I had been using this same script without incident as
late as v14929. Until I can resolve the issue, I have removed the call to
lua from the dialplan and done all I could in it instead.

We sent out a notice about a bug in event socket between revs 15119 and
15134. Update to latest trunk. (FYI, the fix was in r15135, so you just
barely missed it.)

After you update, let us know if the problem persists. 
-MC

 

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From sjmudd at pobox.com  Thu Oct 15 08:03:56 2009
From: sjmudd at pobox.com (Simon J Mudd)
Date: 15 Oct 2009 17:03:56 +0200
Subject: [Freeswitch-users] macros for performing repeated dialplan
	actions?
In-Reply-To: <87f2f3b90910141510n45401995y11e3e380653e9790@mail.gmail.com>
References: <20091014220135.GA12614@mad06.wl0.org>
	<87f2f3b90910141510n45401995y11e3e380653e9790@mail.gmail.com>
Message-ID: <m3eip41yxv.fsf@mad06.wl0.org>

Hi Michael,

msc at freeswitch.org (Michael Collins) writes:

> On Wed, Oct 14, 2009 at 3:01 PM, Simon J Mudd <sjmudd at pobox.com> wrote:
> 
> > I'm trying to build up a soho dialplan to replicate an existing Asterisk
> > setup I want to replace.

...

> > My question is how do I avoid repeatedly using these same 8-9 lines and
> > "merge" this into some macro which given the right parameters allows me
> > to define various extensions using different gateways but all configured
> > "the same way"?
> >
> > Thanks for any pointers to how this should be done.
> 
> Try using the execute_extension app. It's kinda like a gosub in BASIC: it'll
> go and execute an extension and then come back and continue executing. Put
> all those lines into a single extension and then call them from other
> extensions via execute_extension.

OK. That looks clear. However it doesn't look as if it's possible to
pass parameters this way which means that data can only be passed to
the "extension" via variables. That works but isn't so clean. I'll
check this out further shortly. I'm planning later to pass info like
"which gateway to talk to" and perhaps some sort of flag which
indicates "record: yes/no", and even include the bridge application
into this extension/macro.  That reduces the original dialplan pattern
matching to a much simpler and easier to configure setup for me.

excecute_extension does avoid a lot of redundancy, so thanks for the
pointer.  I'll get that working first.

Simon


From larclap at yahoo.com  Thu Oct 15 08:14:28 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Thu, 15 Oct 2009 08:14:28 -0700
Subject: [Freeswitch-users] Outbound sip address call fails
In-Reply-To: <D9915F47-D22D-4D6F-8B27-F76356A7CBBB@freeswitch.org>
References: <00d701ca4d25$b7d615f0$278241d0$@com>	<8A5CCD06-A37C-423B-807B-01F24D537E49@freeswitch.org>	<001e01ca4d65$3814bfc0$a83e3f40$@com>
	<D9915F47-D22D-4D6F-8B27-F76356A7CBBB@freeswitch.org>
Message-ID: <007e01ca4daa$30a74c90$91f5e5b0$@com>

I don't think I have a proxy. It's pretty standard installation.

 

                     Name          Type                               Data
State

============================================================================
=====================

                 internal       profile   sip:mod_sofia at 192.168.10.29:5060
RUNNING (0)

            internal-ipv6       profile           sip:mod_sofia@[::1]:5060
RUNNING (0)

                 external       profile   sip:mod_sofia at 192.168.10.29:5090
RUNNING (0)

                flowroute       gateway        sip:55555 at sip.flowroute.com
REGED

            192.168.10.29         alias                           internal
ALIASED

============================================================================
=====================

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, October 15, 2009 6:55 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Outbound sip address call fails

 

no clue it depends on how you have it configured.  If you have an outbound
proxy set it'll send all calls to the proxy.  If not then most likely it
will go direct to the destination.

 

/b

 

On Oct 15, 2009, at 12:00 AM, Lars Zeb wrote:





Brian,

 

Does this mean that Bria should be sending the invite directly to
sip:888 at conference.freeswitch.org and the From address should be the address
of the Bria itself rather than FS?

 

Lars

 

 

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From chad at apartmentlines.com  Thu Oct 15 09:33:55 2009
From: chad at apartmentlines.com (Chad Phillips -- Apartment Lines)
Date: Thu, 15 Oct 2009 12:33:55 -0400
Subject: [Freeswitch-users] macros for performing repeated dialplan
	actions?
In-Reply-To: <m3eip41yxv.fsf@mad06.wl0.org>
References: <20091014220135.GA12614@mad06.wl0.org>
	<87f2f3b90910141510n45401995y11e3e380653e9790@mail.gmail.com>
	<m3eip41yxv.fsf@mad06.wl0.org>
Message-ID: <1C80FD78-71B7-4D62-8C45-2F5BC8C1564C@apartmentlines.com>


On Oct 15, 2009, at 11:03 AM, Simon J Mudd wrote:

> OK. That looks clear. However it doesn't look as if it's possible to
> pass parameters this way which means that data can only be passed to
> the "extension" via variables. That works but isn't so clean. I'll
> check this out further shortly. I'm planning later to pass info like
> "which gateway to talk to" and perhaps some sort of flag which
> indicates "record: yes/no", and even include the bridge application
> into this extension/macro.  That reduces the original dialplan pattern
> matching to a much simpler and easier to configure setup for me.

here's a tip from a recovering asterisk user: don't feel like you have  
to do everything via the dialplan.  this was a constant mantra in  
asterisk, because calling AGI scripts was expensive.  this is not the  
case with FreeSWITCH.  the general rule of thumb is: as soon as it  
gets hard to do in the dialplan, farm it out to a script.  it's the  
way FreeSWITCH is designed, so don't fight it  ;)

that said, you still might be able to use execute_extension along with  
setting some channel variables before you call it.  those channel  
variables will be available in the executed extension, and can act  
pretty much like the arguments to a macro.


From Russell.Mosemann at cune.org  Thu Oct 15 09:48:10 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Thu, 15 Oct 2009 16:48:10 -0000
Subject: [Freeswitch-users] Outbound sip address call fails
In-Reply-To: <007e01ca4daa$30a74c90$91f5e5b0$@com>
Message-ID: <20091015164810.B6A253F3FA9@mail.cune.org>

Lars Zeb <larclap at yahoo.com> said:

> I don't think I have a proxy. It's pretty standard installation.

Is the phone itself configured to use a proxy?

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From Stefan.Weigel at allianz-warranty.com  Thu Oct 15 08:16:22 2009
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Thu, 15 Oct 2009 17:16:22 +0200
Subject: [Freeswitch-users] Changing display text on phones (sip_from_user)
Message-ID: <5003D7D3E06F514E8C682F18D223265C0465862EAA@AZWSMS03.azwarranty.int>

Hi list,

I'm playing around with this wonderful, mighty solution.
Setup so far:

I created a gateway pointing to my ITSP (sip.qsc.de). Outbound calls getting in an depending on the channel variable

sip_to_user I want to rewrite the sip_from_user (=text which get's displayed in the phones):

      <condition field="${sip_to_user}" expression="^0.*974$" break="never">
        <action application="set" data="sip_from_user='IT_Hotline'"/>
       ...

 My telephone rings but it's always showing the original dialer number. I also tried the variables

username
caller_id_name
sip_contact_user

in the application info log it's showing me:

Caller-Username: [01732898376]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [01732898376]
Caller-Caller-ID-Number: [01732898376]

[..]

variable_sip_from_user: ['IT_Hotline']
variable_sip_from_user_stripped: ['IT_Hotline']
variable_sip_contact_user: ['IT_Hotline']


Is there any chance to rewrite only the part that's getting displayed on the phone  (e.g. From: IT_Hotline sip:123456 at sip.qsc.de) ?


Thanks and best regards

Stefan

Stefan Weigel
Advanced IT-Professional
Tel.: +49 89 2000 48 975
Fax: +49 89 2000 48 566
eMail: Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com>
Allianz Automotive Services GmbH
Einsteinring 28
85609 Aschheim
Germany

http://www.allianz-warranty.com
Gesch?ftsf?hrung: Andreas R?sing, Horst Ziegler
Amtsgericht M?nchen, HRB 175682
F?r Umsatzsteuerzwecke: Ust-ID-Nr.: DE 262 617 720

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From Stefan.Weigel at allianz-warranty.com  Thu Oct 15 08:23:19 2009
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Thu, 15 Oct 2009 17:23:19 +0200
Subject: [Freeswitch-users] WG: Changing display text on phones
	(sip_from_user)
Message-ID: <5003D7D3E06F514E8C682F18D223265C0465862EAC@AZWSMS03.azwarranty.int>

Hi,

just found the answer:

just use

"effective_caller_id_name"

and  it work's :)



Cheers

Stefan

Stefan Weigel
Advanced IT-Professional
Tel.: +49 89 2000 48 975
Fax: +49 89 2000 48 566
eMail: Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com>
Allianz Automotive Services GmbH
Einsteinring 28
85609 Aschheim
Germany

http://www.allianz-warranty.com
Gesch?ftsf?hrung: Andreas R?sing, Horst Ziegler
Amtsgericht M?nchen, HRB 175682
F?r Umsatzsteuerzwecke: Ust-ID-Nr.: DE 262 617 720

Von: Weigel, Stefan
Gesendet: Donnerstag, 15. Oktober 2009 17:16
An: 'freeswitch-users at lists.freeswitch.org'
Betreff: Changing display text on phones (sip_from_user)

Hi list,

I'm playing around with this wonderful, mighty solution.
Setup so far:

I created a gateway pointing to my ITSP (sip.qsc.de). Outbound calls getting in an depending on the channel variable

sip_to_user I want to rewrite the sip_from_user (=text which get's displayed in the phones):

      <condition field="${sip_to_user}" expression="^0.*974$" break="never">
        <action application="set" data="sip_from_user='IT_Hotline'"/>
       ...

 My telephone rings but it's always showing the original dialer number. I also tried the variables

username
caller_id_name
sip_contact_user

in the application info log it's showing me:

Caller-Username: [01732898376]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [01732898376]
Caller-Caller-ID-Number: [01732898376]

[..]

variable_sip_from_user: ['IT_Hotline']
variable_sip_from_user_stripped: ['IT_Hotline']
variable_sip_contact_user: ['IT_Hotline']


Is there any chance to rewrite only the part that's getting displayed on the phone  (e.g. From: IT_Hotline sip:123456 at sip.qsc.de) ?


Thanks and best regards

Stefan

Stefan Weigel
Advanced IT-Professional
Tel.: +49 89 2000 48 975
Fax: +49 89 2000 48 566
eMail: Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com>
Allianz Automotive Services GmbH
Einsteinring 28
85609 Aschheim
Germany

http://www.allianz-warranty.com
Gesch?ftsf?hrung: Andreas R?sing, Horst Ziegler
Amtsgericht M?nchen, HRB 175682
F?r Umsatzsteuerzwecke: Ust-ID-Nr.: DE 262 617 720

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From larclap at yahoo.com  Thu Oct 15 10:17:25 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Thu, 15 Oct 2009 10:17:25 -0700
Subject: [Freeswitch-users] Outbound sip address call fails
In-Reply-To: <20091015164810.B6A253F3FA9@mail.cune.org>
References: <007e01ca4daa$30a74c90$91f5e5b0$@com>
	<20091015164810.B6A253F3FA9@mail.cune.org>
Message-ID: <00be01ca4dbb$5d913160$18b39420$@com>

I don't think Bria is set to use a proxy.

In the SIP Account/Account tab, it is set to "Register with domain and
receive calls" and "Send outbound via" is set to use Domain rather than
"Proxy Address".

The Account ID is set to 1009 at 192.168.10.29, where the IP is the address of
FreeSwitch.

> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-
> bounces at lists.freeswitch.org] On Behalf Of Russell.Mosemann at cune.org
> Sent: Thursday, October 15, 2009 9:48 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Outbound sip address call fails
> 
> Lars Zeb <larclap at yahoo.com> said:
> 
> > I don't think I have a proxy. It's pretty standard installation.
> 
> Is the phone itself configured to use a proxy?
> 
> --
> Russell Mosemann
> 
> 
> 
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From freeswitch at cartissolutions.com  Thu Oct 15 10:30:35 2009
From: freeswitch at cartissolutions.com (Yossi Neiman)
Date: Thu, 15 Oct 2009 12:30:35 -0500
Subject: [Freeswitch-users] apr_queue
In-Reply-To: <f8af5740910090458w6540db9fg972ad1ba8a691c30@mail.gmail.com>
References: <f8af5740910090458w6540db9fg972ad1ba8a691c30@mail.gmail.com>
Message-ID: <4AD75C3B.3030007@cartissolutions.com>

Not sure at all what you are asking.  apr_queue is simply a data 
management container that is threadsafe.

-- 
Yossi Neiman
Cartis Solutions, Inc. - http://www.cartissolutions.com



srinivasula reddy wrote:
> Hi all,
>
> does any know about How apr_queue is maintaing and retriving all 
> registered and all stuff....
>
>
> -- 
> Srinivasula Reddy K
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From Russell.Mosemann at cune.org  Thu Oct 15 10:33:13 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Thu, 15 Oct 2009 17:33:13 -0000
Subject: [Freeswitch-users] Outbound sip address call fails
In-Reply-To: <00be01ca4dbb$5d913160$18b39420$@com>
Message-ID: <20091015173313.3834B199FE2@mail.cune.org>

Lars Zeb <larclap at yahoo.com> said:

> I don't think Bria is set to use a proxy.
> 
> In the SIP Account/Account tab, it is set to "Register with domain and
> receive calls" and "Send outbound via" is set to use Domain rather than
> "Proxy Address".
> 
> The Account ID is set to 1009 at 192.168.10.29, where the IP is the
> address of FreeSwitch.

That's exactly how I have X-Lite configured, and I am able to call the
conference directly by entering the conference URL. I'm sitting behind
NAT, too. Under the Topology tab, it is set to "Discover global address"
and "Discover server" (for STUN). "Enable ICE" is checked. The next thing
to check might be if the phone traffic is going directly to the Internet
and getting past your firewall.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From orien at tx.rr.com  Thu Oct 15 10:43:21 2009
From: orien at tx.rr.com (Orien Love)
Date: Thu, 15 Oct 2009 12:43:21 -0500
Subject: [Freeswitch-users] Subject:, Re:  FreeSWITCH-users Digest,
 Orien L. answer to Mike G.
In-Reply-To: <mailman.1290.1255619700.2495.freeswitch-users@lists.freeswitch.org>
References: <mailman.1290.1255619700.2495.freeswitch-users@lists.freeswitch.org>
Message-ID: <4AD75F39.6010209@tx.rr.com>

Mike,
    I copied the status screen from the freeswitch package in the web 
setup. I you need a console view, I will need instructions on how to 
console into pfsense, Since I have never done that.

Orien L.

                     Name       Type                                  
Data    State
=================================================================================================
                 internal    profile       
sip:mod_sofia at 192.168.10.1:5060    RUNNING (0)
                 external    profile      
sip:mod_sofia at 192.168.2.210:5080    RUNNING (0)
             192.168.10.1      alias                              
internal    ALIASED
            internal-ipv6    profile              
sip:mod_sofia@[::1]:5060    RUNNING (0)
=================================================================================================
3 profiles 1 alias



sofia status profile external    

=================================================================================================
Name                 external
Domain Name          N/A
DBName               sofia_reg_external
Pres Hosts          
Dialplan             XML
Context              public
Challenge Realm      auto_to
RTP-IP               192.168.2.210
Ext-RTP-IP           192.168.2.210
SIP-IP               192.168.2.210
Ext-SIP-IP           192.168.2.210
URL                  sip:mod_sofia at 192.168.2.210:5080
BIND-URL             sip:mod_sofia at 192.168.2.210:5080
HOLD-MUSIC           local_stream://moh
OUTBOUND-PROXY       N/A
CODECS               PCMU,PCMA,GSM
TEL-EVENT            101
DTMF-MODE            rfc2833
CNG                  13
SESSION-TO           0
MAX-DIALOG           0
NOMEDIA              false
LATE-NEG             false
PROXY-MEDIA          false
AGGRESSIVENAT        false
STUN-ENABLED         true
STUN-AUTO-DISABLE    false
CALLS-IN             0
FAILED-CALLS-IN      0
CALLS-OUT            0
FAILED-CALLS-OUT     0

Registrations:
=================================================================================================
=================================================================================================



sofia status profile internal    

=================================================================================================
Name                 internal
Domain Name          N/A
DBName               sofia_reg_internal
Pres Hosts          
Dialplan             XML
Context              public
Challenge Realm      auto_from
RTP-IP               192.168.10.1
Ext-RTP-IP           192.168.10.1
SIP-IP               192.168.10.1
Ext-SIP-IP           192.168.10.1
URL                  sip:mod_sofia at 192.168.10.1:5060
BIND-URL             sip:mod_sofia at 192.168.10.1:5060
HOLD-MUSIC           local_stream://moh
OUTBOUND-PROXY       N/A
CODECS               G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
TEL-EVENT            101
DTMF-MODE            rfc2833
CNG                  13
SESSION-TO           0
MAX-DIALOG           0
NOMEDIA              false
LATE-NEG             false
PROXY-MEDIA          false
AGGRESSIVENAT        false
STUN-ENABLED         true
STUN-AUTO-DISABLE    false
CALLS-IN             6
FAILED-CALLS-IN      3
CALLS-OUT            0
FAILED-CALLS-OUT     0

Registrations:
=================================================================================================
Call-ID:        NjczMzQzNTVlYTJhYjIzYmFiZmFjZGYzNzNlMTVkMTc.
User:           1001 at 192.168.10.1
Contact:        "user"
Agent:          Pangolin v5.2, Build 09182009
Status:         Registered(UDP)(unknown) EXP(2009-10-15 05:03:17)
Host:           orienhome2.local
IP:             192.168.10.243
Port:           3789
Auth-User:      1001
Auth-Realm:     192.168.10.1

=================================================================================================



sofia status profile internal-ipv6    

=================================================================================================
Name                 internal-ipv6
Domain Name          N/A
DBName               sofia_reg_internal-ipv6
Pres Hosts          
Dialplan             XML
Context              public
Challenge Realm      auto_to
RTP-IP               ::1
SIP-IP               ::1
URL                  sip:mod_sofia@[::1]:5060
BIND-URL             sip:mod_sofia@[::1]:5060
HOLD-MUSIC           local_stream://moh
OUTBOUND-PROXY       N/A
CODECS               G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
TEL-EVENT            101
DTMF-MODE            rfc2833
CNG                  13
SESSION-TO           0
MAX-DIALOG           0
NOMEDIA              false
LATE-NEG             false
PROXY-MEDIA          false
AGGRESSIVENAT        false
STUN-ENABLED         true
STUN-AUTO-DISABLE    false
CALLS-IN             0
FAILED-CALLS-IN      0
CALLS-OUT            0
FAILED-CALLS-OUT     0

Registrations:
=================================================================================================
=================================================================================================



status

UP 0 years, 0 days, 6 hours, 55 minutes, 32 seconds, 640 milliseconds, 
638 microseconds
43 session(s) since startup
0 session(s) 0/30
1000 session(s) max


> Subject:
> Re: [Freeswitch-users] FreeSWITCH-users Digest, Mike G. answer to Orien L
> From:
> Michael Gende <mgende at gendesign.com>
> Date:
> Thu, 15 Oct 2009 09:36:55 -0500
> To:
> freeswitch-users at lists.freeswitch.org
>
> To:
> freeswitch-users at lists.freeswitch.org
>
>
> Hey Orien,
>
> OK, let's have a look here. How about going to the console of your FS 
> and posting the output of the following commands:
>
> sofia status profile internal
> sofia status profile external
>
> We'll take it from there. Your problems are just like what I ran into 
> initially, so I'm hopeful we get round them.
>
> Mike G.
>
> On Wed, Oct 14, 2009 at 11:34 PM, Michael Gende <mgende at gendesign.com 
> <mailto:mgende at gendesign.com>> wrote:
>
>     I'll get to you in the morning (where do you live?) and we'll see
>     what's what.
>
>
>
>     On Wed, Oct 14, 2009 at 10:30 PM, Orien Love <orien at tx.rr.com
>     <mailto:orien at tx.rr.com>> wrote:
>
>         Mike,
>             Thank you for your quick response, this has been driving
>         me nuts.
>
>         in my third attempt I tried to follow the mulit home example.
>         I did set
>         the phones to register to the local lan IP. actually had them
>         registering, but it did help it in any way still the same problem.
>         After my post I reinstalled the package and set the firewall
>         to allow
>         phones on the wan to connect, as per
>         http://doc.pfsense.org/index.php/FreeSWITCH#Firewall
>         put my computer with the soft phone on the wan network
>         (192,168.2.xxx)
>         had it registering and it had the same problem. so far all the
>         setups I
>         have tried are not working, the phones register, but cannot
>         dial/connect  another  phone, also does not go to voice mail,
>         maybe it
>         is in my dial plan. which is default.
>
>         Shoot, I am not picky, I just want to get it to work. so any
>         suggestions
>         on how to accomplish this are greatly appreciated.
>
>         My phones are currently a couple of polycom 601's, and a
>         Pangolin Softphone.
>
>         Thanks
>           Orien L.
>          
>
>
>



From msc at freeswitch.org  Thu Oct 15 11:09:22 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 15 Oct 2009 11:09:22 -0700
Subject: [Freeswitch-users] lua script causing FreeSwitch to crash?
In-Reply-To: <007901ca4da7$bafed320$30fc7960$@com>
References: <005a01ca4c3c$c183ceb0$448b6c10$@com>
	<87f2f3b90910131312k7dac8980h70ce8be9c07aa7b2@mail.gmail.com>
	<007901ca4da7$bafed320$30fc7960$@com>
Message-ID: <87f2f3b90910151109g1c317c97n43bed3d90b408548@mail.gmail.com>

On Thu, Oct 15, 2009 at 7:56 AM, Lars Zeb <larclap at yahoo.com> wrote:

>  Thanks, Michael, using ?make current? resolved the issue. I will use only
> that command in the future.
>
Excellent! Thanks for the feedback.
-MC
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From msc at freeswitch.org  Thu Oct 15 11:13:33 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 15 Oct 2009 11:13:33 -0700
Subject: [Freeswitch-users] Alias user mapping
In-Reply-To: <8048ff7f0910150710g810c7e2kbb0f8b0f0a28ef6b@mail.gmail.com>
References: <8048ff7f0909290633t24f15224v40fcfe5aa0c8fdf0@mail.gmail.com>
	<4AC27BC1.2050201@ParcelPool.com>
	<8048ff7f0910150710g810c7e2kbb0f8b0f0a28ef6b@mail.gmail.com>
Message-ID: <87f2f3b90910151113x43871d2fw41e99d638787563a@mail.gmail.com>

On Thu, Oct 15, 2009 at 7:10 AM, Jonathan Barou <jbarou at sqli.com> wrote:

> Thanks, now it's working, but in my user directory I have something like
> this :
>
>   <user id="john" number-alias="1368">
>
> and in my dialplan.xml I have to put the two entry for working :
>
> <condition field="destination_number" expression="^(toto|john|8400|1368)$">
>
> If I only put john, when I try to call the "1368" it's not working. Is it a
> normal way ?
>

Correct. The number-alias in the directory has no relation to the
destination_number matching in the dialplan. The destination_number match
will match whatever the person physically dialed on the telephone, so leave
the 1368 in there.

-MC
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From davis.erwin at gmail.com  Thu Oct 15 11:30:19 2009
From: davis.erwin at gmail.com (Erwin Davis)
Date: Thu, 15 Oct 2009 14:30:19 -0400
Subject: [Freeswitch-users] failed in bridging with SIPp
Message-ID: <a38a91d50910151130s1413cbc8yc39c58f8ca6473c7@mail.gmail.com>

Hi,

I tried to originate a call from x-lite and hope to terminate to SIPp
running in another machine (192.168.1.36:5069)
through FS. My x-lite received service no available and the FS report
NORMAL_TEMPOARY_FAILURE
(http://pastebin.freeswitch.org/10720). My purpose for FS is to simply
forward the call to the destination IP addr,
which is 192.168.1.36. Did I miss anything in the configuration? Thanks,

Below is my configuration,

in dialplan/default.xml,
    <extension name="to-sipp">
      <condition field="destination_number" expression="^6666$">
    <action application="bridge" data="sofia/internal/profile1/$
0 at 192.168.1.36"/>
      </condition>
    </extension>

in sip-profile/profile1.xml

<profile name="profile1">
  <param name="debug" value="1"/>
  <param name="rfc2833-pt" value="101"/>
  <param name="sip-ip" value="192.168.1.36"/>
  <param name="sip-port" value="5069"/>
  <param name="dialplan" value="XML"/>
  <param name="dtmf-duration" value="100"/>
  <param name="codec-ms" value="20"/>
  <param name="use-rtp-timer" value="true"/>
</profile>
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From tculjaga at gmail.com  Thu Oct 15 11:54:34 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 15 Oct 2009 20:54:34 +0200
Subject: [Freeswitch-users] Media got stuck after attended transfer...
In-Reply-To: <191c3a030910150657r668eb5a3q24c641e312d2b113@mail.gmail.com>
References: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330253@srv02.khdev.corp>
	<191c3a030910150657r668eb5a3q24c641e312d2b113@mail.gmail.com>
Message-ID: <65d96fc80910151154w2468ebeie06211d0966b4548@mail.gmail.com>

hi, any clue when can t38 be added?

T.

On Thu, Oct 15, 2009 at 3:57 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> This is a known limitation until we add actual t38 support to the project.
>
>
> On Wed, Oct 14, 2009 at 6:56 PM, Klaus Hochlehnert <Mailings at kh-dev.de>wrote:
>
>>  Hi,
>>
>>
>>
>> sometimes I have the problem that after doing an attended transfer the
>> media got stuck in FS.
>>
>> Meaning the call goes through, but I don?t hear anything and the caller
>> still hears music.
>>
>>
>>
>> Now I found out that setting the sofia parameter
>>  media-option=resume-media-on-hold  helps here.
>>
>> But after setting this parameter I always get the error ?Codec PROXY
>> PASS-THROUGH encoder error!? when using t38modem with proxy_media=true.
>>
>>
>>
>> So I stuck here. I can either use fax or do attended transfers.
>>
>>
>>
>> Does anyone have a solution for this?
>>
>>
>>
>> Thanks, Klaus
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From larclap at yahoo.com  Thu Oct 15 13:09:30 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Thu, 15 Oct 2009 13:09:30 -0700
Subject: [Freeswitch-users] Outbound sip address call fails
In-Reply-To: <20091015173313.3834B199FE2@mail.cune.org>
References: <00be01ca4dbb$5d913160$18b39420$@com>
	<20091015173313.3834B199FE2@mail.cune.org>
Message-ID: <011301ca4dd3$67d486a0$377d93e0$@com>

Russell,

My settings in Bria are just like yours.

I captured the SIP trace from the Bria and it says it is using the
FreeSwitch as the Outbound Proxy. I am very confused.

I tried dial sip:echo at iptel.org. In the pastebin you can see that this
address came over to the cli as "echo". Is it Bria which is messing up, or
my FS configuration?

Where did Bria get the Contact info at lines 79 & 82? 192.168.10.11 is the
address of the softphone, so I don't think sip:1009 at 192.168.10.11 makes
sense.

http://pastebin.freeswitch.org/10723

Lars


> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-
> bounces at lists.freeswitch.org] On Behalf Of Russell.Mosemann at cune.org
> Sent: Thursday, October 15, 2009 10:33 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Outbound sip address call fails
> 
> Lars Zeb <larclap at yahoo.com> said:
> 
> > I don't think Bria is set to use a proxy.
> >
> > In the SIP Account/Account tab, it is set to "Register with domain and
> > receive calls" and "Send outbound via" is set to use Domain rather than
> > "Proxy Address".
> >
> > The Account ID is set to 1009 at 192.168.10.29, where the IP is the
> > address of FreeSwitch.
> 
> That's exactly how I have X-Lite configured, and I am able to call the
> conference directly by entering the conference URL. I'm sitting behind
> NAT, too. Under the Topology tab, it is set to "Discover global address"
> and "Discover server" (for STUN). "Enable ICE" is checked. The next thing
> to check might be if the phone traffic is going directly to the Internet
> and getting past your firewall.
> 
> --
> Russell Mosemann
> 
> 
> 
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From alexagone at gmail.com  Thu Oct 15 12:20:30 2009
From: alexagone at gmail.com (Alexandre Savard)
Date: Thu, 15 Oct 2009 15:20:30 -0400
Subject: [Freeswitch-users] TLS client authentification
Message-ID: <87747c930910151220p474e079bt609019e562698dfb@mail.gmail.com>

Hi,

Does Freeswitch support TLS Client-Authenticated handshake. Openssl does,
but it has to be enabled in order to send the certificate request to the
client.

I tested simple TLS hanshaking and it wotks well.


Thanks.
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From mcampbellsmith at gmail.com  Thu Oct 15 15:19:21 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Fri, 16 Oct 2009 09:19:21 +1100
Subject: [Freeswitch-users] Dingaling / Jingle DTMF support?
Message-ID: <33c87fa30910151519x637618d3p109f1b8edbf3432e@mail.gmail.com>

Hi!

I was wondering if the Dingaling implementation in FS supports DTMF?
This is now supported in the jingle specs (
http://www.jabberforum.org/showthread.php?t=2709 ), even though Google
Talk client does not currently support DTMF.

If not, are there plans to implement this?

Thanks!


From jason at jasonjgw.net  Thu Oct 15 16:22:29 2009
From: jason at jasonjgw.net (Jason White)
Date: Fri, 16 Oct 2009 10:22:29 +1100
Subject: [Freeswitch-users] failed in bridging with SIPp
In-Reply-To: <a38a91d50910151130s1413cbc8yc39c58f8ca6473c7@mail.gmail.com>
References: <a38a91d50910151130s1413cbc8yc39c58f8ca6473c7@mail.gmail.com>
Message-ID: <20091015232229.GA10415@jdc.jasonjgw.net>

Erwin Davis <davis.erwin at gmail.com> wrote:
>     <action application="bridge" data="sofia/internal/profile1/$
> 0 at 192.168.1.36"/>

The above syntax won't work.

data="user/$1 at 192.168.1.36" (assuming that your FreeSWITCH domain is
192.168.1.36) would be better.

If the other end doesn't require authentication (i.e., it isn't registering
with FS) you can use sofia/internal/$1 at domain



From msc at freeswitch.org  Thu Oct 15 16:56:11 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 15 Oct 2009 16:56:11 -0700
Subject: [Freeswitch-users] failed in bridging with SIPp
In-Reply-To: <20091015232229.GA10415@jdc.jasonjgw.net>
References: <a38a91d50910151130s1413cbc8yc39c58f8ca6473c7@mail.gmail.com>
	<20091015232229.GA10415@jdc.jasonjgw.net>
Message-ID: <87f2f3b90910151656tc908855qf1011e84a8075a7b@mail.gmail.com>

On Thu, Oct 15, 2009 at 4:22 PM, Jason White <jason at jasonjgw.net> wrote:

> Erwin Davis <davis.erwin at gmail.com> wrote:
> >     <action application="bridge" data="sofia/internal/profile1/$
> > 0 at 192.168.1.36"/>
>
> The above syntax won't work.
>
> data="user/$1 at 192.168.1.36" (assuming that your FreeSWITCH domain is
> 192.168.1.36) would be better.
>
> If the other end doesn't require authentication (i.e., it isn't registering
> with FS) you can use sofia/internal/$1 at domain
>

Erwin,

Good catch on the sofia syntax. FYI, for those who don't already know, be
sure to check out this wiki page:

http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings

It discusses the various syntaxes for sofia dial strings. Hint: in the
#freeswitch IRC channel you can type "~whatis anatomy" and get this URL.
It's "anatomy" of a SIP dial string.
-MC
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From davis.erwin at gmail.com  Thu Oct 15 17:01:17 2009
From: davis.erwin at gmail.com (Erwin Davis)
Date: Thu, 15 Oct 2009 20:01:17 -0400
Subject: [Freeswitch-users] failed in bridging with SIPp
In-Reply-To: <20091015232229.GA10415@jdc.jasonjgw.net>
References: <a38a91d50910151130s1413cbc8yc39c58f8ca6473c7@mail.gmail.com>
	<20091015232229.GA10415@jdc.jasonjgw.net>
Message-ID: <a38a91d50910151701s4fc8dfd3r513ee41f39e99648@mail.gmail.com>

I prefer the second case. How to define the domain?  Thanks,

e

On Thu, Oct 15, 2009 at 7:22 PM, Jason White <jason at jasonjgw.net> wrote:

> Erwin Davis <davis.erwin at gmail.com> wrote:
> >     <action application="bridge" data="sofia/internal/profile1/$
> > 0 at 192.168.1.36"/>
>
> The above syntax won't work.
>
> data="user/$1 at 192.168.1.36" (assuming that your FreeSWITCH domain is
> 192.168.1.36) would be better.
>
> If the other end doesn't require authentication (i.e., it isn't registering
> with FS) you can use sofia/internal/$1 at domain
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Thu Oct 15 17:10:22 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 15 Oct 2009 17:10:22 -0700
Subject: [Freeswitch-users] Media got stuck after attended transfer...
In-Reply-To: <65d96fc80910151154w2468ebeie06211d0966b4548@mail.gmail.com>
References: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330253@srv02.khdev.corp>
	<191c3a030910150657r668eb5a3q24c641e312d2b113@mail.gmail.com>
	<65d96fc80910151154w2468ebeie06211d0966b4548@mail.gmail.com>
Message-ID: <87f2f3b90910151710k34e4092eg26108dd819d9c041@mail.gmail.com>

On Thu, Oct 15, 2009 at 11:54 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:

> hi, any clue when can t38 be added?
>
>
"Eventually." :)  Of course, if we could get more to add to the bounty it
might grease the wheels of innovation.

http://wiki.freeswitch.org/wiki/Bounty#spanDSP_.2B_t.38_.28origination.2C_termination.2C_.26_gateway.29_in_Freeswitch

Of course, I was listening to my A.M radio the other day and they said that
there was this new invention called the Internet that would let people send
documents to each other electronically. Maybe you should look into that.
Next thing you know they'll come up with telephones that people don't have
to plug into the wall and can take with them in the car. ;)

-MC
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From jason at jasonjgw.net  Thu Oct 15 17:31:51 2009
From: jason at jasonjgw.net (Jason White)
Date: Fri, 16 Oct 2009 11:31:51 +1100
Subject: [Freeswitch-users] failed in bridging with SIPp
In-Reply-To: <a38a91d50910151701s4fc8dfd3r513ee41f39e99648@mail.gmail.com>
References: <a38a91d50910151130s1413cbc8yc39c58f8ca6473c7@mail.gmail.com>
	<20091015232229.GA10415@jdc.jasonjgw.net>
	<a38a91d50910151701s4fc8dfd3r513ee41f39e99648@mail.gmail.com>
Message-ID: <20091016003151.GA26958@jdc.jasonjgw.net>

Erwin Davis <davis.erwin at gmail.com> wrote:
> I prefer the second case. How to define the domain?  Thanks,

It's the domain of FreeSWITCH itself. You could just specify $${domain} if you
aren't using multiple domains.
user/6666@$${domain}

By default, the domain is the IPv4 address of the machine running FreeSWITCH,
which you can change by setting it in vars.xml (assuming that you're using the
default configuration; it will be different if you totally rewrite the
configuration, of course.)



From mike at jerris.com  Thu Oct 15 18:50:40 2009
From: mike at jerris.com (Michael Jerris)
Date: Thu, 15 Oct 2009 21:50:40 -0400
Subject: [Freeswitch-users] Sending an Event to a Session for onInput
In-Reply-To: <4256bf830910112311t409c45a3q2f26cef3464e49e7@mail.gmail.com>
References: <4256bf830910091234t11f4f4d0w28283f341bd1fb06@mail.gmail.com>
	<ACA9A602-0925-4D6F-8B11-0B214A462500@jerris.com>
	<4256bf830910112311t409c45a3q2f26cef3464e49e7@mail.gmail.com>
Message-ID: <0BDCD23D-D551-4A2F-9CD7-CC5A2F1382CB@jerris.com>

It updates the display on a phone if the phone supports this.  This  
works on some sip phones right now including polycom and snom.

Mike

On Oct 12, 2009, at 2:11 AM, Matthew Fong wrote:

> Hi Mike,
>
> I'm just trying to send it an event with some custom event headers,  
> just so an external program can communicate with a session without  
> having to transfer the session to a different program.  I'm curious  
> what uuid_display does...the wiki only gives a brief description and  
> my Google'ing could not find any examples. Thanks for the help.
>
> --matt
> http://www.hellohunter.com
>
> On Mon, Oct 12, 2009 at 5:04 AM, Michael Jerris <mike at jerris.com>  
> wrote:
> We don't have session messages directly exposed, except for things
> like display, respond, and deflect.  What specifically are you trying
> to send ?
>
> Mike
>
> On Oct 9, 2009, at 3:34 PM, Matthew Fong wrote:
>
> > I'm used to using the onInput callbacks inside lua and javascript to
> > listen for dtmf and other events and perform a task accordingly. I'm
> > wondering if there is a way to send an event to a session or channel
> > that can be caught using the setInputCallback inside lua from
> > outside the session program. Maybe an API command that can generate
> > an event for a specific UUID. Does a mechanism exist to do this that
> > I'm over looking? Thanks.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From mcampbellsmith at gmail.com  Thu Oct 15 20:35:20 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Fri, 16 Oct 2009 14:35:20 +1100
Subject: [Freeswitch-users] Dingaling: using a hostname instead of stun
	for rtp
In-Reply-To: <C874E25D-6D73-49AA-8DD3-0A0022617FD2@freeswitch.org>
References: <33c87fa30910131502t2af24801i1c3319518b5db50b@mail.gmail.com>
	<C874E25D-6D73-49AA-8DD3-0A0022617FD2@freeswitch.org>
Message-ID: <33c87fa30910152035w43e260faoc031a7a4ac949d9e@mail.gmail.com>

Thanks Brian.  Is this something that is planned to be implemented?
The workaround is to set the stun server also in the dingaling
configuration, but as I said, for some reason the stun times for me
out occasionally with dingaling.

Thanks!

On Wed, Oct 14, 2009 at 11:33 AM, Brian West <brian at freeswitch.org> wrote:
> I don't think mod_dingaling will do a lookup for host: like sofia will
> as it doesn't have the code for that last I checked... I could be
> wrong but I don't recall it doing that.
>
> /b
>
> On Oct 13, 2009, at 3:02 PM, Mark Campbell-Smith wrote:
>
>> I have a hostname set in vars.conf.xml for the parameters
>> external_rtp_ip and the external_sip_ip instead of the usual stun. ?I
>> found that stun was timing out and was causing some problems. ?And as
>> I have a hostname, it makes sense to use that instead of relying on
>> stun.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mgende at gendesign.com  Thu Oct 15 22:02:16 2009
From: mgende at gendesign.com (Michael Gende)
Date: Fri, 16 Oct 2009 00:02:16 -0500
Subject: [Freeswitch-users] Subject:, Re: FreeSWITCH-users Digest,
	Orien 	L. answer to Mike G.
In-Reply-To: <4AD75F39.6010209@tx.rr.com>
References: <mailman.1290.1255619700.2495.freeswitch-users@lists.freeswitch.org>
	<4AD75F39.6010209@tx.rr.com>
Message-ID: <a87d43fd0910152202k2c7b3300ya139a81385f60bf2@mail.gmail.com>

Hey Orien,

pfsense has a nice interface, but I only use it for adding the FS package,
maybe messing with LAN and WAN addresses. Using the "command line" helped me
to better understand what FS was doing and how. Pfsense just made it easier
to have an OS and really good firewall married to FS without me needing to
do a lot of work. In my pfsense installs, I didn't need to add any firewall
rules to allow SIP ports "right of way", the pfsense folks had taken care of
that detail.


One way to talk "direct" to FS on pfsense is to use the console of the
FreeBSD box you have pfsense installed on. Login as root. You'll get a
character screen with options 1-12 (I believe).

SSHing in gets you the same interface options.

Hit option 8 and you'll go to a shell. I'm assuming you can get around in
Linux or Unix (If not, tell me). "ps aux | grep free" and kill the PID for
FS (pfsense runs FS in the background) Then, invoke FS with
/usr/local/freeswitch/bin/freeswitch.

Now, you're talking direct to FS in the foreground, and it is talking to
you. After starting FS, hit "enter" if there's stuff on the screen - which
there will be - and you'll see the FS prompt. Try the sofia commands.


If you are still configured with:

WAN: 192.168.2.210
LAN: 192.168.3.1
Phones on network 192.168.3.0/24

...then make sure you follow the multi-home wiki instructions in section 4.
I'd include that here, but its better formatted there. I've updated it last
week so have another look if you've not seen it lately.

http://wiki.freeswitch.org/wiki/Multi_home_tutorial

Also, make sure you understand section 3, the FS COMMANDS FOR STARTERS and
WHERE STUFF IS FOR STARTERS. It'll help.

Following those instructions, unless there's some other issue, will allow
for phones on the LAN to talk to one another. Its a start.

If it does not, we'll take it from there.

Regards,

Mike G.






On Thu, Oct 15, 2009 at 12:43 PM, Orien Love <orien at tx.rr.com> wrote:

> Mike,
>    I copied the status screen from the freeswitch package in the web
> setup. I you need a console view, I will need instructions on how to
> console into pfsense, Since I have never done that.
>
> Orien L.
>
>                     Name       Type
> Data    State
>
> =================================================================================================
>                 internal    profile
> sip:mod_sofia at 192.168.10.1:5060    RUNNING (0)
>                 external    profile
> sip:mod_sofia at 192.168.2.210:5080    RUNNING (0)
>             192.168.10.1      alias
> internal    ALIASED
>            internal-ipv6    profile
> sip:mod_sofia@[::1]:5060    RUNNING (0)
>
> =================================================================================================
> 3 profiles 1 alias
>
>
>
> sofia status profile external
>
>
> =================================================================================================
> Name                 external
> Domain Name          N/A
> DBName               sofia_reg_external
> Pres Hosts
> Dialplan             XML
> Context              public
> Challenge Realm      auto_to
> RTP-IP               192.168.2.210
> Ext-RTP-IP           192.168.2.210
> SIP-IP               192.168.2.210
> Ext-SIP-IP           192.168.2.210
> URL                  sip:mod_sofia at 192.168.2.210:5080
> BIND-URL             sip:mod_sofia at 192.168.2.210:5080
> HOLD-MUSIC           local_stream://moh
> OUTBOUND-PROXY       N/A
> CODECS               PCMU,PCMA,GSM
> TEL-EVENT            101
> DTMF-MODE            rfc2833
> CNG                  13
> SESSION-TO           0
> MAX-DIALOG           0
> NOMEDIA              false
> LATE-NEG             false
> PROXY-MEDIA          false
> AGGRESSIVENAT        false
> STUN-ENABLED         true
> STUN-AUTO-DISABLE    false
> CALLS-IN             0
> FAILED-CALLS-IN      0
> CALLS-OUT            0
> FAILED-CALLS-OUT     0
>
> Registrations:
>
> =================================================================================================
>
> =================================================================================================
>
>
>
> sofia status profile internal
>
>
> =================================================================================================
> Name                 internal
> Domain Name          N/A
> DBName               sofia_reg_internal
> Pres Hosts
> Dialplan             XML
> Context              public
> Challenge Realm      auto_from
> RTP-IP               192.168.10.1
> Ext-RTP-IP           192.168.10.1
> SIP-IP               192.168.10.1
> Ext-SIP-IP           192.168.10.1
> URL                  sip:mod_sofia at 192.168.10.1:5060
> BIND-URL             sip:mod_sofia at 192.168.10.1:5060
> HOLD-MUSIC           local_stream://moh
> OUTBOUND-PROXY       N/A
> CODECS               G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
> TEL-EVENT            101
> DTMF-MODE            rfc2833
> CNG                  13
> SESSION-TO           0
> MAX-DIALOG           0
> NOMEDIA              false
> LATE-NEG             false
> PROXY-MEDIA          false
> AGGRESSIVENAT        false
> STUN-ENABLED         true
> STUN-AUTO-DISABLE    false
> CALLS-IN             6
> FAILED-CALLS-IN      3
> CALLS-OUT            0
> FAILED-CALLS-OUT     0
>
> Registrations:
>
> =================================================================================================
> Call-ID:        NjczMzQzNTVlYTJhYjIzYmFiZmFjZGYzNzNlMTVkMTc.
> User:           1001 at 192.168.10.1
> Contact:        "user"
> Agent:          Pangolin v5.2, Build 09182009
> Status:         Registered(UDP)(unknown) EXP(2009-10-15 05:03:17)
> Host:           orienhome2.local
> IP:             192.168.10.243
> Port:           3789
> Auth-User:      1001
> Auth-Realm:     192.168.10.1
>
>
> =================================================================================================
>
>
>
> sofia status profile internal-ipv6
>
>
> =================================================================================================
> Name                 internal-ipv6
> Domain Name          N/A
> DBName               sofia_reg_internal-ipv6
> Pres Hosts
> Dialplan             XML
> Context              public
> Challenge Realm      auto_to
> RTP-IP               ::1
> SIP-IP               ::1
> URL                  sip:mod_sofia@[::1]:5060
> BIND-URL             sip:mod_sofia@[::1]:5060
> HOLD-MUSIC           local_stream://moh
> OUTBOUND-PROXY       N/A
> CODECS               G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
> TEL-EVENT            101
> DTMF-MODE            rfc2833
> CNG                  13
> SESSION-TO           0
> MAX-DIALOG           0
> NOMEDIA              false
> LATE-NEG             false
> PROXY-MEDIA          false
> AGGRESSIVENAT        false
> STUN-ENABLED         true
> STUN-AUTO-DISABLE    false
> CALLS-IN             0
> FAILED-CALLS-IN      0
> CALLS-OUT            0
> FAILED-CALLS-OUT     0
>
> Registrations:
>
> =================================================================================================
>
> =================================================================================================
>
>
>
> status
>
> UP 0 years, 0 days, 6 hours, 55 minutes, 32 seconds, 640 milliseconds,
> 638 microseconds
> 43 session(s) since startup
> 0 session(s) 0/30
> 1000 session(s) max
>
>
> > Subject:
> > Re: [Freeswitch-users] FreeSWITCH-users Digest, Mike G. answer to Orien L
> > From:
> > Michael Gende <mgende at gendesign.com>
> > Date:
> > Thu, 15 Oct 2009 09:36:55 -0500
> > To:
> > freeswitch-users at lists.freeswitch.org
> >
> > To:
> > freeswitch-users at lists.freeswitch.org
> >
> >
> > Hey Orien,
> >
> > OK, let's have a look here. How about going to the console of your FS
> > and posting the output of the following commands:
> >
> > sofia status profile internal
> > sofia status profile external
> >
> > We'll take it from there. Your problems are just like what I ran into
> > initially, so I'm hopeful we get round them.
> >
> > Mike G.
> >
> > On Wed, Oct 14, 2009 at 11:34 PM, Michael Gende <mgende at gendesign.com
> > <mailto:mgende at gendesign.com>> wrote:
> >
> >     I'll get to you in the morning (where do you live?) and we'll see
> >     what's what.
> >
> >
> >
> >     On Wed, Oct 14, 2009 at 10:30 PM, Orien Love <orien at tx.rr.com
> >     <mailto:orien at tx.rr.com>> wrote:
> >
> >         Mike,
> >             Thank you for your quick response, this has been driving
> >         me nuts.
> >
> >         in my third attempt I tried to follow the mulit home example.
> >         I did set
> >         the phones to register to the local lan IP. actually had them
> >         registering, but it did help it in any way still the same
> problem.
> >         After my post I reinstalled the package and set the firewall
> >         to allow
> >         phones on the wan to connect, as per
> >         http://doc.pfsense.org/index.php/FreeSWITCH#Firewall
> >         put my computer with the soft phone on the wan network
> >         (192,168.2.xxx)
> >         had it registering and it had the same problem. so far all the
> >         setups I
> >         have tried are not working, the phones register, but cannot
> >         dial/connect  another  phone, also does not go to voice mail,
> >         maybe it
> >         is in my dial plan. which is default.
> >
> >         Shoot, I am not picky, I just want to get it to work. so any
> >         suggestions
> >         on how to accomplish this are greatly appreciated.
> >
> >         My phones are currently a couple of polycom 601's, and a
> >         Pangolin Softphone.
> >
> >         Thanks
> >           Orien L.
> >
> >
> >
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From rogelio.perez at gmail.com  Thu Oct 15 23:44:23 2009
From: rogelio.perez at gmail.com (Rogelio Perez)
Date: Fri, 16 Oct 2009 03:44:23 -0300
Subject: [Freeswitch-users] SIP CANCEL error
Message-ID: <E1FE097E-599C-479E-9D1C-2308F213109C@gmail.com>

Hi guys,

I'm having a signaling issue on FS when calling from an endpoint to a  
GW.
To reproduce the error I just call and hungup the call right away.
FS gets the CANCEL and replies back with 481 Call/Transaction Does Not  
Exist, so the call continues and eventually connects, even when the A  
side of the call is closed.
I have verified that the Call-ID of the CANCEL matches that from the  
INVITE, so I see no reasons for FS to reject it.

SIP+FS logs in pastebin 10738.

Any ideas?

Thanks,
Rogelio





From codecomplete at free.fr  Fri Oct 16 00:11:16 2009
From: codecomplete at free.fr (Fred-145)
Date: Fri, 16 Oct 2009 00:11:16 -0700 (PDT)
Subject: [Freeswitch-users]  PBX vs. SIP proxy?
Message-ID: <25920788.post@talk.nabble.com>


Hello

I was reading an article the other day about Freeswitch in which the author
used the terms "SIP proxy" and PBX. Is an SIP proxy just a way for SIP
clients to register and connect to each other, while the PBX part adds
services like voice mail, etc.?

Thank you.
-- 
View this message in context: http://www.nabble.com/PBX-vs.-SIP-proxy--tp25920788p25920788.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From Russell.Mosemann at cune.org  Fri Oct 16 00:47:31 2009
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Fri, 16 Oct 2009 02:47:31 -0500
Subject: [Freeswitch-users] Subject:, Re: FreeSWITCH-users Digest,
	Orien 	L. answer to Mike G.
In-Reply-To: <a87d43fd0910152202k2c7b3300ya139a81385f60bf2@mail.gmail.com>
References: <mailman.1290.1255619700.2495.freeswitch-users@lists.freeswitch.org><4AD75F39.6010209@tx.rr.com>
	<a87d43fd0910152202k2c7b3300ya139a81385f60bf2@mail.gmail.com>
Message-ID: <C8C9B2D0FA5E40C4AA2658B34D0092DD@cune.pri>

Michael Gende wrote:

> Hit option 8 and you'll go to a shell. I'm assuming you can get around in
> Linux or Unix (If not, tell me). "ps aux | grep free" and kill the PID for
> FS (pfsense runs FS in the background) Then, invoke FS with
> /usr/local/freeswitch/bin/freeswitch.

You've mentioned this twice, now, but a better way to do it is to _not_ stop FreeSwitch. Instead, run /usr/local/freeswitch/bin/fs_cli, which connects to FS, and then you can do everything that you can do directly in FS. When you are done, enter "\quit" or "\bye" or "\exit". See "\help" and "help" for more information.

--
Russell Mosemann



From Russell.Mosemann at cune.org  Fri Oct 16 00:54:18 2009
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Fri, 16 Oct 2009 02:54:18 -0500
Subject: [Freeswitch-users] Subject:, Re: FreeSWITCH-users Digest,
	Orien 	L. answer to Mike G.
In-Reply-To: <C8C9B2D0FA5E40C4AA2658B34D0092DD@cune.pri>
References: <mailman.1290.1255619700.2495.freeswitch-users@lists.freeswitch.org><4AD75F39.6010209@tx.rr.com><a87d43fd0910152202k2c7b3300ya139a81385f60bf2@mail.gmail.com>
	<C8C9B2D0FA5E40C4AA2658B34D0092DD@cune.pri>
Message-ID: <D328986AA10A4B0796265D08C355EFFC@cune.pri>

> When you are done, enter "\quit" or "\bye" or "\exit". See "\help" and
> "help" for more information.

Yup, I got the slashes wrong, and I was staring right at the commands when I wrote the message. I'll blame it on how late it is here. :-) It should be "/quit", "/bye", "/exit" and "/help".

--
Russell Mosemann



From jason at jasonjgw.net  Fri Oct 16 01:06:08 2009
From: jason at jasonjgw.net (Jason White)
Date: Fri, 16 Oct 2009 19:06:08 +1100
Subject: [Freeswitch-users] Subject:, Re: FreeSWITCH-users Digest,
 Orien 	L. answer to Mike G.
In-Reply-To: <D328986AA10A4B0796265D08C355EFFC@cune.pri>
References: <mailman.1290.1255619700.2495.freeswitch-users@lists.freeswitch.org>
	<4AD75F39.6010209@tx.rr.com>
	<a87d43fd0910152202k2c7b3300ya139a81385f60bf2@mail.gmail.com>
	<C8C9B2D0FA5E40C4AA2658B34D0092DD@cune.pri>
	<D328986AA10A4B0796265D08C355EFFC@cune.pri>
Message-ID: <20091016080608.GA21211@jdc.jasonjgw.net>

Russell Mosemann <Russell.Mosemann at cune.org> wrote:
> 
> Yup, I got the slashes wrong, and I was staring right at the commands when I
> wrote the message. I'll blame it on how late it is here. :-) It should be
> "/quit", "/bye", "/exit" and "/help".

ctrl-d is also recognized and it's easier to type.



From helmut.kuper at ewetel.de  Fri Oct 16 04:31:48 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Fri, 16 Oct 2009 13:31:48 +0200
Subject: [Freeswitch-users] Qustion about INFO messages after Connect/Answer
Message-ID: <4AD859A4.5000101@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

after updating FS to trunk a few days ago I found that callee's display
is updated serveral times to caller's name after callee picked up. The
first two equal INFO messages looks like this:

INFO sip:2850 at 85.16.245.213:1040;line=367hfn9i SIP/2.0
Via: SIP/2.0/UDP 85.16.246.6;rport;branch=z9hG4bKU4803UHXaKD2r
Max-Forwards: 70
From: "Snom1 an PBX" <sip:4918 at 85.16.246.6>;tag=ccc4B2F9SD2rm
To: <sip:2850 at 85.16.245.213:1040;line=367hfn9i>;tag=zbrirvg9ow
Call-ID: 97a0e683-34e6-122d-8db2-00144fe6e330
CSeq: 121727215 INFO
Contact: <sip:mod_sofia at 85.16.246.6:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15137M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Type: message/sipfrag
Content-Length: 27

From:
To: "unknown" 2850


The third one is this:

INFO sip:2850 at 85.16.245.213:1040;line=367hfn9i SIP/2.0
Via: SIP/2.0/UDP 85.16.246.6;rport;branch=z9hG4bK0H7vc35evZvZj
Max-Forwards: 70
From: "Snom1 an PBX" <sip:4918 at 85.16.246.6>;tag=ccc4B2F9SD2rm
To: <sip:2850 at 85.16.245.213:1040;line=367hfn9i>;tag=zbrirvg9ow
Call-ID: 97a0e683-34e6-122d-8db2-00144fe6e330
CSeq: 121727216 INFO
Contact: <sip:mod_sofia at 85.16.246.6:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15137M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Type: message/sipfrag
Content-Length: 32

From:
To: "Snom1 an PBX" 4918


What calle sees is this:
1. Phone rings: caller's displayname
2. Callee picks up: switching from dislayname to unknown
3. Switching from unknown to displayname

After callee has picked up caller's display is also updated with
callee's name (this is a copy of callee's number rather than callee's
name) and number.

First Question:
Can this behaviour be disabled? Or can it be modified by dialplan, so
that there is no "unknown"?




Unfortunately when you do an outgoing call via openzap and callee picks
up the phone caller's display is updated with two equal INFO message
like this:

INFO sip:2850 at 85.16.245.213:1040;line=367hfn9i SIP/2.0
Via: SIP/2.0/UDP 85.16.246.6;rport;branch=z9hG4bK7jFa9Ztg39UUr
Max-Forwards: 70
From: <sip:890327 at 85.16.246.6;user=phone>;tag=9QZamXtmUyXvm
To: "Helmut Kuper" <sip:2850 at 85.16.246.6>;tag=5lf2e0q1on
Call-ID: 3c2a2539a966-0m3i9s85my46
CSeq: 121727707 INFO
Contact: <sip:890327 at 85.16.246.6:5060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15137M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Type: message/sipfrag
Content-Length: 36

From:
To: "1/a/890327" 1/a/890327


Second Question:
Can I change the name and number via dialplan, so that the correkt name
and number is viewed to caller?


regards
helmut
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VSQFKUCWpsxVm3evyfWQT/Y=
=FGsp
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From mike at jerris.com  Fri Oct 16 06:27:21 2009
From: mike at jerris.com (Michael Jerris)
Date: Fri, 16 Oct 2009 09:27:21 -0400
Subject: [Freeswitch-users] sched_api doesn't get launched
In-Reply-To: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
References: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
Message-ID: <3F1747F9-C543-4C6F-BB01-4DD5FA5B7BC5@jerris.com>

sched_api is a fsapi command not a dialplan application, I believe  
sched_hangup is both.

Mike


On Oct 13, 2009, at 6:14 AM, Henry Huang wrote:

> Hi:
>
> I am using mod_java. And in my script I was able to achieve using:
>
> execute("sched_hangup", "+300 alloted_timeout");
>
> However, when I try to run sched_api in the same way, system log  
> returns that it's an invalid application. I have also tried to  
> trigger it with many conditional channel variable api calls , but  
> non of them seemed to execute the api command (because I turned on  
> the highest level of debugging and see no where the sched_api is  
> being called.
>
> The closest thing I got was by using "api_after_bridge" like the  
> following, but it only launches when the bridge is teared down(which  
> is not what I want). I originally thought after bridge means right  
> after the 2 party is connected. All I want is to be able to play  
> some message to leg A at certain time.
>
> setVariable("api_after_bridge", "sched_api +10 none uuid_displace $ 
> {uuid} start /path/to/some.wav 20 mux");
>
> I have been struggling with different combination for a week now..  
> Please shed some light if you know something.
>
> Thanks,
>
> -- 
> Henry Huang
> UniC Solution - Communication Unified
> VoIP & Open Source software Consultant
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From mike at jerris.com  Fri Oct 16 06:29:39 2009
From: mike at jerris.com (Michael Jerris)
Date: Fri, 16 Oct 2009 09:29:39 -0400
Subject: [Freeswitch-users] Building Freeswitch with VPATH (src and obj
	directories are different)?
In-Reply-To: <m3eip65vso.fsf@mad06.wl0.org>
References: <20091013205907.GA7545@mad06.wl0.org>
	<D8EB1A6C-38A6-4BC2-AD8C-46CF11993366@freeswitch.org>
	<m3eip65vso.fsf@mad06.wl0.org>
Message-ID: <8C5898A4-E257-4FBE-A8EC-439402965FDB@jerris.com>

I would love to see this work in tree, but i am pretty sure it has  
never worked.  I would gladly accept patches that implement this.


Mike

On Oct 14, 2009, at 2:33 AM, Simon J Mudd wrote:

> brian at freeswitch.org (Brian West) writes:
>
>> You shouldn't have to make clean usually ... doing so might break  
>> your
>> tree...
>
> Why?
>
> In any case on my mac (Leopard) neither make clean or distclean fully
> cleans up afterwards.  I would prefer to get a completely "untouched"
> source tree after doing this.  If that's supposed to work and I should
> post a bug or report the problem in more detail then I can do that.
>
>> You can usually get by with "make current" that will ensure
>> the critical things are cleaned and built correctly... every now and
>> they you'll hit a snafu but we'll usually tell you about it.
>
> ok. That's one of the reasons for liking to use VPATH builds. It's  
> also
> why many BSD system builds use this so that you can mount the source
> tree readonly (perhaps from another server) and build the binaries
> in a separate directory independently.
>
> If that's not supported then that's fine.
>
> I was trying to build the latest svn version of freeswitch (Revision:
> 15138) and this didn't build. As I'm not too familiar with compiling
> at least on the Mac I thought I'd first try to see if a VPATH build
> would work better. I'll have to try and investigate further.



From mike at jerris.com  Fri Oct 16 06:33:59 2009
From: mike at jerris.com (Michael Jerris)
Date: Fri, 16 Oct 2009 09:33:59 -0400
Subject: [Freeswitch-users] T.38 via UPDATE request
In-Reply-To: <4AD5AE3E.9070206@gmx.net>
References: <4AD5AE3E.9070206@gmx.net>
Message-ID: <0C137031-4C23-4A70-A05F-DE038D005839@jerris.com>

There was just a bunch of work on UPDATE, can you confirm this is the  
same behavior with trunk?

On Oct 14, 2009, at 6:55 AM, Peter P GMX wrote:

> Hello,
>
> we have the following problem.
> 2 Fax machines are communicating via Freeswitch. One is externally
> attached via a Telco who is able to handle T.38. The other one is
> attached locally.
>
> When 2 Fax machines start syncing each other, the Telco sends a SIP
> UPDATE message with T.38 SDP, as it detects fax during the fax  
> negociations.
> Freeswitch answers with an SIP OK message back to the telco, and I can
> see the T.38 SDP on the debug console of freeswitch.
> Then nothing happens any more until one of fax machines detects  
> timeout.
>
> We have set proxy-media to true. However is was done during call setup
> when both machines communicated with G711 SDP.
> The UPDATE message was commited by FS to the telco, but was not sent  
> to
> the other fax, so I think in this case Freeswitch is supposed to
> transcode between T.38 and G711 which it cannot do, as we know.
>
> How can I overcome this scenario? Is this a defect, should freeswitch
> send the UPDATE message to the other fax?  Or is there a workaround?
>
> Best regards
> Peter



From mike at jerris.com  Fri Oct 16 06:36:29 2009
From: mike at jerris.com (Michael Jerris)
Date: Fri, 16 Oct 2009 09:36:29 -0400
Subject: [Freeswitch-users] TLS Audiocodes
In-Reply-To: <4a4d27d30910132316g206580f0ya32287291af00c5b@mail.gmail.com>
References: <4a4d27d30910132316g206580f0ya32287291af00c5b@mail.gmail.com>
Message-ID: <E82402B2-857E-45EC-AC6A-0FF18983F9CE@jerris.com>

Try turning up all the sofia debug to 9.

Mike

On Oct 14, 2009, at 2:16 AM, Szasz Szabolcs wrote:

> Hi,
>
> Did anybody set up TLS between Freeswitch and Audiodes MP11X ? I got  
> to work TLS between freeswitch  and a softphone (phonerlite), but I  
> have problem with Audiocodes during the TLS authentication. I've  
> loaded the certification but it still doesn't work. Can I debug the  
> tls in freeswitch? Please help me.
>


From pedroprado at msn.com  Fri Oct 16 06:48:48 2009
From: pedroprado at msn.com (Pedro Prado)
Date: Fri, 16 Oct 2009 10:48:48 -0300
Subject: [Freeswitch-users] Brazilians (Off-Topic)
Message-ID: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>


Hi,

Do you have a group of Brazilians here?

Thanks,
Pedro Prado
 		 	   		  
_________________________________________________________________
Voc? sabia que com o Hotmail voc? tem espa?o ilimitado para guardar seus e-mails? Come?e a usar j?!
http://www.microsoft.com/brasil/windows/windowslive/products/hotmail.aspx
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From mgende at gendesign.com  Fri Oct 16 07:02:28 2009
From: mgende at gendesign.com (Michael Gende)
Date: Fri, 16 Oct 2009 09:02:28 -0500
Subject: [Freeswitch-users] Subject:, Re: FreeSWITCH-users Digest,
	Orien 	L. answer to Mike G.
In-Reply-To: <C8C9B2D0FA5E40C4AA2658B34D0092DD@cune.pri>
References: <mailman.1290.1255619700.2495.freeswitch-users@lists.freeswitch.org>
	<4AD75F39.6010209@tx.rr.com>
	<a87d43fd0910152202k2c7b3300ya139a81385f60bf2@mail.gmail.com>
	<C8C9B2D0FA5E40C4AA2658B34D0092DD@cune.pri>
Message-ID: <a87d43fd0910160702y5ab70c29w3bb22a903ddc59b7@mail.gmail.com>

I didn't know that. Thanks Russell.


>
>
>
> You've mentioned this twice, now, but a better way to do it is to _not_
> stop FreeSwitch. Instead, run /usr/local/freeswitch/bin/fs_cli, which
> connects to FS, and then you can do everything that you can do directly in
> FS. When you are done, enter "\quit" or "\bye" or "\exit". See "\help" and
> "help" for more information.
>
> --
>
> Russell Mosemann
>
>
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From itamar at ispbrasil.com.br  Fri Oct 16 07:02:28 2009
From: itamar at ispbrasil.com.br (Itamar Reis Peixoto)
Date: Fri, 16 Oct 2009 11:02:28 -0300
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
Message-ID: <d3ceea380910160702x8acbc4ci3bf3cfc1fdf9a9b7@mail.gmail.com>

I think no.


On Fri, Oct 16, 2009 at 10:48 AM, Pedro Prado <pedroprado at msn.com> wrote:
> Hi,
>
> Do you have a group of Brazilians here?
>
> Thanks,
> Pedro Prado



-- 
------------

Itamar Reis Peixoto

e-mail/msn: itamar at ispbrasil.com.br
sip: itamar at ispbrasil.com.br
skype: itamarjp
icq: 81053601
+55 11 4063 5033
+55 34 3221 8599


From jmesquita at freeswitch.org  Fri Oct 16 07:25:25 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 16 Oct 2009 11:25:25 -0300
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <d3ceea380910160702x8acbc4ci3bf3cfc1fdf9a9b7@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<d3ceea380910160702x8acbc4ci3bf3cfc1fdf9a9b7@mail.gmail.com>
Message-ID: <ebbc2d550910160725x3141ce14re1821926d4027d42@mail.gmail.com>

We do. :-)

Jo?o Mesquita

On Fri, Oct 16, 2009 at 11:02 AM, Itamar Reis Peixoto <
itamar at ispbrasil.com.br> wrote:

> I think no.
>
>
> On Fri, Oct 16, 2009 at 10:48 AM, Pedro Prado <pedroprado at msn.com> wrote:
> > Hi,
> >
> > Do you have a group of Brazilians here?
> >
> > Thanks,
> > Pedro Prado
>
>
>
> --
> ------------
>
> Itamar Reis Peixoto
>
> e-mail/msn: itamar at ispbrasil.com.br
> sip: itamar at ispbrasil.com.br
> skype: itamarjp
> icq: 81053601
> +55 11 4063 5033
> +55 34 3221 8599
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Prometheus001 at gmx.net  Fri Oct 16 07:35:22 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 16 Oct 2009 16:35:22 +0200
Subject: [Freeswitch-users] T.38 via UPDATE request
In-Reply-To: <0C137031-4C23-4A70-A05F-DE038D005839@jerris.com>
References: <4AD5AE3E.9070206@gmx.net>
	<0C137031-4C23-4A70-A05F-DE038D005839@jerris.com>
Message-ID: <4AD884AA.1050805@gmx.net>

Hello Michael

this is a productive system, so I can currently not update to the
current trunk.
But the installed SVN is 14741. Were those changes after 14741? Then
I'll try to find a timeslot at night in order to update freeswitch.

But Freeswitch should forward the UPDATE request in proxy-media mode, right?

Best regards
Peter

Michael Jerris schrieb:
> There was just a bunch of work on UPDATE, can you confirm this is the  
> same behavior with trunk?
>
> On Oct 14, 2009, at 6:55 AM, Peter P GMX wrote:
>
>   
>> Hello,
>>
>> we have the following problem.
>> 2 Fax machines are communicating via Freeswitch. One is externally
>> attached via a Telco who is able to handle T.38. The other one is
>> attached locally.
>>
>> When 2 Fax machines start syncing each other, the Telco sends a SIP
>> UPDATE message with T.38 SDP, as it detects fax during the fax  
>> negociations.
>> Freeswitch answers with an SIP OK message back to the telco, and I can
>> see the T.38 SDP on the debug console of freeswitch.
>> Then nothing happens any more until one of fax machines detects  
>> timeout.
>>
>> We have set proxy-media to true. However is was done during call setup
>> when both machines communicated with G711 SDP.
>> The UPDATE message was commited by FS to the telco, but was not sent  
>> to
>> the other fax, so I think in this case Freeswitch is supposed to
>> transcode between T.38 and G711 which it cannot do, as we know.
>>
>> How can I overcome this scenario? Is this a defect, should freeswitch
>> send the UPDATE message to the other fax?  Or is there a workaround?
>>
>> Best regards
>> Peter
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   


From brian at freeswitch.org  Fri Oct 16 07:48:37 2009
From: brian at freeswitch.org (Brian West)
Date: Fri, 16 Oct 2009 09:48:37 -0500
Subject: [Freeswitch-users] Dingaling: using a hostname instead of stun
	for rtp
In-Reply-To: <33c87fa30910152035w43e260faoc031a7a4ac949d9e@mail.gmail.com>
References: <33c87fa30910131502t2af24801i1c3319518b5db50b@mail.gmail.com>
	<C874E25D-6D73-49AA-8DD3-0A0022617FD2@freeswitch.org>
	<33c87fa30910152035w43e260faoc031a7a4ac949d9e@mail.gmail.com>
Message-ID: <65BF398D-E3A6-4A98-83D9-D4E271070CCA@freeswitch.org>

If you setup your own stun server it wouldn't do that....  But the  
hostlookup only solves half the problem .. getting the external IP vs  
poking holes for RTP which is what stun will do.

/b

On Oct 15, 2009, at 10:35 PM, Mark Campbell-Smith wrote:

> Thanks Brian.  Is this something that is planned to be implemented?
> The workaround is to set the stun server also in the dingaling
> configuration, but as I said, for some reason the stun times for me
> out occasionally with dingaling.
>
> Thanks!



From nbelov at abisoft.spb.ru  Fri Oct 16 08:54:47 2009
From: nbelov at abisoft.spb.ru (Nikita Belov)
Date: Fri, 16 Oct 2009 19:54:47 +0400
Subject: [Freeswitch-users] uuid_send_dtmf fails (was: conference call)
In-Reply-To: <b39273530910150504l5c7411b9g39b9ed33c592e85f@mail.gmail.com>
References: <064A5201C10D4D20AC76EB93CA5EA4B5@abisoft.biz><ebbc2d550910140838v3125579ev9fa16e3e83d25770@mail.gmail.com><BE4EDDA84CB64AA98F18D412B51912FC@abisoft.biz>
	<b39273530910150504l5c7411b9g39b9ed33c592e85f@mail.gmail.com>
Message-ID: <02C2EFD4CF8D4ACF8EEC0F77C4D75028@abisoft.biz>

Yes, it is what I need. But now I have problem with sending dtmf. Here what
I've done:
[root at centos4-4-vm ~]# telnet localhost 8021
Trying 127.0.0.1...
Connected to localhost.localdomain (127.0.0.1).
Escape character is '^]'.
Content-Type: auth/request

auth ClueCon

Content-Type: command/reply
Reply-Text: +OK accepted

api originate user/1007 at master.agent.starpoundtech.net &park()

Content-Type: api/response
Content-Length: 41

+OK bba3b45a-4cc1-48af-a15d-1052d5f11371

SendMsg bba3b45a-4cc1-48af-a15d-1052d5f11371
call-command: execute
execute-app-name: eavesdrop
execute-app-arg: cd99f999-9b47-457e-8439-1d366e015b8c

Content-Type: command/reply
Reply-Text: +OK

Here I had started to hear A and B. Here what I saw in FS log:
2009-10-18 03:22:47 [DEBUG] switch_core_session.c:706
switch_core_session_queue_private_event() Send signal
sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes [BREAK]
2009-10-18 03:22:47 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event()
sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes Command Execute
eavesdrop(cd99f999-9b47-457e-8439-1d366e015b8c)
2009-10-18 03:22:47 [DEBUG] switch_core_media_bug.c:297
switch_core_media_bug_add() Attaching BUG to
sofia/internal/1006 at master.agent.starpoundtech.net
2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234
switch_core_session_read_frame()
sofia/internal/1006 at master.agent.starpoundtech.net receive message
[TRANSCODING_NECESSARY]
2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234
switch_core_session_read_frame()
sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes receive message
[TRANSCODING_NECESSARY]

Then I run command:
api uuid_send_dtmf bba3b45a-4cc1-48af-a15d-1052d5f11371 1

Content-Type: api/response
Content-Length: 14

-ERR no reply

Log:
2009-10-18 03:24:01 [DEBUG] switch_core_io.c:1190
switch_core_session_send_dtmf_string()
sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes send dtmf
digit=1 ms=250 samples=2000
2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1282 do_2833() Send start packet
for [1] ts=2241760 dur=160/160/2000 seq=21346
2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=320/320/2000 seq=21347
2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=480/480/2000 seq=21348
2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=640/640/2000 seq=21349
2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=800/800/2000 seq=21350
2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=960/960/2000 seq=21351
2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=1120/1120/2000 seq=21352
2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=1280/1280/2000 seq=21353
2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=1440/1440/2000 seq=21354
2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=1600/1600/2000 seq=21355
2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=1760/1760/2000 seq=21356
2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=1920/1920/2000 seq=21357
2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end packet for
[1] ts=2241760 dur=2080/2080/2000 seq=21358
2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end packet for
[1] ts=2241760 dur=2080/2080/2000 seq=21359
2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end packet for
[1] ts=2241760 dur=2080/2080/2000 seq=21360

But both A and B couldn't hear me. 
Btw, after I had send dtmf "1" manually from my phone. B started to hear me.
There was this record in log:
2009-10-18 03:47:55 [DEBUG] switch_rtp.c:1767 switch_rtp_dequeue_dtmf() RTP
RECV DTMF 1:2240

Does anybody know, what had I done wrong?

___

Thanks, Nikita
    
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker
> Sent: Thursday, October 15, 2009 4:04 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] conference call
> 
> The wiki has a pretty good explanation of how eavesdrop works.
> Enabling a talk path to A or B or both A and B requires dtmf.
> 
> So, if C hits the 1 button on the phone they can talk to the UUID you
> bound the eavesdrop to.   2 talks to the other party.  3 talks to both
> parties.
> 
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop
> 
> Adding more api calls to control eavesdrop via event socket should be
> possible, though I haven't looked at the code to see what it would
> take...
> 
> On Thu, Oct 15, 2009 at 3:16 AM, Nikita Belov <nbelov at abisoft.spb.ru>
> wrote:
> > Thanks for answer.
> >
> >
> >
> > As I've understood eavesdrop allows C to hear A and B, but A and B will
> not
> > hear C. But in my case B SHOULD hear C. In my case A - client, B - agent
> > worked with client and C - coach/supervisor for B. Can I make such
> > conference in FS using eavesdrop or other API command?
> >
> >
> >
> > ___
> >
> >
> >
> > Thanks, Nikita
> >
> >
> >
> > ________________________________
> >
> > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Joao
> > Mesquita
> > Sent: Wednesday, October 14, 2009 7:38 PM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] conference call
> >
> >
> >
> > Look at eavesdrop on the wiki.
> >
> > JM
> >
> > 2009/10/14 Nikita Belov <nbelov at abisoft.spb.ru>
> >
> > HI all,
> >
> >
> >
> > I want to configure FS to make special conference call between three
> users
> > (A, B, C).? In this conference C will hear A and B, but A will hear only
> B.
> > Can I make it using FS API commands? Does anybody know which approach is
> > better to use?
> >
> >
> >
> > ___
> >
> >
> >
> > Thanks, Nikita
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> 
> 
> --
> -Rupa
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From msc at freeswitch.org  Fri Oct 16 08:57:04 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 16 Oct 2009 08:57:04 -0700
Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Starting
	Shortly, Please Start 	Calling In
Message-ID: <87f2f3b90910160857v36acb772s324389b0debb489c@mail.gmail.com>

The weekly conference call is about to start. Please call in. The agenda and
calling instructions are here:

http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_16

Looking forward to hearing from everyone!
-Michael
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From anthony.minessale at gmail.com  Fri Oct 16 09:18:12 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 16 Oct 2009 11:18:12 -0500
Subject: [Freeswitch-users] Qustion about INFO messages after
	Connect/Answer
In-Reply-To: <4AD859A4.5000101@ewetel.de>
References: <4AD859A4.5000101@ewetel.de>
Message-ID: <191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>

1) you should update again there were a few issues.
2) you can set the variable sip_callee_id_name and sip_callee_id number on
the inbound leg before you answer to control what it says.


On Fri, Oct 16, 2009 at 6:31 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> after updating FS to trunk a few days ago I found that callee's display
> is updated serveral times to caller's name after callee picked up. The
> first two equal INFO messages looks like this:
>
> INFO sip:2850 at 85.16.245.213:1040;line=367hfn9i SIP/2.0
> Via: SIP/2.0/UDP 85.16.246.6;rport;branch=z9hG4bKU4803UHXaKD2r
> Max-Forwards: 70
> From: "Snom1 an PBX" <sip:4918 at 85.16.246.6 <sip%3A4918 at 85.16.246.6>
> >;tag=ccc4B2F9SD2rm
> To: <sip:2850 at 85.16.245.213:1040;line=367hfn9i>;tag=zbrirvg9ow
> Call-ID: 97a0e683-34e6-122d-8db2-00144fe6e330
> CSeq: 121727215 INFO
> Contact: <sip:mod_sofia at 85.16.246.6:5060>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15137M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
> REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Content-Type: message/sipfrag
> Content-Length: 27
>
> From:
> To: "unknown" 2850
>
>
> The third one is this:
>
> INFO sip:2850 at 85.16.245.213:1040;line=367hfn9i SIP/2.0
> Via: SIP/2.0/UDP 85.16.246.6;rport;branch=z9hG4bK0H7vc35evZvZj
> Max-Forwards: 70
> From: "Snom1 an PBX" <sip:4918 at 85.16.246.6 <sip%3A4918 at 85.16.246.6>
> >;tag=ccc4B2F9SD2rm
> To: <sip:2850 at 85.16.245.213:1040;line=367hfn9i>;tag=zbrirvg9ow
> Call-ID: 97a0e683-34e6-122d-8db2-00144fe6e330
> CSeq: 121727216 INFO
> Contact: <sip:mod_sofia at 85.16.246.6:5060>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15137M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
> REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Content-Type: message/sipfrag
> Content-Length: 32
>
> From:
> To: "Snom1 an PBX" 4918
>
>
> What calle sees is this:
> 1. Phone rings: caller's displayname
> 2. Callee picks up: switching from dislayname to unknown
> 3. Switching from unknown to displayname
>
> After callee has picked up caller's display is also updated with
> callee's name (this is a copy of callee's number rather than callee's
> name) and number.
>
> First Question:
> Can this behaviour be disabled? Or can it be modified by dialplan, so
> that there is no "unknown"?
>
>
>
>
> Unfortunately when you do an outgoing call via openzap and callee picks
> up the phone caller's display is updated with two equal INFO message
> like this:
>
> INFO sip:2850 at 85.16.245.213:1040;line=367hfn9i SIP/2.0
> Via: SIP/2.0/UDP 85.16.246.6;rport;branch=z9hG4bK7jFa9Ztg39UUr
> Max-Forwards: 70
> From: <sip:890327 at 85.16.246.6 <sip%3A890327 at 85.16.246.6>
> ;user=phone>;tag=9QZamXtmUyXvm
> To: "Helmut Kuper" <sip:2850 at 85.16.246.6 <sip%3A2850 at 85.16.246.6>
> >;tag=5lf2e0q1on
> Call-ID: 3c2a2539a966-0m3i9s85my46
> CSeq: 121727707 INFO
> Contact: <sip:890327 at 85.16.246.6:5060;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15137M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
> REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Content-Type: message/sipfrag
> Content-Length: 36
>
> From:
> To: "1/a/890327" 1/a/890327
>
>
> Second Question:
> Can I change the name and number via dialplan, so that the correkt name
> and number is viewed to caller?
>
>
> regards
> helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
>
> iD8DBQFK2Fmk4tZeNddg3dwRApPUAKCQLBEi/l6mwf1c55r5mlnEqvLLowCgoP9t
> VSQFKUCWpsxVm3evyfWQT/Y=
> =FGsp
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From jerry.richards at teotech.com  Fri Oct 16 09:47:17 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 16 Oct 2009 09:47:17 -0700
Subject: [Freeswitch-users] scripts/contrib/trixter/makemodconf.pl: No such
	file or directory
Message-ID: <43360B9674B446119482FC37680F6F5F@greyhawk.tonecommander.com>

I am building Freeswitch on a Centos 5.3 machine and the last step below
gets an error because there is no scripts/contrib folder.  Anyone know why?

./configure
make
make all install sounds-install uhd-moh-install moh-install
scripts/contrib/trixter/makemodconf.pl modules.conf >
/usr/local/freeswitch/conf/autoload_configs/modules.conf.xml

bash: scripts/contrib/trixter/makemodconf.pl: No such file or directory

Best Regards,
Jerry



From jerry.richards at teotech.com  Fri Oct 16 10:16:04 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 16 Oct 2009 10:16:04 -0700
Subject: [Freeswitch-users] scripts/contrib/trixter/makemodconf.pl: No
	such file or directory
Message-ID: <1E8E134F44F146558C9CC9613FF48C2C@greyhawk.tonecommander.com>


Okay, I think the contrib folder moved up one level.  So the Wiki
installation documentation should probably be updated to reflect that.

Best Regards,
Jerry


-----Original Message-----
From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Friday, October 16, 2009 9:47 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: scripts/contrib/trixter/makemodconf.pl: No such file or directory

I am building Freeswitch on a Centos 5.3 machine and the last step below
gets an error because there is no scripts/contrib folder.  Anyone know why?

./configure
make
make all install sounds-install uhd-moh-install moh-install
scripts/contrib/trixter/makemodconf.pl modules.conf >
/usr/local/freeswitch/conf/autoload_configs/modules.conf.xml

bash: scripts/contrib/trixter/makemodconf.pl: No such file or directory

Best Regards,
Jerry



From miles.chet at gmail.com  Fri Oct 16 10:20:37 2009
From: miles.chet at gmail.com (Roberto Martins)
Date: Fri, 16 Oct 2009 14:20:37 -0300
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <ebbc2d550910160725x3141ce14re1821926d4027d42@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<d3ceea380910160702x8acbc4ci3bf3cfc1fdf9a9b7@mail.gmail.com>
	<ebbc2d550910160725x3141ce14re1821926d4027d42@mail.gmail.com>
Message-ID: <2E845C1E-C084-4025-806D-178A175FF90A@gmail.com>

we do!.

roberto
On Oct 16, 2009, at 11:25 AM, Jo?o Mesquita wrote:

> We do. :-)
>
> Jo?o Mesquita
>
> On Fri, Oct 16, 2009 at 11:02 AM, Itamar Reis Peixoto <itamar at ispbrasil.com.br 
> > wrote:
> I think no.
>
>
> On Fri, Oct 16, 2009 at 10:48 AM, Pedro Prado <pedroprado at msn.com>  
> wrote:
> > Hi,
> >
> > Do you have a group of Brazilians here?
> >
> > Thanks,
> > Pedro Prado
>
>
>
> --
> ------------
>
> Itamar Reis Peixoto
>
> e-mail/msn: itamar at ispbrasil.com.br
> sip: itamar at ispbrasil.com.br
> skype: itamarjp
> icq: 81053601
> +55 11 4063 5033
> +55 34 3221 8599
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From Russell.Mosemann at cune.org  Fri Oct 16 10:29:36 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Fri, 16 Oct 2009 17:29:36 -0000
Subject: [Freeswitch-users] Outbound sip address call fails
In-Reply-To: <011301ca4dd3$67d486a0$377d93e0$@com>
Message-ID: <20091016172936.CCE7C41E8F6@mail.cune.org>

Lars Zeb <larclap at yahoo.com> said:

> I tried dial sip:echo at iptel.org. In the pastebin you can see that this
> address came over to the cli as "echo". Is it Bria which is messing up, or
> my FS configuration?

It looks like a Bria problem. If you open the account information and go
to the Advanced tab, is the box checked to "Send outgoing request
directly to target"?


-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From brian at freeswitch.org  Fri Oct 16 10:38:51 2009
From: brian at freeswitch.org (Brian West)
Date: Fri, 16 Oct 2009 12:38:51 -0500
Subject: [Freeswitch-users] scripts/contrib/trixter/makemodconf.pl: No
	such file or directory
In-Reply-To: <1E8E134F44F146558C9CC9613FF48C2C@greyhawk.tonecommander.com>
References: <1E8E134F44F146558C9CC9613FF48C2C@greyhawk.tonecommander.com>
Message-ID: <803E6637-5D7C-4B88-84B1-9947A88E4B95@freeswitch.org>

You can signup for a wiki account and correct the documentation so  
others don't repeat the same issue.

Thanks,
Brian

On Oct 16, 2009, at 12:16 PM, Jerry Richards wrote:

>
> Okay, I think the contrib folder moved up one level.  So the Wiki
> installation documentation should probably be updated to reflect that.
>
> Best Regards,
> Jerry



From red.rain.seven at gmail.com  Fri Oct 16 11:53:43 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Sat, 17 Oct 2009 02:53:43 +0800
Subject: [Freeswitch-users] sched_api doesn't get launched
In-Reply-To: <3F1747F9-C543-4C6F-BB01-4DD5FA5B7BC5@jerris.com>
References: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
	<3F1747F9-C543-4C6F-BB01-4DD5FA5B7BC5@jerris.com>
Message-ID: <59ad9ca10910161153h5d531533vd844e49b9ca6d896@mail.gmail.com>

So how would you trigger it from a script dialplan? The only time it seemed
to work is when I did setVariable("api_after_bridge", "sched_api blah blah
blah");
but then it gets executed after the channel's been teared down. I thought
api_after_bridge means right after the call gets connected.

I need something to execute an api command right before or right after the
call gets bridged.

On Fri, Oct 16, 2009 at 9:27 PM, Michael Jerris <mike at jerris.com> wrote:

> sched_api is a fsapi command not a dialplan application, I believe
> sched_hangup is both.
> Mike
>
>
> On Oct 13, 2009, at 6:14 AM, Henry Huang wrote:
>
> Hi:
> I am using mod_java. And in my script I was able to achieve using:
>
> execute("sched_hangup", "+300 alloted_timeout");
>
> However, when I try to run sched_api in the same way, system log returns
> that it's an invalid application. I have also tried to trigger it with many
> conditional channel variable api calls , but non of them seemed to execute
> the api command (because I turned on the highest level of debugging and see
> no where the sched_api is being called.
>
> The closest thing I got was by using "api_after_bridge" like the following,
> but it only launches when the bridge is teared down(which is not what I
> want). I originally thought after bridge means right after the 2 party is
> connected. All I want is to be able to play some message to leg A at certain
> time.
>
> setVariable("api_after_bridge", "sched_api +10 none uuid_displace ${uuid}
> start /path/to/some.wav 20 mux");
>
> I have been struggling with different combination for a week now.. Please
> shed some light if you know something.
>
> Thanks,
>
> --
> Henry Huang
> UniC Solution - Communication Unified
> VoIP & Open Source software Consultant
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From msc at freeswitch.org  Fri Oct 16 12:37:20 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 16 Oct 2009 12:37:20 -0700
Subject: [Freeswitch-users] sched_api doesn't get launched
In-Reply-To: <59ad9ca10910161153h5d531533vd844e49b9ca6d896@mail.gmail.com>
References: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
	<3F1747F9-C543-4C6F-BB01-4DD5FA5B7BC5@jerris.com>
	<59ad9ca10910161153h5d531533vd844e49b9ca6d896@mail.gmail.com>
Message-ID: <87f2f3b90910161237h356b7aa2q6ec12c6ecf78dc34@mail.gmail.com>

On Fri, Oct 16, 2009 at 11:53 AM, Henry Huang <red.rain.seven at gmail.com>wrote:

> So how would you trigger it from a script dialplan? The only time it seemed
> to work is when I did setVariable("api_after_bridge", "sched_api blah blah
> blah");
> but then it gets executed after the channel's been teared down. I thought
> api_after_bridge means right after the call gets connected.
>
> I need something to execute an api command right before or right after the
> call gets bridged.
>
> api_after_bridge is a channel variable, so using setVariable works just
fine. If you need to sched_api is an API only. Check these out:
http://wiki.freeswitch.org/wiki/Mod_commands#Misc._Commands

So you need an API object in order to use it. I don't know the syntax for
creating an api obj in Java but in Lua it goes like this:
api = freeswitch.API();
res = api:execute("sched_api","+300 none my_api my_api_args")

Remember, if the method you are using isn't found in the dial plan tools
then it isn't a dial plan application. Make sure it's on the list:
http://wiki.freeswitch.org/wiki/Mod_dptools

On the other hand, API commands are listed here:
http://wiki.freeswitch.org/wiki/Mod_commands

dptools require a session object, api commands require an api object...

-MC
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From fabioferrari at gmail.com  Fri Oct 16 13:02:20 2009
From: fabioferrari at gmail.com (Fabio Ferrari)
Date: Fri, 16 Oct 2009 17:02:20 -0300
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <2E845C1E-C084-4025-806D-178A175FF90A@gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<d3ceea380910160702x8acbc4ci3bf3cfc1fdf9a9b7@mail.gmail.com> 
	<ebbc2d550910160725x3141ce14re1821926d4027d42@mail.gmail.com> 
	<2E845C1E-C084-4025-806D-178A175FF90A@gmail.com>
Message-ID: <f3cdc69d0910161302v6a97ead2xc2af5d73b5a048d7@mail.gmail.com>

Just to let you know, Jeremias Silva from Khomp registered the
freeswitchbrasil.org domain. I think there is space to build an wiki
with portuguese content there.

Fabio Ferrari

On Fri, Oct 16, 2009 at 14:20, Roberto Martins <miles.chet at gmail.com> wrote:
> we do!.
>
> roberto
> On Oct 16, 2009, at 11:25 AM, Jo?o Mesquita wrote:
>
>> We do. :-)
>>
>> Jo?o Mesquita
>>
>> On Fri, Oct 16, 2009 at 11:02 AM, Itamar Reis Peixoto <itamar at ispbrasil.com.br
>> > wrote:
>> I think no.
>>
>>
>> On Fri, Oct 16, 2009 at 10:48 AM, Pedro Prado <pedroprado at msn.com>
>> wrote:
>> > Hi,
>> >
>> > Do you have a group of Brazilians here?
>> >
>> > Thanks,
>> > Pedro Prado
>>
>>
>>
>> --
>> ------------
>>
>> Itamar Reis Peixoto
>>
>> e-mail/msn: itamar at ispbrasil.com.br
>> sip: itamar at ispbrasil.com.br
>> skype: itamarjp
>> icq: 81053601
>> +55 11 4063 5033
>> +55 34 3221 8599
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From tculjaga at gmail.com  Fri Oct 16 13:13:07 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 16 Oct 2009 22:13:07 +0200
Subject: [Freeswitch-users] Media got stuck after attended transfer...
In-Reply-To: <87f2f3b90910151710k34e4092eg26108dd819d9c041@mail.gmail.com>
References: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330253@srv02.khdev.corp>
	<191c3a030910150657r668eb5a3q24c641e312d2b113@mail.gmail.com>
	<65d96fc80910151154w2468ebeie06211d0966b4548@mail.gmail.com>
	<87f2f3b90910151710k34e4092eg26108dd819d9c041@mail.gmail.com>
Message-ID: <65d96fc80910161313p5d0aacecq1f1bb71ad514921e@mail.gmail.com>

> Of course, I was listening to my A.M radio the other day and they said that
> there was this new invention called the Internet that would let people send
> documents to each other electronically. Maybe you should look into that.
> Next thing you know they'll come up with telephones that people don't have
> to plug into the wall and can take with them in the car. ;)
>
>
yes, in that galaxy far far away :P



> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From itamar at ispbrasil.com.br  Fri Oct 16 13:28:30 2009
From: itamar at ispbrasil.com.br (Itamar Reis Peixoto)
Date: Fri, 16 Oct 2009 17:28:30 -0300
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <f3cdc69d0910161302v6a97ead2xc2af5d73b5a048d7@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<d3ceea380910160702x8acbc4ci3bf3cfc1fdf9a9b7@mail.gmail.com>
	<ebbc2d550910160725x3141ce14re1821926d4027d42@mail.gmail.com>
	<2E845C1E-C084-4025-806D-178A175FF90A@gmail.com>
	<f3cdc69d0910161302v6a97ead2xc2af5d73b5a048d7@mail.gmail.com>
Message-ID: <d3ceea380910161328w4c843e9as49b2341dc6f8eb7d@mail.gmail.com>

instead of creating thousand of small sites, why don't put the content
in only place.

freeswitch.org already exist, no need for a new domain.



On Fri, Oct 16, 2009 at 5:02 PM, Fabio Ferrari <fabioferrari at gmail.com> wrote:
> Just to let you know, Jeremias Silva from Khomp registered the
> freeswitchbrasil.org domain. I think there is space to build an wiki
> with portuguese content there.
>
> Fabio Ferrari


------------

Itamar Reis Peixoto

e-mail/msn: itamar at ispbrasil.com.br
sip: itamar at ispbrasil.com.br
skype: itamarjp
icq: 81053601
+55 11 4063 5033
+55 34 3221 8599


From anthony.minessale at gmail.com  Fri Oct 16 14:56:44 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 16 Oct 2009 16:56:44 -0500
Subject: [Freeswitch-users] Dingaling / Jingle DTMF support?
In-Reply-To: <33c87fa30910151519x637618d3p109f1b8edbf3432e@mail.gmail.com>
References: <33c87fa30910151519x637618d3p109f1b8edbf3432e@mail.gmail.com>
Message-ID: <191c3a030910161456y65d49120p8479c7c5df675125@mail.gmail.com>

It already should support at least recv of rfc2833 any time.
as a workaround in googletalk any string that starts with + typed in the
chat box is treated as dtmf by FS
e.g. +1

Once the jingle spec stops being a moving target we will re-investigate
making sure its supported properly.


On Thu, Oct 15, 2009 at 5:19 PM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> Hi!
>
> I was wondering if the Dingaling implementation in FS supports DTMF?
> This is now supported in the jingle specs (
> http://www.jabberforum.org/showthread.php?t=2709 ), even though Google
> Talk client does not currently support DTMF.
>
> If not, are there plans to implement this?
>
> Thanks!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Fri Oct 16 14:59:48 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 16 Oct 2009 16:59:48 -0500
Subject: [Freeswitch-users] T.38 via UPDATE request
In-Reply-To: <4AD884AA.1050805@gmx.net>
References: <4AD5AE3E.9070206@gmx.net>
	<0C137031-4C23-4A70-A05F-DE038D005839@jerris.com>
	<4AD884AA.1050805@gmx.net>
Message-ID: <191c3a030910161459t77162434k3b6a8977b594f393@mail.gmail.com>

I'm sure it won't work because we only implemented forwarding t38 with
invite
it would require adding code to say the least.


On Fri, Oct 16, 2009 at 9:35 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Hello Michael
>
> this is a productive system, so I can currently not update to the
> current trunk.
> But the installed SVN is 14741. Were those changes after 14741? Then
> I'll try to find a timeslot at night in order to update freeswitch.
>
> But Freeswitch should forward the UPDATE request in proxy-media mode,
> right?
>
> Best regards
> Peter
>
> Michael Jerris schrieb:
> > There was just a bunch of work on UPDATE, can you confirm this is the
> > same behavior with trunk?
> >
> > On Oct 14, 2009, at 6:55 AM, Peter P GMX wrote:
> >
> >
> >> Hello,
> >>
> >> we have the following problem.
> >> 2 Fax machines are communicating via Freeswitch. One is externally
> >> attached via a Telco who is able to handle T.38. The other one is
> >> attached locally.
> >>
> >> When 2 Fax machines start syncing each other, the Telco sends a SIP
> >> UPDATE message with T.38 SDP, as it detects fax during the fax
> >> negociations.
> >> Freeswitch answers with an SIP OK message back to the telco, and I can
> >> see the T.38 SDP on the debug console of freeswitch.
> >> Then nothing happens any more until one of fax machines detects
> >> timeout.
> >>
> >> We have set proxy-media to true. However is was done during call setup
> >> when both machines communicated with G711 SDP.
> >> The UPDATE message was commited by FS to the telco, but was not sent
> >> to
> >> the other fax, so I think in this case Freeswitch is supposed to
> >> transcode between T.38 and G711 which it cannot do, as we know.
> >>
> >> How can I overcome this scenario? Is this a defect, should freeswitch
> >> send the UPDATE message to the other fax?  Or is there a workaround?
> >>
> >> Best regards
> >> Peter
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From diego.viola at gmail.com  Fri Oct 16 15:10:35 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 16 Oct 2009 22:10:35 +0000
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <6b65470d0910131009y1cf136c3q8c0932375c19f272@mail.gmail.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
	<891080.87141.qm@web56406.mail.re3.yahoo.com>
	<87f2f3b90910130952r5657c8e2g6e11c22140da919c@mail.gmail.com>
	<6b65470d0910131009y1cf136c3q8c0932375c19f272@mail.gmail.com>
Message-ID: <86a32abc0910161510ma323718g410bcf1029156077@mail.gmail.com>

Can someone please convert this IRC log to documentation form:

http://wiki.freeswitch.org/wiki/Cacti

Try to explain what the IRC log explains in documentation form, and get rid
of the IRC log.

I'm working on fixing typos and stuff like that on the wiki now, then I will
focus on fixing the format, so all pages look the same (titles, choice of
case in text, etc.)

Thanks,

Diego

On Tue, Oct 13, 2009 at 5:09 PM, William Suffill
<william.suffill at gmail.com>wrote:

> Under special pages there is ways to get a list of all the wiki pages. (
> http://wiki.freeswitch.org/wiki/Special:SpecialPages)
>
> http://wiki.freeswitch.org/wiki/Special:AllPages
>
> Due to the number of pages it's broken into sub pages based in alphabetical
> order of the page names.
>
> -- W
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tayeb.meftah at gmail.com  Fri Oct 16 17:51:10 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sat, 17 Oct 2009 00:51:10 +0000
Subject: [Freeswitch-users] PBX vs. SIP proxy?
In-Reply-To: <25920788.post@talk.nabble.com>
References: <25920788.post@talk.nabble.com>
Message-ID: <4AD914FE.4020709@gmail.com>

hello
for sip proxy,
Click here <http://www.opensips.org/>
(http://www.opensips.org)
for a Ip PBX, sted with freeswitch (
http://www.Freeswitch.org)
sip provide a registrare to let phones register and proxy there request 
to other route or something
while PBX let local dialing, conferencing, voicemail, call  parcking and 
other stufs...
thank you and welcome to freeswitch
Fred-145 a ?crit :
> Hello
>
> I was reading an article the other day about Freeswitch in which the author
> used the terms "SIP proxy" and PBX. Is an SIP proxy just a way for SIP
> clients to register and connect to each other, while the PBX part adds
> services like voice mail, etc.?
>
> Thank you.
>   




__________ Information from ESET NOD32 Antivirus, version of virus signature database 4516 (20091016) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From mgende at gendesign.com  Fri Oct 16 17:29:46 2009
From: mgende at gendesign.com (Michael Gende)
Date: Fri, 16 Oct 2009 19:29:46 -0500
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <86a32abc0910161510ma323718g410bcf1029156077@mail.gmail.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
	<891080.87141.qm@web56406.mail.re3.yahoo.com>
	<87f2f3b90910130952r5657c8e2g6e11c22140da919c@mail.gmail.com>
	<6b65470d0910131009y1cf136c3q8c0932375c19f272@mail.gmail.com>
	<86a32abc0910161510ma323718g410bcf1029156077@mail.gmail.com>
Message-ID: <a87d43fd0910161729u42db5ea7qafb5604c0680954@mail.gmail.com>

Hey Diego,

You seem to know your way around the documentation and clearly have many
contributions. Thus, a question:

Can you point me to doc on dial plan applications (such as transfer, bridge,
etc)? It would help me greatly. Sorry if this is in an obvious place and
I've just not seen it despite the nose on my face.

Regards,

Mike G.

On Fri, Oct 16, 2009 at 5:10 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Can someone please convert this IRC log to documentation form:
>
> http://wiki.freeswitch.org/wiki/Cacti
>
> Try to explain what the IRC log explains in documentation form, and get rid
> of the IRC log.
>
> I'm working on fixing typos and stuff like that on the wiki now, then I
> will focus on fixing the format, so all pages look the same (titles, choice
> of case in text, etc.)
>
> Thanks,
>
> Diego
>
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From diego.viola at gmail.com  Fri Oct 16 18:11:42 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 17 Oct 2009 01:11:42 +0000
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <a87d43fd0910161729u42db5ea7qafb5604c0680954@mail.gmail.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
	<891080.87141.qm@web56406.mail.re3.yahoo.com>
	<87f2f3b90910130952r5657c8e2g6e11c22140da919c@mail.gmail.com>
	<6b65470d0910131009y1cf136c3q8c0932375c19f272@mail.gmail.com>
	<86a32abc0910161510ma323718g410bcf1029156077@mail.gmail.com>
	<a87d43fd0910161729u42db5ea7qafb5604c0680954@mail.gmail.com>
Message-ID: <86a32abc0910161811h15eac05dwc3fe0c2533b7397a@mail.gmail.com>

Hi Michael,

You might want to look at the mod_dptools wiki.

http://wiki.freeswitch.org/wiki/Mod_dptools

mod_dptools is the module that contains the dialplan apps.

Regards,

Diego

On Sat, Oct 17, 2009 at 12:29 AM, Michael Gende <mgende at gendesign.com>wrote:

> Hey Diego,
>
> You seem to know your way around the documentation and clearly have many
> contributions. Thus, a question:
>
> Can you point me to doc on dial plan applications (such as transfer,
> bridge, etc)? It would help me greatly. Sorry if this is in an obvious place
> and I've just not seen it despite the nose on my face.
>
> Regards,
>
> Mike G.
>
>
> On Fri, Oct 16, 2009 at 5:10 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> Can someone please convert this IRC log to documentation form:
>>
>> http://wiki.freeswitch.org/wiki/Cacti
>>
>> Try to explain what the IRC log explains in documentation form, and get
>> rid of the IRC log.
>>
>> I'm working on fixing typos and stuff like that on the wiki now, then I
>> will focus on fixing the format, so all pages look the same (titles, choice
>> of case in text, etc.)
>>
>> Thanks,
>>
>> Diego
>>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ruda at ruda.com.br  Fri Oct 16 18:22:19 2009
From: ruda at ruda.com.br (=?ISO-8859-1?Q?Rud=E1_Cunha?=)
Date: Fri, 16 Oct 2009 22:22:19 -0300
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <d3ceea380910161328w4c843e9as49b2341dc6f8eb7d@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<d3ceea380910160702x8acbc4ci3bf3cfc1fdf9a9b7@mail.gmail.com> 
	<ebbc2d550910160725x3141ce14re1821926d4027d42@mail.gmail.com> 
	<2E845C1E-C084-4025-806D-178A175FF90A@gmail.com>
	<f3cdc69d0910161302v6a97ead2xc2af5d73b5a048d7@mail.gmail.com> 
	<d3ceea380910161328w4c843e9as49b2341dc6f8eb7d@mail.gmail.com>
Message-ID: <7600794b0910161822j37848bbfk8b0eaf35eeeae189@mail.gmail.com>

Hello,

I wanted to create a forum so we can discuss FreeSWITCH.

That person who registered the domain, I can create and host the forum, so
we can talk there and ask questions


2009/10/16 Itamar Reis Peixoto <itamar at ispbrasil.com.br>

> instead of creating thousand of small sites, why don't put the content
> in only place.
>
> freeswitch.org already exist, no need for a new domain.
>
>
>
> On Fri, Oct 16, 2009 at 5:02 PM, Fabio Ferrari <fabioferrari at gmail.com>
> wrote:
> > Just to let you know, Jeremias Silva from Khomp registered the
> > freeswitchbrasil.org domain. I think there is space to build an wiki
> > with portuguese content there.
> >
> > Fabio Ferrari
>
>
> ------------
>
> Itamar Reis Peixoto
>
> e-mail/msn: itamar at ispbrasil.com.br
> sip: itamar at ispbrasil.com.br
> skype: itamarjp
> icq: 81053601
> +55 11 4063 5033
> +55 34 3221 8599
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From diego.viola at gmail.com  Fri Oct 16 18:47:56 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 17 Oct 2009 01:47:56 +0000
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <86a32abc0910161811h15eac05dwc3fe0c2533b7397a@mail.gmail.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
	<891080.87141.qm@web56406.mail.re3.yahoo.com>
	<87f2f3b90910130952r5657c8e2g6e11c22140da919c@mail.gmail.com>
	<6b65470d0910131009y1cf136c3q8c0932375c19f272@mail.gmail.com>
	<86a32abc0910161510ma323718g410bcf1029156077@mail.gmail.com>
	<a87d43fd0910161729u42db5ea7qafb5604c0680954@mail.gmail.com>
	<86a32abc0910161811h15eac05dwc3fe0c2533b7397a@mail.gmail.com>
Message-ID: <86a32abc0910161847k415b9722ubc5439ea1b9f6e78@mail.gmail.com>

My plan to improve the FreeSWITCH wiki


   1. Correct all typos from all pages.
   2. Delete unused/irrelevant pages.
   3. Make sure we don't repeat stuff (multi-tenant/multi-companies,
   features/spec-sheet, etc)
   4. Define a size for text, titles/subtitles (so all pages look the same).
   5. Define a case for text, titles/subtitles (so all pages look the same).

More to come...



On Sat, Oct 17, 2009 at 1:11 AM, Diego Viola <diego.viola at gmail.com> wrote:

> Hi Michael,
>
> You might want to look at the mod_dptools wiki.
>
> http://wiki.freeswitch.org/wiki/Mod_dptools
>
> mod_dptools is the module that contains the dialplan apps.
>
> Regards,
>
> Diego
>
> On Sat, Oct 17, 2009 at 12:29 AM, Michael Gende <mgende at gendesign.com>wrote:
>
>> Hey Diego,
>>
>> You seem to know your way around the documentation and clearly have many
>> contributions. Thus, a question:
>>
>> Can you point me to doc on dial plan applications (such as transfer,
>> bridge, etc)? It would help me greatly. Sorry if this is in an obvious place
>> and I've just not seen it despite the nose on my face.
>>
>> Regards,
>>
>> Mike G.
>>
>>
>> On Fri, Oct 16, 2009 at 5:10 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>
>>> Can someone please convert this IRC log to documentation form:
>>>
>>> http://wiki.freeswitch.org/wiki/Cacti
>>>
>>> Try to explain what the IRC log explains in documentation form, and get
>>> rid of the IRC log.
>>>
>>> I'm working on fixing typos and stuff like that on the wiki now, then I
>>> will focus on fixing the format, so all pages look the same (titles, choice
>>> of case in text, etc.)
>>>
>>> Thanks,
>>>
>>> Diego
>>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From jason at jasonjgw.net  Fri Oct 16 18:54:27 2009
From: jason at jasonjgw.net (Jason White)
Date: Sat, 17 Oct 2009 12:54:27 +1100
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <7600794b0910161822j37848bbfk8b0eaf35eeeae189@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<d3ceea380910160702x8acbc4ci3bf3cfc1fdf9a9b7@mail.gmail.com>
	<ebbc2d550910160725x3141ce14re1821926d4027d42@mail.gmail.com>
	<2E845C1E-C084-4025-806D-178A175FF90A@gmail.com>
	<f3cdc69d0910161302v6a97ead2xc2af5d73b5a048d7@mail.gmail.com>
	<d3ceea380910161328w4c843e9as49b2341dc6f8eb7d@mail.gmail.com>
	<7600794b0910161822j37848bbfk8b0eaf35eeeae189@mail.gmail.com>
Message-ID: <20091017015427.GA7044@jdc.jasonjgw.net>

Rud? Cunha <ruda at ruda.com.br> wrote:
> Hello,
> 
> I wanted to create a forum so we can discuss FreeSWITCH.

You're writing to one now. It can be accessed as a mailing list, on the Web at
gmane.org, and via NNTP at gmane.org. I think there's a Web archive somewhere
else, too.



From diego.viola at gmail.com  Fri Oct 16 18:54:16 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 17 Oct 2009 01:54:16 +0000
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <86a32abc0910161847k415b9722ubc5439ea1b9f6e78@mail.gmail.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
	<891080.87141.qm@web56406.mail.re3.yahoo.com>
	<87f2f3b90910130952r5657c8e2g6e11c22140da919c@mail.gmail.com>
	<6b65470d0910131009y1cf136c3q8c0932375c19f272@mail.gmail.com>
	<86a32abc0910161510ma323718g410bcf1029156077@mail.gmail.com>
	<a87d43fd0910161729u42db5ea7qafb5604c0680954@mail.gmail.com>
	<86a32abc0910161811h15eac05dwc3fe0c2533b7397a@mail.gmail.com>
	<86a32abc0910161847k415b9722ubc5439ea1b9f6e78@mail.gmail.com>
Message-ID: <86a32abc0910161854k7fd3fca6pc0686e928c772d2c@mail.gmail.com>

s/so all pages look the same/so all the pages have the same look and feel/

On Sat, Oct 17, 2009 at 1:47 AM, Diego Viola <diego.viola at gmail.com> wrote:

> My plan to improve the FreeSWITCH wiki
>
>
>    1. Correct all typos from all pages.
>    2. Delete unused/irrelevant pages.
>    3. Make sure we don't repeat stuff (multi-tenant/multi-companies,
>    features/spec-sheet, etc)
>    4. Define a size for text, titles/subtitles (so all pages look the
>    same).
>    5. Define a case for text, titles/subtitles (so all pages look the
>    same).
>
> More to come...
>
>
>
>
> On Sat, Oct 17, 2009 at 1:11 AM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> Hi Michael,
>>
>> You might want to look at the mod_dptools wiki.
>>
>> http://wiki.freeswitch.org/wiki/Mod_dptools
>>
>> mod_dptools is the module that contains the dialplan apps.
>>
>> Regards,
>>
>> Diego
>>
>> On Sat, Oct 17, 2009 at 12:29 AM, Michael Gende <mgende at gendesign.com>wrote:
>>
>>> Hey Diego,
>>>
>>> You seem to know your way around the documentation and clearly have many
>>> contributions. Thus, a question:
>>>
>>> Can you point me to doc on dial plan applications (such as transfer,
>>> bridge, etc)? It would help me greatly. Sorry if this is in an obvious place
>>> and I've just not seen it despite the nose on my face.
>>>
>>> Regards,
>>>
>>> Mike G.
>>>
>>>
>>> On Fri, Oct 16, 2009 at 5:10 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>
>>>> Can someone please convert this IRC log to documentation form:
>>>>
>>>> http://wiki.freeswitch.org/wiki/Cacti
>>>>
>>>> Try to explain what the IRC log explains in documentation form, and get
>>>> rid of the IRC log.
>>>>
>>>> I'm working on fixing typos and stuff like that on the wiki now, then I
>>>> will focus on fixing the format, so all pages look the same (titles, choice
>>>> of case in text, etc.)
>>>>
>>>> Thanks,
>>>>
>>>> Diego
>>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
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From jason at jasonjgw.net  Fri Oct 16 19:53:23 2009
From: jason at jasonjgw.net (Jason White)
Date: Sat, 17 Oct 2009 13:53:23 +1100
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <d3ceea380910161328w4c843e9as49b2341dc6f8eb7d@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<d3ceea380910160702x8acbc4ci3bf3cfc1fdf9a9b7@mail.gmail.com>
	<ebbc2d550910160725x3141ce14re1821926d4027d42@mail.gmail.com>
	<2E845C1E-C084-4025-806D-178A175FF90A@gmail.com>
	<f3cdc69d0910161302v6a97ead2xc2af5d73b5a048d7@mail.gmail.com>
	<d3ceea380910161328w4c843e9as49b2341dc6f8eb7d@mail.gmail.com>
Message-ID: <20091017025323.GA15649@jdc.jasonjgw.net>

Itamar Reis Peixoto <itamar at ispbrasil.com.br> wrote:
> instead of creating thousand of small sites, why don't put the content
> in only place.
> 
> freeswitch.org already exist, no need for a new domain.

This mailing list, the development mailing list, the IRC channel and the
FreeSWITCH conference all exist - no need for another "forum".



From moises.silva at gmail.com  Fri Oct 16 20:05:12 2009
From: moises.silva at gmail.com (Moises Silva)
Date: Fri, 16 Oct 2009 23:05:12 -0400
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <20091017025323.GA15649@jdc.jasonjgw.net>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<d3ceea380910160702x8acbc4ci3bf3cfc1fdf9a9b7@mail.gmail.com>
	<ebbc2d550910160725x3141ce14re1821926d4027d42@mail.gmail.com>
	<2E845C1E-C084-4025-806D-178A175FF90A@gmail.com>
	<f3cdc69d0910161302v6a97ead2xc2af5d73b5a048d7@mail.gmail.com>
	<d3ceea380910161328w4c843e9as49b2341dc6f8eb7d@mail.gmail.com>
	<20091017025323.GA15649@jdc.jasonjgw.net>
Message-ID: <c4d05cbe0910162005r3f094fc7ne448b905446282f4@mail.gmail.com>

>
> This mailing list, the development mailing list, the IRC channel and the
> FreeSWITCH conference all exist - no need for another "forum".
>
>
I think he meant a forum in Portuguese. May be it's hard to believe for some
people, but not everybody speaks English. Programmers tend to do it by
default, but there are a lot of Asterisk users in  Brazil or
spanish-speaking countries that post daily to spanish and portuguese forums.
I don't see a problem with creating a forum in portuguese if it helps to
bring adoption of FreeSWTICH.

I understand we want to avoid duplication of efforts, but the language
barrier is there and is not going to change.

-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From chris at fowler.cc  Fri Oct 16 20:06:52 2009
From: chris at fowler.cc (Chris Fowler)
Date: Fri, 16 Oct 2009 20:06:52 -0700
Subject: [Freeswitch-users] NAT problems migrating from Version 1.0.trunk
 (13168M) to Version 1.0.trunk (15166)
Message-ID: <1255748812.24395.1340540355@webmail.messagingengine.com>

Hi,

We've been using 13168M in production for some time now (works great). 
I want to get us onto the latest build but am having problems getting
NAT to work.

Phones can register; can dial 9999 test #, but after 100 seconds the
call is disconnected with error:
2009-10-16 19:52:26.936618 [NOTICE] sofia.c:4038 Hangup
sofia/internal/1001 at myhost.mydomain.com [CS_EXECUTE]
[RECOVERY_ON_TIMER_EXPIRE]

I took the standard internal.xml and vars.xml files from the new build
and made the following modifications - which worked previously:

modify conf/vars.xml and update
  <X-PRE-PROCESS cmd="set" data="domain=myhost.mydomain.com"/>

  <X-PRE-PROCESS cmd="set" data="bind_server_ip=1.2.3.4"/>
  <X-PRE-PROCESS cmd="set" data="external_rtp_ip=1.2.3.4"/>
  <X-PRE-PROCESS cmd="set" data="external_sip_ip=1.2.3.4"/>

Modify conf/sip_profiles/internal.xml
 <param name="aggressive-nat-detection" value="true"/>  <param
 name="ext-rtp-ip" value="$${external_rtp_ip}"/>  <param
 name="ext-sip-ip" value="$${external_sip_ip}"/>  <param
 name="NDLB-received-in-nat-reg-contact" value="true"/>  <param
 name="NDLB-force-rport" value="true"/>  <param
 name="NDLB-broken-auth-hash" value="true"/>

The big difference I note is that on PRODUCTION (which works) "sofia
status profile internal" yields:
URL                     sip:mod_sofia at 1.2.3.4:5060
BIND-URL                sip:mod_sofia at 1.2.3.4:5060;maddr=10.250.35.224

But on Test I see:
URL                     sip:mod_sofia at 10.250.66.210:5060
BIND-URL                sip:mod_sofia at 10.250.66.210:5060

Any ideas?

Thanks, Chris.



From diego.viola at gmail.com  Fri Oct 16 20:19:37 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 17 Oct 2009 03:19:37 +0000
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <c4d05cbe0910162005r3f094fc7ne448b905446282f4@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<d3ceea380910160702x8acbc4ci3bf3cfc1fdf9a9b7@mail.gmail.com>
	<ebbc2d550910160725x3141ce14re1821926d4027d42@mail.gmail.com>
	<2E845C1E-C084-4025-806D-178A175FF90A@gmail.com>
	<f3cdc69d0910161302v6a97ead2xc2af5d73b5a048d7@mail.gmail.com>
	<d3ceea380910161328w4c843e9as49b2341dc6f8eb7d@mail.gmail.com>
	<20091017025323.GA15649@jdc.jasonjgw.net>
	<c4d05cbe0910162005r3f094fc7ne448b905446282f4@mail.gmail.com>
Message-ID: <86a32abc0910162019m6315692bw566e282aaf8e2900@mail.gmail.com>

I'm with Moises and with the other people supporting this initiative.

I'm not Brazilian, but they should be able to do whatever they want, after
all, that's how open source works, if you can do it go ahead and do it.

Diego

On Sat, Oct 17, 2009 at 3:05 AM, Moises Silva <moises.silva at gmail.com>wrote:

> This mailing list, the development mailing list, the IRC channel and the
>> FreeSWITCH conference all exist - no need for another "forum".
>>
>>
> I think he meant a forum in Portuguese. May be it's hard to believe for
> some people, but not everybody speaks English. Programmers tend to do it by
> default, but there are a lot of Asterisk users in  Brazil or
> spanish-speaking countries that post daily to spanish and portuguese forums.
> I don't see a problem with creating a forum in portuguese if it helps to
> bring adoption of FreeSWTICH.
>
> I understand we want to avoid duplication of efforts, but the language
> barrier is there and is not going to change.
>
> --
> Moises Silva
> Software Developer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From diego.viola at gmail.com  Fri Oct 16 20:50:12 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 17 Oct 2009 03:50:12 +0000
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <86a32abc0910161854k7fd3fca6pc0686e928c772d2c@mail.gmail.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
	<891080.87141.qm@web56406.mail.re3.yahoo.com>
	<87f2f3b90910130952r5657c8e2g6e11c22140da919c@mail.gmail.com>
	<6b65470d0910131009y1cf136c3q8c0932375c19f272@mail.gmail.com>
	<86a32abc0910161510ma323718g410bcf1029156077@mail.gmail.com>
	<a87d43fd0910161729u42db5ea7qafb5604c0680954@mail.gmail.com>
	<86a32abc0910161811h15eac05dwc3fe0c2533b7397a@mail.gmail.com>
	<86a32abc0910161847k415b9722ubc5439ea1b9f6e78@mail.gmail.com>
	<86a32abc0910161854k7fd3fca6pc0686e928c772d2c@mail.gmail.com>
Message-ID: <86a32abc0910162050m6bad9a9do171b268e083b3d0d@mail.gmail.com>

This needs to be converted to documentation form too.

http://wiki.freeswitch.org/wiki/Mod_vmd#Related_conversations

Diego

On Sat, Oct 17, 2009 at 1:54 AM, Diego Viola <diego.viola at gmail.com> wrote:

> s/so all pages look the same/so all the pages have the same look and feel/
>
>
> On Sat, Oct 17, 2009 at 1:47 AM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> My plan to improve the FreeSWITCH wiki
>>
>>
>>    1. Correct all typos from all pages.
>>    2. Delete unused/irrelevant pages.
>>    3. Make sure we don't repeat stuff (multi-tenant/multi-companies,
>>    features/spec-sheet, etc)
>>    4. Define a size for text, titles/subtitles (so all pages look the
>>    same).
>>    5. Define a case for text, titles/subtitles (so all pages look the
>>    same).
>>
>> More to come...
>>
>>
>>
>>
>> On Sat, Oct 17, 2009 at 1:11 AM, Diego Viola <diego.viola at gmail.com>wrote:
>>
>>> Hi Michael,
>>>
>>> You might want to look at the mod_dptools wiki.
>>>
>>> http://wiki.freeswitch.org/wiki/Mod_dptools
>>>
>>> mod_dptools is the module that contains the dialplan apps.
>>>
>>> Regards,
>>>
>>> Diego
>>>
>>> On Sat, Oct 17, 2009 at 12:29 AM, Michael Gende <mgende at gendesign.com>wrote:
>>>
>>>> Hey Diego,
>>>>
>>>> You seem to know your way around the documentation and clearly have many
>>>> contributions. Thus, a question:
>>>>
>>>> Can you point me to doc on dial plan applications (such as transfer,
>>>> bridge, etc)? It would help me greatly. Sorry if this is in an obvious place
>>>> and I've just not seen it despite the nose on my face.
>>>>
>>>> Regards,
>>>>
>>>> Mike G.
>>>>
>>>>
>>>> On Fri, Oct 16, 2009 at 5:10 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>>
>>>>> Can someone please convert this IRC log to documentation form:
>>>>>
>>>>> http://wiki.freeswitch.org/wiki/Cacti
>>>>>
>>>>> Try to explain what the IRC log explains in documentation form, and get
>>>>> rid of the IRC log.
>>>>>
>>>>> I'm working on fixing typos and stuff like that on the wiki now, then I
>>>>> will focus on fixing the format, so all pages look the same (titles, choice
>>>>> of case in text, etc.)
>>>>>
>>>>> Thanks,
>>>>>
>>>>> Diego
>>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>
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From jason at jasonjgw.net  Fri Oct 16 20:52:15 2009
From: jason at jasonjgw.net (Jason White)
Date: Sat, 17 Oct 2009 14:52:15 +1100
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <86a32abc0910162019m6315692bw566e282aaf8e2900@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<d3ceea380910160702x8acbc4ci3bf3cfc1fdf9a9b7@mail.gmail.com>
	<ebbc2d550910160725x3141ce14re1821926d4027d42@mail.gmail.com>
	<2E845C1E-C084-4025-806D-178A175FF90A@gmail.com>
	<f3cdc69d0910161302v6a97ead2xc2af5d73b5a048d7@mail.gmail.com>
	<d3ceea380910161328w4c843e9as49b2341dc6f8eb7d@mail.gmail.com>
	<20091017025323.GA15649@jdc.jasonjgw.net>
	<c4d05cbe0910162005r3f094fc7ne448b905446282f4@mail.gmail.com>
	<86a32abc0910162019m6315692bw566e282aaf8e2900@mail.gmail.com>
Message-ID: <20091017035215.GA24308@jdc.jasonjgw.net>

Diego Viola <diego.viola at gmail.com> wrote:
> I'm with Moises and with the other people supporting this initiative.
> 
> I'm not Brazilian, but they should be able to do whatever they want, after
> all, that's how open source works, if you can do it go ahead and do it.

Correct. We have enough of them as far as English-language fora are concerned;
other languages are a different question altogether, though.



From jmesquita at freeswitch.org  Fri Oct 16 21:23:40 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 17 Oct 2009 01:23:40 -0300
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <20091017035215.GA24308@jdc.jasonjgw.net>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<d3ceea380910160702x8acbc4ci3bf3cfc1fdf9a9b7@mail.gmail.com>
	<ebbc2d550910160725x3141ce14re1821926d4027d42@mail.gmail.com>
	<2E845C1E-C084-4025-806D-178A175FF90A@gmail.com>
	<f3cdc69d0910161302v6a97ead2xc2af5d73b5a048d7@mail.gmail.com>
	<d3ceea380910161328w4c843e9as49b2341dc6f8eb7d@mail.gmail.com>
	<20091017025323.GA15649@jdc.jasonjgw.net>
	<c4d05cbe0910162005r3f094fc7ne448b905446282f4@mail.gmail.com>
	<86a32abc0910162019m6315692bw566e282aaf8e2900@mail.gmail.com>
	<20091017035215.GA24308@jdc.jasonjgw.net>
Message-ID: <ebbc2d550910162123p33a80f0fnd8ebac5fc569588d@mail.gmail.com>

Just a heads up, I have talked to Jeremias from Khomp today and he is
setting up the wiki. I will personally be adding contents to the that wiki
if it ever picks up.

Regards,

jm

On Sat, Oct 17, 2009 at 12:52 AM, Jason White <jason at jasonjgw.net> wrote:

> Diego Viola <diego.viola at gmail.com> wrote:
> > I'm with Moises and with the other people supporting this initiative.
> >
> > I'm not Brazilian, but they should be able to do whatever they want,
> after
> > all, that's how open source works, if you can do it go ahead and do it.
>
> Correct. We have enough of them as far as English-language fora are
> concerned;
> other languages are a different question altogether, though.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mgende at gendesign.com  Fri Oct 16 22:23:22 2009
From: mgende at gendesign.com (Michael Gende)
Date: Sat, 17 Oct 2009 00:23:22 -0500
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <86a32abc0910161811h15eac05dwc3fe0c2533b7397a@mail.gmail.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
	<891080.87141.qm@web56406.mail.re3.yahoo.com>
	<87f2f3b90910130952r5657c8e2g6e11c22140da919c@mail.gmail.com>
	<6b65470d0910131009y1cf136c3q8c0932375c19f272@mail.gmail.com>
	<86a32abc0910161510ma323718g410bcf1029156077@mail.gmail.com>
	<a87d43fd0910161729u42db5ea7qafb5604c0680954@mail.gmail.com>
	<86a32abc0910161811h15eac05dwc3fe0c2533b7397a@mail.gmail.com>
Message-ID: <a87d43fd0910162223l45e2adb5xf529f8d56107c786@mail.gmail.com>

That's exactly what I needed. Many thanks.

On Fri, Oct 16, 2009 at 8:11 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Hi Michael,
>
> You might want to look at the mod_dptools wiki.
>
> http://wiki.freeswitch.org/wiki/Mod_dptools
>
> mod_dptools is the module that contains the dialplan apps.
>
> Regards,
>
> Diego
>
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From msc at freeswitch.org  Fri Oct 16 22:54:15 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 16 Oct 2009 22:54:15 -0700
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <86a32abc0910162050m6bad9a9do171b268e083b3d0d@mail.gmail.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
	<891080.87141.qm@web56406.mail.re3.yahoo.com>
	<87f2f3b90910130952r5657c8e2g6e11c22140da919c@mail.gmail.com>
	<6b65470d0910131009y1cf136c3q8c0932375c19f272@mail.gmail.com>
	<86a32abc0910161510ma323718g410bcf1029156077@mail.gmail.com>
	<a87d43fd0910161729u42db5ea7qafb5604c0680954@mail.gmail.com>
	<86a32abc0910161811h15eac05dwc3fe0c2533b7397a@mail.gmail.com>
	<86a32abc0910161847k415b9722ubc5439ea1b9f6e78@mail.gmail.com>
	<86a32abc0910161854k7fd3fca6pc0686e928c772d2c@mail.gmail.com>
	<86a32abc0910162050m6bad9a9do171b268e083b3d0d@mail.gmail.com>
Message-ID: <87f2f3b90910162254q23893e4cx160fe4a5ae64cc91@mail.gmail.com>

Diego, when I get some time I will help convert these from IRC to wiki.
Thanks,
MC

On Fri, Oct 16, 2009 at 8:50 PM, Diego Viola <diego.viola at gmail.com> wrote:

> This needs to be converted to documentation form too.
>
> http://wiki.freeswitch.org/wiki/Mod_vmd#Related_conversations
>
> Diego
>
>
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From msc at freeswitch.org  Fri Oct 16 23:00:03 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 16 Oct 2009 23:00:03 -0700
Subject: [Freeswitch-users] Sending an Event to a Session for onInput
In-Reply-To: <0BDCD23D-D551-4A2F-9CD7-CC5A2F1382CB@jerris.com>
References: <4256bf830910091234t11f4f4d0w28283f341bd1fb06@mail.gmail.com>
	<ACA9A602-0925-4D6F-8B11-0B214A462500@jerris.com>
	<4256bf830910112311t409c45a3q2f26cef3464e49e7@mail.gmail.com>
	<0BDCD23D-D551-4A2F-9CD7-CC5A2F1382CB@jerris.com>
Message-ID: <87f2f3b90910162300x30701935k9f35162fb88838d9@mail.gmail.com>

Thanks for the explanation. I added this to the wiki.
-MC

On Thu, Oct 15, 2009 at 6:50 PM, Michael Jerris <mike at jerris.com> wrote:

> It updates the display on a phone if the phone supports this.  This works
> on some sip phones right now including polycom and snom.
> Mike
>
> On Oct 12, 2009, at 2:11 AM, Matthew Fong wrote:
>
>
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From msc at freeswitch.org  Fri Oct 16 23:06:51 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 16 Oct 2009 23:06:51 -0700
Subject: [Freeswitch-users] Qustion about INFO messages after
	Connect/Answer
In-Reply-To: <191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>
References: <4AD859A4.5000101@ewetel.de>
	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>
Message-ID: <87f2f3b90910162306s14356f1bmbc484761b99bcc42@mail.gmail.com>

On Fri, Oct 16, 2009 at 9:18 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> 1) you should update again there were a few issues.
> 2) you can set the variable sip_callee_id_name and sip_callee_id number on
> the inbound leg before you answer to control what it says.
>
>
> Thanks for the heads up. I added these two vars to the wiki on the chan
vars page, SIP-related vars section.
-MC
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From diego.viola at gmail.com  Fri Oct 16 23:12:38 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 17 Oct 2009 06:12:38 +0000
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <87f2f3b90910162254q23893e4cx160fe4a5ae64cc91@mail.gmail.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
	<87f2f3b90910130952r5657c8e2g6e11c22140da919c@mail.gmail.com>
	<6b65470d0910131009y1cf136c3q8c0932375c19f272@mail.gmail.com>
	<86a32abc0910161510ma323718g410bcf1029156077@mail.gmail.com>
	<a87d43fd0910161729u42db5ea7qafb5604c0680954@mail.gmail.com>
	<86a32abc0910161811h15eac05dwc3fe0c2533b7397a@mail.gmail.com>
	<86a32abc0910161847k415b9722ubc5439ea1b9f6e78@mail.gmail.com>
	<86a32abc0910161854k7fd3fca6pc0686e928c772d2c@mail.gmail.com>
	<86a32abc0910162050m6bad9a9do171b268e083b3d0d@mail.gmail.com>
	<87f2f3b90910162254q23893e4cx160fe4a5ae64cc91@mail.gmail.com>
Message-ID: <86a32abc0910162312le6c08b8jb53ae83bcafadab1@mail.gmail.com>

Thanks, our wiki is getting a lot better :).

Diego

On Sat, Oct 17, 2009 at 5:54 AM, Michael Collins <msc at freeswitch.org> wrote:

> Diego, when I get some time I will help convert these from IRC to wiki.
> Thanks,
> MC
>
>
> On Fri, Oct 16, 2009 at 8:50 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> This needs to be converted to documentation form too.
>>
>> http://wiki.freeswitch.org/wiki/Mod_vmd#Related_conversations
>>
>> Diego
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sjmudd at pobox.com  Fri Oct 16 23:44:14 2009
From: sjmudd at pobox.com (Simon J Mudd)
Date: 17 Oct 2009 08:44:14 +0200
Subject: [Freeswitch-users] macros for performing repeated dialplan
	actions?
In-Reply-To: <1C80FD78-71B7-4D62-8C45-2F5BC8C1564C@apartmentlines.com>
References: <20091014220135.GA12614@mad06.wl0.org>
	<87f2f3b90910141510n45401995y11e3e380653e9790@mail.gmail.com>
	<m3eip41yxv.fsf@mad06.wl0.org>
	<1C80FD78-71B7-4D62-8C45-2F5BC8C1564C@apartmentlines.com>
Message-ID: <m3aazq1pvl.fsf@mad06.wl0.org>

chad at apartmentlines.com (Chad Phillips -- Apartment Lines) writes:

> On Oct 15, 2009, at 11:03 AM, Simon J Mudd wrote:
> 
> > OK. That looks clear. However it doesn't look as if it's possible to
> > pass parameters this way which means that data can only be passed to
> > the "extension" via variables. That works but isn't so clean. I'll
> > check this out further shortly. I'm planning later to pass info like
> > "which gateway to talk to" and perhaps some sort of flag which
> > indicates "record: yes/no", and even include the bridge application
> > into this extension/macro.  That reduces the original dialplan pattern
> > matching to a much simpler and easier to configure setup for me.
> 
> here's a tip from a recovering asterisk user: don't feel like you have  
> to do everything via the dialplan.  this was a constant mantra in  
> asterisk, because calling AGI scripts was expensive.  this is not the  
> case with FreeSWITCH.  the general rule of thumb is: as soon as it  
> gets hard to do in the dialplan, farm it out to a script.  it's the  
> way FreeSWITCH is designed, so don't fight it  ;)

OK. That makes a lot of sense. Perhaps I'm just missing examples to copy
which do some of the things I want to do.

Since there are indications on this list that FreeSWITCH can do many
things the changeover from Asterisk to FreeSWITCH is quite difficult
at least once you start to try and do more complex things.

I've other plans like:

1. choose the cheapest gateway based on some scripts which download
rates from the various gateways web pages and let me compare rates.
Hard partly as the "charges pages" aren't designed to allow me to
match a number to a charge. Shame the VoIP community can't come up
with a common format "charge/rate file" to make this easier.

2. Try one gateway and if that fails try another one. Many of the
gateways I've been using are low cost and don't always work, so to
avoid me having to adjust things each time this might be useful.

3. Remove a lot of the "intelligence" about what to do to a backend
script/program or whatever which decides how to route calls and tells
FreeSWITCH what it decides. This may not scale for a large setup but
for SOHO is never going to be an issue. It also avoids me writing so
many xml configuration files which seems to get spread all over
various places of the conf/ tree.

However I've still got a lot of experimenting to do yet.

Simon


From rupa at rupa.com  Sat Oct 17 00:18:17 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 17 Oct 2009 02:18:17 -0500
Subject: [Freeswitch-users] macros for performing repeated dialplan
	actions?
In-Reply-To: <m3aazq1pvl.fsf@mad06.wl0.org>
References: <20091014220135.GA12614@mad06.wl0.org>
	<87f2f3b90910141510n45401995y11e3e380653e9790@mail.gmail.com>
	<m3eip41yxv.fsf@mad06.wl0.org>
	<1C80FD78-71B7-4D62-8C45-2F5BC8C1564C@apartmentlines.com>
	<m3aazq1pvl.fsf@mad06.wl0.org>
Message-ID: <b39273530910170018h5063c309xa96ab59b15256055@mail.gmail.com>

On Sat, Oct 17, 2009 at 1:44 AM, Simon J Mudd <sjmudd at pobox.com> wrote:
> I've other plans like:
>
> 1. choose the cheapest gateway based on some scripts which download
> rates from the various gateways web pages and let me compare rates.
> Hard partly as the "charges pages" aren't designed to allow me to
> match a number to a charge. Shame the VoIP community can't come up
> with a common format "charge/rate file" to make this easier.

Look at mod_lcr.  The only part you have to write is the part that
loads the database from your providers.

> 2. Try one gateway and if that fails try another one. Many of the
> gateways I've been using are low cost and don't always work, so to
> avoid me having to adjust things each time this might be useful.

bridge to ${lcr_auto_route} after calling the lcr application.

> 3. Remove a lot of the "intelligence" about what to do to a backend
> script/program or whatever which decides how to route calls and tells
> FreeSWITCH what it decides. This may not scale for a large setup but
> for SOHO is never going to be an issue. It also avoids me writing so
> many xml configuration files which seems to get spread all over
> various places of the conf/ tree.

This is for incoming calls to your DIDs?  Or something else?  I'm not
sure what you mean here.

> However I've still got a lot of experimenting to do yet.
>
> Simon



-- 
-Rupa


From ruda at ruda.com.br  Sat Oct 17 04:06:26 2009
From: ruda at ruda.com.br (=?ISO-8859-1?Q?Rud=E1_Cunha?=)
Date: Sat, 17 Oct 2009 08:06:26 -0300
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <ebbc2d550910162123p33a80f0fnd8ebac5fc569588d@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<ebbc2d550910160725x3141ce14re1821926d4027d42@mail.gmail.com> 
	<2E845C1E-C084-4025-806D-178A175FF90A@gmail.com>
	<f3cdc69d0910161302v6a97ead2xc2af5d73b5a048d7@mail.gmail.com> 
	<d3ceea380910161328w4c843e9as49b2341dc6f8eb7d@mail.gmail.com> 
	<20091017025323.GA15649@jdc.jasonjgw.net>
	<c4d05cbe0910162005r3f094fc7ne448b905446282f4@mail.gmail.com> 
	<86a32abc0910162019m6315692bw566e282aaf8e2900@mail.gmail.com> 
	<20091017035215.GA24308@jdc.jasonjgw.net>
	<ebbc2d550910162123p33a80f0fnd8ebac5fc569588d@mail.gmail.com>
Message-ID: <7600794b0910170406m164e3e2od198586f53de5efc@mail.gmail.com>

Jo?o Mesquita,

Could check with Jeremiah or contact me through, so I talk to him about the
forum?

He could redirect the domain to my hosting server and we can put there the
wiki and forums and we can communicate!

Rud? Cunha

2009/10/17 Jo?o Mesquita <jmesquita at freeswitch.org>

> Just a heads up, I have talked to Jeremias from Khomp today and he is
> setting up the wiki. I will personally be adding contents to the that wiki
> if it ever picks up.
>
> Regards,
>
> jm
>
>
> On Sat, Oct 17, 2009 at 12:52 AM, Jason White <jason at jasonjgw.net> wrote:
>
>> Diego Viola <diego.viola at gmail.com> wrote:
>> > I'm with Moises and with the other people supporting this initiative.
>> >
>> > I'm not Brazilian, but they should be able to do whatever they want,
>> after
>> > all, that's how open source works, if you can do it go ahead and do it.
>>
>> Correct. We have enough of them as far as English-language fora are
>> concerned;
>> other languages are a different question altogether, though.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ruda at ruda.com.br  Sat Oct 17 07:38:58 2009
From: ruda at ruda.com.br (=?ISO-8859-1?Q?Rud=E1_Cunha?=)
Date: Sat, 17 Oct 2009 11:38:58 -0300
Subject: [Freeswitch-users] SPA 3000 - Disconnect Tone
Message-ID: <7600794b0910170738l612e3163i567c5522313109ad@mail.gmail.com>

I am using SPA 3000 (FXO) except that the Disconnect Tone is not working for
an operator (GVT) now works in another carrier (OI). I'm from Brazil.

Anybody know how to find the Disconnect Tone Operator?

Att,
Rud? Cunha
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From anthony.minessale at gmail.com  Sat Oct 17 09:25:08 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 17 Oct 2009 11:25:08 -0500
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <7600794b0910170406m164e3e2od198586f53de5efc@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<2E845C1E-C084-4025-806D-178A175FF90A@gmail.com>
	<f3cdc69d0910161302v6a97ead2xc2af5d73b5a048d7@mail.gmail.com>
	<d3ceea380910161328w4c843e9as49b2341dc6f8eb7d@mail.gmail.com>
	<20091017025323.GA15649@jdc.jasonjgw.net>
	<c4d05cbe0910162005r3f094fc7ne448b905446282f4@mail.gmail.com>
	<86a32abc0910162019m6315692bw566e282aaf8e2900@mail.gmail.com>
	<20091017035215.GA24308@jdc.jasonjgw.net>
	<ebbc2d550910162123p33a80f0fnd8ebac5fc569588d@mail.gmail.com>
	<7600794b0910170406m164e3e2od198586f53de5efc@mail.gmail.com>
Message-ID: <191c3a030910170925i7aa7d4f3p9d382e0d855b7c8b@mail.gmail.com>

or someone could just ask us for pg.freeswitch.org or whatever other
sub-domain to avoid trademark infringement of registering 40 domains with
freeswitch in the name for no reason.


On Sat, Oct 17, 2009 at 6:06 AM, Rud? Cunha <ruda at ruda.com.br> wrote:

> Jo?o Mesquita,
>
> Could check with Jeremiah or contact me through, so I talk to him about the
> forum?
>
> He could redirect the domain to my hosting server and we can put there the
> wiki and forums and we can communicate!
>
> Rud? Cunha
>
> 2009/10/17 Jo?o Mesquita <jmesquita at freeswitch.org>
>
> Just a heads up, I have talked to Jeremias from Khomp today and he is
>> setting up the wiki. I will personally be adding contents to the that wiki
>> if it ever picks up.
>>
>> Regards,
>>
>> jm
>>
>>
>> On Sat, Oct 17, 2009 at 12:52 AM, Jason White <jason at jasonjgw.net> wrote:
>>
>>> Diego Viola <diego.viola at gmail.com> wrote:
>>> > I'm with Moises and with the other people supporting this initiative.
>>> >
>>> > I'm not Brazilian, but they should be able to do whatever they want,
>>> after
>>> > all, that's how open source works, if you can do it go ahead and do it.
>>>
>>> Correct. We have enough of them as far as English-language fora are
>>> concerned;
>>> other languages are a different question altogether, though.
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Sat Oct 17 09:30:45 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 17 Oct 2009 11:30:45 -0500
Subject: [Freeswitch-users] Sending an Event to a Session for onInput
In-Reply-To: <ACA9A602-0925-4D6F-8B11-0B214A462500@jerris.com>
References: <4256bf830910091234t11f4f4d0w28283f341bd1fb06@mail.gmail.com>
	<ACA9A602-0925-4D6F-8B11-0B214A462500@jerris.com>
Message-ID: <191c3a030910170930i30c0b428ya415162418db851b@mail.gmail.com>

actually,

If you use sendevent command and add a unique-id header to the event you
send, it will be sent to the channel rather than globally.


On Sun, Oct 11, 2009 at 5:04 PM, Michael Jerris <mike at jerris.com> wrote:

> We don't have session messages directly exposed, except for things
> like display, respond, and deflect.  What specifically are you trying
> to send ?
>
> Mike
>
> On Oct 9, 2009, at 3:34 PM, Matthew Fong wrote:
>
> > I'm used to using the onInput callbacks inside lua and javascript to
> > listen for dtmf and other events and perform a task accordingly. I'm
> > wondering if there is a way to send an event to a session or channel
> > that can be caught using the setInputCallback inside lua from
> > outside the session program. Maybe an API command that can generate
> > an event for a specific UUID. Does a mechanism exist to do this that
> > I'm over looking? Thanks.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Sat Oct 17 09:49:09 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 17 Oct 2009 11:49:09 -0500
Subject: [Freeswitch-users] SIP CANCEL error
In-Reply-To: <E1FE097E-599C-479E-9D1C-2308F213109C@gmail.com>
References: <E1FE097E-599C-479E-9D1C-2308F213109C@gmail.com>
Message-ID: <191c3a030910170949l1b4553dcub5c5d4a6ae529ae6@mail.gmail.com>

I think the phone is making a mistake with the CANCEL packet.

Here is the Via header from the INVITE the phone sends:
Via: SIP/2.0/UDP
190.137.190.131;rport;branch=z9hG4bKac109601000000fb4ad814425106977400000792

Here is the Via header from the CANCEL
Via: SIP/2.0/UDP
10.0.0.8;rport;branch=z9hG4bKac109601000000fb4ad814425106977400000792

it's identical aside from the private address contained in the CANCEL and a
public one in the INVITE

So the anti-nat settings on your phone are not working properly and they
forgot to substitute the public IP for the CANCEL.

That makes Sofia unable to match it to an existing call.



On Fri, Oct 16, 2009 at 1:44 AM, Rogelio Perez <rogelio.perez at gmail.com>wrote:

> Hi guys,
>
> I'm having a signaling issue on FS when calling from an endpoint to a
> GW.
> To reproduce the error I just call and hungup the call right away.
> FS gets the CANCEL and replies back with 481 Call/Transaction Does Not
> Exist, so the call continues and eventually connects, even when the A
> side of the call is closed.
> I have verified that the Call-ID of the CANCEL matches that from the
> INVITE, so I see no reasons for FS to reject it.
>
> SIP+FS logs in pastebin 10738.
>
> Any ideas?
>
> Thanks,
> Rogelio
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From miles.chet at gmail.com  Sat Oct 17 13:01:57 2009
From: miles.chet at gmail.com (Roberto Martins)
Date: Sat, 17 Oct 2009 17:01:57 -0300
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <191c3a030910170925i7aa7d4f3p9d382e0d855b7c8b@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<2E845C1E-C084-4025-806D-178A175FF90A@gmail.com>
	<f3cdc69d0910161302v6a97ead2xc2af5d73b5a048d7@mail.gmail.com>
	<d3ceea380910161328w4c843e9as49b2341dc6f8eb7d@mail.gmail.com>
	<20091017025323.GA15649@jdc.jasonjgw.net>
	<c4d05cbe0910162005r3f094fc7ne448b905446282f4@mail.gmail.com>
	<86a32abc0910162019m6315692bw566e282aaf8e2900@mail.gmail.com>
	<20091017035215.GA24308@jdc.jasonjgw.net>
	<ebbc2d550910162123p33a80f0fnd8ebac5fc569588d@mail.gmail.com>
	<7600794b0910170406m164e3e2od198586f53de5efc@mail.gmail.com>
	<191c3a030910170925i7aa7d4f3p9d382e0d855b7c8b@mail.gmail.com>
Message-ID: <6701786D-A8AF-444E-A5A7-29E69F6E055C@gmail.com>

sub-domain seem to be a very wise choice...
ptbr



On Oct 17, 2009, at 1:25 PM, Anthony Minessale wrote:

> or someone could just ask us for pg.freeswitch.org or whatever other  
> sub-domain to avoid trademark infringement of registering 40 domains  
> with freeswitch in the name for no reason.
>
>
> On Sat, Oct 17, 2009 at 6:06 AM, Rud? Cunha <ruda at ruda.com.br> wrote:
> Jo?o Mesquita,
>
> Could check with Jeremiah or contact me through, so I talk to him  
> about the forum?
>
> He could redirect the domain to my hosting server and we can put  
> there the wiki and forums and we can communicate!
>
> Rud? Cunha
>
> 2009/10/17 Jo?o Mesquita <jmesquita at freeswitch.org>
>
> Just a heads up, I have talked to Jeremias from Khomp today and he  
> is setting up the wiki. I will personally be adding contents to the  
> that wiki if it ever picks up.
>
> Regards,
>
> jm
>
>
> On Sat, Oct 17, 2009 at 12:52 AM, Jason White <jason at jasonjgw.net>  
> wrote:
> Diego Viola <diego.viola at gmail.com> wrote:
> > I'm with Moises and with the other people supporting this  
> initiative.
> >
> > I'm not Brazilian, but they should be able to do whatever they  
> want, after
> > all, that's how open source works, if you can do it go ahead and  
> do it.
>
> Correct. We have enough of them as far as English-language fora are  
> concerned;
> other languages are a different question altogether, though.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From diego.viola at gmail.com  Sat Oct 17 13:24:54 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 17 Oct 2009 20:24:54 +0000
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <191c3a030910170925i7aa7d4f3p9d382e0d855b7c8b@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<f3cdc69d0910161302v6a97ead2xc2af5d73b5a048d7@mail.gmail.com>
	<d3ceea380910161328w4c843e9as49b2341dc6f8eb7d@mail.gmail.com>
	<20091017025323.GA15649@jdc.jasonjgw.net>
	<c4d05cbe0910162005r3f094fc7ne448b905446282f4@mail.gmail.com>
	<86a32abc0910162019m6315692bw566e282aaf8e2900@mail.gmail.com>
	<20091017035215.GA24308@jdc.jasonjgw.net>
	<ebbc2d550910162123p33a80f0fnd8ebac5fc569588d@mail.gmail.com>
	<7600794b0910170406m164e3e2od198586f53de5efc@mail.gmail.com>
	<191c3a030910170925i7aa7d4f3p9d382e0d855b7c8b@mail.gmail.com>
Message-ID: <86a32abc0910171324l454b6719xba3c920d80d35b61@mail.gmail.com>

I'm with Anthony, the sub-domain idea is nice, I wish we could find a way to
do internalization in our wiki in a easy/elegantly way.

Diego

On Sat, Oct 17, 2009 at 4:25 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> or someone could just ask us for pg.freeswitch.org or whatever other
> sub-domain to avoid trademark infringement of registering 40 domains with
> freeswitch in the name for no reason.
>
>
>
> On Sat, Oct 17, 2009 at 6:06 AM, Rud? Cunha <ruda at ruda.com.br> wrote:
>
>> Jo?o Mesquita,
>>
>> Could check with Jeremiah or contact me through, so I talk to him about
>> the forum?
>>
>> He could redirect the domain to my hosting server and we can put there the
>> wiki and forums and we can communicate!
>>
>> Rud? Cunha
>>
>> 2009/10/17 Jo?o Mesquita <jmesquita at freeswitch.org>
>>
>> Just a heads up, I have talked to Jeremias from Khomp today and he is
>>> setting up the wiki. I will personally be adding contents to the that wiki
>>> if it ever picks up.
>>>
>>> Regards,
>>>
>>> jm
>>>
>>>
>>> On Sat, Oct 17, 2009 at 12:52 AM, Jason White <jason at jasonjgw.net>wrote:
>>>
>>>> Diego Viola <diego.viola at gmail.com> wrote:
>>>> > I'm with Moises and with the other people supporting this initiative.
>>>> >
>>>> > I'm not Brazilian, but they should be able to do whatever they want,
>>>> after
>>>> > all, that's how open source works, if you can do it go ahead and do
>>>> it.
>>>>
>>>> Correct. We have enough of them as far as English-language fora are
>>>> concerned;
>>>> other languages are a different question altogether, though.
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Sat Oct 17 13:29:55 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 17 Oct 2009 16:29:55 -0400
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <6701786D-A8AF-444E-A5A7-29E69F6E055C@gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<2E845C1E-C084-4025-806D-178A175FF90A@gmail.com>
	<f3cdc69d0910161302v6a97ead2xc2af5d73b5a048d7@mail.gmail.com>
	<d3ceea380910161328w4c843e9as49b2341dc6f8eb7d@mail.gmail.com>
	<20091017025323.GA15649@jdc.jasonjgw.net>
	<c4d05cbe0910162005r3f094fc7ne448b905446282f4@mail.gmail.com>
	<86a32abc0910162019m6315692bw566e282aaf8e2900@mail.gmail.com>
	<20091017035215.GA24308@jdc.jasonjgw.net>
	<ebbc2d550910162123p33a80f0fnd8ebac5fc569588d@mail.gmail.com>
	<7600794b0910170406m164e3e2od198586f53de5efc@mail.gmail.com>
	<191c3a030910170925i7aa7d4f3p9d382e0d855b7c8b@mail.gmail.com>
	<6701786D-A8AF-444E-A5A7-29E69F6E055C@gmail.com>
Message-ID: <B69597BF-DA4A-440B-8CB0-08F52F33DD17@jerris.com>

I think we strongly lean towards using a sub-domain (if necessary) and  
maintaining other language content in the same wiki in alternate  
pages.  If the wiki software we are using is not effective to create  
multiple languages we should find a way to do it all in one.  We can  
set up an additional mailing list if necessary but also we have no  
issue with going across multiple languages in both the mailing list  
and in irc.

Mike

On Oct 17, 2009, at 4:01 PM, Roberto Martins wrote:

> sub-domain seem to be a very wise choice...
> ptbr
>
>
>
> On Oct 17, 2009, at 1:25 PM, Anthony Minessale wrote:
>
>> or someone could just ask us for pg.freeswitch.org or whatever other
>> sub-domain to avoid trademark infringement of registering 40 domains
>> with freeswitch in the name for no reason.
>>
>>
>> On Sat, Oct 17, 2009 at 6:06 AM, Rud? Cunha <ruda at ruda.com.br> wrote:
>> Jo?o Mesquita,
>>
>> Could check with Jeremiah or contact me through, so I talk to him
>> about the forum?
>>
>> He could redirect the domain to my hosting server and we can put
>> there the wiki and forums and we can communicate!
>>
>> Rud? Cunha
>>
>> 2009/10/17 Jo?o Mesquita <jmesquita at freeswitch.org>
>>
>> Just a heads up, I have talked to Jeremias from Khomp today and he
>> is setting up the wiki. I will personally be adding contents to the
>> that wiki if it ever picks up.
>>
>> Regards,
>>
>> jm
>>
>>
>> On Sat, Oct 17, 2009 at 12:52 AM, Jason White <jason at jasonjgw.net>
>> wrote:
>> Diego Viola <diego.viola at gmail.com> wrote:
>>> I'm with Moises and with the other people supporting this
>> initiative.
>>>
>>> I'm not Brazilian, but they should be able to do whatever they
>> want, after
>>> all, that's how open source works, if you can do it go ahead and
>> do it.
>>
>> Correct. We have enough of them as far as English-language fora are
>> concerned;
>> other languages are a different question altogether, though.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> -- 
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:213-799-1400
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From diego.viola at gmail.com  Sat Oct 17 13:39:53 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 17 Oct 2009 20:39:53 +0000
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <B69597BF-DA4A-440B-8CB0-08F52F33DD17@jerris.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<20091017025323.GA15649@jdc.jasonjgw.net>
	<c4d05cbe0910162005r3f094fc7ne448b905446282f4@mail.gmail.com>
	<86a32abc0910162019m6315692bw566e282aaf8e2900@mail.gmail.com>
	<20091017035215.GA24308@jdc.jasonjgw.net>
	<ebbc2d550910162123p33a80f0fnd8ebac5fc569588d@mail.gmail.com>
	<7600794b0910170406m164e3e2od198586f53de5efc@mail.gmail.com>
	<191c3a030910170925i7aa7d4f3p9d382e0d855b7c8b@mail.gmail.com>
	<6701786D-A8AF-444E-A5A7-29E69F6E055C@gmail.com>
	<B69597BF-DA4A-440B-8CB0-08F52F33DD17@jerris.com>
Message-ID: <86a32abc0910171339o6a300932v66b7f00419a0cb69@mail.gmail.com>

If we find a better wiki software that does internalization in a better way,
I could help porting the documentation over to the new software.

The sub-domain idea is also very nice.

Diego


On Sat, Oct 17, 2009 at 8:29 PM, Michael Jerris <mike at jerris.com> wrote:

> I think we strongly lean towards using a sub-domain (if necessary) and
> maintaining other language content in the same wiki in alternate
> pages.  If the wiki software we are using is not effective to create
> multiple languages we should find a way to do it all in one.  We can
> set up an additional mailing list if necessary but also we have no
> issue with going across multiple languages in both the mailing list
> and in irc.
>
> Mike
>
> On Oct 17, 2009, at 4:01 PM, Roberto Martins wrote:
>
> > sub-domain seem to be a very wise choice...
> > ptbr
> >
> >
> >
> > On Oct 17, 2009, at 1:25 PM, Anthony Minessale wrote:
> >
> >> or someone could just ask us for pg.freeswitch.org or whatever other
> >> sub-domain to avoid trademark infringement of registering 40 domains
> >> with freeswitch in the name for no reason.
> >>
> >>
> >> On Sat, Oct 17, 2009 at 6:06 AM, Rud? Cunha <ruda at ruda.com.br> wrote:
> >> Jo?o Mesquita,
> >>
> >> Could check with Jeremiah or contact me through, so I talk to him
> >> about the forum?
> >>
> >> He could redirect the domain to my hosting server and we can put
> >> there the wiki and forums and we can communicate!
> >>
> >> Rud? Cunha
> >>
> >> 2009/10/17 Jo?o Mesquita <jmesquita at freeswitch.org>
> >>
> >> Just a heads up, I have talked to Jeremias from Khomp today and he
> >> is setting up the wiki. I will personally be adding contents to the
> >> that wiki if it ever picks up.
> >>
> >> Regards,
> >>
> >> jm
> >>
> >>
> >> On Sat, Oct 17, 2009 at 12:52 AM, Jason White <jason at jasonjgw.net>
> >> wrote:
> >> Diego Viola <diego.viola at gmail.com> wrote:
> >>> I'm with Moises and with the other people supporting this
> >> initiative.
> >>>
> >>> I'm not Brazilian, but they should be able to do whatever they
> >> want, after
> >>> all, that's how open source works, if you can do it go ahead and
> >> do it.
> >>
> >> Correct. We have enough of them as far as English-language fora are
> >> concerned;
> >> other languages are a different question altogether, though.
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> iax:guest at conference.freeswitch.org/888
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> pstn:213-799-1400
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Sat Oct 17 14:04:07 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 17 Oct 2009 17:04:07 -0400
Subject: [Freeswitch-users] Dingaling: using a hostname instead of stun
	for rtp
In-Reply-To: <65BF398D-E3A6-4A98-83D9-D4E271070CCA@freeswitch.org>
References: <33c87fa30910131502t2af24801i1c3319518b5db50b@mail.gmail.com>
	<C874E25D-6D73-49AA-8DD3-0A0022617FD2@freeswitch.org>
	<33c87fa30910152035w43e260faoc031a7a4ac949d9e@mail.gmail.com>
	<65BF398D-E3A6-4A98-83D9-D4E271070CCA@freeswitch.org>
Message-ID: <7710644E-CEA1-4D48-8A03-EEE1C18F1B44@jerris.com>

If you don't have working stun, jingle is not going to work very  
well.  It is a required part of the protocol.  You need to be able to  
determine your external ports for media on each call, using a host  
name will not do this for you.

Mike


On Oct 16, 2009, at 10:48 AM, Brian West wrote:

> If you setup your own stun server it wouldn't do that....  But the
> hostlookup only solves half the problem .. getting the external IP vs
> poking holes for RTP which is what stun will do.
>
> /b
>
> On Oct 15, 2009, at 10:35 PM, Mark Campbell-Smith wrote:
>
>> Thanks Brian.  Is this something that is planned to be implemented?
>> The workaround is to set the stun server also in the dingaling
>> configuration, but as I said, for some reason the stun times for me
>> out occasionally with dingaling.
>>
>> Thanks!
>



From diego.viola at gmail.com  Sat Oct 17 15:13:53 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 17 Oct 2009 22:13:53 +0000
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <86a32abc0910171339o6a300932v66b7f00419a0cb69@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<c4d05cbe0910162005r3f094fc7ne448b905446282f4@mail.gmail.com>
	<86a32abc0910162019m6315692bw566e282aaf8e2900@mail.gmail.com>
	<20091017035215.GA24308@jdc.jasonjgw.net>
	<ebbc2d550910162123p33a80f0fnd8ebac5fc569588d@mail.gmail.com>
	<7600794b0910170406m164e3e2od198586f53de5efc@mail.gmail.com>
	<191c3a030910170925i7aa7d4f3p9d382e0d855b7c8b@mail.gmail.com>
	<6701786D-A8AF-444E-A5A7-29E69F6E055C@gmail.com>
	<B69597BF-DA4A-440B-8CB0-08F52F33DD17@jerris.com>
	<86a32abc0910171339o6a300932v66b7f00419a0cb69@mail.gmail.com>
Message-ID: <86a32abc0910171513n35d7d0echff940a16685ae1a4@mail.gmail.com>

Would you be open to test another wiki software, that supports
internalization/translation easier, if i am open to port the entire
documentation to the new wiki software?




On Sat, Oct 17, 2009 at 8:39 PM, Diego Viola <diego.viola at gmail.com> wrote:

> If we find a better wiki software that does internalization in a better
> way, I could help porting the documentation over to the new software.
>
> The sub-domain idea is also very nice.
>
> Diego
>
>
>
> On Sat, Oct 17, 2009 at 8:29 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> I think we strongly lean towards using a sub-domain (if necessary) and
>> maintaining other language content in the same wiki in alternate
>> pages.  If the wiki software we are using is not effective to create
>> multiple languages we should find a way to do it all in one.  We can
>> set up an additional mailing list if necessary but also we have no
>> issue with going across multiple languages in both the mailing list
>> and in irc.
>>
>> Mike
>>
>> On Oct 17, 2009, at 4:01 PM, Roberto Martins wrote:
>>
>> > sub-domain seem to be a very wise choice...
>> > ptbr
>> >
>> >
>> >
>> > On Oct 17, 2009, at 1:25 PM, Anthony Minessale wrote:
>> >
>> >> or someone could just ask us for pg.freeswitch.org or whatever other
>> >> sub-domain to avoid trademark infringement of registering 40 domains
>> >> with freeswitch in the name for no reason.
>> >>
>> >>
>> >> On Sat, Oct 17, 2009 at 6:06 AM, Rud? Cunha <ruda at ruda.com.br> wrote:
>> >> Jo?o Mesquita,
>> >>
>> >> Could check with Jeremiah or contact me through, so I talk to him
>> >> about the forum?
>> >>
>> >> He could redirect the domain to my hosting server and we can put
>> >> there the wiki and forums and we can communicate!
>> >>
>> >> Rud? Cunha
>> >>
>> >> 2009/10/17 Jo?o Mesquita <jmesquita at freeswitch.org>
>> >>
>> >> Just a heads up, I have talked to Jeremias from Khomp today and he
>> >> is setting up the wiki. I will personally be adding contents to the
>> >> that wiki if it ever picks up.
>> >>
>> >> Regards,
>> >>
>> >> jm
>> >>
>> >>
>> >> On Sat, Oct 17, 2009 at 12:52 AM, Jason White <jason at jasonjgw.net>
>> >> wrote:
>> >> Diego Viola <diego.viola at gmail.com> wrote:
>> >>> I'm with Moises and with the other people supporting this
>> >> initiative.
>> >>>
>> >>> I'm not Brazilian, but they should be able to do whatever they
>> >> want, after
>> >>> all, that's how open source works, if you can do it go ahead and
>> >> do it.
>> >>
>> >> Correct. We have enough of them as far as English-language fora are
>> >> concerned;
>> >> other languages are a different question altogether, though.
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> >> users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> >> users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> >> users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH http://www.freeswitch.org/
>> >> ClueCon http://www.cluecon.com/
>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>
>> >> AIM: anthm
>> >> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> >> IRC: irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>> >> iax:guest at conference.freeswitch.org/888
>> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> >> pstn:213-799-1400
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> >> users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> > users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From diego.viola at gmail.com  Sat Oct 17 15:21:32 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 17 Oct 2009 22:21:32 +0000
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <86a32abc0910171513n35d7d0echff940a16685ae1a4@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<86a32abc0910162019m6315692bw566e282aaf8e2900@mail.gmail.com>
	<20091017035215.GA24308@jdc.jasonjgw.net>
	<ebbc2d550910162123p33a80f0fnd8ebac5fc569588d@mail.gmail.com>
	<7600794b0910170406m164e3e2od198586f53de5efc@mail.gmail.com>
	<191c3a030910170925i7aa7d4f3p9d382e0d855b7c8b@mail.gmail.com>
	<6701786D-A8AF-444E-A5A7-29E69F6E055C@gmail.com>
	<B69597BF-DA4A-440B-8CB0-08F52F33DD17@jerris.com>
	<86a32abc0910171339o6a300932v66b7f00419a0cb69@mail.gmail.com>
	<86a32abc0910171513n35d7d0echff940a16685ae1a4@mail.gmail.com>
Message-ID: <86a32abc0910171521n403094bfvf27aa7c2a1c1f798@mail.gmail.com>

Some interesting pages on the subject.

http://en.wikipedia.org/wiki/Comparison_of_wiki_software
http://www.osnews.com/story/21865/Recommend_an_Open_Source_Wiki_Platform



On Sat, Oct 17, 2009 at 10:13 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Would you be open to test another wiki software, that supports
> internalization/translation easier, if i am open to port the entire
> documentation to the new wiki software?
>
>
>
>
>
> On Sat, Oct 17, 2009 at 8:39 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> If we find a better wiki software that does internalization in a better
>> way, I could help porting the documentation over to the new software.
>>
>> The sub-domain idea is also very nice.
>>
>> Diego
>>
>>
>>
>> On Sat, Oct 17, 2009 at 8:29 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> I think we strongly lean towards using a sub-domain (if necessary) and
>>> maintaining other language content in the same wiki in alternate
>>> pages.  If the wiki software we are using is not effective to create
>>> multiple languages we should find a way to do it all in one.  We can
>>> set up an additional mailing list if necessary but also we have no
>>> issue with going across multiple languages in both the mailing list
>>> and in irc.
>>>
>>> Mike
>>>
>>> On Oct 17, 2009, at 4:01 PM, Roberto Martins wrote:
>>>
>>> > sub-domain seem to be a very wise choice...
>>> > ptbr
>>> >
>>> >
>>> >
>>> > On Oct 17, 2009, at 1:25 PM, Anthony Minessale wrote:
>>> >
>>> >> or someone could just ask us for pg.freeswitch.org or whatever other
>>> >> sub-domain to avoid trademark infringement of registering 40 domains
>>> >> with freeswitch in the name for no reason.
>>> >>
>>> >>
>>> >> On Sat, Oct 17, 2009 at 6:06 AM, Rud? Cunha <ruda at ruda.com.br> wrote:
>>> >> Jo?o Mesquita,
>>> >>
>>> >> Could check with Jeremiah or contact me through, so I talk to him
>>> >> about the forum?
>>> >>
>>> >> He could redirect the domain to my hosting server and we can put
>>> >> there the wiki and forums and we can communicate!
>>> >>
>>> >> Rud? Cunha
>>> >>
>>> >> 2009/10/17 Jo?o Mesquita <jmesquita at freeswitch.org>
>>> >>
>>> >> Just a heads up, I have talked to Jeremias from Khomp today and he
>>> >> is setting up the wiki. I will personally be adding contents to the
>>> >> that wiki if it ever picks up.
>>> >>
>>> >> Regards,
>>> >>
>>> >> jm
>>> >>
>>> >>
>>> >> On Sat, Oct 17, 2009 at 12:52 AM, Jason White <jason at jasonjgw.net>
>>> >> wrote:
>>> >> Diego Viola <diego.viola at gmail.com> wrote:
>>> >>> I'm with Moises and with the other people supporting this
>>> >> initiative.
>>> >>>
>>> >>> I'm not Brazilian, but they should be able to do whatever they
>>> >> want, after
>>> >>> all, that's how open source works, if you can do it go ahead and
>>> >> do it.
>>> >>
>>> >> Correct. We have enough of them as far as English-language fora are
>>> >> concerned;
>>> >> other languages are a different question altogether, though.
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> >> users
>>> >> http://www.freeswitch.org
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> >> users
>>> >> http://www.freeswitch.org
>>> >>
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> >> users
>>> >> http://www.freeswitch.org
>>> >>
>>> >>
>>> >>
>>> >>
>>> >> --
>>> >> Anthony Minessale II
>>> >>
>>> >> FreeSWITCH http://www.freeswitch.org/
>>> >> ClueCon http://www.cluecon.com/
>>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>>> >>
>>> >> AIM: anthm
>>> >> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> >> IRC: irc.freenode.net #freeswitch
>>> >>
>>> >> FreeSWITCH Developer Conference
>>> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>> >> iax:guest at conference.freeswitch.org/888
>>> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> >> pstn:213-799-1400
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> >> users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> > users
>>> > http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
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From diego.viola at gmail.com  Sat Oct 17 15:35:43 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 17 Oct 2009 22:35:43 +0000
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <86a32abc0910171521n403094bfvf27aa7c2a1c1f798@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<20091017035215.GA24308@jdc.jasonjgw.net>
	<ebbc2d550910162123p33a80f0fnd8ebac5fc569588d@mail.gmail.com>
	<7600794b0910170406m164e3e2od198586f53de5efc@mail.gmail.com>
	<191c3a030910170925i7aa7d4f3p9d382e0d855b7c8b@mail.gmail.com>
	<6701786D-A8AF-444E-A5A7-29E69F6E055C@gmail.com>
	<B69597BF-DA4A-440B-8CB0-08F52F33DD17@jerris.com>
	<86a32abc0910171339o6a300932v66b7f00419a0cb69@mail.gmail.com>
	<86a32abc0910171513n35d7d0echff940a16685ae1a4@mail.gmail.com>
	<86a32abc0910171521n403094bfvf27aa7c2a1c1f798@mail.gmail.com>
Message-ID: <86a32abc0910171535l606eb9c6id54f7b3b5ae1b670@mail.gmail.com>

We could also do this for a better documentation.


   1. Pick a new wiki engine that does internalization/translation easier.
   2. Set some documentation guidelines from the start.
   3. Move stuff over to the new wiki.
   4. Start with the translation/internalization on the new wiki.


Diego

On Sat, Oct 17, 2009 at 10:21 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Some interesting pages on the subject.
>
> http://en.wikipedia.org/wiki/Comparison_of_wiki_software
> http://www.osnews.com/story/21865/Recommend_an_Open_Source_Wiki_Platform
>
>
>
>
> On Sat, Oct 17, 2009 at 10:13 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> Would you be open to test another wiki software, that supports
>> internalization/translation easier, if i am open to port the entire
>> documentation to the new wiki software?
>>
>>
>>
>>
>>
>> On Sat, Oct 17, 2009 at 8:39 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>
>>> If we find a better wiki software that does internalization in a better
>>> way, I could help porting the documentation over to the new software.
>>>
>>> The sub-domain idea is also very nice.
>>>
>>> Diego
>>>
>>>
>>>
>>> On Sat, Oct 17, 2009 at 8:29 PM, Michael Jerris <mike at jerris.com> wrote:
>>>
>>>> I think we strongly lean towards using a sub-domain (if necessary) and
>>>> maintaining other language content in the same wiki in alternate
>>>> pages.  If the wiki software we are using is not effective to create
>>>> multiple languages we should find a way to do it all in one.  We can
>>>> set up an additional mailing list if necessary but also we have no
>>>> issue with going across multiple languages in both the mailing list
>>>> and in irc.
>>>>
>>>> Mike
>>>>
>>>> On Oct 17, 2009, at 4:01 PM, Roberto Martins wrote:
>>>>
>>>> > sub-domain seem to be a very wise choice...
>>>> > ptbr
>>>> >
>>>> >
>>>> >
>>>> > On Oct 17, 2009, at 1:25 PM, Anthony Minessale wrote:
>>>> >
>>>> >> or someone could just ask us for pg.freeswitch.org or whatever other
>>>> >> sub-domain to avoid trademark infringement of registering 40 domains
>>>> >> with freeswitch in the name for no reason.
>>>> >>
>>>> >>
>>>> >> On Sat, Oct 17, 2009 at 6:06 AM, Rud? Cunha <ruda at ruda.com.br>
>>>> wrote:
>>>> >> Jo?o Mesquita,
>>>> >>
>>>> >> Could check with Jeremiah or contact me through, so I talk to him
>>>> >> about the forum?
>>>> >>
>>>> >> He could redirect the domain to my hosting server and we can put
>>>> >> there the wiki and forums and we can communicate!
>>>> >>
>>>> >> Rud? Cunha
>>>> >>
>>>> >> 2009/10/17 Jo?o Mesquita <jmesquita at freeswitch.org>
>>>> >>
>>>> >> Just a heads up, I have talked to Jeremias from Khomp today and he
>>>> >> is setting up the wiki. I will personally be adding contents to the
>>>> >> that wiki if it ever picks up.
>>>> >>
>>>> >> Regards,
>>>> >>
>>>> >> jm
>>>> >>
>>>> >>
>>>> >> On Sat, Oct 17, 2009 at 12:52 AM, Jason White <jason at jasonjgw.net>
>>>> >> wrote:
>>>> >> Diego Viola <diego.viola at gmail.com> wrote:
>>>> >>> I'm with Moises and with the other people supporting this
>>>> >> initiative.
>>>> >>>
>>>> >>> I'm not Brazilian, but they should be able to do whatever they
>>>> >> want, after
>>>> >>> all, that's how open source works, if you can do it go ahead and
>>>> >> do it.
>>>> >>
>>>> >> Correct. We have enough of them as far as English-language fora are
>>>> >> concerned;
>>>> >> other languages are a different question altogether, though.
>>>> >>
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>> >> users
>>>> >> http://www.freeswitch.org
>>>> >>
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>> >> users
>>>> >> http://www.freeswitch.org
>>>> >>
>>>> >>
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>> >> users
>>>> >> http://www.freeswitch.org
>>>> >>
>>>> >>
>>>> >>
>>>> >>
>>>> >> --
>>>> >> Anthony Minessale II
>>>> >>
>>>> >> FreeSWITCH http://www.freeswitch.org/
>>>> >> ClueCon http://www.cluecon.com/
>>>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>>>> >>
>>>> >> AIM: anthm
>>>> >> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> >> IRC: irc.freenode.net #freeswitch
>>>> >>
>>>> >> FreeSWITCH Developer Conference
>>>> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>> >> iax:guest at conference.freeswitch.org/888
>>>> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> >> pstn:213-799-1400
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>> >> users
>>>> >> http://www.freeswitch.org
>>>> >
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>> > users
>>>> > http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>
>
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From diego.viola at gmail.com  Sat Oct 17 15:45:40 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 17 Oct 2009 22:45:40 +0000
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <86a32abc0910171535l606eb9c6id54f7b3b5ae1b670@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<ebbc2d550910162123p33a80f0fnd8ebac5fc569588d@mail.gmail.com>
	<7600794b0910170406m164e3e2od198586f53de5efc@mail.gmail.com>
	<191c3a030910170925i7aa7d4f3p9d382e0d855b7c8b@mail.gmail.com>
	<6701786D-A8AF-444E-A5A7-29E69F6E055C@gmail.com>
	<B69597BF-DA4A-440B-8CB0-08F52F33DD17@jerris.com>
	<86a32abc0910171339o6a300932v66b7f00419a0cb69@mail.gmail.com>
	<86a32abc0910171513n35d7d0echff940a16685ae1a4@mail.gmail.com>
	<86a32abc0910171521n403094bfvf27aa7c2a1c1f798@mail.gmail.com>
	<86a32abc0910171535l606eb9c6id54f7b3b5ae1b670@mail.gmail.com>
Message-ID: <86a32abc0910171545l17f3f539ldba432db550c3eaa@mail.gmail.com>

Instead of adding million of different domains for the languages, it would
be nice if we only keep wiki.freeswitch.org, and the wiki software provides
a list of languages to choose from.

Then when we click in one of the languages we have the wiki translated to
the desired language.

I believe if we go this way we will be more organized, and there is more
chances that things will get translated, and we wont be repeating too much
information here and there too.

Just a thought... I'm open to do this work to make it happen.

The different domains per language is a good idea too.

Diego

On Sat, Oct 17, 2009 at 10:35 PM, Diego Viola <diego.viola at gmail.com> wrote:

> We could also do this for a better documentation.
>
>
>    1. Pick a new wiki engine that does internalization/translation easier.
>    2. Set some documentation guidelines from the start.
>    3. Move stuff over to the new wiki.
>    4. Start with the translation/internalization on the new wiki.
>
>
> Diego
>
>
> On Sat, Oct 17, 2009 at 10:21 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> Some interesting pages on the subject.
>>
>> http://en.wikipedia.org/wiki/Comparison_of_wiki_software
>> http://www.osnews.com/story/21865/Recommend_an_Open_Source_Wiki_Platform
>>
>>
>>
>>
>> On Sat, Oct 17, 2009 at 10:13 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>
>>> Would you be open to test another wiki software, that supports
>>> internalization/translation easier, if i am open to port the entire
>>> documentation to the new wiki software?
>>>
>>>
>>>
>>>
>>>
>>> On Sat, Oct 17, 2009 at 8:39 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>
>>>> If we find a better wiki software that does internalization in a better
>>>> way, I could help porting the documentation over to the new software.
>>>>
>>>> The sub-domain idea is also very nice.
>>>>
>>>> Diego
>>>>
>>>>
>>>>
>>>> On Sat, Oct 17, 2009 at 8:29 PM, Michael Jerris <mike at jerris.com>wrote:
>>>>
>>>>> I think we strongly lean towards using a sub-domain (if necessary) and
>>>>> maintaining other language content in the same wiki in alternate
>>>>> pages.  If the wiki software we are using is not effective to create
>>>>> multiple languages we should find a way to do it all in one.  We can
>>>>> set up an additional mailing list if necessary but also we have no
>>>>> issue with going across multiple languages in both the mailing list
>>>>> and in irc.
>>>>>
>>>>> Mike
>>>>>
>>>>> On Oct 17, 2009, at 4:01 PM, Roberto Martins wrote:
>>>>>
>>>>> > sub-domain seem to be a very wise choice...
>>>>> > ptbr
>>>>> >
>>>>> >
>>>>> >
>>>>> > On Oct 17, 2009, at 1:25 PM, Anthony Minessale wrote:
>>>>> >
>>>>> >> or someone could just ask us for pg.freeswitch.org or whatever
>>>>> other
>>>>> >> sub-domain to avoid trademark infringement of registering 40 domains
>>>>> >> with freeswitch in the name for no reason.
>>>>> >>
>>>>> >>
>>>>> >> On Sat, Oct 17, 2009 at 6:06 AM, Rud? Cunha <ruda at ruda.com.br>
>>>>> wrote:
>>>>> >> Jo?o Mesquita,
>>>>> >>
>>>>> >> Could check with Jeremiah or contact me through, so I talk to him
>>>>> >> about the forum?
>>>>> >>
>>>>> >> He could redirect the domain to my hosting server and we can put
>>>>> >> there the wiki and forums and we can communicate!
>>>>> >>
>>>>> >> Rud? Cunha
>>>>> >>
>>>>> >> 2009/10/17 Jo?o Mesquita <jmesquita at freeswitch.org>
>>>>> >>
>>>>> >> Just a heads up, I have talked to Jeremias from Khomp today and he
>>>>> >> is setting up the wiki. I will personally be adding contents to the
>>>>> >> that wiki if it ever picks up.
>>>>> >>
>>>>> >> Regards,
>>>>> >>
>>>>> >> jm
>>>>> >>
>>>>> >>
>>>>> >> On Sat, Oct 17, 2009 at 12:52 AM, Jason White <jason at jasonjgw.net>
>>>>> >> wrote:
>>>>> >> Diego Viola <diego.viola at gmail.com> wrote:
>>>>> >>> I'm with Moises and with the other people supporting this
>>>>> >> initiative.
>>>>> >>>
>>>>> >>> I'm not Brazilian, but they should be able to do whatever they
>>>>> >> want, after
>>>>> >>> all, that's how open source works, if you can do it go ahead and
>>>>> >> do it.
>>>>> >>
>>>>> >> Correct. We have enough of them as far as English-language fora are
>>>>> >> concerned;
>>>>> >> other languages are a different question altogether, though.
>>>>> >>
>>>>> >>
>>>>> >> _______________________________________________
>>>>> >> FreeSWITCH-users mailing list
>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>> >> users
>>>>> >> http://www.freeswitch.org
>>>>> >>
>>>>> >>
>>>>> >> _______________________________________________
>>>>> >> FreeSWITCH-users mailing list
>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>> >> users
>>>>> >> http://www.freeswitch.org
>>>>> >>
>>>>> >>
>>>>> >>
>>>>> >> _______________________________________________
>>>>> >> FreeSWITCH-users mailing list
>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>> >> users
>>>>> >> http://www.freeswitch.org
>>>>> >>
>>>>> >>
>>>>> >>
>>>>> >>
>>>>> >> --
>>>>> >> Anthony Minessale II
>>>>> >>
>>>>> >> FreeSWITCH http://www.freeswitch.org/
>>>>> >> ClueCon http://www.cluecon.com/
>>>>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>> >>
>>>>> >> AIM: anthm
>>>>> >> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>> >> IRC: irc.freenode.net #freeswitch
>>>>> >>
>>>>> >> FreeSWITCH Developer Conference
>>>>> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>> >> iax:guest at conference.freeswitch.org/888
>>>>> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>> >> pstn:213-799-1400
>>>>> >> _______________________________________________
>>>>> >> FreeSWITCH-users mailing list
>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>> >> users
>>>>> >> http://www.freeswitch.org
>>>>> >
>>>>> >
>>>>> > _______________________________________________
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>> > users
>>>>> > http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>
>>
>
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From diego.viola at gmail.com  Sat Oct 17 16:01:13 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 17 Oct 2009 23:01:13 +0000
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <86a32abc0910171545l17f3f539ldba432db550c3eaa@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<7600794b0910170406m164e3e2od198586f53de5efc@mail.gmail.com>
	<191c3a030910170925i7aa7d4f3p9d382e0d855b7c8b@mail.gmail.com>
	<6701786D-A8AF-444E-A5A7-29E69F6E055C@gmail.com>
	<B69597BF-DA4A-440B-8CB0-08F52F33DD17@jerris.com>
	<86a32abc0910171339o6a300932v66b7f00419a0cb69@mail.gmail.com>
	<86a32abc0910171513n35d7d0echff940a16685ae1a4@mail.gmail.com>
	<86a32abc0910171521n403094bfvf27aa7c2a1c1f798@mail.gmail.com>
	<86a32abc0910171535l606eb9c6id54f7b3b5ae1b670@mail.gmail.com>
	<86a32abc0910171545l17f3f539ldba432db550c3eaa@mail.gmail.com>
Message-ID: <86a32abc0910171601u6ab06e42xcf303301c85c57b1@mail.gmail.com>

Take a look at this for instance:

http://wiki.rubyonrails.org/start

This uses DokuWiki, check the langauges option at the top (en, de, el, es,
fr, etc) and when you click one of them it translates the actual page to the
desired language... I'd love to have something like that.

Diego

On Sat, Oct 17, 2009 at 10:45 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Instead of adding million of different domains for the languages, it would
> be nice if we only keep wiki.freeswitch.org, and the wiki software
> provides a list of languages to choose from.
>
> Then when we click in one of the languages we have the wiki translated to
> the desired language.
>
> I believe if we go this way we will be more organized, and there is more
> chances that things will get translated, and we wont be repeating too much
> information here and there too.
>
> Just a thought... I'm open to do this work to make it happen.
>
> The different domains per language is a good idea too.
>
> Diego
>
>
> On Sat, Oct 17, 2009 at 10:35 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> We could also do this for a better documentation.
>>
>>
>>    1. Pick a new wiki engine that does internalization/translation
>>    easier.
>>    2. Set some documentation guidelines from the start.
>>    3. Move stuff over to the new wiki.
>>    4. Start with the translation/internalization on the new wiki.
>>
>>
>> Diego
>>
>>
>> On Sat, Oct 17, 2009 at 10:21 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>
>>> Some interesting pages on the subject.
>>>
>>> http://en.wikipedia.org/wiki/Comparison_of_wiki_software
>>> http://www.osnews.com/story/21865/Recommend_an_Open_Source_Wiki_Platform
>>>
>>>
>>>
>>>
>>> On Sat, Oct 17, 2009 at 10:13 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>
>>>> Would you be open to test another wiki software, that supports
>>>> internalization/translation easier, if i am open to port the entire
>>>> documentation to the new wiki software?
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Sat, Oct 17, 2009 at 8:39 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>>
>>>>> If we find a better wiki software that does internalization in a better
>>>>> way, I could help porting the documentation over to the new software.
>>>>>
>>>>> The sub-domain idea is also very nice.
>>>>>
>>>>> Diego
>>>>>
>>>>>
>>>>>
>>>>> On Sat, Oct 17, 2009 at 8:29 PM, Michael Jerris <mike at jerris.com>wrote:
>>>>>
>>>>>> I think we strongly lean towards using a sub-domain (if necessary) and
>>>>>> maintaining other language content in the same wiki in alternate
>>>>>> pages.  If the wiki software we are using is not effective to create
>>>>>> multiple languages we should find a way to do it all in one.  We can
>>>>>> set up an additional mailing list if necessary but also we have no
>>>>>> issue with going across multiple languages in both the mailing list
>>>>>> and in irc.
>>>>>>
>>>>>> Mike
>>>>>>
>>>>>> On Oct 17, 2009, at 4:01 PM, Roberto Martins wrote:
>>>>>>
>>>>>> > sub-domain seem to be a very wise choice...
>>>>>> > ptbr
>>>>>> >
>>>>>> >
>>>>>> >
>>>>>> > On Oct 17, 2009, at 1:25 PM, Anthony Minessale wrote:
>>>>>> >
>>>>>> >> or someone could just ask us for pg.freeswitch.org or whatever
>>>>>> other
>>>>>> >> sub-domain to avoid trademark infringement of registering 40
>>>>>> domains
>>>>>> >> with freeswitch in the name for no reason.
>>>>>> >>
>>>>>> >>
>>>>>> >> On Sat, Oct 17, 2009 at 6:06 AM, Rud? Cunha <ruda at ruda.com.br>
>>>>>> wrote:
>>>>>> >> Jo?o Mesquita,
>>>>>> >>
>>>>>> >> Could check with Jeremiah or contact me through, so I talk to him
>>>>>> >> about the forum?
>>>>>> >>
>>>>>> >> He could redirect the domain to my hosting server and we can put
>>>>>> >> there the wiki and forums and we can communicate!
>>>>>> >>
>>>>>> >> Rud? Cunha
>>>>>> >>
>>>>>> >> 2009/10/17 Jo?o Mesquita <jmesquita at freeswitch.org>
>>>>>> >>
>>>>>> >> Just a heads up, I have talked to Jeremias from Khomp today and he
>>>>>> >> is setting up the wiki. I will personally be adding contents to the
>>>>>> >> that wiki if it ever picks up.
>>>>>> >>
>>>>>> >> Regards,
>>>>>> >>
>>>>>> >> jm
>>>>>> >>
>>>>>> >>
>>>>>> >> On Sat, Oct 17, 2009 at 12:52 AM, Jason White <jason at jasonjgw.net>
>>>>>> >> wrote:
>>>>>> >> Diego Viola <diego.viola at gmail.com> wrote:
>>>>>> >>> I'm with Moises and with the other people supporting this
>>>>>> >> initiative.
>>>>>> >>>
>>>>>> >>> I'm not Brazilian, but they should be able to do whatever they
>>>>>> >> want, after
>>>>>> >>> all, that's how open source works, if you can do it go ahead and
>>>>>> >> do it.
>>>>>> >>
>>>>>> >> Correct. We have enough of them as far as English-language fora are
>>>>>> >> concerned;
>>>>>> >> other languages are a different question altogether, though.
>>>>>> >>
>>>>>> >>
>>>>>> >> _______________________________________________
>>>>>> >> FreeSWITCH-users mailing list
>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> >> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>> >> users
>>>>>> >> http://www.freeswitch.org
>>>>>> >>
>>>>>> >>
>>>>>> >> _______________________________________________
>>>>>> >> FreeSWITCH-users mailing list
>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> >> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>> >> users
>>>>>> >> http://www.freeswitch.org
>>>>>> >>
>>>>>> >>
>>>>>> >>
>>>>>> >> _______________________________________________
>>>>>> >> FreeSWITCH-users mailing list
>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> >> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>> >> users
>>>>>> >> http://www.freeswitch.org
>>>>>> >>
>>>>>> >>
>>>>>> >>
>>>>>> >>
>>>>>> >> --
>>>>>> >> Anthony Minessale II
>>>>>> >>
>>>>>> >> FreeSWITCH http://www.freeswitch.org/
>>>>>> >> ClueCon http://www.cluecon.com/
>>>>>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>> >>
>>>>>> >> AIM: anthm
>>>>>> >> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>> >> IRC: irc.freenode.net #freeswitch
>>>>>> >>
>>>>>> >> FreeSWITCH Developer Conference
>>>>>> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>> >> iax:guest at conference.freeswitch.org/888
>>>>>> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>> >> pstn:213-799-1400
>>>>>> >> _______________________________________________
>>>>>> >> FreeSWITCH-users mailing list
>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> >> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>> >> users
>>>>>> >> http://www.freeswitch.org
>>>>>> >
>>>>>> >
>>>>>> > _______________________________________________
>>>>>> > FreeSWITCH-users mailing list
>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>> > users
>>>>>> > http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>
>>>
>>
>
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From diego.viola at gmail.com  Sat Oct 17 16:05:33 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 17 Oct 2009 23:05:33 +0000
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <86a32abc0910171601u6ab06e42xcf303301c85c57b1@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<191c3a030910170925i7aa7d4f3p9d382e0d855b7c8b@mail.gmail.com>
	<6701786D-A8AF-444E-A5A7-29E69F6E055C@gmail.com>
	<B69597BF-DA4A-440B-8CB0-08F52F33DD17@jerris.com>
	<86a32abc0910171339o6a300932v66b7f00419a0cb69@mail.gmail.com>
	<86a32abc0910171513n35d7d0echff940a16685ae1a4@mail.gmail.com>
	<86a32abc0910171521n403094bfvf27aa7c2a1c1f798@mail.gmail.com>
	<86a32abc0910171535l606eb9c6id54f7b3b5ae1b670@mail.gmail.com>
	<86a32abc0910171545l17f3f539ldba432db550c3eaa@mail.gmail.com>
	<86a32abc0910171601u6ab06e42xcf303301c85c57b1@mail.gmail.com>
Message-ID: <86a32abc0910171605y73ecb75cqfb13923ad19adc56@mail.gmail.com>

With sub-domains you'll need to start all over and it will be a pain to keep
the documentation in-sync.

With a wiki that supports real translation it's easier to add a new line
with the translation tag or whatever.

Diego

On Sat, Oct 17, 2009 at 11:01 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Take a look at this for instance:
>
> http://wiki.rubyonrails.org/start
>
> This uses DokuWiki, check the langauges option at the top (en, de, el, es,
> fr, etc) and when you click one of them it translates the actual page to the
> desired language... I'd love to have something like that.
>
> Diego
>
>
> On Sat, Oct 17, 2009 at 10:45 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> Instead of adding million of different domains for the languages, it would
>> be nice if we only keep wiki.freeswitch.org, and the wiki software
>> provides a list of languages to choose from.
>>
>> Then when we click in one of the languages we have the wiki translated to
>> the desired language.
>>
>> I believe if we go this way we will be more organized, and there is more
>> chances that things will get translated, and we wont be repeating too much
>> information here and there too.
>>
>> Just a thought... I'm open to do this work to make it happen.
>>
>> The different domains per language is a good idea too.
>>
>> Diego
>>
>>
>> On Sat, Oct 17, 2009 at 10:35 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>
>>> We could also do this for a better documentation.
>>>
>>>
>>>    1. Pick a new wiki engine that does internalization/translation
>>>    easier.
>>>    2. Set some documentation guidelines from the start.
>>>    3. Move stuff over to the new wiki.
>>>    4. Start with the translation/internalization on the new wiki.
>>>
>>>
>>> Diego
>>>
>>>
>>> On Sat, Oct 17, 2009 at 10:21 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>
>>>> Some interesting pages on the subject.
>>>>
>>>> http://en.wikipedia.org/wiki/Comparison_of_wiki_software
>>>> http://www.osnews.com/story/21865/Recommend_an_Open_Source_Wiki_Platform
>>>>
>>>>
>>>>
>>>>
>>>> On Sat, Oct 17, 2009 at 10:13 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>>
>>>>> Would you be open to test another wiki software, that supports
>>>>> internalization/translation easier, if i am open to port the entire
>>>>> documentation to the new wiki software?
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Sat, Oct 17, 2009 at 8:39 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>>>
>>>>>> If we find a better wiki software that does internalization in a
>>>>>> better way, I could help porting the documentation over to the new software.
>>>>>>
>>>>>> The sub-domain idea is also very nice.
>>>>>>
>>>>>> Diego
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Sat, Oct 17, 2009 at 8:29 PM, Michael Jerris <mike at jerris.com>wrote:
>>>>>>
>>>>>>> I think we strongly lean towards using a sub-domain (if necessary)
>>>>>>> and
>>>>>>> maintaining other language content in the same wiki in alternate
>>>>>>> pages.  If the wiki software we are using is not effective to create
>>>>>>> multiple languages we should find a way to do it all in one.  We can
>>>>>>> set up an additional mailing list if necessary but also we have no
>>>>>>> issue with going across multiple languages in both the mailing list
>>>>>>> and in irc.
>>>>>>>
>>>>>>> Mike
>>>>>>>
>>>>>>> On Oct 17, 2009, at 4:01 PM, Roberto Martins wrote:
>>>>>>>
>>>>>>> > sub-domain seem to be a very wise choice...
>>>>>>> > ptbr
>>>>>>> >
>>>>>>> >
>>>>>>> >
>>>>>>> > On Oct 17, 2009, at 1:25 PM, Anthony Minessale wrote:
>>>>>>> >
>>>>>>> >> or someone could just ask us for pg.freeswitch.org or whatever
>>>>>>> other
>>>>>>> >> sub-domain to avoid trademark infringement of registering 40
>>>>>>> domains
>>>>>>> >> with freeswitch in the name for no reason.
>>>>>>> >>
>>>>>>> >>
>>>>>>> >> On Sat, Oct 17, 2009 at 6:06 AM, Rud? Cunha <ruda at ruda.com.br>
>>>>>>> wrote:
>>>>>>> >> Jo?o Mesquita,
>>>>>>> >>
>>>>>>> >> Could check with Jeremiah or contact me through, so I talk to him
>>>>>>> >> about the forum?
>>>>>>> >>
>>>>>>> >> He could redirect the domain to my hosting server and we can put
>>>>>>> >> there the wiki and forums and we can communicate!
>>>>>>> >>
>>>>>>> >> Rud? Cunha
>>>>>>> >>
>>>>>>> >> 2009/10/17 Jo?o Mesquita <jmesquita at freeswitch.org>
>>>>>>> >>
>>>>>>> >> Just a heads up, I have talked to Jeremias from Khomp today and he
>>>>>>> >> is setting up the wiki. I will personally be adding contents to
>>>>>>> the
>>>>>>> >> that wiki if it ever picks up.
>>>>>>> >>
>>>>>>> >> Regards,
>>>>>>> >>
>>>>>>> >> jm
>>>>>>> >>
>>>>>>> >>
>>>>>>> >> On Sat, Oct 17, 2009 at 12:52 AM, Jason White <jason at jasonjgw.net
>>>>>>> >
>>>>>>> >> wrote:
>>>>>>> >> Diego Viola <diego.viola at gmail.com> wrote:
>>>>>>> >>> I'm with Moises and with the other people supporting this
>>>>>>> >> initiative.
>>>>>>> >>>
>>>>>>> >>> I'm not Brazilian, but they should be able to do whatever they
>>>>>>> >> want, after
>>>>>>> >>> all, that's how open source works, if you can do it go ahead and
>>>>>>> >> do it.
>>>>>>> >>
>>>>>>> >> Correct. We have enough of them as far as English-language fora
>>>>>>> are
>>>>>>> >> concerned;
>>>>>>> >> other languages are a different question altogether, though.
>>>>>>> >>
>>>>>>> >>
>>>>>>> >> _______________________________________________
>>>>>>> >> FreeSWITCH-users mailing list
>>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>>> >> users
>>>>>>> >> http://www.freeswitch.org
>>>>>>> >>
>>>>>>> >>
>>>>>>> >> _______________________________________________
>>>>>>> >> FreeSWITCH-users mailing list
>>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>>> >> users
>>>>>>> >> http://www.freeswitch.org
>>>>>>> >>
>>>>>>> >>
>>>>>>> >>
>>>>>>> >> _______________________________________________
>>>>>>> >> FreeSWITCH-users mailing list
>>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>>> >> users
>>>>>>> >> http://www.freeswitch.org
>>>>>>> >>
>>>>>>> >>
>>>>>>> >>
>>>>>>> >>
>>>>>>> >> --
>>>>>>> >> Anthony Minessale II
>>>>>>> >>
>>>>>>> >> FreeSWITCH http://www.freeswitch.org/
>>>>>>> >> ClueCon http://www.cluecon.com/
>>>>>>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>> >>
>>>>>>> >> AIM: anthm
>>>>>>> >> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>>> >> IRC: irc.freenode.net #freeswitch
>>>>>>> >>
>>>>>>> >> FreeSWITCH Developer Conference
>>>>>>> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>>> >> iax:guest at conference.freeswitch.org/888
>>>>>>> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>>> >> pstn:213-799-1400
>>>>>>> >> _______________________________________________
>>>>>>> >> FreeSWITCH-users mailing list
>>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>>> >> users
>>>>>>> >> http://www.freeswitch.org
>>>>>>> >
>>>>>>> >
>>>>>>> > _______________________________________________
>>>>>>> > FreeSWITCH-users mailing list
>>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> > UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>>> > users
>>>>>>> > http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>
>>>
>>
>
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From mcampbellsmith at gmail.com  Sat Oct 17 17:10:01 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sun, 18 Oct 2009 11:10:01 +1100
Subject: [Freeswitch-users] Dingaling: using a hostname instead of stun
	for rtp
In-Reply-To: <7710644E-CEA1-4D48-8A03-EEE1C18F1B44@jerris.com>
References: <33c87fa30910131502t2af24801i1c3319518b5db50b@mail.gmail.com>
	<C874E25D-6D73-49AA-8DD3-0A0022617FD2@freeswitch.org>
	<33c87fa30910152035w43e260faoc031a7a4ac949d9e@mail.gmail.com>
	<65BF398D-E3A6-4A98-83D9-D4E271070CCA@freeswitch.org>
	<7710644E-CEA1-4D48-8A03-EEE1C18F1B44@jerris.com>
Message-ID: <33c87fa30910171710t7b9038cap5dcf13259486ec13@mail.gmail.com>

Thanks for the response Mike and Brian.

Using stun is not a problem and I have it working okay now.  Is there
something different in the implementation of the stun lookup between
dingaling and sofia?

I normally use stun.freeswitch.org and sofia never had a problem.

It doesn't matter which stun server I use with dingaling (I have tried
with numerous servers) and the same thing seems to happen.  The first
stun lookup fails with a timeout (on the first inbound call), and then
if another dingaling call is tried directly, the second stun lookup
works.

I noticed a similar behaviour by issuing the stun command from fs_cli

Thanks

On Sun, Oct 18, 2009 at 8:04 AM, Michael Jerris <mike at jerris.com> wrote:
> If you don't have working stun, jingle is not going to work very
> well. ?It is a required part of the protocol. ?You need to be able to
> determine your external ports for media on each call, using a host
> name will not do this for you.
>
> Mike
>
>
> On Oct 16, 2009, at 10:48 AM, Brian West wrote:
>
>> If you setup your own stun server it wouldn't do that.... ?But the
>> hostlookup only solves half the problem .. getting the external IP vs
>> poking holes for RTP which is what stun will do.
>>
>> /b
>>
>> On Oct 15, 2009, at 10:35 PM, Mark Campbell-Smith wrote:
>>
>>> Thanks Brian. ?Is this something that is planned to be implemented?
>>> The workaround is to set the stun server also in the dingaling
>>> configuration, but as I said, for some reason the stun times for me
>>> out occasionally with dingaling.
>>>
>>> Thanks!
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From diego.viola at gmail.com  Sat Oct 17 17:11:40 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 18 Oct 2009 00:11:40 +0000
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <86a32abc0910171605y73ecb75cqfb13923ad19adc56@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<6701786D-A8AF-444E-A5A7-29E69F6E055C@gmail.com>
	<B69597BF-DA4A-440B-8CB0-08F52F33DD17@jerris.com>
	<86a32abc0910171339o6a300932v66b7f00419a0cb69@mail.gmail.com>
	<86a32abc0910171513n35d7d0echff940a16685ae1a4@mail.gmail.com>
	<86a32abc0910171521n403094bfvf27aa7c2a1c1f798@mail.gmail.com>
	<86a32abc0910171535l606eb9c6id54f7b3b5ae1b670@mail.gmail.com>
	<86a32abc0910171545l17f3f539ldba432db550c3eaa@mail.gmail.com>
	<86a32abc0910171601u6ab06e42xcf303301c85c57b1@mail.gmail.com>
	<86a32abc0910171605y73ecb75cqfb13923ad19adc56@mail.gmail.com>
Message-ID: <86a32abc0910171711je1e57bduf2bb672472365ce1@mail.gmail.com>

http://moinmo.in/ is also pretty cool it seems and it's used for the apache
wiki and others.

http://wiki.apache.org/

It also has i18n support, full text search, etc.

Diego

On Sat, Oct 17, 2009 at 11:05 PM, Diego Viola <diego.viola at gmail.com> wrote:

> With sub-domains you'll need to start all over and it will be a pain to
> keep the documentation in-sync.
>
> With a wiki that supports real translation it's easier to add a new line
> with the translation tag or whatever.
>
> Diego
>
>
> On Sat, Oct 17, 2009 at 11:01 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> Take a look at this for instance:
>>
>> http://wiki.rubyonrails.org/start
>>
>> This uses DokuWiki, check the langauges option at the top (en, de, el, es,
>> fr, etc) and when you click one of them it translates the actual page to the
>> desired language... I'd love to have something like that.
>>
>> Diego
>>
>>
>> On Sat, Oct 17, 2009 at 10:45 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>
>>> Instead of adding million of different domains for the languages, it
>>> would be nice if we only keep wiki.freeswitch.org, and the wiki software
>>> provides a list of languages to choose from.
>>>
>>> Then when we click in one of the languages we have the wiki translated to
>>> the desired language.
>>>
>>> I believe if we go this way we will be more organized, and there is more
>>> chances that things will get translated, and we wont be repeating too much
>>> information here and there too.
>>>
>>> Just a thought... I'm open to do this work to make it happen.
>>>
>>> The different domains per language is a good idea too.
>>>
>>> Diego
>>>
>>>
>>> On Sat, Oct 17, 2009 at 10:35 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>
>>>> We could also do this for a better documentation.
>>>>
>>>>
>>>>    1. Pick a new wiki engine that does internalization/translation
>>>>    easier.
>>>>    2. Set some documentation guidelines from the start.
>>>>    3. Move stuff over to the new wiki.
>>>>    4. Start with the translation/internalization on the new wiki.
>>>>
>>>>
>>>> Diego
>>>>
>>>>
>>>> On Sat, Oct 17, 2009 at 10:21 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>>
>>>>> Some interesting pages on the subject.
>>>>>
>>>>> http://en.wikipedia.org/wiki/Comparison_of_wiki_software
>>>>>
>>>>> http://www.osnews.com/story/21865/Recommend_an_Open_Source_Wiki_Platform
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Sat, Oct 17, 2009 at 10:13 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>>>
>>>>>> Would you be open to test another wiki software, that supports
>>>>>> internalization/translation easier, if i am open to port the entire
>>>>>> documentation to the new wiki software?
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Sat, Oct 17, 2009 at 8:39 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>>>>
>>>>>>> If we find a better wiki software that does internalization in a
>>>>>>> better way, I could help porting the documentation over to the new software.
>>>>>>>
>>>>>>> The sub-domain idea is also very nice.
>>>>>>>
>>>>>>> Diego
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Sat, Oct 17, 2009 at 8:29 PM, Michael Jerris <mike at jerris.com>wrote:
>>>>>>>
>>>>>>>> I think we strongly lean towards using a sub-domain (if necessary)
>>>>>>>> and
>>>>>>>> maintaining other language content in the same wiki in alternate
>>>>>>>> pages.  If the wiki software we are using is not effective to create
>>>>>>>> multiple languages we should find a way to do it all in one.  We can
>>>>>>>> set up an additional mailing list if necessary but also we have no
>>>>>>>> issue with going across multiple languages in both the mailing list
>>>>>>>> and in irc.
>>>>>>>>
>>>>>>>> Mike
>>>>>>>>
>>>>>>>> On Oct 17, 2009, at 4:01 PM, Roberto Martins wrote:
>>>>>>>>
>>>>>>>> > sub-domain seem to be a very wise choice...
>>>>>>>> > ptbr
>>>>>>>> >
>>>>>>>> >
>>>>>>>> >
>>>>>>>> > On Oct 17, 2009, at 1:25 PM, Anthony Minessale wrote:
>>>>>>>> >
>>>>>>>> >> or someone could just ask us for pg.freeswitch.org or whatever
>>>>>>>> other
>>>>>>>> >> sub-domain to avoid trademark infringement of registering 40
>>>>>>>> domains
>>>>>>>> >> with freeswitch in the name for no reason.
>>>>>>>> >>
>>>>>>>> >>
>>>>>>>> >> On Sat, Oct 17, 2009 at 6:06 AM, Rud? Cunha <ruda at ruda.com.br>
>>>>>>>> wrote:
>>>>>>>> >> Jo?o Mesquita,
>>>>>>>> >>
>>>>>>>> >> Could check with Jeremiah or contact me through, so I talk to him
>>>>>>>> >> about the forum?
>>>>>>>> >>
>>>>>>>> >> He could redirect the domain to my hosting server and we can put
>>>>>>>> >> there the wiki and forums and we can communicate!
>>>>>>>> >>
>>>>>>>> >> Rud? Cunha
>>>>>>>> >>
>>>>>>>> >> 2009/10/17 Jo?o Mesquita <jmesquita at freeswitch.org>
>>>>>>>> >>
>>>>>>>> >> Just a heads up, I have talked to Jeremias from Khomp today and
>>>>>>>> he
>>>>>>>> >> is setting up the wiki. I will personally be adding contents to
>>>>>>>> the
>>>>>>>> >> that wiki if it ever picks up.
>>>>>>>> >>
>>>>>>>> >> Regards,
>>>>>>>> >>
>>>>>>>> >> jm
>>>>>>>> >>
>>>>>>>> >>
>>>>>>>> >> On Sat, Oct 17, 2009 at 12:52 AM, Jason White <
>>>>>>>> jason at jasonjgw.net>
>>>>>>>> >> wrote:
>>>>>>>> >> Diego Viola <diego.viola at gmail.com> wrote:
>>>>>>>> >>> I'm with Moises and with the other people supporting this
>>>>>>>> >> initiative.
>>>>>>>> >>>
>>>>>>>> >>> I'm not Brazilian, but they should be able to do whatever they
>>>>>>>> >> want, after
>>>>>>>> >>> all, that's how open source works, if you can do it go ahead and
>>>>>>>> >> do it.
>>>>>>>> >>
>>>>>>>> >> Correct. We have enough of them as far as English-language fora
>>>>>>>> are
>>>>>>>> >> concerned;
>>>>>>>> >> other languages are a different question altogether, though.
>>>>>>>> >>
>>>>>>>> >>
>>>>>>>> >> _______________________________________________
>>>>>>>> >> FreeSWITCH-users mailing list
>>>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>>>> >> users
>>>>>>>> >> http://www.freeswitch.org
>>>>>>>> >>
>>>>>>>> >>
>>>>>>>> >> _______________________________________________
>>>>>>>> >> FreeSWITCH-users mailing list
>>>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>>>> >> users
>>>>>>>> >> http://www.freeswitch.org
>>>>>>>> >>
>>>>>>>> >>
>>>>>>>> >>
>>>>>>>> >> _______________________________________________
>>>>>>>> >> FreeSWITCH-users mailing list
>>>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>>>> >> users
>>>>>>>> >> http://www.freeswitch.org
>>>>>>>> >>
>>>>>>>> >>
>>>>>>>> >>
>>>>>>>> >>
>>>>>>>> >> --
>>>>>>>> >> Anthony Minessale II
>>>>>>>> >>
>>>>>>>> >> FreeSWITCH http://www.freeswitch.org/
>>>>>>>> >> ClueCon http://www.cluecon.com/
>>>>>>>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>>> >>
>>>>>>>> >> AIM: anthm
>>>>>>>> >> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>>>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>>>> >> IRC: irc.freenode.net #freeswitch
>>>>>>>> >>
>>>>>>>> >> FreeSWITCH Developer Conference
>>>>>>>> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>>>> >> iax:guest at conference.freeswitch.org/888
>>>>>>>> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>>>> >> pstn:213-799-1400
>>>>>>>> >> _______________________________________________
>>>>>>>> >> FreeSWITCH-users mailing list
>>>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>>>> >> users
>>>>>>>> >> http://www.freeswitch.org
>>>>>>>> >
>>>>>>>> >
>>>>>>>> > _______________________________________________
>>>>>>>> > FreeSWITCH-users mailing list
>>>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> > UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>>>> > users
>>>>>>>> > http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>
>>>
>>
>
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From wiltingtree at gmail.com  Sat Oct 17 17:13:59 2009
From: wiltingtree at gmail.com (Adam Wilt)
Date: Sat, 17 Oct 2009 20:13:59 -0400
Subject: [Freeswitch-users] Problem connecting to Spidermonkey ODBC
Message-ID: <eb7797d70910171713wc3a064fi6231745381bb2d1b@mail.gmail.com>

Hi, I know there are a lot of answered questions about this already, but I
think I read all of the answers and I'm still stuck.
* I'm using FreeSWITCH Version 1.0.trunk (14961).
* I installed UnixODBC
* My odbc.ini and odbcinst.ini files are in /usr/local/etc/
* I have symlinks to /usr/local/etc from /usr/etc and from
/usr/local/freeswitch/etc/
* Im running freeswitch as root, and I copied odbc.ini to /root/.odbc.ini
* I have mod_spidermonkey_odbc uncommented in spidermonkey.conf.xml
* When I run "isql dbname" from the command prompt it works fine.

But when I try to run my script, I get:

2009-10-17 23:56:21.848930 [ERR] switch_odbc.c:188 STATE: IM002 CODE 0
ERROR: [unixODBC][Driver Manager]Data source name not found, and no default
driver specified

2009-10-17 23:56:21.848930 [ERR] mod_spidermonkey_odbc.c:245 Database is not
connected!


Please help!
Thanks.
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From brian at freeswitch.org  Sat Oct 17 17:56:36 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 17 Oct 2009 19:56:36 -0500
Subject: [Freeswitch-users] Problem connecting to Spidermonkey ODBC
In-Reply-To: <eb7797d70910171713wc3a064fi6231745381bb2d1b@mail.gmail.com>
References: <eb7797d70910171713wc3a064fi6231745381bb2d1b@mail.gmail.com>
Message-ID: <3BEC6702-4754-41BB-B458-835250ECD897@freeswitch.org>

How are you using this from your javascript code?

/b

On Oct 17, 2009, at 7:13 PM, Adam Wilt wrote:

>
> 2009-10-17 23:56:21.848930 [ERR] switch_odbc.c:188 STATE: IM002 CODE  
> 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and  
> no default driver specified
>
> 2009-10-17 23:56:21.848930 [ERR] mod_spidermonkey_odbc.c:245  
> Database is not connected!
>



From brian at freeswitch.org  Sat Oct 17 18:01:09 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 17 Oct 2009 20:01:09 -0500
Subject: [Freeswitch-users] Dingaling: using a hostname instead of stun
	for rtp
In-Reply-To: <33c87fa30910171710t7b9038cap5dcf13259486ec13@mail.gmail.com>
References: <33c87fa30910131502t2af24801i1c3319518b5db50b@mail.gmail.com>
	<C874E25D-6D73-49AA-8DD3-0A0022617FD2@freeswitch.org>
	<33c87fa30910152035w43e260faoc031a7a4ac949d9e@mail.gmail.com>
	<65BF398D-E3A6-4A98-83D9-D4E271070CCA@freeswitch.org>
	<7710644E-CEA1-4D48-8A03-EEE1C18F1B44@jerris.com>
	<33c87fa30910171710t7b9038cap5dcf13259486ec13@mail.gmail.com>
Message-ID: <9B8E1ADF-B0EB-495F-8D27-A9D1211A4D2C@freeswitch.org>

They all use the same core API.  I suspect since our stun server is  
just a cname to many public services out there you're just thinking  
its dingaling related but its really not...

/b

On Oct 17, 2009, at 7:10 PM, Mark Campbell-Smith wrote:

> I noticed a similar behaviour by issuing the stun command from fs_cli
>
> Thanks



From wiltingtree at gmail.com  Sat Oct 17 18:11:27 2009
From: wiltingtree at gmail.com (Adam Wilt)
Date: Sat, 17 Oct 2009 21:11:27 -0400
Subject: [Freeswitch-users] Problem connecting to Spidermonkey ODBC
In-Reply-To: <3BEC6702-4754-41BB-B458-835250ECD897@freeswitch.org>
References: <eb7797d70910171713wc3a064fi6231745381bb2d1b@mail.gmail.com>
	<3BEC6702-4754-41BB-B458-835250ECD897@freeswitch.org>
Message-ID: <eb7797d70910171811y57f2d29q2360de42c0ad3c81@mail.gmail.com>

Brian,

Here's how I'm using the ODBC in my .js file:

use("ODBC");

var DSN="mydsn";
var DB_USER="myuser";
var DB_PASS="mypass";
.
.
.
var db = new ODBC(DSN, DB_USER, DB_PASS);
db.connect();
.
.
.

Thanks.


On Sat, Oct 17, 2009 at 8:56 PM, Brian West <brian at freeswitch.org> wrote:

> How are you using this from your javascript code?
>
> /b
>
> On Oct 17, 2009, at 7:13 PM, Adam Wilt wrote:
>
> >
> > 2009-10-17 23:56:21.848930 [ERR] switch_odbc.c:188 STATE: IM002 CODE
> > 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and
> > no default driver specified
> >
> > 2009-10-17 23:56:21.848930 [ERR] mod_spidermonkey_odbc.c:245
> > Database is not connected!
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mcampbellsmith at gmail.com  Sat Oct 17 20:07:50 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sun, 18 Oct 2009 14:07:50 +1100
Subject: [Freeswitch-users] Dingaling / Jingle DTMF support?
In-Reply-To: <191c3a030910161456y65d49120p8479c7c5df675125@mail.gmail.com>
References: <33c87fa30910151519x637618d3p109f1b8edbf3432e@mail.gmail.com>
	<191c3a030910161456y65d49120p8479c7c5df675125@mail.gmail.com>
Message-ID: <33c87fa30910172007p57eec112qb03a9f31c763b139@mail.gmail.com>

Perfect... thanks for the + tip with googletalk...   I had been using
gtalk2voip but didn't want to rely on another service

Wki for dingaling updated.


On Sat, Oct 17, 2009 at 8:56 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> It already should support at least recv of rfc2833 any time.
> as a workaround in googletalk any string that starts with + typed in the
> chat box is treated as dtmf by FS
> e.g. +1
>
> Once the jingle spec stops being a moving target we will re-investigate
> making sure its supported properly.
>
>
> On Thu, Oct 15, 2009 at 5:19 PM, Mark Campbell-Smith
> <mcampbellsmith at gmail.com> wrote:
>>
>> Hi!
>>
>> I was wondering if the Dingaling implementation in FS supports DTMF?
>> This is now supported in the jingle specs (
>> http://www.jabberforum.org/showthread.php?t=2709 ), even though Google
>> Talk client does not currently support DTMF.
>>
>> If not, are there plans to implement this?
>>
>> Thanks!
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From wiltingtree at gmail.com  Sat Oct 17 20:16:05 2009
From: wiltingtree at gmail.com (Adam Wilt)
Date: Sat, 17 Oct 2009 23:16:05 -0400
Subject: [Freeswitch-users] Problem connecting to Spidermonkey ODBC
In-Reply-To: <eb7797d70910171811y57f2d29q2360de42c0ad3c81@mail.gmail.com>
References: <eb7797d70910171713wc3a064fi6231745381bb2d1b@mail.gmail.com>
	<3BEC6702-4754-41BB-B458-835250ECD897@freeswitch.org>
	<eb7797d70910171811y57f2d29q2360de42c0ad3c81@mail.gmail.com>
Message-ID: <eb7797d70910172016i37c6c1c0tfe2464c60e06d62d@mail.gmail.com>

So, where is FreeSWITCH looking for the odbc.ini file?
I keep reading conflicting information, including
/usr/local/freeswitch/etc/,  /usr/local/etc,  /usr/etc,  /home/username/



On Sat, Oct 17, 2009 at 9:11 PM, Adam Wilt <wiltingtree at gmail.com> wrote:

> Brian,
>
> Here's how I'm using the ODBC in my .js file:
>
> use("ODBC");
>
> var DSN="mydsn";
> var DB_USER="myuser";
> var DB_PASS="mypass";
> .
> .
> .
> var db = new ODBC(DSN, DB_USER, DB_PASS);
> db.connect();
> .
> .
> .
>
> Thanks.
>
>
> On Sat, Oct 17, 2009 at 8:56 PM, Brian West <brian at freeswitch.org> wrote:
>
>> How are you using this from your javascript code?
>>
>> /b
>>
>> On Oct 17, 2009, at 7:13 PM, Adam Wilt wrote:
>>
>> >
>> > 2009-10-17 23:56:21.848930 [ERR] switch_odbc.c:188 STATE: IM002 CODE
>> > 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and
>> > no default driver specified
>> >
>> > 2009-10-17 23:56:21.848930 [ERR] mod_spidermonkey_odbc.c:245
>> > Database is not connected!
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From brian at freeswitch.org  Sat Oct 17 20:26:58 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 17 Oct 2009 22:26:58 -0500
Subject: [Freeswitch-users] Problem connecting to Spidermonkey ODBC
In-Reply-To: <eb7797d70910172016i37c6c1c0tfe2464c60e06d62d@mail.gmail.com>
References: <eb7797d70910171713wc3a064fi6231745381bb2d1b@mail.gmail.com>
	<3BEC6702-4754-41BB-B458-835250ECD897@freeswitch.org>
	<eb7797d70910171811y57f2d29q2360de42c0ad3c81@mail.gmail.com>
	<eb7797d70910172016i37c6c1c0tfe2464c60e06d62d@mail.gmail.com>
Message-ID: <4CAB8EB3-6D37-448D-8B33-59C84AB685C3@freeswitch.org>

Its in /etc/ or where ever your default is.. is FS running as root?

/b

On Oct 17, 2009, at 10:16 PM, Adam Wilt wrote:

> So, where is FreeSWITCH looking for the odbc.ini file?
> I keep reading conflicting information, including /usr/local/ 
> freeswitch/etc/,  /usr/local/etc,  /usr/etc,  /home/username/
>
>



From wiltingtree at gmail.com  Sat Oct 17 20:38:45 2009
From: wiltingtree at gmail.com (Adam Wilt)
Date: Sat, 17 Oct 2009 23:38:45 -0400
Subject: [Freeswitch-users] Problem connecting to Spidermonkey ODBC
In-Reply-To: <4CAB8EB3-6D37-448D-8B33-59C84AB685C3@freeswitch.org>
References: <eb7797d70910171713wc3a064fi6231745381bb2d1b@mail.gmail.com>
	<3BEC6702-4754-41BB-B458-835250ECD897@freeswitch.org>
	<eb7797d70910171811y57f2d29q2360de42c0ad3c81@mail.gmail.com>
	<eb7797d70910172016i37c6c1c0tfe2464c60e06d62d@mail.gmail.com>
	<4CAB8EB3-6D37-448D-8B33-59C84AB685C3@freeswitch.org>
Message-ID: <eb7797d70910172038v4e1f7da2ge36679974f010cac@mail.gmail.com>

Yes, I have FreeSWITCH running as root.  So the odbc.ini file should be in
/etc/ ?
How do you set the default?


On Sat, Oct 17, 2009 at 11:26 PM, Brian West <brian at freeswitch.org> wrote:

> Its in /etc/ or where ever your default is.. is FS running as root?
>
> /b
>
> On Oct 17, 2009, at 10:16 PM, Adam Wilt wrote:
>
> > So, where is FreeSWITCH looking for the odbc.ini file?
> > I keep reading conflicting information, including /usr/local/
> > freeswitch/etc/,  /usr/local/etc,  /usr/etc,  /home/username/
> >
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From wiltingtree at gmail.com  Sat Oct 17 20:46:04 2009
From: wiltingtree at gmail.com (Adam Wilt)
Date: Sat, 17 Oct 2009 23:46:04 -0400
Subject: [Freeswitch-users] Problem connecting to Spidermonkey ODBC
In-Reply-To: <4CAB8EB3-6D37-448D-8B33-59C84AB685C3@freeswitch.org>
References: <eb7797d70910171713wc3a064fi6231745381bb2d1b@mail.gmail.com>
	<3BEC6702-4754-41BB-B458-835250ECD897@freeswitch.org>
	<eb7797d70910171811y57f2d29q2360de42c0ad3c81@mail.gmail.com>
	<eb7797d70910172016i37c6c1c0tfe2464c60e06d62d@mail.gmail.com>
	<4CAB8EB3-6D37-448D-8B33-59C84AB685C3@freeswitch.org>
Message-ID: <eb7797d70910172046u6b2435dcqd2fc833bdc8a9b29@mail.gmail.com>

Brian,
Putting the odbc.ini and odbcinst.ini in /etc/ fixed the problem.  Thanks.

Thanks,
Marc


On Sat, Oct 17, 2009 at 11:26 PM, Brian West <brian at freeswitch.org> wrote:

> Its in /etc/ or where ever your default is.. is FS running as root?
>
> /b
>
> On Oct 17, 2009, at 10:16 PM, Adam Wilt wrote:
>
> > So, where is FreeSWITCH looking for the odbc.ini file?
> > I keep reading conflicting information, including /usr/local/
> > freeswitch/etc/,  /usr/local/etc,  /usr/etc,  /home/username/
> >
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From max.bridgewater at gmail.com  Sat Oct 17 20:47:55 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sat, 17 Oct 2009 23:47:55 -0400
Subject: [Freeswitch-users] Detecting Forwarding Device Caller ID (Off Topic)
Message-ID: <c55d46950910172047y6de63448h656a1d889077b8f5@mail.gmail.com>

Hey Guys,

Pleas help me answer the following question. Sorry for being a bit
off-topic.  I have a GSM-VoiP modem with just 4 SIM Cards. Is there a known
mechanism that allows me to use this modem for more than 4 (albeit not at
the same time) GSM numbers?  Let's say the phone numbers of the SIM cards
inserted in the modem are call real phone numbers. My goal is to have some
10 additional virtual phone numbers served by this very same modem.

One way i thought this could be done is by forwarding the virtual phone
numbers the 4 real phone numbers. Then, when somebody calls one of the
virtual numbers, the Telco will forward it to the real phone number and it
will end on Freeswitch box. My only problem is now that on the Freeswitch
box I have no idea what virtual phone number was actually dialed. Any idea
how to recover this phone number? I looked at the GSM specification and
couldn't find a way.

PS. Purchasing DIDs won't solve my underlying problem.

Thanks,
Max.
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From dmitry.bely at gmail.com  Sun Oct 18 02:49:33 2009
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Sun, 18 Oct 2009 13:49:33 +0400
Subject: [Freeswitch-users] Status of ubuntu/debian packages.
In-Reply-To: <4ABA47FB.2050100@gmail.com>
References: <4ABA47FB.2050100@gmail.com>
Message-ID: <90823c940910180249u69f18aaamc383f9a5a0c65848@mail.gmail.com>

William,

When I upgrade freeswitch-config package, all my customized config
files are silently overwritten with the default ones from .deb
package. This bug is quite annoying.

- Dmitry Bely


From mattdfong at gmail.com  Sun Oct 18 07:25:10 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Sun, 18 Oct 2009 21:25:10 +0700
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
In-Reply-To: <3D0B855A-5B2C-4C6F-9306-64B0ECE97091@jerris.com>
References: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
	<177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>
	<4256bf830908041151l55980e3fifb85c75b87535426@mail.gmail.com>
	<3D0B855A-5B2C-4C6F-9306-64B0ECE97091@jerris.com>
Message-ID: <4256bf830910180725y72539d98gb138934afc18d58b@mail.gmail.com>

I'm still having this issue with random EXCHANGE_ROUTING_ERROR disconnects
for users utilizing my inbound DID to connect to my FreeSWITCH server. It's
a predictive dialing application, with one agent session being bridged with
multiple calls and transfered back and forth between extensions in my dial
plan. After a random number of bridging and transferring, FreeSWITCH
suddenly sends a BYE to my DID provider citing an EXCHANGE_ROUTING_ERROR. It
does not happen at any one-point in my dial plan, or applications--it just
randomly disconnects when a call that the Agent is bridged to hangs-up or is
disconnected. It seems to only happen when two external sip profiles are
being bridged together, and not when an internal and external profile is
being bridged.
I turned

sip trace on and
sofia loglevel all 9

below is the the snippet. I've posted the entire Agent session at the
following pastebin http://pastebin.freeswitch.org/10756

tport_deliver(0x18413c0): msg 0x7faeb818ea30 (304 bytes) from udp/
208.76.18.254:5080/sip next=(nil)
nta: received 200 OK for BYE (121818983)
nta: 200 OK is going to a transaction
nta_outgoing: RTT is 84.409 ms
tport_release(0x18413c0): 0x1a15cc0 by 0x1a16a00 with 0x7faeb818ea30
nua(0x1ad6fb0): event r_bye 200 OK
nua(0x1ad6fb0): call state changed: terminating -> terminated
nua(0x1ad6fb0): event i_state 200 to BYE
nua: nua_application_event: entering
nua(0x1ad6fb0): event i_terminated 200 to BYE
nua: nua_handle_magic: entering
nua(0x1ad6fb0): removing session usage
soa_destroy(static::0x1b5ae90) called
nua: nua_application_event: entering
nta_leg_destroy(0x1b594a0)
nua: nua_handle_magic: entering
nua: nua_handle_bind: entering
nua: nua_application_event: entering
nua: nua_handle_magic: entering
nua: terminated session 0x1ad6fb0
nua: nua_handle_destroy: entering
nua(0x1ad6fb0): recv signal r_destroy
nta_leg_destroy((nil))
nua(0x1ad6fb0): sent signal r_destroy
nta: timer set next to 28 ms
nta: timer E fired, retransmit BYE (121818989)
tport_release(0x18413c0): 0x1b5c9b0 by 0x7faeb817d830 with (nil)
tport_tsend(0x18413c0) tpn = */209.216.2.211:5060
tport_resolve addrinfo = 209.216.2.211:5060
tport_by_addrinfo(0x18413c0): not found by name */209.216.2.211:5060
tport_vsend(0x18413c0): 862 bytes of 862 to udp/209.216.2.211:5060
tport_vsend returned 862
send 862 bytes to udp/[209.216.2.211]:5060 at 14:04:11.753690:
   ------------------------------------------------------------------------
   BYE sip:199.173.100.16:5060;transport=UDP SIP/2.0
   Via: SIP/2.0/UDP 67.220.216.146:5080;rport;branch=z9hG4bK02jNX8a4HrNyN
   Route: <sip:209.216.2.211;ftag=4adc7-13c4-1ab03-71ce3705-1ab03;lr=on>
   Route: <sip:65.211.120.237:5060;lr>
   Route: <sip:63.110.102.239;lr>
   Max-Forwards: 70
   From: <sip:+12133304391 at 63.110.102.239:5060;user=phone>;tag=cgBe054jZrt3a
   To: <sip:+14158867717 at 199.173.100.16:5060
;user=phone>;tag=4adc7-13c4-1ab03-71ce3705-1ab03
   Call-ID: a0f656a01064adc713c41ab036840746ee20ca11c06b2d8-0440-5124
   CSeq: 121818989 BYE
   Contact: <sip:+12133304391 at 67.220.216.146:5080;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15135
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER,
UPDATE, NOTIFY
   Supported: timer, precondition, path, replaces
   Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
   Content-Length: 0

   ------------------------------------------------------------------------

Thanks.
--matt

On Sat, Aug 8, 2009 at 12:42 PM, Michael Jerris <mike at jerris.com> wrote:

> http://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP
> turn the logging all the way up and see what it says.
>
> Mike
>
> On Aug 4, 2009, at 2:51 PM, Matthew Fong wrote:
>
> Hi Mathieu, thanks for the reply. I enabled sip trace logging and got the
> logs below, but I am still at a loss at being able to identify the error or
> reproduce it consistently. The below log indicates to me that my FS server
> is initiating sending 2 BYE message to my DID provider (didforsale.com).
> Is there a way I can look further inside FreeSWITCH to see why it is sending
> this BYE packet?
>
> sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:44.679208:
> BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0
> Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD
> Route: <sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on>
> Route: <sip:65.217.40.210:5060;lr>
> Route: <sip:63.110.102.238;lr>
> Max-Forwards: 70
> From: <sip:+1212381XXXX at 63.110.102.238:5060;user=phone>;tag=Ztr5ycrv3QZ1g
> To: <sip:+1909635XXXX at 199.173.100.144:5060
> ;user=phone>;tag=dc7-13c4-2401b7-46dea593-2401b7
> Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441
> CSeq: 118584736 BYE
> Contact: <sip:+1212381XXXX at 66.197.142.69:5080;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO
> Supported: timer, precondition, path, replaces
> Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
> Content-Length: 0
>
>
> sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:45.182589:
> BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0
> Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD
> Route: <sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on>
> Route: <sip:65.217.40.210:5060;lr>
> Route: <sip:63.110.102.238;lr>
> Max-Forwards: 70
> From: <sip:+1212381XXXX at 63.110.102.238:5060;user=phone>;tag=Ztr5ycrv3QZ1g
> To: <sip:+1909635XXXX at 199.173.100.144:5060
> ;user=phone>;tag=dc7-13c4-2401b7-46dea593-2401b7
> Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441
> CSeq: 118584736 BYE
> Contact: <sip:+1212381XXXX at 66.197.142.69:5080;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO
> Supported: timer, precondition, path, replaces
> Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
> Content-Length: 0
>
> On Sun, Aug 2, 2009 at 11:20 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
>
>> Hi,
>>
>> Digging a bit in mod_sofia releaved that it can be caused by a SIP
>> code 482 (loop detected), 483 (too many hops) or 484 (address
>> incomplete).
>>
>> Do a SIP trace to sched more light on what's happening.
>>
>> Mathieu Rene
>> Avant-Garde Solutions Inc
>> Office: + 1 (514) 664-1044 x100
>> Cell: +1 (514) 664-1044 x200
>> mrene at avgs.ca
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From juanbackson at gmail.com  Sun Oct 18 08:29:45 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Sun, 18 Oct 2009 23:29:45 +0800
Subject: [Freeswitch-users] anyway to not to retry
Message-ID: <27c25bc40910180829y6ced15d7hdb75e7dda8daf04c@mail.gmail.com>

Hi,

Is there anyway to configure freeswitch so that it won't retry the SIP
message when 200 OK is not received?

thanks,
jb
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From dave at 3c.co.uk  Sun Oct 18 09:04:32 2009
From: dave at 3c.co.uk (David Knell)
Date: Sun, 18 Oct 2009 09:04:32 -0700
Subject: [Freeswitch-users] Detecting Forwarding Device Caller ID
	(Off	Topic)
In-Reply-To: <c55d46950910172047y6de63448h656a1d889077b8f5@mail.gmail.com>
References: <c55d46950910172047y6de63448h656a1d889077b8f5@mail.gmail.com>
Message-ID: <1255881872.5374.8.camel@localhost.localdomain>

Hi Max,

Some operators (e.g. Orange in the UK) will allow you to have two
numbers attached to one SIM, which might get you 4/10ths of the way to
your goal.

I'm struggling to see why purchasing DIDs won't help, unless your FS box
has no internet connectivity.  A bit more background on the problem
might help with finding a solution?

--Dave

> Hey Guys,
> 
> Pleas help me answer the following question. Sorry for being a bit
> off-topic.  I have a GSM-VoiP modem with just 4 SIM Cards. Is there a
> known mechanism that allows me to use this modem for more than 4
> (albeit not at the same time) GSM numbers?  Let's say the phone
> numbers of the SIM cards inserted in the modem are call real phone
> numbers. My goal is to have some 10 additional virtual phone numbers
> served by this very same modem.
> 
> One way i thought this could be done is by forwarding the virtual
> phone numbers the 4 real phone numbers. Then, when somebody calls one
> of the virtual numbers, the Telco will forward it to the real phone
> number and it will end on Freeswitch box. My only problem is now that
> on the Freeswitch box I have no idea what virtual phone number was
> actually dialed. Any idea how to recover this phone number? I looked
> at the GSM specification and couldn't find a way.
> 
> PS. Purchasing DIDs won't solve my underlying problem.
> 
> Thanks,
> Max.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From frank at carmickle.com  Sun Oct 18 09:40:00 2009
From: frank at carmickle.com (Frank Carmickle)
Date: Sun, 18 Oct 2009 12:40:00 -0400
Subject: [Freeswitch-users] Status of ubuntu/debian packages.
In-Reply-To: <90823c940910180249u69f18aaamc383f9a5a0c65848@mail.gmail.com>
References: <4ABA47FB.2050100@gmail.com>
	<90823c940910180249u69f18aaamc383f9a5a0c65848@mail.gmail.com>
Message-ID: <20091018164000.GD13895@base.carmickle.com>

On Sun, Oct 18, Dmitry Bely wrote:
> William,
> 
> When I upgrade freeswitch-config package, all my customized config
> files are silently overwritten with the default ones from .deb
> package. This bug is quite annoying.

Very sorry.  I can try to include some code that detects if your moving from the old versions that don't have a config package to the new stuff that does.  My way around this is not to install the freeswitch-config package at all.  I extract it some where else and diff my way through it.

--FC


From max.bridgewater at gmail.com  Sun Oct 18 09:51:04 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sun, 18 Oct 2009 12:51:04 -0400
Subject: [Freeswitch-users] Detecting Forwarding Device Caller ID (Off
	Topic)
In-Reply-To: <1255881872.5374.8.camel@localhost.localdomain>
References: <c55d46950910172047y6de63448h656a1d889077b8f5@mail.gmail.com>
	<1255881872.5374.8.camel@localhost.localdomain>
Message-ID: <c55d46950910180951t25ad668et385f6f3d78ce69e9@mail.gmail.com>

Thanks Dave. I'm essentially trying to build a service in one country where
there is no DID provider. If DIDs were available for such countries, that
would have done the job.

Using the two-numbers-per-SIM approach  would help significantly, if I could
only figure out, how to detect the Called Number using AT commands or
Unsolicited Result codes. I was hoping the CLIP code could give me this
information, but I'm still struggling to if it does or not. Any idea?

Max.

On Sun, Oct 18, 2009 at 12:04 PM, David Knell <dave at 3c.co.uk> wrote:

> Hi Max,
>
> Some operators (e.g. Orange in the UK) will allow you to have two
> numbers attached to one SIM, which might get you 4/10ths of the way to
> your goal.
>
> I'm struggling to see why purchasing DIDs won't help, unless your FS box
> has no internet connectivity.  A bit more background on the problem
> might help with finding a solution?
>
> --Dave
>
> > Hey Guys,
> >
> > Pleas help me answer the following question. Sorry for being a bit
> > off-topic.  I have a GSM-VoiP modem with just 4 SIM Cards. Is there a
> > known mechanism that allows me to use this modem for more than 4
> > (albeit not at the same time) GSM numbers?  Let's say the phone
> > numbers of the SIM cards inserted in the modem are call real phone
> > numbers. My goal is to have some 10 additional virtual phone numbers
> > served by this very same modem.
> >
> > One way i thought this could be done is by forwarding the virtual
> > phone numbers the 4 real phone numbers. Then, when somebody calls one
> > of the virtual numbers, the Telco will forward it to the real phone
> > number and it will end on Freeswitch box. My only problem is now that
> > on the Freeswitch box I have no idea what virtual phone number was
> > actually dialed. Any idea how to recover this phone number? I looked
> > at the GSM specification and couldn't find a way.
> >
> > PS. Purchasing DIDs won't solve my underlying problem.
> >
> > Thanks,
> > Max.
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sun Oct 18 11:43:25 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 18 Oct 2009 13:43:25 -0500
Subject: [Freeswitch-users] anyway to not to retry
In-Reply-To: <27c25bc40910180829y6ced15d7hdb75e7dda8daf04c@mail.gmail.com>
References: <27c25bc40910180829y6ced15d7hdb75e7dda8daf04c@mail.gmail.com>
Message-ID: <705F9C29-44F8-4E5E-93E1-0B234DAEECF4@freeswitch.org>

Nope its part of SIP it'll retry if a response isn't received in a  
certain time.

/b

On Oct 18, 2009, at 10:29 AM, Juan Backson wrote:

> Hi,
>
> Is there anyway to configure freeswitch so that it won't retry the  
> SIP message when 200 OK is not received?
>
> thanks,
> jb



From dmitry.bely at gmail.com  Sun Oct 18 11:54:02 2009
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Sun, 18 Oct 2009 22:54:02 +0400
Subject: [Freeswitch-users] Status of ubuntu/debian packages.
In-Reply-To: <20091018164000.GD13895@base.carmickle.com>
References: <4ABA47FB.2050100@gmail.com>
	<90823c940910180249u69f18aaamc383f9a5a0c65848@mail.gmail.com>
	<20091018164000.GD13895@base.carmickle.com>
Message-ID: <90823c940910181154w615b9ac5v89a30a3a3288e619@mail.gmail.com>

On Sun, Oct 18, 2009 at 8:40 PM, Frank Carmickle <frank at carmickle.com> wrote:
> On Sun, Oct 18, Dmitry Bely wrote:
>> William,
>>
>> When I upgrade freeswitch-config package, all my customized config
>> files are silently overwritten with the default ones from .deb
>> package. This bug is quite annoying.
>
> Very sorry. ?I can try to include some code that detects if your moving from the old versions that don't have a config package to the new stuff that does.

I did have the previous version of freeswitch-config installed but
nevertheless the files were lost.

> ?My way around this is not to install the freeswitch-config package at all. ?I extract it
> some where else and diff my way through it.

Yes, but the bug should be fixed.

- Dmitry Bely


From brian at freeswitch.org  Sun Oct 18 12:06:13 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 18 Oct 2009 14:06:13 -0500
Subject: [Freeswitch-users] anyway to not to retry
In-Reply-To: <27c25bc40910180829y6ced15d7hdb75e7dda8daf04c@mail.gmail.com>
References: <27c25bc40910180829y6ced15d7hdb75e7dda8daf04c@mail.gmail.com>
Message-ID: <EDA7684B-73B8-486D-A7AA-B689A268AFA5@freeswitch.org>

You can also try this http://wiki.freeswitch.org/wiki/Channel_Variables#progress_timeout

/b

On Oct 18, 2009, at 10:29 AM, Juan Backson wrote:

> Hi,
>
> Is there anyway to configure freeswitch so that it won't retry the  
> SIP message when 200 OK is not received?
>
> thanks,
> jb
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From tayeb.meftah at gmail.com  Sun Oct 18 13:08:53 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sun, 18 Oct 2009 20:08:53 +0000
Subject: [Freeswitch-users] freeswitch french community
Message-ID: <4ADB75D5.9000705@gmail.com>

hello folk
we are starting a new freeswitch community for french users
the newly created Irc channel is #freeswitch-fr
freeswitch staf, please can we register this channel officialy to the 
freeswitch freenode group?
also can we build a french documentation and host it somehere? in the wiki?
my long time friend, diegoviola i going to help about the freeswitch 
wiki markup
we are waiting for french users to join, and welcome
thank toAnthony Minessale for accepting the community
welcome!


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4520 (20091018) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com




From sjmudd at pobox.com  Sun Oct 18 13:09:07 2009
From: sjmudd at pobox.com (Simon J Mudd)
Date: 18 Oct 2009 22:09:07 +0200
Subject: [Freeswitch-users] macros for performing repeated dialplan
	actions?
In-Reply-To: <b39273530910170018h5063c309xa96ab59b15256055@mail.gmail.com>
References: <20091014220135.GA12614@mad06.wl0.org>
	<87f2f3b90910141510n45401995y11e3e380653e9790@mail.gmail.com>
	<m3eip41yxv.fsf@mad06.wl0.org>
	<1C80FD78-71B7-4D62-8C45-2F5BC8C1564C@apartmentlines.com>
	<m3aazq1pvl.fsf@mad06.wl0.org>
	<b39273530910170018h5063c309xa96ab59b15256055@mail.gmail.com>
Message-ID: <m363ac1n30.fsf@mad06.wl0.org>

rupa at rupa.com (Rupa Schomaker) writes:

> On Sat, Oct 17, 2009 at 1:44 AM, Simon J Mudd <sjmudd at pobox.com> wrote:
> > I've other plans like:
> >
> > 1. choose the cheapest gateway based on some scripts which download
> > rates from the various gateways web pages and let me compare rates.
> > Hard partly as the "charges pages" aren't designed to allow me to
> > match a number to a charge. Shame the VoIP community can't come up
> > with a common format "charge/rate file" to make this easier.
> 
> Look at mod_lcr.  The only part you have to write is the part that
> loads the database from your providers.

Sounds good. Thanks for the pointer.

> > 2. Try one gateway and if that fails try another one. Many of the
> > gateways I've been using are low cost and don't always work, so to
> > avoid me having to adjust things each time this might be useful.
> 
> bridge to ${lcr_auto_route} after calling the lcr application.

You make it sound easy in FreeSWITCH. :-)

> > 3. Remove a lot of the "intelligence" about what to do to a backend
> > script/program or whatever which decides how to route calls and tells
> > FreeSWITCH what it decides. This may not scale for a large setup but
> > for SOHO is never going to be an issue. It also avoids me writing so
> > many xml configuration files which seems to get spread all over
> > various places of the conf/ tree.
> 
> This is for incoming calls to your DIDs?  Or something else?  I'm not
> sure what you mean here.

Well it seems/ed that the dialplan actions and xml files get complex
if you want to do anything complicated. however it seems that I'm
doing things the wrong way and perhaps that's part of the problem.

I'll look at the mod_lcr and see how it works.

Thanks for the pointers.

Simon


From diego.viola at gmail.com  Sun Oct 18 16:13:34 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 18 Oct 2009 23:13:34 +0000
Subject: [Freeswitch-users] freeswitch french community
In-Reply-To: <4ADB75D5.9000705@gmail.com>
References: <4ADB75D5.9000705@gmail.com>
Message-ID: <86a32abc0910181613g321b8315t6204f6e45be7a212@mail.gmail.com>

Hi Meftah,

I'm more than happy to help with the documentation and anything else related
to FreeSWITCH.

As you mention the documentation and translation, there are some things that
I would really like to improve in the documentation.

I propose that we start a new documentation from the ground up, we really
need a better wiki engine that supports inline translation and a better
search engine. I'm open to do all the hard work like porting the entire
documentation to it.

We could put the new wiki on a sub-domain like: newdocs.freeswitch.org and I
do all the work there, without affecting others.

That way we could have inline documentation like this:

<text:en>status - show status information</text>
<text:es>status - mostrar informacion de estado</text>

It's up to others to decide if this is really worth or not.

I'm open to help and to give my time to improve things like documentation,
etc :).

Diego

On Sun, Oct 18, 2009 at 8:08 PM, Meftah Tayeb <tayeb.meftah at gmail.com>wrote:

> hello folk
> we are starting a new freeswitch community for french users
> the newly created Irc channel is #freeswitch-fr
> freeswitch staf, please can we register this channel officialy to the
> freeswitch freenode group?
> also can we build a french documentation and host it somehere? in the wiki?
> my long time friend, diegoviola i going to help about the freeswitch
> wiki markup
> we are waiting for french users to join, and welcome
> thank toAnthony Minessale for accepting the community
> welcome!
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus
> signature database 4520 (20091018) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From diego.viola at gmail.com  Sun Oct 18 16:17:27 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 18 Oct 2009 23:17:27 +0000
Subject: [Freeswitch-users] freeswitch french community
In-Reply-To: <86a32abc0910181613g321b8315t6204f6e45be7a212@mail.gmail.com>
References: <4ADB75D5.9000705@gmail.com>
	<86a32abc0910181613g321b8315t6204f6e45be7a212@mail.gmail.com>
Message-ID: <86a32abc0910181617o212e25e4i78f3b595860b8a32@mail.gmail.com>

The other way we could do is set up a fr.freeswitch.org with mediawiki on it
and you start to translate from the English wiki.

But I think inline translation would be better for this.

Just my two cents.

Diego


On Sun, Oct 18, 2009 at 11:13 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Hi Meftah,
>
> I'm more than happy to help with the documentation and anything else
> related to FreeSWITCH.
>
> As you mention the documentation and translation, there are some things
> that I would really like to improve in the documentation.
>
> I propose that we start a new documentation from the ground up, we really
> need a better wiki engine that supports inline translation and a better
> search engine. I'm open to do all the hard work like porting the entire
> documentation to it.
>
> We could put the new wiki on a sub-domain like: newdocs.freeswitch.org and
> I do all the work there, without affecting others.
>
> That way we could have inline documentation like this:
>
> <text:en>status - show status information</text>
> <text:es>status - mostrar informacion de estado</text>
>
> It's up to others to decide if this is really worth or not.
>
> I'm open to help and to give my time to improve things like documentation,
> etc :).
>
> Diego
>
>
> On Sun, Oct 18, 2009 at 8:08 PM, Meftah Tayeb <tayeb.meftah at gmail.com>wrote:
>
>> hello folk
>> we are starting a new freeswitch community for french users
>> the newly created Irc channel is #freeswitch-fr
>> freeswitch staf, please can we register this channel officialy to the
>> freeswitch freenode group?
>> also can we build a french documentation and host it somehere? in the
>> wiki?
>> my long time friend, diegoviola i going to help about the freeswitch
>> wiki markup
>> we are waiting for french users to join, and welcome
>> thank toAnthony Minessale for accepting the community
>> welcome!
>>
>>
>> __________ Information from ESET NOD32 Antivirus, version of virus
>> signature database 4520 (20091018) __________
>>
>> The message was checked by ESET NOD32 Antivirus.
>>
>> http://www.eset.com
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From diego.viola at gmail.com  Sun Oct 18 16:19:41 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 18 Oct 2009 23:19:41 +0000
Subject: [Freeswitch-users] freeswitch french community
In-Reply-To: <86a32abc0910181617o212e25e4i78f3b595860b8a32@mail.gmail.com>
References: <4ADB75D5.9000705@gmail.com>
	<86a32abc0910181613g321b8315t6204f6e45be7a212@mail.gmail.com>
	<86a32abc0910181617o212e25e4i78f3b595860b8a32@mail.gmail.com>
Message-ID: <86a32abc0910181619oa26ed79l4c55b967fa2a5e95@mail.gmail.com>

Take a look at this for instance, for an example of inline translation:
http://wiki.rubyonrails.org/

Check the: "Translations of this page" -- all in the same page, I'd love to
have something like that for FreeSWITCH.

Along with a good search engine on the wiki (not Google).

Diego

On Sun, Oct 18, 2009 at 11:17 PM, Diego Viola <diego.viola at gmail.com> wrote:

> The other way we could do is set up a fr.freeswitch.org with mediawiki on
> it and you start to translate from the English wiki.
>
> But I think inline translation would be better for this.
>
> Just my two cents.
>
> Diego
>
>
>
> On Sun, Oct 18, 2009 at 11:13 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> Hi Meftah,
>>
>> I'm more than happy to help with the documentation and anything else
>> related to FreeSWITCH.
>>
>> As you mention the documentation and translation, there are some things
>> that I would really like to improve in the documentation.
>>
>> I propose that we start a new documentation from the ground up, we really
>> need a better wiki engine that supports inline translation and a better
>> search engine. I'm open to do all the hard work like porting the entire
>> documentation to it.
>>
>> We could put the new wiki on a sub-domain like: newdocs.freeswitch.organd I do all the work there, without affecting others.
>>
>> That way we could have inline documentation like this:
>>
>> <text:en>status - show status information</text>
>> <text:es>status - mostrar informacion de estado</text>
>>
>> It's up to others to decide if this is really worth or not.
>>
>> I'm open to help and to give my time to improve things like documentation,
>> etc :).
>>
>> Diego
>>
>>
>> On Sun, Oct 18, 2009 at 8:08 PM, Meftah Tayeb <tayeb.meftah at gmail.com>wrote:
>>
>>> hello folk
>>> we are starting a new freeswitch community for french users
>>> the newly created Irc channel is #freeswitch-fr
>>> freeswitch staf, please can we register this channel officialy to the
>>> freeswitch freenode group?
>>> also can we build a french documentation and host it somehere? in the
>>> wiki?
>>> my long time friend, diegoviola i going to help about the freeswitch
>>> wiki markup
>>> we are waiting for french users to join, and welcome
>>> thank toAnthony Minessale for accepting the community
>>> welcome!
>>>
>>>
>>> __________ Information from ESET NOD32 Antivirus, version of virus
>>> signature database 4520 (20091018) __________
>>>
>>> The message was checked by ESET NOD32 Antivirus.
>>>
>>> http://www.eset.com
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
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From jason at jasonjgw.net  Sun Oct 18 16:58:58 2009
From: jason at jasonjgw.net (Jason White)
Date: Mon, 19 Oct 2009 10:58:58 +1100
Subject: [Freeswitch-users] freeswitch french community
In-Reply-To: <86a32abc0910181613g321b8315t6204f6e45be7a212@mail.gmail.com>
References: <4ADB75D5.9000705@gmail.com>
	<86a32abc0910181613g321b8315t6204f6e45be7a212@mail.gmail.com>
Message-ID: <20091018235858.GA19968@jdc.jasonjgw.net>

Diego Viola <diego.viola at gmail.com> wrote:
 
> I propose that we start a new documentation from the ground up, we really
> need a better wiki engine that supports inline translation and a better
> search engine. I'm open to do all the hard work like porting the entire
> documentation to it.

Is there a good wiki engine which allows the documentation to be edited as
files in a file system as well as via a Web interface - something like
Ikiwiki which uses Git or another version control system to store the files,
perhaps?

On occasion I have wished I could add to or update the wiki easily, but there
are two problems.

1. The current wiki requires a captcha to be completed in order to register an
account. I can't see the captcha - this is an accessibility issue.

2. Editing anything through a Web interface is painful. If I could edit the
content as files in Emacs or Vim it would be much easier, and I would be more
likely to do it instead of deciding that the difficulty and inconvenience are
simply too great.

Obviously this is only one person's perspective: other people will have very
different needs from mine.



From larclap at yahoo.com  Sun Oct 18 17:02:57 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Sun, 18 Oct 2009 17:02:57 -0700
Subject: [Freeswitch-users] Outbound sip address call fails
In-Reply-To: <20091016172936.CCE7C41E8F6@mail.cune.org>
References: <011301ca4dd3$67d486a0$377d93e0$@com>
	<20091016172936.CCE7C41E8F6@mail.cune.org>
Message-ID: <01e201ca504f$83bc61c0$8b352540$@com>

No, the checkbox is unchecked.

> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-
> bounces at lists.freeswitch.org] On Behalf Of Russell.Mosemann at cune.org
> Sent: Friday, October 16, 2009 10:30 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Outbound sip address call fails
> 
> Lars Zeb <larclap at yahoo.com> said:
> 
> > I tried dial sip:echo at iptel.org. In the pastebin you can see that this
> > address came over to the cli as "echo". Is it Bria which is messing up,
or
> > my FS configuration?
> 
> It looks like a Bria problem. If you open the account information and go
> to the Advanced tab, is the box checked to "Send outgoing request
> directly to target"?
> 
> 
> --
> Russell Mosemann
> 
> 
> 
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From Russell.Mosemann at cune.org  Sun Oct 18 17:40:09 2009
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Sun, 18 Oct 2009 19:40:09 -0500
Subject: [Freeswitch-users] Outbound sip address call fails
In-Reply-To: <01e201ca504f$83bc61c0$8b352540$@com>
References: <011301ca4dd3$67d486a0$377d93e0$@com><20091016172936.CCE7C41E8F6@mail.cune.org>
	<01e201ca504f$83bc61c0$8b352540$@com>
Message-ID: <0542FF31EAB34BD8977E8E87263AA88A@cune.pri>

> No, the checkbox is unchecked.

Well, check it.

--
Russell Mosemann



From freeswitch-list at puzzled.xs4all.nl  Sun Oct 18 04:31:17 2009
From: freeswitch-list at puzzled.xs4all.nl (Patrick List)
Date: Sun, 18 Oct 2009 13:31:17 +0200
Subject: [Freeswitch-users] Status of ubuntu/debian packages.
In-Reply-To: <90823c940910180249u69f18aaamc383f9a5a0c65848@mail.gmail.com>
References: <4ABA47FB.2050100@gmail.com>
	<90823c940910180249u69f18aaamc383f9a5a0c65848@mail.gmail.com>
Message-ID: <4ADAFC85.6000301@puzzled.xs4all.nl>

On 10/18/2009 11:49 AM, Dmitry Bely wrote:
> William,
>
> When I upgrade freeswitch-config package, all my customized config
> files are silently overwritten with the default ones from .deb
> package. This bug is quite annoying.

Hope you have a recent backup. Years ago when I created rpm packages for 
Asterisk I chose to put the default config files in it's own directory 
so there was no chance of your customized configs getting overwritten. 
Example:
/opt/freeswitch/etc/freeswitch-$svn_version-samples/

Maybe the maintainer of the debs can do something similar.

Regards,
Patrick


From mike at jerris.com  Sun Oct 18 18:56:05 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 18 Oct 2009 21:56:05 -0400
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
In-Reply-To: <4256bf830910180725y72539d98gb138934afc18d58b@mail.gmail.com>
References: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
	<177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>
	<4256bf830908041151l55980e3fifb85c75b87535426@mail.gmail.com>
	<3D0B855A-5B2C-4C6F-9306-64B0ECE97091@jerris.com>
	<4256bf830910180725y72539d98gb138934afc18d58b@mail.gmail.com>
Message-ID: <A5ECDCD9-1A12-4A31-89F0-E6F4C79FE9E9@jerris.com>

FreeSWITCH debug level logs should help tell you exactly what is  
killing the call.


On Oct 18, 2009, at 10:25 AM, Matthew Fong wrote:

> I'm still having this issue with random EXCHANGE_ROUTING_ERROR  
> disconnects for users utilizing my inbound DID to connect to my  
> FreeSWITCH server. It's a predictive dialing application, with one  
> agent session being bridged with multiple calls and transfered back  
> and forth between extensions in my dial plan. After a random number  
> of bridging and transferring, FreeSWITCH suddenly sends a BYE to my  
> DID provider citing an EXCHANGE_ROUTING_ERROR. It does not happen at  
> any one-point in my dial plan, or applications--it just randomly  
> disconnects when a call that the Agent is bridged to hangs-up or is  
> disconnected. It seems to only happen when two external sip profiles  
> are being bridged together, and not when an internal and external  
> profile is being bridged.
>
> I turned
>
> sip trace on and
> sofia loglevel all 9
>
> below is the the snippet. I've posted the entire Agent session at  
> the following pastebin http://pastebin.freeswitch.org/10756
>
> tport_deliver(0x18413c0): msg 0x7faeb818ea30 (304 bytes) from udp/ 
> 208.76.18.254:5080/sip next=(nil)
> nta: received 200 OK for BYE (121818983)
> nta: 200 OK is going to a transaction
> nta_outgoing: RTT is 84.409 ms
> tport_release(0x18413c0): 0x1a15cc0 by 0x1a16a00 with 0x7faeb818ea30
> nua(0x1ad6fb0): event r_bye 200 OK
> nua(0x1ad6fb0): call state changed: terminating -> terminated
> nua(0x1ad6fb0): event i_state 200 to BYE
> nua: nua_application_event: entering
> nua(0x1ad6fb0): event i_terminated 200 to BYE
> nua: nua_handle_magic: entering
> nua(0x1ad6fb0): removing session usage
> soa_destroy(static::0x1b5ae90) called
> nua: nua_application_event: entering
> nta_leg_destroy(0x1b594a0)
> nua: nua_handle_magic: entering
> nua: nua_handle_bind: entering
> nua: nua_application_event: entering
> nua: nua_handle_magic: entering
> nua: terminated session 0x1ad6fb0
> nua: nua_handle_destroy: entering
> nua(0x1ad6fb0): recv signal r_destroy
> nta_leg_destroy((nil))
> nua(0x1ad6fb0): sent signal r_destroy
> nta: timer set next to 28 ms
> nta: timer E fired, retransmit BYE (121818989)
> tport_release(0x18413c0): 0x1b5c9b0 by 0x7faeb817d830 with (nil)
> tport_tsend(0x18413c0) tpn = */209.216.2.211:5060
> tport_resolve addrinfo = 209.216.2.211:5060
> tport_by_addrinfo(0x18413c0): not found by name */209.216.2.211:5060
> tport_vsend(0x18413c0): 862 bytes of 862 to udp/209.216.2.211:5060
> tport_vsend returned 862
> send 862 bytes to udp/[209.216.2.211]:5060 at 14:04:11.753690:
>     
> ------------------------------------------------------------------------
>    BYE sip:199.173.100.16:5060;transport=UDP SIP/2.0
>    Via: SIP/2.0/UDP  
> 67.220.216.146:5080;rport;branch=z9hG4bK02jNX8a4HrNyN
>    Route: <sip:209.216.2.211;ftag=4adc7-13c4-1ab03-71ce3705-1ab03;lr=on 
> >
>    Route: <sip:65.211.120.237:5060;lr>
>    Route: <sip:63.110.102.239;lr>
>    Max-Forwards: 70
>    From: <sip: 
> +12133304391 at 63.110.102.239:5060;user=phone>;tag=cgBe054jZrt3a
>    To: <sip: 
> + 
> 14158867717 
> @199.173.100.16:5060;user=phone>;tag=4adc7-13c4-1ab03-71ce3705-1ab03
>    Call-ID: a0f656a01064adc713c41ab036840746ee20ca11c06b2d8-0440-5124
>    CSeq: 121818989 BYE
>    Contact: <sip:+12133304391 at 67.220.216.146:5080;transport=udp>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15135
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO,  
> REGISTER, REFER, UPDATE, NOTIFY
>    Supported: timer, precondition, path, replaces
>    Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
>    Content-Length: 0
>
>     
> ------------------------------------------------------------------------
>
> Thanks.
> --matt
>
> On Sat, Aug 8, 2009 at 12:42 PM, Michael Jerris <mike at jerris.com>  
> wrote:
> http://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP
>
> turn the logging all the way up and see what it says.
>
> Mike
>
> On Aug 4, 2009, at 2:51 PM, Matthew Fong wrote:
>
>> Hi Mathieu, thanks for the reply. I enabled sip trace logging and  
>> got the logs below, but I am still at a loss at being able to  
>> identify the error or reproduce it consistently. The below log  
>> indicates to me that my FS server is initiating sending 2 BYE  
>> message to my DID provider (didforsale.com). Is there a way I can  
>> look further inside FreeSWITCH to see why it is sending this BYE  
>> packet?
>>
>>
>> sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:44.679208:
>> BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0
>> Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD
>> Route: <sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on>
>> Route: <sip:65.217.40.210:5060;lr>
>> Route: <sip:63.110.102.238;lr>
>> Max-Forwards: 70
>> From: <sip: 
>> +1212381XXXX at 63.110.102.238:5060;user=phone>;tag=Ztr5ycrv3QZ1g
>> To: <sip: 
>> + 
>> 1909635XXXX 
>> @199.173.100.144:5060;user=phone>;tag=dc7-13c4-2401b7-46dea593-2401b7
>> Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441
>> CSeq: 118584736 BYE
>> Contact: <sip:+1212381XXXX at 66.197.142.69:5080;transport=udp>
>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,  
>> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
>> Supported: timer, precondition, path, replaces
>> Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
>> Content-Length: 0
>>
>>
>> sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:45.182589:
>> BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0
>> Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD
>> Route: <sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on>
>> Route: <sip:65.217.40.210:5060;lr>
>> Route: <sip:63.110.102.238;lr>
>> Max-Forwards: 70
>> From: <sip: 
>> +1212381XXXX at 63.110.102.238:5060;user=phone>;tag=Ztr5ycrv3QZ1g
>> To: <sip: 
>> + 
>> 1909635XXXX 
>> @199.173.100.144:5060;user=phone>;tag=dc7-13c4-2401b7-46dea593-2401b7
>> Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441
>> CSeq: 118584736 BYE
>> Contact: <sip:+1212381XXXX at 66.197.142.69:5080;transport=udp>
>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,  
>> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
>> Supported: timer, precondition, path, replaces
>> Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
>> Content-Length: 0
>>
>> On Sun, Aug 2, 2009 at 11:20 PM, Mathieu Rene <mrene_lists at avgs.ca>  
>> wrote:
>> Hi,
>>
>> Digging a bit in mod_sofia releaved that it can be caused by a SIP
>> code 482 (loop detected), 483 (too many hops) or 484 (address
>> incomplete).
>>
>> Do a SIP trace to sched more light on what's happening.
>>
>> Mathieu Rene
>> Avant-Garde Solutions Inc
>> Office: + 1 (514) 664-1044 x100
>> Cell: +1 (514) 664-1044 x200
>> m
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From larclap at yahoo.com  Sun Oct 18 19:22:45 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Sun, 18 Oct 2009 19:22:45 -0700
Subject: [Freeswitch-users] Outbound sip address call fails
In-Reply-To: <0542FF31EAB34BD8977E8E87263AA88A@cune.pri>
References: <011301ca4dd3$67d486a0$377d93e0$@com><20091016172936.CCE7C41E8F6@mail.cune.org>	<01e201ca504f$83bc61c0$8b352540$@com>
	<0542FF31EAB34BD8977E8E87263AA88A@cune.pri>
Message-ID: <01e301ca5063$0c498690$24dc93b0$@com>

Thanks, Russell, that did it. Works perfectly!

> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-
> bounces at lists.freeswitch.org] On Behalf Of Russell Mosemann
> Sent: Sunday, October 18, 2009 5:40 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Outbound sip address call fails
> 
> > No, the checkbox is unchecked.
> 
> Well, check it.
> 
> --
> Russell Mosemann
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From dujinfang at gmail.com  Sun Oct 18 20:04:00 2009
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 19 Oct 2009 11:04:00 +0800
Subject: [Freeswitch-users] freeswitch french community
In-Reply-To: <86a32abc0910181617o212e25e4i78f3b595860b8a32@mail.gmail.com>
References: <4ADB75D5.9000705@gmail.com>
	<86a32abc0910181613g321b8315t6204f6e45be7a212@mail.gmail.com>
	<86a32abc0910181617o212e25e4i78f3b595860b8a32@mail.gmail.com>
Message-ID: <23f91030910182004g6bfdafb6g6198d107220f8320@mail.gmail.com>

There are some plugins for mediawiki to support multilanguage, either inline
or on a separate page. However, both have Pros and Cons. e.g.

multi-language support this with a separate page

some_page #default to en
some_page/fr Franch
some_page/zh Chinese

Pros: clear, available languages can be listed in "Other languages for this
page"
Cons: hard to keep all languages in sync

other plugins can do inline translation:

<@en|hello>
<@fr|Ca Va>
<@zh|??>

Pros: untranslated items remains in default lang, no sync headache
Cons: It breaks a page into sentences or items, also if we have more than 10
languages the source code maybe really a mess.


2009/10/19 Diego Viola <diego.viola at gmail.com>

> The other way we could do is set up a fr.freeswitch.org with mediawiki on
> it and you start to translate from the English wiki.
>
> But I think inline translation would be better for this.
>
> Just my two cents.
>
> Diego
>
>
>
> On Sun, Oct 18, 2009 at 11:13 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> Hi Meftah,
>>
>> I'm more than happy to help with the documentation and anything else
>> related to FreeSWITCH.
>>
>> As you mention the documentation and translation, there are some things
>> that I would really like to improve in the documentation.
>>
>> I propose that we start a new documentation from the ground up, we really
>> need a better wiki engine that supports inline translation and a better
>> search engine. I'm open to do all the hard work like porting the entire
>> documentation to it.
>>
>> We could put the new wiki on a sub-domain like: newdocs.freeswitch.organd I do all the work there, without affecting others.
>>
>> That way we could have inline documentation like this:
>>
>> <text:en>status - show status information</text>
>> <text:es>status - mostrar informacion de estado</text>
>>
>> It's up to others to decide if this is really worth or not.
>>
>> I'm open to help and to give my time to improve things like documentation,
>> etc :).
>>
>> Diego
>>
>>
>> On Sun, Oct 18, 2009 at 8:08 PM, Meftah Tayeb <tayeb.meftah at gmail.com>wrote:
>>
>>> hello folk
>>> we are starting a new freeswitch community for french users
>>> the newly created Irc channel is #freeswitch-fr
>>> freeswitch staf, please can we register this channel officialy to the
>>> freeswitch freenode group?
>>> also can we build a french documentation and host it somehere? in the
>>> wiki?
>>> my long time friend, diegoviola i going to help about the freeswitch
>>> wiki markup
>>> we are waiting for french users to join, and welcome
>>> thank toAnthony Minessale for accepting the community
>>> welcome!
>>>
>>>
>>> __________ Information from ESET NOD32 Antivirus, version of virus
>>> signature database 4520 (20091018) __________
>>>
>>> The message was checked by ESET NOD32 Antivirus.
>>>
>>> http://www.eset.com
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From chris at fowler.cc  Sun Oct 18 20:14:49 2009
From: chris at fowler.cc (Chris Fowler)
Date: Sun, 18 Oct 2009 20:14:49 -0700
Subject: [Freeswitch-users] NAT problems migrating from Version
 1.0.trunk (13168M) to Version 1.0.trunk (15166)
Message-ID: <1255922089.4653.1340747457@webmail.messagingengine.com>

I've tried all sorts of debug and parameter changes over the weekend,
but still can't figure out the correct solution.

If I disable timers on the sip profile then all works fine.
 <param name="enable-timer" value="false"/>

But that seems like a hack; not a correct solution.  With the build
13168M (which is pre the new NAT functionality) everything worked fine.

The SIP trace shows the phones and FreeSWITCH happily exchanging NOTIFY
and "200 OK" messages.  Audio's working - just calls timeout after 100
seconds with RECOVERY_ON_TIMER_EXPIRE.  Is enforcement of this timer new
functionality - and really just exposing a problem I've always had
before?

The config is (50 Polycom Phones -> NAT -> Internet -> Amazon EC2)

I would really appreciate some pointers on what to look for; additional
trace that might reveal something.

Thanks, Chris.


On Fri, 16 Oct 2009 20:06:52 -0700, "Chris Fowler" <chris at fowler.cc>
said:
> Hi,
> 
> We've been using 13168M in production for some time now (works great). 
> I want to get us onto the latest build but am having problems getting
> NAT to work.
> 
> Phones can register; can dial 9999 test #, but after 100 seconds the
> call is disconnected with error:
> 2009-10-16 19:52:26.936618 [NOTICE] sofia.c:4038 Hangup
> sofia/internal/1001 at myhost.mydomain.com [CS_EXECUTE]
> [RECOVERY_ON_TIMER_EXPIRE]
> 
> I took the standard internal.xml and vars.xml files from the new build
> and made the following modifications - which worked previously:
> 
> modify conf/vars.xml and update
>   <X-PRE-PROCESS cmd="set" data="domain=myhost.mydomain.com"/>
> 
>   <X-PRE-PROCESS cmd="set" data="bind_server_ip=1.2.3.4"/>
>   <X-PRE-PROCESS cmd="set" data="external_rtp_ip=1.2.3.4"/>
>   <X-PRE-PROCESS cmd="set" data="external_sip_ip=1.2.3.4"/>
> 
> Modify conf/sip_profiles/internal.xml
>  <param name="aggressive-nat-detection" value="true"/>  <param
>  name="ext-rtp-ip" value="$${external_rtp_ip}"/>  <param
>  name="ext-sip-ip" value="$${external_sip_ip}"/>  <param
>  name="NDLB-received-in-nat-reg-contact" value="true"/>  <param
>  name="NDLB-force-rport" value="true"/>  <param
>  name="NDLB-broken-auth-hash" value="true"/>
> 
> The big difference I note is that on PRODUCTION (which works) "sofia
> status profile internal" yields:
> URL                     sip:mod_sofia at 1.2.3.4:5060
> BIND-URL                sip:mod_sofia at 1.2.3.4:5060;maddr=10.250.35.224
> 
> But on Test I see:
> URL                     sip:mod_sofia at 10.250.66.210:5060
> BIND-URL                sip:mod_sofia at 10.250.66.210:5060
> 
> Any ideas?
> 
> Thanks, Chris.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From msc at freeswitch.org  Sun Oct 18 20:37:16 2009
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 18 Oct 2009 20:37:16 -0700
Subject: [Freeswitch-users] Excellent FreeSWITCH Article For You To Read
Message-ID: <87f2f3b90910182037x5c1d1279ue50f724f0a02f94c@mail.gmail.com>

Hello all,

Just a heads up that there is a great article out there for you to read.
Please visit the main freeswitch.org page here:
http://www.freeswitch.org/node/211

Click the link and digg the article after you read it. Please spread the
word! The twittersphere is starting to warm up already. Let's keep the
momentum going!

Thanks,
Michael
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From mattdfong at gmail.com  Sun Oct 18 21:48:25 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Mon, 19 Oct 2009 11:48:25 +0700
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
In-Reply-To: <A5ECDCD9-1A12-4A31-89F0-E6F4C79FE9E9@jerris.com>
References: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
	<177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>
	<4256bf830908041151l55980e3fifb85c75b87535426@mail.gmail.com>
	<3D0B855A-5B2C-4C6F-9306-64B0ECE97091@jerris.com>
	<4256bf830910180725y72539d98gb138934afc18d58b@mail.gmail.com>
	<A5ECDCD9-1A12-4A31-89F0-E6F4C79FE9E9@jerris.com>
Message-ID: <4256bf830910182148mf7f2bf9vdcccd687e0caa1e7@mail.gmail.com>

The debug level logs to the console, right? The pastebin, had log level
debug, sofia trace on for external and default, and sofia loglevel all 9. Is
there another log enable command I'm missing? It seems loglevel all 9
outputs enter and exit functions, but at least to my novice eye, it's not
too obvious why freeswitch is sending a BYE to my DID provider.

I did do some additional testing, and my prior comment about it working in
the internal profile is incorrect. Even if I put my DID provider in my
internal profile, I still receive the Exchange_Routing_Error after being
bridged with a few channels. However, I purchased a DID from icall, and that
DID worked. So icall YES, didforsale.com NO...It's too bad tho that
didforsale.com doesn't work too well with FreeSWITCH because their 20
unlimited inbound channels can't be beat.

--matt
http://www.hellohunter.com


On Mon, Oct 19, 2009 at 8:56 AM, Michael Jerris <mike at jerris.com> wrote:

> FreeSWITCH debug level logs should help tell you exactly what is killing
> the call.
>
>
> On Oct 18, 2009, at 10:25 AM, Matthew Fong wrote:
>
> I'm still having this issue with random EXCHANGE_ROUTING_ERROR disconnects
> for users utilizing my inbound DID to connect to my FreeSWITCH server. It's
> a predictive dialing application, with one agent session being bridged with
> multiple calls and transfered back and forth between extensions in my dial
> plan. After a random number of bridging and transferring, FreeSWITCH
> suddenly sends a BYE to my DID provider citing an EXCHANGE_ROUTING_ERROR. It
> does not happen at any one-point in my dial plan, or applications--it just
> randomly disconnects when a call that the Agent is bridged to hangs-up or is
> disconnected. It seems to only happen when two external sip profiles are
> being bridged together, and not when an internal and external profile is
> being bridged.
> I turned
>
> sip trace on and
> sofia loglevel all 9
>
> below is the the snippet. I've posted the entire Agent session at the
> following pastebin http://pastebin.freeswitch.org/10756
>
> tport_deliver(0x18413c0): msg 0x7faeb818ea30 (304 bytes) from udp/
> 208.76.18.254:5080/sip next=(nil)
> nta: received 200 OK for BYE (121818983)
> nta: 200 OK is going to a transaction
> nta_outgoing: RTT is 84.409 ms
> tport_release(0x18413c0): 0x1a15cc0 by 0x1a16a00 with 0x7faeb818ea30
> nua(0x1ad6fb0): event r_bye 200 OK
> nua(0x1ad6fb0): call state changed: terminating -> terminated
> nua(0x1ad6fb0): event i_state 200 to BYE
> nua: nua_application_event: entering
> nua(0x1ad6fb0): event i_terminated 200 to BYE
> nua: nua_handle_magic: entering
> nua(0x1ad6fb0): removing session usage
> soa_destroy(static::0x1b5ae90) called
> nua: nua_application_event: entering
> nta_leg_destroy(0x1b594a0)
> nua: nua_handle_magic: entering
> nua: nua_handle_bind: entering
> nua: nua_application_event: entering
> nua: nua_handle_magic: entering
> nua: terminated session 0x1ad6fb0
> nua: nua_handle_destroy: entering
> nua(0x1ad6fb0): recv signal r_destroy
> nta_leg_destroy((nil))
> nua(0x1ad6fb0): sent signal r_destroy
> nta: timer set next to 28 ms
> nta: timer E fired, retransmit BYE (121818989)
> tport_release(0x18413c0): 0x1b5c9b0 by 0x7faeb817d830 with (nil)
> tport_tsend(0x18413c0) tpn = */209.216.2.211:5060
> tport_resolve addrinfo = 209.216.2.211:5060
> tport_by_addrinfo(0x18413c0): not found by name */209.216.2.211:5060
> tport_vsend(0x18413c0): 862 bytes of 862 to udp/209.216.2.211:5060
> tport_vsend returned 862
> send 862 bytes to udp/[209.216.2.211]:5060 at 14:04:11.753690:
>    ------------------------------------------------------------------------
>    BYE sip:199.173.100.16:5060;transport=UDP SIP/2.0
>    Via: SIP/2.0/UDP 67.220.216.146:5080;rport;branch=z9hG4bK02jNX8a4HrNyN
>    Route: <sip:209.216.2.211;ftag=4adc7-13c4-1ab03-71ce3705-1ab03;lr=on>
>    Route: <sip:65.211.120.237:5060;lr>
>    Route: <sip:63.110.102.239;lr>
>    Max-Forwards: 70
>    From: <sip:+12133304391 at 63.110.102.239:5060;user=phone
> >;tag=cgBe054jZrt3a
>    To: <sip:+14158867717 at 199.173.100.16:5060;user=phone
> >;tag=4adc7-13c4-1ab03-71ce3705-1ab03
>    Call-ID: a0f656a01064adc713c41ab036840746ee20ca11c06b2d8-0440-5124
>    CSeq: 121818989 BYE
>    Contact: <sip:+12133304391 at 67.220.216.146:5080;transport=udp>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15135
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
> REFER, UPDATE, NOTIFY
>    Supported: timer, precondition, path, replaces
>    Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
>
> Thanks.
> --matt
>
> On Sat, Aug 8, 2009 at 12:42 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> http://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP
>> turn the logging all the way up and see what it says.
>>
>> Mike
>>
>> On Aug 4, 2009, at 2:51 PM, Matthew Fong wrote:
>>
>> Hi Mathieu, thanks for the reply. I enabled sip trace logging and got the
>> logs below, but I am still at a loss at being able to identify the error or
>> reproduce it consistently. The below log indicates to me that my FS server
>> is initiating sending 2 BYE message to my DID provider (didforsale.com).
>> Is there a way I can look further inside FreeSWITCH to see why it is sending
>> this BYE packet?
>>
>> sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:44.679208:
>> BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0
>> Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD
>> Route: <sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on>
>> Route: <sip:65.217.40.210:5060;lr>
>> Route: <sip:63.110.102.238;lr>
>> Max-Forwards: 70
>> From: <sip:+1212381XXXX at 63.110.102.238:5060;user=phone>;tag=Ztr5ycrv3QZ1g
>> To: <sip:+1909635XXXX at 199.173.100.144:5060;user=phone
>> >;tag=dc7-13c4-2401b7-46dea593-2401b7
>> Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441
>> CSeq: 118584736 BYE
>> Contact: <sip:+1212381XXXX at 66.197.142.69:5080;transport=udp>
>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>> NOTIFY, REFER, UPDATE, REGISTER, INFO
>> Supported: timer, precondition, path, replaces
>> Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
>> Content-Length: 0
>>
>>
>> sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:45.182589:
>> BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0
>> Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD
>> Route: <sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on>
>> Route: <sip:65.217.40.210:5060;lr>
>> Route: <sip:63.110.102.238;lr>
>> Max-Forwards: 70
>> From: <sip:+1212381XXXX at 63.110.102.238:5060;user=phone>;tag=Ztr5ycrv3QZ1g
>> To: <sip:+1909635XXXX at 199.173.100.144:5060;user=phone
>> >;tag=dc7-13c4-2401b7-46dea593-2401b7
>> Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441
>> CSeq: 118584736 BYE
>> Contact: <sip:+1212381XXXX at 66.197.142.69:5080;transport=udp>
>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>> NOTIFY, REFER, UPDATE, REGISTER, INFO
>> Supported: timer, precondition, path, replaces
>> Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
>> Content-Length: 0
>>
>> On Sun, Aug 2, 2009 at 11:20 PM, Mathieu Rene <mrene_lists at avgs.ca>wrote:
>>
>>> Hi,
>>>
>>> Digging a bit in mod_sofia releaved that it can be caused by a SIP
>>> code 482 (loop detected), 483 (too many hops) or 484 (address
>>> incomplete).
>>>
>>> Do a SIP trace to sched more light on what's happening.
>>>
>>> Mathieu Rene
>>> Avant-Garde Solutions Inc
>>> Office: + 1 (514) 664-1044 x100
>>> Cell: +1 (514) 664-1044 x200
>>> m <mrene at avgs.ca>
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mcampbellsmith at gmail.com  Sun Oct 18 21:53:02 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Mon, 19 Oct 2009 15:53:02 +1100
Subject: [Freeswitch-users] validating dtmf digits received
Message-ID: <33c87fa30910182153t23e52fa0p2b967b3843eb1175@mail.gmail.com>

Hi!

I simply want to validate the dtmf digits I read from a user.    From
the wiki, it appears I need to use inline="true" when setting the
variable so it can be used directly within the same extension.

What have I done wrong below?   I have tried many different
alternatives, but the second condition field, which is meant to match
the dtmf digits received (in this case 0000) is never matched, and the
anti-action is called instead.

:
      <some code here>
      <action application="read" data="1 10
ivr/ivr-please_enter_pin_followed_by_pound.wav res 10000 9"/>
       <action application="phrase" data="spell,${res}"/>
       <action inline="true" application="set" data="code=${digits_dialed}"/>
       <!-- <action inline="true" application="set" data="code=${res}"/> -->
  </condition>
  <condition field="digits_dialed" expression="^0000$">
  <!-- <condition field="${code}" expression="^0000$"> -->
  <!-- <condition field="${res}" expression="^0000$"> -->
        <some code here>
       <anti-action application="hangup"/>

Thanks!


From tzury.by at reguluslabs.com  Mon Oct 19 00:23:57 2009
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Mon, 19 Oct 2009 09:23:57 +0200
Subject: [Freeswitch-users] How to enforce freeswitch replying to the source
	port instead of to the one specified in v/m header parametes
	(Symmetric NAT)
Message-ID: <10128ef10910190023t17574fafic892bd4105cc1703@mail.gmail.com>

Hi,

I am struggling with a cellular operator which removes the rport from
the SIP messages sent by  the client.
During the troubleshooting process I have been playing with the NDLB
parameters I found mentioned at the fs-wiki

<param name="NDLB-force-rport" value="true"/>
<param name="NDLB-received-in-nat-reg-contact" value="true"/>

as well as with sip-force-contact at the extension xml config file.

<variables>
    <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/>
</variables>

I Want to make freeswitch replying to the source port instead of the
port supplied by the client, that is
to port 12543 instead of to the 7608 as you can see below some pcap
dumps of the case.

I am not sure if that would solve thep roblem and am open to other
suggestions as well.

Thanks in advance,
Tzury

pcap dumps below:

### REGISTER ###
User Datagram Protocol, Src Port: 12543 (12543), Dst Port: sip (5060)
REGISTER sip:example.net SIP/2.0
v: SIP/2.0/UDP 212.154.128.222:7608;branch=z9hG4bKPj41fa0abd044c35c75598d561b2f93167
Max-Forwards: 70
f: <sip:1013 at example.net>;tag=9194adecbf8e8179530eb589e83631db
t: <sip:1013 at example.net>
i: ebf2e3c254517938f90f33d8fd89326d
CSeq: 17355 REGISTER
m: <sip:1013 at 212.154.128.222:7608>
Expires: 300
l:  0

### 401 UNAUTHORIZED ###
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
212.154.128.222:7608;branch=z9hG4bKPj41fa0abd044c35c75598d561b2f93167
From: <sip:1013 at example.net>;tag=9194adecbf8e8179530eb589e83631db
To: <sip:1013 at example.net>;tag=166HraN0B149g
Call-ID: ebf2e3c254517938f90f33d8fd89326d
CSeq: 17355 REGISTER
User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
WWW-Authenticate: Digest realm="example.net",
nonce="f2f1370c-bc7c-11de-b000-fbe81e221ab4", algorithm=MD5,
qop="auth"
Content-Length: 0


From irmatov at gmail.com  Mon Oct 19 01:13:18 2009
From: irmatov at gmail.com (Timur Irmatov)
Date: Mon, 19 Oct 2009 13:13:18 +0500
Subject: [Freeswitch-users] set up sound on windows
Message-ID: <241d382f0910190113v5d2cc8b2kf28eabd2feec3062@mail.gmail.com>

Hi!


We are currently developing small call-center for our company using
FreeSWITCH. Our demands are pretty low at the moment, and we have
implemented all necessary features, now moving to testing.. We manage
FreeSWITCH through mod_erlang_event. FreeSWITCH and mod_erlang_event
are great, I want to thank all developers for those two great pieces
of technology!

We met strange problem with windows clients though. It is not related
to FreeSWITCH, but I don't know where to ask. Tried searching google,
but found nothing. As soon as we plug headset with microphone into
windows machine, we immediately start hearing environment though
headset. It seems as sound from microphone is transferred to earbuds,
which is pretty annoying.

Does anyone have any hints regarding configuration of audio/mics on Windows?


-- 
Timur Irmatov, xmpp:irmatov at jabber.ru


From durk.debeer at isp.solcon.nl  Mon Oct 19 00:11:24 2009
From: durk.debeer at isp.solcon.nl (Durk de Beer)
Date: Mon, 19 Oct 2009 09:11:24 +0200
Subject: [Freeswitch-users] Freeswitch 1.0.4 problems with music on hold
Message-ID: <mailman.0.1255957601.2492.freeswitch-users@lists.freeswitch.org>

Hello,

This is something I came across on Freeswitch 1.0.4

First let me explain what I'm trying to do.

I want Free-Switch to behave as a proxy so in the settings section of Sofia.conf.xml I use



and



As fare as I am able to follow the RTP stream it is passing by Freeswitch making it a proxy.

So far so good. Now I want to serv music on hold if a call is put on hold.

The only way I am able to find for Freeswitch to serv music on hold (MOH) is for it to be in media.

Ok we've got a command for this situation namely



Ok if I put the last parram command into the Sofia.conf.xml Freeswitch is starting to behave funny.

Let me try to explain this by an example. Let suppose Alice is calling Bob without the last command, the ?resume media on hold?.

If Bob puts Alice on hold al is silent on Bob's and Alice's phone.

If Bob takes Alice of hold there's two way audio.

Ok no I put the command in sofia.onf.xml, the ?resume media on hold?.

Now Alice is calling Bob again and Bob is putting her on hold again.

You are expecting MOH on this moment but it isn't, all there is are the sounds of silence.

Ok mistake made checking local streams, and they are there?. Yes it is.

Ok other mistake, didn't put the ?hold-music?command in sofia.conf.xml. Noop it's in there so there must be MOH. Codecs then?.

Noop there in vars.xml. Ok for me it is time to throw in the towel, I'll find it out later.
Ok Bob is now taking Alice of hold and surprise surprise there is only one way audio, between Alice and Bob.

I didn't have the time to look with Wireshark where al the steams go so sorry for that.
Did anyone came across the same problem with Freeswitch 1.0.4. I've a 1.0.2 running and that is doing everything a ok.



Kind regards,

 Durk

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From ineyaa at gmail.com  Mon Oct 19 05:16:02 2009
From: ineyaa at gmail.com (ineya ineya)
Date: Mon, 19 Oct 2009 14:16:02 +0200
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
Message-ID: <7b8592a0910190516rafa4398q62c7a99ef703540f@mail.gmail.com>

Hi,

I installed the phonenix release, and changed IPv4, IPv6 addresses and SIP
port. Directory and dialplan should be default ones.
My phones are able to register when using just IPv4, and I can call from one
to another.

When I try the same thing with IPv6, the call to other phone fails with [
USER_NOT_REGISTERED], however I'm able to call 9999 and hear MOH.
Here you can find the log: http://pastebin.freeswitch.org/10757
Freeswitch is running on 10.80.62.40/2000:2::1
phones are 2000:2::1001 and 2000:2::1002

Am I doing something wrong? Maybe I forgot to configure something, but I
can't see what it is.

Any hints are appreciated.
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From frank at carmickle.com  Mon Oct 19 06:07:17 2009
From: frank at carmickle.com (Frank Carmickle)
Date: Mon, 19 Oct 2009 09:07:17 -0400
Subject: [Freeswitch-users] Status of ubuntu/debian packages.
In-Reply-To: <4ADAFC85.6000301@puzzled.xs4all.nl>
References: <4ABA47FB.2050100@gmail.com>
	<90823c940910180249u69f18aaamc383f9a5a0c65848@mail.gmail.com>
	<4ADAFC85.6000301@puzzled.xs4all.nl>
Message-ID: <20091019130717.GG13895@base.carmickle.com>

On Sun, Oct 18, Patrick List wrote:
> so there was no chance of your customized configs getting overwritten. 
> Example:
> /opt/freeswitch/etc/freeswitch-$svn_version-samples/
> 
> Maybe the maintainer of the debs can do something similar.

We are planning on putting all of the different configs in /usr/share/doc/freeswitch/examples.  Now if I could find the time to get this done.

--FC


From frank at carmickle.com  Mon Oct 19 06:09:47 2009
From: frank at carmickle.com (Frank Carmickle)
Date: Mon, 19 Oct 2009 09:09:47 -0400
Subject: [Freeswitch-users] Status of ubuntu/debian packages.
In-Reply-To: <90823c940910181154w615b9ac5v89a30a3a3288e619@mail.gmail.com>
References: <4ABA47FB.2050100@gmail.com>
	<90823c940910180249u69f18aaamc383f9a5a0c65848@mail.gmail.com>
	<20091018164000.GD13895@base.carmickle.com>
	<90823c940910181154w615b9ac5v89a30a3a3288e619@mail.gmail.com>
Message-ID: <20091019130946.GH13895@base.carmickle.com>

On Sun, Oct 18, Dmitry Bely wrote:
> On Sun, Oct 18, 2009 at 8:40 PM, Frank Carmickle <frank at carmickle.com> wrote:
> > On Sun, Oct 18, Dmitry Bely wrote:
> >> William,
> >>
> >> When I upgrade freeswitch-config package, all my customized config
> >> files are silently overwritten with the default ones from .deb
> >> package. This bug is quite annoying.
> >
> > Very sorry. ?I can try to include some code that detects if your moving from the old versions that don't have a config package to the new stuff that does.
> 
> I did have the previous version of freeswitch-config installed but
> nevertheless the files were lost.

Well I am very sorry.  I wish I new why this happened for you.  I have not experienced this ever.  I will do some testing.

--FC


From mike at jerris.com  Mon Oct 19 06:10:47 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Oct 2009 09:10:47 -0400
Subject: [Freeswitch-users] NAT problems migrating from Version
	1.0.trunk (13168M) to Version 1.0.trunk (15166)
In-Reply-To: <1255922089.4653.1340747457@webmail.messagingengine.com>
References: <1255922089.4653.1340747457@webmail.messagingengine.com>
Message-ID: <9713E1DA-B66B-4A01-8864-49F71F8FF4EE@jerris.com>

we added some params for new better automatic nat handling, grep the  
new defailt configs for localnet and you will find what you are missing.

Mike

On Oct 18, 2009, at 11:14 PM, Chris Fowler wrote:

> I've tried all sorts of debug and parameter changes over the weekend,
> but still can't figure out the correct solution.
>
> If I disable timers on the sip profile then all works fine.
> <param name="enable-timer" value="false"/>
>
> But that seems like a hack; not a correct solution.  With the build
> 13168M (which is pre the new NAT functionality) everything worked  
> fine.
>
> The SIP trace shows the phones and FreeSWITCH happily exchanging  
> NOTIFY
> and "200 OK" messages.  Audio's working - just calls timeout after 100
> seconds with RECOVERY_ON_TIMER_EXPIRE.  Is enforcement of this timer  
> new
> functionality - and really just exposing a problem I've always had
> before?
>
> The config is (50 Polycom Phones -> NAT -> Internet -> Amazon EC2)
>
> I would really appreciate some pointers on what to look for;  
> additional
> trace that might reveal something.
>
> Thanks, Chris.
>
>
> On Fri, 16 Oct 2009 20:06:52 -0700, "Chris Fowler" <chris at fowler.cc>
> said:
>> Hi,
>>
>> We've been using 13168M in production for some time now (works  
>> great).
>> I want to get us onto the latest build but am having problems getting
>> NAT to work.
>>
>> Phones can register; can dial 9999 test #, but after 100 seconds the
>> call is disconnected with error:
>> 2009-10-16 19:52:26.936618 [NOTICE] sofia.c:4038 Hangup
>> sofia/internal/1001 at myhost.mydomain.com [CS_EXECUTE]
>> [RECOVERY_ON_TIMER_EXPIRE]
>>
>> I took the standard internal.xml and vars.xml files from the new  
>> build
>> and made the following modifications - which worked previously:
>>
>> modify conf/vars.xml and update
>>  <X-PRE-PROCESS cmd="set" data="domain=myhost.mydomain.com"/>
>>
>>  <X-PRE-PROCESS cmd="set" data="bind_server_ip=1.2.3.4"/>
>>  <X-PRE-PROCESS cmd="set" data="external_rtp_ip=1.2.3.4"/>
>>  <X-PRE-PROCESS cmd="set" data="external_sip_ip=1.2.3.4"/>
>>
>> Modify conf/sip_profiles/internal.xml
>> <param name="aggressive-nat-detection" value="true"/>  <param
>> name="ext-rtp-ip" value="$${external_rtp_ip}"/>  <param
>> name="ext-sip-ip" value="$${external_sip_ip}"/>  <param
>> name="NDLB-received-in-nat-reg-contact" value="true"/>  <param
>> name="NDLB-force-rport" value="true"/>  <param
>> name="NDLB-broken-auth-hash" value="true"/>
>>
>> The big difference I note is that on PRODUCTION (which works) "sofia
>> status profile internal" yields:
>> URL                     sip:mod_sofia at 1.2.3.4:5060
>> BIND-URL                 
>> sip:mod_sofia at 1.2.3.4:5060;maddr=10.250.35.224
>>
>> But on Test I see:
>> URL                     sip:mod_sofia at 10.250.66.210:5060
>> BIND-URL                sip:mod_sofia at 10.250.66.210:5060
>>
>> Any ideas?
>>
>> Thanks, Chris.



From srinivas.ksvreddy at gmail.com  Mon Oct 19 06:14:06 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Mon, 19 Oct 2009 18:44:06 +0530
Subject: [Freeswitch-users] Freeswitch.managed
Message-ID: <f8af5740910190614u75a98403icd5908c4a0663469@mail.gmail.com>

Hi,

How can i use freeswitch.managed project.  what are the parameters for
calling Execute method? and how can i call?
any help

-- 
Srinivasula Reddy K
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From mike at jerris.com  Mon Oct 19 06:19:25 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Oct 2009 09:19:25 -0400
Subject: [Freeswitch-users] uuid_send_dtmf fails (was: conference call)
In-Reply-To: <02C2EFD4CF8D4ACF8EEC0F77C4D75028@abisoft.biz>
References: <064A5201C10D4D20AC76EB93CA5EA4B5@abisoft.biz><ebbc2d550910140838v3125579ev9fa16e3e83d25770@mail.gmail.com><BE4EDDA84CB64AA98F18D412B51912FC@abisoft.biz>
	<b39273530910150504l5c7411b9g39b9ed33c592e85f@mail.gmail.com>
	<02C2EFD4CF8D4ACF8EEC0F77C4D75028@abisoft.biz>
Message-ID: <414DC781-A0F6-418E-9AAB-FA4307634932@jerris.com>

There is an event you can send as well to switch them, it your trying  
to switch it via event socket, that should be better, its not on the  
wiki, but

a session message with
eavesdrop-command header with data as the same as dtmf
should do the trick

Mike


On Oct 16, 2009, at 11:54 AM, Nikita Belov wrote:

> Yes, it is what I need. But now I have problem with sending dtmf.  
> Here what
> I've done:
> [root at centos4-4-vm ~]# telnet localhost 8021
> Trying 127.0.0.1...
> Connected to localhost.localdomain (127.0.0.1).
> Escape character is '^]'.
> Content-Type: auth/request
>
> auth ClueCon
>
> Content-Type: command/reply
> Reply-Text: +OK accepted
>
> api originate user/1007 at master.agent.starpoundtech.net &park()
>
> Content-Type: api/response
> Content-Length: 41
>
> +OK bba3b45a-4cc1-48af-a15d-1052d5f11371
>
> SendMsg bba3b45a-4cc1-48af-a15d-1052d5f11371
> call-command: execute
> execute-app-name: eavesdrop
> execute-app-arg: cd99f999-9b47-457e-8439-1d366e015b8c
>
> Content-Type: command/reply
> Reply-Text: +OK
>
> Here I had started to hear A and B. Here what I saw in FS log:
> 2009-10-18 03:22:47 [DEBUG] switch_core_session.c:706
> switch_core_session_queue_private_event() Send signal
> sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes [BREAK]
> 2009-10-18 03:22:47 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event()
> sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes Command Execute
> eavesdrop(cd99f999-9b47-457e-8439-1d366e015b8c)
> 2009-10-18 03:22:47 [DEBUG] switch_core_media_bug.c:297
> switch_core_media_bug_add() Attaching BUG to
> sofia/internal/1006 at master.agent.starpoundtech.net
> 2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234
> switch_core_session_read_frame()
> sofia/internal/1006 at master.agent.starpoundtech.net receive message
> [TRANSCODING_NECESSARY]
> 2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234
> switch_core_session_read_frame()
> sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes receive message
> [TRANSCODING_NECESSARY]
>
> Then I run command:
> api uuid_send_dtmf bba3b45a-4cc1-48af-a15d-1052d5f11371 1
>
> Content-Type: api/response
> Content-Length: 14
>
> -ERR no reply
>
> Log:
> 2009-10-18 03:24:01 [DEBUG] switch_core_io.c:1190
> switch_core_session_send_dtmf_string()
> sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes send dtmf
> digit=1 ms=250 samples=2000
> 2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1282 do_2833() Send start  
> packet
> for [1] ts=2241760 dur=160/160/2000 seq=21346
> 2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle  
> packet
> for [1] ts=2241760 dur=320/320/2000 seq=21347
> 2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle  
> packet
> for [1] ts=2241760 dur=480/480/2000 seq=21348
> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle  
> packet
> for [1] ts=2241760 dur=640/640/2000 seq=21349
> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle  
> packet
> for [1] ts=2241760 dur=800/800/2000 seq=21350
> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle  
> packet
> for [1] ts=2241760 dur=960/960/2000 seq=21351
> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle  
> packet
> for [1] ts=2241760 dur=1120/1120/2000 seq=21352
> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle  
> packet
> for [1] ts=2241760 dur=1280/1280/2000 seq=21353
> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle  
> packet
> for [1] ts=2241760 dur=1440/1440/2000 seq=21354
> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle  
> packet
> for [1] ts=2241760 dur=1600/1600/2000 seq=21355
> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle  
> packet
> for [1] ts=2241760 dur=1760/1760/2000 seq=21356
> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle  
> packet
> for [1] ts=2241760 dur=1920/1920/2000 seq=21357
> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end  
> packet for
> [1] ts=2241760 dur=2080/2080/2000 seq=21358
> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end  
> packet for
> [1] ts=2241760 dur=2080/2080/2000 seq=21359
> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end  
> packet for
> [1] ts=2241760 dur=2080/2080/2000 seq=21360
>
> But both A and B couldn't hear me.
> Btw, after I had send dtmf "1" manually from my phone. B started to  
> hear me.
> There was this record in log:
> 2009-10-18 03:47:55 [DEBUG] switch_rtp.c:1767 switch_rtp_dequeue_dtmf 
> () RTP
> RECV DTMF 1:2240
>
> Does anybody know, what had I done wrong?
>
> ___
>
> Thanks, Nikita
>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org  
>> [mailto:freeswitch-
>> users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker
>> Sent: Thursday, October 15, 2009 4:04 PM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] conference call
>>
>> The wiki has a pretty good explanation of how eavesdrop works.
>> Enabling a talk path to A or B or both A and B requires dtmf.
>>
>> So, if C hits the 1 button on the phone they can talk to the UUID you
>> bound the eavesdrop to.   2 talks to the other party.  3 talks to  
>> both
>> parties.
>>
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop
>>
>> Adding more api calls to control eavesdrop via event socket should be
>> possible, though I haven't looked at the code to see what it would
>> take...
>>
>> On Thu, Oct 15, 2009 at 3:16 AM, Nikita Belov <nbelov at abisoft.spb.ru>
>> wrote:
>>> Thanks for answer.
>>>
>>>
>>>
>>> As I've understood eavesdrop allows C to hear A and B, but A and B  
>>> will
>> not
>>> hear C. But in my case B SHOULD hear C. In my case A - client, B -  
>>> agent
>>> worked with client and C - coach/supervisor for B. Can I make such
>>> conference in FS using eavesdrop or other API command?
>>>
>>>
>>>
>>> ___
>>>
>>>
>>>
>>> Thanks, Nikita
>>>
>>>
>>>
>>> ________________________________
>>>
>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf  
>>> Of Joao
>>> Mesquita
>>> Sent: Wednesday, October 14, 2009 7:38 PM
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] conference call
>>>
>>>
>>>
>>> Look at eavesdrop on the wiki.
>>>
>>> JM
>>>
>>> 2009/10/14 Nikita Belov <nbelov at abisoft.spb.ru>
>>>
>>> HI all,
>>>
>>>
>>>
>>> I want to configure FS to make special conference call between three
>> users
>>> (A, B, C).  In this conference C will hear A and B, but A will  
>>> hear only
>> B.
>>> Can I make it using FS API commands? Does anybody know which  
>>> approach is
>>> better to use?
>>>
>>>
>>>
>>> ___
>>>
>>>
>>>
>>> Thanks, Nikita
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>>> users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>>> users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From srinivas.ksvreddy at gmail.com  Mon Oct 19 06:19:45 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Mon, 19 Oct 2009 18:49:45 +0530
Subject: [Freeswitch-users] Freeswitch.managed
In-Reply-To: <f8af5740910190614u75a98403icd5908c4a0663469@mail.gmail.com>
References: <f8af5740910190614u75a98403icd5908c4a0663469@mail.gmail.com>
Message-ID: <f8af5740910190619i32b83541s1ab238eebb87e757@mail.gmail.com>

Hi,

How can i use freeswitch.managed project.  what are the parameters for
calling Execute method? and how can i call?
any help

-- 
Srinivasula Reddy K
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From mike at jerris.com  Mon Oct 19 06:22:27 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Oct 2009 09:22:27 -0400
Subject: [Freeswitch-users] Freeswitch.managed
In-Reply-To: <f8af5740910190614u75a98403icd5908c4a0663469@mail.gmail.com>
References: <f8af5740910190614u75a98403icd5908c4a0663469@mail.gmail.com>
Message-ID: <7C661CE3-8D8F-4850-B751-D815F17DE49F@jerris.com>

Try starting out reading this.

http://wiki.freeswitch.org/wiki/Mod_managed

Mike

On Oct 19, 2009, at 9:14 AM, srinivasula reddy wrote:

> Hi,
>
> How can i use freeswitch.managed project.  what are the parameters  
> for calling Execute method? and how can i call?
> any help



From mike at jerris.com  Mon Oct 19 06:28:39 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Oct 2009 09:28:39 -0400
Subject: [Freeswitch-users] sofia gateways and linux multipath routing
In-Reply-To: <1255428585.11786.90.camel@francois.tc.commsmundi.com>
References: <1255428585.11786.90.camel@francois.tc.commsmundi.com>
Message-ID: <C0403714-EBBB-4758-8CF9-44E9D7E7F23C@jerris.com>

You need a sofia profile for each identity, if your using multiple  
external ip addresses, you will need a profile for each.  If you are  
using bgp or something of the sort and only using one external ip, you  
can use a single profile and route using standard routing.

Mike

On Oct 13, 2009, at 6:09 AM, Fran?ois Delawarde wrote:

> Hello all,
>
> I'm interested in using mod_sofia with multiple Internet connections
> (configured as a unique load-balancing route using multipath).
>
> One solution would be to define a different profile for each  
> connection,
> but it would be more practical having a unique external profile that
> would automatically handle everything (detecting multiple public IPs  
> and
> selecting the right one for a call, being able to select the router  
> for
> gateway registration...).
>
> Routing table:
> 192.168.10.0/24 dev eth0  proto kernel  scope link  src 192.168.10.1
> 192.168.1.0/24 dev eth1  proto kernel  scope link  src 192.168.1.2
> 192.168.2.0/24 dev eth2  proto kernel  scope link  src 192.168.2.2
> default  proto static
> 	nexthop via 192.168.1.1  dev eth1 weight 1
> 	nexthop via 192.168.2.1  dev eth2 weight 1
>
> Both default routers (192.168.1.1 and 192.168.2.1) would have a  
> distinct
> public IP.
>
> Several questions cross my mind:
> - Can a unique sofia profile be bound to multiple IPs (not 0.0.0.0)?
>
> - How would FS behave with a unique external profile in that  
> situation?
>   * Would FS reply to an incoming call using the same router it came
> from forcing packet source address?
>   * Would FS stick to a unique router for all flows of an outgoing  
> call
> (SIP, RTP, UDPTL)?
>   * Can I force a gateway to use a given router (for calls,
> registration, ...)?
>   * Would the NAT system (using stun or auto-nat) work in that
> situation, or does it assume only one default router (and a unique
> public IP) exists per profile?
>
> - Knowing the above, would it be necessary to use a different profile
> for each router/interface, and define the same gateway in each of  
> these?
>
> - Tricky question: What if multiple routers are on the same
> network/interface (192.168.1.1, 192.168.1.2, ...)?
>
>
> Thanks in advance,
> Fran?ois.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From mike at jerris.com  Mon Oct 19 06:30:02 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Oct 2009 09:30:02 -0400
Subject: [Freeswitch-users] TLS client authentification
In-Reply-To: <87747c930910151220p474e079bt609019e562698dfb@mail.gmail.com>
References: <87747c930910151220p474e079bt609019e562698dfb@mail.gmail.com>
Message-ID: <0FA87111-29F1-44E1-BE3A-2B8B2F841704@jerris.com>

I can't recall if we ever exposed an option for this, take a look at  
sofia-sip and see if they have a tag to enable this, if so it would  
probably be a fairly simple patch to add.

Mike

On Oct 15, 2009, at 3:20 PM, Alexandre Savard wrote:

> Hi,
>
> Does Freeswitch support TLS Client-Authenticated handshake. Openssl  
> does, but it has to be enabled in order to send the certificate  
> request to the client.
>
> I tested simple TLS hanshaking and it wotks well.



From mike at jerris.com  Mon Oct 19 06:32:28 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Oct 2009 09:32:28 -0400
Subject: [Freeswitch-users] validating dtmf digits received
In-Reply-To: <33c87fa30910182153t23e52fa0p2b967b3843eb1175@mail.gmail.com>
References: <33c87fa30910182153t23e52fa0p2b967b3843eb1175@mail.gmail.com>
Message-ID: <81433357-3203-4208-98C1-FF3FE2AC2D1E@jerris.com>

inline is new, it won't work unless your using recent trunk.  That  
being said, read is not being run inline, so the set is actually being  
run before digits_dialed is set.  You will most likely need to use  
transfer in this situation.

Mike

On Oct 19, 2009, at 12:53 AM, Mark Campbell-Smith wrote:

> Hi!
>
> I simply want to validate the dtmf digits I read from a user.    From
> the wiki, it appears I need to use inline="true" when setting the
> variable so it can be used directly within the same extension.
>
> What have I done wrong below?   I have tried many different
> alternatives, but the second condition field, which is meant to match
> the dtmf digits received (in this case 0000) is never matched, and the
> anti-action is called instead.
>
> :
>      <some code here>
>      <action application="read" data="1 10
> ivr/ivr-please_enter_pin_followed_by_pound.wav res 10000 9"/>
>       <action application="phrase" data="spell,${res}"/>
>       <action inline="true" application="set" data="code=$ 
> {digits_dialed}"/>
>       <!-- <action inline="true" application="set" data="code=$ 
> {res}"/> -->
>  </condition>
>  <condition field="digits_dialed" expression="^0000$">
>  <!-- <condition field="${code}" expression="^0000$"> -->
>  <!-- <condition field="${res}" expression="^0000$"> -->
>        <some code here>
>       <anti-action application="hangup"/>
>
> Thanks!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From mike at jerris.com  Mon Oct 19 06:34:47 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Oct 2009 09:34:47 -0400
Subject: [Freeswitch-users] How to enforce freeswitch replying to the
	source port instead of to the one specified in v/m header
	parametes (Symmetric NAT)
In-Reply-To: <10128ef10910190023t17574fafic892bd4105cc1703@mail.gmail.com>
References: <10128ef10910190023t17574fafic892bd4105cc1703@mail.gmail.com>
Message-ID: <06D9255C-2D72-4BD1-A17E-922C55A76C56@jerris.com>

This is what force-rport is supposed to do.  That being said, I can't  
tell from your trace where it is actually going to, just what it says  
in the packet, which can be different.

Mike

On Oct 19, 2009, at 3:23 AM, Tzury Bar Yochay wrote:

> Hi,
>
> I am struggling with a cellular operator which removes the rport from
> the SIP messages sent by  the client.
> During the troubleshooting process I have been playing with the NDLB
> parameters I found mentioned at the fs-wiki
>
> <param name="NDLB-force-rport" value="true"/>
> <param name="NDLB-received-in-nat-reg-contact" value="true"/>
>
> as well as with sip-force-contact at the extension xml config file.
>
> <variables>
>    <variable name="sip-force-contact" value="NDLB-connectile- 
> dysfunction"/>
> </variables>
>
> I Want to make freeswitch replying to the source port instead of the
> port supplied by the client, that is
> to port 12543 instead of to the 7608 as you can see below some pcap
> dumps of the case.
>
> I am not sure if that would solve thep roblem and am open to other
> suggestions as well.
>
> Thanks in advance,
> Tzury
>
> pcap dumps below:
>
> ### REGISTER ###
> User Datagram Protocol, Src Port: 12543 (12543), Dst Port: sip (5060)
> REGISTER sip:example.net SIP/2.0
> v: SIP/2.0/UDP  
> 212.154.128.222:7608;branch=z9hG4bKPj41fa0abd044c35c75598d561b2f93167
> Max-Forwards: 70
> f: <sip:1013 at example.net>;tag=9194adecbf8e8179530eb589e83631db
> t: <sip:1013 at example.net>
> i: ebf2e3c254517938f90f33d8fd89326d
> CSeq: 17355 REGISTER
> m: <sip:1013 at 212.154.128.222:7608>
> Expires: 300
> l:  0
>
> ### 401 UNAUTHORIZED ###
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP
> 212.154.128.222:7608;branch=z9hG4bKPj41fa0abd044c35c75598d561b2f93167
> From: <sip:1013 at example.net>;tag=9194adecbf8e8179530eb589e83631db
> To: <sip:1013 at example.net>;tag=166HraN0B149g
> Call-ID: ebf2e3c254517938f90f33d8fd89326d
> CSeq: 17355 REGISTER
> User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
> Supported: timer, precondition, path, replaces
> WWW-Authenticate: Digest realm="example.net",
> nonce="f2f1370c-bc7c-11de-b000-fbe81e221ab4", algorithm=MD5,
> qop="auth"
> Content-Length: 0



From mike at jerris.com  Mon Oct 19 06:47:41 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Oct 2009 09:47:41 -0400
Subject: [Freeswitch-users] Freeswitch 1.0.4 problems with music on hold
In-Reply-To: <4adc679c.a013f30a.539b.ffffb45dSMTPIN_ADDED@mx.google.com>
References: <4adc679c.a013f30a.539b.ffffb45dSMTPIN_ADDED@mx.google.com>
Message-ID: <359F2E83-6A5A-4CE9-B12D-118D8E9BA325@jerris.com>

Try out trunk and see if this issue is resolved please.

mike

On Oct 19, 2009, at 3:11 AM, Durk de Beer wrote:

> Hello,
> This is something I came across on Freeswitch 1.0.4
> First let me explain what I'm trying to do.
> I want Free-Switch to behave as a proxy so in the settings section  
> of Sofia.conf.xml I use
> <param name="media-option" value="bypass-media-after-att-xfer"/>
> and
> <param name="inbound-bypass-media" value="true"/>
> As fare as I am able to follow the RTP stream it is passing by  
> Freeswitch making it a proxy.
> So far so good. Now I want to serv music on hold if a call is put on  
> hold.
> The only way I am able to find for Freeswitch to serv music on hold  
> (MOH) is for it to be in media.
> Ok we've got a command for this situation namely
> <param name="media-option" value="resume-media-on-hold"/>
> Ok if I put the last parram command into the Sofia.conf.xml  
> Freeswitch is starting to behave funny.
> Let me try to explain this by an example. Let suppose Alice is  
> calling Bob without the last command, the ?resume media on hold?.
> If Bob puts Alice on hold al is silent on Bob's and Alice's phone.
> If Bob takes Alice of hold there's two way audio.
> Ok no I put the command in sofia.onf.xml, the ?resume media on hold?.
> Now Alice is calling Bob again and Bob is putting her on hold again.
> You are expecting MOH on this moment but it isn't, all there is are  
> the sounds of silence.
> Ok mistake made checking local streams, and they are there?. Yes it  
> is.
> Ok other mistake, didn't put the ?hold-music?command in  
> sofia.conf.xml. Noop it's in there so there must be MOH. Codecs then?.
> Noop there in vars.xml. Ok for me it is time to throw in the towel,  
> I'll find it out later.
> Ok Bob is now taking Alice of hold and surprise surprise there is  
> only one way audio, between Alice and Bob.
> I didn't have the time to look with Wireshark where al the steams go  
> so sorry for that.
> Did anyone came across the same problem with Freeswitch 1.0.4. I've  
> a 1.0.2 running and that is doing everything a ok.

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From helmut.kuper at ewetel.de  Mon Oct 19 07:47:37 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Mon, 19 Oct 2009 16:47:37 +0200
Subject: [Freeswitch-users] Qustion about INFO messages
	after	Connect/Answer
In-Reply-To: <191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>
References: <4AD859A4.5000101@ewetel.de>
	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>
Message-ID: <4ADC7C09.6070503@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello Anthony,

I updated and restarted my test FS to "FreeSWITCH Version 1.0.trunk
(15174M)". Callee's experience didn't change:

> 1. Phone rings: caller's displayname
> 2. Callee picks up: switching from dislayname to unknown
> 3. Switching from unknown to displayname

I used the two chvars you mentioned, set them via "set" and as well via
"export" but no change (neither on caller's nor in callee's display nor
in SIP INFO messages)


My dialplan portion for this is:
    <extension name="Local_Extension">
      <condition field="${ET_is_local}" expression="^true$">
        <action application="set"
data="dialed_extension=${destination_number}"/>
        <action application="set" data="sip_callee_id_number=1111"/>
        <action application="set" data="sip_callee_id_name=hubu"/>
        <action application="export" data="sip_callee_id_number=1111"/>
        <action application="export" data="sip_callee_id_name=hubu"/>
        <action application="info"/>
[...]


Here is the output of the info app after setting those chvars:

[INFO] mod_dptools.c:961 CHANNEL_DATA:
Channel-State: [CS_EXECUTE]
Channel-State-Number: [4]
Channel-Name: [sofia/internal/1001 at 85.16.246.12:5061]
Unique-ID: [4b143750-bcbd-11de-9f91-c9cd82739033]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Answer-State: [ringing]
Caller-Username: [1001]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [1001 an PBX1]
Caller-Caller-ID-Number: [1001]
Caller-Network-Addr: [85.16.245.206]
Caller-Destination-Number: [1000]
Caller-Unique-ID: [4b143750-bcbd-11de-9f91-c9cd82739033]
Caller-Source: [mod_sofia]
Caller-Context: [default]
Caller-RDNIS: [1000]
Caller-Channel-Name: [sofia/internal/1001 at 85.16.246.12:5061]
Caller-Profile-Index: [2]
Caller-Profile-Created-Time: [1255963206242587]
Caller-Channel-Created-Time: [1255963206214959]
Caller-Channel-Answered-Time: [0]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
variable_sip_received_ip: [85.16.245.206]
variable_sip_received_port: [1024]
variable_sip_via_protocol: [udp]
variable_sip_authorized: [true]
variable_sip_from_user: [1001]
variable_sip_from_port: [5061]
variable_sip_from_uri: [1001 at 85.16.246.12:5061]
variable_sip_from_host: [85.16.246.12]
variable_sip_from_user_stripped: [1001]
variable_sip_from_tag: [snfuiue6ga]
variable_sofia_profile_name: [internal]
variable_sip_req_params: [user=phone]
variable_sip_req_user: [1000]
variable_sip_req_port: [5061]
variable_sip_req_uri: [1000 at 85.16.246.12:5061]
variable_sip_req_host: [85.16.246.12]
variable_sip_to_params: [user=phone]
variable_sip_to_user: [1000]
variable_sip_to_port: [5061]
variable_sip_to_uri: [1000 at 85.16.246.12:5061]
variable_sip_to_host: [85.16.246.12]
variable_sip_contact_params: [line=eg3wp69a]
variable_sip_contact_user: [1001]
variable_sip_contact_port: [1024]
variable_sip_contact_uri: [1001 at 85.16.245.206:1024]
variable_sip_contact_host: [85.16.245.206]
variable_channel_name: [sofia/internal/1001 at 85.16.246.12:5061]
variable_sip_call_id: [3c2d2d8f9a49-edzr2i2iezjp]
variable_sip_user_agent: [snom820/8.2.16]
variable_sip_via_host: [85.16.245.206]
variable_sip_via_port: [1024]
variable_sip_via_rport: [1024]
variable_presence_id: [1001 at 85.16.246.12]
variable_sip_h_X-Serialnumber: [0004134002CB]
variable_sip_h_P-Key-Flags: [resolution="31x13", keys="4"]
variable_switch_r_sdp: [v=0
o=root 1331667919 1331667919 IN IP4 85.16.245.206
s=call
c=IN IP4 85.16.245.206
t=0 0
m=audio 62882 RTP/SAVP 0 8 9 99 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:m6fas/KsLF57r9RnU7X0WEWeJw9Y6+a66YUIf9Dc
a=ptime:20
m=audio 62882 RTP/AVP 0 8 9 99 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
]
variable_ep_codec_string: [G722 at 8000h@20i,PCMA at 8000h@20i]
variable_endpoint_disposition: [DELAYED NEGOTIATION]
variable_ET_is_local: [true]
variable_max_forwards: [69]
variable_domain_name: [85.16.246.12]
variable_dialed_extension: [1000]
variable_sip_callee_id_number: [1111]
variable_sip_callee_id_name: [hubu]
variable_export_vars: [sip_callee_id_number,sip_callee_id_name]
variable_current_application: [info]


On 16.10.2009 18:18, Anthony Minessale wrote:
> 1) you should update again there were a few issues.
> 2) you can set the variable sip_callee_id_name and sip_callee_id number
> on the inbound leg before you answer to control what it says.

regards
Helmut
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From t.mahe at telemaque.fr  Mon Oct 19 07:55:58 2009
From: t.mahe at telemaque.fr (=?ISO-8859-1?Q?Tristan_Mah=E9?=)
Date: Mon, 19 Oct 2009 16:55:58 +0200
Subject: [Freeswitch-users] Qustion about INFO
	messages	after	Connect/Answer
In-Reply-To: <4ADC7C09.6070503@ewetel.de>
References: <4AD859A4.5000101@ewetel.de>	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>
	<4ADC7C09.6070503@ewetel.de>
Message-ID: <4ADC7DFE.5000305@telemaque.fr>

Hi Helmut,

Just to add my 2 cents to the discussion, I have the same behaviour there...

Regards,

Gled.

Helmut Kuper a ?crit :
> Hello Anthony,
>
> I updated and restarted my test FS to "FreeSWITCH Version 1.0.trunk
> (15174M)". Callee's experience didn't change:
>
> > 1. Phone rings: caller's displayname
> > 2. Callee picks up: switching from dislayname to unknown
> > 3. Switching from unknown to displayname
>
> I used the two chvars you mentioned, set them via "set" and as well via
> "export" but no change (neither on caller's nor in callee's display nor
> in SIP INFO messages)
>
>
> My dialplan portion for this is:
>     <extension name="Local_Extension">
>       <condition field="${ET_is_local}" expression="^true$">
>         <action application="set"
> data="dialed_extension=${destination_number}"/>
>         <action application="set" data="sip_callee_id_number=1111"/>
>         <action application="set" data="sip_callee_id_name=hubu"/>
>         <action application="export" data="sip_callee_id_number=1111"/>
>         <action application="export" data="sip_callee_id_name=hubu"/>
>         <action application="info"/>
> [...]
>
>
> Here is the output of the info app after setting those chvars:
>
> [INFO] mod_dptools.c:961 CHANNEL_DATA:
> Channel-State: [CS_EXECUTE]
> Channel-State-Number: [4]
> Channel-Name: [sofia/internal/1001 at 85.16.246.12:5061]
> Unique-ID: [4b143750-bcbd-11de-9f91-c9cd82739033]
> Call-Direction: [inbound]
> Presence-Call-Direction: [inbound]
> Answer-State: [ringing]
> Caller-Username: [1001]
> Caller-Dialplan: [XML]
> Caller-Caller-ID-Name: [1001 an PBX1]
> Caller-Caller-ID-Number: [1001]
> Caller-Network-Addr: [85.16.245.206]
> Caller-Destination-Number: [1000]
> Caller-Unique-ID: [4b143750-bcbd-11de-9f91-c9cd82739033]
> Caller-Source: [mod_sofia]
> Caller-Context: [default]
> Caller-RDNIS: [1000]
> Caller-Channel-Name: [sofia/internal/1001 at 85.16.246.12:5061]
> Caller-Profile-Index: [2]
> Caller-Profile-Created-Time: [1255963206242587]
> Caller-Channel-Created-Time: [1255963206214959]
> Caller-Channel-Answered-Time: [0]
> Caller-Channel-Progress-Time: [0]
> Caller-Channel-Progress-Media-Time: [0]
> Caller-Channel-Hangup-Time: [0]
> Caller-Channel-Transfer-Time: [0]
> Caller-Screen-Bit: [true]
> Caller-Privacy-Hide-Name: [false]
> Caller-Privacy-Hide-Number: [false]
> variable_sip_received_ip: [85.16.245.206]
> variable_sip_received_port: [1024]
> variable_sip_via_protocol: [udp]
> variable_sip_authorized: [true]
> variable_sip_from_user: [1001]
> variable_sip_from_port: [5061]
> variable_sip_from_uri: [1001 at 85.16.246.12:5061]
> variable_sip_from_host: [85.16.246.12]
> variable_sip_from_user_stripped: [1001]
> variable_sip_from_tag: [snfuiue6ga]
> variable_sofia_profile_name: [internal]
> variable_sip_req_params: [user=phone]
> variable_sip_req_user: [1000]
> variable_sip_req_port: [5061]
> variable_sip_req_uri: [1000 at 85.16.246.12:5061]
> variable_sip_req_host: [85.16.246.12]
> variable_sip_to_params: [user=phone]
> variable_sip_to_user: [1000]
> variable_sip_to_port: [5061]
> variable_sip_to_uri: [1000 at 85.16.246.12:5061]
> variable_sip_to_host: [85.16.246.12]
> variable_sip_contact_params: [line=eg3wp69a]
> variable_sip_contact_user: [1001]
> variable_sip_contact_port: [1024]
> variable_sip_contact_uri: [1001 at 85.16.245.206:1024]
> variable_sip_contact_host: [85.16.245.206]
> variable_channel_name: [sofia/internal/1001 at 85.16.246.12:5061]
> variable_sip_call_id: [3c2d2d8f9a49-edzr2i2iezjp]
> variable_sip_user_agent: [snom820/8.2.16]
> variable_sip_via_host: [85.16.245.206]
> variable_sip_via_port: [1024]
> variable_sip_via_rport: [1024]
> variable_presence_id: [1001 at 85.16.246.12]
> variable_sip_h_X-Serialnumber: [0004134002CB]
> variable_sip_h_P-Key-Flags: [resolution="31x13", keys="4"]
> variable_switch_r_sdp: [v=0
> o=root 1331667919 1331667919 IN IP4 85.16.245.206
> s=call
> c=IN IP4 85.16.245.206
> t=0 0
> m=audio 62882 RTP/SAVP 0 8 9 99 3 18 4 101
> a=rtpmap:0 pcmu/8000
> a=rtpmap:8 pcma/8000
> a=rtpmap:9 g722/8000
> a=rtpmap:99 g726-32/8000
> a=rtpmap:3 gsm/8000
> a=rtpmap:18 g729/8000
> a=rtpmap:4 g723/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=crypto:1 AES_CM_128_HMAC_SHA1_32
> inline:m6fas/KsLF57r9RnU7X0WEWeJw9Y6+a66YUIf9Dc
> a=ptime:20
> m=audio 62882 RTP/AVP 0 8 9 99 3 18 4 101
> a=rtpmap:0 pcmu/8000
> a=rtpmap:8 pcma/8000
> a=rtpmap:9 g722/8000
> a=rtpmap:99 g726-32/8000
> a=rtpmap:3 gsm/8000
> a=rtpmap:18 g729/8000
> a=rtpmap:4 g723/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> ]
> variable_ep_codec_string: [G722 at 8000h@20i,PCMA at 8000h@20i]
> variable_endpoint_disposition: [DELAYED NEGOTIATION]
> variable_ET_is_local: [true]
> variable_max_forwards: [69]
> variable_domain_name: [85.16.246.12]
> variable_dialed_extension: [1000]
> variable_sip_callee_id_number: [1111]
> variable_sip_callee_id_name: [hubu]
> variable_export_vars: [sip_callee_id_number,sip_callee_id_name]
> variable_current_application: [info]
>
>
> On 16.10.2009 18:18, Anthony Minessale wrote:
> > 1) you should update again there were a few issues.
> > 2) you can set the variable sip_callee_id_name and sip_callee_id number
> > on the inbound leg before you answer to control what it says.
>
> regards
> Helmut

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From anthony.minessale at gmail.com  Mon Oct 19 08:00:49 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 19 Oct 2009 10:00:49 -0500
Subject: [Freeswitch-users] Qustion about INFO messages after
	Connect/Answer
In-Reply-To: <4ADC7C09.6070503@ewetel.de>
References: <4AD859A4.5000101@ewetel.de>
	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>
	<4ADC7C09.6070503@ewetel.de>
Message-ID: <191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>

you only need to "set" it on the inbound leg and you must answer and bridge
it somewhere.


On Mon, Oct 19, 2009 at 9:47 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello Anthony,
>
> I updated and restarted my test FS to "FreeSWITCH Version 1.0.trunk
> (15174M)". Callee's experience didn't change:
>
> > 1. Phone rings: caller's displayname
> > 2. Callee picks up: switching from dislayname to unknown
> > 3. Switching from unknown to displayname
>
> I used the two chvars you mentioned, set them via "set" and as well via
> "export" but no change (neither on caller's nor in callee's display nor
> in SIP INFO messages)
>
>
> My dialplan portion for this is:
>    <extension name="Local_Extension">
>      <condition field="${ET_is_local}" expression="^true$">
>        <action application="set"
> data="dialed_extension=${destination_number}"/>
>        <action application="set" data="sip_callee_id_number=1111"/>
>        <action application="set" data="sip_callee_id_name=hubu"/>
>        <action application="export" data="sip_callee_id_number=1111"/>
>        <action application="export" data="sip_callee_id_name=hubu"/>
>        <action application="info"/>
> [...]
>
>
> Here is the output of the info app after setting those chvars:
>
> [INFO] mod_dptools.c:961 CHANNEL_DATA:
> Channel-State: [CS_EXECUTE]
> Channel-State-Number: [4]
> Channel-Name: [sofia/internal/1001 at 85.16.246.12:5061]
> Unique-ID: [4b143750-bcbd-11de-9f91-c9cd82739033]
> Call-Direction: [inbound]
> Presence-Call-Direction: [inbound]
> Answer-State: [ringing]
> Caller-Username: [1001]
> Caller-Dialplan: [XML]
> Caller-Caller-ID-Name: [1001 an PBX1]
> Caller-Caller-ID-Number: [1001]
> Caller-Network-Addr: [85.16.245.206]
> Caller-Destination-Number: [1000]
> Caller-Unique-ID: [4b143750-bcbd-11de-9f91-c9cd82739033]
> Caller-Source: [mod_sofia]
> Caller-Context: [default]
> Caller-RDNIS: [1000]
> Caller-Channel-Name: [sofia/internal/1001 at 85.16.246.12:5061]
> Caller-Profile-Index: [2]
> Caller-Profile-Created-Time: [1255963206242587]
> Caller-Channel-Created-Time: [1255963206214959]
> Caller-Channel-Answered-Time: [0]
> Caller-Channel-Progress-Time: [0]
> Caller-Channel-Progress-Media-Time: [0]
> Caller-Channel-Hangup-Time: [0]
> Caller-Channel-Transfer-Time: [0]
> Caller-Screen-Bit: [true]
> Caller-Privacy-Hide-Name: [false]
> Caller-Privacy-Hide-Number: [false]
> variable_sip_received_ip: [85.16.245.206]
> variable_sip_received_port: [1024]
> variable_sip_via_protocol: [udp]
> variable_sip_authorized: [true]
> variable_sip_from_user: [1001]
> variable_sip_from_port: [5061]
> variable_sip_from_uri: [1001 at 85.16.246.12:5061]
> variable_sip_from_host: [85.16.246.12]
> variable_sip_from_user_stripped: [1001]
> variable_sip_from_tag: [snfuiue6ga]
> variable_sofia_profile_name: [internal]
> variable_sip_req_params: [user=phone]
> variable_sip_req_user: [1000]
> variable_sip_req_port: [5061]
> variable_sip_req_uri: [1000 at 85.16.246.12:5061]
> variable_sip_req_host: [85.16.246.12]
> variable_sip_to_params: [user=phone]
> variable_sip_to_user: [1000]
> variable_sip_to_port: [5061]
> variable_sip_to_uri: [1000 at 85.16.246.12:5061]
> variable_sip_to_host: [85.16.246.12]
> variable_sip_contact_params: [line=eg3wp69a]
> variable_sip_contact_user: [1001]
> variable_sip_contact_port: [1024]
> variable_sip_contact_uri: [1001 at 85.16.245.206:1024]
> variable_sip_contact_host: [85.16.245.206]
> variable_channel_name: [sofia/internal/1001 at 85.16.246.12:5061]
> variable_sip_call_id: [3c2d2d8f9a49-edzr2i2iezjp]
> variable_sip_user_agent: [snom820/8.2.16]
> variable_sip_via_host: [85.16.245.206]
> variable_sip_via_port: [1024]
> variable_sip_via_rport: [1024]
> variable_presence_id: [1001 at 85.16.246.12]
> variable_sip_h_X-Serialnumber: [0004134002CB]
> variable_sip_h_P-Key-Flags: [resolution="31x13", keys="4"]
> variable_switch_r_sdp: [v=0
> o=root 1331667919 1331667919 IN IP4 85.16.245.206
> s=call
> c=IN IP4 85.16.245.206
> t=0 0
> m=audio 62882 RTP/SAVP 0 8 9 99 3 18 4 101
> a=rtpmap:0 pcmu/8000
> a=rtpmap:8 pcma/8000
> a=rtpmap:9 g722/8000
> a=rtpmap:99 g726-32/8000
> a=rtpmap:3 gsm/8000
> a=rtpmap:18 g729/8000
> a=rtpmap:4 g723/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=crypto:1 AES_CM_128_HMAC_SHA1_32
> inline:m6fas/KsLF57r9RnU7X0WEWeJw9Y6+a66YUIf9Dc
> a=ptime:20
> m=audio 62882 RTP/AVP 0 8 9 99 3 18 4 101
> a=rtpmap:0 pcmu/8000
> a=rtpmap:8 pcma/8000
> a=rtpmap:9 g722/8000
> a=rtpmap:99 g726-32/8000
> a=rtpmap:3 gsm/8000
> a=rtpmap:18 g729/8000
> a=rtpmap:4 g723/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> ]
> variable_ep_codec_string: [G722 at 8000h@20i,PCMA at 8000h@20i]
> variable_endpoint_disposition: [DELAYED NEGOTIATION]
> variable_ET_is_local: [true]
> variable_max_forwards: [69]
> variable_domain_name: [85.16.246.12]
> variable_dialed_extension: [1000]
> variable_sip_callee_id_number: [1111]
> variable_sip_callee_id_name: [hubu]
> variable_export_vars: [sip_callee_id_number,sip_callee_id_name]
> variable_current_application: [info]
>
>
> On 16.10.2009 18:18, Anthony Minessale wrote:
> > 1) you should update again there were a few issues.
> > 2) you can set the variable sip_callee_id_name and sip_callee_id number
> > on the inbound leg before you answer to control what it says.
>
> regards
> Helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
>
> iD8DBQFK3HwJ4tZeNddg3dwRAieqAKDCn47dzctaYpweRGzovbMPcKD5KQCgjNx+
> nNTz4r7+N7mI3Wj4GayFdTk=
> =MYOb
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
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Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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From nbelov at abisoft.spb.ru  Mon Oct 19 08:11:03 2009
From: nbelov at abisoft.spb.ru (Nikita Belov)
Date: Mon, 19 Oct 2009 19:11:03 +0400
Subject: [Freeswitch-users] uuid_send_dtmf fails (was: conference call)
In-Reply-To: <414DC781-A0F6-418E-9AAB-FA4307634932@jerris.com>
References: <064A5201C10D4D20AC76EB93CA5EA4B5@abisoft.biz><ebbc2d550910140838v3125579ev9fa16e3e83d25770@mail.gmail.com><BE4EDDA84CB64AA98F18D412B51912FC@abisoft.biz><b39273530910150504l5c7411b9g39b9ed33c592e85f@mail.gmail.com><02C2EFD4CF8D4ACF8EEC0F77C4D75028@abisoft.biz>
	<414DC781-A0F6-418E-9AAB-FA4307634932@jerris.com>
Message-ID: <14E8C50A8F4D41559AFCCD8838350F4F@abisoft.biz>

Thanks, Mike, for idea. But what is the syntax for this session message?
I tried this:

sendmsg e8e4f0ed-a0cc-4dff-b7e1-09eeade5df05
eavesdrop-command: 1

but it doesn't work.


> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
> Sent: Monday, October 19, 2009 5:19 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] uuid_send_dtmf fails (was: conference
> call)
> 
> There is an event you can send as well to switch them, it your trying
> to switch it via event socket, that should be better, its not on the
> wiki, but
> 
> a session message with
> eavesdrop-command header with data as the same as dtmf
> should do the trick
> 
> Mike
> 
> 
> On Oct 16, 2009, at 11:54 AM, Nikita Belov wrote:
> 
> > Yes, it is what I need. But now I have problem with sending dtmf.
> > Here what
> > I've done:
> > [root at centos4-4-vm ~]# telnet localhost 8021
> > Trying 127.0.0.1...
> > Connected to localhost.localdomain (127.0.0.1).
> > Escape character is '^]'.
> > Content-Type: auth/request
> >
> > auth ClueCon
> >
> > Content-Type: command/reply
> > Reply-Text: +OK accepted
> >
> > api originate user/1007 at master.agent.starpoundtech.net &park()
> >
> > Content-Type: api/response
> > Content-Length: 41
> >
> > +OK bba3b45a-4cc1-48af-a15d-1052d5f11371
> >
> > SendMsg bba3b45a-4cc1-48af-a15d-1052d5f11371
> > call-command: execute
> > execute-app-name: eavesdrop
> > execute-app-arg: cd99f999-9b47-457e-8439-1d366e015b8c
> >
> > Content-Type: command/reply
> > Reply-Text: +OK
> >
> > Here I had started to hear A and B. Here what I saw in FS log:
> > 2009-10-18 03:22:47 [DEBUG] switch_core_session.c:706
> > switch_core_session_queue_private_event() Send signal
> > sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes [BREAK]
> > 2009-10-18 03:22:47 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event()
> > sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes Command Execute
> > eavesdrop(cd99f999-9b47-457e-8439-1d366e015b8c)
> > 2009-10-18 03:22:47 [DEBUG] switch_core_media_bug.c:297
> > switch_core_media_bug_add() Attaching BUG to
> > sofia/internal/1006 at master.agent.starpoundtech.net
> > 2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234
> > switch_core_session_read_frame()
> > sofia/internal/1006 at master.agent.starpoundtech.net receive message
> > [TRANSCODING_NECESSARY]
> > 2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234
> > switch_core_session_read_frame()
> > sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes receive message
> > [TRANSCODING_NECESSARY]
> >
> > Then I run command:
> > api uuid_send_dtmf bba3b45a-4cc1-48af-a15d-1052d5f11371 1
> >
> > Content-Type: api/response
> > Content-Length: 14
> >
> > -ERR no reply
> >
> > Log:
> > 2009-10-18 03:24:01 [DEBUG] switch_core_io.c:1190
> > switch_core_session_send_dtmf_string()
> > sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes send dtmf
> > digit=1 ms=250 samples=2000
> > 2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1282 do_2833() Send start
> > packet
> > for [1] ts=2241760 dur=160/160/2000 seq=21346
> > 2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
> > packet
> > for [1] ts=2241760 dur=320/320/2000 seq=21347
> > 2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
> > packet
> > for [1] ts=2241760 dur=480/480/2000 seq=21348
> > 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
> > packet
> > for [1] ts=2241760 dur=640/640/2000 seq=21349
> > 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
> > packet
> > for [1] ts=2241760 dur=800/800/2000 seq=21350
> > 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
> > packet
> > for [1] ts=2241760 dur=960/960/2000 seq=21351
> > 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
> > packet
> > for [1] ts=2241760 dur=1120/1120/2000 seq=21352
> > 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
> > packet
> > for [1] ts=2241760 dur=1280/1280/2000 seq=21353
> > 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
> > packet
> > for [1] ts=2241760 dur=1440/1440/2000 seq=21354
> > 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
> > packet
> > for [1] ts=2241760 dur=1600/1600/2000 seq=21355
> > 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
> > packet
> > for [1] ts=2241760 dur=1760/1760/2000 seq=21356
> > 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
> > packet
> > for [1] ts=2241760 dur=1920/1920/2000 seq=21357
> > 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end
> > packet for
> > [1] ts=2241760 dur=2080/2080/2000 seq=21358
> > 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end
> > packet for
> > [1] ts=2241760 dur=2080/2080/2000 seq=21359
> > 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end
> > packet for
> > [1] ts=2241760 dur=2080/2080/2000 seq=21360
> >
> > But both A and B couldn't hear me.
> > Btw, after I had send dtmf "1" manually from my phone. B started to
> > hear me.
> > There was this record in log:
> > 2009-10-18 03:47:55 [DEBUG] switch_rtp.c:1767 switch_rtp_dequeue_dtmf
> > () RTP
> > RECV DTMF 1:2240
> >
> > Does anybody know, what had I done wrong?
> >
> > ___
> >
> > Thanks, Nikita
> >
> >> -----Original Message-----
> >> From: freeswitch-users-bounces at lists.freeswitch.org
> >> [mailto:freeswitch-
> >> users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker
> >> Sent: Thursday, October 15, 2009 4:04 PM
> >> To: freeswitch-users at lists.freeswitch.org
> >> Subject: Re: [Freeswitch-users] conference call
> >>
> >> The wiki has a pretty good explanation of how eavesdrop works.
> >> Enabling a talk path to A or B or both A and B requires dtmf.
> >>
> >> So, if C hits the 1 button on the phone they can talk to the UUID you
> >> bound the eavesdrop to.   2 talks to the other party.  3 talks to
> >> both
> >> parties.
> >>
> >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop
> >>
> >> Adding more api calls to control eavesdrop via event socket should be
> >> possible, though I haven't looked at the code to see what it would
> >> take...
> >>
> >> On Thu, Oct 15, 2009 at 3:16 AM, Nikita Belov <nbelov at abisoft.spb.ru>
> >> wrote:
> >>> Thanks for answer.
> >>>
> >>>
> >>>
> >>> As I've understood eavesdrop allows C to hear A and B, but A and B
> >>> will
> >> not
> >>> hear C. But in my case B SHOULD hear C. In my case A - client, B -
> >>> agent
> >>> worked with client and C - coach/supervisor for B. Can I make such
> >>> conference in FS using eavesdrop or other API command?
> >>>
> >>>
> >>>
> >>> ___
> >>>
> >>>
> >>>
> >>> Thanks, Nikita
> >>>
> >>>
> >>>
> >>> ________________________________
> >>>
> >>> From: freeswitch-users-bounces at lists.freeswitch.org
> >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf
> >>> Of Joao
> >>> Mesquita
> >>> Sent: Wednesday, October 14, 2009 7:38 PM
> >>> To: freeswitch-users at lists.freeswitch.org
> >>> Subject: Re: [Freeswitch-users] conference call
> >>>
> >>>
> >>>
> >>> Look at eavesdrop on the wiki.
> >>>
> >>> JM
> >>>
> >>> 2009/10/14 Nikita Belov <nbelov at abisoft.spb.ru>
> >>>
> >>> HI all,
> >>>
> >>>
> >>>
> >>> I want to configure FS to make special conference call between three
> >> users
> >>> (A, B, C).  In this conference C will hear A and B, but A will
> >>> hear only
> >> B.
> >>> Can I make it using FS API commands? Does anybody know which
> >>> approach is
> >>> better to use?
> >>>
> >>>
> >>>
> >>> ___
> >>>
> >>>
> >>>
> >>> Thanks, Nikita
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >>> users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >>> users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>
> >>
> >>
> >> --
> >> -Rupa
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Mon Oct 19 08:13:04 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 19 Oct 2009 10:13:04 -0500
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <7b8592a0910190516rafa4398q62c7a99ef703540f@mail.gmail.com>
References: <7b8592a0910190516rafa4398q62c7a99ef703540f@mail.gmail.com>
Message-ID: <5B924C57-B658-44C0-A1CE-00A3ECB5AA07@freeswitch.org>

I think you just have it misconfigured because if ipv6 was broken I  
would have William and Jason jumping on me.

Can you post your xml configs for the profile please?

/b

On Oct 19, 2009, at 7:16 AM, ineya ineya wrote:

> Hi,
>
> I installed the phonenix release, and changed IPv4, IPv6 addresses  
> and SIP port. Directory and dialplan should be default ones.
> My phones are able to register when using just IPv4, and I can call  
> from one to another.
>
> When I try the same thing with IPv6, the call to other phone fails  
> with [USER_NOT_REGISTERED], however I'm able to call 9999 and hear  
> MOH.
> Here you can find the log: http://pastebin.freeswitch.org/10757
> Freeswitch is running on 10.80.62.40/2000:2::1
> phones are 2000:2::1001 and 2000:2::1002
>
> Am I doing something wrong? Maybe I forgot to configure something,  
> but I can't see what it is.
>
> Any hints are appreciated.

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From helmut.kuper at ewetel.de  Mon Oct 19 08:29:04 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Mon, 19 Oct 2009 17:29:04 +0200
Subject: [Freeswitch-users] Qustion about INFO messages
	after	Connect/Answer
In-Reply-To: <191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>
References: <4AD859A4.5000101@ewetel.de>	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>	<4ADC7C09.6070503@ewetel.de>
	<191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>
Message-ID: <4ADC85C0.7060908@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

well, when I add "answer" before I bridge, there is a small change:
There is no INFO message with "unknown" send to callee.

The caller's display isn't affected by "sip_callee_*" chvars.



On 19.10.2009 17:00, Anthony Minessale wrote:
> you only need to "set" it on the inbound leg and you must answer and
> bridge it somewhere.

My current dialplan is this:

    <extension name="Local_Extension">
      <condition field="${ET_is_local}" expression="^true$">
        <action application="set"
data="dialed_extension=${destination_number}"/>
        <action application="answer"/>
        <action application="set" data="sip_callee_id_number=1111"/>
        <action application="set" data="sip_callee_id_name=hubu"/>
        <action application="export" data="sip_callee_id_number=1111"/>
        <action application="export" data="sip_callee_id_name=hubu"/>
        <action application="info"/>
        <action application="export"
data="dialed_extension=${destination_number}"/>
        <action application="set" data="transfer_ringback=$${hold_music}"/>

        <action application="set" data="hangup_after_bridge=true"/>
        <action application="export"
data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name}
var sip_secure_media)}"/>
        <action application="db"
data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
        <action application="set" data="continue_on_fail=false"/>
        <action application="bridge"
data="user/${dialed_extension}@$${domain}"/>
      </condition>
    </extension>



Here is the complete debug log:



2009-10-19 17:27:20.559060 [DEBUG] sofia.c:4906 IP 85.16.245.206
Approved by acl "clients[]". Access Granted.
2009-10-19 17:27:20.560087 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/1001 at 85.16.246.12:5061 [e47af3a6-bcc3-11de-9f91-c9cd82739033]
2009-10-19 17:27:20.560087 [DEBUG] sofia.c:3492 Channel
sofia/internal/1001 at 85.16.246.12:5061 entering state [received][100]
2009-10-19 17:27:20.560087 [DEBUG] sofia.c:3499 Remote SDP:
v=0
o=root 903333206 903333206 IN IP4 85.16.245.206
s=call
c=IN IP4 85.16.245.206
t=0 0
m=audio 54598 RTP/SAVP 0 8 9 99 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:zS0VqvwW7z/dnmeWvtZS0AwhJ1ru1J1tuHJO2JRD
a=ptime:20
m=audio 54598 RTP/AVP 0 8 9 99 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

2009-10-19 17:27:20.560087 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_NEW
2009-10-19 17:27:20.560087 [DEBUG] switch_core_state_machine.c:312
(sofia/internal/1001 at 85.16.246.12:5061) State NEW
2009-10-19 17:27:20.560087 [DEBUG] sofia.c:3620
(sofia/internal/1001 at 85.16.246.12:5061) State Change CS_NEW -> CS_INIT
2009-10-19 17:27:20.560087 [DEBUG] switch_core_session.c:969 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-19 17:27:20.560087 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_INIT
2009-10-19 17:27:20.560087 [DEBUG] switch_core_state_machine.c:330
(sofia/internal/1001 at 85.16.246.12:5061) State INIT
2009-10-19 17:27:20.560087 [DEBUG] mod_sofia.c:83
sofia/internal/1001 at 85.16.246.12:5061 SOFIA INIT
2009-10-19 17:27:20.560087 [DEBUG] mod_sofia.c:111
(sofia/internal/1001 at 85.16.246.12:5061) State Change CS_INIT -> CS_ROUTING
2009-10-19 17:27:20.560087 [DEBUG] switch_core_session.c:969 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-19 17:27:20.560087 [DEBUG] switch_core_state_machine.c:330
(sofia/internal/1001 at 85.16.246.12:5061) State INIT going to sleep
2009-10-19 17:27:20.560087 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_ROUTING
2009-10-19 17:27:20.560087 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/1001 at 85.16.246.12:5061) State ROUTING
2009-10-19 17:27:20.560087 [DEBUG] mod_sofia.c:130
sofia/internal/1001 at 85.16.246.12:5061 SOFIA ROUTING
2009-10-19 17:27:20.560087 [DEBUG] switch_core_state_machine.c:78
sofia/internal/1001 at 85.16.246.12:5061 Standard ROUTING
2009-10-19 17:27:20.560087 [INFO] mod_dialplan_xml.c:391 Processing 1001
an PBX1->1000 in context default
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->anonymous] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [anonymous]
destination_number(1000) =~ /^\*31([0-9]+)$/ break=never
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 ANTI-Action
set(dialed_extension=${destination_number})
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->is_local] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [is_local]
${ET_is_local}() =~ /(true|false)/ break=on-true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 ANTI-Action
lua(ET_is_local.lua)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 ANTI-Action
transfer(${destination_number} XML default)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->set_domain] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (PASS)
[set_domain] destination_number(1000) =~ /^.*$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
set(domain_name=85.16.246.12)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->302_zero_problem] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[302_zero_problem] ${sip_looped_call}() =~ /true/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->fakeTrisko-Servicenumber as callerid] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[fakeTrisko-Servicenumber as callerid] caller_id_number(1001) =~
/^(49[1-2][0-9])$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->outbound-ssw-ddi] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[outbound-ssw-ddi] ${dialed_extension}() =~ /^0([0-9]+)$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing [default->mail]
continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [mail]
destination_number(1000) =~ /^9997$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing [default->pizza]
continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [pizza]
destination_number(1000) =~ /^2006$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing [default->pbook]
continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [pbook]
destination_number(1000) =~ /^2007$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->shout_1] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [shout_1]
${dialed_extension}() =~ /^9123$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->shout_2] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [shout_2]
${dialed_extension}() =~ /^9124$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->speex_32000] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[speex_32000] ${dialed_extension}() =~ /^9125$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->speex_32000_2] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[speex_32000_2] ${dialed_extension}() =~ /^9126$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->jukebox] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [jukebox]
destination_number(1000) =~ /^9111$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->pickup-extension] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[pickup-extension] ${dialed_extension}() =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->Voicemail-Verwaltung] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[Voicemail-Verwaltung] ${dialed_extension}() =~ /^(1001|9999)$/
break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->Local_Extension] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[Local_Extension] ${ET_is_local}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->parallel call] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [parallel
call] ${dialed_extension}() =~ /^(6000)$/ break=on-false
2009-10-19 17:27:20.562064 [DEBUG] switch_core_state_machine.c:114
(sofia/internal/1001 at 85.16.246.12:5061) State Change CS_ROUTING ->
CS_EXECUTE
2009-10-19 17:27:20.562064 [DEBUG] switch_core_session.c:969 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-19 17:27:20.562064 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/1001 at 85.16.246.12:5061) State ROUTING going to sleep
2009-10-19 17:27:20.563061 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_EXECUTE
2009-10-19 17:27:20.563061 [DEBUG] switch_core_state_machine.c:340
(sofia/internal/1001 at 85.16.246.12:5061) State EXECUTE
2009-10-19 17:27:20.563061 [DEBUG] mod_sofia.c:173
sofia/internal/1001 at 85.16.246.12:5061 SOFIA EXECUTE
2009-10-19 17:27:20.563061 [DEBUG] switch_core_state_machine.c:151
sofia/internal/1001 at 85.16.246.12:5061 Standard EXECUTE
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 set(dialed_extension=1000)
2009-10-19 17:27:20.563061 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET [dialed_extension]=[1000]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 lua(ET_is_local.lua)
2009-10-19 17:27:20.588079 [DEBUG] switch_cpp.cpp:925 destroy/unlink
session from object
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 transfer(1000 XML default)
2009-10-19 17:27:20.588079 [DEBUG] switch_ivr.c:1381
(sofia/internal/1001 at 85.16.246.12:5061) State Change CS_EXECUTE ->
CS_ROUTING
2009-10-19 17:27:20.588079 [DEBUG] switch_core_session.c:969 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-19 17:27:20.588079 [DEBUG] switch_ivr.c:1385
sofia/internal/1001 at 85.16.246.12:5061 receive message [TRANSFER]
2009-10-19 17:27:20.588079 [DEBUG] switch_core_session.c:639 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-19 17:27:20.588079 [NOTICE] switch_ivr.c:1387 Transfer
sofia/internal/1001 at 85.16.246.12:5061 to XML[1000 at default]
2009-10-19 17:27:20.588079 [DEBUG] switch_core_state_machine.c:340
(sofia/internal/1001 at 85.16.246.12:5061) State EXECUTE going to sleep
2009-10-19 17:27:20.588079 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_ROUTING
2009-10-19 17:27:20.588079 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/1001 at 85.16.246.12:5061) State ROUTING
2009-10-19 17:27:20.588079 [DEBUG] mod_sofia.c:130
sofia/internal/1001 at 85.16.246.12:5061 SOFIA ROUTING
2009-10-19 17:27:20.588079 [DEBUG] switch_core_state_machine.c:78
sofia/internal/1001 at 85.16.246.12:5061 Standard ROUTING
2009-10-19 17:27:20.588079 [INFO] mod_dialplan_xml.c:391 Processing 1001
an PBX1->1000 in context default
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->anonymous] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [anonymous]
destination_number(1000) =~ /^\*31([0-9]+)$/ break=never
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 ANTI-Action
set(dialed_extension=${destination_number})
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->is_local] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (PASS) [is_local]
${ET_is_local}(true) =~ /(true|false)/ break=on-true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->set_domain] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (PASS)
[set_domain] destination_number(1000) =~ /^.*$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
set(domain_name=85.16.246.12)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->302_zero_problem] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[302_zero_problem] ${sip_looped_call}() =~ /true/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->fakeTrisko-Servicenumber as callerid] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[fakeTrisko-Servicenumber as callerid] caller_id_number(1001) =~
/^(49[1-2][0-9])$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->outbound-ssw-ddi] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[outbound-ssw-ddi] ${dialed_extension}(1000) =~ /^0([0-9]+)$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing [default->mail]
continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [mail]
destination_number(1000) =~ /^9997$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing [default->pizza]
continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [pizza]
destination_number(1000) =~ /^2006$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing [default->pbook]
continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [pbook]
destination_number(1000) =~ /^2007$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->shout_1] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [shout_1]
${dialed_extension}(1000) =~ /^9123$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->shout_2] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [shout_2]
${dialed_extension}(1000) =~ /^9124$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->speex_32000] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[speex_32000] ${dialed_extension}(1000) =~ /^9125$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->speex_32000_2] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[speex_32000_2] ${dialed_extension}(1000) =~ /^9126$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->jukebox] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [jukebox]
destination_number(1000) =~ /^9111$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->pickup-extension] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[pickup-extension] ${dialed_extension}(1000) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->Voicemail-Verwaltung] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[Voicemail-Verwaltung] ${dialed_extension}(1000) =~ /^(1001|9999)$/
break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->Local_Extension] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (PASS)
[Local_Extension] ${ET_is_local}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
set(dialed_extension=${destination_number})
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action answer()
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
set(sip_callee_id_number=1111)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
set(sip_callee_id_name=hubu)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
export(sip_callee_id_number=1111)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
export(sip_callee_id_name=hubu)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action info()
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
export(dialed_extension=${destination_number})
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
set(transfer_ringback=local_stream://moh)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
set(hangup_after_bridge=true)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
export(nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name}
var sip_secure_media)})
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
db(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid})
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
set(continue_on_fail=false)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
bridge(user/${dialed_extension}@85.16.246.12)
2009-10-19 17:27:20.590066 [DEBUG] switch_core_state_machine.c:114
(sofia/internal/1001 at 85.16.246.12:5061) State Change CS_ROUTING ->
CS_EXECUTE
2009-10-19 17:27:20.590066 [DEBUG] switch_core_session.c:969 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-19 17:27:20.590066 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/1001 at 85.16.246.12:5061) State ROUTING going to sleep
2009-10-19 17:27:20.590066 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_EXECUTE
2009-10-19 17:27:20.590066 [DEBUG] switch_core_state_machine.c:340
(sofia/internal/1001 at 85.16.246.12:5061) State EXECUTE
2009-10-19 17:27:20.590066 [DEBUG] mod_sofia.c:173
sofia/internal/1001 at 85.16.246.12:5061 SOFIA EXECUTE
2009-10-19 17:27:20.591072 [DEBUG] switch_core_state_machine.c:151
sofia/internal/1001 at 85.16.246.12:5061 Standard EXECUTE
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 set(dialed_extension=1000)
2009-10-19 17:27:20.591072 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET [dialed_extension]=[1000]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 set(domain_name=85.16.246.12)
2009-10-19 17:27:20.591072 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET [domain_name]=[85.16.246.12]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 set(dialed_extension=1000)
2009-10-19 17:27:20.591072 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET [dialed_extension]=[1000]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 answer()
2009-10-19 17:27:20.591072 [DEBUG] mod_dptools.c:653
sofia/internal/1001 at 85.16.246.12:5061 receive message [ANSWER]
2009-10-19 17:27:20.591072 [DEBUG] sofia_glue.c:3028 Set Remote Key [1
AES_CM_128_HMAC_SHA1_32 inline:zS0VqvwW7z/dnmeWvtZS0AwhJ1ru1J1tuHJO2JRD]
2009-10-19 17:27:20.591072 [DEBUG] sofia_glue.c:2160 Set Local Key [1
AES_CM_128_HMAC_SHA1_32 inline:GteACgi6L6LMWBxdEnW4/XSOUhsmp/iKk9dZjmBJ]
2009-10-19 17:27:20.591072 [DEBUG] sofia_glue.c:3142 Audio Codec Compare
[pcmu:0:8000:20]/[CELT:95:48000:10]
2009-10-19 17:27:20.591072 [DEBUG] sofia_glue.c:3142 Audio Codec Compare
[pcmu:0:8000:20]/[SPEEX:103:32000:20]
2009-10-19 17:27:20.591072 [DEBUG] sofia_glue.c:3142 Audio Codec Compare
[pcmu:0:8000:20]/[G722:9:8000:20]
2009-10-19 17:27:20.591072 [DEBUG] sofia_glue.c:3142 Audio Codec Compare
[pcmu:0:8000:20]/[PCMA:8:8000:20]
2009-10-19 17:27:20.591072 [DEBUG] sofia_glue.c:3142 Audio Codec Compare
[pcma:8:8000:20]/[CELT:95:48000:10]
2009-10-19 17:27:20.591072 [DEBUG] sofia_glue.c:3142 Audio Codec Compare
[pcma:8:8000:20]/[SPEEX:103:32000:20]
2009-10-19 17:27:20.591072 [DEBUG] sofia_glue.c:3142 Audio Codec Compare
[pcma:8:8000:20]/[G722:9:8000:20]
2009-10-19 17:27:20.591072 [DEBUG] sofia_glue.c:3142 Audio Codec Compare
[pcma:8:8000:20]/[PCMA:8:8000:20]
2009-10-19 17:27:20.591072 [DEBUG] sofia_glue.c:2100 Set Codec
sofia/internal/1001 at 85.16.246.12:5061 PCMA/8000 20 ms 160 samples
2009-10-19 17:27:20.591072 [DEBUG] sofia_glue.c:3102 Set 2833 dtmf
payload to 101
2009-10-19 17:27:20.591072 [DEBUG] sofia_glue.c:3076 Our existing sdp is
still good [pcma 85.16.245.206:54598], let's keep it.
2009-10-19 17:27:20.591072 [DEBUG] sofia_glue.c:2334 AUDIO RTP
[sofia/internal/1001 at 85.16.246.12:5061] 85.16.246.12 port 14580 ->
85.16.245.206 port 54598 codec: 8 ms: 20
2009-10-19 17:27:20.591072 [DEBUG] switch_rtp.c:1155 Starting timer
[soft] 160 bytes per 20ms
2009-10-19 17:27:20.594059 [INFO] switch_rtp.c:1019 Activating Secure
RTP SEND
2009-10-19 17:27:20.594059 [INFO] switch_rtp.c:999 Activating Secure RTP
RECV
2009-10-19 17:27:20.594059 [DEBUG] switch_core_sqldb.c:396 Secure Type:
srtp:AES_CM_128_HMAC_SHA1_32
2009-10-19 17:27:20.594059 [DEBUG] switch_core_sqldb.c:396 Secure Type:
srtp:AES_CM_128_HMAC_SHA1_32
2009-10-19 17:27:20.594059 [NOTICE] sofia_glue.c:2769 Pre-Answer
sofia/internal/1001 at 85.16.246.12:5061!
2009-10-19 17:27:20.594059 [DEBUG] mod_sofia.c:543 Local SDP
sofia/internal/1001 at 85.16.246.12:5061:
v=0
o=FreeSWITCH 1255951460 1255951461 IN IP4 85.16.246.12
s=FreeSWITCH
c=IN IP4 85.16.246.12
t=0 0
m=audio 14580 RTP/SAVP 8 101
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:GteACgi6L6LMWBxdEnW4/XSOUhsmp/iKk9dZjmBJ

2009-10-19 17:27:20.594059 [DEBUG] switch_core_session.c:639 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-19 17:27:20.594059 [NOTICE] mod_dptools.c:653 Channel
[sofia/internal/1001 at 85.16.246.12:5061] has been answered
2009-10-19 17:27:20.594059 [DEBUG] switch_channel.c:182
sofia/internal/1001 at 85.16.246.12:5061 receive message [AUDIO_SYNC]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 set(sip_callee_id_number=1111)
2009-10-19 17:27:20.594059 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET [sip_callee_id_number]=[1111]
2009-10-19 17:27:20.595060 [DEBUG] sofia.c:3492 Channel
sofia/internal/1001 at 85.16.246.12:5061 entering state [completed][200]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 set(sip_callee_id_name=hubu)
2009-10-19 17:27:20.595060 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET [sip_callee_id_name]=[hubu]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061
export(sip_callee_id_number=1111)
2009-10-19 17:27:20.595060 [DEBUG] mod_dptools.c:896 EXPORT
[sip_callee_id_number]=[1111]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061
export(sip_callee_id_name=hubu)
2009-10-19 17:27:20.595060 [DEBUG] mod_dptools.c:896 EXPORT
[sip_callee_id_name]=[hubu]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 info()
2009-10-19 17:27:20.596059 [INFO] mod_dptools.c:961 CHANNEL_DATA:
Channel-State: [CS_EXECUTE]
Channel-State-Number: [4]
Channel-Name: [sofia/internal/1001 at 85.16.246.12:5061]
Unique-ID: [e47af3a6-bcc3-11de-9f91-c9cd82739033]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Answer-State: [answered]
Channel-Read-Codec-Name: [PCMA]
Channel-Read-Codec-Rate: [8000]
Channel-Write-Codec-Name: [PCMA]
Channel-Write-Codec-Rate: [8000]
Caller-Username: [1001]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [1001 an PBX1]
Caller-Caller-ID-Number: [1001]
Caller-Network-Addr: [85.16.245.206]
Caller-Destination-Number: [1000]
Caller-Unique-ID: [e47af3a6-bcc3-11de-9f91-c9cd82739033]
Caller-Source: [mod_sofia]
Caller-Context: [default]
Caller-RDNIS: [1000]
Caller-Channel-Name: [sofia/internal/1001 at 85.16.246.12:5061]
Caller-Profile-Index: [2]
Caller-Profile-Created-Time: [1255966040588079]
Caller-Channel-Created-Time: [1255966040560087]
Caller-Channel-Answered-Time: [1255966040594059]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [1255966040594059]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
variable_sip_received_ip: [85.16.245.206]
variable_sip_received_port: [1024]
variable_sip_via_protocol: [udp]
variable_sip_authorized: [true]
variable_sip_from_user: [1001]
variable_sip_from_port: [5061]
variable_sip_from_uri: [1001 at 85.16.246.12:5061]
variable_sip_from_host: [85.16.246.12]
variable_sip_from_user_stripped: [1001]
variable_sip_from_tag: [02p9o5jbg8]
variable_sofia_profile_name: [internal]
variable_sip_req_params: [user=phone]
variable_sip_req_user: [1000]
variable_sip_req_port: [5061]
variable_sip_req_uri: [1000 at 85.16.246.12:5061]
variable_sip_req_host: [85.16.246.12]
variable_sip_to_params: [user=phone]
variable_sip_to_user: [1000]
variable_sip_to_port: [5061]
variable_sip_to_uri: [1000 at 85.16.246.12:5061]
variable_sip_to_host: [85.16.246.12]
variable_sip_contact_params: [line=eg3wp69a]
variable_sip_contact_user: [1001]
variable_sip_contact_port: [1024]
variable_sip_contact_uri: [1001 at 85.16.245.206:1024]
variable_sip_contact_host: [85.16.245.206]
variable_channel_name: [sofia/internal/1001 at 85.16.246.12:5061]
variable_sip_call_id: [3c2d38a1cd82-iphmvbhl46hj]
variable_sip_user_agent: [snom820/8.2.16]
variable_sip_via_host: [85.16.245.206]
variable_sip_via_port: [1024]
variable_sip_via_rport: [1024]
variable_presence_id: [1001 at 85.16.246.12]
variable_sip_h_X-Serialnumber: [0004134002CB]
variable_sip_h_P-Key-Flags: [resolution="31x13", keys="4"]
variable_switch_r_sdp: [v=0
o=root 903333206 903333206 IN IP4 85.16.245.206
s=call
c=IN IP4 85.16.245.206
t=0 0
m=audio 54598 RTP/SAVP 0 8 9 99 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:zS0VqvwW7z/dnmeWvtZS0AwhJ1ru1J1tuHJO2JRD
a=ptime:20
m=audio 54598 RTP/AVP 0 8 9 99 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
]
variable_ep_codec_string: [G722 at 8000h@20i,PCMA at 8000h@20i]
variable_ET_is_local: [true]
variable_max_forwards: [69]
variable_domain_name: [85.16.246.12]
variable_dialed_extension: [1000]
variable_sip_has_crypto: [AES_CM_128_HMAC_SHA1_32]
variable_sip_crypto_mandatory: [true]
variable_sip_secure_media: [true]
variable_remote_media_ip: [85.16.245.206]
variable_remote_media_port: [54598]
variable_read_codec: [PCMA]
variable_read_rate: [8000]
variable_write_codec: [PCMA]
variable_write_rate: [8000]
variable_local_media_ip: [85.16.246.12]
variable_local_media_port: [14580]
variable_sip_secure_media_confirmed: [true]
variable_endpoint_disposition: [ANSWER]
variable_sip_callee_id_number: [1111]
variable_sip_callee_id_name: [hubu]
variable_export_vars: [sip_callee_id_number,sip_callee_id_name]
variable_current_application: [info]


EXECUTE sofia/internal/1001 at 85.16.246.12:5061 export(dialed_extension=1000)
2009-10-19 17:27:20.596059 [DEBUG] mod_dptools.c:896 EXPORT
[dialed_extension]=[1000]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061
set(transfer_ringback=local_stream://moh)
2009-10-19 17:27:20.596059 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET
[transfer_ringback]=[local_stream://moh]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 set(hangup_after_bridge=true)
2009-10-19 17:27:20.597073 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061
export(nolocal:sip_secure_media=)
2009-10-19 17:27:20.621072 [DEBUG] mod_dptools.c:896 EXPORT (REMOTE
ONLY) [sip_secure_media]=[UNDEF]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061
db(insert/85.16.246.12-last_dial//e47af3a6-bcc3-11de-9f91-c9cd82739033)
2009-10-19 17:27:20.832283 [DEBUG] sofia.c:3492 Channel
sofia/internal/1001 at 85.16.246.12:5061 entering state [ready][200]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 set(continue_on_fail=false)
2009-10-19 17:27:20.841267 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET [continue_on_fail]=[false]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 bridge(user/1000 at 85.16.246.12)
2009-10-19 17:27:20.865064 [DEBUG] switch_ivr_originate.c:1162 variable
string 0 = [presence_id=1000 at 85.16.246.12]
2009-10-19 17:27:20.865064 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/sip:1000 at 85.16.245.228:1024
[e4a9a7a0-bcc3-11de-9f91-c9cd82739033]
2009-10-19 17:27:20.866297 [DEBUG] mod_sofia.c:2981
(sofia/internal/sip:1000 at 85.16.245.228:1024) State Change CS_NEW -> CS_INIT
2009-10-19 17:27:20.866297 [DEBUG] switch_core_session.c:969 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-19 17:27:20.866297 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/sip:1000 at 85.16.245.228:1024) Running State Change CS_INIT
2009-10-19 17:27:20.866297 [DEBUG] switch_core_state_machine.c:330
(sofia/internal/sip:1000 at 85.16.245.228:1024) State INIT
2009-10-19 17:27:20.866297 [DEBUG] mod_sofia.c:83
sofia/internal/sip:1000 at 85.16.245.228:1024 SOFIA INIT
2009-10-19 17:27:20.866297 [DEBUG] mod_sofia.c:111
(sofia/internal/sip:1000 at 85.16.245.228:1024) State Change CS_INIT ->
CS_ROUTING
2009-10-19 17:27:20.866297 [DEBUG] switch_core_session.c:969 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-19 17:27:20.866297 [DEBUG] switch_core_state_machine.c:330
(sofia/internal/sip:1000 at 85.16.245.228:1024) State INIT going to sleep
2009-10-19 17:27:20.866297 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/sip:1000 at 85.16.245.228:1024) Running State Change CS_ROUTING
2009-10-19 17:27:20.866297 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/sip:1000 at 85.16.245.228:1024) State ROUTING
2009-10-19 17:27:20.866297 [DEBUG] mod_sofia.c:130
sofia/internal/sip:1000 at 85.16.245.228:1024 SOFIA ROUTING
2009-10-19 17:27:20.866297 [DEBUG] switch_ivr_originate.c:66
(sofia/internal/sip:1000 at 85.16.245.228:1024) State Change CS_ROUTING ->
CS_CONSUME_MEDIA
2009-10-19 17:27:20.866297 [DEBUG] switch_core_session.c:969 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-19 17:27:20.866297 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/sip:1000 at 85.16.245.228:1024) State ROUTING going to sleep
2009-10-19 17:27:20.866297 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/sip:1000 at 85.16.245.228:1024) Running State Change
CS_CONSUME_MEDIA
2009-10-19 17:27:20.866297 [DEBUG] switch_ivr_originate.c:1880 Raw Codec
Activation Success L16 at 8000hz 1 channel 20ms
2009-10-19 17:27:20.866297 [DEBUG] switch_ivr_originate.c:1912 Play
Ringback File [local_stream://moh]
2009-10-19 17:27:20.866297 [DEBUG] mod_local_stream.c:408 Opening Stream
[moh/8000] 8000hz
2009-10-19 17:27:20.866297 [DEBUG] switch_core_state_machine.c:352
(sofia/internal/sip:1000 at 85.16.245.228:1024) State CONSUME_MEDIA
2009-10-19 17:27:20.866297 [DEBUG] switch_core_state_machine.c:352
(sofia/internal/sip:1000 at 85.16.245.228:1024) State CONSUME_MEDIA going
to sleep
2009-10-19 17:27:20.868064 [DEBUG] sofia.c:3492 Channel
sofia/internal/sip:1000 at 85.16.245.228:1024 entering state [calling][0]
2009-10-19 17:27:20.877064 [DEBUG] switch_core_io.c:652
sofia/internal/1001 at 85.16.246.12:5061 receive message
[TRANSCODING_NECESSARY]
2009-10-19 17:27:20.887288 [DEBUG] sofia.c:3492 Channel
sofia/internal/sip:1000 at 85.16.245.228:1024 entering state [proceeding][180]
2009-10-19 17:27:20.887288 [NOTICE] sofia.c:3556 Ring-Ready
sofia/internal/sip:1000 at 85.16.245.228:1024!
2009-10-19 17:27:21.392073 [DEBUG] sofia.c:3492 Channel
sofia/internal/sip:1000 at 85.16.245.228:1024 entering state [proceeding][180]
2009-10-19 17:27:22.403086 [DEBUG] sofia.c:3492 Channel
sofia/internal/sip:1000 at 85.16.245.228:1024 entering state [proceeding][180]
2009-10-19 17:27:22.849092 [DEBUG] sofia.c:3492 Channel
sofia/internal/sip:1000 at 85.16.245.228:1024 entering state [ready][200]
2009-10-19 17:27:22.849092 [DEBUG] sofia.c:3499 Remote SDP:
v=0
o=root 1600007648 1600007649 IN IP4 85.16.245.228
s=call
c=IN IP4 85.16.245.228
t=0 0
m=audio 50468 RTP/AVP 8 9 101
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

2009-10-19 17:27:22.849092 [DEBUG] sofia_glue.c:3142 Audio Codec Compare
[pcma:8:8000:20]/[PCMA:8:8000:20]
2009-10-19 17:27:22.849092 [DEBUG] sofia_glue.c:2100 Set Codec
sofia/internal/sip:1000 at 85.16.245.228:1024 PCMA/8000 20 ms 160 samples
2009-10-19 17:27:22.849092 [DEBUG] sofia_glue.c:3102 Set 2833 dtmf
payload to 101
2009-10-19 17:27:22.849092 [DEBUG] sofia_glue.c:2334 AUDIO RTP
[sofia/internal/sip:1000 at 85.16.245.228:1024] 85.16.246.12 port 16276 ->
85.16.245.228 port 50468 codec: 8 ms: 20
2009-10-19 17:27:22.850096 [DEBUG] switch_rtp.c:1155 Starting timer
[soft] 160 bytes per 20ms
2009-10-19 17:27:22.851093 [DEBUG] switch_channel.c:2046 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-19 17:27:22.851093 [NOTICE] sofia.c:3982 Channel
[sofia/internal/sip:1000 at 85.16.245.228:1024] has been answered
2009-10-19 17:27:22.851093 [DEBUG] switch_channel.c:182
sofia/internal/sip:1000 at 85.16.245.228:1024 receive message [AUDIO_SYNC]
2009-10-19 17:27:22.857294 [DEBUG] switch_core_codec.c:122 Restore
original codec.
2009-10-19 17:27:22.857294 [DEBUG] switch_ivr_originate.c:2363 Originate
Resulted in Success: [sofia/internal/sip:1000 at 85.16.245.228:1024]
2009-10-19 17:27:22.857294 [DEBUG] switch_channel.c:182
sofia/internal/sip:1000 at 85.16.245.228:1024 receive message [AUDIO_SYNC]
2009-10-19 17:27:22.857294 [DEBUG] switch_channel.c:182
sofia/internal/1001 at 85.16.246.12:5061 receive message [AUDIO_SYNC]
2009-10-19 17:27:22.857294 [DEBUG] switch_ivr_originate.c:1880 Raw Codec
Activation Success L16 at 8000hz 1 channel 20ms
2009-10-19 17:27:22.858097 [DEBUG] switch_ivr_originate.c:1912 Play
Ringback File [local_stream://moh]
2009-10-19 17:27:22.858097 [DEBUG] mod_local_stream.c:408 Opening Stream
[moh/8000] 8000hz
2009-10-19 17:27:22.858097 [DEBUG] switch_core_codec.c:122 Restore
original codec.
2009-10-19 17:27:22.858097 [DEBUG] switch_ivr_originate.c:2363 Originate
Resulted in Success: [sofia/internal/sip:1000 at 85.16.245.228:1024]
2009-10-19 17:27:22.858097 [DEBUG] switch_channel.c:182
sofia/internal/sip:1000 at 85.16.245.228:1024 receive message [AUDIO_SYNC]
2009-10-19 17:27:22.858097 [DEBUG] switch_channel.c:182
sofia/internal/1001 at 85.16.246.12:5061 receive message [AUDIO_SYNC]
2009-10-19 17:27:22.858097 [DEBUG] switch_ivr_bridge.c:966
sofia/internal/sip:1000 at 85.16.245.228:1024 receive message [BRIDGE]
2009-10-19 17:27:22.858097 [DEBUG] switch_core_session.c:639 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-19 17:27:22.858097 [DEBUG] switch_ivr_bridge.c:973
sofia/internal/1001 at 85.16.246.12:5061 receive message [BRIDGE]
2009-10-19 17:27:22.858097 [DEBUG] switch_core_session.c:639 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-19 17:27:22.858097 [DEBUG] switch_ivr_bridge.c:1017
(sofia/internal/sip:1000 at 85.16.245.228:1024) State Change
CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2009-10-19 17:27:22.858097 [DEBUG] switch_core_session.c:969 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-19 17:27:22.858097 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/sip:1000 at 85.16.245.228:1024) Running State Change
CS_EXCHANGE_MEDIA
2009-10-19 17:27:22.858097 [DEBUG] switch_core_state_machine.c:343
(sofia/internal/sip:1000 at 85.16.245.228:1024) State EXCHANGE_MEDIA
2009-10-19 17:27:22.858097 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK
2009-10-19 17:27:22.877303 [DEBUG] switch_ivr_bridge.c:115
sofia/internal/sip:1000 at 85.16.245.228:1024 receive message [DISPLAY]
2009-10-19 17:27:22.877303 [DEBUG] switch_ivr_bridge.c:115
sofia/internal/1001 at 85.16.246.12:5061 receive message [DISPLAY]
2009-10-19 17:27:24.568124 [NOTICE] sofia.c:328 Hangup
sofia/internal/sip:1000 at 85.16.245.228:1024 [CS_EXCHANGE_MEDIA]
[NORMAL_CLEARING]
2009-10-19 17:27:24.568124 [DEBUG] switch_channel.c:1837 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [KILL]
2009-10-19 17:27:24.568124 [DEBUG] switch_core_session.c:969 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-19 17:27:24.568124 [DEBUG] switch_core_state_machine.c:449
thread mismatch skipping state handler.
2009-10-19 17:27:24.577124 [DEBUG] switch_ivr_bridge.c:446
sofia/internal/sip:1000 at 85.16.245.228:1024 ending bridge by request from
read function
2009-10-19 17:27:24.577124 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
DONE [sofia/internal/sip:1000 at 85.16.245.228:1024]
2009-10-19 17:27:24.577124 [DEBUG] switch_ivr_bridge.c:523 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-19 17:27:24.577124 [DEBUG] switch_core_state_machine.c:343
(sofia/internal/sip:1000 at 85.16.245.228:1024) State EXCHANGE_MEDIA going
to sleep
2009-10-19 17:27:24.577124 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/sip:1000 at 85.16.245.228:1024) Running State Change CS_HANGUP
2009-10-19 17:27:24.577124 [DEBUG] switch_ivr_bridge.c:440
sofia/internal/sip:1000 at 85.16.245.228:1024 ending bridge by request from
write function
2009-10-19 17:27:24.577124 [DEBUG] switch_ivr_bridge.c:495
sofia/internal/1001 at 85.16.246.12:5061 receive message [UNBRIDGE]
2009-10-19 17:27:24.577124 [DEBUG] switch_core_session.c:639 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-19 17:27:24.577124 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
DONE [sofia/internal/1001 at 85.16.246.12:5061]
2009-10-19 17:27:24.577124 [DEBUG] switch_ivr_bridge.c:523 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-19 17:27:24.577124 [NOTICE] switch_ivr_bridge.c:1093 Hangup
sofia/internal/1001 at 85.16.246.12:5061 [CS_EXECUTE] [NORMAL_CLEARING]
2009-10-19 17:27:24.577124 [DEBUG] switch_channel.c:1837 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [KILL]
2009-10-19 17:27:24.577124 [DEBUG] switch_core_session.c:969 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-19 17:27:24.577124 [DEBUG] switch_core_state_machine.c:476
(sofia/internal/sip:1000 at 85.16.245.228:1024) State HANGUP
2009-10-19 17:27:24.577124 [DEBUG] mod_sofia.c:344 Channel
sofia/internal/sip:1000 at 85.16.245.228:1024 hanging up, cause:
NORMAL_CLEARING
2009-10-19 17:27:24.577124 [DEBUG] switch_core_state_machine.c:46
sofia/internal/sip:1000 at 85.16.245.228:1024 Standard HANGUP, cause:
NORMAL_CLEARING
2009-10-19 17:27:24.577124 [DEBUG] switch_core_state_machine.c:476
(sofia/internal/sip:1000 at 85.16.245.228:1024) State HANGUP going to sleep
2009-10-19 17:27:24.578354 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/sip:1000 at 85.16.245.228:1024) State Change CS_HANGUP ->
CS_REPORTING
2009-10-19 17:27:24.578354 [DEBUG] switch_core_session.c:969 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-19 17:27:24.578354 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/sip:1000 at 85.16.245.228:1024) Running State Change
CS_REPORTING
2009-10-19 17:27:24.578354 [DEBUG] switch_core_state_machine.c:567
(sofia/internal/sip:1000 at 85.16.245.228:1024) State REPORTING
2009-10-19 17:27:24.578354 [DEBUG] switch_core_state_machine.c:53
sofia/internal/sip:1000 at 85.16.245.228:1024 Standard REPORTING, cause:
NORMAL_CLEARING
2009-10-19 17:27:24.578354 [DEBUG] switch_core_state_machine.c:567
(sofia/internal/sip:1000 at 85.16.245.228:1024) State REPORTING going to sleep
2009-10-19 17:27:24.578354 [DEBUG] switch_core_state_machine.c:319
(sofia/internal/sip:1000 at 85.16.245.228:1024) State Change CS_REPORTING
- -> CS_DESTROY
2009-10-19 17:27:24.578354 [DEBUG] switch_core_session.c:969 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-19 17:27:24.578354 [DEBUG] switch_core_state_machine.c:476
(sofia/internal/1001 at 85.16.246.12:5061) State HANGUP
2009-10-19 17:27:24.578354 [DEBUG] mod_sofia.c:344 Channel
sofia/internal/1001 at 85.16.246.12:5061 hanging up, cause: NORMAL_CLEARING
2009-10-19 17:27:24.578354 [DEBUG] switch_core_session.c:1106 Session 46
(sofia/internal/sip:1000 at 85.16.245.228:1024) Locked, Waiting on external
entities
2009-10-19 17:27:24.578354 [DEBUG] mod_sofia.c:382 Sending BYE to
sofia/internal/1001 at 85.16.246.12:5061
2009-10-19 17:27:24.579127 [DEBUG] switch_core_state_machine.c:46
sofia/internal/1001 at 85.16.246.12:5061 Standard HANGUP, cause:
NORMAL_CLEARING
2009-10-19 17:27:24.579127 [DEBUG] switch_core_state_machine.c:476
(sofia/internal/1001 at 85.16.246.12:5061) State HANGUP going to sleep
2009-10-19 17:27:24.579127 [NOTICE] switch_core_session.c:1124 Session
46 (sofia/internal/sip:1000 at 85.16.245.228:1024) Ended
2009-10-19 17:27:24.579127 [NOTICE] switch_core_session.c:1126 Close
Channel sofia/internal/sip:1000 at 85.16.245.228:1024 [CS_DESTROY]
2009-10-19 17:27:24.579127 [DEBUG] switch_core_state_machine.c:413
(sofia/internal/sip:1000 at 85.16.245.228:1024) Running State Change CS_DESTROY
2009-10-19 17:27:24.579127 [DEBUG] switch_core_state_machine.c:340
(sofia/internal/1001 at 85.16.246.12:5061) State EXECUTE going to sleep
2009-10-19 17:27:24.579127 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_HANGUP
2009-10-19 17:27:24.579127 [DEBUG] switch_core_state_machine.c:424
(sofia/internal/sip:1000 at 85.16.245.228:1024) State DESTROY
2009-10-19 17:27:24.579127 [DEBUG] mod_sofia.c:261
sofia/internal/sip:1000 at 85.16.245.228:1024 SOFIA DESTROY
2009-10-19 17:27:24.579127 [DEBUG] switch_core_state_machine.c:60
sofia/internal/sip:1000 at 85.16.245.228:1024 Standard DESTROY
2009-10-19 17:27:24.579127 [DEBUG] switch_core_state_machine.c:454
handler already called, skipping state handler.
2009-10-19 17:27:24.579127 [DEBUG] switch_core_state_machine.c:424
(sofia/internal/sip:1000 at 85.16.245.228:1024) State DESTROY going to sleep
2009-10-19 17:27:24.579127 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/1001 at 85.16.246.12:5061) State Change CS_HANGUP ->
CS_REPORTING
2009-10-19 17:27:24.579127 [DEBUG] switch_core_session.c:969 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-19 17:27:24.579127 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_REPORTING
2009-10-19 17:27:24.579127 [DEBUG] switch_core_state_machine.c:567
(sofia/internal/1001 at 85.16.246.12:5061) State REPORTING
2009-10-19 17:27:24.581127 [DEBUG] switch_core_state_machine.c:53
sofia/internal/1001 at 85.16.246.12:5061 Standard REPORTING, cause:
NORMAL_CLEARING
2009-10-19 17:27:24.581127 [DEBUG] switch_core_state_machine.c:567
(sofia/internal/1001 at 85.16.246.12:5061) State REPORTING going to sleep
2009-10-19 17:27:24.581127 [DEBUG] switch_core_state_machine.c:319
(sofia/internal/1001 at 85.16.246.12:5061) State Change CS_REPORTING ->
CS_DESTROY
2009-10-19 17:27:24.581127 [DEBUG] switch_core_session.c:969 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-19 17:27:24.581127 [DEBUG] switch_core_session.c:1106 Session 45
(sofia/internal/1001 at 85.16.246.12:5061) Locked, Waiting on external entities
2009-10-19 17:27:24.581127 [NOTICE] switch_core_session.c:1124 Session
45 (sofia/internal/1001 at 85.16.246.12:5061) Ended
2009-10-19 17:27:24.581127 [NOTICE] switch_core_session.c:1126 Close
Channel sofia/internal/1001 at 85.16.246.12:5061 [CS_DESTROY]
2009-10-19 17:27:24.581127 [DEBUG] switch_core_state_machine.c:413
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_DESTROY
2009-10-19 17:27:24.581127 [DEBUG] switch_core_state_machine.c:424
(sofia/internal/1001 at 85.16.246.12:5061) State DESTROY
2009-10-19 17:27:24.581127 [DEBUG] mod_sofia.c:261
sofia/internal/1001 at 85.16.246.12:5061 SOFIA DESTROY
2009-10-19 17:27:24.581127 [DEBUG] switch_core_state_machine.c:60
sofia/internal/1001 at 85.16.246.12:5061 Standard DESTROY
2009-10-19 17:27:24.581127 [DEBUG] switch_core_state_machine.c:424
(sofia/internal/1001 at 85.16.246.12:5061) State DESTROY going to sleep





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From mike at jerris.com  Mon Oct 19 08:33:35 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Oct 2009 11:33:35 -0400
Subject: [Freeswitch-users] uuid_send_dtmf fails (was: conference call)
In-Reply-To: <14E8C50A8F4D41559AFCCD8838350F4F@abisoft.biz>
References: <064A5201C10D4D20AC76EB93CA5EA4B5@abisoft.biz><ebbc2d550910140838v3125579ev9fa16e3e83d25770@mail.gmail.com><BE4EDDA84CB64AA98F18D412B51912FC@abisoft.biz><b39273530910150504l5c7411b9g39b9ed33c592e85f@mail.gmail.com><02C2EFD4CF8D4ACF8EEC0F77C4D75028@abisoft.biz>
	<414DC781-A0F6-418E-9AAB-FA4307634932@jerris.com>
	<14E8C50A8F4D41559AFCCD8838350F4F@abisoft.biz>
Message-ID: <5F579783-33FE-4520-9DC9-7C880DFB34EF@jerris.com>

You will want to use sendevent with a unique-id header and a eavesdrop- 
command header.  Also please note you will want to use svn revision  
15175 or later, I just fixed a segfault in that code.

Mike

On Oct 19, 2009, at 11:11 AM, Nikita Belov wrote:

> Thanks, Mike, for idea. But what is the syntax for this session  
> message?
> I tried this:
>
> sendmsg e8e4f0ed-a0cc-4dff-b7e1-09eeade5df05
> eavesdrop-command: 1
>
> but it doesn't work.
>
>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org  
>> [mailto:freeswitch-
>> users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
>> Sent: Monday, October 19, 2009 5:19 PM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] uuid_send_dtmf fails (was: conference
>> call)
>>
>> There is an event you can send as well to switch them, it your trying
>> to switch it via event socket, that should be better, its not on the
>> wiki, but
>>
>> a session message with
>> eavesdrop-command header with data as the same as dtmf
>> should do the trick
>>
>> Mike
>>
>>
>> On Oct 16, 2009, at 11:54 AM, Nikita Belov wrote:
>>
>>> Yes, it is what I need. But now I have problem with sending dtmf.
>>> Here what
>>> I've done:
>>> [root at centos4-4-vm ~]# telnet localhost 8021
>>> Trying 127.0.0.1...
>>> Connected to localhost.localdomain (127.0.0.1).
>>> Escape character is '^]'.
>>> Content-Type: auth/request
>>>
>>> auth ClueCon
>>>
>>> Content-Type: command/reply
>>> Reply-Text: +OK accepted
>>>
>>> api originate user/1007 at master.agent.starpoundtech.net &park()
>>>
>>> Content-Type: api/response
>>> Content-Length: 41
>>>
>>> +OK bba3b45a-4cc1-48af-a15d-1052d5f11371
>>>
>>> SendMsg bba3b45a-4cc1-48af-a15d-1052d5f11371
>>> call-command: execute
>>> execute-app-name: eavesdrop
>>> execute-app-arg: cd99f999-9b47-457e-8439-1d366e015b8c
>>>
>>> Content-Type: command/reply
>>> Reply-Text: +OK
>>>
>>> Here I had started to hear A and B. Here what I saw in FS log:
>>> 2009-10-18 03:22:47 [DEBUG] switch_core_session.c:706
>>> switch_core_session_queue_private_event() Send signal
>>> sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes [BREAK]
>>> 2009-10-18 03:22:47 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event 
>>> ()
>>> sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes Command Execute
>>> eavesdrop(cd99f999-9b47-457e-8439-1d366e015b8c)
>>> 2009-10-18 03:22:47 [DEBUG] switch_core_media_bug.c:297
>>> switch_core_media_bug_add() Attaching BUG to
>>> sofia/internal/1006 at master.agent.starpoundtech.net
>>> 2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234
>>> switch_core_session_read_frame()
>>> sofia/internal/1006 at master.agent.starpoundtech.net receive message
>>> [TRANSCODING_NECESSARY]
>>> 2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234
>>> switch_core_session_read_frame()
>>> sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes receive message
>>> [TRANSCODING_NECESSARY]
>>>
>>> Then I run command:
>>> api uuid_send_dtmf bba3b45a-4cc1-48af-a15d-1052d5f11371 1
>>>
>>> Content-Type: api/response
>>> Content-Length: 14
>>>
>>> -ERR no reply
>>>
>>> Log:
>>> 2009-10-18 03:24:01 [DEBUG] switch_core_io.c:1190
>>> switch_core_session_send_dtmf_string()
>>> sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes send dtmf
>>> digit=1 ms=250 samples=2000
>>> 2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1282 do_2833() Send start
>>> packet
>>> for [1] ts=2241760 dur=160/160/2000 seq=21346
>>> 2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
>>> packet
>>> for [1] ts=2241760 dur=320/320/2000 seq=21347
>>> 2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
>>> packet
>>> for [1] ts=2241760 dur=480/480/2000 seq=21348
>>> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
>>> packet
>>> for [1] ts=2241760 dur=640/640/2000 seq=21349
>>> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
>>> packet
>>> for [1] ts=2241760 dur=800/800/2000 seq=21350
>>> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
>>> packet
>>> for [1] ts=2241760 dur=960/960/2000 seq=21351
>>> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
>>> packet
>>> for [1] ts=2241760 dur=1120/1120/2000 seq=21352
>>> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
>>> packet
>>> for [1] ts=2241760 dur=1280/1280/2000 seq=21353
>>> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
>>> packet
>>> for [1] ts=2241760 dur=1440/1440/2000 seq=21354
>>> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
>>> packet
>>> for [1] ts=2241760 dur=1600/1600/2000 seq=21355
>>> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
>>> packet
>>> for [1] ts=2241760 dur=1760/1760/2000 seq=21356
>>> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
>>> packet
>>> for [1] ts=2241760 dur=1920/1920/2000 seq=21357
>>> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end
>>> packet for
>>> [1] ts=2241760 dur=2080/2080/2000 seq=21358
>>> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end
>>> packet for
>>> [1] ts=2241760 dur=2080/2080/2000 seq=21359
>>> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end
>>> packet for
>>> [1] ts=2241760 dur=2080/2080/2000 seq=21360
>>>
>>> But both A and B couldn't hear me.
>>> Btw, after I had send dtmf "1" manually from my phone. B started to
>>> hear me.
>>> There was this record in log:
>>> 2009-10-18 03:47:55 [DEBUG] switch_rtp.c:1767  
>>> switch_rtp_dequeue_dtmf
>>> () RTP
>>> RECV DTMF 1:2240
>>>
>>> Does anybody know, what had I done wrong?
>>>
>>> ___
>>>
>>> Thanks, Nikita
>>>
>>>> -----Original Message-----
>>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>>> [mailto:freeswitch-
>>>> users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker
>>>> Sent: Thursday, October 15, 2009 4:04 PM
>>>> To: freeswitch-users at lists.freeswitch.org
>>>> Subject: Re: [Freeswitch-users] conference call
>>>>
>>>> The wiki has a pretty good explanation of how eavesdrop works.
>>>> Enabling a talk path to A or B or both A and B requires dtmf.
>>>>
>>>> So, if C hits the 1 button on the phone they can talk to the UUID  
>>>> you
>>>> bound the eavesdrop to.   2 talks to the other party.  3 talks to
>>>> both
>>>> parties.
>>>>
>>>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop
>>>>
>>>> Adding more api calls to control eavesdrop via event socket  
>>>> should be
>>>> possible, though I haven't looked at the code to see what it would
>>>> take...
>>>>
>>>> On Thu, Oct 15, 2009 at 3:16 AM, Nikita Belov <nbelov at abisoft.spb.ru 
>>>> >
>>>> wrote:
>>>>> Thanks for answer.
>>>>>
>>>>>
>>>>>
>>>>> As I've understood eavesdrop allows C to hear A and B, but A and B
>>>>> will
>>>> not
>>>>> hear C. But in my case B SHOULD hear C. In my case A - client, B -
>>>>> agent
>>>>> worked with client and C - coach/supervisor for B. Can I make such
>>>>> conference in FS using eavesdrop or other API command?
>>>>>
>>>>>
>>>>>
>>>>> ___
>>>>>
>>>>>
>>>>>
>>>>> Thanks, Nikita
>>>>>
>>>>>
>>>>>
>>>>> ________________________________
>>>>>
>>>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf
>>>>> Of Joao
>>>>> Mesquita
>>>>> Sent: Wednesday, October 14, 2009 7:38 PM
>>>>> To: freeswitch-users at lists.freeswitch.org
>>>>> Subject: Re: [Freeswitch-users] conference call
>>>>>
>>>>>
>>>>>
>>>>> Look at eavesdrop on the wiki.
>>>>>
>>>>> JM
>>>>>
>>>>> 2009/10/14 Nikita Belov <nbelov at abisoft.spb.ru>
>>>>>
>>>>> HI all,
>>>>>
>>>>>
>>>>>
>>>>> I want to configure FS to make special conference call between  
>>>>> three
>>>> users
>>>>> (A, B, C).  In this conference C will hear A and B, but A will
>>>>> hear only
>>>> B.
>>>>> Can I make it using FS API commands? Does anybody know which
>>>>> approach is
>>>>> better to use?
>>>>>
>>>>>
>>>>>
>>>>> ___
>>>>>
>>>>>
>>>>>
>>>>> Thanks, Nikita
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
>>>>> freeswitch-
>>>>> users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
>>>>> freeswitch-
>>>>> users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> -Rupa
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>> users
>>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From helmut.kuper at ewetel.de  Mon Oct 19 08:53:18 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Mon, 19 Oct 2009 17:53:18 +0200
Subject: [Freeswitch-users] Qustion about INFO
	messages	after	Connect/Answer
In-Reply-To: <4ADC85C0.7060908@ewetel.de>
References: <4AD859A4.5000101@ewetel.de>	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>	<4ADC7C09.6070503@ewetel.de>	<191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>
	<4ADC85C0.7060908@ewetel.de>
Message-ID: <4ADC8B6E.7070404@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Anthony,

just to make it clear: my goal is to avoid to see something (newly
introduced since 1 or 2 weeks) like "1/a/890327" for outgoing in the
caller's display after answering the call (for openzap calls). I want
simply e.g. "89327". I don't want to put the call into "answer" state
before the called side demands it. So I'm not sure if sip_callee_* is
the right way to follow as long as it needs the answer state.

regards
helmut


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Version: GnuPG v1.4.7 (MingW32)

iD8DBQFK3Itu4tZeNddg3dwRAvREAKCrZcRM+TnypdVOczpe2+q26b3BYQCfadrY
tdA2XIc/hpuIv/916eu4GFo=
=y6Md
-----END PGP SIGNATURE-----


From christian.loeschenkohl at xpirio.com  Mon Oct 19 09:07:10 2009
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Mon, 19 Oct 2009 18:07:10 +0200
Subject: [Freeswitch-users] check to see if freeswitch is alive
Message-ID: <4ADC8EAE.8020603@xpirio.com>

hello

we have the problem here that our freeswitch server "freezes" from time
to time (no sip traffic is possible any more).

has somebody here expirience or any idea how to monitor freeswitch to answer
the questions

- is the server alive
- does he process sip messages (invites, registrations)
and so on

the monitoring should work via cron local and remote (with esl for example)

a simple "fs_cli -x 'version'" wouldn't work - it's too simple, am i right?

any ideas would be helpful

br


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From jerry.richards at teotech.com  Mon Oct 19 09:13:48 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 19 Oct 2009 09:13:48 -0700
Subject: [Freeswitch-users] Adding Leading Digits To CALLING Number of
	Outgoing Call
Message-ID: <49D347B4795647BF9C35D28EDD207099@greyhawk.tonecommander.com>


How do I use the dial plan to add leading digits to an outgoing call through
a gateway?

My internal phone number is 5380, but when FS sends the call to the gateway
I want the CALLING party to be 4253495380.

Best Regards,
Jerry



From rob4manhere at gmail.com  Mon Oct 19 09:23:44 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Mon, 19 Oct 2009 11:23:44 -0500
Subject: [Freeswitch-users] check to see if freeswitch is alive
In-Reply-To: <4ADC8EAE.8020603@xpirio.com>
References: <4ADC8EAE.8020603@xpirio.com>
Message-ID: <8A38EE37-E03F-48F3-8B54-220BC2DC2363@gmail.com>

Hi Christian,

You can subscribe and monitor the heartbeat event either locally or  
remotely:
http://wiki.freeswitch.org/wiki/Perl_freeswitch_client_example

You could also send an application level SIP packet, like a ping, to  
Freeswitch externally.  I have a small script if you can't find one.

More importantly, while monitoring is good, its probably worth  
investigating the root cause of the freezes since thats not normal.

Rob


On Oct 19, 2009, at 11:07 AM, Christian L?schenkohl wrote:

> hello
>
> we have the problem here that our freeswitch server "freezes" from  
> time
> to time (no sip traffic is possible any more).
>
> has somebody here expirience or any idea how to monitor freeswitch  
> to answer
> the questions
>
> - is the server alive
> - does he process sip messages (invites, registrations)
> and so on
>
> the monitoring should work via cron local and remote (with esl for  
> example)
>
> a simple "fs_cli -x 'version'" wouldn't work - it's too simple, am i  
> right?
>
> any ideas would be helpful
>
> br
>
>
> -- 
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 (0) 5 77 11 - 1000
> F  +43 (0) 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From woof at iwoof.org  Mon Oct 19 09:28:47 2009
From: woof at iwoof.org (Andy Spitzer)
Date: Mon, 19 Oct 2009 12:28:47 -0400
Subject: [Freeswitch-users] check to see if freeswitch is alive
In-Reply-To: <4ADC8EAE.8020603@xpirio.com>
References: <4ADC8EAE.8020603@xpirio.com>
Message-ID: <op.u112p9fjg0dksq@woof3.fuzzy>

Woof!

On Mon, 19 Oct 2009 12:07:10 -0400, Christian L?schenkohl  
<christian.loeschenkohl at xpirio.com> wrote:

> any ideas would be helpful

We run a perl script that checks if the servers are responding to  
requests.  It can send OPTIONS, and PING requests to various servers  
periodically.  If the response it gets back isn't correct, it can send  
e-mail.  It'll keep testing and send an e-mail again when the system is  
working again.  It's defaults setup for testing sipXecs servers, but it's  
easy enough to set the various command line arguments to point it at  
FreeSWITCH directly.

Code here.
    http://code.sipfoundry.org/browse/~raw,r=14874/sipXecs/main/sipXtools/src/sipx-servtest.in

--Woof!


From woof at iwoof.org  Mon Oct 19 09:28:47 2009
From: woof at iwoof.org (Andy Spitzer)
Date: Mon, 19 Oct 2009 12:28:47 -0400
Subject: [Freeswitch-users] check to see if freeswitch is alive
In-Reply-To: <4ADC8EAE.8020603@xpirio.com>
References: <4ADC8EAE.8020603@xpirio.com>
Message-ID: <op.u112p9fjg0dksq@woof3.fuzzy>

Woof!

On Mon, 19 Oct 2009 12:07:10 -0400, Christian L?schenkohl  
<christian.loeschenkohl at xpirio.com> wrote:

> any ideas would be helpful

We run a perl script that checks if the servers are responding to  
requests.  It can send OPTIONS, and PING requests to various servers  
periodically.  If the response it gets back isn't correct, it can send  
e-mail.  It'll keep testing and send an e-mail again when the system is  
working again.  It's defaults setup for testing sipXecs servers, but it's  
easy enough to set the various command line arguments to point it at  
FreeSWITCH directly.

Code here.
    http://code.sipfoundry.org/browse/~raw,r=14874/sipXecs/main/sipXtools/src/sipx-servtest.in

--Woof!


From msc at freeswitch.org  Mon Oct 19 09:39:53 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 19 Oct 2009 09:39:53 -0700
Subject: [Freeswitch-users] check to see if freeswitch is alive
In-Reply-To: <4ADC8EAE.8020603@xpirio.com>
References: <4ADC8EAE.8020603@xpirio.com>
Message-ID: <87f2f3b90910190939h29ced665gb9681a8946d00212@mail.gmail.com>

2009/10/19 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>

> hello
>
> we have the problem here that our freeswitch server "freezes" from time
> to time (no sip traffic is possible any more).
>

Monitoring is definitely important, but I'm sure the FS devs would like to
know more about what happens when FS "freezes" - is it FS or some other
process that is messing things up? If there's an issue with FS then the
developers would like to know more.
-MC
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From nbelov at abisoft.spb.ru  Mon Oct 19 09:42:26 2009
From: nbelov at abisoft.spb.ru (Nikita Belov)
Date: Mon, 19 Oct 2009 20:42:26 +0400
Subject: [Freeswitch-users] uuid_send_dtmf fails (was: conference call)
In-Reply-To: <5F579783-33FE-4520-9DC9-7C880DFB34EF@jerris.com>
References: <064A5201C10D4D20AC76EB93CA5EA4B5@abisoft.biz><ebbc2d550910140838v3125579ev9fa16e3e83d25770@mail.gmail.com><BE4EDDA84CB64AA98F18D412B51912FC@abisoft.biz><b39273530910150504l5c7411b9g39b9ed33c592e85f@mail.gmail.com><02C2EFD4CF8D4ACF8EEC0F77C4D75028@abisoft.biz><414DC781-A0F6-418E-9AAB-FA4307634932@jerris.com><14E8C50A8F4D41559AFCCD8838350F4F@abisoft.biz>
	<5F579783-33FE-4520-9DC9-7C880DFB34EF@jerris.com>
Message-ID: <FFF06EC411464736B44C44B58646A8C6@abisoft.biz>

And what event name to use for sendevent command? Sorry for importunity.

sendevent ???

Unique-ID: b9d6a35c-ee0c-4203-8d65-ed816e0a9c19

eavesdrop-command: 1

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Monday, October 19, 2009 7:34 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] uuid_send_dtmf fails (was: conference call)

 

You will want to use sendevent with a unique-id header and a
eavesdrop-command header.  Also please note you will want to use svn
revision 15175 or later, I just fixed a segfault in that code.

 

Mike

 

On Oct 19, 2009, at 11:11 AM, Nikita Belov wrote:





Thanks, Mike, for idea. But what is the syntax for this session message?
I tried this:

sendmsg e8e4f0ed-a0cc-4dff-b7e1-09eeade5df05
eavesdrop-command: 1

but it doesn't work.





-----Original Message-----

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-

users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris

Sent: Monday, October 19, 2009 5:19 PM

To: freeswitch-users at lists.freeswitch.org

Subject: Re: [Freeswitch-users] uuid_send_dtmf fails (was: conference

call)

 

There is an event you can send as well to switch them, it your trying

to switch it via event socket, that should be better, its not on the

wiki, but

 

a session message with

eavesdrop-command header with data as the same as dtmf

should do the trick

 

Mike

 

 

On Oct 16, 2009, at 11:54 AM, Nikita Belov wrote:

 

Yes, it is what I need. But now I have problem with sending dtmf.

Here what

I've done:

[root at centos4-4-vm ~]# telnet localhost 8021

Trying 127.0.0.1...

Connected to localhost.localdomain (127.0.0.1).

Escape character is '^]'.

Content-Type: auth/request

 

auth ClueCon

 

Content-Type: command/reply

Reply-Text: +OK accepted

 

api originate user/1007 at master.agent.starpoundtech.net &park()

 

Content-Type: api/response

Content-Length: 41

 

+OK bba3b45a-4cc1-48af-a15d-1052d5f11371

 

SendMsg bba3b45a-4cc1-48af-a15d-1052d5f11371

call-command: execute

execute-app-name: eavesdrop

execute-app-arg: cd99f999-9b47-457e-8439-1d366e015b8c

 

Content-Type: command/reply

Reply-Text: +OK

 

Here I had started to hear A and B. Here what I saw in FS log:

2009-10-18 03:22:47 [DEBUG] switch_core_session.c:706

switch_core_session_queue_private_event() Send signal

sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes [BREAK]

2009-10-18 03:22:47 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event()

sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes Command Execute

eavesdrop(cd99f999-9b47-457e-8439-1d366e015b8c)

2009-10-18 03:22:47 [DEBUG] switch_core_media_bug.c:297

switch_core_media_bug_add() Attaching BUG to

sofia/internal/1006 at master.agent.starpoundtech.net

2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234

switch_core_session_read_frame()

sofia/internal/1006 at master.agent.starpoundtech.net receive message

[TRANSCODING_NECESSARY]

2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234

switch_core_session_read_frame()

sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes receive message

[TRANSCODING_NECESSARY]

 

Then I run command:

api uuid_send_dtmf bba3b45a-4cc1-48af-a15d-1052d5f11371 1

 

Content-Type: api/response

Content-Length: 14

 

-ERR no reply

 

Log:

2009-10-18 03:24:01 [DEBUG] switch_core_io.c:1190

switch_core_session_send_dtmf_string()

sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes send dtmf

digit=1 ms=250 samples=2000

2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1282 do_2833() Send start

packet

for [1] ts=2241760 dur=160/160/2000 seq=21346

2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle

packet

for [1] ts=2241760 dur=320/320/2000 seq=21347

2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle

packet

for [1] ts=2241760 dur=480/480/2000 seq=21348

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle

packet

for [1] ts=2241760 dur=640/640/2000 seq=21349

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle

packet

for [1] ts=2241760 dur=800/800/2000 seq=21350

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle

packet

for [1] ts=2241760 dur=960/960/2000 seq=21351

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle

packet

for [1] ts=2241760 dur=1120/1120/2000 seq=21352

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle

packet

for [1] ts=2241760 dur=1280/1280/2000 seq=21353

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle

packet

for [1] ts=2241760 dur=1440/1440/2000 seq=21354

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle

packet

for [1] ts=2241760 dur=1600/1600/2000 seq=21355

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle

packet

for [1] ts=2241760 dur=1760/1760/2000 seq=21356

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle

packet

for [1] ts=2241760 dur=1920/1920/2000 seq=21357

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end

packet for

[1] ts=2241760 dur=2080/2080/2000 seq=21358

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end

packet for

[1] ts=2241760 dur=2080/2080/2000 seq=21359

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end

packet for

[1] ts=2241760 dur=2080/2080/2000 seq=21360

 

But both A and B couldn't hear me.

Btw, after I had send dtmf "1" manually from my phone. B started to

hear me.

There was this record in log:

2009-10-18 03:47:55 [DEBUG] switch_rtp.c:1767 switch_rtp_dequeue_dtmf

() RTP

RECV DTMF 1:2240

 

Does anybody know, what had I done wrong?

 

___

 

Thanks, Nikita

 

-----Original Message-----

From: freeswitch-users-bounces at lists.freeswitch.org

[mailto:freeswitch-

users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker

Sent: Thursday, October 15, 2009 4:04 PM

To: freeswitch-users at lists.freeswitch.org

Subject: Re: [Freeswitch-users] conference call

 

The wiki has a pretty good explanation of how eavesdrop works.

Enabling a talk path to A or B or both A and B requires dtmf.

 

So, if C hits the 1 button on the phone they can talk to the UUID you

bound the eavesdrop to.   2 talks to the other party.  3 talks to

both

parties.

 

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop

 

Adding more api calls to control eavesdrop via event socket should be

possible, though I haven't looked at the code to see what it would

take...

 

On Thu, Oct 15, 2009 at 3:16 AM, Nikita Belov <nbelov at abisoft.spb.ru>

wrote:

Thanks for answer.

 

 

 

As I've understood eavesdrop allows C to hear A and B, but A and B

will

not

hear C. But in my case B SHOULD hear C. In my case A - client, B -

agent

worked with client and C - coach/supervisor for B. Can I make such

conference in FS using eavesdrop or other API command?

 

 

 

___

 

 

 

Thanks, Nikita

 

 

 

________________________________

 

From: freeswitch-users-bounces at lists.freeswitch.org

[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf

Of Joao

Mesquita

Sent: Wednesday, October 14, 2009 7:38 PM

To: freeswitch-users at lists.freeswitch.org

Subject: Re: [Freeswitch-users] conference call

 

 

 

Look at eavesdrop on the wiki.

 

JM

 

2009/10/14 Nikita Belov <nbelov at abisoft.spb.ru>

 

HI all,

 

 

 

I want to configure FS to make special conference call between three

users

(A, B, C).  In this conference C will hear A and B, but A will

hear only

B.

Can I make it using FS API commands? Does anybody know which

approach is

better to use?

 

 

 

___

 

 

 

Thanks, Nikita

 

_______________________________________________

FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-

users

http://www.freeswitch.org

 

 

 

_______________________________________________

FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-

users

http://www.freeswitch.org

 

 

 

 

 

--

-Rupa

 

_______________________________________________

FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-

users

http://www.freeswitch.org

 

 

 

_______________________________________________

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FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-

users

http://www.freeswitch.org

 

 

_______________________________________________

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FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org




_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From brian at freeswitch.org  Mon Oct 19 09:58:49 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 19 Oct 2009 11:58:49 -0500
Subject: [Freeswitch-users] uuid_send_dtmf fails (was: conference call)
In-Reply-To: <FFF06EC411464736B44C44B58646A8C6@abisoft.biz>
References: <064A5201C10D4D20AC76EB93CA5EA4B5@abisoft.biz><ebbc2d550910140838v3125579ev9fa16e3e83d25770@mail.gmail.com><BE4EDDA84CB64AA98F18D412B51912FC@abisoft.biz><b39273530910150504l5c7411b9g39b9ed33c592e85f@mail.gmail.com><02C2EFD4CF8D4ACF8EEC0F77C4D75028@abisoft.biz><414DC781-A0F6-418E-9AAB-FA4307634932@jerris.com><14E8C50A8F4D41559AFCCD8838350F4F@abisoft.biz>
	<5F579783-33FE-4520-9DC9-7C880DFB34EF@jerris.com>
	<FFF06EC411464736B44C44B58646A8C6@abisoft.biz>
Message-ID: <01AAA3C0-54DE-4BE0-B1D4-35B5B9C493DF@freeswitch.org>

Please review this:

http://wiki.freeswitch.org/wiki/Mod_event_socket#sendmsg

SendMsg <uuid>
call-command: execute
execute-app-name: <one of the applications>
execute-app-arg: <application data>



/b

On Oct 19, 2009, at 11:42 AM, Nikita Belov wrote:

> And what event name to use for sendevent command? Sorry for  
> importunity.
> sendevent ???
> Unique-ID: b9d6a35c-ee0c-4203-8d65-ed816e0a9c19
> eavesdrop-command: 1
>



From jerry.richards at teotech.com  Mon Oct 19 10:15:21 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 19 Oct 2009 10:15:21 -0700
Subject: [Freeswitch-users] Adding Leading Digits To CALLING Number of
	Outgoing Call
Message-ID: <3121A49E9BE549989158DDC0008498E7@greyhawk.tonecommander.com>


Okay, I figured it out.  I added the following line to the default.xml file,
just prior to the "bridge" action:

<action application="set"
data="effective_caller_id_number=425349${caller_id_number}"/>

Now, 425349 is prepended to the outgoing call's caller ID.

Best Regards,
Jerry


-----Original Message-----
From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Monday, October 19, 2009 9:14 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: Adding Leading Digits To CALLING Number of Outgoing Call


How do I use the dial plan to add leading digits to an outgoing call through
a gateway?

My internal phone number is 5380, but when FS sends the call to the gateway
I want the CALLING party to be 4253495380.

Best Regards,
Jerry



From m.sobkow at marketelsystems.com  Mon Oct 19 10:35:10 2009
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Mon, 19 Oct 2009 11:35:10 -0600
Subject: [Freeswitch-users] Is anyone running Ubuntu 8.04/Hardy?
Message-ID: <4ADCA34E.5000700@marketelsystems.com>

Everyone I've emailed with on the dev list is running the current 
release of Ubuntu, not 8.04/Hardy.  We're having some problems with 
loading mod_sofia on Hardy, so unless I get some progress/help by end of 
day Wednesday we're going to have to rebuild our development server with 
the most recent release of Ubuntu on Thursday and Friday.

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 



From nbelov at abisoft.spb.ru  Mon Oct 19 10:36:32 2009
From: nbelov at abisoft.spb.ru (Nikita Belov)
Date: Mon, 19 Oct 2009 21:36:32 +0400
Subject: [Freeswitch-users] uuid_send_dtmf fails (was: conference call)
In-Reply-To: <01AAA3C0-54DE-4BE0-B1D4-35B5B9C493DF@freeswitch.org>
References: <064A5201C10D4D20AC76EB93CA5EA4B5@abisoft.biz><ebbc2d550910140838v3125579ev9fa16e3e83d25770@mail.gmail.com><BE4EDDA84CB64AA98F18D412B51912FC@abisoft.biz><b39273530910150504l5c7411b9g39b9ed33c592e85f@mail.gmail.com><02C2EFD4CF8D4ACF8EEC0F77C4D75028@abisoft.biz><414DC781-A0F6-418E-9AAB-FA4307634932@jerris.com><14E8C50A8F4D41559AFCCD8838350F4F@abisoft.biz><5F579783-33FE-4520-9DC9-7C880DFB34EF@jerris.com><FFF06EC411464736B44C44B58646A8C6@abisoft.biz>
	<01AAA3C0-54DE-4BE0-B1D4-35B5B9C493DF@freeswitch.org>
Message-ID: <3F5FD88C28BF428CAA935EDE7357B686@abisoft.biz>

Yeah. I'm using it for starting eavesdrop:
SendMsg cced4b9a-b6de-4be1-8c12-3d18cc6e8454
call-command: execute
execute-app-name: eavesdrop
execute-app-arg: 723f3dbb-b87b-4cd4-98fc-698eed7f2bdb

But we are discussing how to switch eavesdrop to allow "spy" to speak with
one of the sides. Do you know any application which can I use for this? Mike
said I should use sendevent command to control eavesdrop, but I still
confused how can I do it.

> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> users-bounces at lists.freeswitch.org] On Behalf Of Brian West
> Sent: Monday, October 19, 2009 8:59 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] uuid_send_dtmf fails (was: conference
> call)
> 
> Please review this:
> 
> http://wiki.freeswitch.org/wiki/Mod_event_socket#sendmsg
> 
> SendMsg <uuid>
> call-command: execute
> execute-app-name: <one of the applications>
> execute-app-arg: <application data>
> 
> 
> 
> /b
> 
> On Oct 19, 2009, at 11:42 AM, Nikita Belov wrote:
> 
> > And what event name to use for sendevent command? Sorry for
> > importunity.
> > sendevent ???
> > Unique-ID: b9d6a35c-ee0c-4203-8d65-ed816e0a9c19
> > eavesdrop-command: 1
> >
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From chris at cloudtel.com  Mon Oct 19 10:38:28 2009
From: chris at cloudtel.com (Chris Burns)
Date: Mon, 19 Oct 2009 13:38:28 -0400
Subject: [Freeswitch-users] check to see if freeswitch is alive
In-Reply-To: <87f2f3b90910190939h29ced665gb9681a8946d00212@mail.gmail.com>
References: <4ADC8EAE.8020603@xpirio.com>
	<87f2f3b90910190939h29ced665gb9681a8946d00212@mail.gmail.com>
Message-ID: <200910191338.28439.chris@cloudtel.com>

There's a check_sip plugin for nagios if you're into that sorta thing

On October 19, 2009 12:39:53 pm Michael Collins wrote:
> 2009/10/19 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>
>
> > hello
> >
> > we have the problem here that our freeswitch server "freezes" from time
> > to time (no sip traffic is possible any more).
>
> Monitoring is definitely important, but I'm sure the FS devs would like to
> know more about what happens when FS "freezes" - is it FS or some other
> process that is messing things up? If there's an issue with FS then the
> developers would like to know more.
> -MC




From gmaruzz at celliax.org  Mon Oct 19 10:46:28 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 19 Oct 2009 19:46:28 +0200
Subject: [Freeswitch-users] Is anyone running Ubuntu 8.04/Hardy?
In-Reply-To: <4ADCA34E.5000700@marketelsystems.com>
References: <4ADCA34E.5000700@marketelsystems.com>
Message-ID: <7b197bef0910191046y6aa41986nf1a4f77061385bd7@mail.gmail.com>

8.04 was known to be very stable and reliable with FreeSWITCH.

First time I heard of problems with it...

What kind of problems? Problems you have not found with current version?

-giovanni

On Mon, Oct 19, 2009 at 7:35 PM, Mark Sobkow
<m.sobkow at marketelsystems.com> wrote:
> Everyone I've emailed with on the dev list is running the current
> release of Ubuntu, not 8.04/Hardy. ?We're having some problems with
> loading mod_sofia on Hardy, so unless I get some progress/help by end of
> day Wednesday we're going to have to rebuild our development server with
> the most recent release of Ubuntu on Thursday and Friday.
>
> --
> Mark Sobkow
> Senior Developer
> MarkeTel Multi-Line Dialing Systems LTD.
> 428 Victoria Ave
> Regina, SK S4N-0P6
> Toll-Free: 800-289-8616-X533
> Local: 306-359-6893-X533
> Fax: 306-359-6879
> Email: m.sobkow at marketelsystems.com
> Web: http://www.marketelsystems.com
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From christian.loeschenkohl at xpirio.com  Mon Oct 19 11:07:19 2009
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Mon, 19 Oct 2009 20:07:19 +0200
Subject: [Freeswitch-users] check to see if freeswitch is alive
In-Reply-To: <87f2f3b90910190939h29ced665gb9681a8946d00212@mail.gmail.com>
References: <4ADC8EAE.8020603@xpirio.com>
	<87f2f3b90910190939h29ced665gb9681a8946d00212@mail.gmail.com>
Message-ID: <4ADCAAD7.8010401@xpirio.com>

i know, we tried to to get closer to this with anthony (see thread "stability problems",
started on 03.09.2009).
the old issue described a server that used more profiles (8-10) and some event socket scripts.
we came to the conclusion that it could be a race condition when using multiple profiles.

now on the server we need monitor we use only internal functions and conditions
- no external scripts
- no database
- only on profile
- only freeswitch stock functions

the symptoms are the same
- no more progress displays on the fs_cli console (debug level info)
- invites are answered with trying, but no more communication after this
- customers said that option requests are answered (they could not switch to fallback
   because this worked)

we did use the trunk version and have now updated to 15162, but i don't think our problem is
solved with any update so far.

br

On 2009-10-19 18:39, Michael Collins wrote:
>
>
> 2009/10/19 Christian L?schenkohl <christian.loeschenkohl at xpirio.com
> <mailto:christian.loeschenkohl at xpirio.com>>
>
>     hello
>
>     we have the problem here that our freeswitch server "freezes" from time
>     to time (no sip traffic is possible any more).
>
>
> Monitoring is definitely important, but I'm sure the FS devs would like
> to know more about what happens when FS "freezes" - is it FS or some
> other process that is messing things up? If there's an issue with FS
> then the developers would like to know more.
> -MC
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From christian.loeschenkohl at xpirio.com  Mon Oct 19 11:18:07 2009
From: christian.loeschenkohl at xpirio.com (=?UTF-8?B?Q2hyaXN0aWFuIEzDtnNjaGVua29obA==?=)
Date: Mon, 19 Oct 2009 20:18:07 +0200
Subject: [Freeswitch-users] check to see if freeswitch is alive
In-Reply-To: <op.u112p9fjg0dksq@woof3.fuzzy>
References: <4ADC8EAE.8020603@xpirio.com> <op.u112p9fjg0dksq@woof3.fuzzy>
Message-ID: <4ADCAD5F.7020702@xpirio.com>

thank you for the script, we will try this asap

br

On 2009-10-19 18:28, Andy Spitzer wrote:
> Woof!
>
> On Mon, 19 Oct 2009 12:07:10 -0400, Christian L?schenkohl
> <christian.loeschenkohl at xpirio.com>  wrote:
>
>> any ideas would be helpful
>
> We run a perl script that checks if the servers are responding to
> requests.  It can send OPTIONS, and PING requests to various servers
> periodically.  If the response it gets back isn't correct, it can send
> e-mail.  It'll keep testing and send an e-mail again when the system is
> working again.  It's defaults setup for testing sipXecs servers, but it's
> easy enough to set the various command line arguments to point it at
> FreeSWITCH directly.
>
> Code here.
>      http://code.sipfoundry.org/browse/~raw,r=14874/sipXecs/main/sipXtools/src/sipx-servtest.in
>
> --Woof!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From anthony.minessale at gmail.com  Mon Oct 19 11:19:25 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 19 Oct 2009 13:19:25 -0500
Subject: [Freeswitch-users] check to see if freeswitch is alive
In-Reply-To: <4ADCAAD7.8010401@xpirio.com>
References: <4ADC8EAE.8020603@xpirio.com>
	<87f2f3b90910190939h29ced665gb9681a8946d00212@mail.gmail.com>
	<4ADCAAD7.8010401@xpirio.com>
Message-ID: <191c3a030910191119n3dae38d8y17c5361f74724d50@mail.gmail.com>

do you have a jira ticket open for this?
The mailing list is not a good place to ask for bugs to be fixed because I
can't possibly scour them for responses.

If you can update to latest trunk and still reproduce this problem:
make sure gdb and gcore is installed on your box

from the FS build root enter

./support-d/fscore_pb gcore

and submit the url it generates to me


2009/10/19 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>

> i know, we tried to to get closer to this with anthony (see thread
> "stability problems",
> started on 03.09.2009).
> the old issue described a server that used more profiles (8-10) and some
> event socket scripts.
> we came to the conclusion that it could be a race condition when using
> multiple profiles.
>
> now on the server we need monitor we use only internal functions and
> conditions
> - no external scripts
> - no database
> - only on profile
> - only freeswitch stock functions
>
> the symptoms are the same
> - no more progress displays on the fs_cli console (debug level info)
> - invites are answered with trying, but no more communication after this
> - customers said that option requests are answered (they could not switch
> to fallback
>   because this worked)
>
> we did use the trunk version and have now updated to 15162, but i don't
> think our problem is
> solved with any update so far.
>
> br
>
> On 2009-10-19 18:39, Michael Collins wrote:
> >
> >
> > 2009/10/19 Christian L?schenkohl <christian.loeschenkohl at xpirio.com
> > <mailto:christian.loeschenkohl at xpirio.com>>
> >
> >     hello
> >
> >     we have the problem here that our freeswitch server "freezes" from
> time
> >     to time (no sip traffic is possible any more).
> >
> >
> > Monitoring is definitely important, but I'm sure the FS devs would like
> > to know more about what happens when FS "freezes" - is it FS or some
> > other process that is messing things up? If there's an issue with FS
> > then the developers would like to know more.
> > -MC
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 (0) 5 77 11 - 1000
> F  +43 (0) 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Mon Oct 19 11:44:50 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 19 Oct 2009 11:44:50 -0700
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
In-Reply-To: <4256bf830910182148mf7f2bf9vdcccd687e0caa1e7@mail.gmail.com>
References: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
	<177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>
	<4256bf830908041151l55980e3fifb85c75b87535426@mail.gmail.com>
	<3D0B855A-5B2C-4C6F-9306-64B0ECE97091@jerris.com>
	<4256bf830910180725y72539d98gb138934afc18d58b@mail.gmail.com>
	<A5ECDCD9-1A12-4A31-89F0-E6F4C79FE9E9@jerris.com>
	<4256bf830910182148mf7f2bf9vdcccd687e0caa1e7@mail.gmail.com>
Message-ID: <87f2f3b90910191144q25750bayfd7a7448e82e78ff@mail.gmail.com>

On Sun, Oct 18, 2009 at 9:48 PM, Matthew Fong <mattdfong at gmail.com> wrote:

> The debug level logs to the console, right? The pastebin, had log level
> debug, sofia trace on for external and default, and sofia loglevel all 9. Is
> there another log enable command I'm missing? It seems loglevel all 9
> outputs enter and exit functions, but at least to my novice eye, it's not
> too obvious why freeswitch is sending a BYE to my DID provider.
>
> Evidently the console loglevel debug did not take because there isn't a
single DEBUG level line of output. Please try again. Just remember that
debug level log messages will have A LOT of information so be patient while
sifting through it. Of course, feel free to pastebin the output and add a
link to it in this email thread.

-MC
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From freeswitch-list at puzzled.xs4all.nl  Mon Oct 19 11:45:25 2009
From: freeswitch-list at puzzled.xs4all.nl (Patrick List)
Date: Mon, 19 Oct 2009 20:45:25 +0200
Subject: [Freeswitch-users] Status of ubuntu/debian packages.
In-Reply-To: <20091019130717.GG13895@base.carmickle.com>
References: <4ABA47FB.2050100@gmail.com>	<90823c940910180249u69f18aaamc383f9a5a0c65848@mail.gmail.com>	<4ADAFC85.6000301@puzzled.xs4all.nl>
	<20091019130717.GG13895@base.carmickle.com>
Message-ID: <4ADCB3C5.1070409@puzzled.xs4all.nl>

On 10/19/2009 03:07 PM, Frank Carmickle wrote:
> On Sun, Oct 18, Patrick List wrote:
>> so there was no chance of your customized configs getting overwritten.
>> Example:
>> /opt/freeswitch/etc/freeswitch-$svn_version-samples/
>>
>> Maybe the maintainer of the debs can do something similar.
>
> We are planning on putting all of the different configs in /usr/share/doc/freeswitch/examples.  Now if I could find the time to get this done.

That would obviously work. The advantage of sticking example configs in 
their own folder designating the release (svn rev) they belong to is 
that you can do a diff between two folders to see what has changed. 
Maybe that's useful to some. Just a thought though.

Regards,
Patrick


From msc at freeswitch.org  Mon Oct 19 11:46:54 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 19 Oct 2009 11:46:54 -0700
Subject: [Freeswitch-users] Adding Leading Digits To CALLING Number of
	Outgoing Call
In-Reply-To: <3121A49E9BE549989158DDC0008498E7@greyhawk.tonecommander.com>
References: <3121A49E9BE549989158DDC0008498E7@greyhawk.tonecommander.com>
Message-ID: <87f2f3b90910191146m7a0215abjc2eddda40621eef8@mail.gmail.com>

On Mon, Oct 19, 2009 at 10:15 AM, Jerry Richards <jerry.richards at teotech.com
> wrote:

>
> Okay, I figured it out.  I added the following line to the default.xml
> file,
> just prior to the "bridge" action:
>
Way to go all Chuck Norris on this issue! :)
-MC
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From anthony.minessale at gmail.com  Mon Oct 19 13:07:04 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 19 Oct 2009 15:07:04 -0500
Subject: [Freeswitch-users] Qustion about INFO messages after
	Connect/Answer
In-Reply-To: <4ADC8B6E.7070404@ewetel.de>
References: <4AD859A4.5000101@ewetel.de>
	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>
	<4ADC7C09.6070503@ewetel.de>
	<191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>
	<4ADC85C0.7060908@ewetel.de> <4ADC8B6E.7070404@ewetel.de>
Message-ID: <191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>

please update and test trunk

1) I changed the core to remove the excess data by default in your scenario
2) I added variables you can use to control it origination_callee_id_name
origination_callee_id_number which belong in {} in the dial string eg
{origination_callee_id_number=1234}openzap/1/a/1234


On Mon, Oct 19, 2009 at 10:53 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi Anthony,
>
> just to make it clear: my goal is to avoid to see something (newly
> introduced since 1 or 2 weeks) like "1/a/890327" for outgoing in the
> caller's display after answering the call (for openzap calls). I want
> simply e.g. "89327". I don't want to put the call into "answer" state
> before the called side demands it. So I'm not sure if sip_callee_* is
> the right way to follow as long as it needs the answer state.
>
> regards
> helmut
>
>
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
>
> iD8DBQFK3Itu4tZeNddg3dwRAvREAKCrZcRM+TnypdVOczpe2+q26b3BYQCfadrY
> tdA2XIc/hpuIv/916eu4GFo=
> =y6Md
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From dmitry.bely at gmail.com  Mon Oct 19 13:23:19 2009
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Tue, 20 Oct 2009 00:23:19 +0400
Subject: [Freeswitch-users] Status of ubuntu/debian packages.
In-Reply-To: <4ADCB3C5.1070409@puzzled.xs4all.nl>
References: <4ABA47FB.2050100@gmail.com>
	<90823c940910180249u69f18aaamc383f9a5a0c65848@mail.gmail.com>
	<4ADAFC85.6000301@puzzled.xs4all.nl>
	<20091019130717.GG13895@base.carmickle.com>
	<4ADCB3C5.1070409@puzzled.xs4all.nl>
Message-ID: <90823c940910191323m9497ca6t7b78c9af6ec43294@mail.gmail.com>

On Mon, Oct 19, 2009 at 10:45 PM, Patrick List
<freeswitch-list at puzzled.xs4all.nl> wrote:
> On 10/19/2009 03:07 PM, Frank Carmickle wrote:
>> On Sun, Oct 18, Patrick List wrote:
>>> so there was no chance of your customized configs getting overwritten.
>>> Example:
>>> /opt/freeswitch/etc/freeswitch-$svn_version-samples/
>>>
>>> Maybe the maintainer of the debs can do something similar.
>>
>> We are planning on putting all of the different configs in /usr/share/doc/freeswitch/examples. ?Now if I could find the time to get this done.
>
> That would obviously work. The advantage of sticking example configs in
> their own folder designating the release (svn rev) they belong to is
> that you can do a diff between two folders to see what has changed.
> Maybe that's useful to some. Just a thought though.

For most Debian packages dpkg just asks the user what he/she would
like to do with the newer config file from .deb: keep the old file,
replace it with the new file, diff old with new files etc. Not an
option here?

- Dmitry Bely


From miles.chet at gmail.com  Mon Oct 19 13:24:19 2009
From: miles.chet at gmail.com (Roberto Martins)
Date: Mon, 19 Oct 2009 18:24:19 -0200
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <86a32abc0910171711je1e57bduf2bb672472365ce1@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<6701786D-A8AF-444E-A5A7-29E69F6E055C@gmail.com>
	<B69597BF-DA4A-440B-8CB0-08F52F33DD17@jerris.com>
	<86a32abc0910171339o6a300932v66b7f00419a0cb69@mail.gmail.com>
	<86a32abc0910171513n35d7d0echff940a16685ae1a4@mail.gmail.com>
	<86a32abc0910171521n403094bfvf27aa7c2a1c1f798@mail.gmail.com>
	<86a32abc0910171535l606eb9c6id54f7b3b5ae1b670@mail.gmail.com>
	<86a32abc0910171545l17f3f539ldba432db550c3eaa@mail.gmail.com>
	<86a32abc0910171601u6ab06e42xcf303301c85c57b1@mail.gmail.com>
	<86a32abc0910171605y73ecb75cqfb13923ad19adc56@mail.gmail.com>
	<86a32abc0910171711je1e57bduf2bb672472365ce1@mail.gmail.com>
Message-ID: <00E26046-328F-440A-81F4-671F197F114C@gmail.com>

what about http://www.atlassian.com/software/confluence/ they give  
free licenses to open source project, and FS is using JIRA.

roberto

On Oct 17, 2009, at 9:11 PM, Diego Viola wrote:

> http://moinmo.in/ is also pretty cool it seems and it's used for the  
> apache wiki and others.
>
> http://wiki.apache.org/
>
> It also has i18n support, full text search, etc.
>
> Diego
>
> On Sat, Oct 17, 2009 at 11:05 PM, Diego Viola  
> <diego.viola at gmail.com> wrote:
> With sub-domains you'll need to start all over and it will be a pain  
> to keep the documentation in-sync.
>
> With a wiki that supports real translation it's easier to add a new  
> line with the translation tag or whatever.
>
> Diego
>
>
> On Sat, Oct 17, 2009 at 11:01 PM, Diego Viola  
> <diego.viola at gmail.com> wrote:
> Take a look at this for instance:
>
> http://wiki.rubyonrails.org/start
>
> This uses DokuWiki, check the langauges option at the top (en, de,  
> el, es, fr, etc) and when you click one of them it translates the  
> actual page to the desired language... I'd love to have something  
> like that.
>
> Diego
>
>
> On Sat, Oct 17, 2009 at 10:45 PM, Diego Viola  
> <diego.viola at gmail.com> wrote:
> Instead of adding million of different domains for the languages, it  
> would be nice if we only keep wiki.freeswitch.org, and the wiki  
> software provides a list of languages to choose from.
>
> Then when we click in one of the languages we have the wiki  
> translated to the desired language.
>
> I believe if we go this way we will be more organized, and there is  
> more chances that things will get translated, and we wont be  
> repeating too much information here and there too.
>
> Just a thought... I'm open to do this work to make it happen.
>
> The different domains per language is a good idea too.
>
> Diego
>
>
> On Sat, Oct 17, 2009 at 10:35 PM, Diego Viola  
> <diego.viola at gmail.com> wrote:
> We could also do this for a better documentation.
>
> 	? Pick a new wiki engine that does internalization/translation  
> easier.
> 	? Set some documentation guidelines from the start.
> 	? Move stuff over to the new wiki.
> 	? Start with the translation/internalization on the new wiki.
>
> Diego
>
>
> On Sat, Oct 17, 2009 at 10:21 PM, Diego Viola  
> <diego.viola at gmail.com> wrote:
> Some interesting pages on the subject.
>
> http://en.wikipedia.org/wiki/Comparison_of_wiki_software
> http://www.osnews.com/story/21865/Recommend_an_Open_Source_Wiki_Platform
>
>
>
>
> On Sat, Oct 17, 2009 at 10:13 PM, Diego Viola  
> <diego.viola at gmail.com> wrote:
> Would you be open to test another wiki software, that supports  
> internalization/translation easier, if i am open to port the entire  
> documentation to the new wiki software?
>
>
>
>
>
> On Sat, Oct 17, 2009 at 8:39 PM, Diego Viola <diego.viola at gmail.com>  
> wrote:
> If we find a better wiki software that does internalization in a  
> better way, I could help porting the documentation over to the new  
> software.
>
> The sub-domain idea is also very nice.
>
> Diego
>
>
>
> On Sat, Oct 17, 2009 at 8:29 PM, Michael Jerris <mike at jerris.com>  
> wrote:
> I think we strongly lean towards using a sub-domain (if necessary) and
> maintaining other language content in the same wiki in alternate
> pages.  If the wiki software we are using is not effective to create
> multiple languages we should find a way to do it all in one.  We can
> set up an additional mailing list if necessary but also we have no
> issue with going across multiple languages in both the mailing list
> and in irc.
>
> Mike
>
> On Oct 17, 2009, at 4:01 PM, Roberto Martins wrote:
>
> > sub-domain seem to be a very wise choice...
> > ptbr
> >
> >
> >
> > On Oct 17, 2009, at 1:25 PM, Anthony Minessale wrote:
> >
> >> or someone could just ask us for pg.freeswitch.org or whatever  
> other
> >> sub-domain to avoid trademark infringement of registering 40  
> domains
> >> with freeswitch in the name for no reason.
> >>
> >>
> >> On Sat, Oct 17, 2009 at 6:06 AM, Rud? Cunha <ruda at ruda.com.br>  
> wrote:
> >> Jo?o Mesquita,
> >>
> >> Could check with Jeremiah or contact me through, so I talk to him
> >> about the forum?
> >>
> >> He could redirect the domain to my hosting server and we can put
> >> there the wiki and forums and we can communicate!
> >>
> >> Rud? Cunha
> >>
> >> 2009/10/17 Jo?o Mesquita <jmesquita at freeswitch.org>
> >>
> >> Just a heads up, I have talked to Jeremias from Khomp today and he
> >> is setting up the wiki. I will personally be adding contents to the
> >> that wiki if it ever picks up.
> >>
> >> Regards,
> >>
> >> jm
> >>
> >>
> >> On Sat, Oct 17, 2009 at 12:52 AM, Jason White <jason at jasonjgw.net>
> >> wrote:
> >> Diego Viola <diego.viola at gmail.com> wrote:
> >>> I'm with Moises and with the other people supporting this
> >> initiative.
> >>>
> >>> I'm not Brazilian, but they should be able to do whatever they
> >> want, after
> >>> all, that's how open source works, if you can do it go ahead and
> >> do it.
> >>
> >> Correct. We have enough of them as far as English-language fora are
> >> concerned;
> >> other languages are a different question altogether, though.
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org
> >> iax:guest at conference.freeswitch.org/888
> >> googletalk:conf+888 at conference.freeswitch.org
> >> pstn:213-799-1400
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From dmitry.bely at gmail.com  Mon Oct 19 13:32:04 2009
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Tue, 20 Oct 2009 00:32:04 +0400
Subject: [Freeswitch-users] Status of ubuntu/debian packages.
In-Reply-To: <4ABA47FB.2050100@gmail.com>
References: <4ABA47FB.2050100@gmail.com>
Message-ID: <90823c940910191332r719970f0m2adce06ad40738d1@mail.gmail.com>

Can you build FreeSWITCH .deb package with tls transport support?
Anyway it already depends on openssl package.

Another problem:

2009-10-20 00:28:50.728557 [CRIT] switch_loadable_module.c:871 Error
Loading module /opt/freeswitch/mod/mod_cluechoo.so
**/opt/freeswitch/mod/mod_cluechoo.so: cannot open shared object file:
No such file or directory**
2009-10-20 00:28:50.728604 [CRIT] switch_loadable_module.c:871 Error
Loading module /opt/freeswitch/mod/mod_valet_parking.so
**/opt/freeswitch/mod/mod_valet_parking.so: cannot open shared object
file: No such file or directory**

Modules mod_cluechoo & mod_valet_parking are referenced in the config
but not present in the package.

- Dmitry Bely


From abovebluesky at gmail.com  Mon Oct 19 13:41:37 2009
From: abovebluesky at gmail.com (Uncle Johny)
Date: Mon, 19 Oct 2009 13:41:37 -0700 (PDT)
Subject: [Freeswitch-users]  bind_meta_app problem with custome extension
Message-ID: <25963971.post@talk.nabble.com>


Hi guys,

I hope you can help me with this one since it looks like some FS bug or my
user error :) .I have a simple dialplan in FS which allows to transfer a
person to some 3rd party phone but it works only if you call it from default
context.
If it is not called from default context I get this error all the time:
2009-09-12 11:58:17.484680 [ERR] mod_native_file.c:68 Error opening
/usr/local/freeswitch/conf/dialplan/execute_extension:roar XML features.PCMU

With the following configuration 2 parties will be connected but whenever
you press *7 it just does not work and FS splits the message. Very weird
that this works if I put the extension into default.xml.

Any ideas what is wrong here?

My config:

I tried to create my own context and call it pstn.
So when FS gets a call from proxy it goes to pstn context, since I have this
line in sip_proxies/internal.xml


in dialplan directory I have pstn.xml
<include>
  <context name="pstn">
   <extension name="pstn_enum">
    <condition field="destination_number" expression="^(.*)$">
                <action application="answer"/>
                <action application="bind_meta_app" data="7 ab s
/usr/local/freeswitch/conf/dialplan/execute_extension:roar XML features"/>
                <action application="set" data="hangup_after_bridge=true"/>
                <action application="bridge"
data="sofia/external/${destination_number}@207.88.122.16"/>
    </condition>
  </extension>
</context>
</include>

In features.xml (It can be anything there since FS never reaches this code
and dies before executing it):
<extension name="roar">
     <condition field="destination_number" expression="^roar$">
        <action application="set" data="continue_on_fail=true"/>
        <action application="log" data="INFO transfer context" />
        <action application="read" data="1 1 /root/1.gsm attxfer_callthis
1000 #"/>
        <action application="log" data="INFO blah ${attxfer_callthis}"/>
        <action application="set" data="origination_cancel_key=#"/>
        <action application="log" data="INFO after read"/>
     </condition>
</extension>

 
-- 
View this message in context: http://www.nabble.com/bind_meta_app-problem-with-custome-extension-tp25963971p25963971.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From ineyaa at gmail.com  Mon Oct 19 13:51:07 2009
From: ineyaa at gmail.com (ineya ineya)
Date: Mon, 19 Oct 2009 22:51:07 +0200
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <5B924C57-B658-44C0-A1CE-00A3ECB5AA07@freeswitch.org>
References: <7b8592a0910190516rafa4398q62c7a99ef703540f@mail.gmail.com>
	<5B924C57-B658-44C0-A1CE-00A3ECB5AA07@freeswitch.org>
Message-ID: <7b8592a0910191351g14cbb47dj3f19362ff6197157@mail.gmail.com>

Here is internal.xml
http://pastebin.freeswitch.org/10766

and internal-ipv6.xml
http://pastebin.freeswitch.org/10767



On Mon, Oct 19, 2009 at 5:13 PM, Brian West <brian at freeswitch.org> wrote:

> I think you just have it misconfigured because if ipv6 was broken I would
> have William and Jason jumping on me.
> Can you post your xml configs for the profile please?
>
> /b
>
> On Oct 19, 2009, at 7:16 AM, ineya ineya wrote:
>
> Hi,
>
> I installed the phonenix release, and changed IPv4, IPv6 addresses and SIP
> port. Directory and dialplan should be default ones.
> My phones are able to register when using just IPv4, and I can call from
> one to another.
>
> When I try the same thing with IPv6, the call to other phone fails with [
> USER_NOT_REGISTERED], however I'm able to call 9999 and hear MOH.
> Here you can find the log: http://pastebin.freeswitch.org/10757
> Freeswitch is running on 10.80.62.40/2000:2::1
> phones are 2000:2::1001 and 2000:2::1002
>
> Am I doing something wrong? Maybe I forgot to configure something, but I
> can't see what it is.
>
> Any hints are appreciated.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Mon Oct 19 14:24:55 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 19 Oct 2009 16:24:55 -0500
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <7b8592a0910191351g14cbb47dj3f19362ff6197157@mail.gmail.com>
References: <7b8592a0910190516rafa4398q62c7a99ef703540f@mail.gmail.com>
	<5B924C57-B658-44C0-A1CE-00A3ECB5AA07@freeswitch.org>
	<7b8592a0910191351g14cbb47dj3f19362ff6197157@mail.gmail.com>
Message-ID: <18202C58-EF07-43F6-9649-A40134B30240@freeswitch.org>

And you're sure you have that IP on your interface?

William or Jason care to let me know if latest SVN IPv6 works for you?

/b

On Oct 19, 2009, at 3:51 PM, ineya ineya wrote:

> Here is internal.xml
> http://pastebin.freeswitch.org/10766
>
> and internal-ipv6.xml
> http://pastebin.freeswitch.org/10767
>
>
>
> On Mon, Oct 19, 2009 at 5:13 PM, Brian West <brian at freeswitch.org>  
> wrote:
> I think you just have it misconfigured because if ipv6 was broken I  
> would have William and Jason jumping on me.
>
> Can you post your xml configs for the profile please?
>
> /b

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From quentusrex at gmail.com  Mon Oct 19 14:28:39 2009
From: quentusrex at gmail.com (William King)
Date: Mon, 19 Oct 2009 14:28:39 -0700
Subject: [Freeswitch-users] Status of ubuntu/debian packages.
In-Reply-To: <90823c940910191332r719970f0m2adce06ad40738d1@mail.gmail.com>
References: <4ABA47FB.2050100@gmail.com>
	<90823c940910191332r719970f0m2adce06ad40738d1@mail.gmail.com>
Message-ID: <4ADCDA07.2000001@gmail.com>

Thanks for the catch. I have fixed the valet_parking as of last night.
And cluechoo will be fixed after this upcoming nightly.

Does anyone know how the other packages deal with locally modified
config files? I like the idea of adding the diffs, but prompting first
per file to see if it should be ignored, diffed, or overwritten.

-William King

Dmitry Bely wrote:
> Can you build FreeSWITCH .deb package with tls transport support?
> Anyway it already depends on openssl package.
>
> Another problem:
>
> 2009-10-20 00:28:50.728557 [CRIT] switch_loadable_module.c:871 Error
> Loading module /opt/freeswitch/mod/mod_cluechoo.so
> **/opt/freeswitch/mod/mod_cluechoo.so: cannot open shared object file:
> No such file or directory**
> 2009-10-20 00:28:50.728604 [CRIT] switch_loadable_module.c:871 Error
> Loading module /opt/freeswitch/mod/mod_valet_parking.so
> **/opt/freeswitch/mod/mod_valet_parking.so: cannot open shared object
> file: No such file or directory**
>
> Modules mod_cluechoo & mod_valet_parking are referenced in the config
> but not present in the package.
>
> - Dmitry Bely
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From msc at freeswitch.org  Mon Oct 19 14:35:10 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 19 Oct 2009 14:35:10 -0700
Subject: [Freeswitch-users] Qustion about INFO messages after
	Connect/Answer
In-Reply-To: <191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>
References: <4AD859A4.5000101@ewetel.de>
	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>
	<4ADC7C09.6070503@ewetel.de>
	<191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>
	<4ADC85C0.7060908@ewetel.de> <4ADC8B6E.7070404@ewetel.de>
	<191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>
Message-ID: <87f2f3b90910191435s7528e256r85df5f12fc634f24@mail.gmail.com>

On Mon, Oct 19, 2009 at 1:07 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> please update and test trunk
>
> 1) I changed the core to remove the excess data by default in your scenario
> 2) I added variables you can use to control it origination_callee_id_name
> origination_callee_id_number which belong in {} in the dial string eg
> {origination_callee_id_number=1234}openzap/1/a/1234
>
>
> After you test, please confirm the behavior and then we'll update the wiki
on these two new chan vars.
-MC
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From mcampbellsmith at gmail.com  Mon Oct 19 14:40:26 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 20 Oct 2009 08:40:26 +1100
Subject: [Freeswitch-users] validating dtmf digits received
In-Reply-To: <81433357-3203-4208-98C1-FF3FE2AC2D1E@jerris.com>
References: <33c87fa30910182153t23e52fa0p2b967b3843eb1175@mail.gmail.com>
	<81433357-3203-4208-98C1-FF3FE2AC2D1E@jerris.com>
Message-ID: <33c87fa30910191440ib033cd6l64bded22394915cc@mail.gmail.com>

Thanks Mike,

I have a lateish trunk and inline seems to work okay.

Does the inline statement below set variable ${code} to be used
directly or does it require transfer also?  ie is digits_dialed
available for use right after a read statement (<action
application="read" data="1 10
ivr/ivr-please_enter_pin_followed_by_pound.wav res 10000 9"/> in my
case) or is it not 'set' until after the transfer?

        <action inline="true" application="set" data="code=${digits_dialed}

Thanks!

On Tue, Oct 20, 2009 at 12:32 AM, Michael Jerris <mike at jerris.com> wrote:
> inline is new, it won't work unless your using recent trunk. ?That
> being said, read is not being run inline, so the set is actually being
> run before digits_dialed is set. ?You will most likely need to use
> transfer in this situation.
>
> Mike
>
> On Oct 19, 2009, at 12:53 AM, Mark Campbell-Smith wrote:
>
>> Hi!
>>
>> I simply want to validate the dtmf digits I read from a user. ? ?From
>> the wiki, it appears I need to use inline="true" when setting the
>> variable so it can be used directly within the same extension.
>>
>> What have I done wrong below? ? I have tried many different
>> alternatives, but the second condition field, which is meant to match
>> the dtmf digits received (in this case 0000) is never matched, and the
>> anti-action is called instead.
>>
>> :
>> ? ? ?<some code here>
>> ? ? ?<action application="read" data="1 10
>> ivr/ivr-please_enter_pin_followed_by_pound.wav res 10000 9"/>
>> ? ? ? <action application="phrase" data="spell,${res}"/>
>> ? ? ? <action inline="true" application="set" data="code=$
>> {digits_dialed}"/>
>> ? ? ? <!-- <action inline="true" application="set" data="code=$
>> {res}"/> -->
>> ?</condition>
>> ?<condition field="digits_dialed" expression="^0000$">
>> ?<!-- <condition field="${code}" expression="^0000$"> -->
>> ?<!-- <condition field="${res}" expression="^0000$"> -->
>> ? ? ? ?<some code here>
>> ? ? ? <anti-action application="hangup"/>
>>
>> Thanks!
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From rupa at rupa.com  Mon Oct 19 14:40:30 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 19 Oct 2009 16:40:30 -0500
Subject: [Freeswitch-users] Status of ubuntu/debian packages.
In-Reply-To: <4ADCDA07.2000001@gmail.com>
References: <4ABA47FB.2050100@gmail.com>
	<90823c940910191332r719970f0m2adce06ad40738d1@mail.gmail.com>
	<4ADCDA07.2000001@gmail.com>
Message-ID: <b39273530910191440s6efd081bie3d8b3138dbae55e@mail.gmail.com>

I was under the impression that if you mark the files as conffiles you
get the default behavior which prompts for diff/overwrite/edit
options.  If you want to get real fancy, look into using ucf to do
three-way diffs between installed default, user modifications and new
default.

Finally, I have checked the current package state, but please put all
the confs in a separate package and make it a suggests not a depends.
I absolutely hate how debian packages keep creating files that I've
removed (ex: in our case would be eg: 1000.xml).  Since conf is so
small,  maybe two.  one that puts them in /etc for people that want
quick out of the box setup and one that puts 'em in
/usr/share/doc/freeswitch/examples for people that want to use the
current defaults as a sample rather than as the basis for their
config.

On Mon, Oct 19, 2009 at 4:28 PM, William King <quentusrex at gmail.com> wrote:
> Thanks for the catch. I have fixed the valet_parking as of last night.
> And cluechoo will be fixed after this upcoming nightly.
>
> Does anyone know how the other packages deal with locally modified
> config files? I like the idea of adding the diffs, but prompting first
> per file to see if it should be ignored, diffed, or overwritten.
>
> -William King
>
> Dmitry Bely wrote:
>> Can you build FreeSWITCH .deb package with tls transport support?
>> Anyway it already depends on openssl package.
>>
>> Another problem:
>>
>> 2009-10-20 00:28:50.728557 [CRIT] switch_loadable_module.c:871 Error
>> Loading module /opt/freeswitch/mod/mod_cluechoo.so
>> **/opt/freeswitch/mod/mod_cluechoo.so: cannot open shared object file:
>> No such file or directory**
>> 2009-10-20 00:28:50.728604 [CRIT] switch_loadable_module.c:871 Error
>> Loading module /opt/freeswitch/mod/mod_valet_parking.so
>> **/opt/freeswitch/mod/mod_valet_parking.so: cannot open shared object
>> file: No such file or directory**
>>
>> Modules mod_cluechoo & mod_valet_parking are referenced in the config
>> but not present in the package.
>>
>> - Dmitry Bely
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa


From hads at nice.net.nz  Mon Oct 19 14:43:59 2009
From: hads at nice.net.nz (Hadley Rich)
Date: Tue, 20 Oct 2009 10:43:59 +1300
Subject: [Freeswitch-users] Status of ubuntu/debian packages.
In-Reply-To: <4ADCDA07.2000001@gmail.com>
References: <4ABA47FB.2050100@gmail.com>
	<90823c940910191332r719970f0m2adce06ad40738d1@mail.gmail.com>
	<4ADCDA07.2000001@gmail.com>
Message-ID: <1255988639.3830.1.camel@sodium>

On Mon, 2009-10-19 at 14:28 -0700, William King wrote:
> Does anyone know how the other packages deal with locally modified
> config files? I like the idea of adding the diffs, but prompting first
> per file to see if it should be ignored, diffed, or overwritten.

dpkg handles config files;

http://www.debian.org/doc/debian-policy/ap-pkg-conffiles.html

hads

-- 
http://nicegear.co.nz
New Zealand's Open Source Hardware Supplier



From mike at jerris.com  Mon Oct 19 15:55:54 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Oct 2009 18:55:54 -0400
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <00E26046-328F-440A-81F4-671F197F114C@gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<6701786D-A8AF-444E-A5A7-29E69F6E055C@gmail.com>
	<B69597BF-DA4A-440B-8CB0-08F52F33DD17@jerris.com>
	<86a32abc0910171339o6a300932v66b7f00419a0cb69@mail.gmail.com>
	<86a32abc0910171513n35d7d0echff940a16685ae1a4@mail.gmail.com>
	<86a32abc0910171521n403094bfvf27aa7c2a1c1f798@mail.gmail.com>
	<86a32abc0910171535l606eb9c6id54f7b3b5ae1b670@mail.gmail.com>
	<86a32abc0910171545l17f3f539ldba432db550c3eaa@mail.gmail.com>
	<86a32abc0910171601u6ab06e42xcf303301c85c57b1@mail.gmail.com>
	<86a32abc0910171605y73ecb75cqfb13923ad19adc56@mail.gmail.com>
	<86a32abc0910171711je1e57bduf2bb672472365ce1@mail.gmail.com>
	<00E26046-328F-440A-81F4-671F197F114C@gmail.com>
Message-ID: <7A73AEA0-9138-4E02-897A-85159405A4EE@jerris.com>

we do have a license for this, people didn't seem to like it last time  
we looked at it, I can't recall why.

On Oct 19, 2009, at 4:24 PM, Roberto Martins wrote:

> what about http://www.atlassian.com/software/confluence/ they give
> free licenses to open source project, and FS is using JIRA.
>
> roberto
>


From mike at jerris.com  Mon Oct 19 16:00:13 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Oct 2009 19:00:13 -0400
Subject: [Freeswitch-users] validating dtmf digits received
In-Reply-To: <33c87fa30910191440ib033cd6l64bded22394915cc@mail.gmail.com>
References: <33c87fa30910182153t23e52fa0p2b967b3843eb1175@mail.gmail.com>
	<81433357-3203-4208-98C1-FF3FE2AC2D1E@jerris.com>
	<33c87fa30910191440ib033cd6l64bded22394915cc@mail.gmail.com>
Message-ID: <490A9EB1-0CAD-421A-A66C-FBA4C52E9F22@jerris.com>

inline is run when the dialplan in parsed, everything else is run  
later.  So read sets digits dialed after it is finished parsing the  
dialplan, if you transfer to another extensions after the read you can  
then condition on that value.

Mike

On Oct 19, 2009, at 5:40 PM, Mark Campbell-Smith wrote:

> Thanks Mike,
>
> I have a lateish trunk and inline seems to work okay.
>
> Does the inline statement below set variable ${code} to be used
> directly or does it require transfer also?  ie is digits_dialed
> available for use right after a read statement (<action
> application="read" data="1 10
> ivr/ivr-please_enter_pin_followed_by_pound.wav res 10000 9"/> in my
> case) or is it not 'set' until after the transfer?
>
>        <action inline="true" application="set" data="code=$ 
> {digits_dialed}
>
> Thanks!
>
> On Tue, Oct 20, 2009 at 12:32 AM, Michael Jerris <mike at jerris.com>  
> wrote:
>> inline is new, it won't work unless your using recent trunk.  That
>> being said, read is not being run inline, so the set is actually  
>> being
>> run before digits_dialed is set.  You will most likely need to use
>> transfer in this situation.
>>
>> Mike
>>
>> On Oct 19, 2009, at 12:53 AM, Mark Campbell-Smith wrote:
>>
>>> Hi!
>>>
>>> I simply want to validate the dtmf digits I read from a user.     
>>> From
>>> the wiki, it appears I need to use inline="true" when setting the
>>> variable so it can be used directly within the same extension.
>>>
>>> What have I done wrong below?   I have tried many different
>>> alternatives, but the second condition field, which is meant to  
>>> match
>>> the dtmf digits received (in this case 0000) is never matched, and  
>>> the
>>> anti-action is called instead.
>>>
>>> :
>>>      <some code here>
>>>      <action application="read" data="1 10
>>> ivr/ivr-please_enter_pin_followed_by_pound.wav res 10000 9"/>
>>>       <action application="phrase" data="spell,${res}"/>
>>>       <action inline="true" application="set" data="code=$
>>> {digits_dialed}"/>
>>>       <!-- <action inline="true" application="set" data="code=$
>>> {res}"/> -->
>>>  </condition>
>>>  <condition field="digits_dialed" expression="^0000$">
>>>  <!-- <condition field="${code}" expression="^0000$"> -->
>>>  <!-- <condition field="${res}" expression="^0000$"> -->
>>>        <some code here>
>>>       <anti-action application="hangup"/>
>>>
>>> Thanks!
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From diego.viola at gmail.com  Mon Oct 19 17:35:38 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 20 Oct 2009 00:35:38 +0000
Subject: [Freeswitch-users] Excellent FreeSWITCH Article For You To Read
In-Reply-To: <87f2f3b90910182037x5c1d1279ue50f724f0a02f94c@mail.gmail.com>
References: <87f2f3b90910182037x5c1d1279ue50f724f0a02f94c@mail.gmail.com>
Message-ID: <86a32abc0910191735l53a5f146o8c46844d6902ffb8@mail.gmail.com>

Nice story.

I see you guys submitted to Slashdot as well, but they have not accepted it
yet.

So please vote the story on Slashdot as well so they accept it :D.

http://slashdot.org/submission/1095795/Open-Source-Carrier-Grade-Telephone-Switch

Diego

On Mon, Oct 19, 2009 at 3:37 AM, Michael Collins <msc at freeswitch.org> wrote:

> Hello all,
>
> Just a heads up that there is a great article out there for you to read.
> Please visit the main freeswitch.org page here:
> http://www.freeswitch.org/node/211
>
> Click the link and digg the article after you read it. Please spread the
> word! The twittersphere is starting to warm up already. Let's keep the
> momentum going!
>
> Thanks,
> Michael
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Mon Oct 19 17:41:05 2009
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 20 Oct 2009 08:41:05 +0800
Subject: [Freeswitch-users] uuid_send_dtmf fails (was: conference call)
In-Reply-To: <3F5FD88C28BF428CAA935EDE7357B686@abisoft.biz>
References: <064A5201C10D4D20AC76EB93CA5EA4B5@abisoft.biz>
	<BE4EDDA84CB64AA98F18D412B51912FC@abisoft.biz>
	<b39273530910150504l5c7411b9g39b9ed33c592e85f@mail.gmail.com>
	<02C2EFD4CF8D4ACF8EEC0F77C4D75028@abisoft.biz>
	<414DC781-A0F6-418E-9AAB-FA4307634932@jerris.com>
	<14E8C50A8F4D41559AFCCD8838350F4F@abisoft.biz>
	<5F579783-33FE-4520-9DC9-7C880DFB34EF@jerris.com>
	<FFF06EC411464736B44C44B58646A8C6@abisoft.biz>
	<01AAA3C0-54DE-4BE0-B1D4-35B5B9C493DF@freeswitch.org>
	<3F5FD88C28BF428CAA935EDE7357B686@abisoft.biz>
Message-ID: <23f91030910191741u5c6093e4w73e3351cd8769aa6@mail.gmail.com>

not sure about this, but did you try send dtmf to uuid
723f3dbb-b87b-4cd4-98fc-698eed7f2bdb other than cced4b9a-b6de-4be1-8c12-
>
> 3d18cc6e8454 ?
>


2009/10/20 Nikita Belov <nbelov at abisoft.spb.ru>

> Yeah. I'm using it for starting eavesdrop:
> SendMsg cced4b9a-b6de-4be1-8c12-3d18cc6e8454
> call-command: execute
> execute-app-name: eavesdrop
> execute-app-arg: 723f3dbb-b87b-4cd4-98fc-698eed7f2bdb
>
> But we are discussing how to switch eavesdrop to allow "spy" to speak with
> one of the sides. Do you know any application which can I use for this?
> Mike
> said I should use sendevent command to control eavesdrop, but I still
> confused how can I do it.
>
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> > users-bounces at lists.freeswitch.org] On Behalf Of Brian West
> > Sent: Monday, October 19, 2009 8:59 PM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] uuid_send_dtmf fails (was: conference
> > call)
> >
> > Please review this:
> >
> > http://wiki.freeswitch.org/wiki/Mod_event_socket#sendmsg
> >
> > SendMsg <uuid>
> > call-command: execute
> > execute-app-name: <one of the applications>
> > execute-app-arg: <application data>
> >
> >
> >
> > /b
> >
> > On Oct 19, 2009, at 11:42 AM, Nikita Belov wrote:
> >
> > > And what event name to use for sendevent command? Sorry for
> > > importunity.
> > > sendevent ???
> > > Unique-ID: b9d6a35c-ee0c-4203-8d65-ed816e0a9c19
> > > eavesdrop-command: 1
> > >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Mon Oct 19 17:53:56 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 19 Oct 2009 19:53:56 -0500
Subject: [Freeswitch-users] uuid_send_dtmf fails (was: conference call)
In-Reply-To: <23f91030910191741u5c6093e4w73e3351cd8769aa6@mail.gmail.com>
References: <064A5201C10D4D20AC76EB93CA5EA4B5@abisoft.biz>
	<BE4EDDA84CB64AA98F18D412B51912FC@abisoft.biz>
	<b39273530910150504l5c7411b9g39b9ed33c592e85f@mail.gmail.com>
	<02C2EFD4CF8D4ACF8EEC0F77C4D75028@abisoft.biz>
	<414DC781-A0F6-418E-9AAB-FA4307634932@jerris.com>
	<14E8C50A8F4D41559AFCCD8838350F4F@abisoft.biz>
	<5F579783-33FE-4520-9DC9-7C880DFB34EF@jerris.com>
	<FFF06EC411464736B44C44B58646A8C6@abisoft.biz>
	<01AAA3C0-54DE-4BE0-B1D4-35B5B9C493DF@freeswitch.org>
	<3F5FD88C28BF428CAA935EDE7357B686@abisoft.biz>
	<23f91030910191741u5c6093e4w73e3351cd8769aa6@mail.gmail.com>
Message-ID: <D2161B4F-CFEC-4717-AE64-B447C751C3B6@freeswitch.org>

OK what makes you think it failed?  The fact you don't hear it?

/b

On Oct 19, 2009, at 7:41 PM, Seven Du wrote:

> not sure about this, but did you try send dtmf to uuid   723f3dbb- 
> b87b-4cd4-98fc-
> 698eed7f2bdb other than cced4b9a-b6de-4be1-8c12-
> 3d18cc6e8454 ?

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From brian at freeswitch.org  Mon Oct 19 18:05:09 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 19 Oct 2009 20:05:09 -0500
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <7b8592a0910191351g14cbb47dj3f19362ff6197157@mail.gmail.com>
References: <7b8592a0910190516rafa4398q62c7a99ef703540f@mail.gmail.com>
	<5B924C57-B658-44C0-A1CE-00A3ECB5AA07@freeswitch.org>
	<7b8592a0910191351g14cbb47dj3f19362ff6197157@mail.gmail.com>
Message-ID: <FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>

Bill has confirmed it works fine in latest trunk.  So i'm not sure  
what exactly you're doing wrong can you provide me some debug output  
of what you're doing in your dialplan and the console debug?

/b

On Oct 19, 2009, at 3:51 PM, ineya ineya wrote:

> Here is internal.xml
> http://pastebin.freeswitch.org/10766
>
> and internal-ipv6.xml
> http://pastebin.freeswitch.org/10767

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From djbinter at yahoo.com  Mon Oct 19 19:13:43 2009
From: djbinter at yahoo.com (DJB)
Date: Mon, 19 Oct 2009 19:13:43 -0700 (PDT)
Subject: [Freeswitch-users] CPU Usage Question
Message-ID: <144734.94454.qm@web37508.mail.mud.yahoo.com>

We have about 1500 concurrent calls on the server.  And, the CPU usage is shown below:

02:06:46 AM  CPU   %user   %nice    %sys %iowait    %irq   %soft  %steal   %idle    intr/s
02:06:46 AM  all    0.06    0.00    0.03    0.44    0.00    0.00    0.00   99.47   1028.25
02:06:46 AM    0    0.08    0.03    0.03    0.01    0.00    0.01    0.00   99.84   1007.07
02:06:46 AM    1    0.06    0.00    0.02    0.01    0.00    0.00    0.00   99.91      0.00
02:06:46 AM    2    0.07    0.00    0.08    3.49    0.02    0.00    0.00   96.34     21.18
02:06:46 AM    3    0.05    0.00    0.01    0.01    0.00    0.00    0.00   99.94      0.00
02:06:46 AM    4    0.04    0.00    0.02    0.00    0.00    0.00    0.00   99.93      0.00
02:06:46 AM    5    0.05    0.00    0.02    0.00    0.00    0.00    0.00   99.92      0.00
02:06:46 AM    6    0.05    0.00    0.02    0.02    0.00    0.00    0.00   99.91      0.00
02:06:46 AM    7    0.05    0.00    0.02    0.01    0.00    0.00    0.00   99.93      0.00


1) I am wondering whether it is normal that every time when i checked, only CPU 2 always have the highest %iowait (in this case 3.49%).

2) Shoudn't it spread out and balance them to utilize all CPUs evenly?

Thank you.



      


From gabe at gundy.org  Mon Oct 19 19:17:28 2009
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 19 Oct 2009 20:17:28 -0600
Subject: [Freeswitch-users] Is anyone running Ubuntu 8.04/Hardy?
In-Reply-To: <4ADCA34E.5000700@marketelsystems.com>
References: <4ADCA34E.5000700@marketelsystems.com>
Message-ID: <903da5680910191917m79bf5ecfvdc8001952dbcc339@mail.gmail.com>

On Mon, Oct 19, 2009 at 11:35 AM, Mark Sobkow
<m.sobkow at marketelsystems.com> wrote:
> Everyone I've emailed with on the dev list is running the current
> release of Ubuntu, not 8.04/Hardy.

Well, what issues?

Gabe


From bottleman at icf.org.ru  Mon Oct 19 22:14:13 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Tue, 20 Oct 2009 09:14:13 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910140215y357fede1l37f6cba9725ddd89@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<65d96fc80910130014hc5f5fcet677f87b7fb57559e@mail.gmail.com>
	<65d96fc80910132359i6d53d8ban10f943795481bf7c@mail.gmail.com>
	<Pine.LNX.4.64.0910141134270.11620@srv.icf.org.ru>
	<65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>
	<Pine.LNX.4.64.0910141202500.21535@srv.icf.org.ru>
	<65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>
	<65d96fc80910140215y357fede1l37f6cba9725ddd89@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>

On 2009-10-14 11:15 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>On Wed, Oct 14, 2009 at 10:16 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
TC>
TC>>
TC>>
TC>> 2009/10/14 Georgiewskiy Yuriy <bottleman at icf.org.ru>
TC>>
TC>>> On 2009-10-14 09:58 +0200, Tihomir Culjaga wrote
TC>>> freeswitch-users at lists.fre...:
TC>>>
TC>>> i need trace level 4 from mod_h323 and debug log of entire call, tcpdump
TC>>> may be needed later, i have no way
TC>>> to test it on this time, i do it later.
TC>>>
TC>>>
TC>
TC>Here are the logs:
TC>
TC>call flow is H323_endpoint => FS => SIP_user
TC>
TC>slow start                => http://pastebin.freeswitch.org/10693
TC>slow start w eH245            => http://pastebin.freeswitch.org/10694
TC>fastStart w tunneling true         => http://pastebin.freeswitch.org/10701
TC>fastStart w tunneling false        => http://pastebin.freeswitch.org/10702
TC>fastStart w tunneling true eH245    => http://pastebin.freeswitch.org/10699
TC>fastStart w tunneling false eH245    => http://pastebin.freeswitch.org/10700
TC>
TC>
TC>In all cases, there is no audio on SIP User side... i see the SIP Phone
TC>(terminator) sending RTP to FS but FS is not forwarding it back to the
TC>originator. Also, i see originator (H323) sending RTP to FS but FS doesn't
TC>forward it to the terminator (SIP).
TC>
TC>
TC>
TC>call flow is SIP_user => FS => H323_endpoint is failing .. coredumped....
TC>http://pastebin.freeswitch.org/10703

i fix some bugs now, ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/mod_h323.tar.bz2 this is 
updated version, try it, if you experience no audio try enable rtp proxy in you sip profile.

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From bottleman at icf.org.ru  Mon Oct 19 22:17:07 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Tue, 20 Oct 2009 09:17:07 +0400 (MSD)
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <5F65174C-2CE6-4007-B288-C81227776281@freeswitch.org>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>
	<65d96fc80910122327u54505b44r4ff53e89293bcf9a@mail.gmail.com>
	<5F65174C-2CE6-4007-B288-C81227776281@freeswitch.org>
Message-ID: <Pine.LNX.4.64.0910200915170.2416@srv.icf.org.ru>

On 2009-10-12 23:31 -0700, Brian West wrote freeswitch-users at lists.freeswit...:

I think now is time to start host it in our svn, there is updated version with some 
fixes and improvements - ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/mod_h323.tar.bz2.

BW>I wouldn't call it donating per se... Its just giving it a place to  
BW>live with easy access for end users without having to do anything  
BW>extra go get it!  ;)
BW>
BW>/b
BW>
BW>On Oct 12, 2009, at 11:27 PM, Tihomir Culjaga wrote:
BW>
BW>> this will be perfect ... but it is up to Yuriy if he is willing to  
BW>> donate his work...
BW>>
BW>> T.
BW>
BW>
BW>_______________________________________________
BW>FreeSWITCH-users mailing list
BW>FreeSWITCH-users at lists.freeswitch.org
BW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
BW>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
BW>http://www.freeswitch.org
BW>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From brian at freeswitch.org  Mon Oct 19 22:28:26 2009
From: brian at freeswitch.org (=?utf-8?B?YnJpYW5AZnJlZXN3aXRjaC5vcmc=?=)
Date: Mon, 19 Oct 2009 22:28:26 -0700 (PDT)
Subject: [Freeswitch-users] =?utf-8?q?CPU_Usage_Question?=
Message-ID: <4add4a7a.1408c00a.4283.ffffc11f@mx.google.com>

It's because Sofia used one thread per profile so depending on load you will see one CPU used more.

/b 

Sent from my HTC on the Now Network from Sprint!

----- Reply message -----
From: "DJB" <djbinter at yahoo.com>
Date: Mon, Oct 19, 2009 9:13 PM
Subject: [Freeswitch-users] CPU Usage Question
To: "FREESWITCH-USERS MAILING LIST" <freeswitch-users at lists.freeswitch.org>

We have about 1500 concurrent calls on the server.  And, the CPU usage is shown below:

02:06:46 AM  CPU   %user   %nice    %sys %iowait    %irq   %soft  %steal   %idle    intr/s
02:06:46 AM  all    0.06    0.00    0.03    0.44    0.00    0.00    0.00   99.47   1028.25
02:06:46 AM    0    0.08    0.03    0.03    0.01    0.00    0.01    0.00   99.84   1007.07
02:06:46 AM    1    0.06    0.00    0.02    0.01    0.00    0.00    0.00   99.91      0.00
02:06:46 AM    2    0.07    0.00    0.08    3.49    0.02    0.00    0.00   96.34     21.18
02:06:46 AM    3    0.05    0.00    0.01    0.01    0.00    0.00    0.00   99.94      0.00
02:06:46 AM    4    0.04    0.00    0.02    0.00    0.00    0.00    0.00   99.93      0.00
02:06:46 AM    5    0.05    0.00    0.02    0.00    0.00    0.00    0.00   99.92      0.00
02:06:46 AM    6    0.05    0.00    0.02    0.02    0.00    0.00    0.00   99.91      0.00
02:06:46 AM    7    0.05    0.00    0.02    0.01    0.00    0.00    0.00   99.93      0.00


1) I am wondering whether it is normal that every time when i checked, only CPU 2 always have the highest %iowait (in this case 3.49%).

2) Shoudn't it spread out and balance them to utilize all CPUs evenly?

Thank you.



      

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From ineyaa at gmail.com  Mon Oct 19 22:53:11 2009
From: ineyaa at gmail.com (ineya ineya)
Date: Tue, 20 Oct 2009 07:53:11 +0200
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
References: <7b8592a0910190516rafa4398q62c7a99ef703540f@mail.gmail.com>
	<5B924C57-B658-44C0-A1CE-00A3ECB5AA07@freeswitch.org>
	<7b8592a0910191351g14cbb47dj3f19362ff6197157@mail.gmail.com>
	<FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
Message-ID: <7b8592a0910192253l1313d82j51c933cb5f55ad41@mail.gmail.com>

IP should be OK:

[root at franta /opt/freeswitch]# ifconfig eth1
eth1????? Link encap:Ethernet? HWaddr 00:4f:4e:62:ad:83
????????? inet addr:10.80.62.40? Bcast:10.80.62.255? Mask:255.255.255.0
????????? inet6 addr: 2000:2::1/32 Scope:Global
????????? inet6 addr: fe80::24f:4eff:fe62:ad83/64 Scope:Link
????????? UP BROADCAST RUNNING MULTICAST? MTU:1500? Metric:1
????????? RX packets:4310768 errors:0 dropped:0 overruns:0 frame:0
????????? TX packets:2537585 errors:0 dropped:0 overruns:0 carrier:0
????????? collisions:114862 txqueuelen:1000
????????? RX bytes:1067582578 (1018.1 MiB)? TX bytes:616134369 (587.5 MiB)
????????? Interrupt:19 Base address:0x3400

Console debug: http://pastebin.freeswitch.org/10768

When I do "sofia status profile internal-ipv6", I can see both phones
as registered:
Call-ID:? ? ? ? 9cb5faf2a052867b
User:? ? ? ? ? ?1007 at franta.openstage.net
Contact:? ? ? ? 1007 <sip:1007@[2000:2::1002]:5062;transport=udp>

but when it comes to connecting the call:

EXECUTE sofia/internal-ipv6/1006 at franta.openstage.net
bridge(user/1007 at franta.openstage.net)
2009-10-20 07:36:12.749270 [DEBUG] switch_ivr_originate.c:1027
variable string 0 = [presence_id=1007 at franta.openstage.net]
2009-10-20 07:36:12.749270 [ERR] switch_ivr_originate.c:1510 Cannot
create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]

@Bill: Can you share your IPv6 test config?

On Tue, Oct 20, 2009 at 3:05 AM, Brian West <brian at freeswitch.org> wrote:
>
> Bill has confirmed it works fine in latest trunk. ?So i'm not sure what exactly you're doing wrong can you provide me some debug output of what you're doing in your dialplan and the console debug?
> /b
> On Oct 19, 2009, at 3:51 PM, ineya ineya wrote:
>
> Here is internal.xml
> http://pastebin.freeswitch.org/10766
>
> and internal-ipv6.xml
> http://pastebin.freeswitch.org/10767
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From lakindia89 at gmail.com  Tue Oct 20 00:18:47 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Tue, 20 Oct 2009 12:48:47 +0530
Subject: [Freeswitch-users] Digium TE220 wiki
Message-ID: <7d79b3930910200018s25184c01k838bff05ad308ffb@mail.gmail.com>

Hi all,
    I've created a wiki page, which contains the example configuration for
making Digium TE220 to work.
    I request you people to check this, and give feedbacks.

http://wiki.freeswitch.org/wiki/Configuration_OpenZap-DigiumTE220-Example
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From helmut.kuper at ewetel.de  Tue Oct 20 00:25:14 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 20 Oct 2009 09:25:14 +0200
Subject: [Freeswitch-users] Qustion about INFO messages
	after	Connect/Answer
In-Reply-To: <191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>
References: <4AD859A4.5000101@ewetel.de>	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>	<4ADC7C09.6070503@ewetel.de>	<191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>	<4ADC85C0.7060908@ewetel.de>
	<4ADC8B6E.7070404@ewetel.de>
	<191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>
Message-ID: <4ADD65DA.4050809@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello Anthony,


On 19.10.2009 22:07, Anthony Minessale wrote:
> please update and test trunk
> 
> 1) I changed the core to remove the excess data by default in your scenario
> 2) I added variables you can use to control it
> origination_callee_id_name origination_callee_id_number which belong in
> {} in the dial string eg {origination_callee_id_number=1234}openzap/1/a/1234

updated and testet for SIP calls. "origination_callee_id_number=1234"
works within the dial string and via using "export" application but not
via "set" application. Didn't test it for openzap, yet but I guess it
will work, too.

What left is that "unknown" thing on callee's display (SIP phones only I
guess) ... I would suggest to make sending "unknown" INFO message
optional, but to be honest, I have no idea for what it was invented, so
maybe my suggestion is nonsens.


regards
Helmut
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=7N8O
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From helmut.kuper at ewetel.de  Tue Oct 20 00:35:56 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 20 Oct 2009 09:35:56 +0200
Subject: [Freeswitch-users] Stereo Support
Message-ID: <4ADD685C.1000506@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

I wonder if there is real stereo support planned for FreeSWITCH in terms
of streaming music/video-audio to desktop? I ask, because I heared that
celt codec is supporting stereo.

regards
helmut
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)

iD8DBQFK3Whc4tZeNddg3dwRAkdEAKCLBSV4H8q5JN4NZ39fgSZPxr5PCgCgmpTq
PAu8qozLP5FaVKIwfCXVqjU=
=Zseo
-----END PGP SIGNATURE-----


From helmut.kuper at ewetel.de  Tue Oct 20 00:39:31 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 20 Oct 2009 09:39:31 +0200
Subject: [Freeswitch-users] Is anyone running Ubuntu 8.04/Hardy?
In-Reply-To: <4ADCA34E.5000700@marketelsystems.com>
References: <4ADCA34E.5000700@marketelsystems.com>
Message-ID: <4ADD6933.4040504@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Mark,

I confirm that it works fine. I use it for two years now and there were
no problems compiling or running FS on ubuntu 8.04 caused by the OS.

This is my system:

lsb_release -a
No LSB modules are available.
Distributor ID: Ubuntu
Description:    Ubuntu 8.04.3 LTS
Release:        8.04
Codename:       hardy

uname -a
Linux ippbx-prod-node0 2.6.24-22-server #1 SMP Mon Nov 24 19:14:19 UTC
2008 i686 GNU/Linux


On 19.10.2009 19:35, Mark Sobkow wrote:
> Everyone I've emailed with on the dev list is running the current 
> release of Ubuntu, not 8.04/Hardy.  We're having some problems with 
> loading mod_sofia on Hardy, so unless I get some progress/help by end of 
> day Wednesday we're going to have to rebuild our development server with 
> the most recent release of Ubuntu on Thursday and Friday.
> 

regards
helmut
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From tculjaga at gmail.com  Tue Oct 20 01:17:54 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 20 Oct 2009 10:17:54 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<65d96fc80910130014hc5f5fcet677f87b7fb57559e@mail.gmail.com>
	<65d96fc80910132359i6d53d8ban10f943795481bf7c@mail.gmail.com>
	<Pine.LNX.4.64.0910141134270.11620@srv.icf.org.ru>
	<65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>
	<Pine.LNX.4.64.0910141202500.21535@srv.icf.org.ru>
	<65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>
	<65d96fc80910140215y357fede1l37f6cba9725ddd89@mail.gmail.com>
	<Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>
Message-ID: <65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>

> TC>
> TC>call flow is SIP_user => FS => H323_endpoint is failing ..
> coredumped....
> TC>http://pastebin.freeswitch.org/10703
>
> i fix some bugs now,
> ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/mod_h323.tar.bz2 this
> is
> updated version, try it, if you experience no audio try enable rtp proxy in
> you sip profile.
>
>
Hi,  there are several issues... lets start with top 4 :)



1. I'm still stuck with no audio:

I have this parameter in the sip profile set: <param
name="inbound-proxy-media" value="true"/>
...tried with both with slow start and fast start... any idea ?

pls check: http://pastebin.freeswitch.org/10771





2. outgoing calls still failing in coredumps: what is your dialplan ? ...
how do you call bridge application?

2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3252 Found capability:
UserInput/PointDevice <14>
2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3248 FindCapability: 15
2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external thread
0xb6eb60a0 for id 3048876944
2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:65 Destroyed external thread
0xb6ebafa8 for id 3048876944
2009-10-20 10:08:18.426608 [DEBUG] tlibthrd.cxx:406 Destroyed thread
0xb6ebafa8 PExternalThread:0xb5ba2b90(id = b5ba2b90)
2009-10-20 10:08:18.426608 [DEBUG] h323caps.cxx:3252 Found capability:
UserInput/Modal <15>
2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0,
expiries=0
2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0,
expiries=0
2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external thread
0xb6eba910 for id 3048876944
2009-10-20 10:08:18.426608 [DEBUG] h4601.cxx:1725 Endpoint Attached
Segmentation fault (core dumped)
tculjaga at subZero:~/freeswitch-trunk$

pls check: http://pastebin.freeswitch.org/10769




3.  when you hangup from SIP side, the call is not released end-to-end (the
H323 endpoint doesn't get any releaseComplete message)

2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=2,
expiries=3
2009-10-20 10:10:51.264527 [DEBUG] h323neg.cxx:432 Received
MasterSlaveDeterminationAck: state=Incoming
2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=1,
expiries=3
2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=2,
expiries=4
2009-10-20 10:10:51.264527 [DEBUG] h323.cxx:4138
InternalEstablishedConnectionCheck: connectionState=EstablishedConnection
fastStartState=FastStartAcknowledged
2009-10-20 10:10:51.264527 [DEBUG] h323caps.cxx:3264 FindCapability: "T.120"
2009-10-20 10:10:51.264527 [DEBUG] h323.cxx:4138
InternalEstablishedConnectionCheck: connectionState=EstablishedConnection
fastStartState=FastStartAcknowledged
2009-10-20 10:10:51.264527 [DEBUG] tlibthrd.cxx:1023
PThread::PXBlockOnIO(45,0)
2009-10-20 10:10:54.405479 [NOTICE] sofia.c:328 Hangup sofia/internal/
sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [CS_CONSUME_MEDIA]
[NORMAL_CLEARING]
2009-10-20 10:10:54.405479 [DEBUG] switch_channel.c:1726 Send signal
sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [KILL]
2009-10-20 10:10:54.405479 [DEBUG] switch_core_session.c:932 Send signal
sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [BREAK]
2009-10-20 10:10:54.405479 [DEBUG] switch_core_state_machine.c:437 thread
mismatch skipping state handler.
2009-10-20 10:10:54.405479 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) Running State
Change CS_HANGUP
2009-10-20 10:10:54.406530 [DEBUG] switch_core_state_machine.c:464
(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State HANGUP
2009-10-20 10:10:54.406530 [DEBUG] mod_sofia.c:338 Channel sofia/internal/
sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> hanging up, cause:
NORMAL_CLEARING
2009-10-20 10:10:54.406530 [DEBUG] switch_core_state_machine.c:46
sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> Standard HANGUP,
cause: NORMAL_CLEARING


pls check: http://pastebin.freeswitch.org/10771




4. sometimes when i shutdown FS i get coredimps - from my experience it
looks like you don't wait for a FS thread to finish when you exit...

2009-10-20 10:05:59.493306 [CONSOLE] switch_event.c:508 Stopping queue
thread 2
2009-10-20 10:05:59.493339 [CONSOLE] switch_core.c:1693 Finalizing Shutdown.
2009-10-20 10:05:59.493379 [CONSOLE] switch_log.c:310 Logger Ended.
2009-10-20 10:05:59.494472 [CONSOLE] switch_core_memory.c:567 Stopping
memory pool queue.
Segmentation fault (core dumped)
tculjaga at subZero:~/freeswitch-trunk$
tculjaga at subZero:~/freeswitch-trunk$


Please advice your FS/mod_h323.conf.xml settings...


Tihomir.




> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129-RIPE
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From maciej.aniserowicz at gmail.com  Tue Oct 20 03:30:59 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Tue, 20 Oct 2009 03:30:59 -0700 (PDT)
Subject: [Freeswitch-users] "Can not record session. Media not enabled on
 channel."
Message-ID: <1256034659005-3857858.post@n2.nabble.com>


Hello,
I am using the same set of extensions for testing the system during
development, they include XLite, Cisco sip phone and several extensions that
just play some audio file.
Sometimes, very rarely, this message "Can not record session.  Media not
enabled on channel." appears on FS console.
Like I wrote before, I don't change any codec settings and always use the
same set of devices/emulators.
What can cause this message?

Thanks,
Maciej Aniserowicz
-- 
View this message in context: http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3857858.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From nbelov at abisoft.spb.ru  Tue Oct 20 04:49:05 2009
From: nbelov at abisoft.spb.ru (Nikita Belov)
Date: Tue, 20 Oct 2009 15:49:05 +0400
Subject: [Freeswitch-users] uuid_send_dtmf fails (was: conference call)
In-Reply-To: <D2161B4F-CFEC-4717-AE64-B447C751C3B6@freeswitch.org>
References: <064A5201C10D4D20AC76EB93CA5EA4B5@abisoft.biz><BE4EDDA84CB64AA98F18D412B51912FC@abisoft.biz><b39273530910150504l5c7411b9g39b9ed33c592e85f@mail.gmail.com><02C2EFD4CF8D4ACF8EEC0F77C4D75028@abisoft.biz><414DC781-A0F6-418E-9AAB-FA4307634932@jerris.com><14E8C50A8F4D41559AFCCD8838350F4F@abisoft.biz><5F579783-33FE-4520-9DC9-7C880DFB34EF@jerris.com><FFF06EC411464736B44C44B58646A8C6@abisoft.biz><01AAA3C0-54DE-4BE0-B1D4-35B5B9C493DF@freeswitch.org><3F5FD88C28BF428CAA935EDE7357B686@abisoft.biz><23f91030910191741u5c6093e4w73e3351cd8769aa6@mail.gmail.com>
	<D2161B4F-CFEC-4717-AE64-B447C751C3B6@freeswitch.org>
Message-ID: <D464AFA2661C49C68AF378F8A199998E@abisoft.biz>

Yeah. A call to B and C eavesdrops call. I send dtmf for C to talk with B,
but B can't hear C. Here, what I had done in details:

 

[root at centos4-4-vm ~]# telnet localhost 8021 Trying 127.0.0.1...

Connected to localhost.localdomain (127.0.0.1).

Escape character is '^]'.

Content-Type: auth/request

 

auth ClueCon

 

Content-Type: command/reply

Reply-Text: +OK accepted

 

api originate user/1007 at master.agent.starpoundtech.net &park()

 

Content-Type: api/response

Content-Length: 41

 

+OK bba3b45a-4cc1-48af-a15d-1052d5f11371

 

SendMsg bba3b45a-4cc1-48af-a15d-1052d5f11371

call-command: execute

execute-app-name: eavesdrop

execute-app-arg: cd99f999-9b47-457e-8439-1d366e015b8c

 

Content-Type: command/reply

Reply-Text: +OK

 

Here I had started to hear A and B. Here what I saw in FS log:

2009-10-18 03:22:47 [DEBUG] switch_core_session.c:706

switch_core_session_queue_private_event() Send signal
sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes [BREAK]

2009-10-18 03:22:47 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event()
sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes Command Execute

eavesdrop(cd99f999-9b47-457e-8439-1d366e015b8c)

2009-10-18 03:22:47 [DEBUG] switch_core_media_bug.c:297

switch_core_media_bug_add() Attaching BUG to
sofia/internal/1006 at master.agent.starpoundtech.net

2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234

switch_core_session_read_frame()

sofia/internal/1006 at master.agent.starpoundtech.net receive message
[TRANSCODING_NECESSARY]

2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234

switch_core_session_read_frame()

sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes receive message
[TRANSCODING_NECESSARY]

 

Then I run command:

api uuid_send_dtmf bba3b45a-4cc1-48af-a15d-1052d5f11371 1

 

Content-Type: api/response

Content-Length: 14

 

-ERR no reply

 

Log:

2009-10-18 03:24:01 [DEBUG] switch_core_io.c:1190

switch_core_session_send_dtmf_string()

sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes send dtmf

digit=1 ms=250 samples=2000

2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1282 do_2833() Send start packet
for [1] ts=2241760 dur=160/160/2000 seq=21346

2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=320/320/2000 seq=21347

2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=480/480/2000 seq=21348

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=640/640/2000 seq=21349

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=800/800/2000 seq=21350

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=960/960/2000 seq=21351

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=1120/1120/2000 seq=21352

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=1280/1280/2000 seq=21353

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=1440/1440/2000 seq=21354

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=1600/1600/2000 seq=21355

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=1760/1760/2000 seq=21356

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=1920/1920/2000 seq=21357

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end packet for
[1] ts=2241760 dur=2080/2080/2000 seq=21358

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end packet for
[1] ts=2241760 dur=2080/2080/2000 seq=21359

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end packet for
[1] ts=2241760 dur=2080/2080/2000 seq=21360

 

But both A and B couldn't hear me. 

Btw, after I had send dtmf "1" manually from my phone. B started to hear me.

There was this record in log:

2009-10-18 03:47:55 [DEBUG] switch_rtp.c:1767 switch_rtp_dequeue_dtmf() RTP
RECV DTMF 1:2240

 

 

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Tuesday, October 20, 2009 4:54 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] uuid_send_dtmf fails (was: conference call)

 

OK what makes you think it failed?  The fact you don't hear it?

 

/b

 

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From durk.debeer at isp.solcon.nl  Tue Oct 20 04:55:46 2009
From: durk.debeer at isp.solcon.nl (Durk de Beer)
Date: Tue, 20 Oct 2009 13:55:46 +0200
Subject: [Freeswitch-users] How to retain the B-number in an Call forward
Message-ID: <mailman.1.1256039782.2492.freeswitch-users@lists.freeswitch.org>


 Hello,


 I us a call forward on freeswitch to forward calls to my mobile phone. If now Freeswitch forwards a call, the number information presented to my mobile phone is that of the original caller, because that is what there is appearing in its display. What I want
 to see is that Freeswitch is calling me so I would know it is a business call and not an private call. Now I have no means to distinguishes between them.



 Kind
 regards,



 Durk de Beer


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From ahmedmunir007 at gmail.com  Tue Oct 20 05:33:49 2009
From: ahmedmunir007 at gmail.com (Ahmed Munir)
Date: Tue, 20 Oct 2009 17:33:49 +0500
Subject: [Freeswitch-users] mod_curl + mod_nibblebill
Message-ID: <a60fed5e0910200533h57d3d123maf8fa651801c31b4@mail.gmail.com>

Hi,

Consider below mob_nibblebill configuration,


<configuration name="nibblebill.conf" description="Nibble Billing">
  <settings>

    <param name="db_username" value="root"/>
    <param name="db_password" value="password"/>
    <param name="db_dsn" value="MySQL-freeswitch"/>

    <param name="db_table" value="accounts"/>

    <param name="db_column_cash" value="cash"/>

    <param name="db_column_account" value="id"/>

    <param name="global_heartbeat" value="60"/>

    <param name="lowbal_amt" value="5"/>
    <param name="lowbal_action" value="play ding"/>

     <param name="nobal_amt" value="0"/>
    <param name="nobal_action" value="hangup"/>

     <param name="percall_max_amt" value="100"/>
    <param name="percall_action" value="hangup"/>

  </settings>
</configuration>


When i put this configuration in
/usr/local/freeswitch/conf/autoload_configs, everything works perfect as
expected.

But when i call this configuration using mod_xml_curl, freeswitch does not
loads mod_nibblebill configuration from it. I even debuged mod_xml_curl
using console command,

xml_curl debug_on

and reload mod_nibblebill, I get exact same configuration from xml_curl but
mod_nibblebill does not repect it and try to use default parameter values as
mentioned below,


2009-10-20 16:58:23.396710 [ERR] switch_odbc.c:188 STATE: IM002 CODE 0
ERROR: [unixODBC][Driver Manager]Data source name not found, and no default
driver specified

2009-10-20 16:58:23.396738 [CRIT] mod_nibblebill.c:221 Cannot connect to
ODBC driver/database bandwidth.com (user: bandwidth.com / pass dev)!
2009-10-20 16:58:23.396800 [CONSOLE] switch_loadable_module.c:889
Successfully Loaded [mod_nibblebill]

I am also loading various other configs from xml_curl without any problem,
its only mod_nibblebill that is not honoring configuration provided by
xml_curl.

Please advise if i am doing anything wrong.


-- 
Regards,

Ahmed Munir
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From dome at tel.co.th  Tue Oct 20 06:31:17 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Tue, 20 Oct 2009 20:31:17 +0700
Subject: [Freeswitch-users] mod_odbc_query share memory problem
Message-ID: <8ccbff060910200631l44cc7dcfm2aa9511d86798aa1@mail.gmail.com>

Dear Sir,
I'm using mod_odbc_query and mod_nibble_billing for my calling card
solutoin. i found mod_odbc_query cant work with high load  (200 calls
concurrent)
i got error
[STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory;
Error while executing the query
]
in FS console

How to fix it ?

BG
Dome C.


From dwhite at olp.net  Tue Oct 20 06:34:37 2009
From: dwhite at olp.net (Dan White)
Date: Tue, 20 Oct 2009 08:34:37 -0500
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <7b8592a0910192253l1313d82j51c933cb5f55ad41@mail.gmail.com>
References: <7b8592a0910190516rafa4398q62c7a99ef703540f@mail.gmail.com>
	<5B924C57-B658-44C0-A1CE-00A3ECB5AA07@freeswitch.org>
	<7b8592a0910191351g14cbb47dj3f19362ff6197157@mail.gmail.com>
	<FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
	<7b8592a0910192253l1313d82j51c933cb5f55ad41@mail.gmail.com>
Message-ID: <20091020133437.GA4941@dan.olp.net>

On 20/10/09?07:53?+0200, ineya ineya wrote:
>IP should be OK:
>
>[root at franta /opt/freeswitch]# ifconfig eth1
>eth1????? Link encap:Ethernet? HWaddr 00:4f:4e:62:ad:83
>????????? inet addr:10.80.62.40? Bcast:10.80.62.255? Mask:255.255.255.0
>????????? inet6 addr: 2000:2::1/32 Scope:Global

This address caught my eye. Although valid, a /32 address assignment is
atypical. Is that what you intended? Does assigning a /64 address on both
ends make any difference?

-- 
Dan White


From rob4manhere at gmail.com  Tue Oct 20 06:46:43 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Tue, 20 Oct 2009 08:46:43 -0500
Subject: [Freeswitch-users] mod_odbc_query share memory problem
In-Reply-To: <8ccbff060910200631l44cc7dcfm2aa9511d86798aa1@mail.gmail.com>
References: <8ccbff060910200631l44cc7dcfm2aa9511d86798aa1@mail.gmail.com>
Message-ID: <1BFD340C-EC22-434D-B521-8E4D1572A343@gmail.com>

Hi Dome,

Have you tried increasing global_heartbeat to reduce the frequency of  
odbc calls?  What is it currently set to?

Rob

On Oct 20, 2009, at 8:31 AM, Dome Charoenyost wrote:

> Dear Sir,
> I'm using mod_odbc_query and mod_nibble_billing for my calling card
> solutoin. i found mod_odbc_query cant work with high load  (200 calls
> concurrent)
> i got error
> [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory;
> Error while executing the query
> ]
> in FS console
>
> How to fix it ?
>
> BG
> Dome C.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From rob4manhere at gmail.com  Tue Oct 20 07:00:05 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Tue, 20 Oct 2009 09:00:05 -0500
Subject: [Freeswitch-users] How to retain the B-number in an Call forward
In-Reply-To: <4adda65c.4944f10a.7a9b.ffff9408SMTPIN_ADDED@mx.google.com>
References: <4adda65c.4944f10a.7a9b.ffff9408SMTPIN_ADDED@mx.google.com>
Message-ID: <7394C8E4-C8D8-42C2-8ECD-B95119B604E5@gmail.com>

Have you tried setting the effective_caller_id_number before  
bridging?  Such as:

<action application="set" data="effective_caller_id_number=9185551212"/>
Cheers,
Rob

On Oct 20, 2009, at 6:55 AM, Durk de Beer wrote:

> Hello,
> I us a call forward on freeswitch to forward calls to my mobile  
> phone. If now Freeswitch forwards a call, the number information  
> presented to my mobile phone is that of the original caller, because  
> that is what there is appearing in its display. What I want to see  
> is that Freeswitch is calling me so I would know it is a business  
> call and not an private call. Now I have no means to distinguishes  
> between them.
> Kind regards,
> Durk de Beer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From dome at tel.co.th  Tue Oct 20 07:00:22 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Tue, 20 Oct 2009 21:00:22 +0700
Subject: [Freeswitch-users] mod_odbc_query share memory problem
In-Reply-To: <1BFD340C-EC22-434D-B521-8E4D1572A343@gmail.com>
References: <8ccbff060910200631l44cc7dcfm2aa9511d86798aa1@mail.gmail.com>
	<1BFD340C-EC22-434D-B521-8E4D1572A343@gmail.com>
Message-ID: <8ccbff060910200700o5213a8c2mf95acd9603f96ea@mail.gmail.com>

2009/10/20 Rob Forman <rob4manhere at gmail.com>:
> Hi Dome,
>
> Have you tried increasing global_heartbeat to reduce the frequency of
> odbc calls? ?What is it currently set to?

Now 1 min

>
> Rob
>
> On Oct 20, 2009, at 8:31 AM, Dome Charoenyost wrote:
>
>> Dear Sir,
>> I'm using mod_odbc_query and mod_nibble_billing for my calling card
>> solutoin. i found mod_odbc_query cant work with high load ?(200 calls
>> concurrent)
>> i got error
>> [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory;
>> Error while executing the query
>> ]
>> in FS console
>>
>> How to fix it ?
>>
>> BG
>> Dome C.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From leon at scarlet-internet.nl  Tue Oct 20 07:10:30 2009
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Tue, 20 Oct 2009 16:10:30 +0200
Subject: [Freeswitch-users] mod_odbc_query share memory problem
In-Reply-To: <8ccbff060910200631l44cc7dcfm2aa9511d86798aa1@mail.gmail.com>
References: <8ccbff060910200631l44cc7dcfm2aa9511d86798aa1@mail.gmail.com>
Message-ID: <180CE701-D840-47E6-821C-BB79C9F1945A@scarlet-internet.nl>

Hi,

I haven't run into that problem yet, but did you try increasing the  
maximum shared memory in /proc/sys/kernel/shmmax (sysctl  
kernel.shmmax) ?

regards,

Leon



On Oct 20, 2009, at 3:31 PM, Dome Charoenyost wrote:

> Dear Sir,
> I'm using mod_odbc_query and mod_nibble_billing for my calling card
> solutoin. i found mod_odbc_query cant work with high load  (200 calls
> concurrent)
> i got error
> [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory;
> Error while executing the query
> ]
> in FS console
>
> How to fix it ?
>
> BG
> Dome C.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From rob4manhere at gmail.com  Tue Oct 20 07:11:11 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Tue, 20 Oct 2009 09:11:11 -0500
Subject: [Freeswitch-users] mod_odbc_query share memory problem
In-Reply-To: <8ccbff060910200700o5213a8c2mf95acd9603f96ea@mail.gmail.com>
References: <8ccbff060910200631l44cc7dcfm2aa9511d86798aa1@mail.gmail.com>
	<1BFD340C-EC22-434D-B521-8E4D1572A343@gmail.com>
	<8ccbff060910200700o5213a8c2mf95acd9603f96ea@mail.gmail.com>
Message-ID: <5D397B2B-335C-4460-AAA2-60230F6B8C6A@gmail.com>

Try 300 seconds (5 minutes) and see if it improves.


On Oct 20, 2009, at 9:00 AM, Dome Charoenyost wrote:

> 2009/10/20 Rob Forman <rob4manhere at gmail.com>:
>> Hi Dome,
>>
>> Have you tried increasing global_heartbeat to reduce the frequency of
>> odbc calls?  What is it currently set to?
>
> Now 1 min
>
>>
>> Rob
>>
>> On Oct 20, 2009, at 8:31 AM, Dome Charoenyost wrote:
>>
>>> Dear Sir,
>>> I'm using mod_odbc_query and mod_nibble_billing for my calling card
>>> solutoin. i found mod_odbc_query cant work with high load  (200  
>>> calls
>>> concurrent)
>>> i got error
>>> [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory;
>>> Error while executing the query
>>> ]
>>> in FS console
>>>
>>> How to fix it ?
>>>
>>> BG
>>> Dome C.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From ineyaa at gmail.com  Tue Oct 20 07:16:46 2009
From: ineyaa at gmail.com (ineya ineya)
Date: Tue, 20 Oct 2009 16:16:46 +0200
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <20091020133437.GA4941@dan.olp.net>
References: <7b8592a0910190516rafa4398q62c7a99ef703540f@mail.gmail.com>
	<5B924C57-B658-44C0-A1CE-00A3ECB5AA07@freeswitch.org>
	<7b8592a0910191351g14cbb47dj3f19362ff6197157@mail.gmail.com>
	<FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
	<7b8592a0910192253l1313d82j51c933cb5f55ad41@mail.gmail.com>
	<20091020133437.GA4941@dan.olp.net>
Message-ID: <7b8592a0910200716q4a7db93cyb955595e4993ac96@mail.gmail.com>

No, it's the same thing with /64 on both ends.
I tried to built SVN version and modified just IP addresses in
sip_profiles, but still can't call from one phone to another. It goes
straght to voicemail.

Is there a softphone, which you know works with freeswitch on IPv6?
Maybe the error is in phones I'm currently using.

On Tue, Oct 20, 2009 at 3:34 PM, Dan White <dwhite at olp.net> wrote:

> This address caught my eye. Although valid, a /32 address assignment is
> atypical. Is that what you intended? Does assigning a /64 address on both
> ends make any difference?
>
> --
> Dan White
>


From brian at freeswitch.org  Tue Oct 20 07:21:24 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 20 Oct 2009 09:21:24 -0500
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <7b8592a0910200716q4a7db93cyb955595e4993ac96@mail.gmail.com>
References: <7b8592a0910190516rafa4398q62c7a99ef703540f@mail.gmail.com>
	<5B924C57-B658-44C0-A1CE-00A3ECB5AA07@freeswitch.org>
	<7b8592a0910191351g14cbb47dj3f19362ff6197157@mail.gmail.com>
	<FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
	<7b8592a0910192253l1313d82j51c933cb5f55ad41@mail.gmail.com>
	<20091020133437.GA4941@dan.olp.net>
	<7b8592a0910200716q4a7db93cyb955595e4993ac96@mail.gmail.com>
Message-ID: <858EC056-1D84-4F30-A0B7-9A70FF98695D@freeswitch.org>

Well how are you trying to dial users?

/b

On Oct 20, 2009, at 9:16 AM, ineya ineya wrote:

> No, it's the same thing with /64 on both ends.
> I tried to built SVN version and modified just IP addresses in
> sip_profiles, but still can't call from one phone to another. It goes
> straght to voicemail.
>
> Is there a softphone, which you know works with freeswitch on IPv6?
> Maybe the error is in phones I'm currently using.



From m.sobkow at marketelsystems.com  Tue Oct 20 07:27:36 2009
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Tue, 20 Oct 2009 08:27:36 -0600
Subject: [Freeswitch-users] Is anyone running Ubuntu 8.04/Hardy?
In-Reply-To: <903da5680910191917m79bf5ecfvdc8001952dbcc339@mail.gmail.com>
References: <4ADCA34E.5000700@marketelsystems.com>
	<903da5680910191917m79bf5ecfvdc8001952dbcc339@mail.gmail.com>
Message-ID: <4ADDC8D8.7090401@marketelsystems.com>

Gabriel Gunderson wrote:
> On Mon, Oct 19, 2009 at 11:35 AM, Mark Sobkow
> <m.sobkow at marketelsystems.com> wrote:
>   
>> Everyone I've emailed with on the dev list is running the current
>> release of Ubuntu, not 8.04/Hardy.
>>     
>
> Well, what issues?
>
> Gabe
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   
We had problems with loading mod_sofia.  However, we were running an 
older release, then we'd tried upgrading to the source build from 
launchpad.net.

Yesterday we downloaded the current svn.freeswitch.org, and that 
particular problem has gone away.  Now we need to figure out why our 
Erlang component can synchronize from Windows, but not from Linux.

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 



From dome at tel.co.th  Tue Oct 20 07:30:38 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Tue, 20 Oct 2009 21:30:38 +0700
Subject: [Freeswitch-users] mod_odbc_query share memory problem
In-Reply-To: <5D397B2B-335C-4460-AAA2-60230F6B8C6A@gmail.com>
References: <8ccbff060910200631l44cc7dcfm2aa9511d86798aa1@mail.gmail.com>
	<1BFD340C-EC22-434D-B521-8E4D1572A343@gmail.com>
	<8ccbff060910200700o5213a8c2mf95acd9603f96ea@mail.gmail.com>
	<5D397B2B-335C-4460-AAA2-60230F6B8C6A@gmail.com>
Message-ID: <8ccbff060910200730w6a46ffc7u1ade7e5413b76cf6@mail.gmail.com>

2009/10/20 Rob Forman <rob4manhere at gmail.com>:
> Try 300 seconds (5 minutes) and see if it improves.
Ok. i'll try

>
>
> On Oct 20, 2009, at 9:00 AM, Dome Charoenyost wrote:
>
>> 2009/10/20 Rob Forman <rob4manhere at gmail.com>:
>>> Hi Dome,
>>>
>>> Have you tried increasing global_heartbeat to reduce the frequency of
>>> odbc calls? ?What is it currently set to?
>>
>> Now 1 min
>>
>>>
>>> Rob
>>>
>>> On Oct 20, 2009, at 8:31 AM, Dome Charoenyost wrote:
>>>
>>>> Dear Sir,
>>>> I'm using mod_odbc_query and mod_nibble_billing for my calling card
>>>> solutoin. i found mod_odbc_query cant work with high load ?(200
>>>> calls
>>>> concurrent)
>>>> i got error
>>>> [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory;
>>>> Error while executing the query
>>>> ]
>>>> in FS console
>>>>
>>>> How to fix it ?
>>>>
>>>> BG
>>>> Dome C.
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From dome at tel.co.th  Tue Oct 20 07:35:15 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Tue, 20 Oct 2009 21:35:15 +0700
Subject: [Freeswitch-users] mod_odbc_query share memory problem
In-Reply-To: <180CE701-D840-47E6-821C-BB79C9F1945A@scarlet-internet.nl>
References: <8ccbff060910200631l44cc7dcfm2aa9511d86798aa1@mail.gmail.com>
	<180CE701-D840-47E6-821C-BB79C9F1945A@scarlet-internet.nl>
Message-ID: <8ccbff060910200735h73e9459ex7460fcbeac457d59@mail.gmail.com>

2009/10/20 Leon de Rooij <leon at scarlet-internet.nl>:
> Hi,
>
> I haven't run into that problem yet, but did you try increasing the
> maximum shared memory in /proc/sys/kernel/shmmax (sysctl
> kernel.shmmax) ?
Recommend me please i have 4GB RAM.
is posible to increase to 50% RAM ?

Dome C.
>
> regards,
>
> Leon
>
>
>
> On Oct 20, 2009, at 3:31 PM, Dome Charoenyost wrote:
>
>> Dear Sir,
>> I'm using mod_odbc_query and mod_nibble_billing for my calling card
>> solutoin. i found mod_odbc_query cant work with high load ?(200 calls
>> concurrent)
>> i got error
>> [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory;
>> Error while executing the query
>> ]
>> in FS console
>>
>> How to fix it ?
>>
>> BG
>> Dome C.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From ngay01042005 at gmail.com  Tue Oct 20 07:53:01 2009
From: ngay01042005 at gmail.com (homqua)
Date: Tue, 20 Oct 2009 07:53:01 -0700 (PDT)
Subject: [Freeswitch-users] Connect PHP SOAP Web Server with SQLite database
 of FS
Message-ID: <25976554.post@talk.nabble.com>


Now I am building a PHP SOAP Web Service to access the database of FS. Anyone
has idea about how to access sqlite database of FS through PHP ? I have read
about socket event in FS, but I don't know whether it can response with the
query of database or not.
Thanks for your help.
-- 
View this message in context: http://www.nabble.com/Connect-PHP-SOAP-Web-Server-with-SQLite-database-of-FS-tp25976554p25976554.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From bottleman at icf.org.ru  Tue Oct 20 08:00:57 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Tue, 20 Oct 2009 19:00:57 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<65d96fc80910130014hc5f5fcet677f87b7fb57559e@mail.gmail.com>
	<65d96fc80910132359i6d53d8ban10f943795481bf7c@mail.gmail.com>
	<Pine.LNX.4.64.0910141134270.11620@srv.icf.org.ru>
	<65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>
	<Pine.LNX.4.64.0910141202500.21535@srv.icf.org.ru>
	<65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>
	<65d96fc80910140215y357fede1l37f6cba9725ddd89@mail.gmail.com>
	<Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>
	<65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910201841030.12352@srv.icf.org.ru>

On 2009-10-20 10:17 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>> TC>
TC>> TC>call flow is SIP_user => FS => H323_endpoint is failing ..
TC>> coredumped....
TC>> TC>http://pastebin.freeswitch.org/10703
TC>>
TC>> i fix some bugs now,
TC>> ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/mod_h323.tar.bz2 this
TC>> is
TC>> updated version, try it, if you experience no audio try enable rtp proxy in
TC>> you sip profile.
TC>>
TC>>
TC>Hi,  there are several issues... lets start with top 4 :)
TC>
TC>
TC>
TC>1. I'm still stuck with no audio:
TC>
TC>I have this parameter in the sip profile set: <param
TC>name="inbound-proxy-media" value="true"/>
TC>...tried with both with slow start and fast start... any idea ?
TC>
TC>pls check: http://pastebin.freeswitch.org/10771

try to sniff rtp traffic on fs, is it go thouch him.

TC>
TC>
TC>
TC>
TC>
TC>2. outgoing calls still failing in coredumps: what is your dialplan ? ...
TC>how do you call bridge application?

simple:


<action application="bridge" data="h323/${number}"/>

if fs not registered on gk then data="h323/${number}@xxx.xxx.xxx.xxx".
       

TC>
TC>2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3252 Found capability:
TC>UserInput/PointDevice <14>
TC>2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3248 FindCapability: 15
TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external thread
TC>0xb6eb60a0 for id 3048876944
TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:65 Destroyed external thread
TC>0xb6ebafa8 for id 3048876944
TC>2009-10-20 10:08:18.426608 [DEBUG] tlibthrd.cxx:406 Destroyed thread
TC>0xb6ebafa8 PExternalThread:0xb5ba2b90(id = b5ba2b90)
TC>2009-10-20 10:08:18.426608 [DEBUG] h323caps.cxx:3252 Found capability:
TC>UserInput/Modal <15>
TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0,
TC>expiries=0
TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0,
TC>expiries=0
TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external thread
TC>0xb6eba910 for id 3048876944
TC>2009-10-20 10:08:18.426608 [DEBUG] h4601.cxx:1725 Endpoint Attached
TC>Segmentation fault (core dumped)
TC>tculjaga at subZero:~/freeswitch-trunk$
TC>
TC>pls check: http://pastebin.freeswitch.org/10769

look strange, what version of libpt/h323plus you use and freeswitch itself ?

TC>
TC>
TC>
TC>3.  when you hangup from SIP side, the call is not released end-to-end (the
TC>H323 endpoint doesn't get any releaseComplete message)
TC>
TC>2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=2,
TC>expiries=3
TC>2009-10-20 10:10:51.264527 [DEBUG] h323neg.cxx:432 Received
TC>MasterSlaveDeterminationAck: state=Incoming
TC>2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=1,
TC>expiries=3
TC>2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=2,
TC>expiries=4
TC>2009-10-20 10:10:51.264527 [DEBUG] h323.cxx:4138
TC>InternalEstablishedConnectionCheck: connectionState=EstablishedConnection
TC>fastStartState=FastStartAcknowledged
TC>2009-10-20 10:10:51.264527 [DEBUG] h323caps.cxx:3264 FindCapability: "T.120"
TC>2009-10-20 10:10:51.264527 [DEBUG] h323.cxx:4138
TC>InternalEstablishedConnectionCheck: connectionState=EstablishedConnection
TC>fastStartState=FastStartAcknowledged
TC>2009-10-20 10:10:51.264527 [DEBUG] tlibthrd.cxx:1023
TC>PThread::PXBlockOnIO(45,0)
TC>2009-10-20 10:10:54.405479 [NOTICE] sofia.c:328 Hangup sofia/internal/
TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [CS_CONSUME_MEDIA]
TC>[NORMAL_CLEARING]
TC>2009-10-20 10:10:54.405479 [DEBUG] switch_channel.c:1726 Send signal
TC>sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [KILL]
TC>2009-10-20 10:10:54.405479 [DEBUG] switch_core_session.c:932 Send signal
TC>sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [BREAK]
TC>2009-10-20 10:10:54.405479 [DEBUG] switch_core_state_machine.c:437 thread
TC>mismatch skipping state handler.
TC>2009-10-20 10:10:54.405479 [DEBUG] switch_core_state_machine.c:306
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) Running State
TC>Change CS_HANGUP
TC>2009-10-20 10:10:54.406530 [DEBUG] switch_core_state_machine.c:464
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State HANGUP
TC>2009-10-20 10:10:54.406530 [DEBUG] mod_sofia.c:338 Channel sofia/internal/
TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> hanging up, cause:
TC>NORMAL_CLEARING
TC>2009-10-20 10:10:54.406530 [DEBUG] switch_core_state_machine.c:46
TC>sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> Standard HANGUP,
TC>cause: NORMAL_CLEARING
TC>
TC>
TC>pls check: http://pastebin.freeswitch.org/10771

hm, according this log it send release complete:
2009-10-20 10:11:03.995707 [DEBUG] h323pdu.cxx:609 Sending PDU [ip$10.4.62.7:1720/ip$10.1.14.153:3341]
i think please try to dump full traffic of this call from both endpoints and send me.

TC>
TC>
TC>
TC>
TC>4. sometimes when i shutdown FS i get coredimps - from my experience it
TC>looks like you don't wait for a FS thread to finish when you exit...
TC>
TC>2009-10-20 10:05:59.493306 [CONSOLE] switch_event.c:508 Stopping queue
TC>thread 2
TC>2009-10-20 10:05:59.493339 [CONSOLE] switch_core.c:1693 Finalizing Shutdown.
TC>2009-10-20 10:05:59.493379 [CONSOLE] switch_log.c:310 Logger Ended.
TC>2009-10-20 10:05:59.494472 [CONSOLE] switch_core_memory.c:567 Stopping
TC>memory pool queue.
TC>Segmentation fault (core dumped)
TC>tculjaga at subZero:~/freeswitch-trunk$
TC>tculjaga at subZero:~/freeswitch-trunk$

may be, i don't look on this so close at this time.

TC>Please advice your FS/mod_h323.conf.xml settings...

<configuration name="h323.conf" description="H323 Endpoints">
<settings>
<param name="trace-level" value="4"/>
<param name="context" value="default"/>
<param name="dialplan" value="XML"/>
<param name="codec-prefs" value="PCMA,PCMU,GSM,G729,G726,LPC"/>
<param name="gk-address" value="91.195.204.1"/>    <!-- empty to disable, "*" to search LAN -->
<param name="gk-identifer" value="ICFGK"/>  <!-- optional name of gk -->
<param name="gk-interface" value=""/>  <!-- optional listener interface name -->
</settings>
<listeners>
<listener name="default">
<param name="h323-ip" value="$${local_ip_v4}"/>
<param name="h323-port" value="1720"/>
</listener>
</listeners>
</configuration>

this my current settings.
							

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From bottleman at icf.org.ru  Tue Oct 20 08:01:05 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Tue, 20 Oct 2009 19:01:05 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<65d96fc80910130014hc5f5fcet677f87b7fb57559e@mail.gmail.com>
	<65d96fc80910132359i6d53d8ban10f943795481bf7c@mail.gmail.com>
	<Pine.LNX.4.64.0910141134270.11620@srv.icf.org.ru>
	<65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>
	<Pine.LNX.4.64.0910141202500.21535@srv.icf.org.ru>
	<65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>
	<65d96fc80910140215y357fede1l37f6cba9725ddd89@mail.gmail.com>
	<Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>
	<65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910201900570.12352@srv.icf.org.ru>

On 2009-10-20 10:17 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>> TC>
TC>> TC>call flow is SIP_user => FS => H323_endpoint is failing ..
TC>> coredumped....
TC>> TC>http://pastebin.freeswitch.org/10703
TC>>
TC>> i fix some bugs now,
TC>> ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/mod_h323.tar.bz2 this
TC>> is
TC>> updated version, try it, if you experience no audio try enable rtp proxy in
TC>> you sip profile.
TC>>
TC>>
TC>Hi,  there are several issues... lets start with top 4 :)
TC>
TC>
TC>
TC>1. I'm still stuck with no audio:
TC>
TC>I have this parameter in the sip profile set: <param
TC>name="inbound-proxy-media" value="true"/>
TC>...tried with both with slow start and fast start... any idea ?
TC>
TC>pls check: http://pastebin.freeswitch.org/10771

try to sniff rtp traffic on fs, is it go thouch him.

TC>
TC>
TC>
TC>
TC>
TC>2. outgoing calls still failing in coredumps: what is your dialplan ? ...
TC>how do you call bridge application?

simple:


<action application="bridge" data="h323/${number}"/>

if fs not registered on gk then data="h323/${number}@xxx.xxx.xxx.xxx".
       

TC>
TC>2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3252 Found capability:
TC>UserInput/PointDevice <14>
TC>2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3248 FindCapability: 15
TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external thread
TC>0xb6eb60a0 for id 3048876944
TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:65 Destroyed external thread
TC>0xb6ebafa8 for id 3048876944
TC>2009-10-20 10:08:18.426608 [DEBUG] tlibthrd.cxx:406 Destroyed thread
TC>0xb6ebafa8 PExternalThread:0xb5ba2b90(id = b5ba2b90)
TC>2009-10-20 10:08:18.426608 [DEBUG] h323caps.cxx:3252 Found capability:
TC>UserInput/Modal <15>
TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0,
TC>expiries=0
TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0,
TC>expiries=0
TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external thread
TC>0xb6eba910 for id 3048876944
TC>2009-10-20 10:08:18.426608 [DEBUG] h4601.cxx:1725 Endpoint Attached
TC>Segmentation fault (core dumped)
TC>tculjaga at subZero:~/freeswitch-trunk$
TC>
TC>pls check: http://pastebin.freeswitch.org/10769

look strange, what version of libpt/h323plus you use and freeswitch itself ?

TC>
TC>
TC>
TC>3.  when you hangup from SIP side, the call is not released end-to-end (the
TC>H323 endpoint doesn't get any releaseComplete message)
TC>
TC>2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=2,
TC>expiries=3
TC>2009-10-20 10:10:51.264527 [DEBUG] h323neg.cxx:432 Received
TC>MasterSlaveDeterminationAck: state=Incoming
TC>2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=1,
TC>expiries=3
TC>2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=2,
TC>expiries=4
TC>2009-10-20 10:10:51.264527 [DEBUG] h323.cxx:4138
TC>InternalEstablishedConnectionCheck: connectionState=EstablishedConnection
TC>fastStartState=FastStartAcknowledged
TC>2009-10-20 10:10:51.264527 [DEBUG] h323caps.cxx:3264 FindCapability: "T.120"
TC>2009-10-20 10:10:51.264527 [DEBUG] h323.cxx:4138
TC>InternalEstablishedConnectionCheck: connectionState=EstablishedConnection
TC>fastStartState=FastStartAcknowledged
TC>2009-10-20 10:10:51.264527 [DEBUG] tlibthrd.cxx:1023
TC>PThread::PXBlockOnIO(45,0)
TC>2009-10-20 10:10:54.405479 [NOTICE] sofia.c:328 Hangup sofia/internal/
TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [CS_CONSUME_MEDIA]
TC>[NORMAL_CLEARING]
TC>2009-10-20 10:10:54.405479 [DEBUG] switch_channel.c:1726 Send signal
TC>sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [KILL]
TC>2009-10-20 10:10:54.405479 [DEBUG] switch_core_session.c:932 Send signal
TC>sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [BREAK]
TC>2009-10-20 10:10:54.405479 [DEBUG] switch_core_state_machine.c:437 thread
TC>mismatch skipping state handler.
TC>2009-10-20 10:10:54.405479 [DEBUG] switch_core_state_machine.c:306
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) Running State
TC>Change CS_HANGUP
TC>2009-10-20 10:10:54.406530 [DEBUG] switch_core_state_machine.c:464
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State HANGUP
TC>2009-10-20 10:10:54.406530 [DEBUG] mod_sofia.c:338 Channel sofia/internal/
TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> hanging up, cause:
TC>NORMAL_CLEARING
TC>2009-10-20 10:10:54.406530 [DEBUG] switch_core_state_machine.c:46
TC>sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> Standard HANGUP,
TC>cause: NORMAL_CLEARING
TC>
TC>
TC>pls check: http://pastebin.freeswitch.org/10771

hm, according this log it send release complete:
2009-10-20 10:11:03.995707 [DEBUG] h323pdu.cxx:609 Sending PDU [ip$10.4.62.7:1720/ip$10.1.14.153:3341]
i think please try to dump full traffic of this call from both endpoints and send me.

TC>
TC>
TC>
TC>
TC>4. sometimes when i shutdown FS i get coredimps - from my experience it
TC>looks like you don't wait for a FS thread to finish when you exit...
TC>
TC>2009-10-20 10:05:59.493306 [CONSOLE] switch_event.c:508 Stopping queue
TC>thread 2
TC>2009-10-20 10:05:59.493339 [CONSOLE] switch_core.c:1693 Finalizing Shutdown.
TC>2009-10-20 10:05:59.493379 [CONSOLE] switch_log.c:310 Logger Ended.
TC>2009-10-20 10:05:59.494472 [CONSOLE] switch_core_memory.c:567 Stopping
TC>memory pool queue.
TC>Segmentation fault (core dumped)
TC>tculjaga at subZero:~/freeswitch-trunk$
TC>tculjaga at subZero:~/freeswitch-trunk$

may be, i don't look on this so close at this time.

TC>Please advice your FS/mod_h323.conf.xml settings...

<configuration name="h323.conf" description="H323 Endpoints">
<settings>
<param name="trace-level" value="4"/>
<param name="context" value="default"/>
<param name="dialplan" value="XML"/>
<param name="codec-prefs" value="PCMA,PCMU,GSM,G729,G726,LPC"/>
<param name="gk-address" value="91.195.204.1"/>    <!-- empty to disable, "*" to search LAN -->
<param name="gk-identifer" value="ICFGK"/>  <!-- optional name of gk -->
<param name="gk-interface" value=""/>  <!-- optional listener interface name -->
</settings>
<listeners>
<listener name="default">
<param name="h323-ip" value="$${local_ip_v4}"/>
<param name="h323-port" value="1720"/>
</listener>
</listeners>
</configuration>

this my current settings.
							

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From rob4manhere at gmail.com  Tue Oct 20 08:03:36 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Tue, 20 Oct 2009 10:03:36 -0500
Subject: [Freeswitch-users] mod_odbc_query share memory problem
In-Reply-To: <8ccbff060910200735h73e9459ex7460fcbeac457d59@mail.gmail.com>
References: <8ccbff060910200631l44cc7dcfm2aa9511d86798aa1@mail.gmail.com>
	<180CE701-D840-47E6-821C-BB79C9F1945A@scarlet-internet.nl>
	<8ccbff060910200735h73e9459ex7460fcbeac457d59@mail.gmail.com>
Message-ID: <85AF1814-9B1A-439C-83F5-D46F9BFEC140@gmail.com>

You could, but I would try just doubling whatever it is to see if  
thats improves the issue first.  The default is 32MB.  You could  
double it to 64MB and test again.

What is it currently set to (run: sysctl kernel.shmmax)?  You can  
change it on the fly with sysctl.  Once you're done testing and want  
it to persist across reboots, you'll want to set in /etc/sysctl.conf .

Cheers,
Rob

On Oct 20, 2009, at 9:35 AM, Dome Charoenyost wrote:

> 2009/10/20 Leon de Rooij <leon at scarlet-internet.nl>:
>> Hi,
>>
>> I haven't run into that problem yet, but did you try increasing the
>> maximum shared memory in /proc/sys/kernel/shmmax (sysctl
>> kernel.shmmax) ?
> Recommend me please i have 4GB RAM.
> is posible to increase to 50% RAM ?
>
> Dome C.
>>
>> regards,
>>
>> Leon
>>
>>
>>
>> On Oct 20, 2009, at 3:31 PM, Dome Charoenyost wrote:
>>
>>> Dear Sir,
>>> I'm using mod_odbc_query and mod_nibble_billing for my calling card
>>> solutoin. i found mod_odbc_query cant work with high load  (200  
>>> calls
>>> concurrent)
>>> i got error
>>> [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory;
>>> Error while executing the query
>>> ]
>>> in FS console
>>>
>>> How to fix it ?
>>>
>>> BG
>>> Dome C.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From dome at tel.co.th  Tue Oct 20 08:14:59 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Tue, 20 Oct 2009 22:14:59 +0700
Subject: [Freeswitch-users] mod_odbc_query share memory problem
In-Reply-To: <85AF1814-9B1A-439C-83F5-D46F9BFEC140@gmail.com>
References: <8ccbff060910200631l44cc7dcfm2aa9511d86798aa1@mail.gmail.com>
	<180CE701-D840-47E6-821C-BB79C9F1945A@scarlet-internet.nl>
	<8ccbff060910200735h73e9459ex7460fcbeac457d59@mail.gmail.com>
	<85AF1814-9B1A-439C-83F5-D46F9BFEC140@gmail.com>
Message-ID: <8ccbff060910200814w2fa37866m8e211a73610f03c4@mail.gmail.com>

2009/10/20 Rob Forman <rob4manhere at gmail.com>:
> You could, but I would try just doubling whatever it is to see if
> thats improves the issue first. ?The default is 32MB. ?You could
> double it to 64MB and test again.
Ok' let's me try 64MB first
i found some information
https://www.ishoof.org/project/wiki/index.php/ORACLE_11_G_-_Gentoo_Installation

>
> What is it currently set to (run: sysctl kernel.shmmax)? ?You can
> change it on the fly with sysctl. ?Once you're done testing and want
> it to persist across reboots, you'll want to set in /etc/sysctl.conf .
>
> Cheers,
> Rob
>
> On Oct 20, 2009, at 9:35 AM, Dome Charoenyost wrote:
>
>> 2009/10/20 Leon de Rooij <leon at scarlet-internet.nl>:
>>> Hi,
>>>
>>> I haven't run into that problem yet, but did you try increasing the
>>> maximum shared memory in /proc/sys/kernel/shmmax (sysctl
>>> kernel.shmmax) ?
>> Recommend me please i have 4GB RAM.
>> is posible to increase to 50% RAM ?
>>
>> Dome C.
>>>
>>> regards,
>>>
>>> Leon
>>>
>>>
>>>
>>> On Oct 20, 2009, at 3:31 PM, Dome Charoenyost wrote:
>>>
>>>> Dear Sir,
>>>> I'm using mod_odbc_query and mod_nibble_billing for my calling card
>>>> solutoin. i found mod_odbc_query cant work with high load ?(200
>>>> calls
>>>> concurrent)
>>>> i got error
>>>> [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory;
>>>> Error while executing the query
>>>> ]
>>>> in FS console
>>>>
>>>> How to fix it ?
>>>>
>>>> BG
>>>> Dome C.
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From bottleman at icf.org.ru  Tue Oct 20 08:17:37 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Tue, 20 Oct 2009 19:17:37 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<65d96fc80910130014hc5f5fcet677f87b7fb57559e@mail.gmail.com>
	<65d96fc80910132359i6d53d8ban10f943795481bf7c@mail.gmail.com>
	<Pine.LNX.4.64.0910141134270.11620@srv.icf.org.ru>
	<65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>
	<Pine.LNX.4.64.0910141202500.21535@srv.icf.org.ru>
	<65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>
	<65d96fc80910140215y357fede1l37f6cba9725ddd89@mail.gmail.com>
	<Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>
	<65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910201901050.12352@srv.icf.org.ru>

On 2009-10-20 10:17 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>> TC>
TC>> TC>call flow is SIP_user => FS => H323_endpoint is failing ..
TC>> coredumped....
TC>> TC>http://pastebin.freeswitch.org/10703
TC>>
TC>> i fix some bugs now,
TC>> ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/mod_h323.tar.bz2 this
TC>> is
TC>> updated version, try it, if you experience no audio try enable rtp proxy in
TC>> you sip profile.
TC>>
TC>>
TC>Hi,  there are several issues... lets start with top 4 :)
TC>
TC>
TC>
TC>1. I'm still stuck with no audio:
TC>
TC>I have this parameter in the sip profile set: <param
TC>name="inbound-proxy-media" value="true"/>
TC>...tried with both with slow start and fast start... any idea ?
TC>
TC>pls check: http://pastebin.freeswitch.org/10771

try to sniff rtp traffic on fs, is it go thouch him.

TC>
TC>
TC>
TC>
TC>
TC>2. outgoing calls still failing in coredumps: what is your dialplan ? ...
TC>how do you call bridge application?

simple:


<action application="bridge" data="h323/${number}"/>

if fs not registered on gk then data="h323/${number}@xxx.xxx.xxx.xxx".
       

TC>
TC>2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3252 Found capability:
TC>UserInput/PointDevice <14>
TC>2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3248 FindCapability: 15
TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external thread
TC>0xb6eb60a0 for id 3048876944
TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:65 Destroyed external thread
TC>0xb6ebafa8 for id 3048876944
TC>2009-10-20 10:08:18.426608 [DEBUG] tlibthrd.cxx:406 Destroyed thread
TC>0xb6ebafa8 PExternalThread:0xb5ba2b90(id = b5ba2b90)
TC>2009-10-20 10:08:18.426608 [DEBUG] h323caps.cxx:3252 Found capability:
TC>UserInput/Modal <15>
TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0,
TC>expiries=0
TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0,
TC>expiries=0
TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external thread
TC>0xb6eba910 for id 3048876944
TC>2009-10-20 10:08:18.426608 [DEBUG] h4601.cxx:1725 Endpoint Attached
TC>Segmentation fault (core dumped)
TC>tculjaga at subZero:~/freeswitch-trunk$
TC>
TC>pls check: http://pastebin.freeswitch.org/10769

look strange, what version of libpt/h323plus you use and freeswitch itself ?

TC>
TC>
TC>
TC>3.  when you hangup from SIP side, the call is not released end-to-end (the
TC>H323 endpoint doesn't get any releaseComplete message)
TC>
TC>2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=2,
TC>expiries=3
TC>2009-10-20 10:10:51.264527 [DEBUG] h323neg.cxx:432 Received
TC>MasterSlaveDeterminationAck: state=Incoming
TC>2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=1,
TC>expiries=3
TC>2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=2,
TC>expiries=4
TC>2009-10-20 10:10:51.264527 [DEBUG] h323.cxx:4138
TC>InternalEstablishedConnectionCheck: connectionState=EstablishedConnection
TC>fastStartState=FastStartAcknowledged
TC>2009-10-20 10:10:51.264527 [DEBUG] h323caps.cxx:3264 FindCapability: "T.120"
TC>2009-10-20 10:10:51.264527 [DEBUG] h323.cxx:4138
TC>InternalEstablishedConnectionCheck: connectionState=EstablishedConnection
TC>fastStartState=FastStartAcknowledged
TC>2009-10-20 10:10:51.264527 [DEBUG] tlibthrd.cxx:1023
TC>PThread::PXBlockOnIO(45,0)
TC>2009-10-20 10:10:54.405479 [NOTICE] sofia.c:328 Hangup sofia/internal/
TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [CS_CONSUME_MEDIA]
TC>[NORMAL_CLEARING]
TC>2009-10-20 10:10:54.405479 [DEBUG] switch_channel.c:1726 Send signal
TC>sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [KILL]
TC>2009-10-20 10:10:54.405479 [DEBUG] switch_core_session.c:932 Send signal
TC>sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [BREAK]
TC>2009-10-20 10:10:54.405479 [DEBUG] switch_core_state_machine.c:437 thread
TC>mismatch skipping state handler.
TC>2009-10-20 10:10:54.405479 [DEBUG] switch_core_state_machine.c:306
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) Running State
TC>Change CS_HANGUP
TC>2009-10-20 10:10:54.406530 [DEBUG] switch_core_state_machine.c:464
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State HANGUP
TC>2009-10-20 10:10:54.406530 [DEBUG] mod_sofia.c:338 Channel sofia/internal/
TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> hanging up, cause:
TC>NORMAL_CLEARING
TC>2009-10-20 10:10:54.406530 [DEBUG] switch_core_state_machine.c:46
TC>sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> Standard HANGUP,
TC>cause: NORMAL_CLEARING
TC>
TC>
TC>pls check: http://pastebin.freeswitch.org/10771

hm, according this log it send release complete:
2009-10-20 10:11:03.995707 [DEBUG] h323pdu.cxx:609 Sending PDU [ip$10.4.62.7:1720/ip$10.1.14.153:3341]
i think please try to dump full traffic of this call from both endpoints and send me.

TC>
TC>
TC>
TC>
TC>4. sometimes when i shutdown FS i get coredimps - from my experience it
TC>looks like you don't wait for a FS thread to finish when you exit...
TC>
TC>2009-10-20 10:05:59.493306 [CONSOLE] switch_event.c:508 Stopping queue
TC>thread 2
TC>2009-10-20 10:05:59.493339 [CONSOLE] switch_core.c:1693 Finalizing Shutdown.
TC>2009-10-20 10:05:59.493379 [CONSOLE] switch_log.c:310 Logger Ended.
TC>2009-10-20 10:05:59.494472 [CONSOLE] switch_core_memory.c:567 Stopping
TC>memory pool queue.
TC>Segmentation fault (core dumped)
TC>tculjaga at subZero:~/freeswitch-trunk$
TC>tculjaga at subZero:~/freeswitch-trunk$

may be, i don't look on this so close at this time.

TC>Please advice your FS/mod_h323.conf.xml settings...

<configuration name="h323.conf" description="H323 Endpoints">
<settings>
<param name="trace-level" value="4"/>
<param name="context" value="default"/>
<param name="dialplan" value="XML"/>
<param name="codec-prefs" value="PCMA,PCMU,GSM,G729,G726,LPC"/>
<param name="gk-address" value="91.195.204.1"/>    <!-- empty to disable, "*" to search LAN -->
<param name="gk-identifer" value="ICFGK"/>  <!-- optional name of gk -->
<param name="gk-interface" value=""/>  <!-- optional listener interface name -->
</settings>
<listeners>
<listener name="default">
<param name="h323-ip" value="$${local_ip_v4}"/>
<param name="h323-port" value="1720"/>
</listener>
</listeners>
</configuration>

this my current settings.
							

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From ineyaa at gmail.com  Tue Oct 20 08:21:09 2009
From: ineyaa at gmail.com (ineya ineya)
Date: Tue, 20 Oct 2009 17:21:09 +0200
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <858EC056-1D84-4F30-A0B7-9A70FF98695D@freeswitch.org>
References: <7b8592a0910190516rafa4398q62c7a99ef703540f@mail.gmail.com>
	<5B924C57-B658-44C0-A1CE-00A3ECB5AA07@freeswitch.org>
	<7b8592a0910191351g14cbb47dj3f19362ff6197157@mail.gmail.com>
	<FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
	<7b8592a0910192253l1313d82j51c933cb5f55ad41@mail.gmail.com>
	<20091020133437.GA4941@dan.olp.net>
	<7b8592a0910200716q4a7db93cyb955595e4993ac96@mail.gmail.com>
	<858EC056-1D84-4F30-A0B7-9A70FF98695D@freeswitch.org>
Message-ID: <7b8592a0910200821m18fe9b4dg73dda05a10699af2@mail.gmail.com>

By numbers, I have 2 numbers registered, so I dial "1006" or "1007".
This worked for IPv4, so I haven't thought about doing it differently for IPv6.
SIP messages looked OK to me - you just gave me an idea to compare SIP
messages for IPv4 and IPv6. I'll try that tommorow.


On Tue, Oct 20, 2009 at 4:21 PM, Brian West <brian at freeswitch.org> wrote:
> Well how are you trying to dial users?
>
> /b
>


From rob4manhere at gmail.com  Tue Oct 20 08:23:24 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Tue, 20 Oct 2009 10:23:24 -0500
Subject: [Freeswitch-users] mod_odbc_query share memory problem
In-Reply-To: <8ccbff060910200814w2fa37866m8e211a73610f03c4@mail.gmail.com>
References: <8ccbff060910200631l44cc7dcfm2aa9511d86798aa1@mail.gmail.com>
	<180CE701-D840-47E6-821C-BB79C9F1945A@scarlet-internet.nl>
	<8ccbff060910200735h73e9459ex7460fcbeac457d59@mail.gmail.com>
	<85AF1814-9B1A-439C-83F5-D46F9BFEC140@gmail.com>
	<8ccbff060910200814w2fa37866m8e211a73610f03c4@mail.gmail.com>
Message-ID: <7B94B9BF-B15F-4690-B0E6-74776BD26898@gmail.com>

Those are recommended values for Oracle.  Freeswitch != Oracle.  They  
behave and use a server's resources very differently.

I wouldn't change more than you need for now.  Tweak shmmax then go  
from there.

Cheers,
Rob

On Oct 20, 2009, at 10:14 AM, Dome Charoenyost wrote:

> 2009/10/20 Rob Forman <rob4manhere at gmail.com>:
>> You could, but I would try just doubling whatever it is to see if
>> thats improves the issue first.  The default is 32MB.  You could
>> double it to 64MB and test again.
> Ok' let's me try 64MB first
> i found some information
> https://www.ishoof.org/project/wiki/index.php/ORACLE_11_G_-_Gentoo_Installation
>
>>
>> What is it currently set to (run: sysctl kernel.shmmax)?  You can
>> change it on the fly with sysctl.  Once you're done testing and want
>> it to persist across reboots, you'll want to set in /etc/ 
>> sysctl.conf .
>>
>> Cheers,
>> Rob
>>
>> On Oct 20, 2009, at 9:35 AM, Dome Charoenyost wrote:
>>
>>> 2009/10/20 Leon de Rooij <leon at scarlet-internet.nl>:
>>>> Hi,
>>>>
>>>> I haven't run into that problem yet, but did you try increasing the
>>>> maximum shared memory in /proc/sys/kernel/shmmax (sysctl
>>>> kernel.shmmax) ?
>>> Recommend me please i have 4GB RAM.
>>> is posible to increase to 50% RAM ?
>>>
>>> Dome C.
>>>>
>>>> regards,
>>>>
>>>> Leon
>>>>
>>>>
>>>>
>>>> On Oct 20, 2009, at 3:31 PM, Dome Charoenyost wrote:
>>>>
>>>>> Dear Sir,
>>>>> I'm using mod_odbc_query and mod_nibble_billing for my calling  
>>>>> card
>>>>> solutoin. i found mod_odbc_query cant work with high load  (200
>>>>> calls
>>>>> concurrent)
>>>>> i got error
>>>>> [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory;
>>>>> Error while executing the query
>>>>> ]
>>>>> in FS console
>>>>>
>>>>> How to fix it ?
>>>>>
>>>>> BG
>>>>> Dome C.
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From gmaruzz at celliax.org  Tue Oct 20 08:25:09 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 20 Oct 2009 17:25:09 +0200
Subject: [Freeswitch-users] Is anyone running Ubuntu 8.04/Hardy?
In-Reply-To: <4ADDC8D8.7090401@marketelsystems.com>
References: <4ADCA34E.5000700@marketelsystems.com>
	<903da5680910191917m79bf5ecfvdc8001952dbcc339@mail.gmail.com>
	<4ADDC8D8.7090401@marketelsystems.com>
Message-ID: <7b197bef0910200825v64e1239emf717f357e423794@mail.gmail.com>

maybe next time test and/or search the mailing list before asking. I
was a little worried when I read that it do not works on Hardy. Good
to be reassured, it works. :-)



On 10/20/09, Mark Sobkow <m.sobkow at marketelsystems.com> wrote:
> Gabriel Gunderson wrote:
>> On Mon, Oct 19, 2009 at 11:35 AM, Mark Sobkow
>> <m.sobkow at marketelsystems.com> wrote:
>>
>>> Everyone I've emailed with on the dev list is running the current
>>> release of Ubuntu, not 8.04/Hardy.
>>>
>>
>> Well, what issues?
>>
>> Gabe
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> We had problems with loading mod_sofia.  However, we were running an
> older release, then we'd tried upgrading to the source build from
> launchpad.net.
>
> Yesterday we downloaded the current svn.freeswitch.org, and that
> particular problem has gone away.  Now we need to figure out why our
> Erlang component can synchronize from Windows, but not from Linux.
>
> --
> Mark Sobkow
> Senior Developer
> MarkeTel Multi-Line Dialing Systems LTD.
> 428 Victoria Ave
> Regina, SK S4N-0P6
> Toll-Free: 800-289-8616-X533
> Local: 306-359-6893-X533
> Fax: 306-359-6879
> Email: m.sobkow at marketelsystems.com
> Web: http://www.marketelsystems.com
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From abovebluesky at gmail.com  Tue Oct 20 08:25:48 2009
From: abovebluesky at gmail.com (Uncle Johny)
Date: Tue, 20 Oct 2009 08:25:48 -0700 (PDT)
Subject: [Freeswitch-users] bind_meta_app problem with custome extension
In-Reply-To: <25963971.post@talk.nabble.com>
References: <25963971.post@talk.nabble.com>
Message-ID: <25977224.post@talk.nabble.com>


OMG such a stupid user error!

I have to write
<action application="bind_meta_app" data="7 ab s
/usr/local/freeswitch/conf/dialplan/execute_extension::roar XML features"/>

with double colon instead of a single one.

Thanks everyone for help!


Uncle Johny wrote:
> 
> Hi guys,
> 
> I hope you can help me with this one since it looks like some FS bug or my
> user error :) .I have a simple dialplan in FS which allows to transfer a
> person to some 3rd party phone but it works only if you call it from
> default context.
> If it is not called from default context I get this error all the time:
> 2009-09-12 11:58:17.484680 [ERR] mod_native_file.c:68 Error opening
> /usr/local/freeswitch/conf/dialplan/execute_extension:roar XML
> features.PCMU
> 
> With the following configuration 2 parties will be connected but whenever
> you press *7 it just does not work and FS splits the message. Very weird
> that this works if I put the extension into default.xml.
> 
> Any ideas what is wrong here?
> 
> My config:
> 
> I tried to create my own context and call it pstn.
> So when FS gets a call from proxy it goes to pstn context, since I have
> this line in sip_proxies/internal.xml
> 
> 
> in dialplan directory I have pstn.xml
> <include>
>   <context name="pstn">
>    <extension name="pstn_enum">
>     <condition field="destination_number" expression="^(.*)$">
>                 <action application="answer"/>
>                 <action application="bind_meta_app" data="7 ab s
> /usr/local/freeswitch/conf/dialplan/execute_extension:roar XML features"/>
>                 <action application="set"
> data="hangup_after_bridge=true"/>
>                 <action application="bridge"
> data="sofia/external/${destination_number}@207.88.122.16"/>
>     </condition>
>   </extension>
> </context>
> </include>
> 
> In features.xml (It can be anything there since FS never reaches this code
> and dies before executing it):
> <extension name="roar">
>      <condition field="destination_number" expression="^roar$">
>         <action application="set" data="continue_on_fail=true"/>
>         <action application="log" data="INFO transfer context" />
>         <action application="read" data="1 1 /root/1.gsm attxfer_callthis
> 1000 #"/>
>         <action application="log" data="INFO blah ${attxfer_callthis}"/>
>         <action application="set" data="origination_cancel_key=#"/>
>         <action application="log" data="INFO after read"/>
>      </condition>
> </extension>
> 
>  
> 

-- 
View this message in context: http://www.nabble.com/bind_meta_app-problem-with-custome-extension-tp25963971p25977224.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From anthony.minessale at gmail.com  Tue Oct 20 08:31:49 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 20 Oct 2009 10:31:49 -0500
Subject: [Freeswitch-users] Stereo Support
In-Reply-To: <4ADD685C.1000506@ewetel.de>
References: <4ADD685C.1000506@ewetel.de>
Message-ID: <191c3a030910200831h59a58e2aqf1aee92cc4859786@mail.gmail.com>

There is not enough spec/devices using spec on stereo to try to implement it
at this time.
Maybe some day.


On Tue, Oct 20, 2009 at 2:35 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> I wonder if there is real stereo support planned for FreeSWITCH in terms
> of streaming music/video-audio to desktop? I ask, because I heared that
> celt codec is supporting stereo.
>
> regards
> helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
>
> iD8DBQFK3Whc4tZeNddg3dwRAkdEAKCLBSV4H8q5JN4NZ39fgSZPxr5PCgCgmpTq
> PAu8qozLP5FaVKIwfCXVqjU=
> =Zseo
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From dome at tel.co.th  Tue Oct 20 08:33:25 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Tue, 20 Oct 2009 22:33:25 +0700
Subject: [Freeswitch-users] mod_odbc_query share memory problem
In-Reply-To: <7B94B9BF-B15F-4690-B0E6-74776BD26898@gmail.com>
References: <8ccbff060910200631l44cc7dcfm2aa9511d86798aa1@mail.gmail.com>
	<180CE701-D840-47E6-821C-BB79C9F1945A@scarlet-internet.nl>
	<8ccbff060910200735h73e9459ex7460fcbeac457d59@mail.gmail.com>
	<85AF1814-9B1A-439C-83F5-D46F9BFEC140@gmail.com>
	<8ccbff060910200814w2fa37866m8e211a73610f03c4@mail.gmail.com>
	<7B94B9BF-B15F-4690-B0E6-74776BD26898@gmail.com>
Message-ID: <8ccbff060910200833g892d35fr69c2a4ea5c16562b@mail.gmail.com>

Now i'm setting to  64MB
I found channel problem also. many channel not disconnect
freeswitch at internal> show calls count
87 total.

freeswitch at internal> show channels count
466 total.

freeswitch at internal> status
UP 0 years, 0 days, 1 hour, 53 minutes, 23 seconds, 872 milliseconds,
0 microseconds
11641 session(s) since startup
154 session(s) 0/30
1000 session(s) max

I think channels should be 154 (same session) but 466 is to high. may
be some channels not disconnect

Dome C.



2009/10/20 Rob Forman <rob4manhere at gmail.com>:
> Those are recommended values for Oracle. ?Freeswitch != Oracle. ?They
> behave and use a server's resources very differently.
>
> I wouldn't change more than you need for now. ?Tweak shmmax then go
> from there.
>
> Cheers,
> Rob
>
> On Oct 20, 2009, at 10:14 AM, Dome Charoenyost wrote:
>
>> 2009/10/20 Rob Forman <rob4manhere at gmail.com>:
>>> You could, but I would try just doubling whatever it is to see if
>>> thats improves the issue first. ?The default is 32MB. ?You could
>>> double it to 64MB and test again.
>> Ok' let's me try 64MB first
>> i found some information
>> https://www.ishoof.org/project/wiki/index.php/ORACLE_11_G_-_Gentoo_Installation
>>
>>>
>>> What is it currently set to (run: sysctl kernel.shmmax)? ?You can
>>> change it on the fly with sysctl. ?Once you're done testing and want
>>> it to persist across reboots, you'll want to set in /etc/
>>> sysctl.conf .
>>>
>>> Cheers,
>>> Rob
>>>
>>> On Oct 20, 2009, at 9:35 AM, Dome Charoenyost wrote:
>>>
>>>> 2009/10/20 Leon de Rooij <leon at scarlet-internet.nl>:
>>>>> Hi,
>>>>>
>>>>> I haven't run into that problem yet, but did you try increasing the
>>>>> maximum shared memory in /proc/sys/kernel/shmmax (sysctl
>>>>> kernel.shmmax) ?
>>>> Recommend me please i have 4GB RAM.
>>>> is posible to increase to 50% RAM ?
>>>>
>>>> Dome C.
>>>>>
>>>>> regards,
>>>>>
>>>>> Leon
>>>>>
>>>>>
>>>>>
>>>>> On Oct 20, 2009, at 3:31 PM, Dome Charoenyost wrote:
>>>>>
>>>>>> Dear Sir,
>>>>>> I'm using mod_odbc_query and mod_nibble_billing for my calling
>>>>>> card
>>>>>> solutoin. i found mod_odbc_query cant work with high load ?(200
>>>>>> calls
>>>>>> concurrent)
>>>>>> i got error
>>>>>> [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory;
>>>>>> Error while executing the query
>>>>>> ]
>>>>>> in FS console
>>>>>>
>>>>>> How to fix it ?
>>>>>>
>>>>>> BG
>>>>>> Dome C.
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at freeswitch.org  Tue Oct 20 08:34:21 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 20 Oct 2009 10:34:21 -0500
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <7b8592a0910200821m18fe9b4dg73dda05a10699af2@mail.gmail.com>
References: <7b8592a0910190516rafa4398q62c7a99ef703540f@mail.gmail.com>
	<5B924C57-B658-44C0-A1CE-00A3ECB5AA07@freeswitch.org>
	<7b8592a0910191351g14cbb47dj3f19362ff6197157@mail.gmail.com>
	<FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
	<7b8592a0910192253l1313d82j51c933cb5f55ad41@mail.gmail.com>
	<20091020133437.GA4941@dan.olp.net>
	<7b8592a0910200716q4a7db93cyb955595e4993ac96@mail.gmail.com>
	<858EC056-1D84-4F30-A0B7-9A70FF98695D@freeswitch.org>
	<7b8592a0910200821m18fe9b4dg73dda05a10699af2@mail.gmail.com>
Message-ID: <642B6CED-CDF3-4D40-B493-0F2D6222C30B@freeswitch.org>

The defaults will NOT work with ipv6 out of the box because the  
sofia_contact on the directory only looks at the internal profile NOT  
the internal-ipv6 profile... open up the directory default and change  
the sofia_contact to prepend the internal-ipv6/user at domain

/b

On Oct 20, 2009, at 10:21 AM, ineya ineya wrote:

> By numbers, I have 2 numbers registered, so I dial "1006" or "1007".
> This worked for IPv4, so I haven't thought about doing it  
> differently for IPv6.
> SIP messages looked OK to me - you just gave me an idea to compare SIP
> messages for IPv4 and IPv6. I'll try that tommorow.



From dome at tel.co.th  Tue Oct 20 08:47:37 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Tue, 20 Oct 2009 22:47:37 +0700
Subject: [Freeswitch-users] CS_REPORTING Channel event state
Message-ID: <8ccbff060910200847y5e7ccab6o84d431836970bcb4@mail.gmail.com>

Dear All
What's CS_REPORTING state ?
I found many channels not hang up  ans state is CS_REPORTING
e264f84a-bd87-11de-9a90-2320c02172de,outbound,2009-10-20
21:50:26,1256050226,sofia/external/xxxxxxxxxxxx at xxx.xxx.xx.191:7050,CS_REPORTING,FreeSWITCH,xxxxxxxxxxxxxxxxxxxx,xx.xxx.xxx.xxx,7050,,,XML,public,,,,,

BG
Dome C.


From msc at freeswitch.org  Tue Oct 20 09:19:45 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 20 Oct 2009 09:19:45 -0700
Subject: [Freeswitch-users] Qustion about INFO messages after
	Connect/Answer
In-Reply-To: <4ADD65DA.4050809@ewetel.de>
References: <4AD859A4.5000101@ewetel.de>
	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>
	<4ADC7C09.6070503@ewetel.de>
	<191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>
	<4ADC85C0.7060908@ewetel.de> <4ADC8B6E.7070404@ewetel.de>
	<191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>
	<4ADD65DA.4050809@ewetel.de>
Message-ID: <87f2f3b90910200919i62dac94fq1b6c9289d9fb357c@mail.gmail.com>

On Tue, Oct 20, 2009 at 12:25 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello Anthony,
>
>
> On 19.10.2009 22:07, Anthony Minessale wrote:
> > please update and test trunk
> >
> > 1) I changed the core to remove the excess data by default in your
> scenario
> > 2) I added variables you can use to control it
> > origination_callee_id_name origination_callee_id_number which belong in
> > {} in the dial string eg
> {origination_callee_id_number=1234}openzap/1/a/1234
>
> updated and testet for SIP calls. "origination_callee_id_number=1234"
> works within the dial string and via using "export" application but not
> via "set" application. Didn't test it for openzap, yet but I guess it
> will work, too.
>

Thanks for testing. BTW, according to Tony's instructions the user needs to
put the variable definition inside the {} of the dialstring. This means that
the vars are designed for the new leg, not the local leg, which means that
"export" would work but "set" would not. (Remember, "set" is to set a chan
var on the local channel, "export" will set a chan var locally *and* on the
other call leg. See also the "nolocal" option of export.)

>
> What left is that "unknown" thing on callee's display (SIP phones only I
> guess) ... I would suggest to make sending "unknown" INFO message
> optional, but to be honest, I have no idea for what it was invented, so
> maybe my suggestion is nonsens.
>
>
Under what conditions did you see "unknown"? I'm wondering if the user can
just pick a default other than "unknown" if he wants something else to be
displayed.

Thoughts?
-MC
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From brian at freeswitch.org  Tue Oct 20 09:41:48 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 20 Oct 2009 11:41:48 -0500
Subject: [Freeswitch-users] Qustion about INFO messages after
	Connect/Answer
In-Reply-To: <87f2f3b90910200919i62dac94fq1b6c9289d9fb357c@mail.gmail.com>
References: <4AD859A4.5000101@ewetel.de>
	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>
	<4ADC7C09.6070503@ewetel.de>
	<191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>
	<4ADC85C0.7060908@ewetel.de> <4ADC8B6E.7070404@ewetel.de>
	<191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>
	<4ADD65DA.4050809@ewetel.de>
	<87f2f3b90910200919i62dac94fq1b6c9289d9fb357c@mail.gmail.com>
Message-ID: <CBC6759C-5192-4731-BACA-79C90CD4EE27@freeswitch.org>

Or just set the var to what you want it to say?

/b

On Oct 20, 2009, at 11:19 AM, Michael Collins wrote:

>
> Under what conditions did you see "unknown"? I'm wondering if the  
> user can just pick a default other than "unknown" if he wants  
> something else to be displayed.
>
> Thoughts?
> -MC



From ineyaa at gmail.com  Tue Oct 20 09:47:33 2009
From: ineyaa at gmail.com (ineya ineya)
Date: Tue, 20 Oct 2009 18:47:33 +0200
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <642B6CED-CDF3-4D40-B493-0F2D6222C30B@freeswitch.org>
References: <7b8592a0910190516rafa4398q62c7a99ef703540f@mail.gmail.com>
	<5B924C57-B658-44C0-A1CE-00A3ECB5AA07@freeswitch.org>
	<7b8592a0910191351g14cbb47dj3f19362ff6197157@mail.gmail.com>
	<FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
	<7b8592a0910192253l1313d82j51c933cb5f55ad41@mail.gmail.com>
	<20091020133437.GA4941@dan.olp.net>
	<7b8592a0910200716q4a7db93cyb955595e4993ac96@mail.gmail.com>
	<858EC056-1D84-4F30-A0B7-9A70FF98695D@freeswitch.org>
	<7b8592a0910200821m18fe9b4dg73dda05a10699af2@mail.gmail.com>
	<642B6CED-CDF3-4D40-B493-0F2D6222C30B@freeswitch.org>
Message-ID: <7b8592a0910200947k6ca5a391l35786f0e855ae388@mail.gmail.com>

Hmm, I didn't about this.

so in directory/default.xml I have:
 <param name="dial-string"
value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>

and the modified version for IPv6 would be ?...
 <param name="dial-string"
value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(internal-ipv6/${dialed_user}@${dialed_domain})}"/>



On Tue, Oct 20, 2009 at 5:34 PM, Brian West <brian at freeswitch.org> wrote:
> The defaults will NOT work with ipv6 out of the box because the
> sofia_contact on the directory only looks at the internal profile NOT
> the internal-ipv6 profile... open up the directory default and change
> the sofia_contact to prepend the internal-ipv6/user at domain
>
> /b
>


From msc at freeswitch.org  Tue Oct 20 10:42:07 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 20 Oct 2009 10:42:07 -0700
Subject: [Freeswitch-users] Janitorial Project Update - Wiki Cleanup
Message-ID: <87f2f3b90910201042o75b93127v619d61b84d962677@mail.gmail.com>

Hello all,

As you may know we have a weekly conference call each Friday. This week's
agenda has several wiki cleanup sub-projects. I wanted to crowdsource this
because, as the saying goes, many hands makes the work light. In other
words, if everyone can help a little bit then we won't be dumping a lot of
work on just a few people.

HOW YOU CAN HELP

If you don't already have a wiki account then please go to
http://wiki.freeswitch.org and create one. All wiki users have editing
rights and can make changes to wiki content. Don't worry about making
mistakes! We can always revert a change. Brian West and I review all wiki
edits each day. Standard users can't accidentally delete a page or
permanently remove content. You're totally safe to make edits. By the way,
when you sign in to the wiki you won't see the ads on the left-hand side of
the page and you won't have to scroll down so far to locate the search box.

There are several ways that you can assist with the wiki cleanup. One way is
to help users like Diego Viola who are quite literally going page to page
looking for spelling errors, grammar, etc. If you see an obvious error
please by all means correct it.

Another thing you can do is visit this page and look at the janitorial
section:
http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_23

There are several areas where we can clean up and improve the wiki:

*Dead end pages - these are pages that are linked to by other wiki pages but
do not themselves have links back to anywhere. At the very least each page
should have a link back to some other "see also" kind of content.

*Double redirects - these are redirect pages that point to other redirect
pages. These need to be pointed to the ultimate destination page.

*Orphaned pages - orphans are pages that are not linked to by other pages.
The only way to access these pages is via a search or if you already know
the URL. We need people to review the orphaned pages and add links to them
from other pages.

*Wanted pages and categories - these are pages and categories that have
links but no content. We need people review them and add content or
redirects as appropriate.

*Uncategorized pages - all pages belong to some category and these pages
haven't been added to a category yet. Categories are handy because they act
as natural indexes. Please review pages that have no categories. If you have
an idea for a category that does not yet exist please email me and let me
know. We don't want to have too many categories.

The best way to learn wiki markup is to click the edit button on a page and
see what wiki markup does. You can change the markup and then click Preview
to see what it doesn. Click the cancel link to discard any changes. Play
around and learn!

Thank you for your help. Please feel free to ask for assistance with editing
the wiki. It is a community resource and the more people who know how to add
content, the better off we all are.

Thanks,
Michael
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From dmitry.bely at gmail.com  Tue Oct 20 10:43:03 2009
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Tue, 20 Oct 2009 21:43:03 +0400
Subject: [Freeswitch-users] Status of ubuntu/debian packages.
In-Reply-To: <4ABA47FB.2050100@gmail.com>
References: <4ABA47FB.2050100@gmail.com>
Message-ID: <90823c940910201043g55f51005kc3b90d6b8c8a4f6f@mail.gmail.com>

William,

Where is mod_skypiax.so? Isn't freeswitch-skypiax package expected to
contain it?

- Dmitry Bely


From msc at freeswitch.org  Tue Oct 20 10:44:10 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 20 Oct 2009 10:44:10 -0700
Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Agenda - Oct
	23rd
Message-ID: <87f2f3b90910201044v31a281d0j1c441425ba993786@mail.gmail.com>

FYI,

The conference call agenda for this week is online:
http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_23

Also, we will be discussing whether or not Friday is the best time to be
doing this call. Just remember that the FS devs have lots of work to do and
they can't always accommodate everyone's schedules. Still, they'll do their
best to be flexible wherever possible.

Thanks for supporting the weekly conference call.
-Michael
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From chris at cloudtel.com  Tue Oct 20 11:23:35 2009
From: chris at cloudtel.com (Chris Burns)
Date: Tue, 20 Oct 2009 14:23:35 -0400
Subject: [Freeswitch-users] Connect PHP SOAP Web Server with SQLite
	database of FS
In-Reply-To: <25976554.post@talk.nabble.com>
References: <25976554.post@talk.nabble.com>
Message-ID: <200910201423.35419.chris@cloudtel.com>

If you really wanted: http://php.net/manual/en/book.sqlite.php

But I would recommend you make use of ODBC to use a client/server RDBMS. 
Here's some good reading: 
http://www.sqlite.org/cvstrac/wiki?p=WhenToUseSqlite

On October 20, 2009 10:53:01 am homqua wrote:
> Now I am building a PHP SOAP Web Service to access the database of FS.
> Anyone has idea about how to access sqlite database of FS through PHP ? I
> have read about socket event in FS, but I don't know whether it can
> response with the query of database or not.
> Thanks for your help.




From msc at freeswitch.org  Tue Oct 20 11:59:48 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 20 Oct 2009 11:59:48 -0700
Subject: [Freeswitch-users] Digium TE220 wiki
In-Reply-To: <7d79b3930910200018s25184c01k838bff05ad308ffb@mail.gmail.com>
References: <7d79b3930910200018s25184c01k838bff05ad308ffb@mail.gmail.com>
Message-ID: <87f2f3b90910201159h1721eqe363e9ed397d634b@mail.gmail.com>

Thanks. I added minor changes, plus a category and see also. Thanks for
adding this content!
-MC

On Tue, Oct 20, 2009 at 12:18 AM, lakshmanan ganapathy <lakindia89 at gmail.com
> wrote:

> Hi all,
>     I've created a wiki page, which contains the example configuration for
> making Digium TE220 to work.
>     I request you people to check this, and give feedbacks.
>
> http://wiki.freeswitch.org/wiki/Configuration_OpenZap-DigiumTE220-Example
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kristian.kielhofner at gmail.com  Tue Oct 20 12:56:15 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Tue, 20 Oct 2009 15:56:15 -0400
Subject: [Freeswitch-users] Troubles with proxy media mode
Message-ID: <2d9149cd0910201256j39b01228ufa34f977787f0c79@mail.gmail.com>

Hello everyone,

  I'm trying to use proxy media across two profiles.  The codec
settings are identical, they both have late negotiation enabled, and
they both have inbound-proxy-media set to true (I also tried setting
proxy_media from the dialplan).

  FreeSWITCH ends up clearing the call with TRANSCODING_NECESSARY but
I can't figure out why it thinks it needs to transcode for this call.
I've attached a level 7 debug and an ngrep siptrace showing traffic
from both profiles.

  FreeSWITCH trunk rev. 15180 running on Debian 5.0.2.

  See anything interesting?

Thanks!

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com
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From anthony.minessale at gmail.com  Tue Oct 20 13:11:37 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 20 Oct 2009 15:11:37 -0500
Subject: [Freeswitch-users] Troubles with proxy media mode
In-Reply-To: <2d9149cd0910201256j39b01228ufa34f977787f0c79@mail.gmail.com>
References: <2d9149cd0910201256j39b01228ufa34f977787f0c79@mail.gmail.com>
Message-ID: <191c3a030910201311t31806073y936fac4c6c4ef9c@mail.gmail.com>

Fixed in 15181 =D 1 revision higher doh =D


On Tue, Oct 20, 2009 at 2:56 PM, Kristian Kielhofner <
kristian.kielhofner at gmail.com> wrote:

> Hello everyone,
>
>  I'm trying to use proxy media across two profiles.  The codec
> settings are identical, they both have late negotiation enabled, and
> they both have inbound-proxy-media set to true (I also tried setting
> proxy_media from the dialplan).
>
>  FreeSWITCH ends up clearing the call with TRANSCODING_NECESSARY but
> I can't figure out why it thinks it needs to transcode for this call.
> I've attached a level 7 debug and an ngrep siptrace showing traffic
> from both profiles.
>
>  FreeSWITCH trunk rev. 15180 running on Debian 5.0.2.
>
>  See anything interesting?
>
> Thanks!
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From tculjaga at gmail.com  Tue Oct 20 13:16:42 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 20 Oct 2009 22:16:42 +0200
Subject: [Freeswitch-users] Troubles with proxy media mode
In-Reply-To: <2d9149cd0910201256j39b01228ufa34f977787f0c79@mail.gmail.com>
References: <2d9149cd0910201256j39b01228ufa34f977787f0c79@mail.gmail.com>
Message-ID: <65d96fc80910201316i3e0c4d73q9bf269176a420045@mail.gmail.com>

you are making FS to play wav file when sending a call in G711 or GSM or
some other codec.

you might use mod_native_file<http://wiki.freeswitch.org/wiki/Mod_native_file>to
avoid transcoding.

T.



On Tue, Oct 20, 2009 at 9:56 PM, Kristian Kielhofner <
kristian.kielhofner at gmail.com> wrote:

> Hello everyone,
>
>  I'm trying to use proxy media across two profiles.  The codec
> settings are identical, they both have late negotiation enabled, and
> they both have inbound-proxy-media set to true (I also tried setting
> proxy_media from the dialplan).
>
>  FreeSWITCH ends up clearing the call with TRANSCODING_NECESSARY but
> I can't figure out why it thinks it needs to transcode for this call.
> I've attached a level 7 debug and an ngrep siptrace showing traffic
> from both profiles.
>
>  FreeSWITCH trunk rev. 15180 running on Debian 5.0.2.
>
>  See anything interesting?
>
> Thanks!
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kristian.kielhofner at gmail.com  Tue Oct 20 13:37:24 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Tue, 20 Oct 2009 16:37:24 -0400
Subject: [Freeswitch-users] Troubles with proxy media mode
In-Reply-To: <191c3a030910201311t31806073y936fac4c6c4ef9c@mail.gmail.com>
References: <2d9149cd0910201256j39b01228ufa34f977787f0c79@mail.gmail.com>
	<191c3a030910201311t31806073y936fac4c6c4ef9c@mail.gmail.com>
Message-ID: <2d9149cd0910201337m22a03d1fr9538cc54fbd22b98@mail.gmail.com>

Updated and now working.

Thanks!

On Tue, Oct 20, 2009 at 4:11 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Fixed in 15181 =D 1 revision higher doh =D
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From kristian.kielhofner at gmail.com  Tue Oct 20 13:57:41 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Tue, 20 Oct 2009 16:57:41 -0400
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
Message-ID: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>

Hello everyone,

  Now that proxy media mode is working again I'm trying to figure out
why T.38 with re-INVITE doesn't...

  Everything goes well until my end tries to re-INVITE to T.38:

U 10.16.5.129:5060 -> 65.196.170.191:5060
INVITE sip:mod_sofia at 65.196.170.191:5060 SIP/2.0.
Via: SIP/2.0/UDP 10.16.5.129:5060;branch=z9hG4bK-2656bddb.
From: <sip:fax_9186 at 10.16.5.129:5060>;tag=9f75a042a3ca1f17i0.
To: "WIRELESS CALLER" <sip:19412848354 at 65.196.170.191>;tag=4eUH874j0eUZF.
Remote-Party-ID: fax_9186 <sip:fax_9186 at 65.196.170.191>;screen=yes;party=called.
Call-ID: 2889ab05-385d-122d-07b2-0014221c0392.
CSeq: 101 INVITE.
Max-Forwards: 70.
Contact: fax_9186 <sip:fax_9186 at 10.16.5.129:5060>.
Expires: 30.
User-Agent: Linksys/SPA3102-5.1.10(GW).
Content-Length: 261.
Content-Type: application/sdp.
.
v=0.
o=- 57670 57670 IN IP4 10.16.5.129.
s=-.
c=IN IP4 10.16.5.129.
t=0 0.
m=image 16454 udptl t38.
a=T38FaxVersion:0.
a=T38MaxBitRate:14400.
a=T38FaxRateManagement:transferredTCF.
a=T38FaxMaxBuffer:200.
a=T38FaxMaxDatagram:200.
a=T38FaxUdpEC:t38UDPRedundancy.


U 65.196.170.191:5060 -> 10.16.5.129:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 10.16.5.129:5060;branch=z9hG4bK-2656bddb.
From: <sip:fax_9186 at 10.16.5.129:5060>;tag=9f75a042a3ca1f17i0.
To: "WIRELESS CALLER" <sip:19412848354 at 65.196.170.191>;tag=4eUH874j0eUZF.
Call-ID: 2889ab05-385d-122d-07b2-0014221c0392.
CSeq: 101 INVITE.
User-Agent: Star2Star Media.
Content-Length: 0.
.

U 65.196.170.191:5060 -> 10.16.5.129:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 10.16.5.129:5060;branch=z9hG4bK-2656bddb.
From: <sip:fax_9186 at 10.16.5.129:5060>;tag=9f75a042a3ca1f17i0.
To: "WIRELESS CALLER" <sip:19412848354 at 65.196.170.191>;tag=4eUH874j0eUZF.
Call-ID: 2889ab05-385d-122d-07b2-0014221c0392.
CSeq: 101 INVITE.
Contact: <sip:mod_sofia at 65.196.170.191:5060>.
User-Agent: Star2Star Media.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, precondition, path, replaces.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 242.
.
v=0.
o=Sonus_UAC 881416742673069109 186840485790971511 IN IP4 65.196.170.129.
s=SIP Media Capabilities.
c=IN IP4 65.196.170.191.
t=0 0.
m=audio 19646 RTP/AVP 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.


U 10.16.5.129:5060 -> 65.196.170.191:5060
ACK sip:mod_sofia at 65.196.170.191:5060 SIP/2.0.
Via: SIP/2.0/UDP 10.16.5.129:5060;branch=z9hG4bK-842177a7.
From: <sip:fax_9186 at 10.16.5.129:5060>;tag=9f75a042a3ca1f17i0.
To: "WIRELESS CALLER" <sip:19412848354 at 65.196.170.191>;tag=4eUH874j0eUZF.
Call-ID: 2889ab05-385d-122d-07b2-0014221c0392.
CSeq: 101 ACK.
Max-Forwards: 70.
Contact: fax_9186 <sip:fax_9186 at 10.16.5.129:5060>.
User-Agent: Linksys/SPA3102-5.1.10(GW).
Content-Length: 0.
.

U 10.16.5.129:5060 -> 65.196.170.191:5060
BYE sip:mod_sofia at 65.196.170.191:5060 SIP/2.0.
Via: SIP/2.0/UDP 10.16.5.129:5060;branch=z9hG4bK-9e6b284b.
From: <sip:fax_9186 at 10.16.5.129:5060>;tag=9f75a042a3ca1f17i0.
To: "WIRELESS CALLER" <sip:19412848354 at 65.196.170.191>;tag=4eUH874j0eUZF.
Call-ID: 2889ab05-385d-122d-07b2-0014221c0392.
CSeq: 102 BYE.
Max-Forwards: 70.
User-Agent: Linksys/SPA3102-5.1.10(GW).
Content-Length: 0.
.

U 65.196.170.191:5060 -> 10.16.5.129:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 10.16.5.129:5060;branch=z9hG4bK-9e6b284b.
From: <sip:fax_9186 at 10.16.5.129:5060>;tag=9f75a042a3ca1f17i0.
To: "WIRELESS CALLER" <sip:19412848354 at 65.196.170.191>;tag=4eUH874j0eUZF.
Call-ID: 2889ab05-385d-122d-07b2-0014221c0392.
CSeq: 102 BYE.
User-Agent: Star2Star Media.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, precondition, path, replaces.
Content-Length: 0.
.

U 65.196.170.191:5080 -> 65.196.170.129:5060
BYE sip:gw+sip0 at 65.196.170.129:5080;transport=udp SIP/2.0.
Via: SIP/2.0/UDP 65.196.170.191:5080;rport;branch=z9hG4bKpv9pFF0S5r9ca.
Route: <sip:65.196.170.129;lr=on;ftag=3D535BNy2tS9e;did=699.54d14392>.
Max-Forwards: 70.
From: <sip:+15673861009 at 65.196.170.129>;tag=p4H9tpXS1Qv8m.
To: "WIRELESS CALLER" <sip:+19412848354 at 65.196.170.129>;tag=3D535BNy2tS9e.
Call-ID: 2d4c8faa-385d-122d-a69b-003018ae0bd3.
CSeq: 121917605 BYE.
Contact: <sip:+15673861009 at 65.196.170.191:5080;transport=udp>.
User-Agent: Star2Star Media.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY.
Supported: timer, precondition, path, replaces.
Reason: Q.850;cause=16;text="NORMAL_CLEARING".
Content-Length: 0.
.

  It's bizarre because the re-INVITE never gets transmitted to the
other leg and FreeSWITCH responds with an SDP that only supports
RFC2833.  Here is the FreeSWITCH level 7 log:

2009-10-20 20:54:15.577191 [NOTICE] switch_channel.c:613 New Channel
sofia/s2s/+19412848354 at 65.196.170.129
[ba5b3890-bdba-11de-b004-d7a35c351f6f]
2009-10-20 20:54:15.577191 [DEBUG] switch_core_state_machine.c:306
(sofia/s2s/+19412848354 at 65.196.170.129) Running State Change CS_NEW
2009-10-20 20:54:15.577191 [DEBUG] sofia.c:3493 Channel
sofia/s2s/+19412848354 at 65.196.170.129 entering state [received][100]
2009-10-20 20:54:15.577191 [DEBUG] sofia.c:3500 Remote SDP:
v=0
o=Sonus_UAC 7412556365376427063 4600708733824463060 IN IP4 65.196.170.129
s=SIP Media Capabilities
c=IN IP4 4.55.0.130
t=0 0
m=audio 26636 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20

2009-10-20 20:54:15.577191 [DEBUG] switch_core_state_machine.c:312
(sofia/s2s/+19412848354 at 65.196.170.129) State NEW
2009-10-20 20:54:15.577191 [DEBUG] sofia.c:3615
(sofia/s2s/+19412848354 at 65.196.170.129) State Change CS_NEW -> CS_INIT
2009-10-20 20:54:15.577191 [DEBUG] switch_core_session.c:985 Send
signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:306
(sofia/s2s/+19412848354 at 65.196.170.129) Running State Change CS_INIT
2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:330
(sofia/s2s/+19412848354 at 65.196.170.129) State INIT
2009-10-20 20:54:15.581191 [DEBUG] mod_sofia.c:83
sofia/s2s/+19412848354 at 65.196.170.129 SOFIA INIT
2009-10-20 20:54:15.581191 [DEBUG] mod_sofia.c:111
(sofia/s2s/+19412848354 at 65.196.170.129) State Change CS_INIT ->
CS_ROUTING
2009-10-20 20:54:15.581191 [DEBUG] switch_core_session.c:985 Send
signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:330
(sofia/s2s/+19412848354 at 65.196.170.129) State INIT going to sleep
2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:306
(sofia/s2s/+19412848354 at 65.196.170.129) Running State Change
CS_ROUTING
2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:333
(sofia/s2s/+19412848354 at 65.196.170.129) State ROUTING
2009-10-20 20:54:15.581191 [DEBUG] mod_sofia.c:130
sofia/s2s/+19412848354 at 65.196.170.129 SOFIA ROUTING
2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:78
sofia/s2s/+19412848354 at 65.196.170.129 Standard ROUTING
2009-10-20 20:54:15.581191 [INFO] mod_dialplan_xml.c:391 Processing
WIRELESS CALLER->fax_9186 in context s2s-in
Dialplan: sofia/s2s/+19412848354 at 65.196.170.129 parsing
[s2s-in->unloop] continue=false
Dialplan: sofia/s2s/+19412848354 at 65.196.170.129 Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/s2s/+19412848354 at 65.196.170.129 Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/s2s/+19412848354 at 65.196.170.129 parsing
[s2s-in->faxpass] continue=false
Dialplan: sofia/s2s/+19412848354 at 65.196.170.129 Absolute Condition [faxpass]
Dialplan: sofia/s2s/+19412848354 at 65.196.170.129 Action
bridge(sofia/rmt/fax_9186%rmt.star2star.com)
2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:114
(sofia/s2s/+19412848354 at 65.196.170.129) State Change CS_ROUTING ->
CS_EXECUTE
2009-10-20 20:54:15.581191 [DEBUG] switch_core_session.c:985 Send
signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:333
(sofia/s2s/+19412848354 at 65.196.170.129) State ROUTING going to sleep
2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:306
(sofia/s2s/+19412848354 at 65.196.170.129) Running State Change
CS_EXECUTE
2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:340
(sofia/s2s/+19412848354 at 65.196.170.129) State EXECUTE
2009-10-20 20:54:15.581191 [DEBUG] mod_sofia.c:173
sofia/s2s/+19412848354 at 65.196.170.129 SOFIA EXECUTE
2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:151
sofia/s2s/+19412848354 at 65.196.170.129 Standard EXECUTE
EXECUTE sofia/s2s/+19412848354 at 65.196.170.129
bridge(sofia/rmt/fax_9186%rmt.star2star.com)
2009-10-20 20:54:15.585206 [NOTICE] switch_channel.c:613 New Channel
sofia/rmt/fax_9186 [ba5baee2-bdba-11de-b004-d7a35c351f6f]
2009-10-20 20:54:15.585206 [DEBUG] mod_sofia.c:2981
(sofia/rmt/fax_9186) State Change CS_NEW -> CS_INIT
2009-10-20 20:54:15.585206 [DEBUG] switch_core_session.c:985 Send
signal sofia/rmt/fax_9186 [BREAK]
2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:306
(sofia/rmt/fax_9186) Running State Change CS_INIT
2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:330
(sofia/rmt/fax_9186) State INIT
2009-10-20 20:54:15.585206 [DEBUG] mod_sofia.c:83 sofia/rmt/fax_9186 SOFIA INIT
2009-10-20 20:54:15.585206 [DEBUG] sofia_glue.c:1328
sofia/rmt/fax_9186 Patched SDP
---
v=0
o=Sonus_UAC 7412556365376427063 4600708733824463060 IN IP4 65.196.170.129
s=SIP Media Capabilities
c=IN IP4 4.55.0.130
t=0 0
m=audio 26636 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20

+++
v=0
o=Sonus_UAC 7412556365376427063 4600708733824463060 IN IP4 65.196.170.129
s=SIP Media Capabilities
c=IN IP4 65.196.170.191
t=0 0
m=audio 11842 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20

2009-10-20 20:54:15.585206 [DEBUG] sofia_glue.c:1761
sip:fax_9186 at 10.16.5.129:5060 Setting proxy route to
sofia/rmt/fax_9186
2009-10-20 20:54:15.585206 [DEBUG] mod_sofia.c:111
(sofia/rmt/fax_9186) State Change CS_INIT -> CS_ROUTING
2009-10-20 20:54:15.585206 [DEBUG] switch_core_session.c:985 Send
signal sofia/rmt/fax_9186 [BREAK]
2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:330
(sofia/rmt/fax_9186) State INIT going to sleep
2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:306
(sofia/rmt/fax_9186) Running State Change CS_ROUTING
2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:333
(sofia/rmt/fax_9186) State ROUTING
2009-10-20 20:54:15.585206 [DEBUG] mod_sofia.c:130 sofia/rmt/fax_9186
SOFIA ROUTING
2009-10-20 20:54:15.585206 [DEBUG] switch_ivr_originate.c:66
(sofia/rmt/fax_9186) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2009-10-20 20:54:15.585206 [DEBUG] switch_core_session.c:985 Send
signal sofia/rmt/fax_9186 [BREAK]
2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:333
(sofia/rmt/fax_9186) State ROUTING going to sleep
2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:306
(sofia/rmt/fax_9186) Running State Change CS_CONSUME_MEDIA
2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:352
(sofia/rmt/fax_9186) State CONSUME_MEDIA
2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:352
(sofia/rmt/fax_9186) State CONSUME_MEDIA going to sleep
2009-10-20 20:54:15.585206 [DEBUG] sofia.c:3493 Channel
sofia/rmt/fax_9186 entering state [calling][0]
2009-10-20 20:54:15.601186 [DEBUG] sofia.c:3493 Channel
sofia/rmt/fax_9186 entering state [proceeding][180]
2009-10-20 20:54:15.601186 [NOTICE] sofia.c:3557 Ring-Ready sofia/rmt/fax_9186!
2009-10-20 20:54:15.606802 [DEBUG] switch_ivr_originate.c:388
sofia/s2s/+19412848354 at 65.196.170.129 receive message [RINGING]
2009-10-20 20:54:15.606802 [NOTICE] mod_sofia.c:1525 Ring-Ready
sofia/s2s/+19412848354 at 65.196.170.129!
2009-10-20 20:54:15.606802 [DEBUG] switch_core_session.c:653 Send
signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:15.606802 [NOTICE] switch_ivr_originate.c:388 Ring
Ready sofia/s2s/+19412848354 at 65.196.170.129!
2009-10-20 20:54:15.609957 [DEBUG] sofia.c:3493 Channel
sofia/s2s/+19412848354 at 65.196.170.129 entering state [early][180]
2009-10-20 20:54:16.169195 [DEBUG] sofia.c:3493 Channel
sofia/rmt/fax_9186 entering state [ready][200]
2009-10-20 20:54:16.169195 [DEBUG] sofia.c:3500 Remote SDP:
v=0
o=- 74979 74979 IN IP4 10.16.5.129
s=-
c=IN IP4 10.16.5.129
t=0 0
m=audio 16458 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

2009-10-20 20:54:16.169195 [DEBUG] switch_channel.c:2105 Send signal
sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:16.169195 [NOTICE] sofia.c:3948 Channel
[sofia/rmt/fax_9186] has been answered
2009-10-20 20:54:16.169195 [DEBUG] sofia_glue.c:2102 Set Codec
sofia/rmt/fax_9186 PROXY/8000 20 ms 160 samples
2009-10-20 20:54:16.175057 [DEBUG] sofia_glue.c:2379 PROXY AUDIO RTP
[sofia/rmt/fax_9186] 65.196.170.191:11842->10.16.5.129:16458 codec: 0
ms: 20
2009-10-20 20:54:16.175057 [DEBUG] switch_rtp.c:1163 Not using a timer
2009-10-20 20:54:16.175057 [DEBUG] sofia.c:3961
sofia/s2s/+19412848354 at 65.196.170.129 receive message [ANSWER]
2009-10-20 20:54:16.175057 [DEBUG] sofia_glue.c:1328
sofia/s2s/+19412848354 at 65.196.170.129 Patched SDP
---
v=0
o=- 74979 74979 IN IP4 10.16.5.129
s=-
c=IN IP4 10.16.5.129
t=0 0
m=audio 16458 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

+++
v=0
o=- 74979 74979 IN IP4 10.16.5.129
s=-
c=IN IP4 65.196.170.191
t=0 0
m=audio 11186 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

2009-10-20 20:54:16.175057 [DEBUG] sofia_glue.c:2102 Set Codec
sofia/s2s/+19412848354 at 65.196.170.129 PROXY/8000 20 ms 160 samples
2009-10-20 20:54:16.177257 [DEBUG] sofia_glue.c:2379 PROXY AUDIO RTP
[sofia/s2s/+19412848354 at 65.196.170.129]
65.196.170.191:11186->4.55.0.130:26636 codec: 0 ms: 20
2009-10-20 20:54:16.177257 [DEBUG] switch_rtp.c:1163 Not using a timer
2009-10-20 20:54:16.177257 [DEBUG] switch_ivr_originate.c:2324
sofia/s2s/+19412848354 at 65.196.170.129 receive message [ANSWER]
2009-10-20 20:54:16.182190 [DEBUG] switch_core_session.c:712 Send
signal sofia/rmt/fax_9186 [BREAK]
2009-10-20 20:54:16.182190 [DEBUG] switch_core_session.c:653 Send
signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:16.182190 [DEBUG] sofia.c:407 sofia/rmt/fax_9186
receive message [DISPLAY]
2009-10-20 20:54:16.182190 [DEBUG] switch_core_session.c:712 Send
signal sofia/rmt/fax_9186 [BREAK]
2009-10-20 20:54:16.182190 [DEBUG] switch_core_session.c:653 Send
signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:16.182190 [NOTICE] sofia.c:3961 Channel
[sofia/s2s/+19412848354 at 65.196.170.129] has been answered
2009-10-20 20:54:16.185443 [DEBUG] switch_channel.c:182
sofia/s2s/+19412848354 at 65.196.170.129 receive message [AUDIO_SYNC]
2009-10-20 20:54:16.185443 [DEBUG] switch_ivr_originate.c:2368
Originate Resulted in Success: [sofia/rmt/fax_9186]
2009-10-20 20:54:16.185443 [DEBUG] switch_channel.c:182
sofia/rmt/fax_9186 receive message [AUDIO_SYNC]
2009-10-20 20:54:16.185443 [DEBUG] switch_channel.c:182
sofia/s2s/+19412848354 at 65.196.170.129 receive message [AUDIO_SYNC]
2009-10-20 20:54:16.185443 [DEBUG] sofia.c:407 sofia/rmt/fax_9186
receive message [DISPLAY]
2009-10-20 20:54:16.185443 [DEBUG] mod_sofia.c:1328 Not sending same
id again "unknown" <fax_9186>
2009-10-20 20:54:16.185443 [DEBUG] switch_ivr_bridge.c:975
sofia/rmt/fax_9186 receive message [BRIDGE]
2009-10-20 20:54:16.185443 [DEBUG] switch_core_session.c:653 Send
signal sofia/rmt/fax_9186 [BREAK]
2009-10-20 20:54:16.185443 [DEBUG] sofia.c:3493 Channel
sofia/s2s/+19412848354 at 65.196.170.129 entering state [completed][200]
2009-10-20 20:54:16.185443 [DEBUG] switch_ivr_bridge.c:982
sofia/s2s/+19412848354 at 65.196.170.129 receive message [BRIDGE]
2009-10-20 20:54:16.185443 [DEBUG] switch_core_session.c:653 Send
signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:16.185443 [DEBUG] switch_ivr_bridge.c:1026
(sofia/rmt/fax_9186) State Change CS_CONSUME_MEDIA ->
CS_EXCHANGE_MEDIA
2009-10-20 20:54:16.185443 [DEBUG] switch_core_session.c:985 Send
signal sofia/rmt/fax_9186 [BREAK]
2009-10-20 20:54:16.185443 [DEBUG] switch_core_state_machine.c:306
(sofia/rmt/fax_9186) Running State Change CS_EXCHANGE_MEDIA
2009-10-20 20:54:16.185443 [DEBUG] switch_core_state_machine.c:343
(sofia/rmt/fax_9186) State EXCHANGE_MEDIA
2009-10-20 20:54:16.185443 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK
2009-10-20 20:54:16.242188 [DEBUG] sofia.c:3493 Channel
sofia/s2s/+19412848354 at 65.196.170.129 entering state [ready][200]
2009-10-20 20:54:16.310188 [DEBUG] switch_core_session.c:712 Send
signal sofia/rmt/fax_9186 [BREAK]
2009-10-20 20:54:16.310188 [DEBUG] switch_core_session.c:712 Send
signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:16.310188 [DEBUG] switch_ivr_bridge.c:122
sofia/rmt/fax_9186 receive message [DISPLAY]
2009-10-20 20:54:16.313084 [DEBUG] switch_ivr_bridge.c:122
sofia/s2s/+19412848354 at 65.196.170.129 receive message [DISPLAY]
2009-10-20 20:54:18.993185 [DEBUG] sofia.c:3493 Channel
sofia/rmt/fax_9186 entering state [received][100]
2009-10-20 20:54:18.993185 [DEBUG] sofia.c:3500 Remote SDP:
v=0
o=- 75318 75318 IN IP4 10.16.5.129
s=-
c=IN IP4 10.16.5.129
t=0 0
m=image 16458 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:200
a=T38FaxUdpEC:t38UDPRedundancy

2009-10-20 20:54:18.993185 [DEBUG] switch_core_session.c:861 Send
signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:18.993185 [DEBUG] switch_ivr_bridge.c:301
sofia/s2s/+19412848354 at 65.196.170.129 receive message [UNBRIDGE]
2009-10-20 20:54:18.993185 [DEBUG] switch_core_session.c:653 Send
signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:18.993185 [DEBUG] sofia.c:3493 Channel
sofia/rmt/fax_9186 entering state [completed][200]
2009-10-20 20:54:19.005192 [DEBUG] sofia.c:3493 Channel
sofia/rmt/fax_9186 entering state [ready][200]
2009-10-20 20:54:19.005192 [NOTICE] sofia.c:328 Hangup
sofia/rmt/fax_9186 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2009-10-20 20:54:19.005192 [DEBUG] switch_channel.c:1896 Send signal
sofia/rmt/fax_9186 [KILL]
2009-10-20 20:54:19.005192 [DEBUG] switch_core_session.c:985 Send
signal sofia/rmt/fax_9186 [BREAK]
2009-10-20 20:54:19.005192 [DEBUG] switch_core_state_machine.c:451
thread mismatch skipping state handler.
2009-10-20 20:54:19.009120 [DEBUG] switch_ivr_bridge.c:528 BRIDGE
THREAD DONE [sofia/rmt/fax_9186]
2009-10-20 20:54:19.009120 [DEBUG] switch_ivr_bridge.c:530 Send signal
sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:343
(sofia/rmt/fax_9186) State EXCHANGE_MEDIA going to sleep
2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:306
(sofia/rmt/fax_9186) Running State Change CS_HANGUP
2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:478
(sofia/rmt/fax_9186) State HANGUP
2009-10-20 20:54:19.009120 [DEBUG] mod_sofia.c:344 Channel
sofia/rmt/fax_9186 hanging up, cause: NORMAL_CLEARING
2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:46
sofia/rmt/fax_9186 Standard HANGUP, cause: NORMAL_CLEARING
2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:478
(sofia/rmt/fax_9186) State HANGUP going to sleep
2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:325
(sofia/rmt/fax_9186) State Change CS_HANGUP -> CS_REPORTING
2009-10-20 20:54:19.009120 [DEBUG] switch_core_session.c:985 Send
signal sofia/rmt/fax_9186 [BREAK]
2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:306
(sofia/rmt/fax_9186) Running State Change CS_REPORTING
2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:569
(sofia/rmt/fax_9186) State REPORTING
2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:53
sofia/rmt/fax_9186 Standard REPORTING, cause: NORMAL_CLEARING
2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:569
(sofia/rmt/fax_9186) State REPORTING going to sleep
2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:319
(sofia/rmt/fax_9186) State Change CS_REPORTING -> CS_DESTROY
2009-10-20 20:54:19.009120 [DEBUG] switch_core_session.c:985 Send
signal sofia/rmt/fax_9186 [BREAK]
2009-10-20 20:54:19.009120 [DEBUG] switch_core_session.c:1122 Session
16 (sofia/rmt/fax_9186) Locked, Waiting on external entities
2009-10-20 20:54:19.110187 [DEBUG] switch_ivr.c:540
sofia/s2s/+19412848354 at 65.196.170.129 Command Execute
playback(local_stream://moh)
EXECUTE sofia/s2s/+19412848354 at 65.196.170.129 playback(local_stream://moh)
2009-10-20 20:54:19.112927 [ERR] switch_core_file.c:116 Invalid file
format [local_stream] for [moh]!
2009-10-20 20:54:19.112927 [DEBUG] switch_ivr_bridge.c:304
sofia/s2s/+19412848354 at 65.196.170.129 receive message [BRIDGE]
2009-10-20 20:54:19.112927 [DEBUG] switch_core_session.c:653 Send
signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:19.112927 [DEBUG] switch_ivr_bridge.c:306 Send signal
sofia/rmt/fax_9186 [BREAK]
2009-10-20 20:54:19.112927 [DEBUG] switch_ivr_bridge.c:502
sofia/s2s/+19412848354 at 65.196.170.129 receive message [UNBRIDGE]
2009-10-20 20:54:19.112927 [DEBUG] switch_core_session.c:653 Send
signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:19.112927 [DEBUG] switch_ivr_bridge.c:528 BRIDGE
THREAD DONE [sofia/s2s/+19412848354 at 65.196.170.129]
2009-10-20 20:54:19.112927 [DEBUG] switch_ivr_bridge.c:530 Send signal
sofia/rmt/fax_9186 [BREAK]
2009-10-20 20:54:19.112927 [NOTICE] switch_core_state_machine.c:179
Hangup sofia/s2s/+19412848354 at 65.196.170.129 [CS_EXECUTE]
[NORMAL_CLEARING]
2009-10-20 20:54:19.112927 [DEBUG] switch_channel.c:1896 Send signal
sofia/s2s/+19412848354 at 65.196.170.129 [KILL]
2009-10-20 20:54:19.112927 [DEBUG] switch_core_session.c:985 Send
signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:478
(sofia/s2s/+19412848354 at 65.196.170.129) State HANGUP
2009-10-20 20:54:19.112927 [DEBUG] mod_sofia.c:344 Channel
sofia/s2s/+19412848354 at 65.196.170.129 hanging up, cause:
NORMAL_CLEARING
2009-10-20 20:54:19.112927 [DEBUG] mod_sofia.c:382 Sending BYE to
sofia/s2s/+19412848354 at 65.196.170.129
2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:46
sofia/s2s/+19412848354 at 65.196.170.129 Standard HANGUP, cause:
NORMAL_CLEARING
2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:478
(sofia/s2s/+19412848354 at 65.196.170.129) State HANGUP going to sleep
2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:340
(sofia/s2s/+19412848354 at 65.196.170.129) State EXECUTE going to sleep
2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:306
(sofia/s2s/+19412848354 at 65.196.170.129) Running State Change CS_HANGUP
2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:456
handler already called, skipping state handler.
2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:325
(sofia/s2s/+19412848354 at 65.196.170.129) State Change CS_HANGUP ->
CS_REPORTING
2009-10-20 20:54:19.112927 [DEBUG] switch_core_session.c:985 Send
signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:306
(sofia/s2s/+19412848354 at 65.196.170.129) Running State Change
CS_REPORTING
2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:569
(sofia/s2s/+19412848354 at 65.196.170.129) State REPORTING
2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:53
sofia/s2s/+19412848354 at 65.196.170.129 Standard REPORTING, cause:
NORMAL_CLEARING
2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:569
(sofia/s2s/+19412848354 at 65.196.170.129) State REPORTING going to sleep
2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:319
(sofia/s2s/+19412848354 at 65.196.170.129) State Change CS_REPORTING ->
CS_DESTROY
2009-10-20 20:54:19.112927 [DEBUG] switch_core_session.c:985 Send
signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:19.112927 [DEBUG] switch_core_session.c:1122 Session
15 (sofia/s2s/+19412848354 at 65.196.170.129) Locked, Waiting on external
entities
2009-10-20 20:54:19.242188 [NOTICE] switch_core_session.c:1140 Session
16 (sofia/rmt/fax_9186) Ended
2009-10-20 20:54:19.242188 [NOTICE] switch_core_session.c:1142 Close
Channel sofia/rmt/fax_9186 [CS_DESTROY]
2009-10-20 20:54:19.242188 [DEBUG] switch_core_state_machine.c:415
(sofia/rmt/fax_9186) Running State Change CS_DESTROY
2009-10-20 20:54:19.245025 [DEBUG] switch_core_state_machine.c:426
(sofia/rmt/fax_9186) State DESTROY
2009-10-20 20:54:19.245025 [DEBUG] mod_sofia.c:261 sofia/rmt/fax_9186
SOFIA DESTROY
2009-10-20 20:54:19.245025 [DEBUG] switch_core_state_machine.c:60
sofia/rmt/fax_9186 Standard DESTROY
2009-10-20 20:54:19.245025 [DEBUG] switch_core_state_machine.c:426
(sofia/rmt/fax_9186) State DESTROY going to sleep
2009-10-20 20:54:19.245025 [NOTICE] switch_core_session.c:1140 Session
15 (sofia/s2s/+19412848354 at 65.196.170.129) Ended
2009-10-20 20:54:19.245025 [NOTICE] switch_core_session.c:1142 Close
Channel sofia/s2s/+19412848354 at 65.196.170.129 [CS_DESTROY]
2009-10-20 20:54:19.245025 [DEBUG] switch_core_state_machine.c:415
(sofia/s2s/+19412848354 at 65.196.170.129) Running State Change
CS_DESTROY
2009-10-20 20:54:19.245025 [DEBUG] switch_core_state_machine.c:426
(sofia/s2s/+19412848354 at 65.196.170.129) State DESTROY
2009-10-20 20:54:19.245025 [DEBUG] mod_sofia.c:261
sofia/s2s/+19412848354 at 65.196.170.129 SOFIA DESTROY
2009-10-20 20:54:19.245025 [DEBUG] switch_core_state_machine.c:60
sofia/s2s/+19412848354 at 65.196.170.129 Standard DESTROY
2009-10-20 20:54:19.245025 [DEBUG] switch_core_state_machine.c:426
(sofia/s2s/+19412848354 at 65.196.170.129) State DESTROY going to sleep


-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From anthony.minessale at gmail.com  Tue Oct 20 14:34:38 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 20 Oct 2009 16:34:38 -0500
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
In-Reply-To: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>
References: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>
Message-ID: <191c3a030910201434q4c6e65e1h58c5f0a82addc2d3@mail.gmail.com>

issue:
console loglevel debug
sofia profile internal siptrace on

and put it on pastebin
http://pastebin.freeswitch.org


On Tue, Oct 20, 2009 at 3:57 PM, Kristian Kielhofner <
kristian.kielhofner at gmail.com> wrote:

> Hello everyone,
>
>  Now that proxy media mode is working again I'm trying to figure out
> why T.38 with re-INVITE doesn't...
>
>  Everything goes well until my end tries to re-INVITE to T.38:
>
> U 10.16.5.129:5060 -> 65.196.170.191:5060
> INVITE sip:mod_sofia at 65.196.170.191:5060 SIP/2.0.
> Via: SIP/2.0/UDP 10.16.5.129:5060;branch=z9hG4bK-2656bddb.
> From: <sip:fax_9186 at 10.16.5.129:5060>;tag=9f75a042a3ca1f17i0.
> To: "WIRELESS CALLER" <sip:19412848354 at 65.196.170.191<sip%3A19412848354 at 65.196.170.191>
> >;tag=4eUH874j0eUZF.
> Remote-Party-ID: fax_9186 <sip:fax_9186 at 65.196.170.191<sip%3Afax_9186 at 65.196.170.191>
> >;screen=yes;party=called.
> Call-ID: 2889ab05-385d-122d-07b2-0014221c0392.
> CSeq: 101 INVITE.
> Max-Forwards: 70.
> Contact: fax_9186 <sip:fax_9186 at 10.16.5.129:5060>.
> Expires: 30.
> User-Agent: Linksys/SPA3102-5.1.10(GW).
> Content-Length: 261.
> Content-Type: application/sdp.
> .
> v=0.
> o=- 57670 57670 IN IP4 10.16.5.129.
> s=-.
> c=IN IP4 10.16.5.129.
> t=0 0.
> m=image 16454 udptl t38.
> a=T38FaxVersion:0.
> a=T38MaxBitRate:14400.
> a=T38FaxRateManagement:transferredTCF.
> a=T38FaxMaxBuffer:200.
> a=T38FaxMaxDatagram:200.
> a=T38FaxUdpEC:t38UDPRedundancy.
>
>
> U 65.196.170.191:5060 -> 10.16.5.129:5060
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP 10.16.5.129:5060;branch=z9hG4bK-2656bddb.
> From: <sip:fax_9186 at 10.16.5.129:5060>;tag=9f75a042a3ca1f17i0.
> To: "WIRELESS CALLER" <sip:19412848354 at 65.196.170.191<sip%3A19412848354 at 65.196.170.191>
> >;tag=4eUH874j0eUZF.
> Call-ID: 2889ab05-385d-122d-07b2-0014221c0392.
> CSeq: 101 INVITE.
> User-Agent: Star2Star Media.
> Content-Length: 0.
> .
>
> U 65.196.170.191:5060 -> 10.16.5.129:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 10.16.5.129:5060;branch=z9hG4bK-2656bddb.
> From: <sip:fax_9186 at 10.16.5.129:5060>;tag=9f75a042a3ca1f17i0.
> To: "WIRELESS CALLER" <sip:19412848354 at 65.196.170.191<sip%3A19412848354 at 65.196.170.191>
> >;tag=4eUH874j0eUZF.
> Call-ID: 2889ab05-385d-122d-07b2-0014221c0392.
> CSeq: 101 INVITE.
> Contact: <sip:mod_sofia at 65.196.170.191:5060>.
> User-Agent: Star2Star Media.
> Accept: application/sdp.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
> REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE.
> Supported: timer, precondition, path, replaces.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 242.
> .
> v=0.
> o=Sonus_UAC 881416742673069109 186840485790971511 IN IP4 65.196.170.129.
> s=SIP Media Capabilities.
> c=IN IP4 65.196.170.191.
> t=0 0.
> m=audio 19646 RTP/AVP 101.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=silenceSupp:off - - - -.
>
>
> U 10.16.5.129:5060 -> 65.196.170.191:5060
> ACK sip:mod_sofia at 65.196.170.191:5060 SIP/2.0.
> Via: SIP/2.0/UDP 10.16.5.129:5060;branch=z9hG4bK-842177a7.
> From: <sip:fax_9186 at 10.16.5.129:5060>;tag=9f75a042a3ca1f17i0.
> To: "WIRELESS CALLER" <sip:19412848354 at 65.196.170.191<sip%3A19412848354 at 65.196.170.191>
> >;tag=4eUH874j0eUZF.
> Call-ID: 2889ab05-385d-122d-07b2-0014221c0392.
> CSeq: 101 ACK.
> Max-Forwards: 70.
> Contact: fax_9186 <sip:fax_9186 at 10.16.5.129:5060>.
> User-Agent: Linksys/SPA3102-5.1.10(GW).
> Content-Length: 0.
> .
>
> U 10.16.5.129:5060 -> 65.196.170.191:5060
> BYE sip:mod_sofia at 65.196.170.191:5060 SIP/2.0.
> Via: SIP/2.0/UDP 10.16.5.129:5060;branch=z9hG4bK-9e6b284b.
> From: <sip:fax_9186 at 10.16.5.129:5060>;tag=9f75a042a3ca1f17i0.
> To: "WIRELESS CALLER" <sip:19412848354 at 65.196.170.191<sip%3A19412848354 at 65.196.170.191>
> >;tag=4eUH874j0eUZF.
> Call-ID: 2889ab05-385d-122d-07b2-0014221c0392.
> CSeq: 102 BYE.
> Max-Forwards: 70.
> User-Agent: Linksys/SPA3102-5.1.10(GW).
> Content-Length: 0.
> .
>
> U 65.196.170.191:5060 -> 10.16.5.129:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 10.16.5.129:5060;branch=z9hG4bK-9e6b284b.
> From: <sip:fax_9186 at 10.16.5.129:5060>;tag=9f75a042a3ca1f17i0.
> To: "WIRELESS CALLER" <sip:19412848354 at 65.196.170.191<sip%3A19412848354 at 65.196.170.191>
> >;tag=4eUH874j0eUZF.
> Call-ID: 2889ab05-385d-122d-07b2-0014221c0392.
> CSeq: 102 BYE.
> User-Agent: Star2Star Media.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
> REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE.
> Supported: timer, precondition, path, replaces.
> Content-Length: 0.
> .
>
> U 65.196.170.191:5080 -> 65.196.170.129:5060
> BYE sip:gw+sip0 at 65.196.170.129:5080;transport=udp SIP/2.0.
> Via: SIP/2.0/UDP 65.196.170.191:5080;rport;branch=z9hG4bKpv9pFF0S5r9ca.
> Route: <sip:65.196.170.129;lr=on;ftag=3D535BNy2tS9e;did=699.54d14392>.
> Max-Forwards: 70.
> From: <sip:+15673861009 at 65.196.170.129<sip%3A%2B15673861009 at 65.196.170.129>
> >;tag=p4H9tpXS1Qv8m.
> To: "WIRELESS CALLER" <sip:+19412848354 at 65.196.170.129<sip%3A%2B19412848354 at 65.196.170.129>
> >;tag=3D535BNy2tS9e.
> Call-ID: 2d4c8faa-385d-122d-a69b-003018ae0bd3.
> CSeq: 121917605 BYE.
> Contact: <sip:+15673861009 at 65.196.170.191:5080;transport=udp>.
> User-Agent: Star2Star Media.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
> REFER, UPDATE, NOTIFY.
> Supported: timer, precondition, path, replaces.
> Reason: Q.850;cause=16;text="NORMAL_CLEARING".
> Content-Length: 0.
> .
>
>  It's bizarre because the re-INVITE never gets transmitted to the
> other leg and FreeSWITCH responds with an SDP that only supports
> RFC2833.  Here is the FreeSWITCH level 7 log:
>
> 2009-10-20 20:54:15.577191 [NOTICE] switch_channel.c:613 New Channel
> sofia/s2s/+19412848354 at 65.196.170.129
> [ba5b3890-bdba-11de-b004-d7a35c351f6f]
> 2009-10-20 20:54:15.577191 [DEBUG] switch_core_state_machine.c:306
> (sofia/s2s/+19412848354 at 65.196.170.129) Running State Change CS_NEW
> 2009-10-20 20:54:15.577191 [DEBUG] sofia.c:3493 Channel
> sofia/s2s/+19412848354 at 65.196.170.129 entering state [received][100]
> 2009-10-20 20:54:15.577191 [DEBUG] sofia.c:3500 Remote SDP:
> v=0
> o=Sonus_UAC 7412556365376427063 4600708733824463060 IN IP4 65.196.170.129
> s=SIP Media Capabilities
> c=IN IP4 4.55.0.130
> t=0 0
> m=audio 26636 RTP/AVP 0 8 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=maxptime:20
>
> 2009-10-20 20:54:15.577191 [DEBUG] switch_core_state_machine.c:312
> (sofia/s2s/+19412848354 at 65.196.170.129) State NEW
> 2009-10-20 20:54:15.577191 [DEBUG] sofia.c:3615
> (sofia/s2s/+19412848354 at 65.196.170.129) State Change CS_NEW -> CS_INIT
> 2009-10-20 20:54:15.577191 [DEBUG] switch_core_session.c:985 Send
> signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:306
> (sofia/s2s/+19412848354 at 65.196.170.129) Running State Change CS_INIT
> 2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:330
> (sofia/s2s/+19412848354 at 65.196.170.129) State INIT
> 2009-10-20 20:54:15.581191 [DEBUG] mod_sofia.c:83
> sofia/s2s/+19412848354 at 65.196.170.129 SOFIA INIT
> 2009-10-20 20:54:15.581191 [DEBUG] mod_sofia.c:111
> (sofia/s2s/+19412848354 at 65.196.170.129) State Change CS_INIT ->
> CS_ROUTING
> 2009-10-20 20:54:15.581191 [DEBUG] switch_core_session.c:985 Send
> signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:330
> (sofia/s2s/+19412848354 at 65.196.170.129) State INIT going to sleep
> 2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:306
> (sofia/s2s/+19412848354 at 65.196.170.129) Running State Change
> CS_ROUTING
> 2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:333
> (sofia/s2s/+19412848354 at 65.196.170.129) State ROUTING
> 2009-10-20 20:54:15.581191 [DEBUG] mod_sofia.c:130
> sofia/s2s/+19412848354 at 65.196.170.129 SOFIA ROUTING
> 2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:78
> sofia/s2s/+19412848354 at 65.196.170.129 Standard ROUTING
> 2009-10-20 20:54:15.581191 [INFO] mod_dialplan_xml.c:391 Processing
> WIRELESS CALLER->fax_9186 in context s2s-in
> Dialplan: sofia/s2s/+19412848354 at 65.196.170.129 parsing
> [s2s-in->unloop] continue=false
> Dialplan: sofia/s2s/+19412848354 at 65.196.170.129 Regex (PASS) [unloop]
> ${unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/s2s/+19412848354 at 65.196.170.129 Regex (FAIL) [unloop]
> ${sip_looped_call}() =~ /^true$/ break=on-false
> Dialplan: sofia/s2s/+19412848354 at 65.196.170.129 parsing
> [s2s-in->faxpass] continue=false
> Dialplan: sofia/s2s/+19412848354 at 65.196.170.129 Absolute Condition
> [faxpass]
> Dialplan: sofia/s2s/+19412848354 at 65.196.170.129 Action
> bridge(sofia/rmt/fax_9186%rmt.star2star.com)
> 2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:114
> (sofia/s2s/+19412848354 at 65.196.170.129) State Change CS_ROUTING ->
> CS_EXECUTE
> 2009-10-20 20:54:15.581191 [DEBUG] switch_core_session.c:985 Send
> signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:333
> (sofia/s2s/+19412848354 at 65.196.170.129) State ROUTING going to sleep
> 2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:306
> (sofia/s2s/+19412848354 at 65.196.170.129) Running State Change
> CS_EXECUTE
> 2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:340
> (sofia/s2s/+19412848354 at 65.196.170.129) State EXECUTE
> 2009-10-20 20:54:15.581191 [DEBUG] mod_sofia.c:173
> sofia/s2s/+19412848354 at 65.196.170.129 SOFIA EXECUTE
> 2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:151
> sofia/s2s/+19412848354 at 65.196.170.129 Standard EXECUTE
> EXECUTE sofia/s2s/+19412848354 at 65.196.170.129
> bridge(sofia/rmt/fax_9186%rmt.star2star.com)
> 2009-10-20 20:54:15.585206 [NOTICE] switch_channel.c:613 New Channel
> sofia/rmt/fax_9186 [ba5baee2-bdba-11de-b004-d7a35c351f6f]
> 2009-10-20 20:54:15.585206 [DEBUG] mod_sofia.c:2981
> (sofia/rmt/fax_9186) State Change CS_NEW -> CS_INIT
> 2009-10-20 20:54:15.585206 [DEBUG] switch_core_session.c:985 Send
> signal sofia/rmt/fax_9186 [BREAK]
> 2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:306
> (sofia/rmt/fax_9186) Running State Change CS_INIT
> 2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:330
> (sofia/rmt/fax_9186) State INIT
> 2009-10-20 20:54:15.585206 [DEBUG] mod_sofia.c:83 sofia/rmt/fax_9186 SOFIA
> INIT
> 2009-10-20 20:54:15.585206 [DEBUG] sofia_glue.c:1328
> sofia/rmt/fax_9186 Patched SDP
> ---
> v=0
> o=Sonus_UAC 7412556365376427063 4600708733824463060 IN IP4 65.196.170.129
> s=SIP Media Capabilities
> c=IN IP4 4.55.0.130
> t=0 0
> m=audio 26636 RTP/AVP 0 8 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=maxptime:20
>
> +++
> v=0
> o=Sonus_UAC 7412556365376427063 4600708733824463060 IN IP4 65.196.170.129
> s=SIP Media Capabilities
> c=IN IP4 65.196.170.191
> t=0 0
> m=audio 11842 RTP/AVP 0 8 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=maxptime:20
>
> 2009-10-20 20:54:15.585206 [DEBUG] sofia_glue.c:1761
> sip:fax_9186 at 10.16.5.129:5060 Setting proxy route to
> sofia/rmt/fax_9186
> 2009-10-20 20:54:15.585206 [DEBUG] mod_sofia.c:111
> (sofia/rmt/fax_9186) State Change CS_INIT -> CS_ROUTING
> 2009-10-20 20:54:15.585206 [DEBUG] switch_core_session.c:985 Send
> signal sofia/rmt/fax_9186 [BREAK]
> 2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:330
> (sofia/rmt/fax_9186) State INIT going to sleep
> 2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:306
> (sofia/rmt/fax_9186) Running State Change CS_ROUTING
> 2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:333
> (sofia/rmt/fax_9186) State ROUTING
> 2009-10-20 20:54:15.585206 [DEBUG] mod_sofia.c:130 sofia/rmt/fax_9186
> SOFIA ROUTING
> 2009-10-20 20:54:15.585206 [DEBUG] switch_ivr_originate.c:66
> (sofia/rmt/fax_9186) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2009-10-20 20:54:15.585206 [DEBUG] switch_core_session.c:985 Send
> signal sofia/rmt/fax_9186 [BREAK]
> 2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:333
> (sofia/rmt/fax_9186) State ROUTING going to sleep
> 2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:306
> (sofia/rmt/fax_9186) Running State Change CS_CONSUME_MEDIA
> 2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:352
> (sofia/rmt/fax_9186) State CONSUME_MEDIA
> 2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:352
> (sofia/rmt/fax_9186) State CONSUME_MEDIA going to sleep
> 2009-10-20 20:54:15.585206 [DEBUG] sofia.c:3493 Channel
> sofia/rmt/fax_9186 entering state [calling][0]
> 2009-10-20 20:54:15.601186 [DEBUG] sofia.c:3493 Channel
> sofia/rmt/fax_9186 entering state [proceeding][180]
> 2009-10-20 20:54:15.601186 [NOTICE] sofia.c:3557 Ring-Ready
> sofia/rmt/fax_9186!
> 2009-10-20 20:54:15.606802 [DEBUG] switch_ivr_originate.c:388
> sofia/s2s/+19412848354 at 65.196.170.129 receive message [RINGING]
> 2009-10-20 20:54:15.606802 [NOTICE] mod_sofia.c:1525 Ring-Ready
> sofia/s2s/+19412848354 at 65.196.170.129!
> 2009-10-20 20:54:15.606802 [DEBUG] switch_core_session.c:653 Send
> signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:15.606802 [NOTICE] switch_ivr_originate.c:388 Ring
> Ready sofia/s2s/+19412848354 at 65.196.170.129!
> 2009-10-20 20:54:15.609957 [DEBUG] sofia.c:3493 Channel
> sofia/s2s/+19412848354 at 65.196.170.129 entering state [early][180]
> 2009-10-20 20:54:16.169195 [DEBUG] sofia.c:3493 Channel
> sofia/rmt/fax_9186 entering state [ready][200]
> 2009-10-20 20:54:16.169195 [DEBUG] sofia.c:3500 Remote SDP:
> v=0
> o=- 74979 74979 IN IP4 10.16.5.129
> s=-
> c=IN IP4 10.16.5.129
> t=0 0
> m=audio 16458 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
>
> 2009-10-20 20:54:16.169195 [DEBUG] switch_channel.c:2105 Send signal
> sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:16.169195 [NOTICE] sofia.c:3948 Channel
> [sofia/rmt/fax_9186] has been answered
> 2009-10-20 20:54:16.169195 [DEBUG] sofia_glue.c:2102 Set Codec
> sofia/rmt/fax_9186 PROXY/8000 20 ms 160 samples
> 2009-10-20 20:54:16.175057 [DEBUG] sofia_glue.c:2379 PROXY AUDIO RTP
> [sofia/rmt/fax_9186] 65.196.170.191:11842->10.16.5.129:16458 codec: 0
> ms: 20
> 2009-10-20 20:54:16.175057 [DEBUG] switch_rtp.c:1163 Not using a timer
> 2009-10-20 20:54:16.175057 [DEBUG] sofia.c:3961
> sofia/s2s/+19412848354 at 65.196.170.129 receive message [ANSWER]
> 2009-10-20 20:54:16.175057 [DEBUG] sofia_glue.c:1328
> sofia/s2s/+19412848354 at 65.196.170.129 Patched SDP
> ---
> v=0
> o=- 74979 74979 IN IP4 10.16.5.129
> s=-
> c=IN IP4 10.16.5.129
> t=0 0
> m=audio 16458 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
>
> +++
> v=0
> o=- 74979 74979 IN IP4 10.16.5.129
> s=-
> c=IN IP4 65.196.170.191
> t=0 0
> m=audio 11186 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
>
> 2009-10-20 20:54:16.175057 [DEBUG] sofia_glue.c:2102 Set Codec
> sofia/s2s/+19412848354 at 65.196.170.129 PROXY/8000 20 ms 160 samples
> 2009-10-20 20:54:16.177257 [DEBUG] sofia_glue.c:2379 PROXY AUDIO RTP
> [sofia/s2s/+19412848354 at 65.196.170.129]
> 65.196.170.191:11186->4.55.0.130:26636 codec: 0 ms: 20
> 2009-10-20 20:54:16.177257 [DEBUG] switch_rtp.c:1163 Not using a timer
> 2009-10-20 20:54:16.177257 [DEBUG] switch_ivr_originate.c:2324
> sofia/s2s/+19412848354 at 65.196.170.129 receive message [ANSWER]
> 2009-10-20 20:54:16.182190 [DEBUG] switch_core_session.c:712 Send
> signal sofia/rmt/fax_9186 [BREAK]
> 2009-10-20 20:54:16.182190 [DEBUG] switch_core_session.c:653 Send
> signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:16.182190 [DEBUG] sofia.c:407 sofia/rmt/fax_9186
> receive message [DISPLAY]
> 2009-10-20 20:54:16.182190 [DEBUG] switch_core_session.c:712 Send
> signal sofia/rmt/fax_9186 [BREAK]
> 2009-10-20 20:54:16.182190 [DEBUG] switch_core_session.c:653 Send
> signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:16.182190 [NOTICE] sofia.c:3961 Channel
> [sofia/s2s/+19412848354 at 65.196.170.129] has been answered
> 2009-10-20 20:54:16.185443 [DEBUG] switch_channel.c:182
> sofia/s2s/+19412848354 at 65.196.170.129 receive message [AUDIO_SYNC]
> 2009-10-20 20:54:16.185443 [DEBUG] switch_ivr_originate.c:2368
> Originate Resulted in Success: [sofia/rmt/fax_9186]
> 2009-10-20 20:54:16.185443 [DEBUG] switch_channel.c:182
> sofia/rmt/fax_9186 receive message [AUDIO_SYNC]
> 2009-10-20 20:54:16.185443 [DEBUG] switch_channel.c:182
> sofia/s2s/+19412848354 at 65.196.170.129 receive message [AUDIO_SYNC]
> 2009-10-20 20:54:16.185443 [DEBUG] sofia.c:407 sofia/rmt/fax_9186
> receive message [DISPLAY]
> 2009-10-20 20:54:16.185443 [DEBUG] mod_sofia.c:1328 Not sending same
> id again "unknown" <fax_9186>
> 2009-10-20 20:54:16.185443 [DEBUG] switch_ivr_bridge.c:975
> sofia/rmt/fax_9186 receive message [BRIDGE]
> 2009-10-20 20:54:16.185443 [DEBUG] switch_core_session.c:653 Send
> signal sofia/rmt/fax_9186 [BREAK]
> 2009-10-20 20:54:16.185443 [DEBUG] sofia.c:3493 Channel
> sofia/s2s/+19412848354 at 65.196.170.129 entering state [completed][200]
> 2009-10-20 20:54:16.185443 [DEBUG] switch_ivr_bridge.c:982
> sofia/s2s/+19412848354 at 65.196.170.129 receive message [BRIDGE]
> 2009-10-20 20:54:16.185443 [DEBUG] switch_core_session.c:653 Send
> signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:16.185443 [DEBUG] switch_ivr_bridge.c:1026
> (sofia/rmt/fax_9186) State Change CS_CONSUME_MEDIA ->
> CS_EXCHANGE_MEDIA
> 2009-10-20 20:54:16.185443 [DEBUG] switch_core_session.c:985 Send
> signal sofia/rmt/fax_9186 [BREAK]
> 2009-10-20 20:54:16.185443 [DEBUG] switch_core_state_machine.c:306
> (sofia/rmt/fax_9186) Running State Change CS_EXCHANGE_MEDIA
> 2009-10-20 20:54:16.185443 [DEBUG] switch_core_state_machine.c:343
> (sofia/rmt/fax_9186) State EXCHANGE_MEDIA
> 2009-10-20 20:54:16.185443 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK
> 2009-10-20 20:54:16.242188 [DEBUG] sofia.c:3493 Channel
> sofia/s2s/+19412848354 at 65.196.170.129 entering state [ready][200]
> 2009-10-20 20:54:16.310188 [DEBUG] switch_core_session.c:712 Send
> signal sofia/rmt/fax_9186 [BREAK]
> 2009-10-20 20:54:16.310188 [DEBUG] switch_core_session.c:712 Send
> signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:16.310188 [DEBUG] switch_ivr_bridge.c:122
> sofia/rmt/fax_9186 receive message [DISPLAY]
> 2009-10-20 20:54:16.313084 [DEBUG] switch_ivr_bridge.c:122
> sofia/s2s/+19412848354 at 65.196.170.129 receive message [DISPLAY]
> 2009-10-20 20:54:18.993185 [DEBUG] sofia.c:3493 Channel
> sofia/rmt/fax_9186 entering state [received][100]
> 2009-10-20 20:54:18.993185 [DEBUG] sofia.c:3500 Remote SDP:
> v=0
> o=- 75318 75318 IN IP4 10.16.5.129
> s=-
> c=IN IP4 10.16.5.129
> t=0 0
> m=image 16458 udptl t38
> a=T38FaxVersion:0
> a=T38MaxBitRate:14400
> a=T38FaxRateManagement:transferredTCF
> a=T38FaxMaxBuffer:200
> a=T38FaxMaxDatagram:200
> a=T38FaxUdpEC:t38UDPRedundancy
>
> 2009-10-20 20:54:18.993185 [DEBUG] switch_core_session.c:861 Send
> signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:18.993185 [DEBUG] switch_ivr_bridge.c:301
> sofia/s2s/+19412848354 at 65.196.170.129 receive message [UNBRIDGE]
> 2009-10-20 20:54:18.993185 [DEBUG] switch_core_session.c:653 Send
> signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:18.993185 [DEBUG] sofia.c:3493 Channel
> sofia/rmt/fax_9186 entering state [completed][200]
> 2009-10-20 20:54:19.005192 [DEBUG] sofia.c:3493 Channel
> sofia/rmt/fax_9186 entering state [ready][200]
> 2009-10-20 20:54:19.005192 [NOTICE] sofia.c:328 Hangup
> sofia/rmt/fax_9186 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
> 2009-10-20 20:54:19.005192 [DEBUG] switch_channel.c:1896 Send signal
> sofia/rmt/fax_9186 [KILL]
> 2009-10-20 20:54:19.005192 [DEBUG] switch_core_session.c:985 Send
> signal sofia/rmt/fax_9186 [BREAK]
> 2009-10-20 20:54:19.005192 [DEBUG] switch_core_state_machine.c:451
> thread mismatch skipping state handler.
> 2009-10-20 20:54:19.009120 [DEBUG] switch_ivr_bridge.c:528 BRIDGE
> THREAD DONE [sofia/rmt/fax_9186]
> 2009-10-20 20:54:19.009120 [DEBUG] switch_ivr_bridge.c:530 Send signal
> sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:343
> (sofia/rmt/fax_9186) State EXCHANGE_MEDIA going to sleep
> 2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:306
> (sofia/rmt/fax_9186) Running State Change CS_HANGUP
> 2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:478
> (sofia/rmt/fax_9186) State HANGUP
> 2009-10-20 20:54:19.009120 [DEBUG] mod_sofia.c:344 Channel
> sofia/rmt/fax_9186 hanging up, cause: NORMAL_CLEARING
> 2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:46
> sofia/rmt/fax_9186 Standard HANGUP, cause: NORMAL_CLEARING
> 2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:478
> (sofia/rmt/fax_9186) State HANGUP going to sleep
> 2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:325
> (sofia/rmt/fax_9186) State Change CS_HANGUP -> CS_REPORTING
> 2009-10-20 20:54:19.009120 [DEBUG] switch_core_session.c:985 Send
> signal sofia/rmt/fax_9186 [BREAK]
> 2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:306
> (sofia/rmt/fax_9186) Running State Change CS_REPORTING
> 2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:569
> (sofia/rmt/fax_9186) State REPORTING
> 2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:53
> sofia/rmt/fax_9186 Standard REPORTING, cause: NORMAL_CLEARING
> 2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:569
> (sofia/rmt/fax_9186) State REPORTING going to sleep
> 2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:319
> (sofia/rmt/fax_9186) State Change CS_REPORTING -> CS_DESTROY
> 2009-10-20 20:54:19.009120 [DEBUG] switch_core_session.c:985 Send
> signal sofia/rmt/fax_9186 [BREAK]
> 2009-10-20 20:54:19.009120 [DEBUG] switch_core_session.c:1122 Session
> 16 (sofia/rmt/fax_9186) Locked, Waiting on external entities
> 2009-10-20 20:54:19.110187 [DEBUG] switch_ivr.c:540
> sofia/s2s/+19412848354 at 65.196.170.129 Command Execute
> playback(local_stream://moh)
> EXECUTE sofia/s2s/+19412848354 at 65.196.170.129 playback(local_stream://moh)
> 2009-10-20 20:54:19.112927 [ERR] switch_core_file.c:116 Invalid file
> format [local_stream] for [moh]!
> 2009-10-20 20:54:19.112927 [DEBUG] switch_ivr_bridge.c:304
> sofia/s2s/+19412848354 at 65.196.170.129 receive message [BRIDGE]
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_session.c:653 Send
> signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:19.112927 [DEBUG] switch_ivr_bridge.c:306 Send signal
> sofia/rmt/fax_9186 [BREAK]
> 2009-10-20 20:54:19.112927 [DEBUG] switch_ivr_bridge.c:502
> sofia/s2s/+19412848354 at 65.196.170.129 receive message [UNBRIDGE]
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_session.c:653 Send
> signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:19.112927 [DEBUG] switch_ivr_bridge.c:528 BRIDGE
> THREAD DONE [sofia/s2s/+19412848354 at 65.196.170.129]
> 2009-10-20 20:54:19.112927 [DEBUG] switch_ivr_bridge.c:530 Send signal
> sofia/rmt/fax_9186 [BREAK]
> 2009-10-20 20:54:19.112927 [NOTICE] switch_core_state_machine.c:179
> Hangup sofia/s2s/+19412848354 at 65.196.170.129 [CS_EXECUTE]
> [NORMAL_CLEARING]
> 2009-10-20 20:54:19.112927 [DEBUG] switch_channel.c:1896 Send signal
> sofia/s2s/+19412848354 at 65.196.170.129 [KILL]
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_session.c:985 Send
> signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:478
> (sofia/s2s/+19412848354 at 65.196.170.129) State HANGUP
> 2009-10-20 20:54:19.112927 [DEBUG] mod_sofia.c:344 Channel
> sofia/s2s/+19412848354 at 65.196.170.129 hanging up, cause:
> NORMAL_CLEARING
> 2009-10-20 20:54:19.112927 [DEBUG] mod_sofia.c:382 Sending BYE to
> sofia/s2s/+19412848354 at 65.196.170.129
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:46
> sofia/s2s/+19412848354 at 65.196.170.129 Standard HANGUP, cause:
> NORMAL_CLEARING
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:478
> (sofia/s2s/+19412848354 at 65.196.170.129) State HANGUP going to sleep
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:340
> (sofia/s2s/+19412848354 at 65.196.170.129) State EXECUTE going to sleep
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:306
> (sofia/s2s/+19412848354 at 65.196.170.129) Running State Change CS_HANGUP
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:456
> handler already called, skipping state handler.
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:325
> (sofia/s2s/+19412848354 at 65.196.170.129) State Change CS_HANGUP ->
> CS_REPORTING
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_session.c:985 Send
> signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:306
> (sofia/s2s/+19412848354 at 65.196.170.129) Running State Change
> CS_REPORTING
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:569
> (sofia/s2s/+19412848354 at 65.196.170.129) State REPORTING
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:53
> sofia/s2s/+19412848354 at 65.196.170.129 Standard REPORTING, cause:
> NORMAL_CLEARING
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:569
> (sofia/s2s/+19412848354 at 65.196.170.129) State REPORTING going to sleep
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:319
> (sofia/s2s/+19412848354 at 65.196.170.129) State Change CS_REPORTING ->
> CS_DESTROY
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_session.c:985 Send
> signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_session.c:1122 Session
> 15 (sofia/s2s/+19412848354 at 65.196.170.129) Locked, Waiting on external
> entities
> 2009-10-20 20:54:19.242188 [NOTICE] switch_core_session.c:1140 Session
> 16 (sofia/rmt/fax_9186) Ended
> 2009-10-20 20:54:19.242188 [NOTICE] switch_core_session.c:1142 Close
> Channel sofia/rmt/fax_9186 [CS_DESTROY]
> 2009-10-20 20:54:19.242188 [DEBUG] switch_core_state_machine.c:415
> (sofia/rmt/fax_9186) Running State Change CS_DESTROY
> 2009-10-20 20:54:19.245025 [DEBUG] switch_core_state_machine.c:426
> (sofia/rmt/fax_9186) State DESTROY
> 2009-10-20 20:54:19.245025 [DEBUG] mod_sofia.c:261 sofia/rmt/fax_9186
> SOFIA DESTROY
> 2009-10-20 20:54:19.245025 [DEBUG] switch_core_state_machine.c:60
> sofia/rmt/fax_9186 Standard DESTROY
> 2009-10-20 20:54:19.245025 [DEBUG] switch_core_state_machine.c:426
> (sofia/rmt/fax_9186) State DESTROY going to sleep
> 2009-10-20 20:54:19.245025 [NOTICE] switch_core_session.c:1140 Session
> 15 (sofia/s2s/+19412848354 at 65.196.170.129) Ended
> 2009-10-20 20:54:19.245025 [NOTICE] switch_core_session.c:1142 Close
> Channel sofia/s2s/+19412848354 at 65.196.170.129 [CS_DESTROY]
> 2009-10-20 20:54:19.245025 [DEBUG] switch_core_state_machine.c:415
> (sofia/s2s/+19412848354 at 65.196.170.129) Running State Change
> CS_DESTROY
> 2009-10-20 20:54:19.245025 [DEBUG] switch_core_state_machine.c:426
> (sofia/s2s/+19412848354 at 65.196.170.129) State DESTROY
> 2009-10-20 20:54:19.245025 [DEBUG] mod_sofia.c:261
> sofia/s2s/+19412848354 at 65.196.170.129 SOFIA DESTROY
> 2009-10-20 20:54:19.245025 [DEBUG] switch_core_state_machine.c:60
> sofia/s2s/+19412848354 at 65.196.170.129 Standard DESTROY
> 2009-10-20 20:54:19.245025 [DEBUG] switch_core_state_machine.c:426
> (sofia/s2s/+19412848354 at 65.196.170.129) State DESTROY going to sleep
>
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From brian at freeswitch.org  Tue Oct 20 14:35:17 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 20 Oct 2009 16:35:17 -0500
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
In-Reply-To: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>
References: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>
Message-ID: <6F7186AC-AF0F-4594-8555-7071A758AF39@freeswitch.org>

Fix this or set it to silence.

/b

On Oct 20, 2009, at 3:57 PM, Kristian Kielhofner wrote:

> EXECUTE sofia/s2s/+19412848354 at 65.196.170.129 playback 
> (local_stream://moh)
> 2009-10-20 20:54:19.112927 [ERR] switch_core_file.c:116 Invalid file
> format [local_stream] for [moh]!



From mcampbellsmith at gmail.com  Tue Oct 20 16:07:32 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Wed, 21 Oct 2009 10:07:32 +1100
Subject: [Freeswitch-users] 2 voicemail questions
In-Reply-To: <33c87fa30907100257w29c628cbnb641d09a75bc22b8@mail.gmail.com>
References: <33c87fa30907100257w29c628cbnb641d09a75bc22b8@mail.gmail.com>
Message-ID: <33c87fa30910201607x16eeaf0ek823084bbf14dbb6c@mail.gmail.com>

1. Can I email the voicemail message to multiple email addresses?

I revisited this again after not requiring it for a while.  A comma
separated list in the extension.xml file does work.  The problem was
the template file.  Once I removed < and > in the To: field, I can
send to multple emails no problems:

In 1000.xml (for example)
      <param name="vm-mailto" value="email at number1.com,email at number2.com"/>

And then I editied voicemail.tpl and changed the second line from:
To: <${voicemail_email}>

TO:
To: ${voicemail_email}

Not sure if this is a fault in the mail server or not.  Maybe the
default templates should be changed to handle this?


On Fri, Jul 10, 2009 at 8:57 PM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> Hi!
>>> 1. Can I email the voicemail message to multiple email addresses?
> A comma separated list does not work in the extensions.xml file
> (1000.xml), but it does work if I hard code the email addresses into
> the notify-voicemail.tpl file.
>
> Could this be added to the switch so that it can handle comma separated lists?
>
>>> 2. How can I make Freeswitch dial a number AFTER a voicemail is left?
>
>>api Hangup hook?
>
> i want the 'voicemail' application to appear to call the extension to
> notify the user that there is a waiting message. ?This is an extract
> from my dialplan.xml:
>
> ? ? ?<condition field="destination_number" expression="^(10[01][0-9])$">
> ? ? ? ?<action application="set" data="dialed_extension=$1"/>
> ? ? ? ?<action application="export" data="dialed_extension=$1"/>
> ? ? ? ?<!-- <action application="set" data="ringback=${us-ring}"/> -->
> ? ? ? ?<action application="set" data="call_timeout=30"/>
> ? ? ? ?<action application="set" data="hangup_after_bridge=true"/>
> ? ? ? ?<action application="set" data="continue_on_fail=true"/>
> ? ? ? ?<action application="bridge"
> data="user/${dialed_extension}@${domain_name}"/>
> ? ? ? ?<action application="answer"/>
> ? ? ? ?<action application="sleep" data="1000"/>
> ? ? ? ?<action application="voicemail" data="default ${domain_name}
> ${dialed_extension}"/>
> ? ? ?</condition>
> ? ? ?<condition
> field="${vm_boxcount(${destination_number}@${domain_name})}"
> expression="^(1)$">
> ? ? ? ? <action application="log" data="MSGBOX
> ${vm_boxcount(${dialed_extension}@${domain_name})}"/>
> ? ? ? ?<action application="set" data="api_hangup_hook=originate
> sofia/internal_nat/${dialed_etension}%${domain_name} default default
> Message 4000 4000 3"/>
>
> This only works if the B leg (ie voicemail application) hangs up
> first. ?This would be an unusual situation and does not achieve what I
> want... is there any other way to achieve this?
>
> Thanks
>
>> Hi!
>>
>> I have 2 questions regarding voicemail ...
>>
>> 1. Can I email the voicemail message to multiple email addresses? ?If
>> so, what format is this in?
>> ? ? ?<param name="vm-mailto" value="me at myemail.com"/>
>
> Try a comma sep. list. ?Not sure if it will work.
>
>>
>> 2. How can I make Freeswitch dial a number AFTER a voicemail is left?
>
> api Hangup hook?
>
> I g
>> From: Brian West <brian at freeswitch.org>
>
>> On Jul 7, 2009, at 9:11 AM, Mark Campbell-Smith wrote:
>
>> Hi!
>>
>> I have 2 questions regarding voicemail ...
>>
>> 1. Can I email the voicemail message to multiple email addresses? ?If
>> so, what format is this in?
>> ? ? ?<param name="vm-mailto" value="me at myemail.com"/>
>
>> Try a comma sep. list. ?Not sure if it will work.
>


From msc at freeswitch.org  Tue Oct 20 17:46:27 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 20 Oct 2009 17:46:27 -0700
Subject: [Freeswitch-users] "Can not record session. Media not enabled
	on channel."
In-Reply-To: <1256034659005-3857858.post@n2.nabble.com>
References: <1256034659005-3857858.post@n2.nabble.com>
Message-ID: <87f2f3b90910201746h43a31b32m5c163fd98373f915@mail.gmail.com>

What's in the dialplan for this channel? Is bypass-media or proxy-media set
to true? Do a debug trace and post it in pastebin.
-MC

On Tue, Oct 20, 2009 at 3:30 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> Hello,
> I am using the same set of extensions for testing the system during
> development, they include XLite, Cisco sip phone and several extensions
> that
> just play some audio file.
> Sometimes, very rarely, this message "Can not record session.  Media not
> enabled on channel." appears on FS console.
> Like I wrote before, I don't change any codec settings and always use the
> same set of devices/emulators.
> What can cause this message?
>
> Thanks,
> Maciej Aniserowicz
>
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From mike at jerris.com  Tue Oct 20 18:48:56 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 20 Oct 2009 21:48:56 -0400
Subject: [Freeswitch-users] CS_REPORTING Channel event state
In-Reply-To: <8ccbff060910200847y5e7ccab6o84d431836970bcb4@mail.gmail.com>
References: <8ccbff060910200847y5e7ccab6o84d431836970bcb4@mail.gmail.com>
Message-ID: <08C886FF-6D2E-46CD-AC7D-2AF37F18B170@jerris.com>

REPORTING is the state that it writes to CDR.  If you have calls stuck  
in this state, take one and try to use uuid_kill on it and see if it  
goes away, then get a core off of it and pastebin the thread apply all  
bt (with no other calls up).  What modules are you using for cdr and  
with what configuration?

Mike

On Oct 20, 2009, at 11:47 AM, Dome Charoenyost wrote:

> Dear All
> What's CS_REPORTING state ?
> I found many channels not hang up  ans state is CS_REPORTING
> e264f84a-bd87-11de-9a90-2320c02172de,outbound,2009-10-20
> 21:50:26,1256050226,sofia/external/xxxxxxxxxxxx at xxx.xxx.xx. 
> 191:7050,CS_REPORTING,FreeSWITCH,xxxxxxxxxxxxxxxxxxxx,xx.xxx.xxx.xxx, 
> 7050,,,XML,public,,,,,



From mike at jerris.com  Tue Oct 20 18:56:32 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 20 Oct 2009 21:56:32 -0400
Subject: [Freeswitch-users] Connect PHP SOAP Web Server with SQLite
	database of FS
In-Reply-To: <200910201423.35419.chris@cloudtel.com>
References: <25976554.post@talk.nabble.com>
	<200910201423.35419.chris@cloudtel.com>
Message-ID: <A5C02801-C96A-411B-A6D1-55D6D28617DA@jerris.com>

If you really want to access this information outside I would strongly  
recommend using odbc instead of the internal sqlite db, it does not  
handle locking contention well.  If you need access to things in the  
core db (like show calls and show channels information) you will need  
to write a small daemon that listens on events socket and puts that  
information into a database.

Mike

On Oct 20, 2009, at 2:23 PM, Chris Burns wrote:

> If you really wanted: http://php.net/manual/en/book.sqlite.php
>
> But I would recommend you make use of ODBC to use a client/server  
> RDBMS.
> Here's some good reading:
> http://www.sqlite.org/cvstrac/wiki?p=WhenToUseSqlite
>
> On October 20, 2009 10:53:01 am homqua wrote:
>> Now I am building a PHP SOAP Web Service to access the database of  
>> FS.
>> Anyone has idea about how to access sqlite database of FS through  
>> PHP ? I
>> have read about socket event in FS, but I don't know whether it can
>> response with the query of database or not.
>> Thanks for your help.
>
>



From kristian.kielhofner at gmail.com  Tue Oct 20 19:37:01 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Tue, 20 Oct 2009 22:37:01 -0400
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
In-Reply-To: <6F7186AC-AF0F-4594-8555-7071A758AF39@freeswitch.org>
References: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>
	<6F7186AC-AF0F-4594-8555-7071A758AF39@freeswitch.org>
Message-ID: <2d9149cd0910201937t4ab718b1yfbeb2a1f3d33f893@mail.gmail.com>

Brian,

  It's already set to silence :).

  My guess here is that Sofia is mis-interpreting the re-INVITE as not
having any media (a la RFC 2543 hold).

  I don't want to place the call on hold.  I want to negotiate T.38 ;).

On Tue, Oct 20, 2009 at 5:35 PM, Brian West <brian at freeswitch.org> wrote:
> Fix this or set it to silence.
>
> /b
>
> On Oct 20, 2009, at 3:57 PM, Kristian Kielhofner wrote:
>
>> EXECUTE sofia/s2s/+19412848354 at 65.196.170.129 playback
>> (local_stream://moh)
>> 2009-10-20 20:54:19.112927 [ERR] switch_core_file.c:116 Invalid file
>> format [local_stream] for [moh]!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From kristian.kielhofner at gmail.com  Tue Oct 20 19:37:50 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Tue, 20 Oct 2009 22:37:50 -0400
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
In-Reply-To: <191c3a030910201434q4c6e65e1h58c5f0a82addc2d3@mail.gmail.com>
References: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>
	<191c3a030910201434q4c6e65e1h58c5f0a82addc2d3@mail.gmail.com>
Message-ID: <2d9149cd0910201937v282d7ae9ve137ac396cfe81f3@mail.gmail.com>

Anthony,

  I will the next chance I get.

Thanks!

On Tue, Oct 20, 2009 at 5:34 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> issue:
> console loglevel debug
> sofia profile internal siptrace on
>
> and put it on pastebin
> http://pastebin.freeswitch.org
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From kristian.kielhofner at gmail.com  Tue Oct 20 20:50:55 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Tue, 20 Oct 2009 23:50:55 -0400
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
In-Reply-To: <191c3a030910201434q4c6e65e1h58c5f0a82addc2d3@mail.gmail.com>
References: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>
	<191c3a030910201434q4c6e65e1h58c5f0a82addc2d3@mail.gmail.com>
Message-ID: <2d9149cd0910202050i36a7ea82nc7253d87e38f990a@mail.gmail.com>

Done.

On Tue, Oct 20, 2009 at 5:34 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> issue:
> console loglevel debug
> sofia profile internal siptrace on
>
> and put it on pastebin
> http://pastebin.freeswitch.org
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From jason at jasonjgw.net  Tue Oct 20 21:03:43 2009
From: jason at jasonjgw.net (Jason White)
Date: Wed, 21 Oct 2009 15:03:43 +1100
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <7b8592a0910200947k6ca5a391l35786f0e855ae388@mail.gmail.com>
References: <5B924C57-B658-44C0-A1CE-00A3ECB5AA07@freeswitch.org>
	<7b8592a0910191351g14cbb47dj3f19362ff6197157@mail.gmail.com>
	<FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
	<7b8592a0910192253l1313d82j51c933cb5f55ad41@mail.gmail.com>
	<20091020133437.GA4941@dan.olp.net>
	<7b8592a0910200716q4a7db93cyb955595e4993ac96@mail.gmail.com>
	<858EC056-1D84-4F30-A0B7-9A70FF98695D@freeswitch.org>
	<7b8592a0910200821m18fe9b4dg73dda05a10699af2@mail.gmail.com>
	<642B6CED-CDF3-4D40-B493-0F2D6222C30B@freeswitch.org>
	<7b8592a0910200947k6ca5a391l35786f0e855ae388@mail.gmail.com>
Message-ID: <20091021040343.GA9440@jdc.jasonjgw.net>

ineya ineya <ineyaa at gmail.com> wrote:
> Hmm, I didn't about this.
> 
> so in directory/default.xml I have:
>  <param name="dial-string"
> value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
> 
> and the modified version for IPv6 would be ?...
>  <param name="dial-string"
> value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(internal-ipv6/${dialed_user}@${dialed_domain})}"/>

Yes. Are you suggesting it didn't work?



From q.edward at gmail.com  Tue Oct 20 21:20:28 2009
From: q.edward at gmail.com (Edward Q.)
Date: Wed, 21 Oct 2009 00:20:28 -0400
Subject: [Freeswitch-users] Hi Guys
Message-ID: <89313a90910202120v6c563660s61f2efba22f916f@mail.gmail.com>

Hi to all of you guys (and ladies too).
I would like to bother all of you and take just a little of your time.
I am a total NOOB on freeswitch and would like a little help.
I need to know from all of you guys which would be the hardware recommended
from all of you guys to be able to deal with a DS3.
The thing is i have a CANTATA switch already with the DS3 and making calls.
But i will have to communicate the company international headquarters
through there which in turn there are 21 offices around the world and at a
certain time all of them will be passing from freeswitch server to the land
lines. So could you pleae help out here.
Thanks to all of you and the ones who take the time to reply.
Ed

PS All the calls will be comming from the web through SIP protocol into the
freeswitch going out to cell phones and land lines for the executives to
handle the calls.
Call flow as this.  A sip client on the webserver connects to the
freeswitch, registers and makes the call to the support agent or the
executive. Thats what we are implementing.
So i guess the web server has to be another little animal to handle the
traffic too. Thanks again.
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From sicfslist at gmail.com  Tue Oct 20 22:55:19 2009
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Wed, 21 Oct 2009 00:55:19 -0500
Subject: [Freeswitch-users] Hi Guys
In-Reply-To: <89313a90910202120v6c563660s61f2efba22f916f@mail.gmail.com>
References: <89313a90910202120v6c563660s61f2efba22f916f@mail.gmail.com>
Message-ID: <35b355e90910202255x680232f9x45319ab835f5846b@mail.gmail.com>

Question:

Will the call flow look like this (above was not very clear):

web --> FS --> Cantata --> PSTN (via TDM circuits)

Or are you trying to replace the Cantata?

SDR
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From nagalenoj at gmail.com  Tue Oct 20 23:06:31 2009
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Wed, 21 Oct 2009 11:36:31 +0530
Subject: [Freeswitch-users] UUID of the newly originated call?
Message-ID: <fe67a3340910202306i55a0b9ceve92ab19e000edf11@mail.gmail.com>

I've tried with origination_uuid.

First, I tried with SIP and my program executes successfully as what I
expected. This program initiates a new call when a call comes and let the
new call to eavesdrop the landed call.
When, I experimented with PRI(openzap), I'm facing the following error. And,
I am unable to make any calls(even from CLI). It is reporting the same error
for the subsequent calls.
Error:        2009-10-21 11:08:31.661252 [ERR] mod_openzap.c:1760 Session
for channel 1:1 not found [UUID: 8364eb30-be03-11de-aa07-07e76b7651a9]

Full log is in http://pastebin.freeswitch.org/10780
My script is here, http://pastebin.freeswitch.org/10781

What is this error for and how to avoid this?

Is there any other way to get the uuid of the originated call except
explicitly defining(origination_uuid).?!

-- 
Regards,
Nagalenoj H.
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From q.edward at gmail.com  Tue Oct 20 23:09:12 2009
From: q.edward at gmail.com (Edward Q.)
Date: Wed, 21 Oct 2009 02:09:12 -0400
Subject: [Freeswitch-users] Hi Guys
In-Reply-To: <35b355e90910202255x680232f9x45319ab835f5846b@mail.gmail.com>
References: <89313a90910202120v6c563660s61f2efba22f916f@mail.gmail.com> 
	<35b355e90910202255x680232f9x45319ab835f5846b@mail.gmail.com>
Message-ID: <89313a90910202309r6751b242q6b96a56cd1bfe57@mail.gmail.com>

It will flow exactly like that My friends.Thank you for your help and
clarifying that ..
Ed

On Wed, Oct 21, 2009 at 1:55 AM, Shelby Ramsey <sicfslist at gmail.com> wrote:

> Question:
>
> Will the call flow look like this (above was not very clear):
>
> web --> FS --> Cantata --> PSTN (via TDM circuits)
>
> Or are you trying to replace the Cantata?
>
> SDR
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ineyaa at gmail.com  Tue Oct 20 23:36:10 2009
From: ineyaa at gmail.com (ineya ineya)
Date: Wed, 21 Oct 2009 08:36:10 +0200
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <20091021040343.GA9440@jdc.jasonjgw.net>
References: <5B924C57-B658-44C0-A1CE-00A3ECB5AA07@freeswitch.org>
	<FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
	<7b8592a0910192253l1313d82j51c933cb5f55ad41@mail.gmail.com>
	<20091020133437.GA4941@dan.olp.net>
	<7b8592a0910200716q4a7db93cyb955595e4993ac96@mail.gmail.com>
	<858EC056-1D84-4F30-A0B7-9A70FF98695D@freeswitch.org>
	<7b8592a0910200821m18fe9b4dg73dda05a10699af2@mail.gmail.com>
	<642B6CED-CDF3-4D40-B493-0F2D6222C30B@freeswitch.org>
	<7b8592a0910200947k6ca5a391l35786f0e855ae388@mail.gmail.com>
	<20091021040343.GA9440@jdc.jasonjgw.net>
Message-ID: <7b8592a0910202336x1bf2a8a2nd636219ec7031c5a@mail.gmail.com>

Sort of. I have different error now:

2009-10-21 08:34:08.446939 [NOTICE] mod_sofia.c:1521 Pre-Answer
sofia/internal-ipv6/1006 at franta.openstage.net!
2009-10-21 08:34:08.462967 [NOTICE] sofia.c:3925 Hangup
sofia/internal-ipv6/sip:1007@[2000:2::1002]:5062 [CS_CONSUME_MEDIA]
[INCOMPATIBLE_DESTINATION]

Here is the complete log:
http://pastebin.freeswitch.org/10782

> Yes. Are you suggesting it didn't work?
>
>


From sicfslist at gmail.com  Tue Oct 20 23:49:55 2009
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Wed, 21 Oct 2009 01:49:55 -0500
Subject: [Freeswitch-users] Hi Guys
In-Reply-To: <89313a90910202309r6751b242q6b96a56cd1bfe57@mail.gmail.com>
References: <89313a90910202120v6c563660s61f2efba22f916f@mail.gmail.com>
	<35b355e90910202255x680232f9x45319ab835f5846b@mail.gmail.com>
	<89313a90910202309r6751b242q6b96a56cd1bfe57@mail.gmail.com>
Message-ID: <35b355e90910202349q31bc8f7as6e277008600d81b0@mail.gmail.com>

Well ... the question is pretty generic.

But based on these assumptions:
  -- no media (bypass media)
  -- routing done via XML dialplan

Something along the lines of a quad core machine with 4 gigs of ram would be
overkill for 692 calls.

Things to remember:
  -- the more cores the better (FS is heavily threaded)
  -- the more memory the better
  -- 64 bit is way better than 32 bit

SDR





>
>
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From jason at jasonjgw.net  Tue Oct 20 23:58:41 2009
From: jason at jasonjgw.net (Jason White)
Date: Wed, 21 Oct 2009 17:58:41 +1100
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <7b8592a0910202336x1bf2a8a2nd636219ec7031c5a@mail.gmail.com>
References: <FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
	<7b8592a0910192253l1313d82j51c933cb5f55ad41@mail.gmail.com>
	<20091020133437.GA4941@dan.olp.net>
	<7b8592a0910200716q4a7db93cyb955595e4993ac96@mail.gmail.com>
	<858EC056-1D84-4F30-A0B7-9A70FF98695D@freeswitch.org>
	<7b8592a0910200821m18fe9b4dg73dda05a10699af2@mail.gmail.com>
	<642B6CED-CDF3-4D40-B493-0F2D6222C30B@freeswitch.org>
	<7b8592a0910200947k6ca5a391l35786f0e855ae388@mail.gmail.com>
	<20091021040343.GA9440@jdc.jasonjgw.net>
	<7b8592a0910202336x1bf2a8a2nd636219ec7031c5a@mail.gmail.com>
Message-ID: <20091021065841.GA13150@jdc.jasonjgw.net>

ineya ineya <ineyaa at gmail.com> wrote:
> Sort of. I have different error now:
> 
> 2009-10-21 08:34:08.446939 [NOTICE] mod_sofia.c:1521 Pre-Answer
> sofia/internal-ipv6/1006 at franta.openstage.net!
> 2009-10-21 08:34:08.462967 [NOTICE] sofia.c:3925 Hangup
> sofia/internal-ipv6/sip:1007@[2000:2::1002]:5062 [CS_CONSUME_MEDIA]
> [INCOMPATIBLE_DESTINATION]

I suspect the codec negotiation. Make sure that both ends are offering a
common codec.



From tculjaga at gmail.com  Wed Oct 21 00:34:52 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 21 Oct 2009 09:34:52 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910201841030.12352@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910132359i6d53d8ban10f943795481bf7c@mail.gmail.com>
	<Pine.LNX.4.64.0910141134270.11620@srv.icf.org.ru>
	<65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>
	<Pine.LNX.4.64.0910141202500.21535@srv.icf.org.ru>
	<65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>
	<65d96fc80910140215y357fede1l37f6cba9725ddd89@mail.gmail.com>
	<Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>
	<65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
	<Pine.LNX.4.64.0910201841030.12352@srv.icf.org.ru>
Message-ID: <65d96fc80910210034nd7e8864hfcc421ec6a6c52f@mail.gmail.com>

>
>
>
> simple:
>
>
> <action application="bridge" data="h323/${number}"/>
>
> if fs not registered on gk then data="h323/${number}@xxx.xxx.xxx.xxx".
>
>
> TC>
> TC>2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3252 Found capability:
> TC>UserInput/PointDevice <14>
> TC>2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3248 FindCapability: 15
> TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external
> thread
> TC>0xb6eb60a0 for id 3048876944
> TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:65 Destroyed external
> thread
> TC>0xb6ebafa8 for id 3048876944
> TC>2009-10-20 10:08:18.426608 [DEBUG] tlibthrd.cxx:406 Destroyed thread
> TC>0xb6ebafa8 PExternalThread:0xb5ba2b90(id = b5ba2b90)
> TC>2009-10-20 10:08:18.426608 [DEBUG] h323caps.cxx:3252 Found capability:
> TC>UserInput/Modal <15>
> TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0,
> TC>expiries=0
> TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0,
> TC>expiries=0
> TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external
> thread
> TC>0xb6eba910 for id 3048876944
> TC>2009-10-20 10:08:18.426608 [DEBUG] h4601.cxx:1725 Endpoint Attached
> TC>Segmentation fault (core dumped)
> TC>tculjaga at subZero:~/freeswitch-trunk$
> TC>
> TC>pls check: http://pastebin.freeswitch.org/10769
>
> look strange, what version of libpt/h323plus you use and freeswitch itself
> ?
>
> TC>
> TC>
>



I was using latest libpt.so.2.7-beta1.

Now I went back to libpt.so.2.6-beta6 (can't find 2.6.5 you mentioned...)
and FS is crashing on every call :P .. regardless if it is inbound or
outbound...

FreeSWITCH Version 1.0.trunk (15079M)

H323Plus is from cvs





so, what i did is:


create a directory e.g. h323

mkdir -p ~/h323
cd ~/h323

  svn co
http://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/branches/v2_6ptlib-2.6

  export PKG_CONFIG_PATH=/usr/local/lib/pkgconfig
  export LD_LIBRARY_PATH=/usr/local/lib

cd ptlib-2.6

  ./configure
  make
  sudo make install



cd ~/h323

  cvs -d:pserver:anonymous at h323plus.cvs.sourceforge.net:/cvsroot/h323plus
checkout h323plus

  export PTLIBDIR=~/h323plus/ptlib


cd h323plus

  ./configure
  make
  sudo make install



assuming you have FS src in your home

cd ~/freeswitch-trunk

  make mod_h323-clean
  make mod_h323
  sudo make mod_h323-install



cd /usr/local/freeswitch/lib/

  sudo ln -sf /usr/local/lib/libpt.so.2.6-beta6 libpt.so.2.6-beta6




start FS and load mod_h323





Please, can you advice what exact revisions of ptlib you are using so i can
do "svn so -r xxx", also what exact revision of freeswitch and H323Plus you
are using ?





Now with ptlib-2.6-beta6 can't even.


T.
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From helmut.kuper at ewetel.de  Wed Oct 21 01:26:45 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Wed, 21 Oct 2009 10:26:45 +0200
Subject: [Freeswitch-users] Qustion about INFO messages
	after	Connect/Answer
In-Reply-To: <CBC6759C-5192-4731-BACA-79C90CD4EE27@freeswitch.org>
References: <4AD859A4.5000101@ewetel.de>	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>	<4ADC7C09.6070503@ewetel.de>	<191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>	<4ADC85C0.7060908@ewetel.de>
	<4ADC8B6E.7070404@ewetel.de>	<191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>	<4ADD65DA.4050809@ewetel.de>	<87f2f3b90910200919i62dac94fq1b6c9289d9fb357c@mail.gmail.com>
	<CBC6759C-5192-4731-BACA-79C90CD4EE27@freeswitch.org>
Message-ID: <4ADEC5C5.1060306@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Brian, hi Mike,

On 20.10.2009 18:41, Brian West wrote:
> Or just set the var to what you want it to say?

Yes, understood and it works so far. This means that I must enhance my
dialplan to set this new variable to preserve old behaviour. No big
deal, but at least I have to know it.

> 
> /b
> 
> On Oct 20, 2009, at 11:19 AM, Michael Collins wrote:
> 
>>
>> Under what conditions did you see "unknown"? I'm wondering if the  
>> user can just pick a default other than "unknown" if he wants  
>> something else to be displayed.

I get it for internal calls from Snom to Snom. It seems to be the
default configuration. The sip flow shows two INFO messages sent from FS
to caller after callee picked up. The first INFO messages set the
callee's name to "unknown" on caller's side. The second changed it back
to callee's number.

Maybe there is a plan behind it ... by now it is simply increasing the
sip signalling load.

Any ideas for what the first INFO message is?

regards
helmut
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=jNtK
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From red.rain.seven at gmail.com  Wed Oct 21 02:44:53 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Wed, 21 Oct 2009 17:44:53 +0800
Subject: [Freeswitch-users] sched_api doesn't get launched
In-Reply-To: <87f2f3b90910161237h356b7aa2q6ec12c6ecf78dc34@mail.gmail.com>
References: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
	<3F1747F9-C543-4C6F-BB01-4DD5FA5B7BC5@jerris.com>
	<59ad9ca10910161153h5d531533vd844e49b9ca6d896@mail.gmail.com>
	<87f2f3b90910161237h356b7aa2q6ec12c6ecf78dc34@mail.gmail.com>
Message-ID: <59ad9ca10910210244q5410f2bdq3b93c25f95673d11@mail.gmail.com>

I can't seem to find the right thing to use in mod_java to execute api
commands, only api_after_bridge

2009-10-21 17:42:46.593094 [NOTICE] mod_sofia.c:1509 Pre-Answer
sofia/internal/1688003 at 192.168.1.66!
#
# A fatal error has been detected by the Java Runtime Environment:
#
#  SIGSEGV (0xb) at pc=0x004e4480, pid=1927, tid=16116624
#
# JRE version: 6.0_16-b01
# Java VM: Java HotSpot(TM) Client VM (14.2-b01 mixed mode linux-x86 )
# Problematic frame:
# C  [libc.so.6+0x6f480]  strcpy+0x10
#
# An error report file with more information is saved as:
# /usr/local/freeswitch/bin/hs_err_pid1927.log
2009-10-21 17:42:59.883729 [ERR] switch_core_session.c:1374 Invalid
Application sched_api
2009-10-21 17:42:59.883729 [NOTICE] switch_core_session.c:1375 Hangup
sofia/internal/1688003 at 192.168.1.66 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
#
# If you would like to submit a bug report, please visit:
#   http://java.sun.com/webapps/bugreport/crash.jsp
# The crash happened outside the Java Virtual Machine in native code.
# See problematic frame for where to report the bug.


On Sat, Oct 17, 2009 at 3:37 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Oct 16, 2009 at 11:53 AM, Henry Huang <red.rain.seven at gmail.com>wrote:
>
>> So how would you trigger it from a script dialplan? The only time it
>> seemed to work is when I did setVariable("api_after_bridge", "sched_api blah
>> blah blah");
>> but then it gets executed after the channel's been teared down. I thought
>> api_after_bridge means right after the call gets connected.
>>
>> I need something to execute an api command right before or right after the
>> call gets bridged.
>>
>> api_after_bridge is a channel variable, so using setVariable works just
> fine. If you need to sched_api is an API only. Check these out:
> http://wiki.freeswitch.org/wiki/Mod_commands#Misc._Commands
>
> So you need an API object in order to use it. I don't know the syntax for
> creating an api obj in Java but in Lua it goes like this:
> api = freeswitch.API();
> res = api:execute("sched_api","+300 none my_api my_api_args")
>
> Remember, if the method you are using isn't found in the dial plan tools
> then it isn't a dial plan application. Make sure it's on the list:
> http://wiki.freeswitch.org/wiki/Mod_dptools
>
> On the other hand, API commands are listed here:
> http://wiki.freeswitch.org/wiki/Mod_commands
>
> dptools require a session object, api commands require an api object...
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From bottleman at icf.org.ru  Wed Oct 21 03:32:43 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Wed, 21 Oct 2009 14:32:43 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910210034nd7e8864hfcc421ec6a6c52f@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910132359i6d53d8ban10f943795481bf7c@mail.gmail.com>
	<Pine.LNX.4.64.0910141134270.11620@srv.icf.org.ru>
	<65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>
	<Pine.LNX.4.64.0910141202500.21535@srv.icf.org.ru>
	<65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>
	<65d96fc80910140215y357fede1l37f6cba9725ddd89@mail.gmail.com>
	<Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>
	<65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
	<Pine.LNX.4.64.0910201841030.12352@srv.icf.org.ru>
	<65d96fc80910210034nd7e8864hfcc421ec6a6c52f@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910211427520.9175@srv.icf.org.ru>

On 2009-10-21 09:34 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:


TC>
TC>
TC>
TC>I was using latest libpt.so.2.7-beta1.
TC>
TC>Now I went back to libpt.so.2.6-beta6 (can't find 2.6.5 you mentioned...)
TC>and FS is crashing on every call :P .. regardless if it is inbound or
TC>outbound...

http://www.opalvoip.org/ first link into Lalande Stable 5 Released. 
On some version of cvs ptlib i get crash on module loading:)

TC>
TC>FreeSWITCH Version 1.0.trunk (15079M)

hm, i don't test it on trunk, may be there some isues, try get stack backtrace from core file to 
see where it crash. I use 1.0.4


TC>H323Plus is from cvs

it's ok.

TC>
TC>so, what i did is:
TC>
TC>
TC>create a directory e.g. h323
TC>
TC>mkdir -p ~/h323
TC>cd ~/h323
TC>
TC>  svn co
TC>http://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/branches/v2_6ptlib-2.6
TC>
TC>  export PKG_CONFIG_PATH=/usr/local/lib/pkgconfig
TC>  export LD_LIBRARY_PATH=/usr/local/lib
TC>
TC>cd ptlib-2.6
TC>
TC>  ./configure
TC>  make
TC>  sudo make install
TC>
TC>
TC>
TC>cd ~/h323
TC>
TC>  cvs -d:pserver:anonymous at h323plus.cvs.sourceforge.net:/cvsroot/h323plus
TC>checkout h323plus
TC>
TC>  export PTLIBDIR=~/h323plus/ptlib
TC>
TC>
TC>cd h323plus
TC>
TC>  ./configure
TC>  make
TC>  sudo make install
TC>
TC>
TC>
TC>assuming you have FS src in your home
TC>
TC>cd ~/freeswitch-trunk
TC>
TC>  make mod_h323-clean
TC>  make mod_h323
TC>  sudo make mod_h323-install
TC>
TC>
TC>
TC>cd /usr/local/freeswitch/lib/
TC>
TC>  sudo ln -sf /usr/local/lib/libpt.so.2.6-beta6 libpt.so.2.6-beta6
TC>
TC>
TC>
TC>
TC>start FS and load mod_h323
TC>
TC>
TC>
TC>
TC>
TC>Please, can you advice what exact revisions of ptlib you are using so i can
TC>do "svn so -r xxx", also what exact revision of freeswitch and H323Plus you
TC>are using ?
TC>
TC>
TC>
TC>
TC>
TC>Now with ptlib-2.6-beta6 can't even.
TC>
TC>
TC>T.
TC>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From ahmedmunir007 at gmail.com  Wed Oct 21 03:37:15 2009
From: ahmedmunir007 at gmail.com (Ahmed Munir)
Date: Wed, 21 Oct 2009 15:37:15 +0500
Subject: [Freeswitch-users] Call custom variable in condition
Message-ID: <a60fed5e0910210337n45ffefe2wc3e3d9d0b403a568@mail.gmail.com>

Hi,

I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I want
to use it in condition, as I'm listing down the configuration below;

<context name="SIP_incoming">
   <extension name="call-sip-extensions">
      <condition field="destination_number" expression="^(\d+)$">
          <action application="set" data="AUTHENTICATION_STATUS=0"/>
           <action application="transfer" data="${AUTHENTICATION_STATUS} XML
Authen_Status"/>
      </condition>
   </extension>
</context>

<context name="Authen_Status">
     <extension name="exten-auth-status">
       <condition field="AUTHENTICATION_STATUS" expression="^0$">
          <action application="answer"/>
          <action application="playback" data="play.wav"/>
      </condition>
    </extension>
  </context>




 But unfortunately it is not working. Kindly advise me how to do implement
it(Note: I don't want to call script). And one more thing to ask how can I
transfer the values within the same context?

-- 
Regards,

Ahmed Munir
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From mustafa.pk at gmail.com  Wed Oct 21 03:52:24 2009
From: mustafa.pk at gmail.com (Ghulam Mustafa)
Date: Wed, 21 Oct 2009 15:52:24 +0500
Subject: [Freeswitch-users] Call custom variable in condition
In-Reply-To: <a60fed5e0910210337n45ffefe2wc3e3d9d0b403a568@mail.gmail.com>
References: <a60fed5e0910210337n45ffefe2wc3e3d9d0b403a568@mail.gmail.com>
Message-ID: <4ADEE7E8.9000008@gmail.com>

Ahmed,

you can't use variables set by "set" application within a condition, 
though it doesn't make sense. wondering if there is any logic behind 
this or it's just a simple missing feature. anyone?

-m

Ahmed Munir wrote:
> Hi,
>
> I've declared a variable named AUTHENTICATION_STATUS in dialplan, and 
> I want to use it in condition, as I'm listing down the configuration 
> below;
>
> <context name="SIP_incoming">
>    <extension name="call-sip-extensions">
>       <condition field="destination_number" expression="^(\d+)$">
>           <action application="set" data="AUTHENTICATION_STATUS=0"/>
>            <action application="transfer" 
> data="${AUTHENTICATION_STATUS} XML Authen_Status"/>
>       </condition>
>    </extension>
> </context>
>
> <context name="Authen_Status">
>      <extension name="exten-auth-status">
>        <condition field="AUTHENTICATION_STATUS" expression="^0$">
>           <action application="answer"/>
>           <action application="playback" data="play.wav"/>
>       </condition>
>     </extension>
>   </context>
>
>
>
>
>  But unfortunately it is not working. Kindly advise me how to do 
> implement it(Note: I don't want to call script). And one more thing to 
> ask how can I transfer the values within the same context?
>
> -- 
> Regards,
>
> Ahmed Munir
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From tculjaga at gmail.com  Wed Oct 21 04:13:13 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 21 Oct 2009 13:13:13 +0200
Subject: [Freeswitch-users] Call custom variable in condition
In-Reply-To: <a60fed5e0910210337n45ffefe2wc3e3d9d0b403a568@mail.gmail.com>
References: <a60fed5e0910210337n45ffefe2wc3e3d9d0b403a568@mail.gmail.com>
Message-ID: <65d96fc80910210413t1123dbc0xd5740bc0673d638c@mail.gmail.com>

consider this:



<context name="SIP_incoming">
   <extension name="call-sip-extensions">
      <condition field="destination_number" expression="^(\d+)$">
          <action application="set" data="AUTHENTICATION_STATUS=0"/>
           <action application="transfer" data="${AUTHENTICATION_STATUS} XML
Authen_Status"/>
      </condition>
   </extension>
</context>



<context name="Authen_Status">
     <extension name="exten-auth-status">
       <condition field="${AUTHENTICATION_STATUS}" expression="^0$">
          <action application="answer"/>
          <action application="playback" data="play.wav"/>
      </condition>
    </extension>
  </context>





here is one of my dialplan. I'm using execute_extension but it is quite the
same...



   <extension name="ServiceLookup">
      <condition field="destination_number" expression="(^300030)(.*)">
         <action application="lookup_service_destination" data="in
${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, in $
1, in ${network_addr}:5060, out red_contact, out authResult"/>
         <action application="log" data="INFO ########################
ServiceLookup ########################\n"/>
         <action application="log" data="INFO ########################
contact = '${red_contact}' ##############\n"/>
         <action application="log" data="INFO ########################
CallerNum = '${caller_id_number:6:16}' ##########\n"/>
         <action application="log" data="INFO ########################
RADIUS auth = '${authResult}' ##########\n"/>

         <action application="execute_extension" data="doRedirect XML
public"/>
        </condition>
   </extension>


   <extension name="doRedirect">
      <condition field="destination_number" expression="^doRedirect$"/>
      <condition field="${authResult}" expression="^0$|^60$">
         <action application="log" data="INFO ########################
RADIUS auth OK!!!' ##########\n"/>
         <action application="redirect" data="${red_contact}"/>
         <anti-action application="log" data="INFO ########################
RADIUS auth NOK!! ##########\n"/>
         <anti-action application="respond" data="403 Forbidden"/>
      </condition>

   </extension>




On Wed, Oct 21, 2009 at 12:37 PM, Ahmed Munir <ahmedmunir007 at gmail.com>wrote:

> Hi,
>
> I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I
> want to use it in condition, as I'm listing down the configuration below;
>
> <context name="SIP_incoming">
>    <extension name="call-sip-extensions">
>       <condition field="destination_number" expression="^(\d+)$">
>           <action application="set" data="AUTHENTICATION_STATUS=0"/>
>            <action application="transfer" data="${AUTHENTICATION_STATUS}
> XML Authen_Status"/>
>       </condition>
>    </extension>
> </context>
>
> <context name="Authen_Status">
>      <extension name="exten-auth-status">
>        <condition field="AUTHENTICATION_STATUS" expression="^0$">
>           <action application="answer"/>
>           <action application="playback" data="play.wav"/>
>       </condition>
>     </extension>
>   </context>
>
>
>
>
>  But unfortunately it is not working. Kindly advise me how to do implement
> it(Note: I don't want to call script). And one more thing to ask how can I
> transfer the values within the same context?
>
> --
> Regards,
>
> Ahmed Munir
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mcampbellsmith at gmail.com  Wed Oct 21 04:34:55 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Wed, 21 Oct 2009 22:34:55 +1100
Subject: [Freeswitch-users] NOT in dialplan expression
Message-ID: <33c87fa30910210434t268e8026pfd2b5439f26069fd@mail.gmail.com>

Hi!

How do I do a NOT equal to in a dialplan expression

Normaly in regex I would use the ! character.  This doesn't seem to work in FS..

ie
      <condition field="${variable}" expression="!^1">

Shouldn't that match when the variable is not starting with one?


From tculjaga at gmail.com  Wed Oct 21 04:39:17 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 21 Oct 2009 13:39:17 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910211427520.9175@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>
	<Pine.LNX.4.64.0910141202500.21535@srv.icf.org.ru>
	<65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>
	<65d96fc80910140215y357fede1l37f6cba9725ddd89@mail.gmail.com>
	<Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>
	<65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
	<Pine.LNX.4.64.0910201841030.12352@srv.icf.org.ru>
	<65d96fc80910210034nd7e8864hfcc421ec6a6c52f@mail.gmail.com>
	<Pine.LNX.4.64.0910211427520.9175@srv.icf.org.ru>
Message-ID: <65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>

2009/10/21 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-21 09:34 +0200, Tihomir Culjaga wrote
> freeswitch-users at lists.fre...:
>
>
> TC>
> TC>
> TC>
> TC>I was using latest libpt.so.2.7-beta1.
> TC>
> TC>Now I went back to libpt.so.2.6-beta6 (can't find 2.6.5 you
> mentioned...)
> TC>and FS is crashing on every call :P .. regardless if it is inbound or
> TC>outbound...
>
> http://www.opalvoip.org/ first link into Lalande Stable 5 Released.
> On some version of cvs ptlib i get crash on module loading:)
>
> TC>
> TC>FreeSWITCH Version 1.0.trunk (15079M)
>
> hm, i don't test it on trunk, may be there some isues, try get stack
> backtrace from core file to
> see where it crash. I use 1.0.4
>
>
>

module load crash: http://pastebin.freeswitch.org/10783
FreeSWITCH backtrace: http://pastebin.freeswitch.org/10784


now, the only different thing is FS trunk ...

:P
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From tculjaga at gmail.com  Wed Oct 21 04:48:24 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 21 Oct 2009 13:48:24 +0200
Subject: [Freeswitch-users] NOT in dialplan expression
In-Reply-To: <33c87fa30910210434t268e8026pfd2b5439f26069fd@mail.gmail.com>
References: <33c87fa30910210434t268e8026pfd2b5439f26069fd@mail.gmail.com>
Message-ID: <65d96fc80910210448w14d52508j7793fafd930cc8dc@mail.gmail.com>

it depends of what you are trying to acheave.... one approach is with regex

check this: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_regex

you can set a different variable and have it true or false ... than you can
compare for false state...


well .. it is up to you ...

T.


On Wed, Oct 21, 2009 at 1:34 PM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> Hi!
>
> How do I do a NOT equal to in a dialplan expression
>
> Normaly in regex I would use the ! character.  This doesn't seem to work in
> FS..
>
> ie
>      <condition field="${variable}" expression="!^1">
>
> Shouldn't that match when the variable is not starting with one?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ineyaa at gmail.com  Wed Oct 21 04:58:36 2009
From: ineyaa at gmail.com (ineya ineya)
Date: Wed, 21 Oct 2009 13:58:36 +0200
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <20091021065841.GA13150@jdc.jasonjgw.net>
References: <FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
	<20091020133437.GA4941@dan.olp.net>
	<7b8592a0910200716q4a7db93cyb955595e4993ac96@mail.gmail.com>
	<858EC056-1D84-4F30-A0B7-9A70FF98695D@freeswitch.org>
	<7b8592a0910200821m18fe9b4dg73dda05a10699af2@mail.gmail.com>
	<642B6CED-CDF3-4D40-B493-0F2D6222C30B@freeswitch.org>
	<7b8592a0910200947k6ca5a391l35786f0e855ae388@mail.gmail.com>
	<20091021040343.GA9440@jdc.jasonjgw.net>
	<7b8592a0910202336x1bf2a8a2nd636219ec7031c5a@mail.gmail.com>
	<20091021065841.GA13150@jdc.jasonjgw.net>
Message-ID: <7b8592a0910210458s28ddffbdi77cf116d33c28405@mail.gmail.com>

Codecs are fine. I spent much time experimenting with codecs and
completely missed, that freeswitch is modifiyng the SDP record.

When phone A is making a call the SDP contains candidate media attributes:

a=candidate:123abc 1 UDP 9 2000:2::1001 5012 typ host
a=candidate:123abc 1 UDP 8 10.80.62.92 5010 typ host

But when freeswitch makes the INVITE on phone B, these 2 are missing
and phone is looking for it, so the INVITE gets rejected by phone with
"448 Not acceptable here"

So the question is, how can I make the freeswitch to pass these
candidate media attributes?

On Wed, Oct 21, 2009 at 8:58 AM, Jason White <jason at jasonjgw.net> wrote:

>
> I suspect the codec negotiation. Make sure that both ends are offering a
> common codec.
>


From leon at scarlet-internet.nl  Wed Oct 21 05:00:05 2009
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Wed, 21 Oct 2009 14:00:05 +0200
Subject: [Freeswitch-users] NOT in dialplan expression
In-Reply-To: <33c87fa30910210434t268e8026pfd2b5439f26069fd@mail.gmail.com>
References: <33c87fa30910210434t268e8026pfd2b5439f26069fd@mail.gmail.com>
Message-ID: <F813C725-7F27-4FDC-94AE-105F36BF6258@scarlet-internet.nl>

Hi,

Negating is done with [^...] in a regex, so 'not 1' is matched with:

/^[^1]$/

If you want to match on a longer sequence, you can do that with  
negative lookahead, for example 'not 123' can be matched like this:

/^(?!123$)\d{3}$/

regards,

Leon


On Oct 21, 2009, at 1:34 PM, Mark Campbell-Smith wrote:

> Hi!
>
> How do I do a NOT equal to in a dialplan expression
>
> Normaly in regex I would use the ! character.  This doesn't seem to  
> work in FS..
>
> ie
>      <condition field="${variable}" expression="!^1">
>
> Shouldn't that match when the variable is not starting with one?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From bottleman at icf.org.ru  Wed Oct 21 05:24:00 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Wed, 21 Oct 2009 16:24:00 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>
	<Pine.LNX.4.64.0910141202500.21535@srv.icf.org.ru>
	<65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>
	<65d96fc80910140215y357fede1l37f6cba9725ddd89@mail.gmail.com>
	<Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>
	<65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
	<Pine.LNX.4.64.0910201841030.12352@srv.icf.org.ru>
	<65d96fc80910210034nd7e8864hfcc421ec6a6c52f@mail.gmail.com>
	<Pine.LNX.4.64.0910211427520.9175@srv.icf.org.ru>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>

On 2009-10-21 13:39 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>2009/10/21 Georgiewskiy Yuriy <bottleman at icf.org.ru>
TC>
TC>> On 2009-10-21 09:34 +0200, Tihomir Culjaga wrote
TC>> freeswitch-users at lists.fre...:
TC>>
TC>>
TC>> TC>
TC>> TC>
TC>> TC>
TC>> TC>I was using latest libpt.so.2.7-beta1.
TC>> TC>
TC>> TC>Now I went back to libpt.so.2.6-beta6 (can't find 2.6.5 you
TC>> mentioned...)
TC>> TC>and FS is crashing on every call :P .. regardless if it is inbound or
TC>> TC>outbound...
TC>>
TC>> http://www.opalvoip.org/ first link into Lalande Stable 5 Released.
TC>> On some version of cvs ptlib i get crash on module loading:)
TC>>
TC>> TC>
TC>> TC>FreeSWITCH Version 1.0.trunk (15079M)
TC>>
TC>> hm, i don't test it on trunk, may be there some isues, try get stack
TC>> backtrace from core file to
TC>> see where it crash. I use 1.0.4
TC>>
TC>>
TC>>
TC>
TC>module load crash: http://pastebin.freeswitch.org/10783
TC>FreeSWITCH backtrace: http://pastebin.freeswitch.org/10784
TC>
TC>
TC>now, the only different thing is FS trunk ...

i have no trunk at this time and cannot test it, have you enabled crash-protection in fs?

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From woodydickson at gmail.com  Wed Oct 21 06:26:57 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Wed, 21 Oct 2009 21:26:57 +0800
Subject: [Freeswitch-users] check sip client availability
Message-ID: <a44688a0910210626rfd6de48m1232862f026d6b3c@mail.gmail.com>

Hi,

Is there any API to tell freeswitch to send a SIP OPTION message to check
the availability of a SIP client?

thanks,
woody
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From brian at freeswitch.org  Wed Oct 21 07:25:17 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Oct 2009 09:25:17 -0500
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <7b8592a0910210458s28ddffbdi77cf116d33c28405@mail.gmail.com>
References: <FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
	<20091020133437.GA4941@dan.olp.net>
	<7b8592a0910200716q4a7db93cyb955595e4993ac96@mail.gmail.com>
	<858EC056-1D84-4F30-A0B7-9A70FF98695D@freeswitch.org>
	<7b8592a0910200821m18fe9b4dg73dda05a10699af2@mail.gmail.com>
	<642B6CED-CDF3-4D40-B493-0F2D6222C30B@freeswitch.org>
	<7b8592a0910200947k6ca5a391l35786f0e855ae388@mail.gmail.com>
	<20091021040343.GA9440@jdc.jasonjgw.net>
	<7b8592a0910202336x1bf2a8a2nd636219ec7031c5a@mail.gmail.com>
	<20091021065841.GA13150@jdc.jasonjgw.net>
	<7b8592a0910210458s28ddffbdi77cf116d33c28405@mail.gmail.com>
Message-ID: <BCA761FE-0B4B-4B3C-9BC4-607A92A99E22@freeswitch.org>

What are you using to make this call?

/b

On Oct 21, 2009, at 6:58 AM, ineya ineya wrote:

> Codecs are fine. I spent much time experimenting with codecs and
> completely missed, that freeswitch is modifiyng the SDP record.
>
> When phone A is making a call the SDP contains candidate media  
> attributes:
>
> a=candidate:123abc 1 UDP 9 2000:2::1001 5012 typ host
> a=candidate:123abc 1 UDP 8 10.80.62.92 5010 typ host
>
> But when freeswitch makes the INVITE on phone B, these 2 are missing
> and phone is looking for it, so the INVITE gets rejected by phone with
> "448 Not acceptable here"
>
> So the question is, how can I make the freeswitch to pass these
> candidate media attributes?



From brian at freeswitch.org  Wed Oct 21 07:26:03 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Oct 2009 09:26:03 -0500
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>
	<Pine.LNX.4.64.0910141202500.21535@srv.icf.org.ru>
	<65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>
	<65d96fc80910140215y357fede1l37f6cba9725ddd89@mail.gmail.com>
	<Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>
	<65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
	<Pine.LNX.4.64.0910201841030.12352@srv.icf.org.ru>
	<65d96fc80910210034nd7e8864hfcc421ec6a6c52f@mail.gmail.com>
	<Pine.LNX.4.64.0910211427520.9175@srv.icf.org.ru>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
Message-ID: <DE165593-BA7B-4B0A-AAC2-429831D2ECDC@freeswitch.org>

Your SVN Account will be done soon and the directory in endpoints is  
already created for you to start importing your work.

Thanks,

/b

On Oct 21, 2009, at 7:24 AM, Georgiewskiy Yuriy wrote:

> On 2009-10-21 13:39 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre 
> ...:
>
> TC>2009/10/21 Georgiewskiy Yuriy <bottleman at icf.org.ru>
> TC>
> TC>> On 2009-10-21 09:34 +0200, Tihomir Culjaga wrote
> TC>> freeswitch-users at lists.fre...:
> TC>>
> TC>>
> TC>> TC>
> TC>> TC>
> TC>> TC>
> TC>> TC>I was using latest libpt.so.2.7-beta1.
> TC>> TC>
> TC>> TC>Now I went back to libpt.so.2.6-beta6 (can't find 2.6.5 you
> TC>> mentioned...)
> TC>> TC>and FS is crashing on every call :P .. regardless if it is  
> inbound or
> TC>> TC>outbound...
> TC>>
> TC>> http://www.opalvoip.org/ first link into Lalande Stable 5  
> Released.
> TC>> On some version of cvs ptlib i get crash on module loading:)
> TC>>
> TC>> TC>
> TC>> TC>FreeSWITCH Version 1.0.trunk (15079M)
> TC>>
> TC>> hm, i don't test it on trunk, may be there some isues, try get  
> stack
> TC>> backtrace from core file to
> TC>> see where it crash. I use 1.0.4
> TC>>
> TC>>
> TC>>
> TC>
> TC>module load crash: http://pastebin.freeswitch.org/10783
> TC>FreeSWITCH backtrace: http://pastebin.freeswitch.org/10784
> TC>
> TC>
> TC>now, the only different thing is FS trunk ...
>
> i have no trunk at this time and cannot test it, have you enabled  
> crash-protection in fs?
>
> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129- 
> RIPE_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From bottleman at icf.org.ru  Wed Oct 21 07:37:01 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Wed, 21 Oct 2009 18:37:01 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <DE165593-BA7B-4B0A-AAC2-429831D2ECDC@freeswitch.org>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>
	<Pine.LNX.4.64.0910141202500.21535@srv.icf.org.ru>
	<65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>
	<65d96fc80910140215y357fede1l37f6cba9725ddd89@mail.gmail.com>
	<Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>
	<65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
	<Pine.LNX.4.64.0910201841030.12352@srv.icf.org.ru>
	<65d96fc80910210034nd7e8864hfcc421ec6a6c52f@mail.gmail.com>
	<Pine.LNX.4.64.0910211427520.9175@srv.icf.org.ru>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
	<DE165593-BA7B-4B0A-AAC2-429831D2ECDC@freeswitch.org>
Message-ID: <Pine.LNX.4.64.0910211836450.30865@srv.icf.org.ru>

On 2009-10-21 09:26 -0500, Brian West wrote freeswitch-users at lists.freeswit...:

??.

BW>Your SVN Account will be done soon and the directory in endpoints is  
BW>already created for you to start importing your work.
BW>
BW>Thanks,
BW>
BW>/b
BW>
BW>On Oct 21, 2009, at 7:24 AM, Georgiewskiy Yuriy wrote:
BW>
BW>> On 2009-10-21 13:39 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre 
BW>> ...:
BW>>
BW>> TC>2009/10/21 Georgiewskiy Yuriy <bottleman at icf.org.ru>
BW>> TC>
BW>> TC>> On 2009-10-21 09:34 +0200, Tihomir Culjaga wrote
BW>> TC>> freeswitch-users at lists.fre...:
BW>> TC>>
BW>> TC>>
BW>> TC>> TC>
BW>> TC>> TC>
BW>> TC>> TC>
BW>> TC>> TC>I was using latest libpt.so.2.7-beta1.
BW>> TC>> TC>
BW>> TC>> TC>Now I went back to libpt.so.2.6-beta6 (can't find 2.6.5 you
BW>> TC>> mentioned...)
BW>> TC>> TC>and FS is crashing on every call :P .. regardless if it is  
BW>> inbound or
BW>> TC>> TC>outbound...
BW>> TC>>
BW>> TC>> http://www.opalvoip.org/ first link into Lalande Stable 5  
BW>> Released.
BW>> TC>> On some version of cvs ptlib i get crash on module loading:)
BW>> TC>>
BW>> TC>> TC>
BW>> TC>> TC>FreeSWITCH Version 1.0.trunk (15079M)
BW>> TC>>
BW>> TC>> hm, i don't test it on trunk, may be there some isues, try get  
BW>> stack
BW>> TC>> backtrace from core file to
BW>> TC>> see where it crash. I use 1.0.4
BW>> TC>>
BW>> TC>>
BW>> TC>>
BW>> TC>
BW>> TC>module load crash: http://pastebin.freeswitch.org/10783
BW>> TC>FreeSWITCH backtrace: http://pastebin.freeswitch.org/10784
BW>> TC>
BW>> TC>
BW>> TC>now, the only different thing is FS trunk ...
BW>>
BW>> i have no trunk at this time and cannot test it, have you enabled  
BW>> crash-protection in fs?
BW>>
BW>> C ?????????                       With Best Regards
BW>> ???????????? ????.                Georgiewskiy Yuriy
BW>> +7 4872 711666                    +7 4872 711666
BW>> ???? +7 4872 711143               fax +7 4872 711143
BW>> ???????? ??? "?? ?? ??????"       IT Service Ltd
BW>> http://nkoort.ru                  http://nkoort.ru
BW>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
BW>> YG129-RIPE                        YG129- 
BW>> RIPE_______________________________________________
BW>> FreeSWITCH-users mailing list
BW>> FreeSWITCH-users at lists.freeswitch.org
BW>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
BW>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
BW>> users
BW>> http://www.freeswitch.org
BW>
BW>
BW>_______________________________________________
BW>FreeSWITCH-users mailing list
BW>FreeSWITCH-users at lists.freeswitch.org
BW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
BW>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
BW>http://www.freeswitch.org
BW>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From tculjaga at gmail.com  Wed Oct 21 07:43:53 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 21 Oct 2009 16:43:53 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>
	<65d96fc80910140215y357fede1l37f6cba9725ddd89@mail.gmail.com>
	<Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>
	<65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
	<Pine.LNX.4.64.0910201841030.12352@srv.icf.org.ru>
	<65d96fc80910210034nd7e8864hfcc421ec6a6c52f@mail.gmail.com>
	<Pine.LNX.4.64.0910211427520.9175@srv.icf.org.ru>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
Message-ID: <65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>

>
> TC>
> TC>module load crash: http://pastebin.freeswitch.org/10783
> TC>FreeSWITCH backtrace: http://pastebin.freeswitch.org/10784
> TC>
> TC>
> TC>now, the only different thing is FS trunk ...
>
> i have no trunk at this time and cannot test it, have you enabled
> crash-protection in fs?
>
>
Ok i moved back FS to 1.0.4 and i'm back on the first issues,

1. no audio after i answer the call (call flow is H323 => FS => SIP).. i
hear the ringback on the H323 side bit when i answer the call, nothing!

FS = 10.4.62.7
SIP phone = 10.4.62.89
H323 endpoint = 10.1.14.153



 53.696259  10.1.14.153 -> 10.4.62.7    H.225.0 CS: setup
 53.705415    10.4.62.7 -> 10.1.14.153  H.225.0 CS: callProceeding
 53.716368    10.4.62.7 -> 10.4.62.89   SIP/SDP Request: INVITE
sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>;transport=udp, with session
description
 53.717697    10.4.62.7 -> 10.1.14.153  H.225.0 CS: alerting
 53.720132    10.4.62.7 -> 10.1.14.153  H.225.0 CS: facility
 53.725453    10.4.62.7 -> 10.1.14.153  H.245 terminalCapabilitySet
 53.725957    10.4.62.7 -> 10.1.14.153  H.245 masterSlaveDetermination
 53.728129   10.4.62.89 -> 10.4.62.7    SIP Status: 100 Trying
 53.741112   10.4.62.89 -> 10.4.62.7    SIP Status: 180 Ringing
 53.743199  10.1.14.153 -> 10.4.62.7    H.245 terminalCapabilitySet
 53.744221  10.1.14.153 -> 10.4.62.7    H.245 masterSlaveDetermination
 53.749443    10.4.62.7 -> 10.1.14.153  H.245 terminalCapabilitySetAck
 53.751624    10.4.62.7 -> 10.1.14.153  H.245 masterSlaveDeterminationAck
 53.758710  10.1.14.153 -> 10.4.62.7    H.245 terminalCapabilitySetAck
 53.761241  10.1.14.153 -> 10.4.62.7    H.245 masterSlaveDeterminationAck
 53.763919    10.4.62.7 -> 10.1.14.153  H.245 openLogicalChannel (g711A)
#3712: OLC found 10.1.14.153/10.4.62.7/101
 53.777464  10.1.14.153 -> 10.4.62.7    H.245 openLogicalChannelAck
 53.799864    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1305, Time=240, Mark
 53.828940    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1306, Time=480
 53.859180    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1307, Time=720
 53.889379    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1308, Time=960
 53.919611    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1309, Time=1200
 53.949833    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1310, Time=1440
 53.978964    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1311, Time=1680
 54.009218    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1312, Time=1920
 54.039424    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1313, Time=2160
 54.069640    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1314, Time=2400
 54.099795    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1315, Time=2640
 54.129064    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1316, Time=2880

<------------------ snip ----------------> this is a ringback sent from FS
=> H323endpoint

 60.189074    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1518, Time=51360
 60.219197    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1519, Time=51600
 60.249451    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1520, Time=51840
 60.279684    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1521, Time=52080
 60.309689    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1522, Time=52320
 60.338812    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1523, Time=52560
 60.368997    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1524, Time=52800
 60.399231    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1525, Time=53040
 60.429445    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1526, Time=53280
 60.459679    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1527, Time=53520
 60.488917    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1528, Time=53760
 60.494975   10.4.62.89 -> 10.4.62.7    SIP/SDP Status: 200 OK, with session
description
 60.495478    10.4.62.7 -> 10.4.62.89   SIP Request: ACK
sip:1001 at 10.4.62.89<sip%3A1001 at 10.4.62.89>
;transport=udp
 60.496114    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 1
 60.519467    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1529, Time=54000
 60.520574    10.4.62.7 -> 10.1.14.153  H.225.0 CS: connect
 60.531284  10.1.14.153 -> 10.4.62.7    H.245 openLogicalChannel (g711A)
#4045: OLC found 10.4.62.7/10.1.14.153/101
 60.535298    10.4.62.7 -> 10.1.14.153  H.245 openLogicalChannelAck
 60.595098  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=39691, Time=160, Mark
 60.598967  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=39692, Time=320
 60.616538  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=39693, Time=480
 60.639583  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=39694, Time=640
 60.657185  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=39695, Time=800
 60.673118   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21152, Time=98705487
 60.680623  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=39696, Time=960
 60.693033   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21153, Time=98705647
 60.698727  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=39697, Time=1120
 60.712984   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21154, Time=98705807
 60.716184  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=39698, Time=1280
 60.733016   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21155, Time=98705967
 60.739264  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=39699, Time=1440
 60.752987   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21156, Time=98706127
 60.756820  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=39700, Time=1600
 60.772977   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21157, Time=98706287
 60.780507  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=39701, Time=1760
 60.793029   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21158, Time=98706447
 60.798172  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=39702, Time=1920
 60.812980   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21159, Time=98706607
 60.815755  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=39703, Time=2080
 60.833107   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21160, Time=98706767
 60.838943  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=39704, Time=2240
 60.852999   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21161, Time=98706927
 60.856476  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=39705, Time=2400

<------------------ snip ----------------> this is after after SIP phone
answered

 68.245488  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=40079, Time=62240
 68.252993   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21531, Time=98766127
 68.268524  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=40080, Time=62400
 68.272988   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21532, Time=98766287
 68.286129  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=40081, Time=62560
 68.292978   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21533, Time=98766447
 68.309595  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=40082, Time=62720
 68.312988   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21534, Time=98766607
 68.327216  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=40083, Time=62880
 68.332982   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21535, Time=98766767
 68.350940  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=40084, Time=63040
 68.352977   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21536, Time=98766927
 68.368506  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=40085, Time=63200
 68.374916   10.4.62.89 -> 10.4.62.7    SIP Request: BYE
sip:mod_sofia at 10.4.62.7:5060
 68.375342    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 3
 68.375620    10.4.62.7 -> 10.4.62.89   SIP Status: 200 OK
 68.385785  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=40086, Time=63360
 68.409172  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=40087, Time=63520

<------------------ snip ----------------> This is after SIP Phone released
the call

 70.495427  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=40193, Time=80480
 70.518592  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=40194, Time=80640
 70.536159  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=40195, Time=80800
 70.552157  10.1.14.153 -> 10.4.62.7    H.245 endSessionCommand
 70.552401  10.1.14.153 -> 10.4.62.7    H.225.0 CS: releaseComplete
 70.553981    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 3
 70.554488    10.4.62.7 -> 10.1.14.153  H.245 endSessionCommand
 70.555056    10.4.62.7 -> 10.1.14.153  H.225.0 CS: releaseComplete



Well, something gets wrong hrer ...




2. hangup from sip side doesn't release the h323 leg (now the difference is
that FS is not complaining about thread mismatch ant it looks clean but FS
doesn't send any releasecomplete message... strange)
3. coredumps when i place outgoing calls
-------------- next part --------------
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From rupa at rupa.com  Wed Oct 21 08:07:46 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 21 Oct 2009 10:07:46 -0500
Subject: [Freeswitch-users] NOT in dialplan expression
In-Reply-To: <33c87fa30910210434t268e8026pfd2b5439f26069fd@mail.gmail.com>
References: <33c87fa30910210434t268e8026pfd2b5439f26069fd@mail.gmail.com>
Message-ID: <b39273530910210807j1f8a2fc6l4ff2bf1c154497b9@mail.gmail.com>

You can also look at using anti-action rather than action after the condition.

condition == if
action == then
anti-action == else

On Wed, Oct 21, 2009 at 6:34 AM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> Hi!
>
> How do I do a NOT equal to in a dialplan expression
>
> Normaly in regex I would use the ! character. ?This doesn't seem to work in FS..
>
> ie
> ? ? ?<condition field="${variable}" expression="!^1">
>
> Shouldn't that match when the variable is not starting with one?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa


From rupa at rupa.com  Wed Oct 21 08:09:35 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 21 Oct 2009 10:09:35 -0500
Subject: [Freeswitch-users] check sip client availability
In-Reply-To: <a44688a0910210626rfd6de48m1232862f026d6b3c@mail.gmail.com>
References: <a44688a0910210626rfd6de48m1232862f026d6b3c@mail.gmail.com>
Message-ID: <b39273530910210809g22cf8829mc4918590c849c669@mail.gmail.com>

This applies only to clients that are detected as nat, but maybe:

http://wiki.freeswitch.org/wiki/Sofia.conf.xml#nat-options-ping

This is applied to the profile, not an app or a var setting.

On Wed, Oct 21, 2009 at 8:26 AM, Woody Dickson <woodydickson at gmail.com> wrote:
> Hi,
>
> Is there any API to tell freeswitch to send a SIP OPTION message to check
> the availability of a SIP client?
>
> thanks,
> woody
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa


From bottleman at icf.org.ru  Wed Oct 21 08:23:59 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Wed, 21 Oct 2009 19:23:59 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>
	<65d96fc80910140215y357fede1l37f6cba9725ddd89@mail.gmail.com>
	<Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>
	<65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
	<Pine.LNX.4.64.0910201841030.12352@srv.icf.org.ru>
	<65d96fc80910210034nd7e8864hfcc421ec6a6c52f@mail.gmail.com>
	<Pine.LNX.4.64.0910211427520.9175@srv.icf.org.ru>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>

On 2009-10-21 16:43 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>>
TC>> TC>
TC>> TC>module load crash: http://pastebin.freeswitch.org/10783
TC>> TC>FreeSWITCH backtrace: http://pastebin.freeswitch.org/10784
TC>> TC>
TC>> TC>
TC>> TC>now, the only different thing is FS trunk ...
TC>>
TC>> i have no trunk at this time and cannot test it, have you enabled
TC>> crash-protection in fs?
TC>>
TC>>
TC>Ok i moved back FS to 1.0.4 and i'm back on the first issues,
TC>
TC>1. no audio after i answer the call (call flow is H323 => FS => SIP).. i
TC>hear the ringback on the H323 side bit when i answer the call, nothing!

TC>FS = 10.4.62.7
TC>SIP phone = 10.4.62.89
TC>H323 endpoint = 10.1.14.153

TC> 53.696259  10.1.14.153 -> 10.4.62.7    H.225.0 CS: setup
TC> 53.705415    10.4.62.7 -> 10.1.14.153  H.225.0 CS: callProceeding
TC> 53.716368    10.4.62.7 -> 10.4.62.89   SIP/SDP Request: INVITE
TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>;transport=udp, with session
TC>description
TC> 53.717697    10.4.62.7 -> 10.1.14.153  H.225.0 CS: alerting
TC> 53.720132    10.4.62.7 -> 10.1.14.153  H.225.0 CS: facility
TC> 53.725453    10.4.62.7 -> 10.1.14.153  H.245 terminalCapabilitySet
TC> 53.725957    10.4.62.7 -> 10.1.14.153  H.245 masterSlaveDetermination
TC> 53.728129   10.4.62.89 -> 10.4.62.7    SIP Status: 100 Trying
TC> 53.741112   10.4.62.89 -> 10.4.62.7    SIP Status: 180 Ringing
TC> 53.743199  10.1.14.153 -> 10.4.62.7    H.245 terminalCapabilitySet
TC> 53.744221  10.1.14.153 -> 10.4.62.7    H.245 masterSlaveDetermination
TC> 53.749443    10.4.62.7 -> 10.1.14.153  H.245 terminalCapabilitySetAck
TC> 53.751624    10.4.62.7 -> 10.1.14.153  H.245 masterSlaveDeterminationAck
TC> 53.758710  10.1.14.153 -> 10.4.62.7    H.245 terminalCapabilitySetAck
TC> 53.761241  10.1.14.153 -> 10.4.62.7    H.245 masterSlaveDeterminationAck
TC> 53.763919    10.4.62.7 -> 10.1.14.153  H.245 openLogicalChannel (g711A)
TC>#3712: OLC found 10.1.14.153/10.4.62.7/101
TC> 53.777464  10.1.14.153 -> 10.4.62.7    H.245 openLogicalChannelAck
TC> 53.799864    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1305, Time=240, Mark
TC> 53.828940    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1306, Time=480
TC> 53.859180    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1307, Time=720
TC> 53.889379    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1308, Time=960
TC> 53.919611    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1309, Time=1200
TC> 53.949833    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1310, Time=1440
TC> 53.978964    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1311, Time=1680
TC> 54.009218    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1312, Time=1920
TC> 54.039424    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1313, Time=2160
TC> 54.069640    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1314, Time=2400
TC> 54.099795    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1315, Time=2640
TC> 54.129064    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1316, Time=2880
TC>
TC><------------------ snip ----------------> this is a ringback sent from FS
TC>=> H323endpoint
TC>
TC> 60.189074    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1518, Time=51360
TC> 60.219197    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1519, Time=51600
TC> 60.249451    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1520, Time=51840
TC> 60.279684    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1521, Time=52080
TC> 60.309689    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1522, Time=52320
TC> 60.338812    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1523, Time=52560
TC> 60.368997    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1524, Time=52800
TC> 60.399231    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1525, Time=53040
TC> 60.429445    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1526, Time=53280
TC> 60.459679    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1527, Time=53520
TC> 60.488917    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1528, Time=53760
TC> 60.494975   10.4.62.89 -> 10.4.62.7    SIP/SDP Status: 200 OK, with session
TC>description
TC> 60.495478    10.4.62.7 -> 10.4.62.89   SIP Request: ACK
TC>sip:1001 at 10.4.62.89<sip%3A1001 at 10.4.62.89>
TC>;transport=udp
TC> 60.496114    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 1
TC> 60.519467    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1529, Time=54000
TC> 60.520574    10.4.62.7 -> 10.1.14.153  H.225.0 CS: connect
TC> 60.531284  10.1.14.153 -> 10.4.62.7    H.245 openLogicalChannel (g711A)
TC>#4045: OLC found 10.4.62.7/10.1.14.153/101
TC> 60.535298    10.4.62.7 -> 10.1.14.153  H.245 openLogicalChannelAck
TC> 60.595098  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=39691, Time=160, Mark
TC> 60.598967  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=39692, Time=320
TC> 60.616538  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=39693, Time=480
TC> 60.639583  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=39694, Time=640
TC> 60.657185  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=39695, Time=800
TC> 60.673118   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21152, Time=98705487
TC> 60.680623  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=39696, Time=960
TC> 60.693033   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21153, Time=98705647
TC> 60.698727  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=39697, Time=1120
TC> 60.712984   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21154, Time=98705807
TC> 60.716184  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=39698, Time=1280
TC> 60.733016   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21155, Time=98705967
TC> 60.739264  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=39699, Time=1440
TC> 60.752987   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21156, Time=98706127
TC> 60.756820  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=39700, Time=1600
TC> 60.772977   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21157, Time=98706287
TC> 60.780507  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=39701, Time=1760
TC> 60.793029   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21158, Time=98706447
TC> 60.798172  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=39702, Time=1920
TC> 60.812980   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21159, Time=98706607
TC> 60.815755  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=39703, Time=2080
TC> 60.833107   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21160, Time=98706767
TC> 60.838943  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=39704, Time=2240
TC> 60.852999   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21161, Time=98706927
TC> 60.856476  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=39705, Time=2400
TC>
TC><------------------ snip ----------------> this is after after SIP phone
TC>answered
TC>
TC> 68.245488  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=40079, Time=62240
TC> 68.252993   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21531, Time=98766127
TC> 68.268524  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=40080, Time=62400
TC> 68.272988   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21532, Time=98766287
TC> 68.286129  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=40081, Time=62560
TC> 68.292978   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21533, Time=98766447
TC> 68.309595  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=40082, Time=62720
TC> 68.312988   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21534, Time=98766607
TC> 68.327216  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=40083, Time=62880
TC> 68.332982   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21535, Time=98766767
TC> 68.350940  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=40084, Time=63040
TC> 68.352977   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21536, Time=98766927
TC> 68.368506  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=40085, Time=63200
TC> 68.374916   10.4.62.89 -> 10.4.62.7    SIP Request: BYE
TC>sip:mod_sofia at 10.4.62.7:5060
TC> 68.375342    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 3
TC> 68.375620    10.4.62.7 -> 10.4.62.89   SIP Status: 200 OK
TC> 68.385785  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=40086, Time=63360
TC> 68.409172  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=40087, Time=63520
TC>
TC><------------------ snip ----------------> This is after SIP Phone released
TC>the call
TC>
TC> 70.495427  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=40193, Time=80480
TC> 70.518592  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=40194, Time=80640
TC> 70.536159  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=40195, Time=80800
TC> 70.552157  10.1.14.153 -> 10.4.62.7    H.245 endSessionCommand
TC> 70.552401  10.1.14.153 -> 10.4.62.7    H.225.0 CS: releaseComplete
TC> 70.553981    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 3
TC> 70.554488    10.4.62.7 -> 10.1.14.153  H.245 endSessionCommand
TC> 70.555056    10.4.62.7 -> 10.1.14.153  H.225.0 CS: releaseComplete
TC>
TC>
TC>
TC>Well, something gets wrong hrer ...
TC>
TC>
TC>
TC>
TC>2. hangup from sip side doesn't release the h323 leg (now the difference is
TC>that FS is not complaining about thread mismatch ant it looks clean but FS
TC>doesn't send any releasecomplete message... strange)
TC>3. coredumps when i place outgoing calls

btw, 

TC> 70.552157  10.1.14.153 -> 10.4.62.7    H.245 endSessionCommand                                             
TC> 70.552401  10.1.14.153 -> 10.4.62.7    H.225.0 CS: releaseComplete                                         
TC> 70.553981    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 3                                           
TC> 70.554488    10.4.62.7 -> 10.1.14.153  H.245 endSessionCommand                                             
TC> 70.555056    10.4.62.7 -> 10.1.14.153  H.225.0 CS: releaseComplete 

it send, now i have no way to test h323->sip transit, i will have it tomorow.
sip->h323 for me work fine now, give backtrace from code dump of 1.0.4 where it die?

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From ineyaa at gmail.com  Wed Oct 21 08:40:13 2009
From: ineyaa at gmail.com (ineya ineya)
Date: Wed, 21 Oct 2009 17:40:13 +0200
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <BCA761FE-0B4B-4B3C-9BC4-607A92A99E22@freeswitch.org>
References: <FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
	<858EC056-1D84-4F30-A0B7-9A70FF98695D@freeswitch.org>
	<7b8592a0910200821m18fe9b4dg73dda05a10699af2@mail.gmail.com>
	<642B6CED-CDF3-4D40-B493-0F2D6222C30B@freeswitch.org>
	<7b8592a0910200947k6ca5a391l35786f0e855ae388@mail.gmail.com>
	<20091021040343.GA9440@jdc.jasonjgw.net>
	<7b8592a0910202336x1bf2a8a2nd636219ec7031c5a@mail.gmail.com>
	<20091021065841.GA13150@jdc.jasonjgw.net>
	<7b8592a0910210458s28ddffbdi77cf116d33c28405@mail.gmail.com>
	<BCA761FE-0B4B-4B3C-9BC4-607A92A99E22@freeswitch.org>
Message-ID: <7b8592a0910210840u57a664f3l9c65ee75f457ea6a@mail.gmail.com>

the phone is called OpenStage

On Wed, Oct 21, 2009 at 4:25 PM, Brian West <brian at freeswitch.org> wrote:
> What are you using to make this call?
>
> /b
>


From msc at freeswitch.org  Wed Oct 21 08:43:58 2009
From: msc at freeswitch.org (Michael S Collins)
Date: Wed, 21 Oct 2009 08:43:58 -0700
Subject: [Freeswitch-users] sched_api doesn't get launched
In-Reply-To: <59ad9ca10910210244q5410f2bdq3b93c25f95673d11@mail.gmail.com>
References: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
	<3F1747F9-C543-4C6F-BB01-4DD5FA5B7BC5@jerris.com>
	<59ad9ca10910161153h5d531533vd844e49b9ca6d896@mail.gmail.com>
	<87f2f3b90910161237h356b7aa2q6ec12c6ecf78dc34@mail.gmail.com>
	<59ad9ca10910210244q5410f2bdq3b93c25f95673d11@mail.gmail.com>
Message-ID: <38B84C14-FFC3-43F4-BAAE-61B1F1D762C7@freeswitch.org>

Judging by this error I would assume that you're still calling  
sched_api as a Dialplan application and not as an FS API command.

You need to figure out how to create an API obj in java and call  
sched_api from that object.

-MC

Sent from my iPhone

On Oct 21, 2009, at 2:44 AM, Henry Huang <red.rain.seven at gmail.com>  
wrote:

> I can't seem to find the right thing to use in mod_java to execute  
> api commands, only api_after_bridge
>
>
> 2009-10-21 17:42:46.593094 [NOTICE] mod_sofia.c:1509 Pre-Answer  
> sofia/internal/1688003 at 192.168.1.66!
> #
> # A fatal error has been detected by the Java Runtime Environment:
> #
> #  SIGSEGV (0xb) at pc=0x004e4480, pid=1927, tid=16116624
> #
> # JRE version: 6.0_16-b01
> # Java VM: Java HotSpot(TM) Client VM (14.2-b01 mixed mode linux-x86 )
> # Problematic frame:
> # C  [libc.so.6+0x6f480]  strcpy+0x10
> #
> # An error report file with more information is saved as:
> # /usr/local/freeswitch/bin/hs_err_pid1927.log
> 2009-10-21 17:42:59.883729 [ERR] switch_core_session.c:1374 Invalid  
> Application sched_api
> 2009-10-21 17:42:59.883729 [NOTICE] switch_core_session.c:1375  
> Hangup sofia/internal/1688003 at 192.168.1.66 [CS_EXECUTE]  
> [DESTINATION_OUT_OF_ORDER]
> #
> # If you would like to submit a bug report, please visit:
> #   http://java.sun.com/webapps/bugreport/crash.jsp
> # The crash happened outside the Java Virtual Machine in native code.
> # See problematic frame for where to report the bug.
>
>
> On Sat, Oct 17, 2009 at 3:37 AM, Michael Collins  
> <msc at freeswitch.org> wrote:
>
>
> On Fri, Oct 16, 2009 at 11:53 AM, Henry Huang <red.rain.seven at gmail.com 
> > wrote:
> So how would you trigger it from a script dialplan? The only time it  
> seemed to work is when I did setVariable("api_after_bridge",  
> "sched_api blah blah blah");
> but then it gets executed after the channel's been teared down. I  
> thought api_after_bridge means right after the call gets connected.
>
> I need something to execute an api command right before or right  
> after the call gets bridged.
>
> api_after_bridge is a channel variable, so using setVariable works  
> just fine. If you need to sched_api is an API only. Check these out:
> http://wiki.freeswitch.org/wiki/Mod_commands#Misc._Commands
>
> So you need an API object in order to use it. I don't know the  
> syntax for creating an api obj in Java but in Lua it goes like this:
> api = freeswitch.API();
> res = api:execute("sched_api","+300 none my_api my_api_args")
>
> Remember, if the method you are using isn't found in the dial plan  
> tools then it isn't a dial plan application. Make sure it's on the  
> list:
> http://wiki.freeswitch.org/wiki/Mod_dptools
>
> On the other hand, API commands are listed here:
>
> http://wiki.freeswitch.org/wiki/Mod_commands
>
> dptools require a session object, api commands require an api  
> object...
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Henry Huang
> UniC Solution - Communication Unified
> VoIP & Open Source software Consultant
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From tculjaga at gmail.com  Wed Oct 21 08:48:47 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 21 Oct 2009 17:48:47 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>
	<65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
	<Pine.LNX.4.64.0910201841030.12352@srv.icf.org.ru>
	<65d96fc80910210034nd7e8864hfcc421ec6a6c52f@mail.gmail.com>
	<Pine.LNX.4.64.0910211427520.9175@srv.icf.org.ru>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
Message-ID: <65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>

FS = 10.4.62.7
SIP phone = 10.4.62.89
H323 endpoint = 10.1.14.153



> TC>2. hangup from sip side doesn't release the h323 leg (now the difference
> is
> TC>that FS is not complaining about thread mismatch ant it looks clean but
> FS
> TC>doesn't send any releasecomplete message... strange)
> TC>3. coredumps when i place outgoing calls
>
> btw,
>
> TC> 70.552157  10.1.14.153 -> 10.4.62.7    H.245 endSessionCommand
> TC> 70.552401  10.1.14.153 -> 10.4.62.7    H.225.0 CS: releaseComplete
> TC> 70.553981    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 3
> TC> 70.554488    10.4.62.7 -> 10.1.14.153  H.245 endSessionCommand
> TC> 70.555056    10.4.62.7 -> 10.1.14.153  H.225.0 CS: releaseComplete
>
> it send, now i have no way to test h323->sip transit, i will have it
> tomorow.
> sip->h323 for me work fine now, give backtrace from code dump of 1.0.4
> where it die?
>
>

this endSession is when i hangup from H232 side as well :P ... if i don't
hangup on H323 side the H323 leg is not released. Pls chec the time the
packets were sent ...



Here i hangup on the SIP Phone:

68.374916   10.4.62.89 -> 10.4.62.7    SIP Request: BYE
sip:mod_sofia at 10.4.62.7:5060
68.375342    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 3
68.375620    10.4.62.7 -> 10.4.62.89   SIP Status: 200 OK

2 sec delay

Here i hangup on the H323 endpoint (releaseComplete comes from H323 endpoint
first here )
70.552157  10.1.14.153 -> 10.4.62.7    H.245 endSessionCommand
70.552401  10.1.14.153 -> 10.4.62.7    H.225.0 CS: releaseComplete


FS just acknowlages it here:
70.554488    10.4.62.7 -> 10.1.14.153  H.245 endSessionCommand
70.555056    10.4.62.7 -> 10.1.14.153  H.225.0 CS: releaseComplete



I have enabled crash-protection and when i do SIP => H323 call it doesn't
generate coredumps... it is just this thread that is crashing ... pls check
the log downbelow:

Dialplan: sofia/internal/1001 at singtel Regex (FAIL) [EMERGENCY]
destination_number(05492122) =~ /^0(112|9[23456])$/ break=on-false
Dialplan: sofia/internal/1001 at singtel parsing [default->SPECIAL_SERVICES]
continue=false
Dialplan: sofia/internal/1001 at singtel Regex (FAIL) [SPECIAL_SERVICES]
destination_number(05492122) =~ /^0(9[01789]\d{3,4})$/ break=on-false
Dialplan: sofia/internal/1001 at singtel parsing [default->ENYTHING_ELSE]
continue=false
Dialplan: sofia/internal/1001 at singtel Regex (PASS) [ENYTHING_ELSE]
destination_number(05492122) =~
/^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$/ break=on-false
Dialplan: sofia/internal/1001 at singtel Action
set(effective_caller_id_number=1001282122)
Dialplan: sofia/internal/1001 at singtel Action set(NCX_IP=10.4.4.254)
Dialplan: sofia/internal/1001 at singtel Action set(call_timeout=30)
Dialplan: sofia/internal/1001 at singtel Action set(hangup_after_bridge=true)
Dialplan: sofia/internal/1001 at singtel Action set(bypass_media=false)
Dialplan: sofia/internal/1001 at singtel Action set(proxy_media=true)
Dialplan: sofia/internal/1001 at singtel Action bridge(h323/05492122@${NCX_IP})
2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:114
(sofia/internal/1001 at singtel) State Change CS_ROUTING -> CS_EXECUTE
2009-10-21 17:35:28.682475 [DEBUG] switch_core_session.c:932 Send signal
sofia/internal/1001 at singtel [BREAK]
2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:484
(sofia/internal/1001 at singtel) State ROUTING going to sleep
2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/1001 at singtel) Running State Change CS_EXECUTE
2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:491
(sofia/internal/1001 at singtel) State EXECUTE
2009-10-21 17:35:28.682475 [DEBUG] mod_sofia.c:173
sofia/internal/1001 at singtel SOFIA EXECUTE
2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:151
sofia/internal/1001 at singtel Standard EXECUTE
EXECUTE sofia/internal/1001 at singtel set(open=true)
2009-10-21 17:35:28.682475 [DEBUG] mod_dptools.c:748
sofia/internal/1001 at singtel SET [open]=[true]
EXECUTE sofia/internal/1001 at singtelhash(insert/10.4.62.7-spymap/1001/5c3ebda2-be57-11de-a6dd-e7de0b74bdc9)
EXECUTE sofia/internal/1001 at singtelhash(insert/10.4.62.7-last_dial/1001/05492122)
EXECUTE sofia/internal/1001 at singtelhash(insert/10.4.62.7-last_dial/global/5c3ebda2-be57-11de-a6dd-e7de0b74bdc9)
EXECUTE sofia/internal/1001 at singtelset(effective_caller_id_number=1001282122)
2009-10-21 17:35:28.685219 [DEBUG] mod_dptools.c:748
sofia/internal/1001 at singtel SET [effective_caller_id_number]=[1001282122]
EXECUTE sofia/internal/1001 at singtel set(NCX_IP=10.4.4.254)
2009-10-21 17:35:28.685219 [DEBUG] mod_dptools.c:748
sofia/internal/1001 at singtel SET [NCX_IP]=[10.4.4.254]
EXECUTE sofia/internal/1001 at singtel set(call_timeout=30)
2009-10-21 17:35:28.686292 [DEBUG] mod_dptools.c:748
sofia/internal/1001 at singtel SET [call_timeout]=[30]
EXECUTE sofia/internal/1001 at singtel set(hangup_after_bridge=true)
2009-10-21 17:35:28.686292 [DEBUG] mod_dptools.c:748
sofia/internal/1001 at singtel SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/1001 at singtel set(bypass_media=false)
2009-10-21 17:35:28.686292 [DEBUG] mod_dptools.c:748
sofia/internal/1001 at singtel SET [bypass_media]=[false]
EXECUTE sofia/internal/1001 at singtel set(proxy_media=true)
2009-10-21 17:35:28.686292 [DEBUG] mod_dptools.c:748
sofia/internal/1001 at singtel SET [proxy_media]=[true]
EXECUTE sofia/internal/1001 at singtel bridge(h323/05492122 at 10.4.4.254)
2009-10-21 17:35:28.687357 [DEBUG] mod_h323.cpp:1380
======>create_outgoing_channel DST NUMBER = 05492122 at 10.4.4.254
2009-10-21 17:35:28.687357 [DEBUG] mod_h323.cpp:1336
======>FSH323EndPoint::FSMakeCall DST NUMBER =
05492122 at 10.4.4.254[FSH323EndPoint]
2009-10-21 17:35:28.687357 [DEBUG] mod_h323.cpp:1348 ----> Using 10.4.62.7
for outbound call
2009-10-21 17:35:28.687357 [DEBUG] pdns.cxx:498 SRV Lookup 10.4.4.254
service _h323cs._tcp.
2009-10-21 17:35:28.689538 [DEBUG] pdns.cxx:471 SRV Lookup Fail no domain
h323:05492122 at 10.4.4.254 <h323%3A05492122 at 10.4.4.254>
2009-10-21 17:35:28.689538 [DEBUG] h323ep.cxx:2315 DNS SRV Cannot resolve
remote party 05492122 at 10.4.4.254
2009-10-21 17:35:28.689538 [DEBUG] h323ep.cxx:2103 Making call to:
05492122 at 10.4.4.254
2009-10-21 17:35:28.689538 [DEBUG] mod_h323.cpp:483
======>FSH323EndPoint::CreateConnection callReference = 12651 userDate =
0xb5e0dac0 [FSH323EndPoint]
2009-10-21 17:35:28.689538 [DEBUG] mod_h323.cpp:486 ------>
SWITCH_CALL_DIRECTION_OUTBOUND
2009-10-21 17:35:28.689538 [DEBUG] mod_h323.cpp:496 ------> fsSession =
0xb7569548
2009-10-21 17:35:28.689538 [DEBUG] h323caps.cxx:3186 Added capability:
G.711-ALaw-64k <1>
2009-10-21 17:35:28.689538 [DEBUG] h323caps.cxx:3186 Added capability:
G.711-uLaw-64k <2>
2009-10-21 17:35:28.689538 [DEBUG] h323caps.cxx:3186 Added capability:
GSM-06.10{sw} <3>
2009-10-21 17:35:28.689538 [DEBUG] h323caps.cxx:3186 Added capability:
G.729A/B{sw} <4>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
G.729A{sw} <5>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
G.729B{sw} <6>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
G.729{sw} <7>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
UserInput/hookflash <8>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
UserInput/basicString <9>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
UserInput/dtmf <10>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
UserInput/RFC2833 <11>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
UserInput/Navigation <12>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
UserInput/Softkey <13>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
UserInput/PointDevice <14>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
UserInput/Modal <15>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 1
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
G.711-ALaw-64k <1>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 2
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
G.711-uLaw-64k <2>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 3
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
GSM-06.10{sw} <3>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 4
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
G.729A/B{sw} <4>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 5
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
G.729A{sw} <5>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 6
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
G.729B{sw} <6>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 7
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
G.729{sw} <7>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 8
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
UserInput/hookflash <8>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 9
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
UserInput/basicString <9>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 10
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
UserInput/dtmf <10>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 11
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
UserInput/RFC2833 <11>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 12
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
UserInput/Navigation <12>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 13
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
UserInput/Softkey <13>
2009-10-21 17:35:28.691688 [DEBUG] h323caps.cxx:3248 FindCapability: 14
2009-10-21 17:35:28.691688 [DEBUG] h323caps.cxx:3252 Found capability:
UserInput/PointDevice <14>
2009-10-21 17:35:28.691688 [DEBUG] h323caps.cxx:3248 FindCapability: 15
2009-10-21 17:35:28.691688 [DEBUG] h323caps.cxx:3252 Found capability:
UserInput/Modal <15>
2009-10-21 17:35:28.691688 [DEBUG] h4601.cxx:1725 Endpoint Attached
2009-10-21 17:35:28.691688 [DEBUG] rfc2833.cxx:87 Handler created
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:525
======>FSH323Connection::FSH323Connection FSH323Connection
2009-10-21 17:35:28.691688 [NOTICE] switch_channel.c:602 New Channel h323/
05492122 at 10.4.4.254 [5c413af0-be57-11de-a6dd-e7de0b74bdc9]
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:546 (h323/
05492122 at 10.4.4.254) State Change CS_NEW -> CS_INIT
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:958
======>FSH323Connection::state_change FSH323Connection
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:959 State changed on
connection FSH323Connection
2009-10-21 17:35:28.691688 [DEBUG] switch_core_session.c:932 Send signal
h323/05492122 at 10.4.4.254 [BREAK]
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:831
======>FSH323Connection::kill_channel FSH323Connection
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:832 Kill 3 on connection
FSH323Connection
2009-10-21 17:35:28.691688 [DEBUG] h323ep.cxx:2183 Created new connection:
ip$localhost/12651
2009-10-21 17:35:28.691688 [DEBUG] osutil.cxx:188 File handle high water
mark set: 58 Thread unblock pipe
2009-10-21 17:35:28.691688 [DEBUG] h323ep.cxx:1101 Started call thread
2009-10-21 17:35:28.691688 [CRIT] switch_core_state_machine.c:239 Obtained
10 stack frames.
/usr/local/freeswitch/lib/libfreeswitch.so.1 [0xb7f511f5]
[0xb80be400]
/usr/local/freeswitch/mod/mod_h323.so [0xb6953cc9]
/usr/local/freeswitch/lib/libfreeswitch.so.1(switch_core_session_outgoing_channel+0x1ad)
[0xb7f4f59d]
/usr/local/freeswitch/lib/libfreeswitch.so.1(switch_ivr_originate+0x1b2b)
[0xb7f7aa1b]
/usr/local/freeswitch/mod/mod_dptools.so [0xb70ea2d5]
/usr/local/freeswitch/lib/libfreeswitch.so.1(switch_core_session_exec+0x284)
[0xb7f4c5a4]
/usr/local/freeswitch/lib/libfreeswitch.so.1(switch_core_session_execute_application+0x8b)
[0xb7f4cacb]
/usr/local/freeswitch/lib/libfreeswitch.so.1(switch_core_session_run+0x12b8)
[0xb7f52658]
/usr/local/freeswitch/lib/libfreeswitch.so.1 [0xb7f4e6c5]
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:509
======>FSH323EndPoint::OnSetGatewayPrefixes FSH323EndPoint
2009-10-21 17:35:28.691688 [CRIT] switch_core_state_machine.c:352 Thread has
crashed for channel sofia/internal/1001 at singtel
2009-10-21 17:35:28.691688 [NOTICE] switch_core_state_machine.c:353 Hangup
sofia/internal/1001 at singtel [CS_EXECUTE] [CRASH]
2009-10-21 17:35:28.691688 [DEBUG] transports.cxx:1828 Connecting to
10.4.4.254:1720 (local port=0)
2009-10-21 17:35:28.691688 [DEBUG] osutil.cxx:188 File handle high water
mark set: 59 PTCPSocket
2009-10-21 17:35:28.691688 [DEBUG] switch_channel.c:1683 Send signal
sofia/internal/1001 at singtel [KILL]
2009-10-21 17:35:28.691688 [DEBUG] switch_core_session.c:932 Send signal
sofia/internal/1001 at singtel [BREAK]
2009-10-21 17:35:28.691688 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/1001 at singtel) Running State Change CS_HANGUP
2009-10-21 17:35:28.691688 [DEBUG] transports.cxx:1681 Started connection:
host=10.4.4.254:1720, if=10.4.62.7:52984, handle=59
2009-10-21 17:35:28.691688 [DEBUG] h323.cxx:2680 Sending Setup PDU
2009-10-21 17:35:28.691688 [DEBUG] switch_core_state_machine.c:434
(sofia/internal/1001 at singtel) State HANGUP
2009-10-21 17:35:28.691688 [DEBUG] mod_sofia.c:338 Channel
sofia/internal/1001 at singtel hanging up, cause: CRASH
2009-10-21 17:35:28.691688 [DEBUG] mod_sofia.c:417 Responding to INVITE
with: 480
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:559
======>FSH323Connection::OnSetLocalCapabilities() [FSH323Connection]
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:565
======>FSH323Connection::SetLocalCapabilities() Size local capability = 15
[FSH323Connection]
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
======>FSH323Connection::decodeCapability
2009-10-21 17:35:28.691688 [INFO] mod_h323.cpp:619 capability
'G.711-ALaw-64k' format 'PCMU' payload 8
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
======>FSH323Connection::decodeCapability
2009-10-21 17:35:28.691688 [INFO] mod_h323.cpp:619 capability
'G.711-uLaw-64k' format 'PCMA' payload 0
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
======>FSH323Connection::decodeCapability
2009-10-21 17:35:28.691688 [INFO] mod_h323.cpp:619 capability 'GSM-06.10'
format 'GSM' payload 3
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
======>FSH323Connection::decodeCapability
2009-10-21 17:35:28.691688 [INFO] mod_h323.cpp:619 capability 'G.729A/B'
format 'G729' payload 18
2009-10-21 17:35:28.691688 [DEBUG] switch_core_state_machine.c:46
sofia/internal/1001 at singtel Standard HANGUP, cause: CRASH
2009-10-21 17:35:28.691688 [DEBUG] switch_core_state_machine.c:434
(sofia/internal/1001 at singtel) State HANGUP going to sleep
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
======>FSH323Connection::decodeCapability
2009-10-21 17:35:28.691688 [INFO] mod_h323.cpp:619 capability 'G.729A'
format 'G729' payload 18
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
======>FSH323Connection::decodeCapability
2009-10-21 17:35:28.691688 [INFO] mod_h323.cpp:619 capability 'G.729B'
format 'G729b' payload 18
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
======>FSH323Connection::decodeCapability



Well, I'm not sure if the backtrace is from 1.0.4 ... i will disable
crass-protection and will send new logs to you.


Also, if you like i can give you access to the machine itself...

T.
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From jerry.richards at teotech.com  Wed Oct 21 09:11:04 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 21 Oct 2009 09:11:04 -0700
Subject: [Freeswitch-users] Stop/Restart of Freeswitch Causes Crash
Message-ID: <08086CB11D744A97A3FD87B383F43EA6@greyhawk.tonecommander.com>


Sometimes if I stop (using "..." command) and then restart freeswitch (using
"./freeswitch" command), the program will crash and return to the Linux
(CentOS 5.3) prompt.  I am using version 1.0.4.

I just pasted the freeswitch/terminal log into the Pastebin.

Best Regards,
Jerry



From diego.viola at gmail.com  Wed Oct 21 09:31:32 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Wed, 21 Oct 2009 16:31:32 +0000
Subject: [Freeswitch-users] Stop/Restart of Freeswitch Causes Crash
In-Reply-To: <08086CB11D744A97A3FD87B383F43EA6@greyhawk.tonecommander.com>
References: <08086CB11D744A97A3FD87B383F43EA6@greyhawk.tonecommander.com>
Message-ID: <86a32abc0910210931l36072b47ic92d8550f190bdeb@mail.gmail.com>

You should debug FreeSWITCH, check this out:

http://wiki.freeswitch.org/wiki/Debugging_Freeswitch

And then open a Jira.

Diego

On Wed, Oct 21, 2009 at 4:11 PM, Jerry Richards
<jerry.richards at teotech.com> wrote:
>
> Sometimes if I stop (using "..." command) and then restart freeswitch (using
> "./freeswitch" command), the program will crash and return to the Linux
> (CentOS 5.3) prompt. ?I am using version 1.0.4.
>
> I just pasted the freeswitch/terminal log into the Pastebin.
>
> Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From anthony.minessale at gmail.com  Wed Oct 21 09:33:55 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Oct 2009 11:33:55 -0500
Subject: [Freeswitch-users] Stop/Restart of Freeswitch Causes Crash
In-Reply-To: <08086CB11D744A97A3FD87B383F43EA6@greyhawk.tonecommander.com>
References: <08086CB11D744A97A3FD87B383F43EA6@greyhawk.tonecommander.com>
Message-ID: <191c3a030910210933l6217179bk866fedef1007d716@mail.gmail.com>

That problem is fixed if you update to SVN trunk or latest snapshot.


On Wed, Oct 21, 2009 at 11:11 AM, Jerry Richards <jerry.richards at teotech.com
> wrote:

>
> Sometimes if I stop (using "..." command) and then restart freeswitch
> (using
> "./freeswitch" command), the program will crash and return to the Linux
> (CentOS 5.3) prompt.  I am using version 1.0.4.
>
> I just pasted the freeswitch/terminal log into the Pastebin.
>
> Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Wed Oct 21 09:39:31 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Oct 2009 11:39:31 -0500
Subject: [Freeswitch-users] Call custom variable in condition
In-Reply-To: <65d96fc80910210413t1123dbc0xd5740bc0673d638c@mail.gmail.com>
References: <a60fed5e0910210337n45ffefe2wc3e3d9d0b403a568@mail.gmail.com>
	<65d96fc80910210413t1123dbc0xd5740bc0673d638c@mail.gmail.com>
Message-ID: <191c3a030910210939k7dbf1494gf713df8770356450@mail.gmail.com>

It not only makes sense it's well documented on the wiki page.
The set line is not happening right when it's encountered, the set line is
copied into the channel and executed later after the whole dialplan is
parsed.  The dialplan is a pre-processor not a runtime engine.

Here is a new feature in pre-1.0.5 (svn trunk)

Some applications like set can now be executed within the dialplan but you
should use it sparingly.
<action application="set" data="testing=true" inline="true"/>

The inline=true makes it execute inside the dialplan and it's never copied
into your resulting extension because it's executed immediately.



On Wed, Oct 21, 2009 at 6:13 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> consider this:
>
>
>
> <context name="SIP_incoming">
>    <extension name="call-sip-extensions">
>       <condition field="destination_number" expression="^(\d+)$">
>           <action application="set" data="AUTHENTICATION_STATUS=0"/>
>            <action application="transfer" data="${AUTHENTICATION_STATUS}
> XML Authen_Status"/>
>       </condition>
>    </extension>
> </context>
>
>
>
> <context name="Authen_Status">
>      <extension name="exten-auth-status">
>        <condition field="${AUTHENTICATION_STATUS}" expression="^0$">
>           <action application="answer"/>
>           <action application="playback" data="play.wav"/>
>       </condition>
>     </extension>
>   </context>
>
>
>
>
>
> here is one of my dialplan. I'm using execute_extension but it is quite the
> same...
>
>
>
>    <extension name="ServiceLookup">
>       <condition field="destination_number" expression="(^300030)(.*)">
>          <action application="lookup_service_destination" data="in
> ${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, in $
> 1, in ${network_addr}:5060, out red_contact, out authResult"/>
>          <action application="log" data="INFO ########################
> ServiceLookup ########################\n"/>
>          <action application="log" data="INFO ########################
> contact = '${red_contact}' ##############\n"/>
>          <action application="log" data="INFO ########################
> CallerNum = '${caller_id_number:6:16}' ##########\n"/>
>          <action application="log" data="INFO ########################
> RADIUS auth = '${authResult}' ##########\n"/>
>
>          <action application="execute_extension" data="doRedirect XML
> public"/>
>         </condition>
>    </extension>
>
>
>    <extension name="doRedirect">
>       <condition field="destination_number" expression="^doRedirect$"/>
>       <condition field="${authResult}" expression="^0$|^60$">
>          <action application="log" data="INFO ########################
> RADIUS auth OK!!!' ##########\n"/>
>          <action application="redirect" data="${red_contact}"/>
>          <anti-action application="log" data="INFO ########################
> RADIUS auth NOK!! ##########\n"/>
>          <anti-action application="respond" data="403 Forbidden"/>
>       </condition>
>
>    </extension>
>
>
>
>
> On Wed, Oct 21, 2009 at 12:37 PM, Ahmed Munir <ahmedmunir007 at gmail.com>wrote:
>
>> Hi,
>>
>> I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I
>> want to use it in condition, as I'm listing down the configuration below;
>>
>> <context name="SIP_incoming">
>>    <extension name="call-sip-extensions">
>>       <condition field="destination_number" expression="^(\d+)$">
>>           <action application="set" data="AUTHENTICATION_STATUS=0"/>
>>            <action application="transfer" data="${AUTHENTICATION_STATUS}
>> XML Authen_Status"/>
>>       </condition>
>>    </extension>
>> </context>
>>
>> <context name="Authen_Status">
>>      <extension name="exten-auth-status">
>>        <condition field="AUTHENTICATION_STATUS" expression="^0$">
>>           <action application="answer"/>
>>           <action application="playback" data="play.wav"/>
>>       </condition>
>>     </extension>
>>   </context>
>>
>>
>>
>>
>>  But unfortunately it is not working. Kindly advise me how to do implement
>> it(Note: I don't want to call script). And one more thing to ask how can I
>> transfer the values within the same context?
>>
>> --
>> Regards,
>>
>> Ahmed Munir
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From q.edward at gmail.com  Wed Oct 21 09:48:52 2009
From: q.edward at gmail.com (Edward Q.)
Date: Wed, 21 Oct 2009 12:48:52 -0400
Subject: [Freeswitch-users] Hi Guys
In-Reply-To: <35b355e90910202349q31bc8f7as6e277008600d81b0@mail.gmail.com>
References: <89313a90910202120v6c563660s61f2efba22f916f@mail.gmail.com> 
	<35b355e90910202255x680232f9x45319ab835f5846b@mail.gmail.com> 
	<89313a90910202309r6751b242q6b96a56cd1bfe57@mail.gmail.com> 
	<35b355e90910202349q31bc8f7as6e277008600d81b0@mail.gmail.com>
Message-ID: <89313a90910210948w5410e16ai1b7b48ee8fe1ac25@mail.gmail.com>

Thank you guysEd

On Wed, Oct 21, 2009 at 2:49 AM, Shelby Ramsey <sicfslist at gmail.com> wrote:

> Well ... the question is pretty generic.
>
> But based on these assumptions:
>   -- no media (bypass media)
>   -- routing done via XML dialplan
>
> Something along the lines of a quad core machine with 4 gigs of ram would
> be overkill for 692 calls.
>
> Things to remember:
>   -- the more cores the better (FS is heavily threaded)
>   -- the more memory the better
>   -- 64 bit is way better than 32 bit
>
> SDR
>
>
>
>
>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Wed Oct 21 09:54:34 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Oct 2009 11:54:34 -0500
Subject: [Freeswitch-users] Stop/Restart of Freeswitch Causes Crash
In-Reply-To: <86a32abc0910210931l36072b47ic92d8550f190bdeb@mail.gmail.com>
References: <08086CB11D744A97A3FD87B383F43EA6@greyhawk.tonecommander.com>
	<86a32abc0910210931l36072b47ic92d8550f190bdeb@mail.gmail.com>
Message-ID: <191c3a030910210954g74c2d5f8p3451841a8836147@mail.gmail.com>

no you shouldnt
we don't take jiras about release revisions only trunk.


On Wed, Oct 21, 2009 at 11:31 AM, Diego Viola <diego.viola at gmail.com> wrote:

> You should debug FreeSWITCH, check this out:
>
> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch
>
> And then open a Jira.
>
> Diego
>
> On Wed, Oct 21, 2009 at 4:11 PM, Jerry Richards
> <jerry.richards at teotech.com> wrote:
> >
> > Sometimes if I stop (using "..." command) and then restart freeswitch
> (using
> > "./freeswitch" command), the program will crash and return to the Linux
> > (CentOS 5.3) prompt.  I am using version 1.0.4.
> >
> > I just pasted the freeswitch/terminal log into the Pastebin.
> >
> > Best Regards,
> > Jerry
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From kaosfrack at gmail.com  Wed Oct 21 04:11:47 2009
From: kaosfrack at gmail.com (Joey Carter)
Date: Wed, 21 Oct 2009 16:11:47 +0500
Subject: [Freeswitch-users] Is background sound|music during call possible?
Message-ID: <700164f40910210411k6ce2795cud47202a25ce27f87@mail.gmail.com>

Hello

How can I add background music that will play during call?

Thanks


From keith at laaks.com  Wed Oct 21 09:45:19 2009
From: keith at laaks.com (Keith Laaks)
Date: Wed, 21 Oct 2009 18:45:19 +0200
Subject: [Freeswitch-users] Calls dropping on SIP timer expiry due to
	UPDATE's being ignored.
Message-ID: <1256143519.7836.107.camel@keithl-lt>

Hi,

Hope someone knows how I am able to get around this one. Here goes...

Did an upgrade to trunk (from a July vintage build) last week and
noticed calls out to a provider were now failing after about 30 seconds
or so - post answer. Tried latest (15183) - same thing.

Analysing, I see that I have multiple UPDATE messages now being sent to
the provider, but no response being sent back to FS. So FS times out and
eventually kills the call.
Interestingly, it only drops the A-leg; the B-leg remains up till the B
party hangs up.

I cant recall seeing these UPDATE messages before...

The intent of the UPDATE seems to be to send the callee name & number to
the B-leg.

If its the provider's sip stack that's broken w.r.t. handling UPDATE -
is there any way to get around it by doing something in my config to
ensure these UPDATE's are not 'triggered' ?


Some traces below. 


Any suggestions welcomed...

Best Regards

Keith
Pretoria, South Africa.

--------------------------------------------------------------------------------------------------------------------------------------------------


send 1048 bytes to udp/[196.10.11.12]:5060 at 13:24:04.249269:

------------------------------------------------------------------------
   INVITE sip:27835551111 at 196.10.11.12 SIP/2.0
   Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bKyQepDXQ5H8g5m
   Max-Forwards: 67
   From: "Keith PhoneADSL" <sip:878050000 at 10.17.10.10>;tag=Upa3NvXpBB1eF
   To: <sip:27835551111 at 196.10.11.12>
   Call-ID: d821359d-38e7-122d-a38e-002264cc9b93
   CSeq: 121947386 INVITE
   Contact: <sip:gw+vprov at 10.17.10.10:5060;transport=udp;gw=vprov>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 247
   X-Actually-Support: UPDATE
   Remote-Party-ID: "Keith PhoneADSL"
<sip:878050000 at 10.17.10.10>;party=calling;screen=yes;privacy=off
   
   v=0
   o=FreeSWITCH 1256118582 1256118583 IN IP4 10.17.10.10
   s=FreeSWITCH
   c=IN IP4 10.17.10.10
   t=0 0
   m=audio 12862 RTP/AVP 18 101
   a=rtpmap:18 G729/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20

------------------------------------------------------------------------
2009-10-21 15:24:04.248448 [DEBUG] sofia.c:3493 Channel
sofia/vvrf/27835551111 entering state [calling][0]
recv 601 bytes from udp/[196.10.11.12]:5060 at 13:24:04.307690:

------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bKyQepDXQ5H8g5m
   From: "Keith PhoneADSL" <sip:878050000 at 10.17.10.10>;tag=Upa3NvXpBB1eF
   To: <sip:27835551111 at 196.10.11.12>;tag=GR52RWG346-34
   Call-ID: d821359d-38e7-122d-a38e-002264cc9b93
   CSeq: 121947386 INVITE
   Contact: "vprov C5CM" <sip:196.10.11.12:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M
   Allow-Events: talk
   Allow-Events: refer
   Content-Disposition: session
   X-Actually-Support: UPDATE
   Remote-Party-ID: "Keith PhoneADSL"
<sip:878050000 at 10.17.10.10>;party=calling;screen=yes;privacy=off
   Content-Length: 0
   

------------------------------------------------------------------------
recv 879 bytes from udp/[196.10.11.12]:5060 at 13:24:08.508162:

------------------------------------------------------------------------
   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bKyQepDXQ5H8g5m
   From: "Keith PhoneADSL" <sip:878050000 at 10.17.10.10>;tag=Upa3NvXpBB1eF
   To: <sip:27835551111 at 196.10.11.12>;tag=GR52RWG346-34
   Call-ID: d821359d-38e7-122d-a38e-002264cc9b93
   CSeq: 121947386 INVITE
   Contact: "vprov C5CM" <sip:196.10.11.12:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M
   Allow-Events: talk
   Allow-Events: refer
   Content-Disposition: session
   X-Actually-Support: UPDATE
   Remote-Party-ID: "Keith PhoneADSL"
<sip:878050000 at 10.17.10.10>;party=calling;screen=yes;privacy=off
   Content-Type: application/sdp
   Content-Length:   233
   
   v=0
   o=Clarent 152602 152603 IN IP4 196.10.11.15
   s=Clarent C5CM
   c=IN IP4 196.10.11.15
   t=0 0
   m=audio 5230 RTP/AVP 18 101
   a=rtpmap:18 G729/8000
   a=ptime:20
   a=fmtp:18 annexb=no
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15

------------------------------------------------------------------------
2009-10-21 15:24:08.507506 [DEBUG] sofia.c:3493 Channel
sofia/vvrf/27835551111 entering state [proceeding][183]
2009-10-21 15:24:08.507506 [DEBUG] sofia.c:3500 Remote SDP:
v=0
o=Clarent 152602 152603 IN IP4 196.10.11.15
s=Clarent C5CM
c=IN IP4 196.10.11.15
t=0 0
m=audio 5230 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

2009-10-21 15:24:08.507506 [DEBUG] sofia_glue.c:3144 Audio Codec Compare
[G729:18:8000:20]/[G729:18:8000:20]
2009-10-21 15:24:08.508561 [DEBUG] sofia_glue.c:2102 Set Codec
sofia/vvrf/27835551111 G729/8000 20 ms 160 samples
2009-10-21 15:24:08.508561 [DEBUG] sofia_glue.c:3104 Set 2833 dtmf
payload to 101
2009-10-21 15:24:08.508561 [DEBUG] sofia_glue.c:2336 AUDIO RTP
[sofia/vvrf/27835551111] 10.17.10.10 port 12862 -> 196.10.11.15 port
5230 codec: 18 ms: 20
2009-10-21 15:24:08.508561 [DEBUG] switch_rtp.c:1155 Starting timer
[soft] 160 bytes per 20ms
2009-10-21 15:24:08.509616 [NOTICE] sofia_glue.c:2771 Pre-Answer
sofia/vvrf/27835551111!
2009-10-21 15:24:08.509616 [DEBUG] switch_channel.c:1992 Send signal
sofia/inetticky/27878050000 at 196.222.3.4 [BREAK]
2009-10-21 15:24:08.510671 [DEBUG] switch_ivr_originate.c:615 Setting
codec string on sofia/inetticky/27878050000 at 196.222.3.4 to
G729 at 8000h@20i
2009-10-21 15:24:08.510671 [DEBUG] switch_ivr_originate.c:2326
sofia/inetticky/27878050000 at 196.222.3.4 receive message [PROGRESS]
2009-10-21 15:24:08.510671 [INFO] switch_ivr_originate.c:2326 Sending
early media
2009-10-21 15:24:08.510671 [DEBUG] sofia_glue.c:3144 Audio Codec Compare
[G729:18:8000:0]/[G729:18:8000:20]
2009-10-21 15:24:08.510671 [DEBUG] sofia_glue.c:2102 Set Codec
sofia/inetticky/27878050000 at 196.222.3.4 G729/8000 20 ms 160 samples
2009-10-21 15:24:08.510671 [DEBUG] sofia_glue.c:3104 Set 2833 dtmf
payload to 101
2009-10-21 15:24:08.510671 [DEBUG] sofia_glue.c:2336 AUDIO RTP
[sofia/inetticky/27878050000 at 196.222.3.4] 196.222.3.4 port 16840 ->
41.242.6.167 port 62936 codec: 18 ms: 20
2009-10-21 15:24:08.510671 [DEBUG] switch_rtp.c:1155 Starting timer
[soft] 160 bytes per 20ms
2009-10-21 15:24:08.511769 [NOTICE] sofia_glue.c:2771 Pre-Answer
sofia/inetticky/27878050000 at 196.222.3.4!
2009-10-21 15:24:08.511769 [INFO] mod_sofia.c:1582 Ring SDP:
v=0
o=FreeSWITCH 1256114608 1256114609 IN IP4 196.222.3.4
s=FreeSWITCH
c=IN IP4 196.222.3.4
t=0 0
m=audio 16840 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2009-10-21 15:24:08.511769 [DEBUG] switch_core_session.c:712 Send signal
sofia/vvrf/27835551111 [BREAK]
2009-10-21 15:24:08.511769 [DEBUG] switch_core_session.c:653 Send signal
sofia/inetticky/27878050000 at 196.222.3.4 [BREAK]
2009-10-21 15:24:08.511769 [DEBUG] sofia.c:407 sofia/vvrf/27835551111
receive message [DISPLAY]
2009-10-21 15:24:08.511769 [DEBUG] switch_ivr_originate.c:2368 Originate
Resulted in Success: [sofia/vvrf/27835551111]
2009-10-21 15:24:08.511769 [DEBUG] switch_channel.c:182
sofia/vvrf/27835551111 receive message [AUDIO_SYNC]
2009-10-21 15:24:08.511769 [DEBUG] switch_channel.c:182
sofia/inetticky/27878050000 at 196.222.3.4 receive message [AUDIO_SYNC]
2009-10-21 15:24:08.512832 [DEBUG] switch_ivr_bridge.c:975
sofia/vvrf/27835551111 receive message [BRIDGE]
2009-10-21 15:24:08.512832 [DEBUG] switch_core_session.c:653 Send signal
sofia/vvrf/27835551111 [BREAK]
2009-10-21 15:24:08.512832 [DEBUG] switch_ivr_bridge.c:982
sofia/inetticky/27878050000 at 196.222.3.4 receive message [BRIDGE]
send 973 bytes to udp/[41.242.6.167]:62928 at 13:24:08.512974:

------------------------------------------------------------------------
   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP
41.242.6.167:56675;branch=z9hG4bK2234c0462cdad03fc318bd6c7438282;rport=629282009-10-21 15:24:08.512832 [DEBUG] switch_core_session.c:653 Send signal sofia/inetticky/27878050000 at 196.222.3.4 [BREAK]
2009-10-21 15:24:08.512832 [DEBUG] switch_ivr_bridge.c:1026
(sofia/vvrf/27835551111) State Change CS_CONSUME_MEDIA ->
CS_EXCHANGE_MEDIA
2009-10-21 15:24:08.512832 [DEBUG] switch_core_session.c:985 Send signal
sofia/vvrf/27835551111 [BREAK]
2009-10-21 15:24:08.512832 [DEBUG] switch_core_state_machine.c:306
(sofia/vvrf/27835551111) Running State Change CS_EXCHANGE_MEDIA
2009-10-21 15:24:08.512832 [DEBUG] switch_core_state_machine.c:343
(sofia/vvrf/27835551111) State EXCHANGE_MEDIA
2009-10-21 15:24:08.512832 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK

   From: "Keith PhoneADSL" <sip:27878050000 at 196.222.3.4>;tag=3283506127
   To: <sip:0835551111 at 196.222.3.4>;user=phone;tag=g2DFFpS0QN4cc
   Call-ID: 4252084844 at 192_168_1_65
   CSeq: 3 INVITE
   Contact: <sip:0835551111 at 196.222.3.4:5060;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 249
   X-Actually-Support: UPDATE
   
   v=0
   o=FreeSWITCH 1256114608 1256114609 IN IP4 196.222.3.4
   s=FreeSWITCH
   c=IN IP4 196.222.3.4
   t=0 0
   m=audio 16840 RTP/AVP 18 101
   a=rtpmap:18 G729/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20

------------------------------------------------------------------------
2009-10-21 15:24:08.512832 [DEBUG] sofia.c:3493 Channel
sofia/inetticky/27878050000 at 196.222.3.4 entering state [early][183]
recv 928 bytes from udp/[196.10.11.12]:5060 at 13:24:14.188781:

------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bKyQepDXQ5H8g5m
   From: "Keith PhoneADSL" <sip:878050000 at 10.17.10.10>;tag=Upa3NvXpBB1eF
   To: <sip:27835551111 at 196.10.11.12>;tag=GR52RWG346-34
   Call-ID: d821359d-38e7-122d-a38e-002264cc9b93
   CSeq: 121947386 INVITE
   Contact: "vprov C5CM" <sip:196.10.11.12:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M
   Allow-Events: talk
   Allow-Events: refer
   Content-Disposition: session
   X-Actually-Support: UPDATE
   Remote-Party-ID: "Keith PhoneADSL"
<sip:878050000 at 10.17.10.10>;party=calling;screen=yes;privacy=off
   Allow: ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE
   Content-Type: application/sdp
   Content-Length:   233
   
   v=0
   o=Clarent 153288 153288 IN IP4 196.10.11.15
   s=Clarent C5CM
   c=IN IP4 196.10.11.15
   t=0 0
   m=audio 5230 RTP/AVP 18 101
   a=rtpmap:18 G729/8000
   a=ptime:20
   a=fmtp:18 annexb=no
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15

------------------------------------------------------------------------
send 396 bytes to udp/[196.10.11.12]:5060 at 13:24:14.189032:

------------------------------------------------------------------------
   ACK sip:196.10.11.12:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bKZ07eFr88eH7Qg
   Max-Forwards: 70
   From: "Keith PhoneADSL" <sip:878050000 at 10.17.10.10>;tag=Upa3NvXpBB1eF
   To: <sip:27835551111 at 196.10.11.12>;tag=GR52RWG346-34
   Call-ID: d821359d-38e7-122d-a38e-002264cc9b93
   CSeq: 121947386 ACK
   Contact: <sip:gw+vprov at 10.17.10.10:5060;transport=udp;gw=vprov>
   Content-Length: 0
   

------------------------------------------------------------------------
2009-10-21 15:24:14.188910 [DEBUG] sofia.c:3493 Channel
sofia/vvrf/27835551111 entering state [ready][200]
2009-10-21 15:24:14.188910 [DEBUG] switch_channel.c:2105 Send signal
sofia/inetticky/27878050000 at 196.222.3.4 [BREAK]
2009-10-21 15:24:14.188910 [NOTICE] sofia.c:3928 Channel
[sofia/vvrf/27835551111] has been answered
2009-10-21 15:24:14.188910 [DEBUG] switch_channel.c:182
sofia/vvrf/27835551111 receive message [AUDIO_SYNC]
2009-10-21 15:24:14.204823 [DEBUG] switch_ivr_bridge.c:378
sofia/inetticky/27878050000 at 196.222.3.4 receive message [ANSWER]
2009-10-21 15:24:14.204823 [DEBUG] mod_sofia.c:543 Local SDP
sofia/inetticky/27878050000 at 196.222.3.4:
v=0
o=FreeSWITCH 1256114608 1256114610 IN IP4 196.222.3.4
s=FreeSWITCH
c=IN IP4 196.222.3.4
t=0 0
m=audio 16840 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2009-10-21 15:24:14.204823 [DEBUG] switch_core_session.c:712 Send signal
sofia/vvrf/27835551111 [BREAK]
2009-10-21 15:24:14.204823 [DEBUG] switch_core_session.c:653 Send signal
sofia/inetticky/27878050000 at 196.222.3.4 [BREAK]
2009-10-21 15:24:14.204823 [NOTICE] switch_ivr_bridge.c:378 Channel
[sofia/inetticky/27878050000 at 196.222.3.4] has been answered
2009-10-21 15:24:14.204823 [DEBUG] switch_channel.c:182
sofia/inetticky/27878050000 at 196.222.3.4 receive message [AUDIO_SYNC]
send 983 bytes to udp/[41.242.6.167]:62928 at 13:24:14.205173:

------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP
41.242.6.167:56675;branch=z9hG4bK2234c0462cdad03fc318bd6c7438282;rport=62928
   From: "Keith PhoneADSL" <sip:27878050000 at 196.222.3.4>;tag=3283506127
   To: <sip:0835551111 at 196.222.3.4>;user=phone;tag=g2DFFpS0QN4cc
   Call-ID: 4252084844 at 192_168_1_65
   CSeq: 3 INVITE
   Contact: <sip:0835551111 at 196.222.3.4:5060;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, refer
   Session-Expires: 120;refresher=uas
   Min-SE: 120
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 249
   X-Actually-Support: UPDATE
   
   v=0
   o=FreeSWITCH 1256114608 1256114609 IN IP4 196.222.3.4
   s=FreeSWITCH
   c=IN IP4 196.222.3.4
   t=0 0
   m=audio 16840 RTP/AVP 18 101
   a=rtpmap:18 G729/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20

------------------------------------------------------------------------
2009-10-21 15:24:14.204823 [DEBUG] sofia.c:3493 Channel
sofia/inetticky/27878050000 at 196.222.3.4 entering state [completed][200]
2009-10-21 15:24:14.224959 [DEBUG] sofia.c:407 sofia/vvrf/27835551111
receive message [DISPLAY]
send 947 bytes to udp/[196.10.11.12]:5060 at 13:24:14.225162:

------------------------------------------------------------------------
   UPDATE sip:196.10.11.12:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bK0907gKScctXac
   Max-Forwards: 70
   From: "Keith PhoneADSL" <sip:878050000 at 10.17.10.10>;tag=Upa3NvXpBB1eF
   To: <sip:27835551111 at 196.10.11.12>;tag=GR52RWG346-34
   Call-ID: d821359d-38e7-122d-a38e-002264cc9b93
   CSeq: 121947387 UPDATE
   Contact: <sip:gw+vprov at 10.17.10.10:5060;transport=udp;gw=vprov>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY
   Supported: timer, precondition, path, replaces
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 247
   P-Asserted-Identity: "unknown" <27835551111>
   
   v=0
   o=FreeSWITCH 1256118582 1256118583 IN IP4 10.17.10.10
   s=FreeSWITCH
   c=IN IP4 10.17.10.10
   t=0 0
   m=audio 12862 RTP/AVP 18 101
   a=rtpmap:18 G729/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20

------------------------------------------------------------------------
2009-10-21 15:24:14.224959 [DEBUG] sofia.c:3493 Channel
sofia/vvrf/27835551111 entering state [calling][0]
recv 733 bytes from udp/[41.242.6.167]:62928 at 13:24:14.533488:

------------------------------------------------------------------------
   ACK sip:0835551111 at 196.222.3.4:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP
41.242.6.167:56675;branch=z9hG4bKd2c91e1afbf0880332fbeba5fc9c04a6;rport
   From: "Keith PhoneADSL" <sip:27878050000 at 196.222.3.4>;tag=3283506127
   To: <sip:0835551111 at 196.222.3.4>;user=phone;tag=g2DFFpS0QN4cc
   Call-ID: 4252084844 at 192_168_1_65
   CSeq: 3 ACK
   Contact: <sip:27878050000 at 41.242.6.167:56675>
   Proxy-Authorization: Digest username="27878050000",
realm="196.222.3.4", qop=auth, algorithm=MD5,
uri="sip:0835551111 at 196.222.3.4",
nonce="0092f192-be45-11de-a2c1-a70e5037c20e", nc=00000001,
cnonce="ab2d49987adb74997fc6b236fc16d23",
response="445ae74c1f910e45d9aa1043402b4dbb"
   Max-Forwards: 70
   User-Agent: C455 IP021910000000
   Content-Length: 0
   

------------------------------------------------------------------------
2009-10-21 15:24:14.533538 [DEBUG] sofia.c:3493 Channel
sofia/inetticky/27878050000 at 196.222.3.4 entering state [ready][200]
2009-10-21 15:24:14.545161 [DEBUG] switch_core_session.c:712 Send signal
sofia/vvrf/27835551111 [BREAK]
2009-10-21 15:24:14.545161 [DEBUG] switch_core_session.c:712 Send signal
sofia/inetticky/27878050000 at 196.222.3.4 [BREAK]
2009-10-21 15:24:14.565231 [DEBUG] switch_ivr_bridge.c:122
sofia/vvrf/27835551111 receive message [DISPLAY]
2009-10-21 15:24:14.565231 [DEBUG] switch_ivr_bridge.c:122
sofia/inetticky/27878050000 at 196.222.3.4 receive message [DISPLAY]
send 947 bytes to udp/[196.10.11.12]:5060 at 13:24:14.727678:

------------------------------------------------------------------------
   UPDATE sip:196.10.11.12:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bK0907gKScctXac
   Max-Forwards: 70
   From: "Keith PhoneADSL" <sip:878050000 at 10.17.10.10>;tag=Upa3NvXpBB1eF
   To: <sip:27835551111 at 196.10.11.12>;tag=GR52RWG346-34
   Call-ID: d821359d-38e7-122d-a38e-002264cc9b93
   CSeq: 121947387 UPDATE
   Contact: <sip:gw+vprov at 10.17.10.10:5060;transport=udp;gw=vprov>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY
   Supported: timer, precondition, path, replaces
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 247
   P-Asserted-Identity: "unknown" <27835551111>
   
   v=0
   o=FreeSWITCH 1256118582 1256118583 IN IP4 10.17.10.10
   s=FreeSWITCH
   c=IN IP4 10.17.10.10
   t=0 0
   m=audio 12862 RTP/AVP 18 101
   a=rtpmap:18 G729/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20

------------------------------------------------------------------------
send 947 bytes to udp/[196.10.11.12]:5060 at 13:24:15.728255:

------------------------------------------------------------------------
   UPDATE sip:196.10.11.12:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bK0907gKScctXac
   Max-Forwards: 70
   From: "Keith PhoneADSL" <sip:878050000 at 10.17.10.10>;tag=Upa3NvXpBB1eF
   To: <sip:27835551111 at 196.10.11.12>;tag=GR52RWG346-34
   Call-ID: d821359d-38e7-122d-a38e-002264cc9b93
   CSeq: 121947387 UPDATE
   Contact: <sip:gw+vprov at 10.17.10.10:5060;transport=udp;gw=vprov>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY
   Supported: timer, precondition, path, replaces
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 247
   P-Asserted-Identity: "unknown" <27835551111>
   
   v=0
   o=FreeSWITCH 1256118582 1256118583 IN IP4 10.17.10.10
   s=FreeSWITCH
   c=IN IP4 10.17.10.10
   t=0 0
   m=audio 12862 RTP/AVP 18 101
   a=rtpmap:18 G729/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20

[Many more copies of the UPDATE's continue  ... then ]

2009-10-21 15:24:46.228024 [DEBUG] sofia.c:3493 Channel
sofia/vvrf/27835551111 entering state [terminated][408]
2009-10-21 15:24:46.228024 [NOTICE] sofia.c:4039 Hangup
sofia/vvrf/27835551111 [CS_EXCHANGE_MEDIA] [RECOVERY_ON_TIMER_EXPIRE]
2009-10-21 15:24:46.228024 [DEBUG] switch_channel.c:1896 Send signal
sofia/vvrf/27835551111 [KILL]  

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From msc at freeswitch.org  Wed Oct 21 10:02:48 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 21 Oct 2009 10:02:48 -0700
Subject: [Freeswitch-users] Call custom variable in condition
In-Reply-To: <191c3a030910210939k7dbf1494gf713df8770356450@mail.gmail.com>
References: <a60fed5e0910210337n45ffefe2wc3e3d9d0b403a568@mail.gmail.com>
	<65d96fc80910210413t1123dbc0xd5740bc0673d638c@mail.gmail.com>
	<191c3a030910210939k7dbf1494gf713df8770356450@mail.gmail.com>
Message-ID: <87f2f3b90910211002y7c35c622g4c76472fd8c7bada@mail.gmail.com>

On Wed, Oct 21, 2009 at 9:39 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> It not only makes sense it's well documented on the wiki page.
> The set line is not happening right when it's encountered, the set line is
> copied into the channel and executed later after the whole dialplan is
> parsed.  The dialplan is a pre-processor not a runtime engine.
>
> Here is a new feature in pre-1.0.5 (svn trunk)
>
> Some applications like set can now be executed within the dialplan but you
> should use it sparingly.
> <action application="set" data="testing=true" inline="true"/>
>

I'm getting ready to document this feature. For the sake of edification, why
is it best to use this sparingly, other than wide-spread use making
dialplans all cluttered?
-MC


>
> The inline=true makes it execute inside the dialplan and it's never copied
> into your resulting extension because it's executed immediately.
>
>
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From diego.viola at gmail.com  Wed Oct 21 10:04:07 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Wed, 21 Oct 2009 17:04:07 +0000
Subject: [Freeswitch-users] Stop/Restart of Freeswitch Causes Crash
In-Reply-To: <191c3a030910210954g74c2d5f8p3451841a8836147@mail.gmail.com>
References: <08086CB11D744A97A3FD87B383F43EA6@greyhawk.tonecommander.com>
	<86a32abc0910210931l36072b47ic92d8550f190bdeb@mail.gmail.com>
	<191c3a030910210954g74c2d5f8p3451841a8836147@mail.gmail.com>
Message-ID: <86a32abc0910211004u1acfb9a2hb963d63e0c7eeab9@mail.gmail.com>

Oh ok, I didn't knew that Anthony, sorry.

Do as Anthony said, update to trunk :).

On Wed, Oct 21, 2009 at 4:54 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> no you shouldnt
> we don't take jiras about release revisions only trunk.
>
>
> On Wed, Oct 21, 2009 at 11:31 AM, Diego Viola <diego.viola at gmail.com> wrote:
>>
>> You should debug FreeSWITCH, check this out:
>>
>> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch
>>
>> And then open a Jira.
>>
>> Diego
>>
>> On Wed, Oct 21, 2009 at 4:11 PM, Jerry Richards
>> <jerry.richards at teotech.com> wrote:
>> >
>> > Sometimes if I stop (using "..." command) and then restart freeswitch
>> > (using
>> > "./freeswitch" command), the program will crash and return to the Linux
>> > (CentOS 5.3) prompt. ?I am using version 1.0.4.
>> >
>> > I just pasted the freeswitch/terminal log into the Pastebin.
>> >
>> > Best Regards,
>> > Jerry
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From anthony.minessale at gmail.com  Wed Oct 21 10:10:08 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Oct 2009 12:10:08 -0500
Subject: [Freeswitch-users] Call custom variable in condition
In-Reply-To: <87f2f3b90910211002y7c35c622g4c76472fd8c7bada@mail.gmail.com>
References: <a60fed5e0910210337n45ffefe2wc3e3d9d0b403a568@mail.gmail.com>
	<65d96fc80910210413t1123dbc0xd5740bc0673d638c@mail.gmail.com>
	<191c3a030910210939k7dbf1494gf713df8770356450@mail.gmail.com>
	<87f2f3b90910211002y7c35c622g4c76472fd8c7bada@mail.gmail.com>
Message-ID: <191c3a030910211010sfb5fceco31775d3dbacfa181@mail.gmail.com>

you should not abuse it is all i mean, we have measures to limit what apps
you can use in this manner but usually requiring a more complicated dialplan
is a hint you are doing something wrong ;)

On Wed, Oct 21, 2009 at 12:02 PM, Michael Collins <msc at freeswitch.org>wrote:

>
>
> On Wed, Oct 21, 2009 at 9:39 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> It not only makes sense it's well documented on the wiki page.
>> The set line is not happening right when it's encountered, the set line is
>> copied into the channel and executed later after the whole dialplan is
>> parsed.  The dialplan is a pre-processor not a runtime engine.
>>
>> Here is a new feature in pre-1.0.5 (svn trunk)
>>
>> Some applications like set can now be executed within the dialplan but you
>> should use it sparingly.
>> <action application="set" data="testing=true" inline="true"/>
>>
>
> I'm getting ready to document this feature. For the sake of edification,
> why is it best to use this sparingly, other than wide-spread use making
> dialplans all cluttered?
> -MC
>
>
>>
>> The inline=true makes it execute inside the dialplan and it's never copied
>> into your resulting extension because it's executed immediately.
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mike at jerris.com  Wed Oct 21 10:29:32 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 21 Oct 2009 13:29:32 -0400
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <7b8592a0910210458s28ddffbdi77cf116d33c28405@mail.gmail.com>
References: <FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
	<20091020133437.GA4941@dan.olp.net>
	<7b8592a0910200716q4a7db93cyb955595e4993ac96@mail.gmail.com>
	<858EC056-1D84-4F30-A0B7-9A70FF98695D@freeswitch.org>
	<7b8592a0910200821m18fe9b4dg73dda05a10699af2@mail.gmail.com>
	<642B6CED-CDF3-4D40-B493-0F2D6222C30B@freeswitch.org>
	<7b8592a0910200947k6ca5a391l35786f0e855ae388@mail.gmail.com>
	<20091021040343.GA9440@jdc.jasonjgw.net>
	<7b8592a0910202336x1bf2a8a2nd636219ec7031c5a@mail.gmail.com>
	<20091021065841.GA13150@jdc.jasonjgw.net>
	<7b8592a0910210458s28ddffbdi77cf116d33c28405@mail.gmail.com>
Message-ID: <67234FAF-73DE-462E-B7E8-4BDE922D7CB8@jerris.com>

This appears to be some sort of ice implementation?  We don't support  
sip ice at this time.

Mike

On Oct 21, 2009, at 7:58 AM, ineya ineya wrote:

> Codecs are fine. I spent much time experimenting with codecs and
> completely missed, that freeswitch is modifiyng the SDP record.
>
> When phone A is making a call the SDP contains candidate media  
> attributes:
>
> a=candidate:123abc 1 UDP 9 2000:2::1001 5012 typ host
> a=candidate:123abc 1 UDP 8 10.80.62.92 5010 typ host
>
> But when freeswitch makes the INVITE on phone B, these 2 are missing
> and phone is looking for it, so the INVITE gets rejected by phone with
> "448 Not acceptable here"
>
> So the question is, how can I make the freeswitch to pass these
> candidate media attributes?
>
> On Wed, Oct 21, 2009 at 8:58 AM, Jason White <jason at jasonjgw.net>  
> wrote:
>
>>
>> I suspect the codec negotiation. Make sure that both ends are  
>> offering a
>> common codec.
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From bottleman at icf.org.ru  Wed Oct 21 10:28:51 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Wed, 21 Oct 2009 21:28:51 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>
	<65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
	<Pine.LNX.4.64.0910201841030.12352@srv.icf.org.ru>
	<65d96fc80910210034nd7e8864hfcc421ec6a6c52f@mail.gmail.com>
	<Pine.LNX.4.64.0910211427520.9175@srv.icf.org.ru>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>

On 2009-10-21 17:48 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>FS = 10.4.62.7
TC>SIP phone = 10.4.62.89
TC>H323 endpoint = 10.1.14.153
TC>
TC>
TC>
TC>> TC>2. hangup from sip side doesn't release the h323 leg (now the difference
TC>> is
TC>> TC>that FS is not complaining about thread mismatch ant it looks clean but
TC>> FS
TC>> TC>doesn't send any releasecomplete message... strange)
TC>> TC>3. coredumps when i place outgoing calls
TC>>
TC>> btw,
TC>>
TC>> TC> 70.552157  10.1.14.153 -> 10.4.62.7    H.245 endSessionCommand
TC>> TC> 70.552401  10.1.14.153 -> 10.4.62.7    H.225.0 CS: releaseComplete
TC>> TC> 70.553981    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 3
TC>> TC> 70.554488    10.4.62.7 -> 10.1.14.153  H.245 endSessionCommand
TC>> TC> 70.555056    10.4.62.7 -> 10.1.14.153  H.225.0 CS: releaseComplete
TC>>
TC>> it send, now i have no way to test h323->sip transit, i will have it
TC>> tomorow.
TC>> sip->h323 for me work fine now, give backtrace from code dump of 1.0.4
TC>> where it die?
TC>>
TC>>
TC>
TC>this endSession is when i hangup from H232 side as well :P ... if i don't
TC>hangup on H323 side the H323 leg is not released. Pls chec the time the
TC>packets were sent ...
TC>
TC>
TC>
TC>Here i hangup on the SIP Phone:
TC>
TC>68.374916   10.4.62.89 -> 10.4.62.7    SIP Request: BYE
TC>sip:mod_sofia at 10.4.62.7:5060
TC>68.375342    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 3
TC>68.375620    10.4.62.7 -> 10.4.62.89   SIP Status: 200 OK
TC>
TC>2 sec delay
TC>
TC>Here i hangup on the H323 endpoint (releaseComplete comes from H323 endpoint
TC>first here )
TC>70.552157  10.1.14.153 -> 10.4.62.7    H.245 endSessionCommand
TC>70.552401  10.1.14.153 -> 10.4.62.7    H.225.0 CS: releaseComplete
TC>
TC>
TC>FS just acknowlages it here:
TC>70.554488    10.4.62.7 -> 10.1.14.153  H.245 endSessionCommand
TC>70.555056    10.4.62.7 -> 10.1.14.153  H.225.0 CS: releaseComplete
TC>
TC>
TC>
TC>I have enabled crash-protection and when i do SIP => H323 call it doesn't
TC>generate coredumps... it is just this thread that is crashing ... pls check
TC>the log downbelow:

core dump in case enabled crash-protection may be unusable, i have a case then 
my module crash silently, after this crash-protection is killing sip leg and after
this i get core dump where backtrace show me segfault in libc6, i spent one day to
understand this situation, and then i disable crash-protection i see there is actualy 
it crashes. disable crash-protection and show backtrace of crash, i think result will
be different.

TC>
TC>Dialplan: sofia/internal/1001 at singtel Regex (FAIL) [EMERGENCY]
TC>destination_number(05492122) =~ /^0(112|9[23456])$/ break=on-false
TC>Dialplan: sofia/internal/1001 at singtel parsing [default->SPECIAL_SERVICES]
TC>continue=false
TC>Dialplan: sofia/internal/1001 at singtel Regex (FAIL) [SPECIAL_SERVICES]
TC>destination_number(05492122) =~ /^0(9[01789]\d{3,4})$/ break=on-false
TC>Dialplan: sofia/internal/1001 at singtel parsing [default->ENYTHING_ELSE]
TC>continue=false
TC>Dialplan: sofia/internal/1001 at singtel Regex (PASS) [ENYTHING_ELSE]
TC>destination_number(05492122) =~
TC>/^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$/ break=on-false
TC>Dialplan: sofia/internal/1001 at singtel Action
TC>set(effective_caller_id_number=1001282122)
TC>Dialplan: sofia/internal/1001 at singtel Action set(NCX_IP=10.4.4.254)
TC>Dialplan: sofia/internal/1001 at singtel Action set(call_timeout=30)
TC>Dialplan: sofia/internal/1001 at singtel Action set(hangup_after_bridge=true)
TC>Dialplan: sofia/internal/1001 at singtel Action set(bypass_media=false)
TC>Dialplan: sofia/internal/1001 at singtel Action set(proxy_media=true)
TC>Dialplan: sofia/internal/1001 at singtel Action bridge(h323/05492122@${NCX_IP})
TC>2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:114
TC>(sofia/internal/1001 at singtel) State Change CS_ROUTING -> CS_EXECUTE
TC>2009-10-21 17:35:28.682475 [DEBUG] switch_core_session.c:932 Send signal
TC>sofia/internal/1001 at singtel [BREAK]
TC>2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:484
TC>(sofia/internal/1001 at singtel) State ROUTING going to sleep
TC>2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:398
TC>(sofia/internal/1001 at singtel) Running State Change CS_EXECUTE
TC>2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:491
TC>(sofia/internal/1001 at singtel) State EXECUTE
TC>2009-10-21 17:35:28.682475 [DEBUG] mod_sofia.c:173
TC>sofia/internal/1001 at singtel SOFIA EXECUTE
TC>2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:151
TC>sofia/internal/1001 at singtel Standard EXECUTE
TC>EXECUTE sofia/internal/1001 at singtel set(open=true)
TC>2009-10-21 17:35:28.682475 [DEBUG] mod_dptools.c:748
TC>sofia/internal/1001 at singtel SET [open]=[true]
TC>EXECUTE sofia/internal/1001 at singtelhash(insert/10.4.62.7-spymap/1001/5c3ebda2-be57-11de-a6dd-e7de0b74bdc9)
TC>EXECUTE sofia/internal/1001 at singtelhash(insert/10.4.62.7-last_dial/1001/05492122)
TC>EXECUTE sofia/internal/1001 at singtelhash(insert/10.4.62.7-last_dial/global/5c3ebda2-be57-11de-a6dd-e7de0b74bdc9)
TC>EXECUTE sofia/internal/1001 at singtelset(effective_caller_id_number=1001282122)
TC>2009-10-21 17:35:28.685219 [DEBUG] mod_dptools.c:748
TC>sofia/internal/1001 at singtel SET [effective_caller_id_number]=[1001282122]
TC>EXECUTE sofia/internal/1001 at singtel set(NCX_IP=10.4.4.254)
TC>2009-10-21 17:35:28.685219 [DEBUG] mod_dptools.c:748
TC>sofia/internal/1001 at singtel SET [NCX_IP]=[10.4.4.254]
TC>EXECUTE sofia/internal/1001 at singtel set(call_timeout=30)
TC>2009-10-21 17:35:28.686292 [DEBUG] mod_dptools.c:748
TC>sofia/internal/1001 at singtel SET [call_timeout]=[30]
TC>EXECUTE sofia/internal/1001 at singtel set(hangup_after_bridge=true)
TC>2009-10-21 17:35:28.686292 [DEBUG] mod_dptools.c:748
TC>sofia/internal/1001 at singtel SET [hangup_after_bridge]=[true]
TC>EXECUTE sofia/internal/1001 at singtel set(bypass_media=false)
TC>2009-10-21 17:35:28.686292 [DEBUG] mod_dptools.c:748
TC>sofia/internal/1001 at singtel SET [bypass_media]=[false]
TC>EXECUTE sofia/internal/1001 at singtel set(proxy_media=true)
TC>2009-10-21 17:35:28.686292 [DEBUG] mod_dptools.c:748
TC>sofia/internal/1001 at singtel SET [proxy_media]=[true]
TC>EXECUTE sofia/internal/1001 at singtel bridge(h323/05492122 at 10.4.4.254)
TC>2009-10-21 17:35:28.687357 [DEBUG] mod_h323.cpp:1380
TC>======>create_outgoing_channel DST NUMBER = 05492122 at 10.4.4.254
TC>2009-10-21 17:35:28.687357 [DEBUG] mod_h323.cpp:1336
TC>======>FSH323EndPoint::FSMakeCall DST NUMBER =
TC>05492122 at 10.4.4.254[FSH323EndPoint]
TC>2009-10-21 17:35:28.687357 [DEBUG] mod_h323.cpp:1348 ----> Using 10.4.62.7
TC>for outbound call
TC>2009-10-21 17:35:28.687357 [DEBUG] pdns.cxx:498 SRV Lookup 10.4.4.254
TC>service _h323cs._tcp.
TC>2009-10-21 17:35:28.689538 [DEBUG] pdns.cxx:471 SRV Lookup Fail no domain
TC>h323:05492122 at 10.4.4.254 <h323%3A05492122 at 10.4.4.254>
TC>2009-10-21 17:35:28.689538 [DEBUG] h323ep.cxx:2315 DNS SRV Cannot resolve
TC>remote party 05492122 at 10.4.4.254
TC>2009-10-21 17:35:28.689538 [DEBUG] h323ep.cxx:2103 Making call to:
TC>05492122 at 10.4.4.254
TC>2009-10-21 17:35:28.689538 [DEBUG] mod_h323.cpp:483
TC>======>FSH323EndPoint::CreateConnection callReference = 12651 userDate =
TC>0xb5e0dac0 [FSH323EndPoint]
TC>2009-10-21 17:35:28.689538 [DEBUG] mod_h323.cpp:486 ------>
TC>SWITCH_CALL_DIRECTION_OUTBOUND
TC>2009-10-21 17:35:28.689538 [DEBUG] mod_h323.cpp:496 ------> fsSession =
TC>0xb7569548
TC>2009-10-21 17:35:28.689538 [DEBUG] h323caps.cxx:3186 Added capability:
TC>G.711-ALaw-64k <1>
TC>2009-10-21 17:35:28.689538 [DEBUG] h323caps.cxx:3186 Added capability:
TC>G.711-uLaw-64k <2>
TC>2009-10-21 17:35:28.689538 [DEBUG] h323caps.cxx:3186 Added capability:
TC>GSM-06.10{sw} <3>
TC>2009-10-21 17:35:28.689538 [DEBUG] h323caps.cxx:3186 Added capability:
TC>G.729A/B{sw} <4>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
TC>G.729A{sw} <5>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
TC>G.729B{sw} <6>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
TC>G.729{sw} <7>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
TC>UserInput/hookflash <8>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
TC>UserInput/basicString <9>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
TC>UserInput/dtmf <10>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
TC>UserInput/RFC2833 <11>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
TC>UserInput/Navigation <12>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
TC>UserInput/Softkey <13>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
TC>UserInput/PointDevice <14>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
TC>UserInput/Modal <15>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 1
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
TC>G.711-ALaw-64k <1>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 2
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
TC>G.711-uLaw-64k <2>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 3
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
TC>GSM-06.10{sw} <3>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 4
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
TC>G.729A/B{sw} <4>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 5
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
TC>G.729A{sw} <5>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 6
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
TC>G.729B{sw} <6>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 7
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
TC>G.729{sw} <7>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 8
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
TC>UserInput/hookflash <8>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 9
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
TC>UserInput/basicString <9>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 10
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
TC>UserInput/dtmf <10>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 11
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
TC>UserInput/RFC2833 <11>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 12
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
TC>UserInput/Navigation <12>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 13
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
TC>UserInput/Softkey <13>
TC>2009-10-21 17:35:28.691688 [DEBUG] h323caps.cxx:3248 FindCapability: 14
TC>2009-10-21 17:35:28.691688 [DEBUG] h323caps.cxx:3252 Found capability:
TC>UserInput/PointDevice <14>
TC>2009-10-21 17:35:28.691688 [DEBUG] h323caps.cxx:3248 FindCapability: 15
TC>2009-10-21 17:35:28.691688 [DEBUG] h323caps.cxx:3252 Found capability:
TC>UserInput/Modal <15>
TC>2009-10-21 17:35:28.691688 [DEBUG] h4601.cxx:1725 Endpoint Attached
TC>2009-10-21 17:35:28.691688 [DEBUG] rfc2833.cxx:87 Handler created
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:525
TC>======>FSH323Connection::FSH323Connection FSH323Connection
TC>2009-10-21 17:35:28.691688 [NOTICE] switch_channel.c:602 New Channel h323/
TC>05492122 at 10.4.4.254 [5c413af0-be57-11de-a6dd-e7de0b74bdc9]
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:546 (h323/
TC>05492122 at 10.4.4.254) State Change CS_NEW -> CS_INIT
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:958
TC>======>FSH323Connection::state_change FSH323Connection
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:959 State changed on
TC>connection FSH323Connection
TC>2009-10-21 17:35:28.691688 [DEBUG] switch_core_session.c:932 Send signal
TC>h323/05492122 at 10.4.4.254 [BREAK]
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:831
TC>======>FSH323Connection::kill_channel FSH323Connection
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:832 Kill 3 on connection
TC>FSH323Connection
TC>2009-10-21 17:35:28.691688 [DEBUG] h323ep.cxx:2183 Created new connection:
TC>ip$localhost/12651
TC>2009-10-21 17:35:28.691688 [DEBUG] osutil.cxx:188 File handle high water
TC>mark set: 58 Thread unblock pipe
TC>2009-10-21 17:35:28.691688 [DEBUG] h323ep.cxx:1101 Started call thread
TC>2009-10-21 17:35:28.691688 [CRIT] switch_core_state_machine.c:239 Obtained
TC>10 stack frames.
TC>/usr/local/freeswitch/lib/libfreeswitch.so.1 [0xb7f511f5]
TC>[0xb80be400]
TC>/usr/local/freeswitch/mod/mod_h323.so [0xb6953cc9]
TC>/usr/local/freeswitch/lib/libfreeswitch.so.1(switch_core_session_outgoing_channel+0x1ad)
TC>[0xb7f4f59d]
TC>/usr/local/freeswitch/lib/libfreeswitch.so.1(switch_ivr_originate+0x1b2b)
TC>[0xb7f7aa1b]
TC>/usr/local/freeswitch/mod/mod_dptools.so [0xb70ea2d5]
TC>/usr/local/freeswitch/lib/libfreeswitch.so.1(switch_core_session_exec+0x284)
TC>[0xb7f4c5a4]
TC>/usr/local/freeswitch/lib/libfreeswitch.so.1(switch_core_session_execute_application+0x8b)
TC>[0xb7f4cacb]
TC>/usr/local/freeswitch/lib/libfreeswitch.so.1(switch_core_session_run+0x12b8)
TC>[0xb7f52658]
TC>/usr/local/freeswitch/lib/libfreeswitch.so.1 [0xb7f4e6c5]
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:509
TC>======>FSH323EndPoint::OnSetGatewayPrefixes FSH323EndPoint
TC>2009-10-21 17:35:28.691688 [CRIT] switch_core_state_machine.c:352 Thread has
TC>crashed for channel sofia/internal/1001 at singtel
TC>2009-10-21 17:35:28.691688 [NOTICE] switch_core_state_machine.c:353 Hangup
TC>sofia/internal/1001 at singtel [CS_EXECUTE] [CRASH]
TC>2009-10-21 17:35:28.691688 [DEBUG] transports.cxx:1828 Connecting to
TC>10.4.4.254:1720 (local port=0)
TC>2009-10-21 17:35:28.691688 [DEBUG] osutil.cxx:188 File handle high water
TC>mark set: 59 PTCPSocket
TC>2009-10-21 17:35:28.691688 [DEBUG] switch_channel.c:1683 Send signal
TC>sofia/internal/1001 at singtel [KILL]
TC>2009-10-21 17:35:28.691688 [DEBUG] switch_core_session.c:932 Send signal
TC>sofia/internal/1001 at singtel [BREAK]
TC>2009-10-21 17:35:28.691688 [DEBUG] switch_core_state_machine.c:398
TC>(sofia/internal/1001 at singtel) Running State Change CS_HANGUP
TC>2009-10-21 17:35:28.691688 [DEBUG] transports.cxx:1681 Started connection:
TC>host=10.4.4.254:1720, if=10.4.62.7:52984, handle=59
TC>2009-10-21 17:35:28.691688 [DEBUG] h323.cxx:2680 Sending Setup PDU
TC>2009-10-21 17:35:28.691688 [DEBUG] switch_core_state_machine.c:434
TC>(sofia/internal/1001 at singtel) State HANGUP
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_sofia.c:338 Channel
TC>sofia/internal/1001 at singtel hanging up, cause: CRASH
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_sofia.c:417 Responding to INVITE
TC>with: 480
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:559
TC>======>FSH323Connection::OnSetLocalCapabilities() [FSH323Connection]
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:565
TC>======>FSH323Connection::SetLocalCapabilities() Size local capability = 15
TC>[FSH323Connection]
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
TC>======>FSH323Connection::decodeCapability
TC>2009-10-21 17:35:28.691688 [INFO] mod_h323.cpp:619 capability
TC>'G.711-ALaw-64k' format 'PCMU' payload 8
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
TC>======>FSH323Connection::decodeCapability
TC>2009-10-21 17:35:28.691688 [INFO] mod_h323.cpp:619 capability
TC>'G.711-uLaw-64k' format 'PCMA' payload 0
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
TC>======>FSH323Connection::decodeCapability
TC>2009-10-21 17:35:28.691688 [INFO] mod_h323.cpp:619 capability 'GSM-06.10'
TC>format 'GSM' payload 3
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
TC>======>FSH323Connection::decodeCapability
TC>2009-10-21 17:35:28.691688 [INFO] mod_h323.cpp:619 capability 'G.729A/B'
TC>format 'G729' payload 18
TC>2009-10-21 17:35:28.691688 [DEBUG] switch_core_state_machine.c:46
TC>sofia/internal/1001 at singtel Standard HANGUP, cause: CRASH
TC>2009-10-21 17:35:28.691688 [DEBUG] switch_core_state_machine.c:434
TC>(sofia/internal/1001 at singtel) State HANGUP going to sleep
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
TC>======>FSH323Connection::decodeCapability
TC>2009-10-21 17:35:28.691688 [INFO] mod_h323.cpp:619 capability 'G.729A'
TC>format 'G729' payload 18
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
TC>======>FSH323Connection::decodeCapability
TC>2009-10-21 17:35:28.691688 [INFO] mod_h323.cpp:619 capability 'G.729B'
TC>format 'G729b' payload 18
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
TC>======>FSH323Connection::decodeCapability
TC>
TC>
TC>
TC>Well, I'm not sure if the backtrace is from 1.0.4 ... i will disable
TC>crass-protection and will send new logs to you.
TC>
TC>
TC>Also, if you like i can give you access to the machine itself...
TC>
TC>T.
TC>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From brian at freeswitch.org  Wed Oct 21 10:30:59 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Oct 2009 12:30:59 -0500
Subject: [Freeswitch-users] Calls dropping on SIP timer expiry due to
	UPDATE's being ignored.
In-Reply-To: <1256143519.7836.107.camel@keithl-lt>
References: <1256143519.7836.107.camel@keithl-lt>
Message-ID: <4FEF9AF8-14F2-4248-ACD5-E711CDE7D017@freeswitch.org>

This will be fixed soon. Watch SVN.

/b

On Oct 21, 2009, at 11:45 AM, Keith Laaks wrote:

> Hi,


From brian at freeswitch.org  Wed Oct 21 10:31:25 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Oct 2009 12:31:25 -0500
Subject: [Freeswitch-users] Is background sound|music during call
	possible?
In-Reply-To: <700164f40910210411k6ce2795cud47202a25ce27f87@mail.gmail.com>
References: <700164f40910210411k6ce2795cud47202a25ce27f87@mail.gmail.com>
Message-ID: <3893EF02-247C-4C38-8F56-15E2C6A7D19D@freeswitch.org>

uuid_displace with mux option.

/b

On Oct 21, 2009, at 6:11 AM, Joey Carter wrote:

> Hello
>
> How can I add background music that will play during call?
>
> Thanks



From helmut.kuper at ewetel.de  Wed Oct 21 10:43:23 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Wed, 21 Oct 2009 19:43:23 +0200
Subject: [Freeswitch-users] Qustion about INFO messages
	after	Connect/Answer
In-Reply-To: <87f2f3b90910191435s7528e256r85df5f12fc634f24@mail.gmail.com>
References: <4AD859A4.5000101@ewetel.de>	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>	<4ADC7C09.6070503@ewetel.de>	<191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>	<4ADC85C0.7060908@ewetel.de>
	<4ADC8B6E.7070404@ewetel.de>	<191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>
	<87f2f3b90910191435s7528e256r85df5f12fc634f24@mail.gmail.com>
Message-ID: <4ADF483B.4050807@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello Mike,

just updated my prod system. The "1/a/" problem is solved with Anthony's
"originate_callee_id_name" chvar.

thanks alot :)

So, last thing of this thread is still the "unknown" thing on callee's
display, which is (by now) NOT affected by the new chvars.

regards
Helmut


On 19.10.2009 23:35, Michael Collins wrote:
> 
> 
> On Mon, Oct 19, 2009 at 1:07 PM, Anthony Minessale
> <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>> wrote:
> 
>     please update and test trunk
> 
>     1) I changed the core to remove the excess data by default in your
>     scenario
>     2) I added variables you can use to control it
>     origination_callee_id_name origination_callee_id_number which belong
>     in {} in the dial string eg
>     {origination_callee_id_number=1234}openzap/1/a/1234
> 
> 
> After you test, please confirm the behavior and then we'll update the
> wiki on these two new chan vars.
> -MC
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From msc at freeswitch.org  Wed Oct 21 10:44:58 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 21 Oct 2009 10:44:58 -0700
Subject: [Freeswitch-users] NOT in dialplan expression
In-Reply-To: <b39273530910210807j1f8a2fc6l4ff2bf1c154497b9@mail.gmail.com>
References: <33c87fa30910210434t268e8026pfd2b5439f26069fd@mail.gmail.com>
	<b39273530910210807j1f8a2fc6l4ff2bf1c154497b9@mail.gmail.com>
Message-ID: <87f2f3b90910211044t1a02a653ya3bb9e1911099daa@mail.gmail.com>

On Wed, Oct 21, 2009 at 8:07 AM, Rupa Schomaker <rupa at rupa.com> wrote:

> You can also look at using anti-action rather than action after the
> condition.
>
> condition == if
> action == then
> anti-action == else
>

Rupa & Leon,

Nice job of explaining the options. Sometimes we forget about the powerful
constructs that are available in FreeSWITCH and PCRE. karma++ for both of
you. :)
-MC
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From msc at freeswitch.org  Wed Oct 21 10:48:06 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 21 Oct 2009 10:48:06 -0700
Subject: [Freeswitch-users] Qustion about INFO messages after
	Connect/Answer
In-Reply-To: <4ADF483B.4050807@ewetel.de>
References: <4AD859A4.5000101@ewetel.de>
	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>
	<4ADC7C09.6070503@ewetel.de>
	<191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>
	<4ADC85C0.7060908@ewetel.de> <4ADC8B6E.7070404@ewetel.de>
	<191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>
	<87f2f3b90910191435s7528e256r85df5f12fc634f24@mail.gmail.com>
	<4ADF483B.4050807@ewetel.de>
Message-ID: <87f2f3b90910211048s208daf1eyfae501091ab75530@mail.gmail.com>

On Wed, Oct 21, 2009 at 10:43 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello Mike,
>
> just updated my prod system. The "1/a/" problem is solved with Anthony's
> "originate_callee_id_name" chvar.
>
> thanks alot :)
>
> So, last thing of this thread is still the "unknown" thing on callee's
> display, which is (by now) NOT affected by the new chvars.
>
Okay, you are able to reproduce that "unknown" thing? Can you pastebin a
fresh debug log w/ SIP trace on, plus and relevant dp changes from the
default dialplan?
Thanks,
MC
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From jerry.richards at teotech.com  Wed Oct 21 11:37:46 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 21 Oct 2009 11:37:46 -0700
Subject: [Freeswitch-users] 3rd Party Dial Plan Tool
Message-ID: <840EFA6312584A7498E5AB88459851DC@greyhawk.tonecommander.com>


Can anyone recommend a good 3rd party dialplan tool that will work with
Freeswitch?

Best Regards,
Jerry



From anthony.minessale at gmail.com  Wed Oct 21 11:54:17 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Oct 2009 13:54:17 -0500
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
In-Reply-To: <2d9149cd0910202050i36a7ea82nc7253d87e38f990a@mail.gmail.com>
References: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>
	<191c3a030910201434q4c6e65e1h58c5f0a82addc2d3@mail.gmail.com>
	<2d9149cd0910202050i36a7ea82nc7253d87e38f990a@mail.gmail.com>
Message-ID: <191c3a030910211154gc8412b5gd999fbedffa02b53@mail.gmail.com>

can you try trunk and let me know right away,
if it's still not working i may need ssh access and call you on the phone.


On Tue, Oct 20, 2009 at 10:50 PM, Kristian Kielhofner <
kristian.kielhofner at gmail.com> wrote:

> Done.
>
> On Tue, Oct 20, 2009 at 5:34 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > issue:
> > console loglevel debug
> > sofia profile internal siptrace on
> >
> > and put it on pastebin
> > http://pastebin.freeswitch.org
> >
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From brian at freeswitch.org  Wed Oct 21 12:00:12 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Oct 2009 14:00:12 -0500
Subject: [Freeswitch-users] 3rd Party Dial Plan Tool
In-Reply-To: <840EFA6312584A7498E5AB88459851DC@greyhawk.tonecommander.com>
References: <840EFA6312584A7498E5AB88459851DC@greyhawk.tonecommander.com>
Message-ID: <5067C824-5FB3-412A-9653-9A872DA9204A@freeswitch.org>

none specifically exist... good ole trusty editor?

/b
PS: 


On Oct 21, 2009, at 1:37 PM, Jerry Richards wrote:

>
> Can anyone recommend a good 3rd party dialplan tool that will work  
> with
> Freeswitch?
>
> Best Regards,
> Jerry



From sprice at gmail.com  Wed Oct 21 12:10:25 2009
From: sprice at gmail.com (SP)
Date: Wed, 21 Oct 2009 14:10:25 -0500
Subject: [Freeswitch-users] 3rd Party Dial Plan Tool
In-Reply-To: <5067C824-5FB3-412A-9653-9A872DA9204A@freeswitch.org>
References: <840EFA6312584A7498E5AB88459851DC@greyhawk.tonecommander.com>
	<5067C824-5FB3-412A-9653-9A872DA9204A@freeswitch.org>
Message-ID: <7e2ac3270910211210r2ea53cafr66058b38c393fe92@mail.gmail.com>

You could also try consulting at freeswitch.org

On Wed, Oct 21, 2009 at 14:00, Brian West <brian at freeswitch.org> wrote:
> none specifically exist... good ole trusty editor?
>
> /b
> PS: 
>
>
> On Oct 21, 2009, at 1:37 PM, Jerry Richards wrote:
>
>>
>> Can anyone recommend a good 3rd party dialplan tool that will work
>> with
>> Freeswitch?
>>
>> Best Regards,
>> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Shannon


From mike at jerris.com  Wed Oct 21 12:50:51 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 21 Oct 2009 15:50:51 -0400
Subject: [Freeswitch-users] sched_api doesn't get launched
In-Reply-To: <59ad9ca10910210244q5410f2bdq3b93c25f95673d11@mail.gmail.com>
References: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
	<3F1747F9-C543-4C6F-BB01-4DD5FA5B7BC5@jerris.com>
	<59ad9ca10910161153h5d531533vd844e49b9ca6d896@mail.gmail.com>
	<87f2f3b90910161237h356b7aa2q6ec12c6ecf78dc34@mail.gmail.com>
	<59ad9ca10910210244q5410f2bdq3b93c25f95673d11@mail.gmail.com>
Message-ID: <033669F7-EE8B-455C-A8DE-FD4EE564F34C@jerris.com>

The syntax is different, but the api is the same as lua:

> So you need an API object in order to use it. I don't know the  
> syntax for creating an api obj in Java but in Lua it goes like this:
> api = freeswitch.API();
> res = api:execute("sched_api","+300 none my_api my_api_args")

create the API object and use the execute method of it.

Mike


On Oct 21, 2009, at 5:44 AM, Henry Huang wrote:

> I can't seem to find the right thing to use in mod_java to execute  
> api commands, only api_after_bridge
>
>
> 2009-10-21 17:42:46.593094 [NOTICE] mod_sofia.c:1509 Pre-Answer  
> sofia/internal/1688003 at 192.168.1.66!
> #
> # A fatal error has been detected by the Java Runtime Environment:
> #
> #  SIGSEGV (0xb) at pc=0x004e4480, pid=1927, tid=16116624
> #
> # JRE version: 6.0_16-b01
> # Java VM: Java HotSpot(TM) Client VM (14.2-b01 mixed mode linux-x86 )
> # Problematic frame:
> # C  [libc.so.6+0x6f480]  strcpy+0x10
> #
> # An error report file with more information is saved as:
> # /usr/local/freeswitch/bin/hs_err_pid1927.log
> 2009-10-21 17:42:59.883729 [ERR] switch_core_session.c:1374 Invalid  
> Application sched_api
> 2009-10-21 17:42:59.883729 [NOTICE] switch_core_session.c:1375  
> Hangup sofia/internal/1688003 at 192.168.1.66 [CS_EXECUTE]  
> [DESTINATION_OUT_OF_ORDER]
> #
> # If you would like to submit a bug report, please visit:
> #   http://java.sun.com/webapps/bugreport/crash.jsp
> # The crash happened outside the Java Virtual Machine in native code.
> # See problematic frame for where to report the bug.
>
>
> On Sat, Oct 17, 2009 at 3:37 AM, Michael Collins  
> <msc at freeswitch.org> wrote:
>
>
> On Fri, Oct 16, 2009 at 11:53 AM, Henry Huang <red.rain.seven at gmail.com 
> > wrote:
> So how would you trigger it from a script dialplan? The only time it  
> seemed to work is when I did setVariable("api_after_bridge",  
> "sched_api blah blah blah");
> but then it gets executed after the channel's been teared down. I  
> thought api_after_bridge means right after the call gets connected.
>
> I need something to execute an api command right before or right  
> after the call gets bridged.
>
> api_after_bridge is a channel variable, so using setVariable works  
> just fine. If you need to sched_api is an API only. Check these out:
> http://wiki.freeswitch.org/wiki/Mod_commands#Misc._Commands
>
> So you need an API object in order to use it. I don't know the  
> syntax for creating an api obj in Java but in Lua it goes like this:
> api = freeswitch.API();
> res = api:execute("sched_api","+300 none my_api my_api_args")
>
> Remember, if the method you are using isn't found in the dial plan  
> tools then it isn't a dial plan application. Make sure it's on the  
> list:
> http://wiki.freeswitch.org/wiki/Mod_dptools
>
> On the other hand, API commands are listed here:
>
> http://wiki.freeswitch.org/wiki/Mod_commands
>
> dptools require a session object, api commands require an api  
> object...
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Henry Huang
> UniC Solution - Communication Unified
> VoIP & Open Source software Consultant
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From ineyaa at gmail.com  Wed Oct 21 13:08:20 2009
From: ineyaa at gmail.com (ineya ineya)
Date: Wed, 21 Oct 2009 22:08:20 +0200
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <67234FAF-73DE-462E-B7E8-4BDE922D7CB8@jerris.com>
References: <FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
	<858EC056-1D84-4F30-A0B7-9A70FF98695D@freeswitch.org>
	<7b8592a0910200821m18fe9b4dg73dda05a10699af2@mail.gmail.com>
	<642B6CED-CDF3-4D40-B493-0F2D6222C30B@freeswitch.org>
	<7b8592a0910200947k6ca5a391l35786f0e855ae388@mail.gmail.com>
	<20091021040343.GA9440@jdc.jasonjgw.net>
	<7b8592a0910202336x1bf2a8a2nd636219ec7031c5a@mail.gmail.com>
	<20091021065841.GA13150@jdc.jasonjgw.net>
	<7b8592a0910210458s28ddffbdi77cf116d33c28405@mail.gmail.com>
	<67234FAF-73DE-462E-B7E8-4BDE922D7CB8@jerris.com>
Message-ID: <7b8592a0910211308w597c29a3m3afc9953266ca41b@mail.gmail.com>

Yes, I think you are right, the sip stack comes from company called
M5T as far as I know, and there are references to ICE on their site.

On Wed, Oct 21, 2009 at 7:29 PM, Michael Jerris <mike at jerris.com> wrote:
> This appears to be some sort of ice implementation? ?We don't support
> sip ice at this time.
>


From anthony.minessale at gmail.com  Wed Oct 21 13:09:24 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Oct 2009 15:09:24 -0500
Subject: [Freeswitch-users] sched_api doesn't get launched
In-Reply-To: <59ad9ca10910210244q5410f2bdq3b93c25f95673d11@mail.gmail.com>
References: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
	<3F1747F9-C543-4C6F-BB01-4DD5FA5B7BC5@jerris.com>
	<59ad9ca10910161153h5d531533vd844e49b9ca6d896@mail.gmail.com>
	<87f2f3b90910161237h356b7aa2q6ec12c6ecf78dc34@mail.gmail.com>
	<59ad9ca10910210244q5410f2bdq3b93c25f95673d11@mail.gmail.com>
Message-ID: <191c3a030910211309sb727b37g6324e1833f2a1e8c@mail.gmail.com>

Yes you need an API object as described in other email.

Which line of code from java caused that segfault
It looks like a simple NULL string issue that we may want to hunt down.

On Wed, Oct 21, 2009 at 4:44 AM, Henry Huang <red.rain.seven at gmail.com>wrote:

> I can't seem to find the right thing to use in mod_java to execute api
> commands, only api_after_bridge
>
> 2009-10-21 17:42:46.593094 [NOTICE] mod_sofia.c:1509 Pre-Answer
> sofia/internal/1688003 at 192.168.1.66!
> #
> # A fatal error has been detected by the Java Runtime Environment:
> #
> #  SIGSEGV (0xb) at pc=0x004e4480, pid=1927, tid=16116624
> #
> # JRE version: 6.0_16-b01
> # Java VM: Java HotSpot(TM) Client VM (14.2-b01 mixed mode linux-x86 )
> # Problematic frame:
> # C  [libc.so.6+0x6f480]  strcpy+0x10
> #
> # An error report file with more information is saved as:
> # /usr/local/freeswitch/bin/hs_err_pid1927.log
> 2009-10-21 17:42:59.883729 [ERR] switch_core_session.c:1374 Invalid
> Application sched_api
> 2009-10-21 17:42:59.883729 [NOTICE] switch_core_session.c:1375 Hangup
> sofia/internal/1688003 at 192.168.1.66 [CS_EXECUTE]
> [DESTINATION_OUT_OF_ORDER]
> #
> # If you would like to submit a bug report, please visit:
> #   http://java.sun.com/webapps/bugreport/crash.jsp
> # The crash happened outside the Java Virtual Machine in native code.
> # See problematic frame for where to report the bug.
>
>
> On Sat, Oct 17, 2009 at 3:37 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Fri, Oct 16, 2009 at 11:53 AM, Henry Huang <red.rain.seven at gmail.com>wrote:
>>
>>> So how would you trigger it from a script dialplan? The only time it
>>> seemed to work is when I did setVariable("api_after_bridge", "sched_api blah
>>> blah blah");
>>> but then it gets executed after the channel's been teared down. I thought
>>> api_after_bridge means right after the call gets connected.
>>>
>>> I need something to execute an api command right before or right after
>>> the call gets bridged.
>>>
>>> api_after_bridge is a channel variable, so using setVariable works just
>> fine. If you need to sched_api is an API only. Check these out:
>> http://wiki.freeswitch.org/wiki/Mod_commands#Misc._Commands
>>
>> So you need an API object in order to use it. I don't know the syntax for
>> creating an api obj in Java but in Lua it goes like this:
>> api = freeswitch.API();
>> res = api:execute("sched_api","+300 none my_api my_api_args")
>>
>> Remember, if the method you are using isn't found in the dial plan tools
>> then it isn't a dial plan application. Make sure it's on the list:
>> http://wiki.freeswitch.org/wiki/Mod_dptools
>>
>> On the other hand, API commands are listed here:
>> http://wiki.freeswitch.org/wiki/Mod_commands
>>
>> dptools require a session object, api commands require an api object...
>>
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Henry Huang
> UniC Solution - Communication Unified
> VoIP & Open Source software Consultant
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mike at jerris.com  Wed Oct 21 15:05:56 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 21 Oct 2009 18:05:56 -0400
Subject: [Freeswitch-users] Calls dropping on SIP timer expiry due to
	UPDATE's being ignored.
In-Reply-To: <1256143519.7836.107.camel@keithl-lt>
References: <1256143519.7836.107.camel@keithl-lt>
Message-ID: <94C69E51-6577-43A7-AB5B-70BF1AFE12DD@jerris.com>

This should now be fixed in latest svn trunk.

Mike

On Oct 21, 2009, at 12:45 PM, Keith Laaks wrote:

> Hi,
>
> Hope someone knows how I am able to get around this one. Here goes...
>
> Did an upgrade to trunk (from a July vintage build) last week and  
> noticed calls out to a provider were now failing after about 30  
> seconds or so - post answer. Tried latest (15183) - same thing.
>
> Analysing, I see that I have multiple UPDATE messages now being sent  
> to the provider, but no response being sent back to FS. So FS times  
> out and eventually kills the call.
> Interestingly, it only drops the A-leg; the B-leg remains up till  
> the B party hangs up.
>
> I cant recall seeing these UPDATE messages before...
>
> The intent of the UPDATE seems to be to send the callee name &  
> number to the B-leg.
>
> If its the provider's sip stack that's broken w.r.t. handling UPDATE  
> - is there any way to get around it by doing something in my config  
> to ensure these UPDATE's are not 'triggered' ?



From mcampbellsmith at gmail.com  Wed Oct 21 15:51:47 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 22 Oct 2009 09:51:47 +1100
Subject: [Freeswitch-users] NOT in dialplan expression
In-Reply-To: <87f2f3b90910211044t1a02a653ya3bb9e1911099daa@mail.gmail.com>
References: <33c87fa30910210434t268e8026pfd2b5439f26069fd@mail.gmail.com>
	<b39273530910210807j1f8a2fc6l4ff2bf1c154497b9@mail.gmail.com>
	<87f2f3b90910211044t1a02a653ya3bb9e1911099daa@mail.gmail.com>
Message-ID: <33c87fa30910211551m40a21daai3405b7c32da6655d@mail.gmail.com>

Thanks Guys for your suggestions!  Very much appreciated.

The reason I asked was that I want to check if a variable is set or
not.  I don't think your suggestions will work because I actually have
to check to see an undefined variable.  Any way to do this?

I want to have the same extension for checking digits read from DTMF.
The problem is that the digits are not set to the variable until the
transfer statement, which means I would require 2 extensions.   It
will make things simpler if I can have the same extension that reads
the digits and then checks them.

For example:

 <extension name="checkdigits">
      <condition field="destination_number" expression="^checkdigits$"/>
       <!--The next condition should be true if the variable $digits
is not set/undefined -->
        <condition field="${digits}" expression="^[^.+]$" />
-----<<<< what should be here to check for undefined variable $digits
?
           <action application="read" data="1 10 ivr/ivr-hello.wav
digits 10000 #"/>
           <action application="phrase" data="spell,${digits}"/>
           <action application="transfer" data="checkdigits digits XML
features"/>
      </condition>

      <condition field="${pincode}" expression="^${some_variable}$">
             < SOME ACTIONS HERE >
      </condition>
 </extension>

On Thu, Oct 22, 2009 at 4:44 AM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Wed, Oct 21, 2009 at 8:07 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>
>> You can also look at using anti-action rather than action after the
>> condition.
>>
>> condition == if
>> action == then
>> anti-action == else
>
> Rupa & Leon,
>
> Nice job of explaining the options. Sometimes we forget about the powerful
> constructs that are available in FreeSWITCH and PCRE. karma++ for both of
> you. :)
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Wed Oct 21 17:02:29 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 21 Oct 2009 17:02:29 -0700
Subject: [Freeswitch-users] NOT in dialplan expression
In-Reply-To: <33c87fa30910211551m40a21daai3405b7c32da6655d@mail.gmail.com>
References: <33c87fa30910210434t268e8026pfd2b5439f26069fd@mail.gmail.com>
	<b39273530910210807j1f8a2fc6l4ff2bf1c154497b9@mail.gmail.com>
	<87f2f3b90910211044t1a02a653ya3bb9e1911099daa@mail.gmail.com>
	<33c87fa30910211551m40a21daai3405b7c32da6655d@mail.gmail.com>
Message-ID: <87f2f3b90910211702x159836f3p6b3d222341856bc6@mail.gmail.com>

On Wed, Oct 21, 2009 at 3:51 PM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> Thanks Guys for your suggestions!  Very much appreciated.
>
> The reason I asked was that I want to check if a variable is set or
> not.  I don't think your suggestions will work because I actually have
> to check to see an undefined variable.  Any way to do this?
>

Not in the dialplan itself. Better off using a script for this.

>
> I want to have the same extension for checking digits read from DTMF.
> The problem is that the digits are not set to the variable until the
> transfer statement, which means I would require 2 extensions.   It
> will make things simpler if I can have the same extension that reads
> the digits and then checks them.
>

You need either to use multiple extensions or use a script. The dialplan was
specifically designed not to be a programming language and the functionality
you seek is best served by using a script or by using transfer to drop the
call through the diaplan again and use another extension.
-MC


>
> For example:
>
>  <extension name="checkdigits">
>      <condition field="destination_number" expression="^checkdigits$"/>
>       <!--The next condition should be true if the variable $digits
> is not set/undefined -->
>        <condition field="${digits}" expression="^[^.+]$" />
> -----<<<< what should be here to check for undefined variable $digits
> ?
>           <action application="read" data="1 10 ivr/ivr-hello.wav
> digits 10000 #"/>
>           <action application="phrase" data="spell,${digits}"/>
>           <action application="transfer" data="checkdigits digits XML
> features"/>
>      </condition>
>
>      <condition field="${pincode}" expression="^${some_variable}$">
>             < SOME ACTIONS HERE >
>      </condition>
>  </extension>
>
>
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From anthony.minessale at gmail.com  Wed Oct 21 17:07:56 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Oct 2009 19:07:56 -0500
Subject: [Freeswitch-users] NOT in dialplan expression
In-Reply-To: <33c87fa30910211551m40a21daai3405b7c32da6655d@mail.gmail.com>
References: <33c87fa30910210434t268e8026pfd2b5439f26069fd@mail.gmail.com>
	<b39273530910210807j1f8a2fc6l4ff2bf1c154497b9@mail.gmail.com>
	<87f2f3b90910211044t1a02a653ya3bb9e1911099daa@mail.gmail.com>
	<33c87fa30910211551m40a21daai3405b7c32da6655d@mail.gmail.com>
Message-ID: <191c3a030910211707m6f6bf2c6q61fa5932f905cc85@mail.gmail.com>

Like rupa said,
use anti-action?

Remember FreeSWITCH XML dialplan is not a programming language it's a
pre-processor routing markup

Se potential use of anti-action below:

 <extension name="checkdigits">
     <condition field="destination_number" expression="^checkdigits$"/>
      <!--The next condition should be true if the variable $digits is not
set/undefined -->
       <condition field="${digits}" expression="^[^.+]$" />
--        <action application="read" data="1 10 ivr/ivr-hello.wav digits
10000 #"/>
          <action application="phrase" data="spell,${digits}"/>
          <action application="transfer" data="checkdigits digits XML
features"/>
          <anti-action application="set" data="not_true=true"/>
     </condition>

     <condition field="${pincode}" expression="^${some_variable}$">
            < SOME ACTIONS HERE >
     </condition>
 </extension>



On Wed, Oct 21, 2009 at 5:51 PM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> Thanks Guys for your suggestions!  Very much appreciated.
>
> The reason I asked was that I want to check if a variable is set or
> not.  I don't think your suggestions will work because I actually have
> to check to see an undefined variable.  Any way to do this?
>
> I want to have the same extension for checking digits read from DTMF.
> The problem is that the digits are not set to the variable until the
> transfer statement, which means I would require 2 extensions.   It
> will make things simpler if I can have the same extension that reads
> the digits and then checks them.
>
> For example:
>
>  <extension name="checkdigits">
>      <condition field="destination_number" expression="^checkdigits$"/>
>       <!--The next condition should be true if the variable $digits
> is not set/undefined -->
>        <condition field="${digits}" expression="^[^.+]$" />
> -----<<<< what should be here to check for undefined variable $digits
> ?
>           <action application="read" data="1 10 ivr/ivr-hello.wav
> digits 10000 #"/>
>           <action application="phrase" data="spell,${digits}"/>
>           <action application="transfer" data="checkdigits digits XML
> features"/>
>      </condition>
>
>      <condition field="${pincode}" expression="^${some_variable}$">
>             < SOME ACTIONS HERE >
>      </condition>
>  </extension>
>
> On Thu, Oct 22, 2009 at 4:44 AM, Michael Collins <msc at freeswitch.org>
> wrote:
> >
> >
> > On Wed, Oct 21, 2009 at 8:07 AM, Rupa Schomaker <rupa at rupa.com> wrote:
> >>
> >> You can also look at using anti-action rather than action after the
> >> condition.
> >>
> >> condition == if
> >> action == then
> >> anti-action == else
> >
> > Rupa & Leon,
> >
> > Nice job of explaining the options. Sometimes we forget about the
> powerful
> > constructs that are available in FreeSWITCH and PCRE. karma++ for both of
> > you. :)
> > -MC
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From anthony.minessale at gmail.com  Wed Oct 21 17:20:41 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Oct 2009 19:20:41 -0500
Subject: [Freeswitch-users] UUID of the newly originated call?
In-Reply-To: <fe67a3340910202306i55a0b9ceve92ab19e000edf11@mail.gmail.com>
References: <fe67a3340910202306i55a0b9ceve92ab19e000edf11@mail.gmail.com>
Message-ID: <191c3a030910211720l26bd0303i2cb11e9adb28037d@mail.gmail.com>

which revision of FS are you using?

On Wed, Oct 21, 2009 at 1:06 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> I've tried with origination_uuid.
>
> First, I tried with SIP and my program executes successfully as what I
> expected. This program initiates a new call when a call comes and let the
> new call to eavesdrop the landed call.
> When, I experimented with PRI(openzap), I'm facing the following error.
> And, I am unable to make any calls(even from CLI). It is reporting the same
> error for the subsequent calls.
> Error:        2009-10-21 11:08:31.661252 [ERR] mod_openzap.c:1760 Session
> for channel 1:1 not found [UUID: 8364eb30-be03-11de-aa07-07e76b7651a9]
>
> Full log is in http://pastebin.freeswitch.org/10780
> My script is here, http://pastebin.freeswitch.org/10781
>
> What is this error for and how to avoid this?
>
> Is there any other way to get the uuid of the originated call except
> explicitly defining(origination_uuid).?!
>
> --
> Regards,
> Nagalenoj H.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Wed Oct 21 17:31:24 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Oct 2009 19:31:24 -0500
Subject: [Freeswitch-users] CS_REPORTING Channel event state
In-Reply-To: <08C886FF-6D2E-46CD-AC7D-2AF37F18B170@jerris.com>
References: <8ccbff060910200847y5e7ccab6o84d431836970bcb4@mail.gmail.com>
	<08C886FF-6D2E-46CD-AC7D-2AF37F18B170@jerris.com>
Message-ID: <191c3a030910211731y3e4de866qfc8fd0c9edd1db65@mail.gmail.com>

better still,

1) update to trunk with "make current"

2) Try to reproduce, if you can:
 a) open a new shell
 b) make sure you have gdb and gcore installed
 c) run ./support-d/fscore_pb gcore
 d) report the URL




On Tue, Oct 20, 2009 at 8:48 PM, Michael Jerris <mike at jerris.com> wrote:

> REPORTING is the state that it writes to CDR.  If you have calls stuck
> in this state, take one and try to use uuid_kill on it and see if it
> goes away, then get a core off of it and pastebin the thread apply all
> bt (with no other calls up).  What modules are you using for cdr and
> with what configuration?
>
> Mike
>
> On Oct 20, 2009, at 11:47 AM, Dome Charoenyost wrote:
>
> > Dear All
> > What's CS_REPORTING state ?
> > I found many channels not hang up  ans state is CS_REPORTING
> > e264f84a-bd87-11de-9a90-2320c02172de,outbound,2009-10-20
> > 21:50:26,1256050226,sofia/external/xxxxxxxxxxxx at xxx.xxx.xx.
> > 191:7050,CS_REPORTING,FreeSWITCH,xxxxxxxxxxxxxxxxxxxx,xx.xxx.xxx.xxx,
> > 7050,,,XML,public,,,,,
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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IRC: irc.freenode.net #freeswitch

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sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
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From ahmedmunir007 at gmail.com  Wed Oct 21 21:08:44 2009
From: ahmedmunir007 at gmail.com (Ahmed Munir)
Date: Thu, 22 Oct 2009 09:08:44 +0500
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179
In-Reply-To: <mailman.1516.1256123625.2495.freeswitch-users@lists.freeswitch.org>
References: <mailman.1516.1256123625.2495.freeswitch-users@lists.freeswitch.org>
Message-ID: <a60fed5e0910212108v4755203y693b8bdec5501f5f@mail.gmail.com>

Hi,

Thanks for reply, it really helped me. One more thing to ask, how can we
make decision against >,<, >=, <= in condition header? Like we use == for
action and != for anti-action.

Kindly highlight it.




>
> ---------- Forwarded message ----------
> From: Ahmed Munir <ahmedmunir007 at gmail.com>
> To: FreeSwitch <freeswitch-users at lists.freeswitch.org>
> Date: Wed, 21 Oct 2009 15:37:15 +0500
> Subject: [Freeswitch-users] Call custom variable in condition
> Hi,
>
> I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I
> want to use it in condition, as I'm listing down the configuration below;
>
> <context name="SIP_incoming">
>    <extension name="call-sip-extensions">
>       <condition field="destination_number" expression="^(\d+)$">
>           <action application="set" data="AUTHENTICATION_STATUS=0"/>
>            <action application="transfer" data="${AUTHENTICATION_STATUS}
> XML Authen_Status"/>
>       </condition>
>    </extension>
> </context>
>
> <context name="Authen_Status">
>      <extension name="exten-auth-status">
>        <condition field="AUTHENTICATION_STATUS" expression="^0$">
>           <action application="answer"/>
>           <action application="playback" data="play.wav"/>
>       </condition>
>     </extension>
>   </context>
>
>
>
>
>  But unfortunately it is not working. Kindly advise me how to do implement
> it(Note: I don't want to call script). And one more thing to ask how can I
> transfer the values within the same context?
>
> --
> Regards,
>
> Ahmed Munir
>
>
>
>
> ---------- Forwarded message ----------
> From: Ghulam Mustafa <mustafa.pk at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Wed, 21 Oct 2009 15:52:24 +0500
> Subject: Re: [Freeswitch-users] Call custom variable in condition
> Ahmed,
>
> you can't use variables set by "set" application within a condition, though
> it doesn't make sense. wondering if there is any logic behind this or it's
> just a simple missing feature. anyone?
>
> -m
>
> Ahmed Munir wrote:
>
>> Hi,
>>
>> I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I
>> want to use it in condition, as I'm listing down the configuration below;
>>
>> <context name="SIP_incoming">
>>   <extension name="call-sip-extensions">
>>      <condition field="destination_number" expression="^(\d+)$">
>>          <action application="set" data="AUTHENTICATION_STATUS=0"/>
>>           <action application="transfer" data="${AUTHENTICATION_STATUS}
>> XML Authen_Status"/>
>>      </condition>
>>   </extension>
>> </context>
>>
>> <context name="Authen_Status">
>>     <extension name="exten-auth-status">
>>       <condition field="AUTHENTICATION_STATUS" expression="^0$">
>>          <action application="answer"/>
>>          <action application="playback" data="play.wav"/>
>>      </condition>
>>    </extension>
>>  </context>
>>
>>
>>
>>
>>  But unfortunately it is not working. Kindly advise me how to do implement
>> it(Note: I don't want to call script). And one more thing to ask how can I
>> transfer the values within the same context?
>>
>> --
>> Regards,
>>
>> Ahmed Munir
>>
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
>
>
> ---------- Forwarded message ----------
> From: Tihomir Culjaga <tculjaga at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Wed, 21 Oct 2009 13:13:13 +0200
> Subject: Re: [Freeswitch-users] Call custom variable in condition
> consider this:
>
>
>
> <context name="SIP_incoming">
>    <extension name="call-sip-extensions">
>       <condition field="destination_number" expression="^(\d+)$">
>           <action application="set" data="AUTHENTICATION_STATUS=0"/>
>            <action application="transfer" data="${AUTHENTICATION_STATUS}
> XML Authen_Status"/>
>       </condition>
>    </extension>
> </context>
>
>
>
> <context name="Authen_Status">
>      <extension name="exten-auth-status">
>        <condition field="${AUTHENTICATION_STATUS}" expression="^0$">
>           <action application="answer"/>
>           <action application="playback" data="play.wav"/>
>       </condition>
>     </extension>
>   </context>
>
>
>
>
>
> here is one of my dialplan. I'm using execute_extension but it is quite the
> same...
>
>
>
>    <extension name="ServiceLookup">
>       <condition field="destination_number" expression="(^300030)(.*)">
>          <action application="lookup_service_destination" data="in
> ${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, in $
> 1, in ${network_addr}:5060, out red_contact, out authResult"/>
>          <action application="log" data="INFO ########################
> ServiceLookup ########################\n"/>
>          <action application="log" data="INFO ########################
> contact = '${red_contact}' ##############\n"/>
>          <action application="log" data="INFO ########################
> CallerNum = '${caller_id_number:6:16}' ##########\n"/>
>          <action application="log" data="INFO ########################
> RADIUS auth = '${authResult}' ##########\n"/>
>
>          <action application="execute_extension" data="doRedirect XML
> public"/>
>         </condition>
>    </extension>
>
>
>    <extension name="doRedirect">
>       <condition field="destination_number" expression="^doRedirect$"/>
>       <condition field="${authResult}" expression="^0$|^60$">
>          <action application="log" data="INFO ########################
> RADIUS auth OK!!!' ##########\n"/>
>          <action application="redirect" data="${red_contact}"/>
>          <anti-action application="log" data="INFO ########################
> RADIUS auth NOK!! ##########\n"/>
>          <anti-action application="respond" data="403 Forbidden"/>
>       </condition>
>
>    </extension>
>
>
>
>
> On Wed, Oct 21, 2009 at 12:37 PM, Ahmed Munir <ahmedmunir007 at gmail.com>wrote:
>
>> Hi,
>>
>> I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I
>> want to use it in condition, as I'm listing down the configuration below;
>>
>> <context name="SIP_incoming">
>>    <extension name="call-sip-extensions">
>>       <condition field="destination_number" expression="^(\d+)$">
>>           <action application="set" data="AUTHENTICATION_STATUS=0"/>
>>            <action application="transfer" data="${AUTHENTICATION_STATUS}
>> XML Authen_Status"/>
>>       </condition>
>>    </extension>
>> </context>
>>
>> <context name="Authen_Status">
>>      <extension name="exten-auth-status">
>>        <condition field="AUTHENTICATION_STATUS" expression="^0$">
>>           <action application="answer"/>
>>           <action application="playback" data="play.wav"/>
>>       </condition>
>>     </extension>
>>   </context>
>>
>>
>>
>>
>>  But unfortunately it is not working. Kindly advise me how to do implement
>> it(Note: I don't want to call script). And one more thing to ask how can I
>> transfer the values within the same context?
>>
>> --
>> Regards,
>>
>> Ahmed Munir
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,

Ahmed Munir
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From mcampbellsmith at gmail.com  Wed Oct 21 21:28:41 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 22 Oct 2009 15:28:41 +1100
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179
In-Reply-To: <a60fed5e0910212108v4755203y693b8bdec5501f5f@mail.gmail.com>
References: <mailman.1516.1256123625.2495.freeswitch-users@lists.freeswitch.org>
	<a60fed5e0910212108v4755203y693b8bdec5501f5f@mail.gmail.com>
Message-ID: <33c87fa30910212128g1c167e6s8b847184921b21d1@mail.gmail.com>

Can't you use the inline statement to set a variable so that it can be
used directly in a condition?

http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions


On Thu, Oct 22, 2009 at 3:08 PM, Ahmed Munir <ahmedmunir007 at gmail.com> wrote:
> Hi,
>
> Thanks for reply, it really helped me. One more thing to ask, how can we
> make decision against >,<, >=, <= in condition header? Like we use == for
> action and != for anti-action.
>
> Kindly highlight it.
>
>
>
>>
>> ---------- Forwarded message ----------
>> From:?Ahmed Munir <ahmedmunir007 at gmail.com>
>> To:?FreeSwitch <freeswitch-users at lists.freeswitch.org>
>> Date:?Wed, 21 Oct 2009 15:37:15 +0500
>> Subject:?[Freeswitch-users] Call custom variable in condition
>> Hi,
>>
>> I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I
>> want to use it in condition, as I'm listing down the configuration below;
>>
>> <context name="SIP_incoming">
>> ?? <extension name="call-sip-extensions">
>> ????? <condition field="destination_number" expression="^(\d+)$">
>> ????????? <action application="set" data="AUTHENTICATION_STATUS=0"/>
>> ?????????? <action application="transfer" data="${AUTHENTICATION_STATUS}
>> XML Authen_Status"/>
>> ????? </condition>
>> ?? </extension>
>> </context>
>>
>> <context name="Authen_Status">
>> ???? <extension name="exten-auth-status">
>> ?????? <condition field="AUTHENTICATION_STATUS" expression="^0$">
>> ????????? <action application="answer"/>
>> ????????? <action application="playback" data="play.wav"/>
>> ????? </condition>
>> ??? </extension>
>> ? </context>
>>
>>
>>
>>
>> ?But unfortunately it is not working. Kindly advise me how to do implement
>> it(Note: I don't want to call script). And one more thing to ask how can I
>> transfer the values within the same context?
>>
>> --
>> Regards,
>>
>> Ahmed Munir
>>
>>
>>
>>
>> ---------- Forwarded message ----------
>> From:?Ghulam Mustafa <mustafa.pk at gmail.com>
>> To:?freeswitch-users at lists.freeswitch.org
>> Date:?Wed, 21 Oct 2009 15:52:24 +0500
>> Subject:?Re: [Freeswitch-users] Call custom variable in condition
>> Ahmed,
>>
>> you can't use variables set by "set" application within a condition,
>> though it doesn't make sense. wondering if there is any logic behind this or
>> it's just a simple missing feature. anyone?
>>
>> -m
>>
>> Ahmed Munir wrote:
>>>
>>> Hi,
>>>
>>> I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I
>>> want to use it in condition, as I'm listing down the configuration below;
>>>
>>> <context name="SIP_incoming">
>>> ? <extension name="call-sip-extensions">
>>> ? ? ?<condition field="destination_number" expression="^(\d+)$">
>>> ? ? ? ? ?<action application="set" data="AUTHENTICATION_STATUS=0"/>
>>> ? ? ? ? ? <action application="transfer" data="${AUTHENTICATION_STATUS}
>>> XML Authen_Status"/>
>>> ? ? ?</condition>
>>> ? </extension>
>>> </context>
>>>
>>> <context name="Authen_Status">
>>> ? ? <extension name="exten-auth-status">
>>> ? ? ? <condition field="AUTHENTICATION_STATUS" expression="^0$">
>>> ? ? ? ? ?<action application="answer"/>
>>> ? ? ? ? ?<action application="playback" data="play.wav"/>
>>> ? ? ?</condition>
>>> ? ?</extension>
>>> ?</context>
>>>
>>>
>>>
>>>
>>> ?But unfortunately it is not working. Kindly advise me how to do
>>> implement it(Note: I don't want to call script). And one more thing to ask
>>> how can I transfer the values within the same context?
>>>
>>> --
>>> Regards,
>>>
>>> Ahmed Munir
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>>
>>
>> ---------- Forwarded message ----------
>> From:?Tihomir Culjaga <tculjaga at gmail.com>
>> To:?freeswitch-users at lists.freeswitch.org
>> Date:?Wed, 21 Oct 2009 13:13:13 +0200
>> Subject:?Re: [Freeswitch-users] Call custom variable in condition
>> consider this:
>>
>>
>>
>> <context name="SIP_incoming">
>> ?? <extension name="call-sip-extensions">
>> ????? <condition field="destination_number" expression="^(\d+)$">
>> ????????? <action application="set" data="AUTHENTICATION_STATUS=0"/>
>> ?????????? <action application="transfer" data="${AUTHENTICATION_STATUS}
>> XML Authen_Status"/>
>> ????? </condition>
>> ?? </extension>
>> </context>
>>
>>
>>
>> <context name="Authen_Status">
>> ???? <extension name="exten-auth-status">
>> ?????? <condition field="${AUTHENTICATION_STATUS}" expression="^0$">
>> ????????? <action application="answer"/>
>> ????????? <action application="playback" data="play.wav"/>
>> ????? </condition>
>> ??? </extension>
>> ? </context>
>>
>>
>>
>>
>>
>> here is one of my dialplan. I'm using execute_extension but it is quite
>> the same...
>>
>>
>>
>> ?? <extension name="ServiceLookup">
>> ????? <condition field="destination_number" expression="(^300030)(.*)">
>> ???????? <action application="lookup_service_destination" data="in
>> ${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, in $
>> 1, in ${network_addr}:5060, out red_contact, out authResult"/>
>> ???????? <action application="log" data="INFO ########################
>> ServiceLookup ########################\n"/>
>> ???????? <action application="log" data="INFO ########################
>> contact = '${red_contact}' ##############\n"/>
>> ???????? <action application="log" data="INFO ########################
>> CallerNum = '${caller_id_number:6:16}' ##########\n"/>
>> ???????? <action application="log" data="INFO ########################
>> RADIUS auth = '${authResult}' ##########\n"/>
>>
>> ???????? <action application="execute_extension" data="doRedirect XML
>> public"/>
>> ??????? </condition>
>> ?? </extension>
>>
>>
>> ?? <extension name="doRedirect">
>> ????? <condition field="destination_number" expression="^doRedirect$"/>
>> ????? <condition field="${authResult}" expression="^0$|^60$">
>> ???????? <action application="log" data="INFO ########################
>> RADIUS auth OK!!!' ##########\n"/>
>> ???????? <action application="redirect" data="${red_contact}"/>
>> ???????? <anti-action application="log" data="INFO
>> ######################## RADIUS auth NOK!! ##########\n"/>
>> ???????? <anti-action application="respond" data="403 Forbidden"/>
>> ????? </condition>
>>
>> ?? </extension>
>>
>>
>>
>>
>> On Wed, Oct 21, 2009 at 12:37 PM, Ahmed Munir <ahmedmunir007 at gmail.com>
>> wrote:
>>>
>>> Hi,
>>>
>>> I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I
>>> want to use it in condition, as I'm listing down the configuration below;
>>>
>>> <context name="SIP_incoming">
>>> ?? <extension name="call-sip-extensions">
>>> ????? <condition field="destination_number" expression="^(\d+)$">
>>> ????????? <action application="set" data="AUTHENTICATION_STATUS=0"/>
>>> ?????????? <action application="transfer" data="${AUTHENTICATION_STATUS}
>>> XML Authen_Status"/>
>>> ????? </condition>
>>> ?? </extension>
>>> </context>
>>>
>>> <context name="Authen_Status">
>>> ???? <extension name="exten-auth-status">
>>> ?????? <condition field="AUTHENTICATION_STATUS" expression="^0$">
>>> ????????? <action application="answer"/>
>>> ????????? <action application="playback" data="play.wav"/>
>>> ????? </condition>
>>> ??? </extension>
>>> ? </context>
>>>
>>>
>>>
>>>
>>> ?But unfortunately it is not working. Kindly advise me how to do
>>> implement it(Note: I don't want to call script). And one more thing to ask
>>> how can I transfer the values within the same context?
>>>
>>> --
>>> Regards,
>>>
>>> Ahmed Munir
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Regards,
>
> Ahmed Munir
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Wed Oct 21 21:36:31 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 21 Oct 2009 21:36:31 -0700
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179
In-Reply-To: <33c87fa30910212128g1c167e6s8b847184921b21d1@mail.gmail.com>
References: <mailman.1516.1256123625.2495.freeswitch-users@lists.freeswitch.org>
	<a60fed5e0910212108v4755203y693b8bdec5501f5f@mail.gmail.com>
	<33c87fa30910212128g1c167e6s8b847184921b21d1@mail.gmail.com>
Message-ID: <87f2f3b90910212136wce5c22dy9fc23a78e8946541@mail.gmail.com>

On Wed, Oct 21, 2009 at 9:28 PM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> Can't you use the inline statement to set a variable so that it can be
> used directly in a condition?
>
> http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
>
>
Yes you can. Just don't abuse it like Tony said. If you find yourself doing
it a lot then it's a sure sign that you're taking the wrong approach.
-MC
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From msc at freeswitch.org  Wed Oct 21 21:41:31 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 21 Oct 2009 21:41:31 -0700
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179
In-Reply-To: <a60fed5e0910212108v4755203y693b8bdec5501f5f@mail.gmail.com>
References: <mailman.1516.1256123625.2495.freeswitch-users@lists.freeswitch.org>
	<a60fed5e0910212108v4755203y693b8bdec5501f5f@mail.gmail.com>
Message-ID: <87f2f3b90910212141w72b8f5a2oce032adb974d71e2@mail.gmail.com>

On Wed, Oct 21, 2009 at 9:08 PM, Ahmed Munir <ahmedmunir007 at gmail.com>wrote:

> Hi,
>
> Thanks for reply, it really helped me. One more thing to ask, how can we
> make decision against >,<, >=, <= in condition header? Like we use == for
> action and != for anti-action.
>
> Kindly highlight it.
>
>
You can only do greater than and less than in the date/time matching. See
the date/time example in the default.xml dialplan file.

You can also use regular expressions if you're in a pinch. For example, if
you need to match numbers >= 1100 and <= 1500 you could just use this regex:

^(1[1234]\d\d|1500)$

The real question, though, is this: what types of values do you need to
match for GT or LT? Date/time? Money? Other? That will determine if you need
to use a script or just the dialplan.
-MC
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From shaheryarkh at googlemail.com  Wed Oct 21 23:57:42 2009
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Thu, 22 Oct 2009 12:57:42 +0600
Subject: [Freeswitch-users] Sofia SIP external profile gateway names
Message-ID: <b16156850910212357s15fe6116tc22b6709f7110a49@mail.gmail.com>

Hi,

I am facing a problem with Sofia SIP external profile. Basically i have 10+
accounts, say a101 - a110, to a single service provider, say xyz.com. For
each account a have created a gateway in external profile's directory i.e.
/usr/local/freeswitch/conf/sip_profiles/external/a101.xml up to a110.xml.

At first the problem was none of the accounts were registering since, FS was
trying to send registration requests to host "a101" instead of xyz.com, but
later when i set xyz.com as proxy and register proxy address, it
successfully registered all accounts.

Now i am facing a similar problem in dialplan, for example if i try to
dialout via gateway a101, call immediately fails with
NORMAL_TEMPORARY_FAILURE. When i trun on sip tracing i don't see any INVITE
message sent to provider xyz.com.

2009-10-22 06:51:55.325393 [NOTICE] switch_channel.c:613 New Channel
sofia/external/00923344224088 at a101 [b7663caf-cd29-4213-9059-ae880d49b0ca]
2009-10-22 06:51:55.516382 [NOTICE] sofia.c:4039 Hangup
sofia/external/00923344224088 at a101 [CS_CONSUME_MEDIA]
[NORMAL_TEMPORARY_FAILURE]
API CALL [originate(sofia/external/00923344224088 at a101 9999)] output:
-ERR NORMAL_TEMPORARY_FAILURE
2009-10-22 06:51:55.527381 [NOTICE] switch_core_session.c:1132 Session 2
(sofia/external/00923344224088 at a101) Ended
2009-10-22 06:51:55.527381 [NOTICE] switch_core_session.c:1134 Close Channel
sofia/external/00923344224088 at a101 [CS_DESTROY]

I think its trying to look up a101 instead of xyz.com to send INVITE. Kindly
help.

Thank you.


-- 
________________________________________________________
|
                     |
| FATAL ERROR                                                           ---
O X |
|_______________________________________________________|
|                        You have moved the mouse.
 |
| Windows must be restarted for the changes to take effect.   |
|                                        <OK>
               |
####################################/


Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From leon at scarlet-internet.nl  Thu Oct 22 01:27:54 2009
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Thu, 22 Oct 2009 10:27:54 +0200
Subject: [Freeswitch-users] Call custom variable in condition
In-Reply-To: <87f2f3b90910211002y7c35c622g4c76472fd8c7bada@mail.gmail.com>
References: <a60fed5e0910210337n45ffefe2wc3e3d9d0b403a568@mail.gmail.com>
	<65d96fc80910210413t1123dbc0xd5740bc0673d638c@mail.gmail.com>
	<191c3a030910210939k7dbf1494gf713df8770356450@mail.gmail.com>
	<87f2f3b90910211002y7c35c622g4c76472fd8c7bada@mail.gmail.com>
Message-ID: <CF72DAFC-61F5-4E8A-B1D9-C0171E823D3E@scarlet-internet.nl>

Hi Michael,

The feature is already documented here:

http://wiki.freeswitch.org/wiki/Dialplan_XML#Clarification
http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions

Perhaps the reason *why* it's the way it is can be expanded a bit ?

regards,

Leon


On Oct 21, 2009, at 7:02 PM, Michael Collins wrote:

>
>
> On Wed, Oct 21, 2009 at 9:39 AM, Anthony Minessale <anthony.minessale at gmail.com 
> > wrote:
> It not only makes sense it's well documented on the wiki page.
> The set line is not happening right when it's encountered, the set  
> line is copied into the channel and executed later after the whole  
> dialplan is parsed.  The dialplan is a pre-processor not a runtime  
> engine.
>
> Here is a new feature in pre-1.0.5 (svn trunk)
>
> Some applications like set can now be executed within the dialplan  
> but you should use it sparingly.
> <action application="set" data="testing=true" inline="true"/>
>
> I'm getting ready to document this feature. For the sake of  
> edification, why is it best to use this sparingly, other than wide- 
> spread use making dialplans all cluttered?
> -MC
>
>
> The inline=true makes it execute inside the dialplan and it's never  
> copied into your resulting extension because it's executed  
> immediately.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From nagalenoj at gmail.com  Thu Oct 22 02:08:55 2009
From: nagalenoj at gmail.com (Nagalenoj)
Date: Thu, 22 Oct 2009 02:08:55 -0700 (PDT)
Subject: [Freeswitch-users] UUID of the newly originated call?
In-Reply-To: <191c3a030910211720l26bd0303i2cb11e9adb28037d@mail.gmail.com>
References: <fe67a3340910202306i55a0b9ceve92ab19e000edf11@mail.gmail.com>
	<191c3a030910211720l26bd0303i2cb11e9adb28037d@mail.gmail.com>
Message-ID: <26006565.post@talk.nabble.com>


I'm using 'FreeSWITCH Version 1.0.trunk (15106)'.


Anthony Minessale-2 wrote:
> 
> which revision of FS are you using?
> 
> On Wed, Oct 21, 2009 at 1:06 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:
> 
>> I've tried with origination_uuid.
>>
>> First, I tried with SIP and my program executes successfully as what I
>> expected. This program initiates a new call when a call comes and let the
>> new call to eavesdrop the landed call.
>> When, I experimented with PRI(openzap), I'm facing the following error.
>> And, I am unable to make any calls(even from CLI). It is reporting the
>> same
>> error for the subsequent calls.
>> Error:        2009-10-21 11:08:31.661252 [ERR] mod_openzap.c:1760 Session
>> for channel 1:1 not found [UUID: 8364eb30-be03-11de-aa07-07e76b7651a9]
>>
>> Full log is in http://pastebin.freeswitch.org/10780
>> My script is here, http://pastebin.freeswitch.org/10781
>>
>> What is this error for and how to avoid this?
>>
>> Is there any other way to get the uuid of the originated call except
>> explicitly defining(origination_uuid).?!
>>
>> --
>> Regards,
>> Nagalenoj H.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/Re%3A-UUID-of-the-newly-originated-call--tp25987024p26006565.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From ahmedmunir007 at gmail.com  Thu Oct 22 02:47:52 2009
From: ahmedmunir007 at gmail.com (Ahmed Munir)
Date: Thu, 22 Oct 2009 14:47:52 +0500
Subject: [Freeswitch-users] Expression in dialplan
Message-ID: <a60fed5e0910220247q54a6a4dfr4b589e76a78d1d52@mail.gmail.com>

Thanks for reply. Actually I want to match values and strings. Like
IP_CALL=="true" or IP_CALL != "true" or Date >= "22/10/09" and vice versa.
Last thing I forgot to mention, how can I implement condition in dial plan
xml when using operation and/or  i.e. IP_CALL == "false" and PST_CALL ==
"false" / IP_CALL == "false" or PST_CALL == "false".

Note: I only want to implement it in dial plan not in any script.


Kindly advise the solution.

---------- Forwarded message ----------
From: Michael Collins <msc at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Date: Wed, 21 Oct 2009 21:36:31 -0700
Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179


On Wed, Oct 21, 2009 at 9:28 PM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> Can't you use the inline statement to set a variable so that it can be
> used directly in a condition?
>
> http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
>
>
Yes you can. Just don't abuse it like Tony said. If you find yourself doing
it a lot then it's a sure sign that you're taking the wrong approach.
-MC



---------- Forwarded message ----------
From: Michael Collins <msc at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Date: Wed, 21 Oct 2009 21:41:31 -0700
Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179


On Wed, Oct 21, 2009 at 9:08 PM, Ahmed Munir <ahmedmunir007 at gmail.com>wrote:

> Hi,
>
> Thanks for reply, it really helped me. One more thing to ask, how can we
> make decision against >,<, >=, <= in condition header? Like we use == for
> action and != for anti-action.
>
> Kindly highlight it.
>
>
You can only do greater than and less than in the date/time matching. See
the date/time example in the default.xml dialplan file.

You can also use regular expressions if you're in a pinch. For example, if
you need to match numbers >= 1100 and <= 1500 you could just use this regex:

^(1[1234]\d\d|1500)$

The real question, though, is this: what types of values do you need to
match for GT or LT? Date/time? Money? Other? That will determine if you need
to use a script or just the dialplan.
-MC
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From rupa at rupa.com  Thu Oct 22 05:51:04 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 22 Oct 2009 07:51:04 -0500
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179
In-Reply-To: <87f2f3b90910212141w72b8f5a2oce032adb974d71e2@mail.gmail.com>
References: <mailman.1516.1256123625.2495.freeswitch-users@lists.freeswitch.org>
	<a60fed5e0910212108v4755203y693b8bdec5501f5f@mail.gmail.com>
	<87f2f3b90910212141w72b8f5a2oce032adb974d71e2@mail.gmail.com>
Message-ID: <b39273530910220551q14d22150ldf8d4bff0b6fd011@mail.gmail.com>

cond would be helpful here?  I updated the wiki on this one just now
with a bit more detail.  It is a api call. so, you'd use it like:

${cond(eval ? trueval : falseval)}

so to get a value of ERR if the var my myvar is > 15 you could:

${cond(${myvar} > 15 ? ERR : OK)}

If both sides of the comparison operator are numeric then it does
numeric comparison otherwise it does lexical string comparison.


On Wed, Oct 21, 2009 at 11:41 PM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Wed, Oct 21, 2009 at 9:08 PM, Ahmed Munir <ahmedmunir007 at gmail.com>
> wrote:
>>
>> Hi,
>>
>> Thanks for reply, it really helped me. One more thing to ask, how can we
>> make decision against >,<, >=, <= in condition header? Like we use == for
>> action and != for anti-action.
>>
>> Kindly highlight it.
>>
>
> You can only do greater than and less than in the date/time matching. See
> the date/time example in the default.xml dialplan file.
>
> You can also use regular expressions if you're in a pinch. For example, if
> you need to match numbers >= 1100 and <= 1500 you could just use this regex:
>
> ^(1[1234]\d\d|1500)$
>
> The real question, though, is this: what types of values do you need to
> match for GT or LT? Date/time? Money? Other? That will determine if you need
> to use a script or just the dialplan.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa


From tculjaga at gmail.com  Thu Oct 22 06:30:34 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 22 Oct 2009 15:30:34 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910201841030.12352@srv.icf.org.ru>
	<65d96fc80910210034nd7e8864hfcc421ec6a6c52f@mail.gmail.com>
	<Pine.LNX.4.64.0910211427520.9175@srv.icf.org.ru>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
Message-ID: <65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>

>
>
> TC>
> TC>I have enabled crash-protection and when i do SIP => H323 call it
> doesn't
> TC>generate coredumps... it is just this thread that is crashing ... pls
> check
> TC>the log downbelow:
>
> core dump in case enabled crash-protection may be unusable, i have a case
> then
> my module crash silently, after this crash-protection is killing sip leg
> and after
> this i get core dump where backtrace show me segfault in libc6, i spent one
> day to
> understand this situation, and then i disable crash-protection i see there
> is actualy
> it crashes. disable crash-protection and show backtrace of crash, i think
> result will
> be different.
>
>
> TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
> TC>======>FSH323Connection::decodeCapability
> TC>
> TC>
> TC>
> TC>Well, I'm not sure if the backtrace is from 1.0.4 ... i will disable
> TC>crass-protection and will send new logs to you.
> TC>
> TC>
> TC>Also, if you like i can give you access to the machine itself...
> TC>
> TC>T.
> TC>
>



Hi, here is the FS log without crash-protection:
http://pastebin.freeswitch.org/10796 and here is the backtrace:
http://pastebin.freeswitch.org/10797



my dialplan looks ok, so i guess it is up to the module.


  <extension name="ENYTHING_ELSE">
    <condition field="destination_number"
expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
      <action application="set"
data="effective_caller_id_number=1001282122"/>
      <action application="set" data="NCX_IP=10.4.4.254"/>
      <action application="set" data="call_timeout=30"/>
      <action application="set" data="hangup_after_bridge=true"/>
      <!--action application="set" data="bypass_media=false"/-->
      <action application="set" data="proxy_media=true"/>

      <!--action application="bridge" data="opal/h323:0$1@${NCX_IP}"/-->
      <action application="bridge" data="h323/0$1@${NCX_IP}"/>
    </condition>
  </extension>



please advice,

T.



> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129-RIPE
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bottleman at icf.org.ru  Thu Oct 22 06:41:01 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Thu, 22 Oct 2009 17:41:01 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910201841030.12352@srv.icf.org.ru>
	<65d96fc80910210034nd7e8864hfcc421ec6a6c52f@mail.gmail.com>
	<Pine.LNX.4.64.0910211427520.9175@srv.icf.org.ru>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>

On 2009-10-22 15:30 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>>
TC>>
TC>> TC>
TC>> TC>I have enabled crash-protection and when i do SIP => H323 call it
TC>> doesn't
TC>> TC>generate coredumps... it is just this thread that is crashing ... pls
TC>> check
TC>> TC>the log downbelow:
TC>>
TC>> core dump in case enabled crash-protection may be unusable, i have a case
TC>> then
TC>> my module crash silently, after this crash-protection is killing sip leg
TC>> and after
TC>> this i get core dump where backtrace show me segfault in libc6, i spent one
TC>> day to
TC>> understand this situation, and then i disable crash-protection i see there
TC>> is actualy
TC>> it crashes. disable crash-protection and show backtrace of crash, i think
TC>> result will
TC>> be different.
TC>>
TC>>
TC>> TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
TC>> TC>======>FSH323Connection::decodeCapability
TC>> TC>
TC>> TC>
TC>> TC>
TC>> TC>Well, I'm not sure if the backtrace is from 1.0.4 ... i will disable
TC>> TC>crass-protection and will send new logs to you.
TC>> TC>
TC>> TC>
TC>> TC>Also, if you like i can give you access to the machine itself...
TC>> TC>
TC>> TC>T.
TC>> TC>
TC>>
TC>
TC>
TC>
TC>Hi, here is the FS log without crash-protection:
TC>http://pastebin.freeswitch.org/10796 and here is the backtrace:
TC>http://pastebin.freeswitch.org/10797

i fix this crash already, please download latest version from same link 
as previous, recompile and try again.

TC>my dialplan looks ok, so i guess it is up to the module.
TC>
TC>
TC>  <extension name="ENYTHING_ELSE">
TC>    <condition field="destination_number"
TC>expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
TC>      <action application="set"
TC>data="effective_caller_id_number=1001282122"/>
TC>      <action application="set" data="NCX_IP=10.4.4.254"/>
TC>      <action application="set" data="call_timeout=30"/>
TC>      <action application="set" data="hangup_after_bridge=true"/>
TC>      <!--action application="set" data="bypass_media=false"/-->
TC>      <action application="set" data="proxy_media=true"/>
TC>
TC>      <!--action application="bridge" data="opal/h323:0$1@${NCX_IP}"/-->
TC>      <action application="bridge" data="h323/0$1@${NCX_IP}"/>
TC>    </condition>
TC>  </extension>
TC>
TC>
TC>
TC>please advice,
TC>
TC>T.
TC>
TC>
TC>
TC>> C ?????????                       With Best Regards
TC>> ???????????? ????.                Georgiewskiy Yuriy
TC>> +7 4872 711666                    +7 4872 711666
TC>> ???? +7 4872 711143               fax +7 4872 711143
TC>> ???????? ??? "?? ?? ??????"       IT Service Ltd
TC>> http://nkoort.ru                  http://nkoort.ru
TC>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
TC>> YG129-RIPE                        YG129-RIPE
TC>>
TC>> _______________________________________________
TC>> FreeSWITCH-users mailing list
TC>> FreeSWITCH-users at lists.freeswitch.org
TC>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> http://www.freeswitch.org
TC>>
TC>>
TC>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From szasz.szabolcs at gmail.com  Thu Oct 22 06:56:29 2009
From: szasz.szabolcs at gmail.com (Szasz Szabolcs)
Date: Thu, 22 Oct 2009 16:56:29 +0300
Subject: [Freeswitch-users] srtp g729 bypass
Message-ID: <4a4d27d30910220656u46e584b3h94c357e6d218462@mail.gmail.com>

Hi

I have set up a freeswitch with TLS and SRTP support. I'm sending  encrypted
calls to Freswitch and the Freeswitch  forwards the calls to an asterisk
unencrypted. I have issue by using G729 in this scenario. My UA supports
g279, asterisk supports g729 transcoding and I understood that freeswitch
supports the g729 in bypass mode. Should it work? What I have to change in
config? So the scenario, I would like to do:
 UA|======TLS+SRTP(G729)=====|FREESWITCH|------------ClearSIP+RTP(G729)------------|Asterisk

Thank you in advance.

Szasz Szabolcs
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From tculjaga at gmail.com  Thu Oct 22 06:59:52 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 22 Oct 2009 15:59:52 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910211427520.9175@srv.icf.org.ru>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
Message-ID: <65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>

>
>
> TC>Hi, here is the FS log without crash-protection:
> TC>http://pastebin.freeswitch.org/10796 and here is the backtrace:
> TC>http://pastebin.freeswitch.org/10797
>
> i fix this crash already, please download latest version from same link
> as previous, recompile and try again.
>
>
outgoing works, I can place an end-to-end call ... except the RTP is realy
delayed ... after approx 30 sec of conversation the audio is delayed more
than 10 seconds.... but i have 2 way audio for outgoing calls:)

Do you need some logs ?


Inbound cals still the same... i suppose you didn't have a chance working on
that as well ...

T.
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From tculjaga at gmail.com  Thu Oct 22 07:04:54 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 22 Oct 2009 16:04:54 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
Message-ID: <65d96fc80910220704t283bfca3w2129ff3575659f97@mail.gmail.com>

On Thu, Oct 22, 2009 at 3:59 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

>
>> TC>Hi, here is the FS log without crash-protection:
>> TC>http://pastebin.freeswitch.org/10796 and here is the backtrace:
>> TC>http://pastebin.freeswitch.org/10797
>>
>> i fix this crash already, please download latest version from same link
>> as previous, recompile and try again.
>>
>>
> outgoing works, I can place an end-to-end call ... except the RTP is realy
> delayed ... after approx 30 sec of conversation the audio is delayed more
> than 10 seconds.... but i have 2 way audio for outgoing calls:)
>
>
one more thing ... it is H323 endpoint => SIP phone audio that is delayed.
SIP phone => H323 endpoint is ok!

T.


> Do you need some logs ?
>
>
> Inbound cals still the same... i suppose you didn't have a chance working
> on that as well ...
>
> T.
>
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From bottleman at icf.org.ru  Thu Oct 22 07:11:45 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Thu, 22 Oct 2009 18:11:45 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910211427520.9175@srv.icf.org.ru>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>

On 2009-10-22 15:59 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>>
TC>>
TC>> TC>Hi, here is the FS log without crash-protection:
TC>> TC>http://pastebin.freeswitch.org/10796 and here is the backtrace:
TC>> TC>http://pastebin.freeswitch.org/10797
TC>>
TC>> i fix this crash already, please download latest version from same link
TC>> as previous, recompile and try again.
TC>>
TC>>
TC>outgoing works, I can place an end-to-end call ... except the RTP is realy
TC>delayed ... after approx 30 sec of conversation the audio is delayed more
TC>than 10 seconds.... but i have 2 way audio for outgoing calls:)
TC>
TC>Do you need some logs ?

try disable medai-proxy, there is issue with rtp now then medai-proxy or transcoding enabled.

TC>Inbound cals still the same... i suppose you didn't have a chance working on
TC>that as well ...

sorry i don't remember what the same, show extension and logs of inbound call. at this time
at this time i have working transit h323<->sip in both direction, ivr is work too, there is some 
issues present but basicaly it work.

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From bottleman at icf.org.ru  Thu Oct 22 07:13:09 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Thu, 22 Oct 2009 18:13:09 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910220704t283bfca3w2129ff3575659f97@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<65d96fc80910220704t283bfca3w2129ff3575659f97@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910221811590.1929@srv.icf.org.ru>

On 2009-10-22 16:04 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>On Thu, Oct 22, 2009 at 3:59 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
TC>
TC>>
TC>>> TC>Hi, here is the FS log without crash-protection:
TC>>> TC>http://pastebin.freeswitch.org/10796 and here is the backtrace:
TC>>> TC>http://pastebin.freeswitch.org/10797
TC>>>
TC>>> i fix this crash already, please download latest version from same link
TC>>> as previous, recompile and try again.
TC>>>
TC>>>
TC>> outgoing works, I can place an end-to-end call ... except the RTP is realy
TC>> delayed ... after approx 30 sec of conversation the audio is delayed more
TC>> than 10 seconds.... but i have 2 way audio for outgoing calls:)
TC>>
TC>>
TC>one more thing ... it is H323 endpoint => SIP phone audio that is delayed.
TC>SIP phone => H323 endpoint is ok!

hm, i have such issue but in reverce direction now.

TC>> Do you need some logs ?
TC>>
TC>>
TC>> Inbound cals still the same... i suppose you didn't have a chance working
TC>> on that as well ...
TC>>
TC>> T.
TC>>
TC>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From anthony.minessale at gmail.com  Thu Oct 22 07:27:54 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 22 Oct 2009 09:27:54 -0500
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910221811590.1929@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<65d96fc80910220704t283bfca3w2129ff3575659f97@mail.gmail.com>
	<Pine.LNX.4.64.0910221811590.1929@srv.icf.org.ru>
Message-ID: <191c3a030910220727v7e62f47dhf172ba8682d5d52a@mail.gmail.com>

crash protection has been completely removed from FreeSWITCH, I certianly
hope you are working on this against SVN trunk?  Also you have been given an
svn area and a jira category for this so you should move all the info from
this thread to jira http://jira.freeswitch.org

It's much easier to collaberate this kind of development when you have the
code in SVN.


2009/10/22 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-22 16:04 +0200, Tihomir Culjaga wrote
> freeswitch-users at lists.fre...:
>
> TC>On Thu, Oct 22, 2009 at 3:59 PM, Tihomir Culjaga <tculjaga at gmail.com>
> wrote:
> TC>
> TC>>
> TC>>> TC>Hi, here is the FS log without crash-protection:
> TC>>> TC>http://pastebin.freeswitch.org/10796 and here is the backtrace:
> TC>>> TC>http://pastebin.freeswitch.org/10797
> TC>>>
> TC>>> i fix this crash already, please download latest version from same
> link
> TC>>> as previous, recompile and try again.
> TC>>>
> TC>>>
> TC>> outgoing works, I can place an end-to-end call ... except the RTP is
> realy
> TC>> delayed ... after approx 30 sec of conversation the audio is delayed
> more
> TC>> than 10 seconds.... but i have 2 way audio for outgoing calls:)
> TC>>
> TC>>
> TC>one more thing ... it is H323 endpoint => SIP phone audio that is
> delayed.
> TC>SIP phone => H323 endpoint is ok!
>
> hm, i have such issue but in reverce direction now.
>
> TC>> Do you need some logs ?
> TC>>
> TC>>
> TC>> Inbound cals still the same... i suppose you didn't have a chance
> working
> TC>> on that as well ...
> TC>>
> TC>> T.
> TC>>
> TC>
>
> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129-RIPE
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From brian at freeswitch.org  Thu Oct 22 07:30:00 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 22 Oct 2009 09:30:00 -0500
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910221811590.1929@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<65d96fc80910220704t283bfca3w2129ff3575659f97@mail.gmail.com>
	<Pine.LNX.4.64.0910221811590.1929@srv.icf.org.ru>
Message-ID: <DE0920EE-584A-4DF9-B8D8-F2C0DF94E8DF@freeswitch.org>

Have you started moving the code into our SVN and using our  
ticketing / issue tracker to help you manage issues?

/b

On Oct 22, 2009, at 9:13 AM, Georgiewskiy Yuriy wrote:

> On 2009-10-22 16:04 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre 
> ...:
>
> TC>On Thu, Oct 22, 2009 at 3:59 PM, Tihomir Culjaga <tculjaga at gmail.com 
> > wrote:
> TC>
> TC>>
> TC>>> TC>Hi, here is the FS log without crash-protection:
> TC>>> TC>http://pastebin.freeswitch.org/10796 and here is the  
> backtrace:
> TC>>> TC>http://pastebin.freeswitch.org/10797
> TC>>>
> TC>>> i fix this crash already, please download latest version from  
> same link
> TC>>> as previous, recompile and try again.
> TC>>>
> TC>>>
> TC>> outgoing works, I can place an end-to-end call ... except the  
> RTP is realy
> TC>> delayed ... after approx 30 sec of conversation the audio is  
> delayed more
> TC>> than 10 seconds.... but i have 2 way audio for outgoing calls:)
> TC>>
> TC>>
> TC>one more thing ... it is H323 endpoint => SIP phone audio that is  
> delayed.
> TC>SIP phone => H323 endpoint is ok!
>
> hm, i have such issue but in reverce direction now.
>
> TC>> Do you need some logs ?
> TC>>
> TC>>
> TC>> Inbound cals still the same... i suppose you didn't have a  
> chance working
> TC>> on that as well ...
> TC>>
> TC>> T.
> TC>>
> TC>
>
> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129- 
> RIPE_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From bottleman at icf.org.ru  Thu Oct 22 07:34:53 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Thu, 22 Oct 2009 18:34:53 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <DE0920EE-584A-4DF9-B8D8-F2C0DF94E8DF@freeswitch.org>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<65d96fc80910220704t283bfca3w2129ff3575659f97@mail.gmail.com>
	<Pine.LNX.4.64.0910221811590.1929@srv.icf.org.ru>
	<DE0920EE-584A-4DF9-B8D8-F2C0DF94E8DF@freeswitch.org>
Message-ID: <Pine.LNX.4.64.0910221833420.6837@srv.icf.org.ru>

On 2009-10-22 09:30 -0500, Brian West wrote freeswitch-users at lists.freeswit...:

hm, you not tell me what account created, and i don't try to do this.

BW>Have you started moving the code into our SVN and using our  
BW>ticketing / issue tracker to help you manage issues?
BW>
BW>/b
BW>
BW>On Oct 22, 2009, at 9:13 AM, Georgiewskiy Yuriy wrote:
BW>
BW>> On 2009-10-22 16:04 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre 
BW>> ...:
BW>>
BW>> TC>On Thu, Oct 22, 2009 at 3:59 PM, Tihomir Culjaga <tculjaga at gmail.com 
BW>> > wrote:
BW>> TC>
BW>> TC>>
BW>> TC>>> TC>Hi, here is the FS log without crash-protection:
BW>> TC>>> TC>http://pastebin.freeswitch.org/10796 and here is the  
BW>> backtrace:
BW>> TC>>> TC>http://pastebin.freeswitch.org/10797
BW>> TC>>>
BW>> TC>>> i fix this crash already, please download latest version from  
BW>> same link
BW>> TC>>> as previous, recompile and try again.
BW>> TC>>>
BW>> TC>>>
BW>> TC>> outgoing works, I can place an end-to-end call ... except the  
BW>> RTP is realy
BW>> TC>> delayed ... after approx 30 sec of conversation the audio is  
BW>> delayed more
BW>> TC>> than 10 seconds.... but i have 2 way audio for outgoing calls:)
BW>> TC>>
BW>> TC>>
BW>> TC>one more thing ... it is H323 endpoint => SIP phone audio that is  
BW>> delayed.
BW>> TC>SIP phone => H323 endpoint is ok!
BW>>
BW>> hm, i have such issue but in reverce direction now.
BW>>
BW>> TC>> Do you need some logs ?
BW>> TC>>
BW>> TC>>
BW>> TC>> Inbound cals still the same... i suppose you didn't have a  
BW>> chance working
BW>> TC>> on that as well ...
BW>> TC>>
BW>> TC>> T.
BW>> TC>>
BW>> TC>
BW>>
BW>> C ?????????                       With Best Regards
BW>> ???????????? ????.                Georgiewskiy Yuriy
BW>> +7 4872 711666                    +7 4872 711666
BW>> ???? +7 4872 711143               fax +7 4872 711143
BW>> ???????? ??? "?? ?? ??????"       IT Service Ltd
BW>> http://nkoort.ru                  http://nkoort.ru
BW>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
BW>> YG129-RIPE                        YG129- 
BW>> RIPE_______________________________________________
BW>> FreeSWITCH-users mailing list
BW>> FreeSWITCH-users at lists.freeswitch.org
BW>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
BW>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
BW>> users
BW>> http://www.freeswitch.org
BW>
BW>
BW>_______________________________________________
BW>FreeSWITCH-users mailing list
BW>FreeSWITCH-users at lists.freeswitch.org
BW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
BW>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
BW>http://www.freeswitch.org
BW>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From brian at freeswitch.org  Thu Oct 22 07:41:59 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 22 Oct 2009 09:41:59 -0500
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910221833420.6837@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<65d96fc80910220704t283bfca3w2129ff3575659f97@mail.gmail.com>
	<Pine.LNX.4.64.0910221811590.1929@srv.icf.org.ru>
	<DE0920EE-584A-4DF9-B8D8-F2C0DF94E8DF@freeswitch.org>
	<Pine.LNX.4.64.0910221833420.6837@srv.icf.org.ru>
Message-ID: <4EA71BA6-3E47-4A08-9812-2FE6627394E6@freeswitch.org>

AS per the email you and I exchanged we created the account and the  
mod_h323 folder in endpoints....

/b

On Oct 22, 2009, at 9:34 AM, Georgiewskiy Yuriy wrote:

> hm, you not tell me what account created, and i don't try to do this.



From bottleman at icf.org.ru  Thu Oct 22 07:44:42 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Thu, 22 Oct 2009 18:44:42 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <191c3a030910220727v7e62f47dhf172ba8682d5d52a@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<65d96fc80910220704t283bfca3w2129ff3575659f97@mail.gmail.com>
	<Pine.LNX.4.64.0910221811590.1929@srv.icf.org.ru>
	<191c3a030910220727v7e62f47dhf172ba8682d5d52a@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910221842540.8225@srv.icf.org.ru>

On 2009-10-22 09:27 -0500, Anthony Minessale wrote freeswitch-users at lists.f...:

AM>crash protection has been completely removed from FreeSWITCH, I certianly
AM>hope you are working on this against SVN trunk?  

i don't have trunk at this time, my current work is based on 1.0.4 version.

AM>Also you have been given an
AM>svn area and a jira category for this so you should move all the info from
AM>this thread to jira http://jira.freeswitch.org
AM>
AM>It's much easier to collaberate this kind of development when you have the
AM>code in SVN.
AM>
AM>
AM>2009/10/22 Georgiewskiy Yuriy <bottleman at icf.org.ru>
AM>
AM>> On 2009-10-22 16:04 +0200, Tihomir Culjaga wrote
AM>> freeswitch-users at lists.fre...:
AM>>
AM>> TC>On Thu, Oct 22, 2009 at 3:59 PM, Tihomir Culjaga <tculjaga at gmail.com>
AM>> wrote:
AM>> TC>
AM>> TC>>
AM>> TC>>> TC>Hi, here is the FS log without crash-protection:
AM>> TC>>> TC>http://pastebin.freeswitch.org/10796 and here is the backtrace:
AM>> TC>>> TC>http://pastebin.freeswitch.org/10797
AM>> TC>>>
AM>> TC>>> i fix this crash already, please download latest version from same
AM>> link
AM>> TC>>> as previous, recompile and try again.
AM>> TC>>>
AM>> TC>>>
AM>> TC>> outgoing works, I can place an end-to-end call ... except the RTP is
AM>> realy
AM>> TC>> delayed ... after approx 30 sec of conversation the audio is delayed
AM>> more
AM>> TC>> than 10 seconds.... but i have 2 way audio for outgoing calls:)
AM>> TC>>
AM>> TC>>
AM>> TC>one more thing ... it is H323 endpoint => SIP phone audio that is
AM>> delayed.
AM>> TC>SIP phone => H323 endpoint is ok!
AM>>
AM>> hm, i have such issue but in reverce direction now.
AM>>
AM>> TC>> Do you need some logs ?
AM>> TC>>
AM>> TC>>
AM>> TC>> Inbound cals still the same... i suppose you didn't have a chance
AM>> working
AM>> TC>> on that as well ...
AM>> TC>>
AM>> TC>> T.
AM>> TC>>
AM>> TC>
AM>>
AM>> C ?????????                       With Best Regards
AM>> ???????????? ????.                Georgiewskiy Yuriy
AM>> +7 4872 711666                    +7 4872 711666
AM>> ???? +7 4872 711143               fax +7 4872 711143
AM>> ???????? ??? "?? ?? ??????"       IT Service Ltd
AM>> http://nkoort.ru                  http://nkoort.ru
AM>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
AM>> YG129-RIPE                        YG129-RIPE
AM>>
AM>> _______________________________________________
AM>> FreeSWITCH-users mailing list
AM>> FreeSWITCH-users at lists.freeswitch.org
AM>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
AM>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
AM>> http://www.freeswitch.org
AM>>
AM>>
AM>
AM>
AM>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From codecomplete at free.fr  Thu Oct 22 07:48:40 2009
From: codecomplete at free.fr (Fred-145)
Date: Thu, 22 Oct 2009 07:48:40 -0700 (PDT)
Subject: [Freeswitch-users] Porting Freeswitch to ARM?
In-Reply-To: <737802AA-46AB-4327-8F26-E0F7D3E6D921@jerris.com>
References: <25531239.post@talk.nabble.com>
	<737802AA-46AB-4327-8F26-E0F7D3E6D921@jerris.com>
Message-ID: <26011319.post@talk.nabble.com>


Thanks Mike for the link. I'll investigate more whether running FS on
ARM-based devices is a good idea.

For those interested, another thread on the subject:

http://www.nabble.com/Freeswitch-vs.-Asterisk-speed-on-ARM-td25086585.html
-- 
View this message in context: http://www.nabble.com/Porting-Freeswitch-to-ARM--tp25531239p26011319.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From tculjaga at gmail.com  Thu Oct 22 07:53:06 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 22 Oct 2009 16:53:06 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>
Message-ID: <65d96fc80910220753n6d2efa39l470791f75db9a640@mail.gmail.com>

>
> TC>
> TC>Do you need some logs ?
>
> try disable medai-proxy, there is issue with rtp now then medai-proxy or
> transcoding enabled.
>
>
Outbound calls:

disabled rtp proxy and it is still the same issue ... audio delay H323 =>
SIP endpoint.






Inbound calls:

This is the extension i use to register my Avaya SIP phone to FS.


<include>
  <user id="1001">
    <params>
      <param name="password" value="$${default_password}"/>
      <param name="vm-password" value="1001"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="1001"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="Extension 1001"/>
      <variable name="effective_caller_id_number" value="1001"/>
      <variable name="outbound_caller_id_name"
value="$${outbound_caller_name}"/>
      <variable name="outbound_caller_id_number"
value="$${outbound_caller_id}"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>


This is my h323.conf.xml


<configuration name="h323.conf" description="H323 Endpoints">
  <settings>
    <param name="trace-level" value="4"/>
    <param name="context" value="default"/>
    <param name="dialplan" value="XML"/>
    <param name="codec-prefs" value="PCMU,PCMA,GSM,G729,G726"/>
    <param name="gk-address" value=""/>    <!-- empty to disable, "*" to
search LAN -->
    <param name="gk-identifer" value=""/>  <!-- optional name of gk -->
    <param name="gk-interface" value=""/>  <!-- optional listener interface
name -->
  </settings>
  <listeners>
    <listener name="default">
      <param name="h323-ip" value="10.4.62.7"/>
      <param name="h323-port" value="1720"/>
    </listener>
  </listeners>
</configuration>

I'm using default context and an inbound call looks for a registered user in
default context where 1001 user is registered to.



here is the log for an outgoing call:
http://pastebin.freeswitch.org/10799and here is a tshark output:
http://pastebin.freeswitch.org/10800


there are 2 thing that are not working here:


1. no audio at all!
2. hangup from SIP User side doesn't release the H323 leg











two points for your reference in the logs:


1. Here, SIP User disconnected the SIP leg, but nothing was triggered in
mod_h323 ... as the callback function "on_hangup" (in mod_h323.cpp) was
never triggered!

freeswitch at subZero>
freeswitch at subZero>
freeswitch at subZero> recv 371 bytes from udp/[10.4.62.89]:5060 at
14:39:36.714521:
   ------------------------------------------------------------------------
   BYE sip:mod_sofia at 10.4.62.7:5060 SIP/2.0
   From: <sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>
;transport=udp>;tag=-22166b474ae08abf-7_T10.4.62.89
   To: sip:1001282166 at 10.4.62.7 <sip%3A1001282166 at 10.4.62.7>
;tag=Qpc53NZ2cZc1N
   Call-ID: 8aa825c6-39bb-122d-bb89-00110a5be1f0
   CSeq: 127 BYE
   Via: SIP/2.0/UDP
10.4.62.89;branch=z9hG4bK-7e5dc720_442d0f8-2d8bf1174f235bec_B
   Content-Length: 0
   Max-Forwards: 70
   Supported: replaces

   ------------------------------------------------------------------------
2009-10-22 16:39:36.714604 [NOTICE] sofia.c:322 Hangup sofia/internal/
sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [CS_CONSUME_MEDIA]
[NORMAL_CLEARING]
2009-10-22 16:39:36.714604 [DEBUG] switch_channel.c:1683 Send signal
sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [KILL]
2009-10-22 16:39:36.714604 [DEBUG] switch_core_session.c:932 Send signal
sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [BREAK]
send 520 bytes to udp/[10.4.62.89]:5060 at 14:39:36.715258:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP
10.4.62.89;branch=z9hG4bK-7e5dc720_442d0f8-2d8bf1174f235bec_B
   From: <sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>
;transport=udp>;tag=-22166b474ae08abf-7_T10.4.62.89
   To: sip:1001282166 at 10.4.62.7 <sip%3A1001282166 at 10.4.62.7>
;tag=Qpc53NZ2cZc1N
   Call-ID: 8aa825c6-39bb-122d-bb89-00110a5be1f0
   CSeq: 127 BYE
   User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Content-Length: 0

   ------------------------------------------------------------------------
2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:503
(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State
CONSUME_MEDIA going to sleep
2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) Running State
Change CS_HANGUP
2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:434
(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State HANGUP
2009-10-22 16:39:36.721097 [DEBUG] mod_sofia.c:338 Channel sofia/internal/
sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> hanging up, cause:
NORMAL_CLEARING
2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:46
sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> Standard HANGUP,
cause: NORMAL_CLEARING
2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:434
(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State HANGUP
going to sleep
2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:476
(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State Change
CS_HANGUP -> CS_REPORTING
2009-10-22 16:39:36.721097 [DEBUG] switch_core_session.c:932 Send signal
sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [BREAK]
2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) Running State
Change CS_REPORTING
2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:612
(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State REPORTING
2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:53
sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> Standard
REPORTING, cause: NORMAL_CLEARING
2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:612
(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State REPORTING
going to sleep
2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:411
(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State Change
CS_REPORTING -> CS_DESTROY
2009-10-22 16:39:36.721097 [DEBUG] switch_core_session.c:1068 Session 10
(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) Locked, Waiting
on external entities

freeswitch at subZero>





2. switch_ivr_originate.c complains => 2009-10-22 16:39:41.630243 [ERR]
switch_ivr_originate.c:1510 Cannot create outgoing channel of type [user]
cause: [NORMAL_CLEARING]


freeswitch at subZero>
freeswitch at subZero>
freeswitch at subZero> 2009-10-22 16:39:41.630243 [DEBUG] h323pdu.cxx:609
Receiving PDU [ip$10.4.62.7:1720/ip$10.4.62.31:16013] :
  {
    q931pdu = {
      protocolDiscriminator = 8
      callReference = 23
      from = originator
      messageType = ReleaseComplete
      IE: Cause - Normal ca
ll clearing = {
        80 90                                              ..
      }
      IE: User-User = {
        25 80 06 00 08 91 4a 00  04 11 00 11 00 96 40 1f   %.....J....... at .
        4e be 4f 11 de 80 76 ae  99 96 03 ba ff 10 80 01   N.O.
..v.........
        ...
      }
    }
    h225pdu = {
      h323_uu_pdu = {
        h323_message_body = releaseComplete {
          protocolIdentifier = 0.0.8.2250.0.4
          callIdentifier = {
            guid =  16 octets {
              96 40
 1f 4e be 4f 11 de  80 76 ae 99 96 03 ba ff   . at .N.O...v......
            }
          }
        }
        h245Tunneling = true
      }
    }
  }
2009-10-22 16:39:41.630243 [DEBUG] h323.cxx:1292 Handling PDU:
ReleaseComplete callRef=23
2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:745
======>FSH323Connection::OnReceivedReleaseComplete cause = Normal call
clearing value = 16
2009-10-22 16:39:41.630243 [NOTICE] mod_h323.cpp:749 Hangup h323/1001
[CS_EXECUTE] [NORMAL_CLEARING]
2009-10-22 16:39:41.630243 [DEBUG] switch_channel.c:1683 Send signal
h323/1001 [KILL]
2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:840
======>FSH323Connection::kill_channel FSH323Connection
2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:841 Kill 1 on connection
FSH323Connection
2009-10-22 16:39:41.630243 [DEBUG] switch_core_session.c:630 Send signal
h323/1001 [BREAK]
2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:840
======>FSH323Connection::kill_channel FSH323Connection
2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:841 Kill 3 on connection
FSH323Connection
2009-10-22 16:39:41.630243 [DEBUG] switch_ivr_originate.c:2138 Originate
Resulted in Error Cause: 16 [NORMAL_CLEARING]
2009-10-22 16:39:41.630243 [ERR] switch_ivr_originate.c:1510 Cannot create
outgoing channel of type [user] cause: [NORMAL_CLEARING]
2009-10-22 16:39:41.630243 [DEBUG] switch_ivr_originate.c:2138 Originate
Resulted in Error Cause: 16 [NORMAL_CLEARING]
2009-10-22 16:39:41.630243 [INFO] mod_dptools.c:2093 Originate Failed.
Cause: NORMAL_CLEARING
2009-10-22 16:39:41.630243 [DEBUG] mod_dptools.c:2120 Continue on fail
[true]:  Cause: NORMAL_CLEARING
2009-10-22 16:39:41.630243 [DEBUG] switch_core_state_machine.c:491
(h323/1001) State EXECUTE going to sleep
2009-10-22 16:39:41.630243 [DEBUG] switch_core_state_machine.c:398
(h323/1001) Running State Change CS_HANGUP
2009-10-22 16:39:41.630243 [DEBUG] switch_core_state_machine.c:434
(h323/1001) State HANGUP
2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:1454 ======>switch_status_t
on_hangup
2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:1459 ----->
2009-10-22 16:39:41.630243 [DEBUG] h323pdu.cxx:1272 Call End Reason Normal
call clearing
2009-10-22 16:39:41.630243 [DEBUG] h323ep.cxx:2694 Clearing connection ip$
10.4.62.31:16013/23 reason=EndedByRemoteUser
2009-10-22 16:39:41.630243 [DEBUG] h323.cxx:1034 Call end reason for ip$
10.4.62.31:16013/23 set to EndedByRemoteUser
2009-10-22 16:39:41.630243 [DEBUG] h323.cxx:1052 Sending release complete
PDU: callRef=23
2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:764
======>FSH323Connection::OnSendReleaseComplete cause = Normal call clearing
value = 16
2009-10-22 16:39:41.630243 [DEBUG] h323pdu.cxx:609 Sending PDU
[(noaddr)/(noaddr)] :
  command endSessionCommand disconnect <<null>>
2009-10-22 16:39:41.630243 [DEBUG] h323pdu.cxx:609 Sending PDU [ip$
10.4.62.7:1720/ip$10.4.62.31:16013] :
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From tculjaga at gmail.com  Thu Oct 22 08:01:39 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 22 Oct 2009 17:01:39 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910221842540.8225@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<65d96fc80910220704t283bfca3w2129ff3575659f97@mail.gmail.com>
	<Pine.LNX.4.64.0910221811590.1929@srv.icf.org.ru>
	<191c3a030910220727v7e62f47dhf172ba8682d5d52a@mail.gmail.com>
	<Pine.LNX.4.64.0910221842540.8225@srv.icf.org.ru>
Message-ID: <65d96fc80910220801m53996d32h531d1e65aca77f0@mail.gmail.com>

2009/10/22 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-22 09:27 -0500, Anthony Minessale wrote
> freeswitch-users at lists.f...:
>
> AM>crash protection has been completely removed from FreeSWITCH, I
> certianly
> AM>hope you are working on this against SVN trunk?
>
> i don't have trunk at this time, my current work is based on 1.0.4 version.
>
>
Yuriy,

it is better if we move this through a jira ticket, this way it is a mess.
So if you agree, we can open a ticket where we can follow up all issues with
mod_h323.
Also, the same applies to FS trunk... first i wanted to see if i was doing
something wrong when i tried your module. Now, when you fixed outgoing calls
it is time to go on trunk as when we finish this 1.0.4 will be outdated and
obsolete.


so, to continue on this topic i suggest:

1. open a jira ticket
2. move to fs-trunk
3. upload the current src of mod_h323 to the FSSVN


do you agree ?



Tihomir.
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From pippyduck1127 at hotmail.com  Thu Oct 22 08:34:22 2009
From: pippyduck1127 at hotmail.com (Pajongjit Buntaokit)
Date: Thu, 22 Oct 2009 15:34:22 +0000
Subject: [Freeswitch-users] Problem when load CDR Data on the database in
	Real Time
Message-ID: <BAY119-W443376A5BEC1012E7888BDB3BE0@phx.gbl>


Hello Anyone,

Now, I'm attempting to load CDR Data on the database in Real Time.
by following the instruction on this link 
http://wiki.freeswitch.org/wiki/CDR


However, when trying to send create_table.rb to the database,
I'm still struggling with connecting to mySQL database

Note: 

my Operation System is WindowXP 2002.

the ruby version that I used is 1.9.1 p129.

Below is the error message show in the Command Prompt:
------------------------------------------------------------------------- 
C:\cdr>ruby create_table.rb
C:/Ruby19/lib/ruby/gems/1.9.1/gems/sequel-3.5.0/lib/sequel/adapters/mysql.rb:157
:in `query': Mysql::Error: Commands out of sync; you can't run this command now
(Sequel::DatabaseDisconnectError)
        from C:/Ruby19/lib/ruby/gems/1.9.1/gems/sequel-3.5.0/lib/sequel/adapters
/mysql.rb:157:in `_execute'
        from C:/Ruby19/lib/ruby/gems/1.9.1/gems/sequel-3.5.0/lib/sequel/adapters
/mysql.rb:140:in `block in execute'
        from C:/Ruby19/lib/ruby/gems/1.9.1/gems/sequel-3.5.0/lib/sequel/connecti
on_pool.rb:112:in `hold'
        from C:/Ruby19/lib/ruby/gems/1.9.1/gems/sequel-3.5.0/lib/sequel/database
.rb:482:in `synchronize'
        from C:/Ruby19/lib/ruby/gems/1.9.1/gems/sequel-3.5.0/lib/sequel/adapters
/mysql.rb:140:in `execute'
        from C:/Ruby19/lib/ruby/gems/1.9.1/gems/sequel-3.5.0/lib/sequel/database
.rb:313:in `execute_dui'
        from C:/Ruby19/lib/ruby/gems/1.9.1/gems/sequel-3.5.0/lib/sequel/database
.rb:306:in `execute_ddl'
        from C:/Ruby19/lib/ruby/gems/1.9.1/gems/sequel-3.5.0/lib/sequel/database
/schema_methods.rb:188:in `create_table_from_generator'
        from C:/Ruby19/lib/ruby/gems/1.9.1/gems/sequel-3.5.0/lib/sequel/database
/schema_methods.rb:73:in `create_table'
        from create_table.rb:6:in `<main>'
<main>:306: [BUG] Segmentation fault
ruby 1.9.1p129 (2009-05-12 revision 23412) [i386-mswin32]

-- control frame ----------
c:0001 p:0000 s:0002 b:0002 l:0010c4 d:0010c4 TOP    <main>:306
---------------------------
-- Ruby level backtrace information-----------------------------------------

[NOTE]
You may encounter a bug of Ruby interpreter. Bug reports are welcome.
For details: http://www.ruby-lang.org/bugreport.html


This application has requested the Runtime to terminate it in an unusual way.
Please contact the application's support team for more information.

C:\cdr>
------------------------------------------------------------------------- 

Has anyone ever had this problem? Pleaes advise.

PB.






 		 	   		  
_________________________________________________________________
Windows 7: Simplify your PC. Learn more.
http://www.microsoft.com/Windows/windows-7/default.aspx?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_evergreen1:102009
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From freeswitch.noob at gmail.com  Thu Oct 22 08:42:11 2009
From: freeswitch.noob at gmail.com (freeswitch noob)
Date: Thu, 22 Oct 2009 10:42:11 -0500
Subject: [Freeswitch-users] Domain Question
Message-ID: <7f783b530910220842q5c01ddbcj28a6157c30badd52@mail.gmail.com>

I have a noobish question about setting up FS.

I have it installed and running.

I setup a soft client on the machine fs is on and point it to the ip address
of the FS instance and it registers with no issues.

I then setup an entry in my etc/hosts files "mydomain.localhost" and changed
the domain in the soft client.

The registration now fails. with the soft phone giving a 503 error "service
unavailable"

And I am watching the cli and I see no errors come through with the domain
but I see it register with the IP :(

I change it back to the ip address and it works.

Any direction you guys could point me in would be great.
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From kristian.kielhofner at gmail.com  Thu Oct 22 08:44:30 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Thu, 22 Oct 2009 11:44:30 -0400
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
In-Reply-To: <191c3a030910211154gc8412b5gd999fbedffa02b53@mail.gmail.com>
References: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>
	<191c3a030910201434q4c6e65e1h58c5f0a82addc2d3@mail.gmail.com>
	<2d9149cd0910202050i36a7ea82nc7253d87e38f990a@mail.gmail.com>
	<191c3a030910211154gc8412b5gd999fbedffa02b53@mail.gmail.com>
Message-ID: <2d9149cd0910220844v2aa73502j4b88fdbd75a0d833@mail.gmail.com>

An update for Tony, Brian, Mike, and everyone on the list...

I was able to get some phone time with the team yesterday.  Tony
worked on my machine, found the issue, and had it committed within 30
minutes.

I've been testing T.38 all morning between the fax machines in the
office with few issues.

THANKS AGAIN GUYS!

On Wed, Oct 21, 2009 at 2:54 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> can you try trunk and let me know right away,
> if it's still not working i may need ssh access and call you on the phone.
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From helmut.kuper at ewetel.de  Thu Oct 22 08:44:52 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Thu, 22 Oct 2009 17:44:52 +0200
Subject: [Freeswitch-users] Qustion about INFO messages
	after	Connect/Answer
In-Reply-To: <87f2f3b90910211048s208daf1eyfae501091ab75530@mail.gmail.com>
References: <4AD859A4.5000101@ewetel.de>	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>	<4ADC7C09.6070503@ewetel.de>	<191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>	<4ADC85C0.7060908@ewetel.de>
	<4ADC8B6E.7070404@ewetel.de>	<191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>	<87f2f3b90910191435s7528e256r85df5f12fc634f24@mail.gmail.com>	<4ADF483B.4050807@ewetel.de>
	<87f2f3b90910211048s208daf1eyfae501091ab75530@mail.gmail.com>
Message-ID: <4AE07DF4.5040803@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Mike,

here it is:


Dialplan:


    <extension name="Local_Extension">
      <condition field="destination_number" expression="^(10[01][0-9])$">
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
        <action application="set" data="transfer_ringback=$${hold_music}"/>
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="bridge"
data="user/${dialed_extension}@${domain_name}"/>
      </condition>
    </extension>






Debug-Log:


recv 1521 bytes from udp/[85.16.245.206]:1024 at 15:41:02.405834:
   ------------------------------------------------------------------------
   INVITE sip:1000 at 85.16.246.12:5061;user=phone SIP/2.0
   Via: SIP/2.0/UDP 85.16.245.206:1024;branch=z9hG4bK-i48n75d62qts;rport
   From: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=7xpim4o1go
   To: <sip:1000 at 85.16.246.12:5061;user=phone>
   Call-ID: 3c31304b7a80-no9xsnjj0bol
   CSeq: 1 INVITE
   Max-Forwards: 70
   Contact: <sip:1001 at 85.16.245.206:1024;line=eg3wp69a>;reg-id=1
   X-Serialnumber: 0004134002CB
   P-Key-Flags: resolution="31x13", keys="4"
   User-Agent: snom820/8.2.16
   Accept: application/sdp
   Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE
   Allow-Events: talk, hold, refer, call-info
   Supported: timer, 100rel, replaces, from-change
   Session-Expires: 3600;refresher=uas
   Min-SE: 90
   Content-Type: application/sdp
   Content-Length: 732

   v=0
   o=root 411395140 411395140 IN IP4 85.16.245.206
   s=call
   c=IN IP4 85.16.245.206
   t=0 0
   m=audio 49852 RTP/SAVP 0 8 9 99 3 18 4 101
   a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:Um06txYC7vwXdZDohXJ15te5hZ/iWmPd4voRbdby
   a=rtpmap:0 pcmu/8000
   a=rtpmap:8 pcma/8000
   a=rtpmap:9 g722/8000
   a=rtpmap:99 g726-32/8000
   a=rtpmap:3 gsm/8000
   a=rtpmap:18 g729/8000
   a=rtpmap:4 g723/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   a=sendrecv
   m=audio 49852 RTP/AVP 0 8 9 99 3 18 4 101
   a=rtpmap:0 pcmu/8000
   a=rtpmap:8 pcma/8000
   a=rtpmap:9 g722/8000
   a=rtpmap:99 g726-32/8000
   a=rtpmap:3 gsm/8000
   a=rtpmap:18 g729/8000
   a=rtpmap:4 g723/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   a=sendrecv
   ------------------------------------------------------------------------
send 331 bytes to udp/[85.16.245.206]:1024 at 15:41:02.406500:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP
85.16.245.206:1024;branch=z9hG4bK-i48n75d62qts;rport=1024
   From: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=7xpim4o1go
   To: <sip:1000 at 85.16.246.12:5061;user=phone>
   Call-ID: 3c31304b7a80-no9xsnjj0bol
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15180M
   Content-Length: 0

   ------------------------------------------------------------------------
2009-10-22 17:41:02.406509 [DEBUG] sofia.c:4907 IP 85.16.245.206
Approved by acl "clients[]". Access Granted.
2009-10-22 17:41:02.406509 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/1001 at 85.16.246.12:5061 [4d941750-bf21-11de-9c3f-adfc1789590a]
2009-10-22 17:41:02.407509 [DEBUG] sofia.c:3493 Channel
sofia/internal/1001 at 85.16.246.12:5061 entering state [received][100]
2009-10-22 17:41:02.407509 [DEBUG] sofia.c:3500 Remote SDP:
v=0
o=root 411395140 411395140 IN IP4 85.16.245.206
s=call
c=IN IP4 85.16.245.206
t=0 0
m=audio 49852 RTP/SAVP 0 8 9 99 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:Um06txYC7vwXdZDohXJ15te5hZ/iWmPd4voRbdby
a=ptime:20
m=audio 49852 RTP/AVP 0 8 9 99 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

2009-10-22 17:41:02.407509 [DEBUG] sofia.c:3621
(sofia/internal/1001 at 85.16.246.12:5061) State Change CS_NEW -> CS_INIT
2009-10-22 17:41:02.407509 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-22 17:41:02.407509 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_INIT
2009-10-22 17:41:02.407509 [DEBUG] switch_core_state_machine.c:330
(sofia/internal/1001 at 85.16.246.12:5061) State INIT
2009-10-22 17:41:02.407509 [DEBUG] mod_sofia.c:83
sofia/internal/1001 at 85.16.246.12:5061 SOFIA INIT
2009-10-22 17:41:02.407509 [DEBUG] mod_sofia.c:111
(sofia/internal/1001 at 85.16.246.12:5061) State Change CS_INIT -> CS_ROUTING
2009-10-22 17:41:02.407509 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-22 17:41:02.407509 [DEBUG] switch_core_state_machine.c:330
(sofia/internal/1001 at 85.16.246.12:5061) State INIT going to sleep
2009-10-22 17:41:02.407509 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_ROUTING
2009-10-22 17:41:02.407509 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/1001 at 85.16.246.12:5061) State ROUTING
2009-10-22 17:41:02.407509 [DEBUG] mod_sofia.c:130
sofia/internal/1001 at 85.16.246.12:5061 SOFIA ROUTING
2009-10-22 17:41:02.407509 [DEBUG] switch_core_state_machine.c:78
sofia/internal/1001 at 85.16.246.12:5061 Standard ROUTING
2009-10-22 17:41:02.407509 [INFO] mod_dialplan_xml.c:391 Processing 1001
an PBX1->1000 in context default
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->anonymous] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [anonymous]
destination_number(1000) =~ /^\*31([0-9]+)$/ break=never
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 ANTI-Action
set(dialed_extension=${destination_number})
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->is_local] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [is_local]
${ET_is_local}() =~ /(true|false)/ break=on-true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 ANTI-Action
lua(ET_is_local.lua)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 ANTI-Action
transfer(${destination_number} XML default)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->set_domain] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (PASS)
[set_domain] destination_number(1000) =~ /^.*$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
set(domain_name=85.16.246.12)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->302_zero_problem] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[302_zero_problem] ${sip_looped_call}() =~ /true/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->fakeTrisko-Servicenumber as callerid] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[fakeTrisko-Servicenumber as callerid] caller_id_number(1001) =~
/^(49[1-2][0-9])$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->outbound-ssw-ddi] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[outbound-ssw-ddi] ${dialed_extension}() =~ /^0([0-9]+)$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing [default->mail]
continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [mail]
destination_number(1000) =~ /^9997$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing [default->pizza]
continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [pizza]
destination_number(1000) =~ /^2006$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing [default->pbook]
continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [pbook]
destination_number(1000) =~ /^2007$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->shout_1] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [shout_1]
${dialed_extension}() =~ /^9123$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->shout_2] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [shout_2]
${dialed_extension}() =~ /^9124$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->speex_32000] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[speex_32000] ${dialed_extension}() =~ /^9125$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->speex_32000_2] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[speex_32000_2] ${dialed_extension}() =~ /^9126$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->jukebox] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [jukebox]
destination_number(1000) =~ /^9111$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->pickup-extension] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[pickup-extension] ${dialed_extension}() =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->Voicemail-Verwaltung] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[Voicemail-Verwaltung] ${dialed_extension}() =~ /^(1001|9999)$/
break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->Local_Extension] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[Local_Extension] ${ET_is_local}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->parallel call] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [parallel
call] ${dialed_extension}() =~ /^(6000)$/ break=on-false
2009-10-22 17:41:02.409510 [DEBUG] switch_core_state_machine.c:114
(sofia/internal/1001 at 85.16.246.12:5061) State Change CS_ROUTING ->
CS_EXECUTE
2009-10-22 17:41:02.409510 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-22 17:41:02.409510 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/1001 at 85.16.246.12:5061) State ROUTING going to sleep
2009-10-22 17:41:02.409510 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_EXECUTE
2009-10-22 17:41:02.409510 [DEBUG] switch_core_state_machine.c:340
(sofia/internal/1001 at 85.16.246.12:5061) State EXECUTE
2009-10-22 17:41:02.409510 [DEBUG] mod_sofia.c:173
sofia/internal/1001 at 85.16.246.12:5061 SOFIA EXECUTE
2009-10-22 17:41:02.409510 [DEBUG] switch_core_state_machine.c:151
sofia/internal/1001 at 85.16.246.12:5061 Standard EXECUTE
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 set(dialed_extension=1000)
2009-10-22 17:41:02.409510 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET [dialed_extension]=[1000]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 lua(ET_is_local.lua)
2009-10-22 17:41:02.434510 [DEBUG] switch_cpp.cpp:925 destroy/unlink
session from object
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 transfer(1000 XML default)
2009-10-22 17:41:02.434510 [DEBUG] switch_ivr.c:1381
(sofia/internal/1001 at 85.16.246.12:5061) State Change CS_EXECUTE ->
CS_ROUTING
2009-10-22 17:41:02.434510 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-22 17:41:02.434510 [DEBUG] switch_ivr.c:1385
sofia/internal/1001 at 85.16.246.12:5061 receive message [TRANSFER]
2009-10-22 17:41:02.434510 [DEBUG] switch_core_session.c:647 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-22 17:41:02.434510 [NOTICE] switch_ivr.c:1387 Transfer
sofia/internal/1001 at 85.16.246.12:5061 to XML[1000 at default]
2009-10-22 17:41:02.435510 [DEBUG] switch_core_state_machine.c:340
(sofia/internal/1001 at 85.16.246.12:5061) State EXECUTE going to sleep
2009-10-22 17:41:02.435510 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_ROUTING
2009-10-22 17:41:02.435510 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/1001 at 85.16.246.12:5061) State ROUTING
2009-10-22 17:41:02.435510 [DEBUG] mod_sofia.c:130
sofia/internal/1001 at 85.16.246.12:5061 SOFIA ROUTING
2009-10-22 17:41:02.435510 [DEBUG] switch_core_state_machine.c:78
sofia/internal/1001 at 85.16.246.12:5061 Standard ROUTING
2009-10-22 17:41:02.435510 [INFO] mod_dialplan_xml.c:391 Processing 1001
an PBX1->1000 in context default
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->anonymous] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [anonymous]
destination_number(1000) =~ /^\*31([0-9]+)$/ break=never
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 ANTI-Action
set(dialed_extension=${destination_number})
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->is_local] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (PASS) [is_local]
${ET_is_local}(true) =~ /(true|false)/ break=on-true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->set_domain] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (PASS)
[set_domain] destination_number(1000) =~ /^.*$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
set(domain_name=85.16.246.12)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->302_zero_problem] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[302_zero_problem] ${sip_looped_call}() =~ /true/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->fakeTrisko-Servicenumber as callerid] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[fakeTrisko-Servicenumber as callerid] caller_id_number(1001) =~
/^(49[1-2][0-9])$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->outbound-ssw-ddi] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[outbound-ssw-ddi] ${dialed_extension}(1000) =~ /^0([0-9]+)$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing [default->mail]
continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [mail]
destination_number(1000) =~ /^9997$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing [default->pizza]
continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [pizza]
destination_number(1000) =~ /^2006$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing [default->pbook]
continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [pbook]
destination_number(1000) =~ /^2007$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->shout_1] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [shout_1]
${dialed_extension}(1000) =~ /^9123$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->shout_2] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [shout_2]
${dialed_extension}(1000) =~ /^9124$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->speex_32000] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[speex_32000] ${dialed_extension}(1000) =~ /^9125$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->speex_32000_2] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[speex_32000_2] ${dialed_extension}(1000) =~ /^9126$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->jukebox] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [jukebox]
destination_number(1000) =~ /^9111$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->pickup-extension] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[pickup-extension] ${dialed_extension}(1000) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->Voicemail-Verwaltung] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[Voicemail-Verwaltung] ${dialed_extension}(1000) =~ /^(1001|9999)$/
break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->Local_Extension] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (PASS)
[Local_Extension] ${ET_is_local}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
set(dialed_extension=${destination_number})
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
export(origination_callee_id_name=hubu)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
export(dialed_extension=${destination_number})
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
set(transfer_ringback=local_stream://moh)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
set(hangup_after_bridge=true)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
export(nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name}
var sip_secure_media)})
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
db(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid})
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
set(continue_on_fail=false)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
bridge(user/${dialed_extension}@85.16.246.12)
2009-10-22 17:41:02.436510 [DEBUG] switch_core_state_machine.c:114
(sofia/internal/1001 at 85.16.246.12:5061) State Change CS_ROUTING ->
CS_EXECUTE
2009-10-22 17:41:02.436510 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-22 17:41:02.436510 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/1001 at 85.16.246.12:5061) State ROUTING going to sleep
2009-10-22 17:41:02.436510 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_EXECUTE
2009-10-22 17:41:02.436510 [DEBUG] switch_core_state_machine.c:340
(sofia/internal/1001 at 85.16.246.12:5061) State EXECUTE
2009-10-22 17:41:02.436510 [DEBUG] mod_sofia.c:173
sofia/internal/1001 at 85.16.246.12:5061 SOFIA EXECUTE
2009-10-22 17:41:02.436510 [DEBUG] switch_core_state_machine.c:151
sofia/internal/1001 at 85.16.246.12:5061 Standard EXECUTE
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 set(dialed_extension=1000)
2009-10-22 17:41:02.436510 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET [dialed_extension]=[1000]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 set(domain_name=85.16.246.12)
2009-10-22 17:41:02.437526 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET [domain_name]=[85.16.246.12]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 set(dialed_extension=1000)
2009-10-22 17:41:02.437526 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET [dialed_extension]=[1000]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061
export(origination_callee_id_name=hubu)
2009-10-22 17:41:02.437526 [DEBUG] mod_dptools.c:835 EXPORT
[origination_callee_id_name]=[hubu]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 export(dialed_extension=1000)
2009-10-22 17:41:02.437526 [DEBUG] mod_dptools.c:835 EXPORT
[dialed_extension]=[1000]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061
set(transfer_ringback=local_stream://moh)
2009-10-22 17:41:02.438511 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET
[transfer_ringback]=[local_stream://moh]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 set(hangup_after_bridge=true)
2009-10-22 17:41:02.438511 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061
export(nolocal:sip_secure_media=)
2009-10-22 17:41:02.461511 [DEBUG] mod_dptools.c:835 EXPORT (REMOTE
ONLY) [sip_secure_media]=[UNDEF]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061
db(insert/85.16.246.12-last_dial//4d941750-bf21-11de-9c3f-adfc1789590a)
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 set(continue_on_fail=false)
2009-10-22 17:41:02.654513 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET [continue_on_fail]=[false]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 bridge(user/1000 at 85.16.246.12)
2009-10-22 17:41:02.679515 [DEBUG] switch_ivr_originate.c:1163 variable
string 0 = [presence_id=1000 at 85.16.246.12]
2009-10-22 17:41:02.679515 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/sip:1000 at 85.16.245.228:1024
[4dbdc384-bf21-11de-9c3f-adfc1789590a]
2009-10-22 17:41:02.679515 [DEBUG] mod_sofia.c:2981
(sofia/internal/sip:1000 at 85.16.245.228:1024) State Change CS_NEW -> CS_INIT
2009-10-22 17:41:02.679515 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-22 17:41:02.679515 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/sip:1000 at 85.16.245.228:1024) Running State Change CS_INIT
2009-10-22 17:41:02.680516 [DEBUG] switch_core_state_machine.c:330
(sofia/internal/sip:1000 at 85.16.245.228:1024) State INIT
2009-10-22 17:41:02.680516 [DEBUG] mod_sofia.c:83
sofia/internal/sip:1000 at 85.16.245.228:1024 SOFIA INIT
send 1310 bytes to udp/[85.16.245.228]:1024 at 15:41:02.681508:
   ------------------------------------------------------------------------
   INVITE sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
   Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKKXH41BcZjB8FS
   Max-Forwards: 68
   From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=HKSNX8D1pKy6K
   To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>
   Call-ID: 251c312d-39c4-122d-3fbf-00144fe6907a
   CSeq: 121994695 INVITE
   Contact: <sip:mod_sofia at 85.16.246.12:5061>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15180M
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
2009-10-22 17:41:02.681515 [DEBUG] mod_sofia.c:111
(sofia/internal/sip:1000 at 85.16.245.228:1024) State Change CS_INIT ->
CS_ROUTING
2009-10-22 17:41:02.681515 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
   Content-Type: application/sdp
   Content-Disposition: session
2009-10-22 17:41:02.681515 [DEBUG] switch_core_state_machine.c:330
(sofia/internal/sip:1000 at 85.16.245.228:1024) State INIT going to sleep
   Content-Length: 325
   X-Serialnumber: 0004134002CB
2009-10-22 17:41:02.681515 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/sip:1000 at 85.16.245.228:1024) Running State Change CS_ROUTING
   P-Key-Flags: resolution="31x13", keys="4"
   X-Actually-Support: UPDATE
   Remote-Party-ID: "1001 an PBX1"
<sip:1001 at 85.16.246.12>;party=calling;screen=yes;privacy=off

   v=0
   o=FreeSWITCH 1256212358 1256212359 IN IP4 85.16.246.12
   s=FreeSWITCH
   c=IN IP4 85.16.246.12
   t=0 0
   m=audio 13704 RTP/AVP 95 103 9 8 101 13
   a=rtpmap:95 CELT/48000
   a=rtpmap:103 SPEEX/32000
   a=rtpmap:9 G722/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:13 CN/8000
   a=ptime:10
   ------------------------------------------------------------------------
2009-10-22 17:41:02.681515 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/sip:1000 at 85.16.245.228:1024) State ROUTING
2009-10-22 17:41:02.681515 [DEBUG] mod_sofia.c:130
sofia/internal/sip:1000 at 85.16.245.228:1024 SOFIA ROUTING
2009-10-22 17:41:02.681515 [DEBUG] switch_ivr_originate.c:66
(sofia/internal/sip:1000 at 85.16.245.228:1024) State Change CS_ROUTING ->
CS_CONSUME_MEDIA
2009-10-22 17:41:02.681515 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-22 17:41:02.681515 [DEBUG] sofia.c:3493 Channel
sofia/internal/sip:1000 at 85.16.245.228:1024 entering state [calling][0]
2009-10-22 17:41:02.681515 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/sip:1000 at 85.16.245.228:1024) State ROUTING going to sleep
2009-10-22 17:41:02.681515 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/sip:1000 at 85.16.245.228:1024) Running State Change
CS_CONSUME_MEDIA
2009-10-22 17:41:02.681515 [DEBUG] switch_core_state_machine.c:352
(sofia/internal/sip:1000 at 85.16.245.228:1024) State CONSUME_MEDIA
2009-10-22 17:41:02.681515 [DEBUG] switch_core_state_machine.c:352
(sofia/internal/sip:1000 at 85.16.245.228:1024) State CONSUME_MEDIA going
to sleep
recv 521 bytes from udp/[85.16.245.228]:1024 at 15:41:02.706318:
   ------------------------------------------------------------------------
   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKKXH41BcZjB8FS
   From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=HKSNX8D1pKy6K
   To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=sdeidvhpxx
   Call-ID: 251c312d-39c4-122d-3fbf-00144fe6907a
   CSeq: 121994695 INVITE
   Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
   Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE
   Allow-Events: talk, hold, refer, call-info
   Content-Length: 0

   ------------------------------------------------------------------------
2009-10-22 17:41:02.706514 [DEBUG] sofia.c:3493 Channel
sofia/internal/sip:1000 at 85.16.245.228:1024 entering state [proceeding][180]
2009-10-22 17:41:02.706514 [NOTICE] sofia.c:3557 Ring-Ready
sofia/internal/sip:1000 at 85.16.245.228:1024!
2009-10-22 17:41:02.706514 [DEBUG] switch_ivr_originate.c:388
sofia/internal/1001 at 85.16.246.12:5061 receive message [RINGING]
2009-10-22 17:41:02.706514 [NOTICE] mod_sofia.c:1525 Ring-Ready
sofia/internal/1001 at 85.16.246.12:5061!
2009-10-22 17:41:02.706514 [DEBUG] switch_core_session.c:647 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-22 17:41:02.706514 [NOTICE] switch_ivr_originate.c:388 Ring
Ready sofia/internal/1001 at 85.16.246.12:5061!
send 737 bytes to udp/[85.16.245.206]:1024 at 15:41:02.706945:
   ------------------------------------------------------------------------
   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP
85.16.245.206:1024;branch=z9hG4bK-i48n75d62qts;rport=1024
   From: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=7xpim4o1go
   To: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=ga0vUDXXSa8Kr
   Call-ID: 3c31304b7a80-no9xsnjj0bol
   CSeq: 1 INVITE
   Contact: <sip:1000 at 85.16.246.12:5061;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15180M
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   X-Actually-Support: UPDATE

   ------------------------------------------------------------------------
2009-10-22 17:41:02.706514 [DEBUG] sofia.c:3493 Channel
sofia/internal/1001 at 85.16.246.12:5061 entering state [early][180]
recv 521 bytes from udp/[85.16.245.228]:1024 at 15:41:03.203506:
   ------------------------------------------------------------------------
   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKKXH41BcZjB8FS
   From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=HKSNX8D1pKy6K
   To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=sdeidvhpxx
   Call-ID: 251c312d-39c4-122d-3fbf-00144fe6907a
   CSeq: 121994695 INVITE
   Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
   Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE
   Allow-Events: talk, hold, refer, call-info
   Content-Length: 0

   ------------------------------------------------------------------------
2009-10-22 17:41:03.203522 [DEBUG] sofia.c:3493 Channel
sofia/internal/sip:1000 at 85.16.245.228:1024 entering state [proceeding][180]
recv 521 bytes from udp/[85.16.245.228]:1024 at 15:41:04.203621:
   ------------------------------------------------------------------------
   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKKXH41BcZjB8FS
   From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=HKSNX8D1pKy6K
   To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=sdeidvhpxx
   Call-ID: 251c312d-39c4-122d-3fbf-00144fe6907a
   CSeq: 121994695 INVITE
   Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
   Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE
   Allow-Events: talk, hold, refer, call-info
   Content-Length: 0

   ------------------------------------------------------------------------
2009-10-22 17:41:04.203541 [DEBUG] sofia.c:3493 Channel
sofia/internal/sip:1000 at 85.16.245.228:1024 entering state [proceeding][180]
recv 888 bytes from udp/[85.16.245.228]:1024 at 15:41:04.773115:
   ------------------------------------------------------------------------
   SIP/2.0 200 Ok
   Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKKXH41BcZjB8FS
   From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=HKSNX8D1pKy6K
   To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=sdeidvhpxx
   Call-ID: 251c312d-39c4-122d-3fbf-00144fe6907a
   CSeq: 121994695 INVITE
   Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
   User-Agent: snom820/LAB_HD_HEAD_11_09_09_11_20
   Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE
   Allow-Events: talk, hold, refer, call-info
   Supported: timer, 100rel, replaces, from-change
   Content-Type: application/sdp
   Content-Length: 242

   v=0
   o=root 45628059 45628060 IN IP4 85.16.245.228
   s=call
   c=IN IP4 85.16.245.228
   t=0 0
   m=audio 53220 RTP/AVP 9 8 101
   a=rtpmap:9 g722/8000
   a=rtpmap:8 pcma/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:10
   a=sendrecv
   ------------------------------------------------------------------------
send 404 bytes to udp/[85.16.245.228]:1024 at 15:41:04.773445:
   ------------------------------------------------------------------------
   ACK sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
   Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKm6aX36v2Fmy2m
   Max-Forwards: 70
   From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=HKSNX8D1pKy6K
   To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=sdeidvhpxx
   Call-ID: 251c312d-39c4-122d-3fbf-00144fe6907a
   CSeq: 121994695 ACK
   Contact: <sip:mod_sofia at 85.16.246.12:5061>
   Content-Length: 0

   ------------------------------------------------------------------------
2009-10-22 17:41:04.773545 [DEBUG] sofia.c:3493 Channel
sofia/internal/sip:1000 at 85.16.245.228:1024 entering state [ready][200]
2009-10-22 17:41:04.773545 [DEBUG] sofia.c:3500 Remote SDP:
v=0
o=root 45628059 45628060 IN IP4 85.16.245.228
s=call
c=IN IP4 85.16.245.228
t=0 0
m=audio 53220 RTP/AVP 9 8 101
a=rtpmap:9 g722/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:10

2009-10-22 17:41:04.773545 [DEBUG] sofia_glue.c:3143 Audio Codec Compare
[g722:9:8000:10]/[CELT:95:48000:10]
2009-10-22 17:41:04.773545 [DEBUG] sofia_glue.c:3143 Audio Codec Compare
[g722:9:8000:10]/[SPEEX:103:32000:20]
2009-10-22 17:41:04.773545 [DEBUG] sofia_glue.c:3143 Audio Codec Compare
[g722:9:8000:10]/[G722:9:8000:20]
2009-10-22 17:41:04.773545 [DEBUG] sofia_glue.c:3143 Audio Codec Compare
[g722:9:8000:10]/[PCMA:8:8000:20]
2009-10-22 17:41:04.773545 [DEBUG] sofia_glue.c:3191 Substituting codec
G722 at 10i@8000h
2009-10-22 17:41:04.773545 [DEBUG] sofia_glue.c:2101 Set Codec
sofia/internal/sip:1000 at 85.16.245.228:1024 G722/8000 10 ms 80 samples
2009-10-22 17:41:04.773545 [DEBUG] sofia_glue.c:3103 Set 2833 dtmf
payload to 101
2009-10-22 17:41:04.773545 [DEBUG] sofia_glue.c:2335 AUDIO RTP
[sofia/internal/sip:1000 at 85.16.245.228:1024] 85.16.246.12 port 13704 ->
85.16.245.228 port 53220 codec: 9 ms: 10
2009-10-22 17:41:04.773545 [DEBUG] switch_rtp.c:1155 Starting timer
[soft] 80 bytes per 10ms
2009-10-22 17:41:04.774778 [DEBUG] switch_channel.c:2105 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-22 17:41:04.774778 [NOTICE] sofia.c:3983 Channel
[sofia/internal/sip:1000 at 85.16.245.228:1024] has been answered
2009-10-22 17:41:04.774778 [DEBUG] switch_channel.c:182
sofia/internal/sip:1000 at 85.16.245.228:1024 receive message [AUDIO_SYNC]
2009-10-22 17:41:04.776545 [DEBUG] switch_ivr_originate.c:2324
sofia/internal/1001 at 85.16.246.12:5061 receive message [ANSWER]
2009-10-22 17:41:04.776545 [DEBUG] sofia_glue.c:3029 Set Remote Key [1
AES_CM_128_HMAC_SHA1_32 inline:Um06txYC7vwXdZDohXJ15te5hZ/iWmPd4voRbdby]
2009-10-22 17:41:04.776545 [DEBUG] sofia_glue.c:2161 Set Local Key [1
AES_CM_128_HMAC_SHA1_32 inline:EKycEU2CPLFUVM4vFRATL21nluVg3RJhmVwJVwA9]
2009-10-22 17:41:04.776545 [DEBUG] sofia_glue.c:3143 Audio Codec Compare
[pcmu:0:8000:20]/[CELT:95:48000:10]
2009-10-22 17:41:04.776545 [DEBUG] sofia_glue.c:3143 Audio Codec Compare
[pcmu:0:8000:20]/[SPEEX:103:32000:20]
2009-10-22 17:41:04.776545 [DEBUG] sofia_glue.c:3143 Audio Codec Compare
[pcmu:0:8000:20]/[G722:9:8000:20]
2009-10-22 17:41:04.776545 [DEBUG] sofia_glue.c:3143 Audio Codec Compare
[pcmu:0:8000:20]/[PCMA:8:8000:20]
2009-10-22 17:41:04.776545 [DEBUG] sofia_glue.c:3143 Audio Codec Compare
[pcma:8:8000:20]/[CELT:95:48000:10]
2009-10-22 17:41:04.776545 [DEBUG] sofia_glue.c:3143 Audio Codec Compare
[pcma:8:8000:20]/[SPEEX:103:32000:20]
2009-10-22 17:41:04.776545 [DEBUG] sofia_glue.c:3143 Audio Codec Compare
[pcma:8:8000:20]/[G722:9:8000:20]
2009-10-22 17:41:04.776545 [DEBUG] sofia_glue.c:3143 Audio Codec Compare
[pcma:8:8000:20]/[PCMA:8:8000:20]
2009-10-22 17:41:04.776545 [DEBUG] sofia_glue.c:2101 Set Codec
sofia/internal/1001 at 85.16.246.12:5061 PCMA/8000 20 ms 160 samples
2009-10-22 17:41:04.776545 [DEBUG] sofia_glue.c:3103 Set 2833 dtmf
payload to 101
2009-10-22 17:41:04.776545 [DEBUG] sofia_glue.c:3077 Our existing sdp is
still good [pcma 85.16.245.206:49852], let's keep it.
2009-10-22 17:41:04.776545 [DEBUG] sofia_glue.c:2335 AUDIO RTP
[sofia/internal/1001 at 85.16.246.12:5061] 85.16.246.12 port 10062 ->
85.16.245.206 port 49852 codec: 8 ms: 20
2009-10-22 17:41:04.776545 [DEBUG] switch_rtp.c:1155 Starting timer
[soft] 160 bytes per 20ms
2009-10-22 17:41:04.778545 [INFO] switch_rtp.c:1019 Activating Secure
RTP SEND
2009-10-22 17:41:04.778545 [INFO] switch_rtp.c:999 Activating Secure RTP
RECV
2009-10-22 17:41:04.778545 [DEBUG] switch_core_sqldb.c:396 Secure Type:
srtp:AES_CM_128_HMAC_SHA1_32
2009-10-22 17:41:04.778545 [DEBUG] switch_core_sqldb.c:396 Secure Type:
srtp:AES_CM_128_HMAC_SHA1_32
2009-10-22 17:41:04.778545 [NOTICE] sofia_glue.c:2770 Pre-Answer
sofia/internal/1001 at 85.16.246.12:5061!
2009-10-22 17:41:04.778545 [DEBUG] mod_sofia.c:543 Local SDP
sofia/internal/1001 at 85.16.246.12:5061:
v=0
o=FreeSWITCH 1256216002 1256216003 IN IP4 85.16.246.12
s=FreeSWITCH
c=IN IP4 85.16.246.12
t=0 0
m=audio 10062 RTP/SAVP 8 101
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:EKycEU2CPLFUVM4vFRATL21nluVg3RJhmVwJVwA9

2009-10-22 17:41:04.778545 [DEBUG] switch_core_session.c:647 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-22 17:41:04.778545 [NOTICE] switch_ivr_originate.c:2324 Channel
[sofia/internal/1001 at 85.16.246.12:5061] has been answered
2009-10-22 17:41:04.778545 [DEBUG] switch_channel.c:182
sofia/internal/1001 at 85.16.246.12:5061 receive message [AUDIO_SYNC]
2009-10-22 17:41:04.778545 [DEBUG] switch_ivr_originate.c:2368 Originate
Resulted in Success: [sofia/internal/sip:1000 at 85.16.245.228:1024]
2009-10-22 17:41:04.778545 [DEBUG] switch_channel.c:182
sofia/internal/sip:1000 at 85.16.245.228:1024 receive message [AUDIO_SYNC]
2009-10-22 17:41:04.778545 [DEBUG] switch_channel.c:182
sofia/internal/1001 at 85.16.246.12:5061 receive message [AUDIO_SYNC]
send 1174 bytes to udp/[85.16.245.206]:1024 at 15:41:04.779182:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP
85.16.245.206:1024;branch=z9hG4bK-i48n75d62qts;rport=1024
   From: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=7xpim4o1go
   To: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=ga0vUDXXSa8Kr
   Call-ID: 3c31304b7a80-no9xsnjj0bol
   CSeq: 1 INVITE
   Contact: <sip:1000 at 85.16.246.12:5061;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15180M
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Session-Expires: 3600;refresher=uas
   Min-SE: 120
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 354
   X-Actually-Support: UPDATE

   v=0
   o=FreeSWITCH 1256216002 1256216003 IN IP4 85.16.246.12
   s=FreeSWITCH
   c=IN IP4 85.16.246.12
   t=0 0
   m=audio 10062 RTP/SAVP 8 101
   a=rtpmap:8 pcma/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20
   a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:EKycEU2CPLFUVM4vFRATL21nluVg3RJhmVwJVwA9
   m=audio 0 RTP/AVP 19
   ------------------------------------------------------------------------
2009-10-22 17:41:04.778545 [DEBUG] sofia.c:3493 Channel
sofia/internal/1001 at 85.16.246.12:5061 entering state [completed][200]
2009-10-22 17:41:04.778545 [DEBUG] sofia.c:407
sofia/internal/sip:1000 at 85.16.245.228:1024 receive message [DISPLAY]
send 674 bytes to udp/[85.16.245.228]:1024 at 15:41:04.779630:
   ------------------------------------------------------------------------
   INFO sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
   Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKNF4N51D6cXmNg
   Max-Forwards: 70
   From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=HKSNX8D1pKy6K
   To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=sdeidvhpxx
2009-10-22 17:41:04.779555 [DEBUG] switch_ivr_originate.c:2368 Originate
Resulted in Success: [sofia/internal/sip:1000 at 85.16.245.228:1024]
   Call-ID: 251c312d-39c4-122d-3fbf-00144fe6907a
   CSeq: 121994696 INFO
   Contact: <sip:mod_sofia at 85.16.246.12:5061>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15180M
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Content-Type: message/sipfrag
   Content-Length: 27
   2009-10-22 17:41:04.779555 [DEBUG] switch_channel.c:182
sofia/internal/sip:1000 at 85.16.245.228:1024 receive message [AUDIO_SYNC]

   From:
   To: "unknown" 1000
   ------------------------------------------------------------------------
2009-10-22 17:41:04.779555 [DEBUG] switch_channel.c:182
sofia/internal/1001 at 85.16.246.12:5061 receive message [AUDIO_SYNC]
2009-10-22 17:41:04.779555 [DEBUG] switch_ivr_bridge.c:975
sofia/internal/sip:1000 at 85.16.245.228:1024 receive message [BRIDGE]
2009-10-22 17:41:04.779555 [DEBUG] switch_core_session.c:647 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-22 17:41:04.779555 [DEBUG] switch_ivr_bridge.c:982
sofia/internal/1001 at 85.16.246.12:5061 receive message [BRIDGE]
2009-10-22 17:41:04.779555 [DEBUG] switch_core_session.c:647 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-22 17:41:04.779555 [DEBUG] switch_ivr_bridge.c:1026
(sofia/internal/sip:1000 at 85.16.245.228:1024) State Change
CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2009-10-22 17:41:04.779555 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-22 17:41:04.779555 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/sip:1000 at 85.16.245.228:1024) Running State Change
CS_EXCHANGE_MEDIA
2009-10-22 17:41:04.780562 [DEBUG] switch_core_state_machine.c:343
(sofia/internal/sip:1000 at 85.16.245.228:1024) State EXCHANGE_MEDIA
2009-10-22 17:41:04.780562 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK
recv 405 bytes from udp/[85.16.245.206]:1024 at 15:41:04.819062:
   ------------------------------------------------------------------------
   ACK sip:1000 at 85.16.246.12:5061;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 85.16.245.206:1024;branch=z9hG4bK-ltlzbi82kr2t;rport
   From: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=7xpim4o1go
   To: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=ga0vUDXXSa8Kr
   Call-ID: 3c31304b7a80-no9xsnjj0bol
   CSeq: 1 ACK
   Max-Forwards: 70
   Contact: <sip:1001 at 85.16.245.206:1024;line=eg3wp69a>;reg-id=1
   Content-Length: 0

   ------------------------------------------------------------------------
2009-10-22 17:41:04.818545 [DEBUG] sofia.c:3493 Channel
sofia/internal/1001 at 85.16.246.12:5061 entering state [ready][200]
2009-10-22 17:41:04.830545 [DEBUG] switch_ivr_bridge.c:122
sofia/internal/sip:1000 at 85.16.245.228:1024 receive message [DISPLAY]
2009-10-22 17:41:04.840545 [DEBUG] switch_ivr_bridge.c:122
sofia/internal/1001 at 85.16.246.12:5061 receive message [DISPLAY]
send 670 bytes to udp/[85.16.245.206]:1024 at 15:41:04.840803:
   ------------------------------------------------------------------------
   INFO sip:1001 at 85.16.245.206:1024;line=eg3wp69a SIP/2.0
   Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKprXe7vy995a8B
   Max-Forwards: 70
   From: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=ga0vUDXXSa8Kr
   To: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=7xpim4o1go
   Call-ID: 3c31304b7a80-no9xsnjj0bol
   CSeq: 121994696 INFO
   Contact: <sip:1000 at 85.16.246.12:5061;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15180M
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Content-Type: message/sipfrag
   Content-Length: 24

   From:
   To: "hubu" 1000
   ------------------------------------------------------------------------
2009-10-22 17:41:04.860546 [DEBUG] switch_core_io.c:652
sofia/internal/sip:1000 at 85.16.245.228:1024 receive message
[TRANSCODING_NECESSARY]
2009-10-22 17:41:04.860546 [DEBUG] switch_core_io.c:891 Engaging Write
Buffer at 320 bytes to accommodate 640->320
2009-10-22 17:41:04.930547 [DEBUG] switch_core_io.c:652
sofia/internal/1001 at 85.16.246.12:5061 receive message
[TRANSCODING_NECESSARY]
2009-10-22 17:41:04.930547 [DEBUG] switch_core_io.c:891 Engaging Write
Buffer at 320 bytes to accommodate 320->320
2009-10-22 17:41:05.000550 [WARNING] mod_sofia.c:808 We were told to use
ptime 10 but what they meant to say was 20
This issue has so far been identified to happen on the following broken
platforms/devices:
Linksys/Sipura aka Cisco
ShoreTel
Sonus/L3
We will try to fix it but some of the devices on this list are so broken
who knows what will happen..
2009-10-22 17:41:05.000550 [DEBUG] sofia_glue.c:2101 Set Codec
sofia/internal/sip:1000 at 85.16.245.228:1024 G722/8000 20 ms 160 samples
2009-10-22 17:41:05.000550 [DEBUG] switch_rtp.c:1064 RE-Starting timer
[soft] 160 bytes per 20000ms
recv 362 bytes from udp/[85.16.245.206]:1024 at 15:41:05.019892:
   ------------------------------------------------------------------------
   SIP/2.0 200 Ok
   Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKprXe7vy995a8B
   From: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=ga0vUDXXSa8Kr
   To: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=7xpim4o1go
   Call-ID: 3c31304b7a80-no9xsnjj0bol
   CSeq: 121994696 INFO
   Contact: <sip:1001 at 85.16.245.206:1024;line=eg3wp69a>;reg-id=1
   Content-Length: 0

   ------------------------------------------------------------------------
recv 372 bytes from udp/[85.16.245.228]:1024 at 15:41:05.177982:
   ------------------------------------------------------------------------
   SIP/2.0 200 Ok
   Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKNF4N51D6cXmNg
   From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=HKSNX8D1pKy6K
   To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=sdeidvhpxx
   Call-ID: 251c312d-39c4-122d-3fbf-00144fe6907a
   CSeq: 121994696 INFO
   Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
   Content-Length: 0

   ------------------------------------------------------------------------
send 679 bytes to udp/[85.16.245.228]:1024 at 15:41:05.178268:
   ------------------------------------------------------------------------
   INFO sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
   Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKQ1p78QFD7e1tQ
   Max-Forwards: 70
   From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=HKSNX8D1pKy6K
   To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=sdeidvhpxx
   Call-ID: 251c312d-39c4-122d-3fbf-00144fe6907a
   CSeq: 121994697 INFO
   Contact: <sip:mod_sofia at 85.16.246.12:5061>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15180M
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Content-Type: message/sipfrag
   Content-Length: 32

   From:
   To: "1001 an PBX1" 1001
   ------------------------------------------------------------------------
recv 372 bytes from udp/[85.16.245.228]:1024 at 15:41:05.271275:
   ------------------------------------------------------------------------
   SIP/2.0 200 Ok
   Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKQ1p78QFD7e1tQ
   From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=HKSNX8D1pKy6K
   To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=sdeidvhpxx
   Call-ID: 251c312d-39c4-122d-3fbf-00144fe6907a
   CSeq: 121994697 INFO
   Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
   Content-Length: 0

   ------------------------------------------------------------------------
recv 589 bytes from udp/[85.16.245.228]:1024 at 15:41:06.336858:
   ------------------------------------------------------------------------
   BYE sip:mod_sofia at 85.16.246.12:5061 SIP/2.0
   Via: SIP/2.0/UDP 85.16.245.228:1024;branch=z9hG4bK-quadh3ddon5i;rport
   From: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=sdeidvhpxx
   To: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=HKSNX8D1pKy6K
   Call-ID: 251c312d-39c4-122d-3fbf-00144fe6907a
   CSeq: 1 BYE
   Max-Forwards: 70
   Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
   User-Agent: snom820/LAB_HD_HEAD_11_09_09_11_20
   RTP-RxStat:
Total_Rx_Pkts=0,Rx_Pkts=66,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0
   RTP-TxStat: Total_Tx_Pkts=67,Tx_Pkts=62,Remote_Tx_Pkts=0
   Content-Length: 0

   ------------------------------------------------------------------------
2009-10-22 17:41:06.336568 [NOTICE] sofia.c:328 Hangup
sofia/internal/sip:1000 at 85.16.245.228:1024 [CS_EXCHANGE_MEDIA]
[NORMAL_CLEARING]
2009-10-22 17:41:06.336568 [DEBUG] switch_channel.c:1896 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [KILL]
2009-10-22 17:41:06.336568 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-22 17:41:06.336568 [DEBUG] switch_core_state_machine.c:449
thread mismatch skipping state handler.
send 510 bytes to udp/[85.16.245.228]:1024 at 15:41:06.337536:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP
85.16.245.228:1024;branch=z9hG4bK-quadh3ddon5i;rport=1024
   From: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=sdeidvhpxx
   To: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=HKSNX8D1pKy6K
   Call-ID: 251c312d-39c4-122d-3fbf-00144fe6907a
   CSeq: 1 BYE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15180M
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Content-Length: 0

   ------------------------------------------------------------------------
2009-10-22 17:41:06.340568 [DEBUG] switch_ivr_bridge.c:453
sofia/internal/sip:1000 at 85.16.245.228:1024 ending bridge by request from
read function
2009-10-22 17:41:06.340568 [DEBUG] switch_ivr_bridge.c:447
sofia/internal/sip:1000 at 85.16.245.228:1024 ending bridge by request from
write function
2009-10-22 17:41:06.340568 [DEBUG] switch_ivr_bridge.c:528 BRIDGE THREAD
DONE [sofia/internal/sip:1000 at 85.16.245.228:1024]
2009-10-22 17:41:06.340568 [DEBUG] switch_ivr_bridge.c:530 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-22 17:41:06.340568 [DEBUG] switch_core_state_machine.c:343
(sofia/internal/sip:1000 at 85.16.245.228:1024) State EXCHANGE_MEDIA going
to sleep
2009-10-22 17:41:06.340568 [DEBUG] switch_ivr_bridge.c:502
sofia/internal/1001 at 85.16.246.12:5061 receive message [UNBRIDGE]
2009-10-22 17:41:06.340568 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/sip:1000 at 85.16.245.228:1024) Running State Change CS_HANGUP
2009-10-22 17:41:06.340568 [DEBUG] switch_core_session.c:647 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-22 17:41:06.340568 [DEBUG] switch_ivr_bridge.c:528 BRIDGE THREAD
DONE [sofia/internal/1001 at 85.16.246.12:5061]
2009-10-22 17:41:06.340568 [DEBUG] switch_ivr_bridge.c:530 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-22 17:41:06.340568 [NOTICE] switch_ivr_bridge.c:1102 Hangup
sofia/internal/1001 at 85.16.246.12:5061 [CS_EXECUTE] [NORMAL_CLEARING]
2009-10-22 17:41:06.340568 [DEBUG] switch_channel.c:1896 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [KILL]
2009-10-22 17:41:06.340568 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-22 17:41:06.340568 [DEBUG] switch_core_state_machine.c:476
(sofia/internal/sip:1000 at 85.16.245.228:1024) State HANGUP
2009-10-22 17:41:06.340568 [DEBUG] mod_sofia.c:344 Channel
sofia/internal/sip:1000 at 85.16.245.228:1024 hanging up, cause:
NORMAL_CLEARING
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:46
sofia/internal/sip:1000 at 85.16.245.228:1024 Standard HANGUP, cause:
NORMAL_CLEARING
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:476
(sofia/internal/sip:1000 at 85.16.245.228:1024) State HANGUP going to sleep
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/sip:1000 at 85.16.245.228:1024) State Change CS_HANGUP ->
CS_REPORTING
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:476
(sofia/internal/1001 at 85.16.246.12:5061) State HANGUP
2009-10-22 17:41:06.341609 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-22 17:41:06.341609 [DEBUG] mod_sofia.c:344 Channel
sofia/internal/1001 at 85.16.246.12:5061 hanging up, cause: NORMAL_CLEARING
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/sip:1000 at 85.16.245.228:1024) Running State Change
CS_REPORTING
2009-10-22 17:41:06.341609 [DEBUG] mod_sofia.c:382 Sending BYE to
sofia/internal/1001 at 85.16.246.12:5061
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:567
(sofia/internal/sip:1000 at 85.16.245.228:1024) State REPORTING
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:53
sofia/internal/sip:1000 at 85.16.245.228:1024 Standard REPORTING, cause:
NORMAL_CLEARING
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:567
(sofia/internal/sip:1000 at 85.16.245.228:1024) State REPORTING going to sleep
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:319
(sofia/internal/sip:1000 at 85.16.245.228:1024) State Change CS_REPORTING
- -> CS_DESTROY
2009-10-22 17:41:06.341609 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-22 17:41:06.341609 [DEBUG] switch_core_session.c:1114 Session 42
(sofia/internal/sip:1000 at 85.16.245.228:1024) Locked, Waiting on external
entities
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:46
sofia/internal/1001 at 85.16.246.12:5061 Standard HANGUP, cause:
NORMAL_CLEARING
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:476
(sofia/internal/1001 at 85.16.246.12:5061) State HANGUP going to sleep
2009-10-22 17:41:06.341609 [NOTICE] switch_core_session.c:1132 Session
42 (sofia/internal/sip:1000 at 85.16.245.228:1024) Ended
2009-10-22 17:41:06.341609 [NOTICE] switch_core_session.c:1134 Close
Channel sofia/internal/sip:1000 at 85.16.245.228:1024 [CS_DESTROY]
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:413
(sofia/internal/sip:1000 at 85.16.245.228:1024) Running State Change CS_DESTROY
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:424
(sofia/internal/sip:1000 at 85.16.245.228:1024) State DESTROY
2009-10-22 17:41:06.341609 [DEBUG] mod_sofia.c:261
sofia/internal/sip:1000 at 85.16.245.228:1024 SOFIA DESTROY
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:340
(sofia/internal/1001 at 85.16.246.12:5061) State EXECUTE going to sleep
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_HANGUP
2009-10-22 17:41:06.342645 [DEBUG] switch_core_state_machine.c:454
handler already called, skipping state handler.
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:60
sofia/internal/sip:1000 at 85.16.245.228:1024 Standard DESTROY
2009-10-22 17:41:06.342645 [DEBUG] switch_core_state_machine.c:424
(sofia/internal/sip:1000 at 85.16.245.228:1024) State DESTROY going to sleep
send 659 bytes to udp/[85.16.245.206]:1024 at 15:41:06.342919:
   ------------------------------------------------------------------------
   BYE sip:1001 at 85.16.245.206:1024;line=eg3wp69a SIP/2.02009-10-22
17:41:06.342645 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/1001 at 85.16.246.12:5061) State Change CS_HANGUP ->
CS_REPORTING
2009-10-22 17:41:06.342645 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-22 17:41:06.342645 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_REPORTING
2009-10-22 17:41:06.342645 [DEBUG] switch_core_state_machine.c:567
(sofia/internal/1001 at 85.16.246.12:5061) State REPORTING

   Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKrag0aK0g4QQDK
   Max-Forwards: 70
   From: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=ga0vUDXXSa8Kr
   To: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=7xpim4o1go
   Call-ID: 3c31304b7a80-no9xsnjj0bol
   CSeq: 121994697 BYE
   Contact: <sip:1000 at 85.16.246.12:5061;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15180M
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Reason: Q.850;cause=16;text="NORMAL_CLEARING"
   Content-Length: 0

   ------------------------------------------------------------------------
2009-10-22 17:41:06.343570 [DEBUG] switch_core_state_machine.c:53
sofia/internal/1001 at 85.16.246.12:5061 Standard REPORTING, cause:
NORMAL_CLEARING
2009-10-22 17:41:06.343570 [DEBUG] switch_core_state_machine.c:567
(sofia/internal/1001 at 85.16.246.12:5061) State REPORTING going to sleep
2009-10-22 17:41:06.343570 [DEBUG] switch_core_state_machine.c:319
(sofia/internal/1001 at 85.16.246.12:5061) State Change CS_REPORTING ->
CS_DESTROY
2009-10-22 17:41:06.343570 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-22 17:41:06.343570 [DEBUG] switch_core_session.c:1114 Session 41
(sofia/internal/1001 at 85.16.246.12:5061) Locked, Waiting on external entities
2009-10-22 17:41:06.343570 [NOTICE] switch_core_session.c:1132 Session
41 (sofia/internal/1001 at 85.16.246.12:5061) Ended
2009-10-22 17:41:06.343570 [NOTICE] switch_core_session.c:1134 Close
Channel sofia/internal/1001 at 85.16.246.12:5061 [CS_DESTROY]
2009-10-22 17:41:06.343570 [DEBUG] switch_core_state_machine.c:413
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_DESTROY
2009-10-22 17:41:06.343570 [DEBUG] switch_core_state_machine.c:424
(sofia/internal/1001 at 85.16.246.12:5061) State DESTROY
2009-10-22 17:41:06.343570 [DEBUG] mod_sofia.c:261
sofia/internal/1001 at 85.16.246.12:5061 SOFIA DESTROY
2009-10-22 17:41:06.343570 [DEBUG] switch_core_state_machine.c:60
sofia/internal/1001 at 85.16.246.12:5061 Standard DESTROY
2009-10-22 17:41:06.343570 [DEBUG] switch_core_state_machine.c:424
(sofia/internal/1001 at 85.16.246.12:5061) State DESTROY going to sleep
recv 524 bytes from udp/[85.16.245.206]:1024 at 15:41:06.357848:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKrag0aK0g4QQDK
   From: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=ga0vUDXXSa8Kr
   To: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=7xpim4o1go
   Call-ID: 3c31304b7a80-no9xsnjj0bol
   CSeq: 121994697 BYE
   Contact: <sip:1001 at 85.16.245.206:1024;line=eg3wp69a>;reg-id=1
   User-Agent: snom820/8.2.16
   RTP-RxStat:
Total_Rx_Pkts=61,Rx_Pkts=61,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0
   RTP-TxStat: Total_Tx_Pkts=0,Tx_Pkts=90,Remote_Tx_Pkts=0
   Content-Length: 0

   ------------------------------------------------------------------------









On 21.10.2009 19:48, Michael Collins wrote:
> 
> Okay, you are able to reproduce that "unknown" thing? Can you pastebin a
> fresh debug log w/ SIP trace on, plus and relevant dp changes from the
> default dialplan?
> Thanks,
> MC
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)

iD8DBQFK4H3z4tZeNddg3dwRAtR3AJ4xUr8Tp4ctmhQn2DIQSvrBSv59QQCeLSsH
5inLJYyajrsYT2+kXcgd22w=
=5B1x
-----END PGP SIGNATURE-----


From tculjaga at gmail.com  Thu Oct 22 08:58:24 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 22 Oct 2009 17:58:24 +0200
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
In-Reply-To: <2d9149cd0910220844v2aa73502j4b88fdbd75a0d833@mail.gmail.com>
References: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>
	<191c3a030910201434q4c6e65e1h58c5f0a82addc2d3@mail.gmail.com>
	<2d9149cd0910202050i36a7ea82nc7253d87e38f990a@mail.gmail.com>
	<191c3a030910211154gc8412b5gd999fbedffa02b53@mail.gmail.com>
	<2d9149cd0910220844v2aa73502j4b88fdbd75a0d833@mail.gmail.com>
Message-ID: <65d96fc80910220858u788c6146sced38e19418e681a@mail.gmail.com>

On Thu, Oct 22, 2009 at 5:44 PM, Kristian Kielhofner <
kristian.kielhofner at gmail.com> wrote:

> An update for Tony, Brian, Mike, and everyone on the list...
>
> I was able to get some phone time with the team yesterday.  Tony
> worked on my machine, found the issue, and had it committed within 30
> minutes.
>
> I've been testing T.38 all morning between the fax machines in the
> office with few issues.
>
>
and what these few issues are? :P



> THANKS AGAIN GUYS!
>
>
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From brian at freeswitch.org  Thu Oct 22 09:23:27 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 22 Oct 2009 11:23:27 -0500
Subject: [Freeswitch-users] Domain Question
In-Reply-To: <7f783b530910220842q5c01ddbcj28a6157c30badd52@mail.gmail.com>
References: <7f783b530910220842q5c01ddbcj28a6157c30badd52@mail.gmail.com>
Message-ID: <57CAD19F-F048-4491-83AB-A0A07D9A989E@freeswitch.org>

I"m going to guess its because mydomain.localhost doesn't resolve  
outside the machine itself so the softphone never ends up knowing wtf  
to do.

/b

On Oct 22, 2009, at 10:42 AM, freeswitch noob wrote:

> I have a noobish question about setting up FS.
>
> I have it installed and running.
>
> I setup a soft client on the machine fs is on and point it to the ip  
> address of the FS instance and it registers with no issues.
>
> I then setup an entry in my etc/hosts files "mydomain.localhost" and  
> changed the domain in the soft client.
>
> The registration now fails. with the soft phone giving a 503 error  
> "service unavailable"
>
> And I am watching the cli and I see no errors come through with the  
> domain but I see it register with the IP :(
>
> I change it back to the ip address and it works.
>
> Any direction you guys could point me in would be great.



From brian at freeswitch.org  Thu Oct 22 09:25:22 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 22 Oct 2009 11:25:22 -0500
Subject: [Freeswitch-users] Qustion about INFO messages after
	Connect/Answer
In-Reply-To: <4AE07DF4.5040803@ewetel.de>
References: <4AD859A4.5000101@ewetel.de>	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>	<4ADC7C09.6070503@ewetel.de>	<191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>	<4ADC85C0.7060908@ewetel.de>
	<4ADC8B6E.7070404@ewetel.de>	<191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>	<87f2f3b90910191435s7528e256r85df5f12fc634f24@mail.gmail.com>	<4ADF483B.4050807@ewetel.de>
	<87f2f3b90910211048s208daf1eyfae501091ab75530@mail.gmail.com>
	<4AE07DF4.5040803@ewetel.de>
Message-ID: <085B77EE-DA3F-4ADD-86DC-E59D1B6553EA@freeswitch.org>

I can't get what exactly you re talking about. Can you clarify ? Also  
please include the packets of interest only not the full trace if its  
not relevant to the bug.

/b

On Oct 22, 2009, at 10:44 AM, Helmut Kuper wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi Mike,
>
> here it is:
>
>
> Dialplan:
>
>
>    <extension name="Local_Extension">
>      <condition field="destination_number" expression="^(10[01][0-9]) 
> $">
>        <action application="set" data="dialed_extension=$1"/>
>        <action application="export" data="dialed_extension=$1"/>
>        <action application="set" data="transfer_ringback=$$ 
> {hold_music}"/>
>        <action application="set" data="hangup_after_bridge=true"/>
>        <action application="bridge"
> data="user/${dialed_extension}@${domain_name}"/>
>      </condition>
>    </extension>
>



From pippyduck1127 at hotmail.com  Thu Oct 22 09:30:35 2009
From: pippyduck1127 at hotmail.com (Pajongjit Buntaokit)
Date: Thu, 22 Oct 2009 16:30:35 +0000
Subject: [Freeswitch-users] Address Rupa: Database for Audio Data
In-Reply-To: <b39273530910130857jb7dec2eidd71e950d3902b25@mail.gmail.com>
References: <BAY119-W52BD912544FAA1DE624B88B3C70@phx.gbl>
Message-ID: <BAY119-W4307982F0031E8BE965E54B3BE0@phx.gbl>


Hi Rupa,

Thanks again for your advice.

I have been searching for the method to record in the freeswitch documentation but I'm still not sure which command and method to make the record for every call automatically.

Which command or method do you use? 
And to make the recording start and stop automatically every time when the calls is started and end,  where should I insert this command? 

Did you use the Mod commands 'uuid_record'?
If so, where to place this commands? 
Please show me some clues?

Thank you very much!

PB
 



> Date: Tue, 13 Oct 2009 09:57:05 -0600
> From: rupa at rupa.comajongjit Buntaokit
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Database for Audio Data
> 
> What I do is record all calls and store the call with the UUID as the
> filename.  Then when the call is hung up a CDR entry is sent to my web
> server.  This CDR contains callerid and other info I might want to
> query by.  The service on the web server inserts appropriate record(s)
> into the database.  The recordings are available to the webserver.
> When one clicks on the "listen" link, the web server serves up the
> recording by UUID in the recording directory.  I have a process that
> periodically removes old recordings from that dir.  I don't purge the
> CDRs, though that is certainly possible.
> 
> On Tue, Oct 13, 2009 at 7:58 AM, Pajongjit Buntaokit
> <pippyduck1127 at hotmail.com> wrote:
> > Hi,
> >
> > Does anyone know whether FreeSWITCH has a function to automatically record
> > every call as an audio file in a server
> > or forward them to be stored in a database with additional parameters such
> > as caller ID, date, starting time and ending time?
> >
> > So that these recorded audio data can be queried and retrieved with the
> > caller ID, date and time.
> >
> > Any suggestion or guidance, please advise.
> >
> > Thank you very much!
> >
> > ________________________________
> > Hotmail: Trusted email with Microsoft?s powerful SPAM protection. Sign up
> > now.
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> 
> 
> -- 
> -Rupa
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
_________________________________________________________________
Windows 7: It works the way you want. Learn more.
http://www.microsoft.com/Windows/windows-7/default.aspx?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_evergreen2:102009
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From anthony.minessale at gmail.com  Thu Oct 22 10:08:00 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 22 Oct 2009 12:08:00 -0500
Subject: [Freeswitch-users] Qustion about INFO messages after
	Connect/Answer
In-Reply-To: <085B77EE-DA3F-4ADD-86DC-E59D1B6553EA@freeswitch.org>
References: <4AD859A4.5000101@ewetel.de>
	<191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>
	<4ADC85C0.7060908@ewetel.de> <4ADC8B6E.7070404@ewetel.de>
	<191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>
	<87f2f3b90910191435s7528e256r85df5f12fc634f24@mail.gmail.com>
	<4ADF483B.4050807@ewetel.de>
	<87f2f3b90910211048s208daf1eyfae501091ab75530@mail.gmail.com>
	<4AE07DF4.5040803@ewetel.de>
	<085B77EE-DA3F-4ADD-86DC-E59D1B6553EA@freeswitch.org>
Message-ID: <191c3a030910221008n60341314hc7a7e6a55a50ef15@mail.gmail.com>

I'm sure that problem is gone in svn trunk.


On Thu, Oct 22, 2009 at 11:25 AM, Brian West <brian at freeswitch.org> wrote:

> I can't get what exactly you re talking about. Can you clarify ? Also
> please include the packets of interest only not the full trace if its
> not relevant to the bug.
>
> /b
>
> On Oct 22, 2009, at 10:44 AM, Helmut Kuper wrote:
>
> > -----BEGIN PGP SIGNED MESSAGE-----
> > Hash: SHA1
> >
> > Hi Mike,
> >
> > here it is:
> >
> >
> > Dialplan:
> >
> >
> >    <extension name="Local_Extension">
> >      <condition field="destination_number" expression="^(10[01][0-9])
> > $">
> >        <action application="set" data="dialed_extension=$1"/>
> >        <action application="export" data="dialed_extension=$1"/>
> >        <action application="set" data="transfer_ringback=$$
> > {hold_music}"/>
> >        <action application="set" data="hangup_after_bridge=true"/>
> >        <action application="bridge"
> > data="user/${dialed_extension}@${domain_name}"/>
> >      </condition>
> >    </extension>
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Thu Oct 22 10:12:53 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 22 Oct 2009 12:12:53 -0500
Subject: [Freeswitch-users] UUID of the newly originated call?
In-Reply-To: <26006565.post@talk.nabble.com>
References: <fe67a3340910202306i55a0b9ceve92ab19e000edf11@mail.gmail.com>
	<191c3a030910211720l26bd0303i2cb11e9adb28037d@mail.gmail.com>
	<26006565.post@talk.nabble.com>
Message-ID: <191c3a030910221012w125a15d4m24f5fbe72c8a859d@mail.gmail.com>

please update to svn trunk with "make current" and try again.


On Thu, Oct 22, 2009 at 4:08 AM, Nagalenoj <nagalenoj at gmail.com> wrote:

>
> I'm using 'FreeSWITCH Version 1.0.trunk (15106)'.
>
>
> Anthony Minessale-2 wrote:
> >
> > which revision of FS are you using?
> >
> > On Wed, Oct 21, 2009 at 1:06 AM, Nagalenoj H. <nagalenoj at gmail.com>
> wrote:
> >
> >> I've tried with origination_uuid.
> >>
> >> First, I tried with SIP and my program executes successfully as what I
> >> expected. This program initiates a new call when a call comes and let
> the
> >> new call to eavesdrop the landed call.
> >> When, I experimented with PRI(openzap), I'm facing the following error.
> >> And, I am unable to make any calls(even from CLI). It is reporting the
> >> same
> >> error for the subsequent calls.
> >> Error:        2009-10-21 11:08:31.661252 [ERR] mod_openzap.c:1760
> Session
> >> for channel 1:1 not found [UUID: 8364eb30-be03-11de-aa07-07e76b7651a9]
> >>
> >> Full log is in http://pastebin.freeswitch.org/10780
> >> My script is here, http://pastebin.freeswitch.org/10781
> >>
> >> What is this error for and how to avoid this?
> >>
> >> Is there any other way to get the uuid of the originated call except
> >> explicitly defining(origination_uuid).?!
> >>
> >> --
> >> Regards,
> >> Nagalenoj H.
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com><
> MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org><
> sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> > pstn:213-799-1400
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> --
> View this message in context:
> http://www.nabble.com/Re%3A-UUID-of-the-newly-originated-call--tp25987024p26006565.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mattdfong at gmail.com  Thu Oct 22 10:42:29 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Fri, 23 Oct 2009 00:42:29 +0700
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
In-Reply-To: <87f2f3b90910191144q25750bayfd7a7448e82e78ff@mail.gmail.com>
References: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
	<177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>
	<4256bf830908041151l55980e3fifb85c75b87535426@mail.gmail.com>
	<3D0B855A-5B2C-4C6F-9306-64B0ECE97091@jerris.com>
	<4256bf830910180725y72539d98gb138934afc18d58b@mail.gmail.com>
	<A5ECDCD9-1A12-4A31-89F0-E6F4C79FE9E9@jerris.com>
	<4256bf830910182148mf7f2bf9vdcccd687e0caa1e7@mail.gmail.com>
	<87f2f3b90910191144q25750bayfd7a7448e82e78ff@mail.gmail.com>
Message-ID: <4256bf830910221042l68ea769cg532bbd339bae0790@mail.gmail.com>

I figured out how to turn debug mode on. For some reason F8 didn't work, so
I just set the default debug mode to DEBUG in switch.conf.xml

http://pastebin.freeswitch.org/10805
<http://pastebin.freeswitch.org/10805>So my program does more or less the
same sequence for an Agent every time when connected to another party. They
uuid_kill the bridged parties uuid, and then are transfered to an extension
that park's the agent (using the transfer_after_bridge variable -- in my
dialplan extn 1998). In some instances the agent is then transfered to
another extension, played a wav file, and then park'ed again

Sometimes, FreeSWITCH unexpectidly hangsup the Agent channel citing an
EXCHANGE_ROUTING_ERROR. It seems (to me to happen at random times)...and
it's more prevelent with some did providers than others. With
didforsale.orgit happens about one ever 5 bridges. With
icall.com it only happens about once every 30 bridges. With IPKall it
happens rarely. I can't figure it out, and it's driving me crazy :)


A GOOD UUID_KILL SEQUENCE:
2009-10-22 17:22:31.741845 [DEBUG] switch_ivr_play_say.c:1432 done playing
file
2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:340
(sofia/external/+14158867717 at 199.173.100.16:5060) State EXECUTE going to
sleep
2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:306
(sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
CS_HIBERNATE
2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:355
(sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE
2009-10-22 17:22:31.741845 [DEBUG] mod_sofia.c:160 sofia/external/+
14158867717 at 199.173.100.16:5060 SOFIA HIBERNATE
2009-10-22 17:22:31.741845 [DEBUG] switch_ivr_bridge.c:650 (sofia/external/+
14158867717 at 199.173.100.16:5060) State Change CS_HIBERNATE -> CS_RESET
2009-10-22 17:22:31.741845 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:355
(sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE going to
sleep
2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:306
(sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
CS_RESET
2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:336
(sofia/external/+14158867717 at 199.173.100.16:5060) State RESET
2009-10-22 17:22:31.741845 [DEBUG] mod_sofia.c:145 sofia/external/+
14158867717 at 199.173.100.16:5060 SOFIA RESET
2009-10-22 17:22:31.741845 [DEBUG] switch_ivr_bridge.c:637 sofia/external/+
14158867717 at 199.173.100.16:5060 CUSTOM RESET
2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:67
sofia/external/+14158867717 at 199.173.100.16:5060 Standard RESET
2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:336
(sofia/external/+14158867717 at 199.173.100.16:5060) State RESET going to sleep
2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:340
(sofia/external/14159927717) State EXECUTE going to sleep
2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:306
(sofia/external/14159927717) Running State Change CS_HIBERNATE
2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:355
(sofia/external/14159927717) State HIBERNATE
2009-10-22 17:22:31.748086 [DEBUG] mod_sofia.c:160
sofia/external/14159927717 SOFIA HIBERNATE
2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:650
(sofia/external/14159927717) State Change CS_HIBERNATE -> CS_RESET
2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/14159927717 [BREAK]
2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:355
(sofia/external/14159927717) State HIBERNATE going to sleep
2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:306
(sofia/external/14159927717) Running State Change CS_RESET
2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:336
(sofia/external/14159927717) State RESET
2009-10-22 17:22:31.748086 [DEBUG] mod_sofia.c:145
sofia/external/14159927717 SOFIA RESET
2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:637
sofia/external/14159927717 CUSTOM RESET
2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:642
(sofia/external/14159927717) State Change CS_RESET -> CS_SOFT_EXECUTE
2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/14159927717 [BREAK]
2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:336
(sofia/external/14159927717) State RESET going to sleep
2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:306
(sofia/external/14159927717) Running State Change CS_SOFT_EXECUTE
2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:346
(sofia/external/14159927717) State SOFT_EXECUTE
2009-10-22 17:22:31.748086 [DEBUG] mod_sofia.c:442 SOFIA TRANSMIT
2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:660
sofia/external/14159927717 CUSTOM SOFT_EXECUTE
2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_originate.c:674
(sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_RESET ->
CS_SOFT_EXECUTE
2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:120 sofia/external/+
14158867717 at 199.173.100.16:5060 receive message [DISPLAY]
2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:120
sofia/external/14159927717 receive message [DISPLAY]
2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:924 (sofia/external/+
14158867717 at 199.173.100.16:5060) State Change CS_SOFT_EXECUTE ->
CS_CONSUME_MEDIA
2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:968 sofia/external/+
14158867717 at 199.173.100.16:5060 receive message [BRIDGE]
2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:630 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:975
sofia/external/14159927717 receive message [BRIDGE]
2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:630 Send signal
sofia/external/14159927717 [BREAK]
2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:1019
(sofia/external/+14158867717 at 199.173.100.16:5060) State Change
CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:22:31.749553 [DEBUG] switch_core_state_machine.c:306
(sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
CS_EXCHANGE_MEDIA
2009-10-22 17:22:31.749553 [DEBUG] switch_core_state_machine.c:343
(sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
2009-10-22 17:22:31.749553 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK
multipler is being overrided to1
2009-10-22 17:22:34.300662 [INFO] switch_cpp.cpp:1116 BG_AG_GEN_CALLS,
Agents total:1 avail:0 OutChan:0 AR:0 +-Calls:0:Q Call NYNoc
multipler is being overrided to1
2009-10-22 17:22:39.303041 [INFO] switch_cpp.cpp:1116 BG_AG_GEN_CALLS,
Agents total:1 avail:0 OutChan:0 AR:0 +-Calls:0:Q Call NYNoc
2009-10-22 17:22:40.626379 [NOTICE] sofia.c:328 Hangup
sofia/external/14159927717 [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
2009-10-22 17:22:40.626379 [DEBUG] switch_channel.c:1810 Send signal
sofia/external/14159927717 [KILL]
2009-10-22 17:22:40.626379 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/14159927717 [BREAK]
active_true=1
--contact NOT in playmsg or conference, update AGENT
2009-10-22 17:22:40.626379 [DEBUG] switch_core_state_machine.c:437 thread
mismatch skipping state handler.
2009-10-22 17:22:40.626379 [INFO] switch_cpp.cpp:1116 EVENTC_HANGUP_CONTACT,
Q_CONTACT fired for agent:1/1 for qcall:1 contact:4801442/4
2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:440
sofia/external/14159927717 ending bridge by request from write function
2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:495 sofia/external/+
14158867717 at 199.173.100.16:5060 receive message [UNBRIDGE]
2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:630 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
DONE [sofia/external/+14158867717 at 199.173.100.16:5060]
2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:523 Send signal
sofia/external/14159927717 [BREAK]
2009-10-22 17:22:40.639760 [DEBUG] switch_ivr.c:1361 (sofia/external/+
14158867717 at 199.173.100.16:5060) State Change CS_EXCHANGE_MEDIA ->
CS_ROUTING
2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:22:40.639760 [DEBUG] switch_ivr.c:1365 sofia/external/+
14158867717 at 199.173.100.16:5060 receive message [TRANSFER]
2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:630 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:22:40.639760 [NOTICE] switch_ivr.c:1367 Transfer
sofia/external/+14158867717 at 199.173.100.16:5060 to xml[1998 at default]
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:343
(sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA going
to sleep
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
(sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
CS_ROUTING
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:333
(sofia/external/+14158867717 at 199.173.100.16:5060) State ROUTING
2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:130 sofia/external/+
14158867717 at 199.173.100.16:5060 SOFIA ROUTING
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:78
sofia/external/+14158867717 at 199.173.100.16:5060 Standard ROUTING
2009-10-22 17:22:40.639760 [INFO] mod_dialplan_xml.c:391 Processing
+14158867717->1998 in context default
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->tod_example] continue=true
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (PASS)
[tod_example] ${strftime(%w)}(4) =~ /^([1-5])$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (PASS)
[tod_example] ${strftime(%H%M)}(1722) =~ /^((09|1[0-7])[0-5][0-9]|1800)$/
break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Action
set(open=true)
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->ivr_demo] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[ivr_demo] destination_number(1998) =~ /^5000$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1975] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1975] destination_number(1998) =~ /^1975$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1971] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1971] destination_number(1998) =~ /^1971$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1972] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1972] destination_number(1998) =~ /^1972$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1920] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1920] destination_number(1998) =~ /^1920$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1981] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1981] destination_number(1998) =~ /^1981$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1982] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1982] destination_number(1998) =~ /^1982$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1481] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1481] destination_number(1998) =~ /^1481$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1989] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1989] destination_number(1998) =~ /^1989$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1990] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1990] destination_number(1998) =~ /^1990$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1991] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1991] destination_number(1998) =~ /^1991$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1992] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1992] destination_number(1998) =~ /^1992$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1993] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1993] destination_number(1998) =~ /^1993$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1994] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1994] destination_number(1998) =~ /^1994$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1995] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1995] destination_number(1998) =~ /^1995$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1996] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1996] destination_number(1998) =~ /^1996$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1997] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1997] destination_number(1998) =~ /^1997$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1998] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (PASS)
[1998] destination_number(1998) =~ /^1998$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Action
set(transfer_after_bridge=1998)
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Action park()
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:114
(sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_ROUTING ->
CS_EXECUTE
2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:333
(sofia/external/+14158867717 at 199.173.100.16:5060) State ROUTING going to
sleep
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
(sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
CS_EXECUTE
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:340
(sofia/external/+14158867717 at 199.173.100.16:5060) State EXECUTE
2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:173 sofia/external/+
14158867717 at 199.173.100.16:5060 SOFIA EXECUTE
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:151
sofia/external/+14158867717 at 199.173.100.16:5060 Standard EXECUTE
EXECUTE sofia/external/+14158867717 at 199.173.100.16:5060 set(open=true)
2009-10-22 17:22:40.639760 [DEBUG] mod_dptools.c:752 sofia/external/+
14158867717 at 199.173.100.16:5060 SET [open]=[true]
EXECUTE sofia/external/+14158867717 at 199.173.100.16:5060set(transfer_after_bridge=1998)
2009-10-22 17:22:40.639760 [DEBUG] mod_dptools.c:752 sofia/external/+
14158867717 at 199.173.100.16:5060 SET [transfer_after_bridge]=[1998]
EXECUTE sofia/external/+14158867717 at 199.173.100.16:5060 park()
2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:446
sofia/external/14159927717 ending bridge by request from read function
2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
DONE [sofia/external/14159927717]
2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:523 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
subtracting 9 seconds for hh_call: 1095422
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:346
(sofia/external/14159927717) State SOFT_EXECUTE going to sleep
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
(sofia/external/14159927717) Running State Change CS_HANGUP
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:464
(sofia/external/14159927717) State HANGUP
2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:344 Channel
sofia/external/14159927717 hanging up, cause: NORMAL_CLEARING
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:46
sofia/external/14159927717 Standard HANGUP, cause: NORMAL_CLEARING
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:464
(sofia/external/14159927717) State HANGUP going to sleep
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:325
(sofia/external/14159927717) State Change CS_HANGUP -> CS_REPORTING
2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/14159927717 [BREAK]
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
(sofia/external/14159927717) Running State Change CS_REPORTING
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:555
(sofia/external/14159927717) State REPORTING
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:53
sofia/external/14159927717 Standard REPORTING, cause: NORMAL_CLEARING
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:555
(sofia/external/14159927717) State REPORTING going to sleep
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:319
(sofia/external/14159927717) State Change CS_REPORTING -> CS_DESTROY
2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/14159927717 [BREAK]
2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:1069 Session 6
(sofia/external/14159927717) Locked, Waiting on external entities
2009-10-22 17:22:40.639760 [NOTICE] switch_core_session.c:1087 Session 6
(sofia/external/14159927717) Ended
2009-10-22 17:22:40.639760 [NOTICE] switch_core_session.c:1089 Close Channel
sofia/external/14159927717 [CS_DESTROY]
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:401
(sofia/external/14159927717) Running State Change CS_DESTROY
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:412
(sofia/external/14159927717) State DESTROY
2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:261
sofia/external/14159927717 SOFIA DESTROY
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:60
sofia/external/14159927717 Standard DESTROY
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:412
(sofia/external/14159927717) State DESTROY going to sleep
2009-10-22 17:22:40.639760 [INFO] switch_cpp.cpp:1116 PCHangup
gw:teliax.comhc:NORMAL_CLEARING du:9 cn:sofia/external/14159927717



A BAD UUID_KILL SEQUENCE: (where the agent is hungup, unexpectedly)

--bridged in: 0.005821 seconds
2009-10-22 17:23:26.718930 [INFO] switch_cpp.cpp:1116 AG_CBRIDGE, connected
agent:1/1 contact:4801439/22/1/1
2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_play_say.c:1432 done playing
file
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:340
(sofia/external/+14158867717 at 199.173.100.16:5060) State EXECUTE going to
sleep
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
(sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
CS_HIBERNATE
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
(sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE
2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:160 sofia/external/+
14158867717 at 199.173.100.16:5060 SOFIA HIBERNATE
2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:650 (sofia/external/+
14158867717 at 199.173.100.16:5060) State Change CS_HIBERNATE -> CS_RESET
2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
(sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE going to
sleep
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
(sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
CS_RESET
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
(sofia/external/+14158867717 at 199.173.100.16:5060) State RESET
2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:145 sofia/external/+
14158867717 at 199.173.100.16:5060 SOFIA RESET
2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:637 sofia/external/+
14158867717 at 199.173.100.16:5060 CUSTOM RESET
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:67
sofia/external/+14158867717 at 199.173.100.16:5060 Standard RESET
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
(sofia/external/+14158867717 at 199.173.100.16:5060) State RESET going to sleep
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:340
(sofia/external/14159927717) State EXECUTE going to sleep
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
(sofia/external/14159927717) Running State Change CS_HIBERNATE
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
(sofia/external/14159927717) State HIBERNATE
2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:160
sofia/external/14159927717 SOFIA HIBERNATE
2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:650
(sofia/external/14159927717) State Change CS_HIBERNATE -> CS_RESET
2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/14159927717 [BREAK]
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
(sofia/external/14159927717) State HIBERNATE going to sleep
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
(sofia/external/14159927717) Running State Change CS_RESET
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
(sofia/external/14159927717) State RESET
2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:145
sofia/external/14159927717 SOFIA RESET
2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:637
sofia/external/14159927717 CUSTOM RESET
2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:642
(sofia/external/14159927717) State Change CS_RESET -> CS_SOFT_EXECUTE
2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/14159927717 [BREAK]
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
(sofia/external/14159927717) State RESET going to sleep
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
(sofia/external/14159927717) Running State Change CS_SOFT_EXECUTE
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:346
(sofia/external/14159927717) State SOFT_EXECUTE
2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:442 SOFIA TRANSMIT
2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:660
sofia/external/14159927717 CUSTOM SOFT_EXECUTE
2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_originate.c:674
(sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_RESET ->
CS_SOFT_EXECUTE
2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:120 sofia/external/+
14158867717 at 199.173.100.16:5060 receive message [DISPLAY]
2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:120
sofia/external/14159927717 receive message [DISPLAY]
2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:924 (sofia/external/+
14158867717 at 199.173.100.16:5060) State Change CS_SOFT_EXECUTE ->
CS_CONSUME_MEDIA
2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:968 sofia/external/+
14158867717 at 199.173.100.16:5060 receive message [BRIDGE]
2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:630 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:975
sofia/external/14159927717 receive message [BRIDGE]
2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:630 Send signal
sofia/external/14159927717 [BREAK]
2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:1019
(sofia/external/+14158867717 at 199.173.100.16:5060) State Change
CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
(sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
CS_EXCHANGE_MEDIA
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:343
(sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK
2009-10-22 17:23:26.740065 [DEBUG] switch_core_io.c:234 sofia/external/+
14158867717 at 199.173.100.16:5060 receive message [TRANSCODING_NECESSARY]
2009-10-22 17:23:27.319618 [DEBUG] switch_core_media_bug.c:393 Removing BUG
from sofia/external/+14158867717 at 199.173.100.16:5060
multipler is being overrided to1
2009-10-22 17:23:29.406414 [INFO] switch_cpp.cpp:1116 BG_AG_GEN_CALLS,
Agents total:1 avail:0 OutChan:0 AR:0 +-Calls:0:Q Call NYNoc
2009-10-22 17:23:31.873352 [NOTICE] mod_commands.c:1523 Hangup
sofia/external/14159927717 [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
2009-10-22 17:23:31.874411 [INFO] switch_cpp.cpp:1116 EVENTC_HANGUP_CONTACT,
Q_CONTACT fired for agent:1/1 for qcall:1 contact:4801439/1
active_true=1
--contact NOT in playmsg or conference, update AGENT
2009-10-22 17:23:31.875475 [DEBUG] switch_channel.c:1810 Send signal
sofia/external/14159927717 [KILL]
2009-10-22 17:23:31.875475 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/14159927717 [BREAK]
2009-10-22 17:23:31.875475 [DEBUG] switch_core_state_machine.c:437 thread
mismatch skipping state handler.
2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:446
sofia/external/14159927717 ending bridge by request from read function
2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
DONE [sofia/external/14159927717]
2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:523 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:495 sofia/external/+
14158867717 at 199.173.100.16:5060 receive message [UNBRIDGE]
2009-10-22 17:23:31.882153 [DEBUG] switch_core_session.c:630 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
DONE [sofia/external/+14158867717 at 199.173.100.16:5060]
2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:523 Send signal
sofia/external/14159927717 [BREAK]
2009-10-22 17:23:31.882153 [NOTICE] switch_ivr.c:1268 Hangup
sofia/external/+14158867717 at 199.173.100.16:5060 [CS_EXCHANGE_MEDIA]
[EXCHANGE_ROUTING_ERROR]
2009-10-22 17:23:31.882153 [DEBUG] switch_channel.c:1810 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [KILL]
2009-10-22 17:23:31.882153 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:23:31.882153 [DEBUG] switch_core_state_machine.c:464
(sofia/external/+14158867717 at 199.173.100.16:5060) State HANGUP
2009-10-22 17:23:31.882153 [DEBUG] mod_sofia.c:344 Channel sofia/external/+
14158867717 at 199.173.100.16:5060 hanging up, cause: EXCHANGE_ROUTING_ERROR
2009-10-22 17:23:31.882153 [DEBUG] mod_sofia.c:382 Sending BYE to
sofia/external/+14158867717 at 199.173.100.16:5060
2009-10-22 17:23:31.882153 [DEBUG] switch_core_state_machine.c:46
sofia/external/+14158867717 at 199.173.100.16:5060 Standard HANGUP, cause:
EXCHANGE_ROUTING_ERROR
2009-10-22 17:23:31.882153 [DEBUG] switch_core_state_machine.c:464
(sofia/external/+14158867717 at 199.173.100.16:5060) State HANGUP going to
sleep
2009-10-22 17:23:31.884357 [INFO] switch_cpp.cpp:1116 HH_HANGUP, agent:1/1
uuid: f8604694-4e44-4c4c-9805-722b3e9d6b68
2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:346
(sofia/external/14159927717) State SOFT_EXECUTE going to sleep
2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:306
(sofia/external/14159927717) Running State Change CS_HANGUP
2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:464
(sofia/external/14159927717) State HANGUP
2009-10-22 17:23:31.885412 [DEBUG] mod_sofia.c:344 Channel
sofia/external/14159927717 hanging up, cause: NORMAL_CLEARING
2009-10-22 17:23:31.885412 [DEBUG] mod_sofia.c:382 Sending BYE to
sofia/external/14159927717
2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:46
sofia/external/14159927717 Standard HANGUP, cause: NORMAL_CLEARING
2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:464
(sofia/external/14159927717) State HANGUP going to sleep
2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:325
(sofia/external/14159927717) State Change CS_HANGUP -> CS_REPORTING
2009-10-22 17:23:31.885412 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/14159927717 [BREAK]
subtracting 5 seconds for hh_call: 1095423
2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:306
(sofia/external/14159927717) Running State Change CS_REPORTING
2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:555
(sofia/external/14159927717) State REPORTING
2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:53
sofia/external/14159927717 Standard REPORTING, cause: NORMAL_CLEARING
2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:555
(sofia/external/14159927717) State REPORTING going to sleep
2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:319
(sofia/external/14159927717) State Change CS_REPORTING -> CS_DESTROY
2009-10-22 17:23:31.885412 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/14159927717 [BREAK]
2009-10-22 17:23:31.885412 [DEBUG] switch_core_session.c:1069 Session 7
(sofia/external/14159927717) Locked, Waiting on external entities
2009-10-22 17:23:31.885412 [NOTICE] switch_core_session.c:1087 Session 7
(sofia/external/14159927717) Ended
2009-10-22 17:23:31.885412 [NOTICE] switch_core_session.c:1089 Close Channel
sofia/external/14159927717 [CS_DESTROY]
2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:401
(sofia/external/14159927717) Running State Change CS_DESTROY
2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:412
(sofia/external/14159927717) State DESTROY
2009-10-22 17:23:31.885412 [DEBUG] mod_sofia.c:261
sofia/external/14159927717 SOFIA DESTROY
2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:60
sofia/external/14159927717 Standard DESTROY
2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:412
(sofia/external/14159927717) State DESTROY going to sleep
2009-10-22 17:23:31.885412 [INFO] switch_cpp.cpp:1116 PCHangup
gw:teliax.comhc:NORMAL_CLEARING du:5 cn:sofia/external/14159927717
2009-10-22 17:23:31.885412 [INFO] switch_cpp.cpp:1116 EVENTC_BILLING_CEIL,
subtracting 5 for hh_call:1095423
2009-10-22 17:23:31.901620 [DEBUG] switch_core_state_machine.c:494 Hangup
Command luarun(hh_hangup.lua f8604694-4e44-4c4c-9805-722b3e9d6b68 1 1):
+OK

2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:343
(sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA going
to sleep
2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:306
(sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
CS_HANGUP
2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:442 handler
already called, skipping state handler.
2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:325
(sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_HANGUP ->
CS_REPORTING
2009-10-22 17:23:31.902676 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:306
(sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
CS_REPORTING
2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:555
(sofia/external/+14158867717 at 199.173.100.16:5060) State REPORTING
2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:53
sofia/external/+14158867717 at 199.173.100.16:5060 Standard REPORTING, cause:
EXCHANGE_ROUTING_ERROR
2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:555
(sofia/external/+14158867717 at 199.173.100.16:5060) State REPORTING going to
sleep
2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:319
(sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_REPORTING
-> CS_DESTROY
2009-10-22 17:23:31.903742 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:23:31.903742 [DEBUG] switch_core_session.c:1069 Session 4
(sofia/external/+14158867717 at 199.173.100.16:5060) Locked, Waiting on
external entities
2009-10-22 17:23:31.903742 [NOTICE] switch_core_session.c:1087 Session 4
(sofia/external/+14158867717 at 199.173.100.16:5060) Ended
2009-10-22 17:23:31.903742 [NOTICE] switch_core_session.c:1089 Close Channel
sofia/external/+14158867717 at 199.173.100.16:5060 [CS_DESTROY]
2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:401
(sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
CS_DESTROY
2009-10-22 17:23:31.904799 [DEBUG] switch_core_state_machine.c:412
(sofia/external/+14158867717 at 199.173.100.16:5060) State DESTROY
2009-10-22 17:23:31.904799 [DEBUG] mod_sofia.c:261 sofia/external/+
14158867717 at 199.173.100.16:5060 SOFIA DESTROY
2009-10-22 17:23:31.904799 [DEBUG] switch_core_state_machine.c:60
sofia/external/+14158867717 at 199.173.100.16:5060 Standard DESTROY
2009-10-22 17:23:31.904799 [DEBUG] switch_core_state_machine.c:412
(sofia/external/+14158867717 at 199.173.100.16:5060) State DESTROY going to
sleep



On Tue, Oct 20, 2009 at 1:44 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Sun, Oct 18, 2009 at 9:48 PM, Matthew Fong <mattdfong at gmail.com> wrote:
>
>> The debug level logs to the console, right? The pastebin, had log level
>> debug, sofia trace on for external and default, and sofia loglevel all 9. Is
>> there another log enable command I'm missing? It seems loglevel all 9
>> outputs enter and exit functions, but at least to my novice eye, it's not
>> too obvious why freeswitch is sending a BYE to my DID provider.
>>
>> Evidently the console loglevel debug did not take because there isn't a
> single DEBUG level line of output. Please try again. Just remember that
> debug level log messages will have A LOT of information so be patient while
> sifting through it. Of course, feel free to pastebin the output and add a
> link to it in this email thread.
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
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From saeedahmad1981 at gmail.com  Thu Oct 22 10:47:44 2009
From: saeedahmad1981 at gmail.com (Saeed Ahmad)
Date: Thu, 22 Oct 2009 19:47:44 +0200
Subject: [Freeswitch-users] Heartbeat question
In-Reply-To: <86a32abc0910091501y40dd34c4r1bca3eeb2b6fa693@mail.gmail.com>
References: <86a32abc0910081956k91b3600sb8c556e589f5b089@mail.gmail.com> 
	<86a32abc0910082003v3f5d4e6dua8e7eae18fd2f939@mail.gmail.com> 
	<6b65470d0910082036g204d5dcbw26e1e43ef80c7c84@mail.gmail.com> 
	<86a32abc0910082245j36b8b3e9ncaf1cdd358fce319@mail.gmail.com> 
	<191c3a030910090902t38282f72l6b6d1733a758836@mail.gmail.com> 
	<86a32abc0910091427i2395fd80t3bc557186ece3972@mail.gmail.com> 
	<86a32abc0910091430hc151ed8of085fcc3af2f1de4@mail.gmail.com> 
	<86a32abc0910091501y40dd34c4r1bca3eeb2b6fa693@mail.gmail.com>
Message-ID: <c96b95650910221047i545601a4m9e0a645341068d15@mail.gmail.com>

What is heartbeat and what are the uses cases?

Sorry i didn't find much information on wiki.

Thanks.



On Sat, Oct 10, 2009 at 12:01 AM, Diego Viola <diego.viola at gmail.com> wrote:

> Here's my heartbeat script now.
>
> #!/usr/bin/env ruby
>
> require 'rubygems'
> require 'fsr'
> require "fsr/listener/inbound"
>
> def custom_channel_heartbeat_handler(event)
>   puts "Got a SESSION_HEARTBEAT at #{Time.now.strftime('%H:%M:%S')}"
> end
> FSL::Inbound.add_event_hook(:SESSION_HEARTBEAT) {|event|
> custom_channel_heartbeat_handler(event) }
> FSR.start_ies!(FSL::Inbound, :host => "localhost", :port => 8021)
>
> Thanks again.
>
> Diego
>
>
> On Fri, Oct 9, 2009 at 9:30 PM, Diego Viola <diego.viola at gmail.com> wrote:
>
>> Here is on two seconds ;)
>>
>> Got a SESSION_HEARTBEAT at 17:17:13
>> Got a SESSION_HEARTBEAT at 17:17:15
>> Got a SESSION_HEARTBEAT at 17:17:17
>> Got a SESSION_HEARTBEAT at 17:17:19
>> Got a SESSION_HEARTBEAT at 17:17:21
>> Got a SESSION_HEARTBEAT at 17:17:23
>> Got a SESSION_HEARTBEAT at 17:17:25
>> Got a SESSION_HEARTBEAT at 17:17:27
>> Got a SESSION_HEARTBEAT at 17:17:29
>> Got a SESSION_HEARTBEAT at 17:17:31
>> Got a SESSION_HEARTBEAT at 17:17:33
>> Got a SESSION_HEARTBEAT at 17:17:35
>> Got a SESSION_HEARTBEAT at 17:17:37
>> Got a SESSION_HEARTBEAT at 17:17:39
>> Got a SESSION_HEARTBEAT at 17:17:41
>> Got a SESSION_HEARTBEAT at 17:17:43
>> Got a SESSION_HEARTBEAT at 17:17:45
>> Got a SESSION_HEARTBEAT at 17:17:47
>> Got a SESSION_HEARTBEAT at 17:17:49
>> Got a SESSION_HEARTBEAT at 17:17:51
>> Got a SESSION_HEARTBEAT at 17:17:53
>> Got a SESSION_HEARTBEAT at 17:17:55
>> Got a SESSION_HEARTBEAT at 17:17:57
>> Got a SESSION_HEARTBEAT at 17:17:59
>> Got a SESSION_HEARTBEAT at 17:18:01
>> Got a SESSION_HEARTBEAT at 17:18:03
>> Got a SESSION_HEARTBEAT at 17:18:05
>> Got a SESSION_HEARTBEAT at 17:18:07
>> Got a SESSION_HEARTBEAT at 17:18:09
>> Got a SESSION_HEARTBEAT at 17:18:11
>> Got a SESSION_HEARTBEAT at 17:18:13
>> Got a SESSION_HEARTBEAT at 17:18:15
>> Got a SESSION_HEARTBEAT at 17:18:17
>>
>>
>>
>> On Fri, Oct 9, 2009 at 9:27 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>
>>> Thanks Anthony, this solved it. You rock :)
>>>
>>> My program now outputs:
>>>
>>> Got a SESSION_HEARTBEAT at 17:14:59
>>> Got a SESSION_HEARTBEAT at 17:15:00
>>> Got a SESSION_HEARTBEAT at 17:15:02
>>> Got a SESSION_HEARTBEAT at 17:15:03
>>> Got a SESSION_HEARTBEAT at 17:15:04
>>> Got a SESSION_HEARTBEAT at 17:15:05
>>> Got a SESSION_HEARTBEAT at 17:15:06
>>> Got a SESSION_HEARTBEAT at 17:15:07
>>> Got a SESSION_HEARTBEAT at 17:15:08
>>> Got a SESSION_HEARTBEAT at 17:15:09
>>> Got a SESSION_HEARTBEAT at 17:15:10
>>> Got a SESSION_HEARTBEAT at 17:15:11
>>> Got a SESSION_HEARTBEAT at 17:15:12
>>> Got a SESSION_HEARTBEAT at 17:15:13
>>> Got a SESSION_HEARTBEAT at 17:15:14
>>> Got a SESSION_HEARTBEAT at 17:15:15
>>> Got a SESSION_HEARTBEAT at 17:15:16
>>> Got a SESSION_HEARTBEAT at 17:15:17
>>> Got a SESSION_HEARTBEAT at 17:15:18
>>> Got a SESSION_HEARTBEAT at 17:15:19
>>> Got a SESSION_HEARTBEAT at 17:15:20
>>> Got a SESSION_HEARTBEAT at 17:15:21
>>> Got a SESSION_HEARTBEAT at 17:15:22
>>> Got a SESSION_HEARTBEAT at 17:15:23
>>> Got a SESSION_HEARTBEAT at 17:15:24
>>> Got a SESSION_HEARTBEAT at 17:15:25
>>> Got a SESSION_HEARTBEAT at 17:15:26
>>> Got a SESSION_HEARTBEAT at 17:15:27
>>> Got a SESSION_HEARTBEAT at 17:15:28
>>> Got a SESSION_HEARTBEAT at 17:15:29
>>> Got a SESSION_HEARTBEAT at 17:15:30
>>>
>>>
>>>
>>>
>>>
>>> On Fri, Oct 9, 2009 at 4:02 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> Update to trunk and try it with fs_cli it for sure will let you do every
>>>> 1 second
>>>>
>>>> in fs_cli type
>>>>
>>>> /events plain all
>>>>
>>>> if you make that call you will see one every 1 second
>>>>
>>>>
>>>>
>>>> On Fri, Oct 9, 2009 at 12:45 AM, Diego Viola <diego.viola at gmail.com>wrote:
>>>>
>>>>> Nope, I was just wondering why it didn't work at 1 second exactly...
>>>>>
>>>>>
>>>>> On Fri, Oct 9, 2009 at 3:36 AM, William Suffill <
>>>>> william.suffill at gmail.com> wrote:
>>>>>
>>>>>> Why do you need it every second? If you want real time channel counts
>>>>>> you would  be able to track each create/destroy even instead of
>>>>>> relying on the heartbeat summary.
>>>>>>
>>>>>> -- W
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:213-799-1400
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Oct 22 10:54:40 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 22 Oct 2009 10:54:40 -0700
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179
In-Reply-To: <b39273530910220551q14d22150ldf8d4bff0b6fd011@mail.gmail.com>
References: <mailman.1516.1256123625.2495.freeswitch-users@lists.freeswitch.org>
	<a60fed5e0910212108v4755203y693b8bdec5501f5f@mail.gmail.com>
	<87f2f3b90910212141w72b8f5a2oce032adb974d71e2@mail.gmail.com>
	<b39273530910220551q14d22150ldf8d4bff0b6fd011@mail.gmail.com>
Message-ID: <87f2f3b90910221054l8abc3far9a18a66f0959f950@mail.gmail.com>

On Thu, Oct 22, 2009 at 5:51 AM, Rupa Schomaker <rupa at rupa.com> wrote:

> cond would be helpful here?  I updated the wiki on this one just now
> with a bit more detail.  It is a api call. so, you'd use it like:
>
> ${cond(eval ? trueval : falseval)}
>
> so to get a value of ERR if the var my myvar is > 15 you could:
>
> ${cond(${myvar} > 15 ? ERR : OK)}
>
> If both sides of the comparison operator are numeric then it does
> numeric comparison otherwise it does lexical string comparison.
>

Rupa,

Yes, you can do the set/cond API trick but you can only do it in the action
or anti-action tags, not in the condition tags. I'm sure you know that but I
want all those reading this thread to make the connection.
-MC
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From kristian.kielhofner at gmail.com  Thu Oct 22 10:56:15 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Thu, 22 Oct 2009 13:56:15 -0400
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
In-Reply-To: <65d96fc80910220858u788c6146sced38e19418e681a@mail.gmail.com>
References: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>
	<191c3a030910201434q4c6e65e1h58c5f0a82addc2d3@mail.gmail.com>
	<2d9149cd0910202050i36a7ea82nc7253d87e38f990a@mail.gmail.com>
	<191c3a030910211154gc8412b5gd999fbedffa02b53@mail.gmail.com>
	<2d9149cd0910220844v2aa73502j4b88fdbd75a0d833@mail.gmail.com>
	<65d96fc80910220858u788c6146sced38e19418e681a@mail.gmail.com>
Message-ID: <2d9149cd0910221056y3716d8c4s708fbdb6a5563733@mail.gmail.com>

On Thu, Oct 22, 2009 at 11:58 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>
> and what these few issues are? :P
>

  One fax machine here in the office (still testing others) correctly
sends all fax pages.  A minute or so after the fax is marked
"successful" on both sides it hangs up, redials, and resends the last
page...  It never did it while connected to the PSTN but then again my
other fax machine isn't doing it either.  I'm going to test with more
fax machines to see if it's an issue with that specific machine.

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From gkuri at ieee.org  Thu Oct 22 11:16:29 2009
From: gkuri at ieee.org (Gabriel Kuri)
Date: Thu, 22 Oct 2009 11:16:29 -0700
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
In-Reply-To: <2d9149cd0910221056y3716d8c4s708fbdb6a5563733@mail.gmail.com>
References: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>	<191c3a030910201434q4c6e65e1h58c5f0a82addc2d3@mail.gmail.com>	<2d9149cd0910202050i36a7ea82nc7253d87e38f990a@mail.gmail.com>	<191c3a030910211154gc8412b5gd999fbedffa02b53@mail.gmail.com>	<2d9149cd0910220844v2aa73502j4b88fdbd75a0d833@mail.gmail.com>	<65d96fc80910220858u788c6146sced38e19418e681a@mail.gmail.com>
	<2d9149cd0910221056y3716d8c4s708fbdb6a5563733@mail.gmail.com>
Message-ID: <4AE0A17D.3000301@ieee.org>

Out of curiosity, is it a Super G3 (ie v.34) capable FAX? We've had
nothing but intermittent problems with Super G3 FAXes over T.38, unless
v.34 is strictly turned off on the machine.

Gabe


Kristian Kielhofner wrote:
> On Thu, Oct 22, 2009 at 11:58 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>
>   One fax machine here in the office (still testing others) correctly
> sends all fax pages.  A minute or so after the fax is marked
> "successful" on both sides it hangs up, redials, and resends the last
> page...  It never did it while connected to the PSTN but then again my
> other fax machine isn't doing it either.  I'm going to test with more
> fax machines to see if it's an issue with that specific machine.
> 



From msc at freeswitch.org  Thu Oct 22 11:18:41 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 22 Oct 2009 11:18:41 -0700
Subject: [Freeswitch-users] Heartbeat question
In-Reply-To: <c96b95650910221047i545601a4m9e0a645341068d15@mail.gmail.com>
References: <86a32abc0910081956k91b3600sb8c556e589f5b089@mail.gmail.com>
	<86a32abc0910082003v3f5d4e6dua8e7eae18fd2f939@mail.gmail.com>
	<6b65470d0910082036g204d5dcbw26e1e43ef80c7c84@mail.gmail.com>
	<86a32abc0910082245j36b8b3e9ncaf1cdd358fce319@mail.gmail.com>
	<191c3a030910090902t38282f72l6b6d1733a758836@mail.gmail.com>
	<86a32abc0910091427i2395fd80t3bc557186ece3972@mail.gmail.com>
	<86a32abc0910091430hc151ed8of085fcc3af2f1de4@mail.gmail.com>
	<86a32abc0910091501y40dd34c4r1bca3eeb2b6fa693@mail.gmail.com>
	<c96b95650910221047i545601a4m9e0a645341068d15@mail.gmail.com>
Message-ID: <87f2f3b90910221118n5d9fa0b9u92b45227c1d7ec97@mail.gmail.com>

On Thu, Oct 22, 2009 at 10:47 AM, Saeed Ahmad <saeedahmad1981 at gmail.com>wrote:

> What is heartbeat and what are the uses cases?
>
> Sorry i didn't find much information on wiki.
>
> Thanks.
>
> A session heartbeat is just an event that is sent to your script and gives
you updated information about the call in progress. It is *VERY* different
from the system heartbeat, which is sent out to all event socket
subscribers, not just a particular session.

-MC
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From ujjval at simplesignal.com  Thu Oct 22 11:42:49 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Thu, 22 Oct 2009 11:42:49 -0700
Subject: [Freeswitch-users] Connecting to FS CLI...just hangs..
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701E84092@EXMBXCLUS01.citservers.local>

It just hangs....and I CTRL-C out of it.

[root at ss]# ./fs_cli -H 127.0.0.1


^C
[ERROR] libs/esl/fs_cli.c:652 main() Error Connecting [Connection Error]


Freeswitch process is running:

[root at ss bin]# ps -ef|grep free
root      8889 31039  0 12:36 pts/2    00:00:00 ./freeswitch
root      8952 31039  0 12:42 pts/2    00:00:00 grep free


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From tculjaga at gmail.com  Thu Oct 22 12:08:36 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 22 Oct 2009 21:08:36 +0200
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
In-Reply-To: <4AE0A17D.3000301@ieee.org>
References: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>
	<191c3a030910201434q4c6e65e1h58c5f0a82addc2d3@mail.gmail.com>
	<2d9149cd0910202050i36a7ea82nc7253d87e38f990a@mail.gmail.com>
	<191c3a030910211154gc8412b5gd999fbedffa02b53@mail.gmail.com>
	<2d9149cd0910220844v2aa73502j4b88fdbd75a0d833@mail.gmail.com>
	<65d96fc80910220858u788c6146sced38e19418e681a@mail.gmail.com>
	<2d9149cd0910221056y3716d8c4s708fbdb6a5563733@mail.gmail.com>
	<4AE0A17D.3000301@ieee.org>
Message-ID: <65d96fc80910221208h70c86457s4ca5f68289bacad2@mail.gmail.com>

indeed, this looks like a "dialect" problem between your fax machine and
your T.38 device.
Anyhow, T.38  doesn't work well with SG3... I Always have to disable v.34 in
order to have a reliable fax service.

Also, cisco uses to suppress CM so that SG3 timeouts on ANSam the
communication fallbacks to ordinary G3.

Kristian, just for fun, what are you using to send the fax ?

T.

On Thu, Oct 22, 2009 at 8:16 PM, Gabriel Kuri <gkuri at ieee.org> wrote:

> Out of curiosity, is it a Super G3 (ie v.34) capable FAX? We've had
> nothing but intermittent problems with Super G3 FAXes over T.38, unless
> v.34 is strictly turned off on the machine.
>
> Gabe
>
>
> Kristian Kielhofner wrote:
> > On Thu, Oct 22, 2009 at 11:58 AM, Tihomir Culjaga <tculjaga at gmail.com>
> wrote:
> >
> >   One fax machine here in the office (still testing others) correctly
> > sends all fax pages.  A minute or so after the fax is marked
> > "successful" on both sides it hangs up, redials, and resends the last
> > page...  It never did it while connected to the PSTN but then again my
> > other fax machine isn't doing it either.  I'm going to test with more
> > fax machines to see if it's an issue with that specific machine.
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jmesquita at freeswitch.org  Thu Oct 22 12:11:09 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 22 Oct 2009 17:11:09 -0200
Subject: [Freeswitch-users] Connecting to FS CLI...just hangs..
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71701E84092@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84092@EXMBXCLUS01.citservers.local>
Message-ID: <ebbc2d550910221211k18171473p7f4d6f597f6b21d2@mail.gmail.com>

Hangs for how long? Are you sure you are not just waiting on a timeout?

JM

On Thu, Oct 22, 2009 at 4:42 PM, Ujjval Karihaloo
<ujjval at simplesignal.com>wrote:

>  It just hangs?.and I CTRL-C out of it.
>
>
>
> [root at ss]# ./fs_cli -H 127.0.0.1
>
>
>
>
>
> ^C
>
> [ERROR] libs/esl/fs_cli.c:652 main() Error Connecting [Connection Error]
>
>
>
>
>
> Freeswitch process is running:
>
>
>
> [root at ss bin]# ps -ef|grep free
>
> root      8889 31039  0 12:36 pts/2    00:00:00 ./freeswitch
>
> root      8952 31039  0 12:42 pts/2    00:00:00 grep free
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kristian.kielhofner at gmail.com  Thu Oct 22 12:13:58 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Thu, 22 Oct 2009 15:13:58 -0400
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
In-Reply-To: <4AE0A17D.3000301@ieee.org>
References: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>
	<191c3a030910201434q4c6e65e1h58c5f0a82addc2d3@mail.gmail.com>
	<2d9149cd0910202050i36a7ea82nc7253d87e38f990a@mail.gmail.com>
	<191c3a030910211154gc8412b5gd999fbedffa02b53@mail.gmail.com>
	<2d9149cd0910220844v2aa73502j4b88fdbd75a0d833@mail.gmail.com>
	<65d96fc80910220858u788c6146sced38e19418e681a@mail.gmail.com>
	<2d9149cd0910221056y3716d8c4s708fbdb6a5563733@mail.gmail.com>
	<4AE0A17D.3000301@ieee.org>
Message-ID: <2d9149cd0910221213h20661e11pdccafdf407c79b02@mail.gmail.com>

Gabe,

  I don't think any of them are plus the T.38 SDP tells me the bitrate
is 14400, certainly not V.34 speed.

  Are you saying the machine even trying to negotiate V.34 poses a problem?

On Thu, Oct 22, 2009 at 2:16 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
> Out of curiosity, is it a Super G3 (ie v.34) capable FAX? We've had
> nothing but intermittent problems with Super G3 FAXes over T.38, unless
> v.34 is strictly turned off on the machine.
>
> Gabe
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From bottleman at icf.org.ru  Thu Oct 22 12:21:39 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Thu, 22 Oct 2009 23:21:39 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910220753n6d2efa39l470791f75db9a640@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>
	<65d96fc80910220753n6d2efa39l470791f75db9a640@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>

On 2009-10-22 16:53 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

finally i fix this rtp bug, check new wersion please.

TC>>
TC>> TC>
TC>> TC>Do you need some logs ?
TC>>
TC>> try disable medai-proxy, there is issue with rtp now then medai-proxy or
TC>> transcoding enabled.
TC>>
TC>>
TC>Outbound calls:
TC>
TC>disabled rtp proxy and it is still the same issue ... audio delay H323 =>
TC>SIP endpoint.
TC>
TC>
TC>
TC>
TC>
TC>
TC>Inbound calls:
TC>
TC>This is the extension i use to register my Avaya SIP phone to FS.
TC>
TC>
TC><include>
TC>  <user id="1001">
TC>    <params>
TC>      <param name="password" value="$${default_password}"/>
TC>      <param name="vm-password" value="1001"/>
TC>    </params>
TC>    <variables>
TC>      <variable name="toll_allow" value="domestic,international,local"/>
TC>      <variable name="accountcode" value="1001"/>
TC>      <variable name="user_context" value="default"/>
TC>      <variable name="effective_caller_id_name" value="Extension 1001"/>
TC>      <variable name="effective_caller_id_number" value="1001"/>
TC>      <variable name="outbound_caller_id_name"
TC>value="$${outbound_caller_name}"/>
TC>      <variable name="outbound_caller_id_number"
TC>value="$${outbound_caller_id}"/>
TC>      <variable name="callgroup" value="techsupport"/>
TC>    </variables>
TC>  </user>
TC></include>
TC>
TC>
TC>This is my h323.conf.xml
TC>
TC>
TC><configuration name="h323.conf" description="H323 Endpoints">
TC>  <settings>
TC>    <param name="trace-level" value="4"/>
TC>    <param name="context" value="default"/>
TC>    <param name="dialplan" value="XML"/>
TC>    <param name="codec-prefs" value="PCMU,PCMA,GSM,G729,G726"/>
TC>    <param name="gk-address" value=""/>    <!-- empty to disable, "*" to
TC>search LAN -->
TC>    <param name="gk-identifer" value=""/>  <!-- optional name of gk -->
TC>    <param name="gk-interface" value=""/>  <!-- optional listener interface
TC>name -->
TC>  </settings>
TC>  <listeners>
TC>    <listener name="default">
TC>      <param name="h323-ip" value="10.4.62.7"/>
TC>      <param name="h323-port" value="1720"/>
TC>    </listener>
TC>  </listeners>
TC></configuration>
TC>
TC>I'm using default context and an inbound call looks for a registered user in
TC>default context where 1001 user is registered to.
TC>
TC>
TC>
TC>here is the log for an outgoing call:
TC>http://pastebin.freeswitch.org/10799and here is a tshark output:
TC>http://pastebin.freeswitch.org/10800
TC>
TC>
TC>there are 2 thing that are not working here:
TC>
TC>
TC>1. no audio at all!
TC>2. hangup from SIP User side doesn't release the H323 leg
TC>
TC>
TC>
TC>
TC>
TC>
TC>
TC>
TC>
TC>
TC>
TC>two points for your reference in the logs:
TC>
TC>
TC>1. Here, SIP User disconnected the SIP leg, but nothing was triggered in
TC>mod_h323 ... as the callback function "on_hangup" (in mod_h323.cpp) was
TC>never triggered!
TC>
TC>freeswitch at subZero>
TC>freeswitch at subZero>
TC>freeswitch at subZero> recv 371 bytes from udp/[10.4.62.89]:5060 at
TC>14:39:36.714521:
TC>   ------------------------------------------------------------------------
TC>   BYE sip:mod_sofia at 10.4.62.7:5060 SIP/2.0
TC>   From: <sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>
TC>;transport=udp>;tag=-22166b474ae08abf-7_T10.4.62.89
TC>   To: sip:1001282166 at 10.4.62.7 <sip%3A1001282166 at 10.4.62.7>
TC>;tag=Qpc53NZ2cZc1N
TC>   Call-ID: 8aa825c6-39bb-122d-bb89-00110a5be1f0
TC>   CSeq: 127 BYE
TC>   Via: SIP/2.0/UDP
TC>10.4.62.89;branch=z9hG4bK-7e5dc720_442d0f8-2d8bf1174f235bec_B
TC>   Content-Length: 0
TC>   Max-Forwards: 70
TC>   Supported: replaces
TC>
TC>   ------------------------------------------------------------------------
TC>2009-10-22 16:39:36.714604 [NOTICE] sofia.c:322 Hangup sofia/internal/
TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [CS_CONSUME_MEDIA]
TC>[NORMAL_CLEARING]
TC>2009-10-22 16:39:36.714604 [DEBUG] switch_channel.c:1683 Send signal
TC>sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [KILL]
TC>2009-10-22 16:39:36.714604 [DEBUG] switch_core_session.c:932 Send signal
TC>sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [BREAK]
TC>send 520 bytes to udp/[10.4.62.89]:5060 at 14:39:36.715258:
TC>   ------------------------------------------------------------------------
TC>   SIP/2.0 200 OK
TC>   Via: SIP/2.0/UDP
TC>10.4.62.89;branch=z9hG4bK-7e5dc720_442d0f8-2d8bf1174f235bec_B
TC>   From: <sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>
TC>;transport=udp>;tag=-22166b474ae08abf-7_T10.4.62.89
TC>   To: sip:1001282166 at 10.4.62.7 <sip%3A1001282166 at 10.4.62.7>
TC>;tag=Qpc53NZ2cZc1N
TC>   Call-ID: 8aa825c6-39bb-122d-bb89-00110a5be1f0
TC>   CSeq: 127 BYE
TC>   User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported
TC>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
TC>NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
TC>   Supported: timer, precondition, path, replaces
TC>   Content-Length: 0
TC>
TC>   ------------------------------------------------------------------------
TC>2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:503
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State
TC>CONSUME_MEDIA going to sleep
TC>2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:398
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) Running State
TC>Change CS_HANGUP
TC>2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:434
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State HANGUP
TC>2009-10-22 16:39:36.721097 [DEBUG] mod_sofia.c:338 Channel sofia/internal/
TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> hanging up, cause:
TC>NORMAL_CLEARING
TC>2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:46
TC>sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> Standard HANGUP,
TC>cause: NORMAL_CLEARING
TC>2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:434
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State HANGUP
TC>going to sleep
TC>2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:476
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State Change
TC>CS_HANGUP -> CS_REPORTING
TC>2009-10-22 16:39:36.721097 [DEBUG] switch_core_session.c:932 Send signal
TC>sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [BREAK]
TC>2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:398
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) Running State
TC>Change CS_REPORTING
TC>2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:612
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State REPORTING
TC>2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:53
TC>sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> Standard
TC>REPORTING, cause: NORMAL_CLEARING
TC>2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:612
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State REPORTING
TC>going to sleep
TC>2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:411
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State Change
TC>CS_REPORTING -> CS_DESTROY
TC>2009-10-22 16:39:36.721097 [DEBUG] switch_core_session.c:1068 Session 10
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) Locked, Waiting
TC>on external entities
TC>
TC>freeswitch at subZero>
TC>
TC>
TC>
TC>
TC>
TC>2. switch_ivr_originate.c complains => 2009-10-22 16:39:41.630243 [ERR]
TC>switch_ivr_originate.c:1510 Cannot create outgoing channel of type [user]
TC>cause: [NORMAL_CLEARING]
TC>
TC>
TC>freeswitch at subZero>
TC>freeswitch at subZero>
TC>freeswitch at subZero> 2009-10-22 16:39:41.630243 [DEBUG] h323pdu.cxx:609
TC>Receiving PDU [ip$10.4.62.7:1720/ip$10.4.62.31:16013] :
TC>  {
TC>    q931pdu = {
TC>      protocolDiscriminator = 8
TC>      callReference = 23
TC>      from = originator
TC>      messageType = ReleaseComplete
TC>      IE: Cause - Normal ca
TC>ll clearing = {
TC>        80 90                                              ..
TC>      }
TC>      IE: User-User = {
TC>        25 80 06 00 08 91 4a 00  04 11 00 11 00 96 40 1f   %.....J....... at .
TC>        4e be 4f 11 de 80 76 ae  99 96 03 ba ff 10 80 01   N.O.
TC>..v.........
TC>        ...
TC>      }
TC>    }
TC>    h225pdu = {
TC>      h323_uu_pdu = {
TC>        h323_message_body = releaseComplete {
TC>          protocolIdentifier = 0.0.8.2250.0.4
TC>          callIdentifier = {
TC>            guid =  16 octets {
TC>              96 40
TC> 1f 4e be 4f 11 de  80 76 ae 99 96 03 ba ff   . at .N.O...v......
TC>            }
TC>          }
TC>        }
TC>        h245Tunneling = true
TC>      }
TC>    }
TC>  }
TC>2009-10-22 16:39:41.630243 [DEBUG] h323.cxx:1292 Handling PDU:
TC>ReleaseComplete callRef=23
TC>2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:745
TC>======>FSH323Connection::OnReceivedReleaseComplete cause = Normal call
TC>clearing value = 16
TC>2009-10-22 16:39:41.630243 [NOTICE] mod_h323.cpp:749 Hangup h323/1001
TC>[CS_EXECUTE] [NORMAL_CLEARING]
TC>2009-10-22 16:39:41.630243 [DEBUG] switch_channel.c:1683 Send signal
TC>h323/1001 [KILL]
TC>2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:840
TC>======>FSH323Connection::kill_channel FSH323Connection
TC>2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:841 Kill 1 on connection
TC>FSH323Connection
TC>2009-10-22 16:39:41.630243 [DEBUG] switch_core_session.c:630 Send signal
TC>h323/1001 [BREAK]
TC>2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:840
TC>======>FSH323Connection::kill_channel FSH323Connection
TC>2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:841 Kill 3 on connection
TC>FSH323Connection
TC>2009-10-22 16:39:41.630243 [DEBUG] switch_ivr_originate.c:2138 Originate
TC>Resulted in Error Cause: 16 [NORMAL_CLEARING]
TC>2009-10-22 16:39:41.630243 [ERR] switch_ivr_originate.c:1510 Cannot create
TC>outgoing channel of type [user] cause: [NORMAL_CLEARING]
TC>2009-10-22 16:39:41.630243 [DEBUG] switch_ivr_originate.c:2138 Originate
TC>Resulted in Error Cause: 16 [NORMAL_CLEARING]
TC>2009-10-22 16:39:41.630243 [INFO] mod_dptools.c:2093 Originate Failed.
TC>Cause: NORMAL_CLEARING
TC>2009-10-22 16:39:41.630243 [DEBUG] mod_dptools.c:2120 Continue on fail
TC>[true]:  Cause: NORMAL_CLEARING
TC>2009-10-22 16:39:41.630243 [DEBUG] switch_core_state_machine.c:491
TC>(h323/1001) State EXECUTE going to sleep
TC>2009-10-22 16:39:41.630243 [DEBUG] switch_core_state_machine.c:398
TC>(h323/1001) Running State Change CS_HANGUP
TC>2009-10-22 16:39:41.630243 [DEBUG] switch_core_state_machine.c:434
TC>(h323/1001) State HANGUP
TC>2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:1454 ======>switch_status_t
TC>on_hangup
TC>2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:1459 ----->
TC>2009-10-22 16:39:41.630243 [DEBUG] h323pdu.cxx:1272 Call End Reason Normal
TC>call clearing
TC>2009-10-22 16:39:41.630243 [DEBUG] h323ep.cxx:2694 Clearing connection ip$
TC>10.4.62.31:16013/23 reason=EndedByRemoteUser
TC>2009-10-22 16:39:41.630243 [DEBUG] h323.cxx:1034 Call end reason for ip$
TC>10.4.62.31:16013/23 set to EndedByRemoteUser
TC>2009-10-22 16:39:41.630243 [DEBUG] h323.cxx:1052 Sending release complete
TC>PDU: callRef=23
TC>2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:764
TC>======>FSH323Connection::OnSendReleaseComplete cause = Normal call clearing
TC>value = 16
TC>2009-10-22 16:39:41.630243 [DEBUG] h323pdu.cxx:609 Sending PDU
TC>[(noaddr)/(noaddr)] :
TC>  command endSessionCommand disconnect <<null>>
TC>2009-10-22 16:39:41.630243 [DEBUG] h323pdu.cxx:609 Sending PDU [ip$
TC>10.4.62.7:1720/ip$10.4.62.31:16013] :
TC>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From anthony.minessale at gmail.com  Thu Oct 22 12:26:44 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 22 Oct 2009 14:26:44 -0500
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
In-Reply-To: <4256bf830910221042l68ea769cg532bbd339bae0790@mail.gmail.com>
References: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
	<177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>
	<4256bf830908041151l55980e3fifb85c75b87535426@mail.gmail.com>
	<3D0B855A-5B2C-4C6F-9306-64B0ECE97091@jerris.com>
	<4256bf830910180725y72539d98gb138934afc18d58b@mail.gmail.com>
	<A5ECDCD9-1A12-4A31-89F0-E6F4C79FE9E9@jerris.com>
	<4256bf830910182148mf7f2bf9vdcccd687e0caa1e7@mail.gmail.com>
	<87f2f3b90910191144q25750bayfd7a7448e82e78ff@mail.gmail.com>
	<4256bf830910221042l68ea769cg532bbd339bae0790@mail.gmail.com>
Message-ID: <191c3a030910221226j167e0adw5f631f759cd7cfa0@mail.gmail.com>

1) please do not flood emails with traces reference pastebins
2) the error is a routing loop being prevented so you are looping the call
into your same box too many times.

You could have figured this out your self by looking at one of the lines of
code where it reported the hangup.

 if (!switch_strlen_zero(forwardvar)) {
        forwardval = atoi(forwardvar) - 1;
    }
    if (forwardval <= 0) {
        switch_channel_hangup(channel, SWITCH_CAUSE_EXCHANGE_ROUTING_ERROR);
        return SWITCH_STATUS_FALSE;
    }


On Thu, Oct 22, 2009 at 12:42 PM, Matthew Fong <mattdfong at gmail.com> wrote:

> I figured out how to turn debug mode on. For some reason F8 didn't work, so
> I just set the default debug mode to DEBUG in switch.conf.xml
>
> http://pastebin.freeswitch.org/10805
> <http://pastebin.freeswitch.org/10805>So my program does more or less the
> same sequence for an Agent every time when connected to another party. They
> uuid_kill the bridged parties uuid, and then are transfered to an extension
> that park's the agent (using the transfer_after_bridge variable -- in my
> dialplan extn 1998). In some instances the agent is then transfered to
> another extension, played a wav file, and then park'ed again
>
> Sometimes, FreeSWITCH unexpectidly hangsup the Agent channel citing an
> EXCHANGE_ROUTING_ERROR. It seems (to me to happen at random times)...and
> it's more prevelent with some did providers than others. With
> didforsale.org it happens about one ever 5 bridges. With icall.com it only
> happens about once every 30 bridges. With IPKall it happens rarely. I can't
> figure it out, and it's driving me crazy :)
>
>
> A GOOD UUID_KILL SEQUENCE:
> 2009-10-22 17:22:31.741845 [DEBUG] switch_ivr_play_say.c:1432 done playing
> file
> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:340
> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXECUTE going to
> sleep
> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
> CS_HIBERNATE
> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:355
> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE
> 2009-10-22 17:22:31.741845 [DEBUG] mod_sofia.c:160 sofia/external/+
> 14158867717 at 199.173.100.16:5060 SOFIA HIBERNATE
> 2009-10-22 17:22:31.741845 [DEBUG] switch_ivr_bridge.c:650
> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
> CS_HIBERNATE -> CS_RESET
> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:355
> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE going to
> sleep
> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
> CS_RESET
> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:336
> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET
> 2009-10-22 17:22:31.741845 [DEBUG] mod_sofia.c:145 sofia/external/+
> 14158867717 at 199.173.100.16:5060 SOFIA RESET
> 2009-10-22 17:22:31.741845 [DEBUG] switch_ivr_bridge.c:637 sofia/external/+
> 14158867717 at 199.173.100.16:5060 CUSTOM RESET
> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:67
> sofia/external/+14158867717 at 199.173.100.16:5060 Standard RESET
> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:336
> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET going to
> sleep
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:340
> (sofia/external/14159927717) State EXECUTE going to sleep
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/14159927717) Running State Change CS_HIBERNATE
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:355
> (sofia/external/14159927717) State HIBERNATE
> 2009-10-22 17:22:31.748086 [DEBUG] mod_sofia.c:160
> sofia/external/14159927717 SOFIA HIBERNATE
> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:650
> (sofia/external/14159927717) State Change CS_HIBERNATE -> CS_RESET
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/14159927717 [BREAK]
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:355
> (sofia/external/14159927717) State HIBERNATE going to sleep
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/14159927717) Running State Change CS_RESET
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:336
> (sofia/external/14159927717) State RESET
> 2009-10-22 17:22:31.748086 [DEBUG] mod_sofia.c:145
> sofia/external/14159927717 SOFIA RESET
> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:637
> sofia/external/14159927717 CUSTOM RESET
> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:642
> (sofia/external/14159927717) State Change CS_RESET -> CS_SOFT_EXECUTE
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/14159927717 [BREAK]
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:336
> (sofia/external/14159927717) State RESET going to sleep
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/14159927717) Running State Change CS_SOFT_EXECUTE
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:346
> (sofia/external/14159927717) State SOFT_EXECUTE
> 2009-10-22 17:22:31.748086 [DEBUG] mod_sofia.c:442 SOFIA TRANSMIT
> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:660
> sofia/external/14159927717 CUSTOM SOFT_EXECUTE
> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_originate.c:674
> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_RESET ->
> CS_SOFT_EXECUTE
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:120 sofia/external/+
> 14158867717 at 199.173.100.16:5060 receive message [DISPLAY]
> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:120
> sofia/external/14159927717 receive message [DISPLAY]
> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:924
> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
> CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:968 sofia/external/+
> 14158867717 at 199.173.100.16:5060 receive message [BRIDGE]
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:630 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:975
> sofia/external/14159927717 receive message [BRIDGE]
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:630 Send signal
> sofia/external/14159927717 [BREAK]
> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:1019
> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:22:31.749553 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
> CS_EXCHANGE_MEDIA
> 2009-10-22 17:22:31.749553 [DEBUG] switch_core_state_machine.c:343
> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
> 2009-10-22 17:22:31.749553 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK
> multipler is being overrided to1
> 2009-10-22 17:22:34.300662 [INFO] switch_cpp.cpp:1116 BG_AG_GEN_CALLS,
> Agents total:1 avail:0 OutChan:0 AR:0 +-Calls:0:Q Call NYNoc
> multipler is being overrided to1
> 2009-10-22 17:22:39.303041 [INFO] switch_cpp.cpp:1116 BG_AG_GEN_CALLS,
> Agents total:1 avail:0 OutChan:0 AR:0 +-Calls:0:Q Call NYNoc
> 2009-10-22 17:22:40.626379 [NOTICE] sofia.c:328 Hangup
> sofia/external/14159927717 [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
> 2009-10-22 17:22:40.626379 [DEBUG] switch_channel.c:1810 Send signal
> sofia/external/14159927717 [KILL]
> 2009-10-22 17:22:40.626379 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/14159927717 [BREAK]
> active_true=1
> --contact NOT in playmsg or conference, update AGENT
> 2009-10-22 17:22:40.626379 [DEBUG] switch_core_state_machine.c:437 thread
> mismatch skipping state handler.
> 2009-10-22 17:22:40.626379 [INFO] switch_cpp.cpp:1116
> EVENTC_HANGUP_CONTACT, Q_CONTACT fired for agent:1/1 for qcall:1
> contact:4801442/4
> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:440
> sofia/external/14159927717 ending bridge by request from write function
> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:495 sofia/external/+
> 14158867717 at 199.173.100.16:5060 receive message [UNBRIDGE]
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:630 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
> DONE [sofia/external/+14158867717 at 199.173.100.16:5060]
> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:523 Send signal
> sofia/external/14159927717 [BREAK]
> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr.c:1361 (sofia/external/+
> 14158867717 at 199.173.100.16:5060) State Change CS_EXCHANGE_MEDIA ->
> CS_ROUTING
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr.c:1365 sofia/external/+
> 14158867717 at 199.173.100.16:5060 receive message [TRANSFER]
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:630 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:22:40.639760 [NOTICE] switch_ivr.c:1367 Transfer
> sofia/external/+14158867717 at 199.173.100.16:5060 to xml[1998 at default]
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:343
> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
> going to sleep
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
> CS_ROUTING
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:333
> (sofia/external/+14158867717 at 199.173.100.16:5060) State ROUTING
> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:130 sofia/external/+
> 14158867717 at 199.173.100.16:5060 SOFIA ROUTING
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:78
> sofia/external/+14158867717 at 199.173.100.16:5060 Standard ROUTING
> 2009-10-22 17:22:40.639760 [INFO] mod_dialplan_xml.c:391 Processing
> +14158867717->1998 in context default
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->tod_example] continue=true
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (PASS)
> [tod_example] ${strftime(%w)}(4) =~ /^([1-5])$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (PASS)
> [tod_example] ${strftime(%H%M)}(1722) =~ /^((09|1[0-7])[0-5][0-9]|1800)$/
> break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Action
> set(open=true)
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->ivr_demo] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [ivr_demo] destination_number(1998) =~ /^5000$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1975] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1975] destination_number(1998) =~ /^1975$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1971] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1971] destination_number(1998) =~ /^1971$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1972] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1972] destination_number(1998) =~ /^1972$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1920] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1920] destination_number(1998) =~ /^1920$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1981] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1981] destination_number(1998) =~ /^1981$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1982] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1982] destination_number(1998) =~ /^1982$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1481] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1481] destination_number(1998) =~ /^1481$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1989] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1989] destination_number(1998) =~ /^1989$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1990] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1990] destination_number(1998) =~ /^1990$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1991] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1991] destination_number(1998) =~ /^1991$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1992] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1992] destination_number(1998) =~ /^1992$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1993] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1993] destination_number(1998) =~ /^1993$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1994] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1994] destination_number(1998) =~ /^1994$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1995] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1995] destination_number(1998) =~ /^1995$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1996] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1996] destination_number(1998) =~ /^1996$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1997] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1997] destination_number(1998) =~ /^1997$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1998] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (PASS)
> [1998] destination_number(1998) =~ /^1998$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Action
> set(transfer_after_bridge=1998)
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Action park()
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:114
> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_ROUTING
> -> CS_EXECUTE
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:333
> (sofia/external/+14158867717 at 199.173.100.16:5060) State ROUTING going to
> sleep
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
> CS_EXECUTE
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:340
> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXECUTE
> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:173 sofia/external/+
> 14158867717 at 199.173.100.16:5060 SOFIA EXECUTE
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:151
> sofia/external/+14158867717 at 199.173.100.16:5060 Standard EXECUTE
> EXECUTE sofia/external/+14158867717 at 199.173.100.16:5060 set(open=true)
> 2009-10-22 17:22:40.639760 [DEBUG] mod_dptools.c:752 sofia/external/+
> 14158867717 at 199.173.100.16:5060 SET [open]=[true]
> EXECUTE sofia/external/+14158867717 at 199.173.100.16:5060set(transfer_after_bridge=1998)
> 2009-10-22 17:22:40.639760 [DEBUG] mod_dptools.c:752 sofia/external/+
> 14158867717 at 199.173.100.16:5060 SET [transfer_after_bridge]=[1998]
> EXECUTE sofia/external/+14158867717 at 199.173.100.16:5060 park()
> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:446
> sofia/external/14159927717 ending bridge by request from read function
> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
> DONE [sofia/external/14159927717]
> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:523 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> subtracting 9 seconds for hh_call: 1095422
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:346
> (sofia/external/14159927717) State SOFT_EXECUTE going to sleep
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/14159927717) Running State Change CS_HANGUP
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:464
> (sofia/external/14159927717) State HANGUP
> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:344 Channel
> sofia/external/14159927717 hanging up, cause: NORMAL_CLEARING
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:46
> sofia/external/14159927717 Standard HANGUP, cause: NORMAL_CLEARING
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:464
> (sofia/external/14159927717) State HANGUP going to sleep
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:325
> (sofia/external/14159927717) State Change CS_HANGUP -> CS_REPORTING
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/14159927717 [BREAK]
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/14159927717) Running State Change CS_REPORTING
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:555
> (sofia/external/14159927717) State REPORTING
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:53
> sofia/external/14159927717 Standard REPORTING, cause: NORMAL_CLEARING
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:555
> (sofia/external/14159927717) State REPORTING going to sleep
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:319
> (sofia/external/14159927717) State Change CS_REPORTING -> CS_DESTROY
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/14159927717 [BREAK]
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:1069 Session 6
> (sofia/external/14159927717) Locked, Waiting on external entities
> 2009-10-22 17:22:40.639760 [NOTICE] switch_core_session.c:1087 Session 6
> (sofia/external/14159927717) Ended
> 2009-10-22 17:22:40.639760 [NOTICE] switch_core_session.c:1089 Close
> Channel sofia/external/14159927717 [CS_DESTROY]
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:401
> (sofia/external/14159927717) Running State Change CS_DESTROY
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:412
> (sofia/external/14159927717) State DESTROY
> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:261
> sofia/external/14159927717 SOFIA DESTROY
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:60
> sofia/external/14159927717 Standard DESTROY
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:412
> (sofia/external/14159927717) State DESTROY going to sleep
> 2009-10-22 17:22:40.639760 [INFO] switch_cpp.cpp:1116 PCHangup gw:
> teliax.com hc:NORMAL_CLEARING du:9 cn:sofia/external/14159927717
>
>
>
> A BAD UUID_KILL SEQUENCE: (where the agent is hungup, unexpectedly)
>
> --bridged in: 0.005821 seconds
> 2009-10-22 17:23:26.718930 [INFO] switch_cpp.cpp:1116 AG_CBRIDGE, connected
> agent:1/1 contact:4801439/22/1/1
> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_play_say.c:1432 done playing
> file
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:340
> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXECUTE going to
> sleep
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
> CS_HIBERNATE
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE
> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:160 sofia/external/+
> 14158867717 at 199.173.100.16:5060 SOFIA HIBERNATE
> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:650
> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
> CS_HIBERNATE -> CS_RESET
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE going to
> sleep
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
> CS_RESET
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET
> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:145 sofia/external/+
> 14158867717 at 199.173.100.16:5060 SOFIA RESET
> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:637 sofia/external/+
> 14158867717 at 199.173.100.16:5060 CUSTOM RESET
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:67
> sofia/external/+14158867717 at 199.173.100.16:5060 Standard RESET
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET going to
> sleep
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:340
> (sofia/external/14159927717) State EXECUTE going to sleep
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/14159927717) Running State Change CS_HIBERNATE
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
> (sofia/external/14159927717) State HIBERNATE
> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:160
> sofia/external/14159927717 SOFIA HIBERNATE
> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:650
> (sofia/external/14159927717) State Change CS_HIBERNATE -> CS_RESET
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/14159927717 [BREAK]
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
> (sofia/external/14159927717) State HIBERNATE going to sleep
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/14159927717) Running State Change CS_RESET
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
> (sofia/external/14159927717) State RESET
> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:145
> sofia/external/14159927717 SOFIA RESET
> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:637
> sofia/external/14159927717 CUSTOM RESET
> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:642
> (sofia/external/14159927717) State Change CS_RESET -> CS_SOFT_EXECUTE
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/14159927717 [BREAK]
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
> (sofia/external/14159927717) State RESET going to sleep
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/14159927717) Running State Change CS_SOFT_EXECUTE
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:346
> (sofia/external/14159927717) State SOFT_EXECUTE
> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:442 SOFIA TRANSMIT
> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:660
> sofia/external/14159927717 CUSTOM SOFT_EXECUTE
> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_originate.c:674
> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_RESET ->
> CS_SOFT_EXECUTE
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:120 sofia/external/+
> 14158867717 at 199.173.100.16:5060 receive message [DISPLAY]
> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:120
> sofia/external/14159927717 receive message [DISPLAY]
> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:924
> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
> CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:968 sofia/external/+
> 14158867717 at 199.173.100.16:5060 receive message [BRIDGE]
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:630 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:975
> sofia/external/14159927717 receive message [BRIDGE]
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:630 Send signal
> sofia/external/14159927717 [BREAK]
> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:1019
> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
> CS_EXCHANGE_MEDIA
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:343
> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK
> 2009-10-22 17:23:26.740065 [DEBUG] switch_core_io.c:234 sofia/external/+
> 14158867717 at 199.173.100.16:5060 receive message [TRANSCODING_NECESSARY]
> 2009-10-22 17:23:27.319618 [DEBUG] switch_core_media_bug.c:393 Removing BUG
> from sofia/external/+14158867717 at 199.173.100.16:5060
> multipler is being overrided to1
> 2009-10-22 17:23:29.406414 [INFO] switch_cpp.cpp:1116 BG_AG_GEN_CALLS,
> Agents total:1 avail:0 OutChan:0 AR:0 +-Calls:0:Q Call NYNoc
> 2009-10-22 17:23:31.873352 [NOTICE] mod_commands.c:1523 Hangup
> sofia/external/14159927717 [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
> 2009-10-22 17:23:31.874411 [INFO] switch_cpp.cpp:1116
> EVENTC_HANGUP_CONTACT, Q_CONTACT fired for agent:1/1 for qcall:1
> contact:4801439/1
> active_true=1
> --contact NOT in playmsg or conference, update AGENT
> 2009-10-22 17:23:31.875475 [DEBUG] switch_channel.c:1810 Send signal
> sofia/external/14159927717 [KILL]
> 2009-10-22 17:23:31.875475 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/14159927717 [BREAK]
> 2009-10-22 17:23:31.875475 [DEBUG] switch_core_state_machine.c:437 thread
> mismatch skipping state handler.
> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:446
> sofia/external/14159927717 ending bridge by request from read function
> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
> DONE [sofia/external/14159927717]
> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:523 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:495 sofia/external/+
> 14158867717 at 199.173.100.16:5060 receive message [UNBRIDGE]
> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_session.c:630 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
> DONE [sofia/external/+14158867717 at 199.173.100.16:5060]
> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:523 Send signal
> sofia/external/14159927717 [BREAK]
> 2009-10-22 17:23:31.882153 [NOTICE] switch_ivr.c:1268 Hangup
> sofia/external/+14158867717 at 199.173.100.16:5060 [CS_EXCHANGE_MEDIA]
> [EXCHANGE_ROUTING_ERROR]
> 2009-10-22 17:23:31.882153 [DEBUG] switch_channel.c:1810 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [KILL]
> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_state_machine.c:464
> (sofia/external/+14158867717 at 199.173.100.16:5060) State HANGUP
> 2009-10-22 17:23:31.882153 [DEBUG] mod_sofia.c:344 Channel sofia/external/+
> 14158867717 at 199.173.100.16:5060 hanging up, cause: EXCHANGE_ROUTING_ERROR
> 2009-10-22 17:23:31.882153 [DEBUG] mod_sofia.c:382 Sending BYE to
> sofia/external/+14158867717 at 199.173.100.16:5060
> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_state_machine.c:46
> sofia/external/+14158867717 at 199.173.100.16:5060 Standard HANGUP, cause:
> EXCHANGE_ROUTING_ERROR
> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_state_machine.c:464
> (sofia/external/+14158867717 at 199.173.100.16:5060) State HANGUP going to
> sleep
> 2009-10-22 17:23:31.884357 [INFO] switch_cpp.cpp:1116 HH_HANGUP, agent:1/1
> uuid: f8604694-4e44-4c4c-9805-722b3e9d6b68
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:346
> (sofia/external/14159927717) State SOFT_EXECUTE going to sleep
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/14159927717) Running State Change CS_HANGUP
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:464
> (sofia/external/14159927717) State HANGUP
> 2009-10-22 17:23:31.885412 [DEBUG] mod_sofia.c:344 Channel
> sofia/external/14159927717 hanging up, cause: NORMAL_CLEARING
> 2009-10-22 17:23:31.885412 [DEBUG] mod_sofia.c:382 Sending BYE to
> sofia/external/14159927717
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:46
> sofia/external/14159927717 Standard HANGUP, cause: NORMAL_CLEARING
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:464
> (sofia/external/14159927717) State HANGUP going to sleep
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:325
> (sofia/external/14159927717) State Change CS_HANGUP -> CS_REPORTING
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/14159927717 [BREAK]
> subtracting 5 seconds for hh_call: 1095423
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/14159927717) Running State Change CS_REPORTING
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:555
> (sofia/external/14159927717) State REPORTING
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:53
> sofia/external/14159927717 Standard REPORTING, cause: NORMAL_CLEARING
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:555
> (sofia/external/14159927717) State REPORTING going to sleep
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:319
> (sofia/external/14159927717) State Change CS_REPORTING -> CS_DESTROY
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/14159927717 [BREAK]
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_session.c:1069 Session 7
> (sofia/external/14159927717) Locked, Waiting on external entities
> 2009-10-22 17:23:31.885412 [NOTICE] switch_core_session.c:1087 Session 7
> (sofia/external/14159927717) Ended
> 2009-10-22 17:23:31.885412 [NOTICE] switch_core_session.c:1089 Close
> Channel sofia/external/14159927717 [CS_DESTROY]
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:401
> (sofia/external/14159927717) Running State Change CS_DESTROY
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:412
> (sofia/external/14159927717) State DESTROY
> 2009-10-22 17:23:31.885412 [DEBUG] mod_sofia.c:261
> sofia/external/14159927717 SOFIA DESTROY
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:60
> sofia/external/14159927717 Standard DESTROY
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:412
> (sofia/external/14159927717) State DESTROY going to sleep
> 2009-10-22 17:23:31.885412 [INFO] switch_cpp.cpp:1116 PCHangup gw:
> teliax.com hc:NORMAL_CLEARING du:5 cn:sofia/external/14159927717
> 2009-10-22 17:23:31.885412 [INFO] switch_cpp.cpp:1116 EVENTC_BILLING_CEIL,
> subtracting 5 for hh_call:1095423
> 2009-10-22 17:23:31.901620 [DEBUG] switch_core_state_machine.c:494 Hangup
> Command luarun(hh_hangup.lua f8604694-4e44-4c4c-9805-722b3e9d6b68 1 1):
> +OK
>
> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:343
> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
> going to sleep
> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
> CS_HANGUP
> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:442 handler
> already called, skipping state handler.
> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:325
> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_HANGUP
> -> CS_REPORTING
> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
> CS_REPORTING
> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:555
> (sofia/external/+14158867717 at 199.173.100.16:5060) State REPORTING
> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:53
> sofia/external/+14158867717 at 199.173.100.16:5060 Standard REPORTING, cause:
> EXCHANGE_ROUTING_ERROR
> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:555
> (sofia/external/+14158867717 at 199.173.100.16:5060) State REPORTING going to
> sleep
> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:319
> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
> CS_REPORTING -> CS_DESTROY
> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_session.c:1069 Session 4
> (sofia/external/+14158867717 at 199.173.100.16:5060) Locked, Waiting on
> external entities
> 2009-10-22 17:23:31.903742 [NOTICE] switch_core_session.c:1087 Session 4
> (sofia/external/+14158867717 at 199.173.100.16:5060) Ended
> 2009-10-22 17:23:31.903742 [NOTICE] switch_core_session.c:1089 Close
> Channel sofia/external/+14158867717 at 199.173.100.16:5060 [CS_DESTROY]
> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:401
> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
> CS_DESTROY
> 2009-10-22 17:23:31.904799 [DEBUG] switch_core_state_machine.c:412
> (sofia/external/+14158867717 at 199.173.100.16:5060) State DESTROY
> 2009-10-22 17:23:31.904799 [DEBUG] mod_sofia.c:261 sofia/external/+
> 14158867717 at 199.173.100.16:5060 SOFIA DESTROY
> 2009-10-22 17:23:31.904799 [DEBUG] switch_core_state_machine.c:60
> sofia/external/+14158867717 at 199.173.100.16:5060 Standard DESTROY
> 2009-10-22 17:23:31.904799 [DEBUG] switch_core_state_machine.c:412
> (sofia/external/+14158867717 at 199.173.100.16:5060) State DESTROY going to
> sleep
>
>
>
> On Tue, Oct 20, 2009 at 1:44 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Sun, Oct 18, 2009 at 9:48 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>>
>>> The debug level logs to the console, right? The pastebin, had log level
>>> debug, sofia trace on for external and default, and sofia loglevel all 9. Is
>>> there another log enable command I'm missing? It seems loglevel all 9
>>> outputs enter and exit functions, but at least to my novice eye, it's not
>>> too obvious why freeswitch is sending a BYE to my DID provider.
>>>
>>> Evidently the console loglevel debug did not take because there isn't a
>> single DEBUG level line of output. Please try again. Just remember that
>> debug level log messages will have A LOT of information so be patient while
>> sifting through it. Of course, feel free to pastebin the output and add a
>> link to it in this email thread.
>>
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From gkuri at ieee.org  Thu Oct 22 12:29:07 2009
From: gkuri at ieee.org (Gabriel Kuri)
Date: Thu, 22 Oct 2009 12:29:07 -0700
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
In-Reply-To: <2d9149cd0910221213h20661e11pdccafdf407c79b02@mail.gmail.com>
References: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>	<191c3a030910201434q4c6e65e1h58c5f0a82addc2d3@mail.gmail.com>	<2d9149cd0910202050i36a7ea82nc7253d87e38f990a@mail.gmail.com>	<191c3a030910211154gc8412b5gd999fbedffa02b53@mail.gmail.com>	<2d9149cd0910220844v2aa73502j4b88fdbd75a0d833@mail.gmail.com>	<65d96fc80910220858u788c6146sced38e19418e681a@mail.gmail.com>	<2d9149cd0910221056y3716d8c4s708fbdb6a5563733@mail.gmail.com>	<4AE0A17D.3000301@ieee.org>
	<2d9149cd0910221213h20661e11pdccafdf407c79b02@mail.gmail.com>
Message-ID: <4AE0B283.3070008@ieee.org>

AFAIK T.38 v2 supports a max speed of 14.4 anyhow, so that's the max
speed you'll ever see in the SDP. T.38 v3 supposedly supports v.34
speeds, however no one that I've seen has implemented it yet - not sure
it's even an official standard?

Yes, in my experience, v.34 capable FAXes do not properly negotiate down
to 14.4 speeds, we've always had to disable v.34 (or whatever the option
is called on your particular make/model to force it down to 14.4 or
slower speeds). We use the Cisco/Linksys ATAs - I've heard there's
better ATAs out there that properly negotiate the FAX down to 14.4 on
the T.30 side of the connection, but never had a chance to play with them.

I'd certainly be interested to hear anyone's experience with T.38
reliability and various combinations of FAX machines and make/model of ATAs.

Gabe

Kristian Kielhofner wrote:
> Gabe,
> 
>   I don't think any of them are plus the T.38 SDP tells me the bitrate
> is 14400, certainly not V.34 speed.
> 
>   Are you saying the machine even trying to negotiate V.34 poses a problem?
> 
> On Thu, Oct 22, 2009 at 2:16 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>> Out of curiosity, is it a Super G3 (ie v.34) capable FAX? We've had
>> nothing but intermittent problems with Super G3 FAXes over T.38, unless
>> v.34 is strictly turned off on the machine.
>>
>> Gabe
>>
> 



From rupa at rupa.com  Thu Oct 22 12:31:58 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 22 Oct 2009 14:31:58 -0500
Subject: [Freeswitch-users] Address Rupa: Database for Audio Data
In-Reply-To: <BAY119-W4307982F0031E8BE965E54B3BE0@phx.gbl>
References: <BAY119-W52BD912544FAA1DE624B88B3C70@phx.gbl>
	<b39273530910130857jb7dec2eidd71e950d3902b25@mail.gmail.com>
	<BAY119-W4307982F0031E8BE965E54B3BE0@phx.gbl>
Message-ID: <b39273530910221231u93d3560nbb27036620c46e9b@mail.gmail.com>

I use the dialplan app session_record:

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session

I call that in the appropriate parts before bridging to call.  For
incoming it is just before the bridge to user/username.  For outgoing
it is before I bridge to the outgoing provider.

On Thu, Oct 22, 2009 at 11:30 AM, Pajongjit Buntaokit
<pippyduck1127 at hotmail.com> wrote:
> Hi Rupa,
>
> Thanks again for your advice.
>
> I have been searching for the method to record in the freeswitch
> documentation but I'm still not sure which command and method to make the
> record for every call automatically.
>
> Which command or method do you use?
> And to make the recording start and stop automatically every time when the
> calls is started and end, where should I insert this command?
>
> Did you use the Mod commands 'uuid_record'?
> If so, where to place this commands?
> Please show me some clues?
>
> Thank you very much!
>
> PB
>
>
>
>
>> Date: Tue, 13 Oct 2009 09:57:05 -0600
>> From: rupa at rupa.comajongjit Buntaokit
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Database for Audio Data
>>
>> What I do is record all calls and store the call with the UUID as the
>> filename. Then when the call is hung up a CDR entry is sent to my web
>> server. This CDR contains callerid and other info I might want to
>> query by. The service on the web server inserts appropriate record(s)
>> into the database. The recordings are available to the webserver.
>> When one clicks on the "listen" link, the web server serves up the
>> recording by UUID in the recording directory. I have a process that
>> periodically removes old recordings from that dir. I don't purge the
>> CDRs, though that is certainly possible.
>>
>> On Tue, Oct 13, 2009 at 7:58 AM, Pajongjit Buntaokit
>> <pippyduck1127 at hotmail.com> wrote:
>> > Hi,
>> >
>> > Does anyone know whether FreeSWITCH has a function to automatically
>> > record
>> > every call as an audio file in a server
>> > or forward them to be stored in a database with additional parameters
>> > such
>> > as caller ID, date, starting time and ending time?
>> >
>> > So that these recorded audio data can be queried and retrieved with the
>> > caller ID, date and time.
>> >
>> > Any suggestion or guidance, please advise.
>> >
>> > Thank you very much!
>> >
>> > ________________________________
>> > Hotmail: Trusted email with Microsoft?s powerful SPAM protection. Sign
>> > up
>> > now.
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> ________________________________
> Windows 7: It works the way you want. Learn more.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa


From abovebluesky at gmail.com  Thu Oct 22 12:35:16 2009
From: abovebluesky at gmail.com (Uncle Johny)
Date: Thu, 22 Oct 2009 12:35:16 -0700 (PDT)
Subject: [Freeswitch-users]  3 way conferencing question
Message-ID: <26015903.post@talk.nabble.com>


Hello gents,

I know that it is possible to make a 3-way conferencing using att_xfer and
by pressing 0.
But I'm more interested in the way of doing 3-way conferencing using mod
conference and bind_app. Saying in other words, I want to do the same thing
what 3-way conferencing using att_xfer but with mod conference (eventually
it will be more then 3 people there)

  <extension name="mystuff_enum">
    <condition field="destination_number" expression="^(.*)$">
                <action application="answer"/>
                <action application="bind_meta_app" data="7 ab s
execute_extension::conference XML features"/>
                <action application="set" data="hangup_after_bridge=true"/>
                <action application="bridge"
data="sofia/external/${destination_number}@$${dialoutserver}"/>
    </condition>
  </extension>

in features.xml :
<extension name="conference">
     <condition field="destination_number" expression="^conference$">
        <action application="set" data="continue_on_fail=true"/>
        <action application="read" data="3 10
/usr/local/freeswitch/sounds/en/us/callie/conference/8000/conf-welcome.wav
thirdperson 30000 #"/>
       <!-- I guess some logic here which will put original caller, person
whom we dialed and thirdperson(numbers which we pressed) into a conference
room with unique id -->
     </condition>
</extension>

I will be glad to hear any ideas.

-- 
View this message in context: http://www.nabble.com/3-way-conferencing-question-tp26015903p26015903.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From tculjaga at gmail.com  Thu Oct 22 12:44:02 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 22 Oct 2009 21:44:02 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>
	<65d96fc80910220753n6d2efa39l470791f75db9a640@mail.gmail.com>
	<Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>
Message-ID: <65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>

2009/10/22 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-22 16:53 +0200, Tihomir Culjaga wrote
> freeswitch-users at lists.fre...:
>
> finally i fix this rtp bug, check new wersion please.
>
>
if course i can do that, but tomorrow morning ... i'm not in the office
anymore.
BTW: can we please move the tickets to jira?


it is gonna be easier to track.

Tomorrow i will test on 1.0.4 but please lets move to trunk.

T.
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From ujjval at simplesignal.com  Thu Oct 22 12:45:12 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Thu, 22 Oct 2009 12:45:12 -0700
Subject: [Freeswitch-users] Core Dump  question!
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701E840EB@EXMBXCLUS01.citservers.local>


freeswitch at ss_freeswitch> sofia_gateway_data
Segmentation fault (core dumped)


Just ran the gateway command above w/o any parameters,,, and it core dumped..

I am sure mistakes like that happen...but I not sure if it should core dump  and shutdown.....



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From ujjval at simplesignal.com  Thu Oct 22 12:53:56 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Thu, 22 Oct 2009 12:53:56 -0700
Subject: [Freeswitch-users] Connecting to FS CLI...just hangs..
In-Reply-To: <ebbc2d550910221211k18171473p7f4d6f597f6b21d2@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84092@EXMBXCLUS01.citservers.local>
	<ebbc2d550910221211k18171473p7f4d6f597f6b21d2@mail.gmail.com>
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701E840F9@EXMBXCLUS01.citservers.local>


If I run .freeswitch& ,  get back to the ROOT prompt and then from same window type in fs_cli...it fails...hangs forever

However, if I open another new ssh session and fs_cli from the new session, it works.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jo?o Mesquita
Sent: Thursday, October 22, 2009 1:11 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Connecting to FS CLI...just hangs..

Hangs for how long? Are you sure you are not just waiting on a timeout?

JM
On Thu, Oct 22, 2009 at 4:42 PM, Ujjval Karihaloo <ujjval at simplesignal.com<mailto:ujjval at simplesignal.com>> wrote:
It just hangs....and I CTRL-C out of it.

[root at ss]# ./fs_cli -H 127.0.0.1


^C
[ERROR] libs/esl/fs_cli.c:652 main() Error Connecting [Connection Error]


Freeswitch process is running:

[root at ss bin]# ps -ef|grep free
root      8889 31039  0 12:36 pts/2    00:00:00 ./freeswitch
root      8952 31039  0 12:42 pts/2    00:00:00 grep free



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From msc at freeswitch.org  Thu Oct 22 12:58:26 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 22 Oct 2009 12:58:26 -0700
Subject: [Freeswitch-users] Core Dump question!
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71701E840EB@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701E840EB@EXMBXCLUS01.citservers.local>
Message-ID: <87f2f3b90910221258o6936000dn6eb2e22189c45435@mail.gmail.com>

What SVN rev of FS? What operating system? If you're not on the latest then
do a "make current" and get to the latest SVN and see if you can replicate
the issue.

-MC

On Thu, Oct 22, 2009 at 12:45 PM, Ujjval Karihaloo
<ujjval at simplesignal.com>wrote:

>
>
> freeswitch at ss_freeswitch> sofia_gateway_data
>
> Segmentation fault (core dumped)
>
>
>
>
>
> Just ran the gateway command above w/o any parameters,,, and it core
> dumped..
>
>
>
> I am sure mistakes like that happen?but I not sure if it should core dump
> and shutdown?..
>
>
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From ujjval at simplesignal.com  Thu Oct 22 13:00:21 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Thu, 22 Oct 2009 13:00:21 -0700
Subject: [Freeswitch-users] FS Registration Contact
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701E84100@EXMBXCLUS01.citservers.local>

Hi All,

    I have FS registered to an ITSP. The contact is showing as follows..

Contact: sip:gw+itsp at 1.1.1.1:5080;transport=udp

Itsp is the name of the SIP gatway and its IP is changed to 1.1.1.1

I want the Phone number (FromUser)to show in the contact header in the REGISTER msg going to the ITSP.

How can I do that?


Thx,
Ujjval.
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From bottleman at icf.org.ru  Thu Oct 22 13:07:09 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Fri, 23 Oct 2009 00:07:09 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>
	<65d96fc80910220753n6d2efa39l470791f75db9a640@mail.gmail.com>
	<Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>
	<65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>

On 2009-10-22 21:44 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>2009/10/22 Georgiewskiy Yuriy <bottleman at icf.org.ru>
TC>
TC>> On 2009-10-22 16:53 +0200, Tihomir Culjaga wrote
TC>> freeswitch-users at lists.fre...:
TC>>
TC>> finally i fix this rtp bug, check new wersion please.
TC>>
TC>>
TC>if course i can do that, but tomorrow morning ... i'm not in the office
TC>anymore.
TC>BTW: can we please move the tickets to jira?
TC>
TC>
TC>it is gonna be easier to track.
TC>
TC>Tomorrow i will test on 1.0.4 but please lets move to trunk.

i make it a bit later, to move tickets to jira and source to svn i 
need some time to undertand how this system is works, especially jira.

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From william.suffill at gmail.com  Thu Oct 22 13:13:15 2009
From: william.suffill at gmail.com (William Suffill)
Date: Thu, 22 Oct 2009 16:13:15 -0400
Subject: [Freeswitch-users] Connecting to FS CLI...just hangs..
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71701E840F9@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84092@EXMBXCLUS01.citservers.local>
	<ebbc2d550910221211k18171473p7f4d6f597f6b21d2@mail.gmail.com> 
	<3C04B27FC880044F8FCD735D0D952FF71701E840F9@EXMBXCLUS01.citservers.local>
Message-ID: <6b65470d0910221313i2e87d8f1h2a7b00c98c0089fa@mail.gmail.com>

If it's not windows probably be safer to just do freeswitch -nc  for
no console.  Give it a little to start up then fs_cli should be fine.
netstat -anp can also be used to see that the ports have binded
correctly.

-- W


From tculjaga at gmail.com  Thu Oct 22 13:20:00 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 22 Oct 2009 22:20:00 +0200
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
In-Reply-To: <4AE0B283.3070008@ieee.org>
References: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>
	<191c3a030910201434q4c6e65e1h58c5f0a82addc2d3@mail.gmail.com>
	<2d9149cd0910202050i36a7ea82nc7253d87e38f990a@mail.gmail.com>
	<191c3a030910211154gc8412b5gd999fbedffa02b53@mail.gmail.com>
	<2d9149cd0910220844v2aa73502j4b88fdbd75a0d833@mail.gmail.com>
	<65d96fc80910220858u788c6146sced38e19418e681a@mail.gmail.com>
	<2d9149cd0910221056y3716d8c4s708fbdb6a5563733@mail.gmail.com>
	<4AE0A17D.3000301@ieee.org>
	<2d9149cd0910221213h20661e11pdccafdf407c79b02@mail.gmail.com>
	<4AE0B283.3070008@ieee.org>
Message-ID: <65d96fc80910221320y1512c1f9xb23f734adc6b2972@mail.gmail.com>

well, my experience with T.38 is that we switched it off within our network
and now use passthrough instead :P.

the reason:

   - new fax machines that defaults to v.34 and on some you cannot disable
   SG3 at all... just imagine what pain it is to produce a how to disable V.34
   manual for every damn fax machine
   - a variety of T.30 <> T.38 mapping issues, broken GWs/ATAs that don't do
   a correct encoding/decoding expectantly when it comes to a terminating fax
   server that doesn't send a proper EOP... God save our souls...
   - our network has very low jitter - up to 40 ms , so passthrough works
   perfect
   - bandwidth become cheap :=)
   - and ... don't even remember any fax issues after shutdown T.38

i could go with the list... but i suppose you have better things to do  :P


T.

On Thu, Oct 22, 2009 at 9:29 PM, Gabriel Kuri <gkuri at ieee.org> wrote:

> AFAIK T.38 v2 supports a max speed of 14.4 anyhow, so that's the max
> speed you'll ever see in the SDP. T.38 v3 supposedly supports v.34
> speeds, however no one that I've seen has implemented it yet - not sure
> it's even an official standard?
>
> Yes, in my experience, v.34 capable FAXes do not properly negotiate down
> to 14.4 speeds, we've always had to disable v.34 (or whatever the option
> is called on your particular make/model to force it down to 14.4 or
> slower speeds). We use the Cisco/Linksys ATAs - I've heard there's
> better ATAs out there that properly negotiate the FAX down to 14.4 on
> the T.30 side of the connection, but never had a chance to play with them.
>
> I'd certainly be interested to hear anyone's experience with T.38
> reliability and various combinations of FAX machines and make/model of
> ATAs.
>
> Gabe
>
> Kristian Kielhofner wrote:
> > Gabe,
> >
> >   I don't think any of them are plus the T.38 SDP tells me the bitrate
> > is 14400, certainly not V.34 speed.
> >
> >   Are you saying the machine even trying to negotiate V.34 poses a
> problem?
> >
> > On Thu, Oct 22, 2009 at 2:16 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
> >> Out of curiosity, is it a Super G3 (ie v.34) capable FAX? We've had
> >> nothing but intermittent problems with Super G3 FAXes over T.38, unless
> >> v.34 is strictly turned off on the machine.
> >>
> >> Gabe
> >>
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ujjval at simplesignal.com  Thu Oct 22 13:34:41 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Thu, 22 Oct 2009 13:34:41 -0700
Subject: [Freeswitch-users] FS Registration Contact
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71701E84100@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84100@EXMBXCLUS01.citservers.local>
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701E8412B@EXMBXCLUS01.citservers.local>


Ok I got this one...just put ext-in-contact setting and then define the extension to be same as FromUser in my provider.xml in /conf/sip-profile/external/

  <!--/// extension for inbound calls: *optional* same as username, if blank ///-->
  <param name="extension" value="xxxxxxxxxx"/>
  <!--extra sip params to send in the contact-->
  <param name="extension-in-contact" value="true"/>

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ujjval Karihaloo
Sent: Thursday, October 22, 2009 2:00 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] FS Registration Contact

Hi All,

    I have FS registered to an ITSP. The contact is showing as follows..

Contact: sip:gw+itsp at 1.1.1.1:5080;transport=udp

Itsp is the name of the SIP gatway and its IP is changed to 1.1.1.1

I want the Phone number (FromUser)to show in the contact header in the REGISTER msg going to the ITSP.

How can I do that?


Thx,
Ujjval.
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From peder at networkoblivion.com  Thu Oct 22 13:52:03 2009
From: peder at networkoblivion.com (Peder)
Date: Thu, 22 Oct 2009 15:52:03 -0500
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
In-Reply-To: <4AE0B283.3070008@ieee.org>
References: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>	<191c3a030910201434q4c6e65e1h58c5f0a82addc2d3@mail.gmail.com>	<2d9149cd0910202050i36a7ea82nc7253d87e38f990a@mail.gmail.com>	<191c3a030910211154gc8412b5gd999fbedffa02b53@mail.gmail.com>	<2d9149cd0910220844v2aa73502j4b88fdbd75a0d833@mail.gmail.com>	<65d96fc80910220858u788c6146sced38e19418e681a@mail.gmail.com>	<2d9149cd0910221056y3716d8c4s708fbdb6a5563733@mail.gmail.com>	<4AE0A17D.3000301@ieee.org>	<2d9149cd0910221213h20661e11pdccafdf407c79b02@mail.gmail.com>
	<4AE0B283.3070008@ieee.org>
Message-ID: <029a01ca5359$823e0da0$86ba28e0$@com>

We have probably 30-40 fax machines running T.38 between Linksys SPA-3102
and Cisco gateways.  We have a pretty good success rate and we have been
doing this for probably 2-3 years.  We have a couple of them going thru an *
1.4 box and it seems to work ok.  One of my projects is to try and get them
all converted to go thru an FS box, I just haven't had time.  Our biggest
issues have always been error correction.  Once we disable that on the fax
machine, it seems to work pretty good.  As a general rule, we have found
that older cheaper machines work better.  New fancy machines are unreliable
and that could certainly be related to v.34 issues, but we haven't bothered
to get that into it to find out.

Peder

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gabriel
Kuri
Sent: Thursday, October 22, 2009 2:29 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Proxy media mode with T.38 re-invite

AFAIK T.38 v2 supports a max speed of 14.4 anyhow, so that's the max
speed you'll ever see in the SDP. T.38 v3 supposedly supports v.34
speeds, however no one that I've seen has implemented it yet - not sure
it's even an official standard?

Yes, in my experience, v.34 capable FAXes do not properly negotiate down
to 14.4 speeds, we've always had to disable v.34 (or whatever the option
is called on your particular make/model to force it down to 14.4 or
slower speeds). We use the Cisco/Linksys ATAs - I've heard there's
better ATAs out there that properly negotiate the FAX down to 14.4 on
the T.30 side of the connection, but never had a chance to play with them.

I'd certainly be interested to hear anyone's experience with T.38
reliability and various combinations of FAX machines and make/model of ATAs.

Gabe

Kristian Kielhofner wrote:
> Gabe,
> 
>   I don't think any of them are plus the T.38 SDP tells me the bitrate
> is 14400, certainly not V.34 speed.
> 
>   Are you saying the machine even trying to negotiate V.34 poses a
problem?
> 
> On Thu, Oct 22, 2009 at 2:16 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>> Out of curiosity, is it a Super G3 (ie v.34) capable FAX? We've had
>> nothing but intermittent problems with Super G3 FAXes over T.38, unless
>> v.34 is strictly turned off on the machine.
>>
>> Gabe
>>
> 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From mike at jerris.com  Thu Oct 22 13:53:11 2009
From: mike at jerris.com (Michael Jerris)
Date: Thu, 22 Oct 2009 16:53:11 -0400
Subject: [Freeswitch-users] Core Dump question!
In-Reply-To: <87f2f3b90910221258o6936000dn6eb2e22189c45435@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701E840EB@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221258o6936000dn6eb2e22189c45435@mail.gmail.com>
Message-ID: <DF11CCC6-5C89-4489-908C-92A0FCC94AE4@jerris.com>

I did test this on trunk and it seems to work right:

freeswitch at default> sofia_gateway_data
-ERR Parameter missing

Mike

On Oct 22, 2009, at 3:58 PM, Michael Collins wrote:

> What SVN rev of FS? What operating system? If you're not on the  
> latest then do a "make current" and get to the latest SVN and see if  
> you can replicate the issue.
>
> -MC
>
> On Thu, Oct 22, 2009 at 12:45 PM, Ujjval Karihaloo <ujjval at simplesignal.com 
> > wrote:
>
> freeswitch at ss_freeswitch> sofia_gateway_data
>
> Segmentation fault (core dumped)
>
>
>
> Just ran the gateway command above w/o any parameters,,, and it core  
> dumped..
>
>
> I am sure mistakes like that happen?but I not sure if it should core  
> dump  and shutdown?..
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From ujjval at simplesignal.com  Thu Oct 22 17:04:39 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Thu, 22 Oct 2009 17:04:39 -0700
Subject: [Freeswitch-users] Core Dump question!
In-Reply-To: <DF11CCC6-5C89-4489-908C-92A0FCC94AE4@jerris.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701E840EB@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221258o6936000dn6eb2e22189c45435@mail.gmail.com>
	<DF11CCC6-5C89-4489-908C-92A0FCC94AE4@jerris.com>
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701E841E8@EXMBXCLUS01.citservers.local>

I do have the core dump, should I open a ticket.
I am running latest Freeswitch 1.0.4 and had done a make current just before it happened.

Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690

SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO  80112
[cid:image001.jpg at 01CA5342.1F8A8880]<http://www.simplesignal.com/>

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Thursday, October 22, 2009 2:53 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Core Dump question!

I did test this on trunk and it seems to work right:

freeswitch at default> sofia_gateway_data
-ERR Parameter missing

Mike

On Oct 22, 2009, at 3:58 PM, Michael Collins wrote:


What SVN rev of FS? What operating system? If you're not on the latest then do a "make current" and get to the latest SVN and see if you can replicate the issue.

-MC
On Thu, Oct 22, 2009 at 12:45 PM, Ujjval Karihaloo <ujjval at simplesignal.com<mailto:ujjval at simplesignal.com>> wrote:

freeswitch at ss_freeswitch> sofia_gateway_data
Segmentation fault (core dumped)


Just ran the gateway command above w/o any parameters,,, and it core dumped..

I am sure mistakes like that happen...but I not sure if it should core dump  and shutdown.....


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From msc at freeswitch.org  Thu Oct 22 17:25:34 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 22 Oct 2009 17:25:34 -0700
Subject: [Freeswitch-users] Core Dump question!
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71701E841E8@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701E840EB@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221258o6936000dn6eb2e22189c45435@mail.gmail.com>
	<DF11CCC6-5C89-4489-908C-92A0FCC94AE4@jerris.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E841E8@EXMBXCLUS01.citservers.local>
Message-ID: <87f2f3b90910221725g6233ec20w49a73864400426d6@mail.gmail.com>

Yes, if this is latest SVN (after a make current) then open a jira.
-MC

On Thu, Oct 22, 2009 at 5:04 PM, Ujjval Karihaloo
<ujjval at simplesignal.com>wrote:

>  I do have the core dump, should I open a ticket.
>
> I am running latest Freeswitch 1.0.4 and had done a make current just
> before it happened.
>
>
>
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From ujjval at simplesignal.com  Thu Oct 22 19:52:12 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Thu, 22 Oct 2009 19:52:12 -0700
Subject: [Freeswitch-users] Core Dump question!
In-Reply-To: <87f2f3b90910221725g6233ec20w49a73864400426d6@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701E840EB@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221258o6936000dn6eb2e22189c45435@mail.gmail.com>
	<DF11CCC6-5C89-4489-908C-92A0FCC94AE4@jerris.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E841E8@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221725g6233ec20w49a73864400426d6@mail.gmail.com>
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701E84200@EXMBXCLUS01.citservers.local>

How do I tell if it's the latest...I downloaded is yesterday..and installed it from freeswitch.org

Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690

SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO  80112
[cid:image001.jpg at 01CA5359.87CB41C0]<http://www.simplesignal.com/>

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Thursday, October 22, 2009 6:26 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Core Dump question!

Yes, if this is latest SVN (after a make current) then open a jira.
-MC
On Thu, Oct 22, 2009 at 5:04 PM, Ujjval Karihaloo <ujjval at simplesignal.com<mailto:ujjval at simplesignal.com>> wrote:
I do have the core dump, should I open a ticket.
I am running latest Freeswitch 1.0.4 and had done a make current just before it happened.


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From ujjval at simplesignal.com  Thu Oct 22 20:01:57 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Thu, 22 Oct 2009 20:01:57 -0700
Subject: [Freeswitch-users] Setting up Conference with Moderator
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local>

Hi

I have the Basic Conferencing working. Here is my Dial Plan.

 I want to be able to setup a Moderator PIN different from other participants, when I add the moderator flag it logs me in directly w/o asking for a PIN..
action application="conference" data="conference.conf at wideband+flags{moderator}+159753<mailto:conference.conf at wideband+flags%7bmoderator%7d+159753>"/>

DialPlan is below for the normal user, and it asks for the PIN with below settings.

       Ujj Inbound from SS - start
    -->
    <extension name="simplesignal">   <!-- your provider or any name you'd like to call it -->
     <condition field="destination_number" expression="2142349127">  <!-- your DID for this gateway-->
        <action application="conference" data="conference.conf at wideband+159753<mailto:conference.conf at wideband+159753>"/>
     </condition>
</extension>

    <!--
       Ujj Inbound from SS - end




And I am using the existing conference.conf.xml file in the auto_loads directory.


Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690

SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO  80112
[cid:image001.jpg at 01CA535A.E43E9870]<http://www.simplesignal.com/>

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From diego.viola at gmail.com  Thu Oct 22 20:04:24 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 23 Oct 2009 03:04:24 +0000
Subject: [Freeswitch-users] Core Dump question!
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71701E84200@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701E840EB@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221258o6936000dn6eb2e22189c45435@mail.gmail.com>
	<DF11CCC6-5C89-4489-908C-92A0FCC94AE4@jerris.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E841E8@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221725g6233ec20w49a73864400426d6@mail.gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E84200@EXMBXCLUS01.citservers.local>
Message-ID: <86a32abc0910222004y46bf08b6jda75b6b8f1979420@mail.gmail.com>

Type "version" on the CLI.

On Fri, Oct 23, 2009 at 2:52 AM, Ujjval Karihaloo
<ujjval at simplesignal.com>wrote:

>  How do I tell if it?s the latest?I downloaded is yesterday..and installed
> it from freeswitch.org
>
>
>
> Ujjval Karihaloo
>
> VP Voice Engineering
>
> IP Phone: +13032428610
>
> E-Fax: +17202391690
>
>
>
> SimpleSignal Inc.
>
> 88 Inverness Circle East
>
> Suite K105
>
> Englewood, CO  80112
>
> [image: bvoip] <http://www.simplesignal.com/>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, October 22, 2009 6:26 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Core Dump question!
>
>
>
> Yes, if this is latest SVN (after a make current) then open a jira.
> -MC
>
> On Thu, Oct 22, 2009 at 5:04 PM, Ujjval Karihaloo <ujjval at simplesignal.com>
> wrote:
>
> I do have the core dump, should I open a ticket.
>
> I am running latest Freeswitch 1.0.4 and had done a make current just
> before it happened.
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mcampbellsmith at gmail.com  Thu Oct 22 20:20:21 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Fri, 23 Oct 2009 14:20:21 +1100
Subject: [Freeswitch-users] Core Dump question!
In-Reply-To: <86a32abc0910222004y46bf08b6jda75b6b8f1979420@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701E840EB@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221258o6936000dn6eb2e22189c45435@mail.gmail.com>
	<DF11CCC6-5C89-4489-908C-92A0FCC94AE4@jerris.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E841E8@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221725g6233ec20w49a73864400426d6@mail.gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E84200@EXMBXCLUS01.citservers.local>
	<86a32abc0910222004y46bf08b6jda75b6b8f1979420@mail.gmail.com>
Message-ID: <33c87fa30910222020s4bd03f5flec7bc7df3ca6876c@mail.gmail.com>

If you downloaded a precomipled version from the web, you do not (probably)
have the latest version.

As Diego said, type 'version' (or press F12) on the cli and compare to the
latest trunk at http://svn.freeswitch.org/svn/freeswitch/trunk/


On Fri, Oct 23, 2009 at 2:04 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Type "version" on the CLI.
>
> On Fri, Oct 23, 2009 at 2:52 AM, Ujjval Karihaloo <ujjval at simplesignal.com
> > wrote:
>
>>  How do I tell if it?s the latest?I downloaded is yesterday..and
>> installed it from freeswitch.org
>>
>>
>>
>> Ujjval Karihaloo
>>
>> VP Voice Engineering
>>
>> IP Phone: +13032428610
>>
>> E-Fax: +17202391690
>>
>>
>>
>> SimpleSignal Inc.
>>
>> 88 Inverness Circle East
>>
>> Suite K105
>>
>> Englewood, CO  80112
>>
>> [image: bvoip] <http://www.simplesignal.com/>
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
>> Collins
>> *Sent:* Thursday, October 22, 2009 6:26 PM
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* Re: [Freeswitch-users] Core Dump question!
>>
>>
>>
>> Yes, if this is latest SVN (after a make current) then open a jira.
>> -MC
>>
>> On Thu, Oct 22, 2009 at 5:04 PM, Ujjval Karihaloo <
>> ujjval at simplesignal.com> wrote:
>>
>> I do have the core dump, should I open a ticket.
>>
>> I am running latest Freeswitch 1.0.4 and had done a make current just
>> before it happened.
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ujjval at simplesignal.com  Thu Oct 22 20:25:24 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Thu, 22 Oct 2009 20:25:24 -0700
Subject: [Freeswitch-users] Core Dump question!
In-Reply-To: <86a32abc0910222004y46bf08b6jda75b6b8f1979420@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701E840EB@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221258o6936000dn6eb2e22189c45435@mail.gmail.com>
	<DF11CCC6-5C89-4489-908C-92A0FCC94AE4@jerris.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E841E8@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221725g6233ec20w49a73864400426d6@mail.gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E84200@EXMBXCLUS01.citservers.local>
	<86a32abc0910222004y46bf08b6jda75b6b8f1979420@mail.gmail.com>
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701E84203@EXMBXCLUS01.citservers.local>

freeswitch at internal> version
FreeSWITCH Version 1.0.4 (exported)

freeswitch at internal>

Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690

SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO  80112
[cid:image001.jpg at 01CA535E.2ACD2880]<http://www.simplesignal.com/>

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Diego Viola
Sent: Thursday, October 22, 2009 9:04 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Core Dump question!

Type "version" on the CLI.
On Fri, Oct 23, 2009 at 2:52 AM, Ujjval Karihaloo <ujjval at simplesignal.com<mailto:ujjval at simplesignal.com>> wrote:
How do I tell if it's the latest...I downloaded is yesterday..and installed it from freeswitch.org<http://freeswitch.org>

Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690

SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO  80112
[cid:image001.jpg at 01CA535E.2ACD2880]<http://www.simplesignal.com/>

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Collins
Sent: Thursday, October 22, 2009 6:26 PM

To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Core Dump question!

Yes, if this is latest SVN (after a make current) then open a jira.
-MC
On Thu, Oct 22, 2009 at 5:04 PM, Ujjval Karihaloo <ujjval at simplesignal.com<mailto:ujjval at simplesignal.com>> wrote:
I do have the core dump, should I open a ticket.
I am running latest Freeswitch 1.0.4 and had done a make current just before it happened.



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From mattdfong at gmail.com  Thu Oct 22 20:19:07 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Fri, 23 Oct 2009 10:19:07 +0700
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
In-Reply-To: <191c3a030910221226j167e0adw5f631f759cd7cfa0@mail.gmail.com>
References: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
	<177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>
	<4256bf830908041151l55980e3fifb85c75b87535426@mail.gmail.com>
	<3D0B855A-5B2C-4C6F-9306-64B0ECE97091@jerris.com>
	<4256bf830910180725y72539d98gb138934afc18d58b@mail.gmail.com>
	<A5ECDCD9-1A12-4A31-89F0-E6F4C79FE9E9@jerris.com>
	<4256bf830910182148mf7f2bf9vdcccd687e0caa1e7@mail.gmail.com>
	<87f2f3b90910191144q25750bayfd7a7448e82e78ff@mail.gmail.com>
	<4256bf830910221042l68ea769cg532bbd339bae0790@mail.gmail.com>
	<191c3a030910221226j167e0adw5f631f759cd7cfa0@mail.gmail.com>
Message-ID: <4256bf830910222019x4bd7929eoadcb757294cc37d2@mail.gmail.com>

Hi Anthm,
Thanks for the response. I'll use pastebin from now on.

How would I go about fixing this so my calls do not get hung-up?

--matt

On Fri, Oct 23, 2009 at 2:26 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> 1) please do not flood emails with traces reference pastebins
> 2) the error is a routing loop being prevented so you are looping the call
> into your same box too many times.
>
> You could have figured this out your self by looking at one of the lines of
> code where it reported the hangup.
>
>  if (!switch_strlen_zero(forwardvar)) {
>         forwardval = atoi(forwardvar) - 1;
>     }
>     if (forwardval <= 0) {
>         switch_channel_hangup(channel,
> SWITCH_CAUSE_EXCHANGE_ROUTING_ERROR);
>         return SWITCH_STATUS_FALSE;
>
>     }
>
>
> On Thu, Oct 22, 2009 at 12:42 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>
>> I figured out how to turn debug mode on. For some reason F8 didn't work,
>> so I just set the default debug mode to DEBUG in switch.conf.xml
>>
>> http://pastebin.freeswitch.org/10805
>> <http://pastebin.freeswitch.org/10805>So my program does more or less the
>> same sequence for an Agent every time when connected to another party. They
>> uuid_kill the bridged parties uuid, and then are transfered to an extension
>> that park's the agent (using the transfer_after_bridge variable -- in my
>> dialplan extn 1998). In some instances the agent is then transfered to
>> another extension, played a wav file, and then park'ed again
>>
>> Sometimes, FreeSWITCH unexpectidly hangsup the Agent channel citing an
>> EXCHANGE_ROUTING_ERROR. It seems (to me to happen at random times)...and
>> it's more prevelent with some did providers than others. With
>> didforsale.org it happens about one ever 5 bridges. With icall.com it
>> only happens about once every 30 bridges. With IPKall it happens rarely. I
>> can't figure it out, and it's driving me crazy :)
>>
>>
>> A GOOD UUID_KILL SEQUENCE:
>> 2009-10-22 17:22:31.741845 [DEBUG] switch_ivr_play_say.c:1432 done playing
>> file
>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:340
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXECUTE going to
>> sleep
>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>> CS_HIBERNATE
>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:355
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE
>> 2009-10-22 17:22:31.741845 [DEBUG] mod_sofia.c:160 sofia/external/+
>> 14158867717 at 199.173.100.16:5060 SOFIA HIBERNATE
>> 2009-10-22 17:22:31.741845 [DEBUG] switch_ivr_bridge.c:650
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>> CS_HIBERNATE -> CS_RESET
>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:355
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE going
>> to sleep
>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>> CS_RESET
>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:336
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET
>> 2009-10-22 17:22:31.741845 [DEBUG] mod_sofia.c:145 sofia/external/+
>> 14158867717 at 199.173.100.16:5060 SOFIA RESET
>> 2009-10-22 17:22:31.741845 [DEBUG] switch_ivr_bridge.c:637
>> sofia/external/+14158867717 at 199.173.100.16:5060 CUSTOM RESET
>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:67
>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard RESET
>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:336
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET going to
>> sleep
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:340
>> (sofia/external/14159927717) State EXECUTE going to sleep
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/14159927717) Running State Change CS_HIBERNATE
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:355
>> (sofia/external/14159927717) State HIBERNATE
>> 2009-10-22 17:22:31.748086 [DEBUG] mod_sofia.c:160
>> sofia/external/14159927717 SOFIA HIBERNATE
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:650
>> (sofia/external/14159927717) State Change CS_HIBERNATE -> CS_RESET
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/14159927717 [BREAK]
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:355
>> (sofia/external/14159927717) State HIBERNATE going to sleep
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/14159927717) Running State Change CS_RESET
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:336
>> (sofia/external/14159927717) State RESET
>> 2009-10-22 17:22:31.748086 [DEBUG] mod_sofia.c:145
>> sofia/external/14159927717 SOFIA RESET
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:637
>> sofia/external/14159927717 CUSTOM RESET
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:642
>> (sofia/external/14159927717) State Change CS_RESET -> CS_SOFT_EXECUTE
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/14159927717 [BREAK]
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:336
>> (sofia/external/14159927717) State RESET going to sleep
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/14159927717) Running State Change CS_SOFT_EXECUTE
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:346
>> (sofia/external/14159927717) State SOFT_EXECUTE
>> 2009-10-22 17:22:31.748086 [DEBUG] mod_sofia.c:442 SOFIA TRANSMIT
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:660
>> sofia/external/14159927717 CUSTOM SOFT_EXECUTE
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_originate.c:674
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_RESET
>> -> CS_SOFT_EXECUTE
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:120
>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message [DISPLAY]
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:120
>> sofia/external/14159927717 receive message [DISPLAY]
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:924
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>> CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:968
>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message [BRIDGE]
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:630 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:975
>> sofia/external/14159927717 receive message [BRIDGE]
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:630 Send signal
>> sofia/external/14159927717 [BREAK]
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:1019
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:22:31.749553 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>> CS_EXCHANGE_MEDIA
>> 2009-10-22 17:22:31.749553 [DEBUG] switch_core_state_machine.c:343
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
>> 2009-10-22 17:22:31.749553 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK
>> multipler is being overrided to1
>> 2009-10-22 17:22:34.300662 [INFO] switch_cpp.cpp:1116 BG_AG_GEN_CALLS,
>> Agents total:1 avail:0 OutChan:0 AR:0 +-Calls:0:Q Call NYNoc
>> multipler is being overrided to1
>> 2009-10-22 17:22:39.303041 [INFO] switch_cpp.cpp:1116 BG_AG_GEN_CALLS,
>> Agents total:1 avail:0 OutChan:0 AR:0 +-Calls:0:Q Call NYNoc
>> 2009-10-22 17:22:40.626379 [NOTICE] sofia.c:328 Hangup
>> sofia/external/14159927717 [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
>> 2009-10-22 17:22:40.626379 [DEBUG] switch_channel.c:1810 Send signal
>> sofia/external/14159927717 [KILL]
>> 2009-10-22 17:22:40.626379 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/14159927717 [BREAK]
>> active_true=1
>> --contact NOT in playmsg or conference, update AGENT
>> 2009-10-22 17:22:40.626379 [DEBUG] switch_core_state_machine.c:437 thread
>> mismatch skipping state handler.
>> 2009-10-22 17:22:40.626379 [INFO] switch_cpp.cpp:1116
>> EVENTC_HANGUP_CONTACT, Q_CONTACT fired for agent:1/1 for qcall:1
>> contact:4801442/4
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:440
>> sofia/external/14159927717 ending bridge by request from write function
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:495
>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message
>> [UNBRIDGE]
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:630 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
>> DONE [sofia/external/+14158867717 at 199.173.100.16:5060]
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:523 Send signal
>> sofia/external/14159927717 [BREAK]
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr.c:1361 (sofia/external/+
>> 14158867717 at 199.173.100.16:5060) State Change CS_EXCHANGE_MEDIA ->
>> CS_ROUTING
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr.c:1365 sofia/external/+
>> 14158867717 at 199.173.100.16:5060 receive message [TRANSFER]
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:630 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:22:40.639760 [NOTICE] switch_ivr.c:1367 Transfer
>> sofia/external/+14158867717 at 199.173.100.16:5060 to xml[1998 at default]
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:343
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
>> going to sleep
>>  2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>> CS_ROUTING
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:333
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State ROUTING
>> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:130 sofia/external/+
>> 14158867717 at 199.173.100.16:5060 SOFIA ROUTING
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:78
>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard ROUTING
>> 2009-10-22 17:22:40.639760 [INFO] mod_dialplan_xml.c:391 Processing
>> +14158867717->1998 in context default
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->tod_example] continue=true
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (PASS)
>> [tod_example] ${strftime(%w)}(4) =~ /^([1-5])$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (PASS)
>> [tod_example] ${strftime(%H%M)}(1722) =~ /^((09|1[0-7])[0-5][0-9]|1800)$/
>> break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Action
>> set(open=true)
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->ivr_demo] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [ivr_demo] destination_number(1998) =~ /^5000$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1975] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1975] destination_number(1998) =~ /^1975$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1971] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1971] destination_number(1998) =~ /^1971$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1972] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1972] destination_number(1998) =~ /^1972$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1920] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1920] destination_number(1998) =~ /^1920$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1981] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1981] destination_number(1998) =~ /^1981$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1982] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1982] destination_number(1998) =~ /^1982$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1481] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1481] destination_number(1998) =~ /^1481$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1989] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1989] destination_number(1998) =~ /^1989$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1990] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1990] destination_number(1998) =~ /^1990$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1991] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1991] destination_number(1998) =~ /^1991$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1992] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1992] destination_number(1998) =~ /^1992$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1993] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1993] destination_number(1998) =~ /^1993$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1994] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1994] destination_number(1998) =~ /^1994$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1995] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1995] destination_number(1998) =~ /^1995$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1996] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1996] destination_number(1998) =~ /^1996$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1997] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1997] destination_number(1998) =~ /^1997$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1998] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (PASS)
>> [1998] destination_number(1998) =~ /^1998$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Action
>> set(transfer_after_bridge=1998)
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Action park()
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:114
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_ROUTING
>> -> CS_EXECUTE
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:333
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State ROUTING going to
>> sleep
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>> CS_EXECUTE
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:340
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXECUTE
>> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:173 sofia/external/+
>> 14158867717 at 199.173.100.16:5060 SOFIA EXECUTE
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:151
>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard EXECUTE
>> EXECUTE sofia/external/+14158867717 at 199.173.100.16:5060 set(open=true)
>> 2009-10-22 17:22:40.639760 [DEBUG] mod_dptools.c:752 sofia/external/+
>> 14158867717 at 199.173.100.16:5060 SET [open]=[true]
>> EXECUTE sofia/external/+14158867717 at 199.173.100.16:5060set(transfer_after_bridge=1998)
>> 2009-10-22 17:22:40.639760 [DEBUG] mod_dptools.c:752 sofia/external/+
>> 14158867717 at 199.173.100.16:5060 SET [transfer_after_bridge]=[1998]
>> EXECUTE sofia/external/+14158867717 at 199.173.100.16:5060 park()
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:446
>> sofia/external/14159927717 ending bridge by request from read function
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
>> DONE [sofia/external/14159927717]
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:523 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> subtracting 9 seconds for hh_call: 1095422
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:346
>> (sofia/external/14159927717) State SOFT_EXECUTE going to sleep
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/14159927717) Running State Change CS_HANGUP
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:464
>> (sofia/external/14159927717) State HANGUP
>> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:344 Channel
>> sofia/external/14159927717 hanging up, cause: NORMAL_CLEARING
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:46
>> sofia/external/14159927717 Standard HANGUP, cause: NORMAL_CLEARING
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:464
>> (sofia/external/14159927717) State HANGUP going to sleep
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:325
>> (sofia/external/14159927717) State Change CS_HANGUP -> CS_REPORTING
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/14159927717 [BREAK]
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/14159927717) Running State Change CS_REPORTING
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:555
>> (sofia/external/14159927717) State REPORTING
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:53
>> sofia/external/14159927717 Standard REPORTING, cause: NORMAL_CLEARING
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:555
>> (sofia/external/14159927717) State REPORTING going to sleep
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:319
>> (sofia/external/14159927717) State Change CS_REPORTING -> CS_DESTROY
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/14159927717 [BREAK]
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:1069 Session 6
>> (sofia/external/14159927717) Locked, Waiting on external entities
>> 2009-10-22 17:22:40.639760 [NOTICE] switch_core_session.c:1087 Session 6
>> (sofia/external/14159927717) Ended
>> 2009-10-22 17:22:40.639760 [NOTICE] switch_core_session.c:1089 Close
>> Channel sofia/external/14159927717 [CS_DESTROY]
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:401
>> (sofia/external/14159927717) Running State Change CS_DESTROY
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:412
>> (sofia/external/14159927717) State DESTROY
>> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:261
>> sofia/external/14159927717 SOFIA DESTROY
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:60
>> sofia/external/14159927717 Standard DESTROY
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:412
>> (sofia/external/14159927717) State DESTROY going to sleep
>> 2009-10-22 17:22:40.639760 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>> teliax.com hc:NORMAL_CLEARING du:9 cn:sofia/external/14159927717
>>
>>
>>
>> A BAD UUID_KILL SEQUENCE: (where the agent is hungup, unexpectedly)
>>
>> --bridged in: 0.005821 seconds
>> 2009-10-22 17:23:26.718930 [INFO] switch_cpp.cpp:1116 AG_CBRIDGE,
>> connected agent:1/1 contact:4801439/22/1/1
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_play_say.c:1432 done playing
>> file
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:340
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXECUTE going to
>> sleep
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>> CS_HIBERNATE
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE
>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:160 sofia/external/+
>> 14158867717 at 199.173.100.16:5060 SOFIA HIBERNATE
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:650
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>> CS_HIBERNATE -> CS_RESET
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE going
>> to sleep
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>> CS_RESET
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET
>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:145 sofia/external/+
>> 14158867717 at 199.173.100.16:5060 SOFIA RESET
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:637
>> sofia/external/+14158867717 at 199.173.100.16:5060 CUSTOM RESET
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:67
>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard RESET
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET going to
>> sleep
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:340
>> (sofia/external/14159927717) State EXECUTE going to sleep
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/14159927717) Running State Change CS_HIBERNATE
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
>> (sofia/external/14159927717) State HIBERNATE
>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:160
>> sofia/external/14159927717 SOFIA HIBERNATE
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:650
>> (sofia/external/14159927717) State Change CS_HIBERNATE -> CS_RESET
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/14159927717 [BREAK]
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
>> (sofia/external/14159927717) State HIBERNATE going to sleep
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/14159927717) Running State Change CS_RESET
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
>> (sofia/external/14159927717) State RESET
>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:145
>> sofia/external/14159927717 SOFIA RESET
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:637
>> sofia/external/14159927717 CUSTOM RESET
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:642
>> (sofia/external/14159927717) State Change CS_RESET -> CS_SOFT_EXECUTE
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/14159927717 [BREAK]
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
>> (sofia/external/14159927717) State RESET going to sleep
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/14159927717) Running State Change CS_SOFT_EXECUTE
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:346
>> (sofia/external/14159927717) State SOFT_EXECUTE
>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:442 SOFIA TRANSMIT
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:660
>> sofia/external/14159927717 CUSTOM SOFT_EXECUTE
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_originate.c:674
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_RESET
>> -> CS_SOFT_EXECUTE
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:120
>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message [DISPLAY]
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:120
>> sofia/external/14159927717 receive message [DISPLAY]
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:924
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>> CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:968
>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message [BRIDGE]
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:630 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:975
>> sofia/external/14159927717 receive message [BRIDGE]
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:630 Send signal
>> sofia/external/14159927717 [BREAK]
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:1019
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>> CS_EXCHANGE_MEDIA
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:343
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK
>> 2009-10-22 17:23:26.740065 [DEBUG] switch_core_io.c:234 sofia/external/+
>> 14158867717 at 199.173.100.16:5060 receive message [TRANSCODING_NECESSARY]
>> 2009-10-22 17:23:27.319618 [DEBUG] switch_core_media_bug.c:393 Removing
>> BUG from sofia/external/+14158867717 at 199.173.100.16:5060
>> multipler is being overrided to1
>> 2009-10-22 17:23:29.406414 [INFO] switch_cpp.cpp:1116 BG_AG_GEN_CALLS,
>> Agents total:1 avail:0 OutChan:0 AR:0 +-Calls:0:Q Call NYNoc
>> 2009-10-22 17:23:31.873352 [NOTICE] mod_commands.c:1523 Hangup
>> sofia/external/14159927717 [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
>> 2009-10-22 17:23:31.874411 [INFO] switch_cpp.cpp:1116
>> EVENTC_HANGUP_CONTACT, Q_CONTACT fired for agent:1/1 for qcall:1
>> contact:4801439/1
>> active_true=1
>> --contact NOT in playmsg or conference, update AGENT
>> 2009-10-22 17:23:31.875475 [DEBUG] switch_channel.c:1810 Send signal
>> sofia/external/14159927717 [KILL]
>> 2009-10-22 17:23:31.875475 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/14159927717 [BREAK]
>> 2009-10-22 17:23:31.875475 [DEBUG] switch_core_state_machine.c:437 thread
>> mismatch skipping state handler.
>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:446
>> sofia/external/14159927717 ending bridge by request from read function
>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
>> DONE [sofia/external/14159927717]
>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:523 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:495
>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message
>> [UNBRIDGE]
>> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_session.c:630 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
>> DONE [sofia/external/+14158867717 at 199.173.100.16:5060]
>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:523 Send signal
>> sofia/external/14159927717 [BREAK]
>> 2009-10-22 17:23:31.882153 [NOTICE] switch_ivr.c:1268 Hangup
>> sofia/external/+14158867717 at 199.173.100.16:5060 [CS_EXCHANGE_MEDIA]
>> [EXCHANGE_ROUTING_ERROR]
>>  2009-10-22 17:23:31.882153 [DEBUG] switch_channel.c:1810 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [KILL]
>> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_state_machine.c:464
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HANGUP
>> 2009-10-22 17:23:31.882153 [DEBUG] mod_sofia.c:344 Channel
>> sofia/external/+14158867717 at 199.173.100.16:5060 hanging up, cause:
>> EXCHANGE_ROUTING_ERROR
>> 2009-10-22 17:23:31.882153 [DEBUG] mod_sofia.c:382 Sending BYE to
>> sofia/external/+14158867717 at 199.173.100.16:5060
>> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_state_machine.c:46
>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard HANGUP, cause:
>> EXCHANGE_ROUTING_ERROR
>> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_state_machine.c:464
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HANGUP going to
>> sleep
>> 2009-10-22 17:23:31.884357 [INFO] switch_cpp.cpp:1116 HH_HANGUP, agent:1/1
>> uuid: f8604694-4e44-4c4c-9805-722b3e9d6b68
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:346
>> (sofia/external/14159927717) State SOFT_EXECUTE going to sleep
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/14159927717) Running State Change CS_HANGUP
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:464
>> (sofia/external/14159927717) State HANGUP
>> 2009-10-22 17:23:31.885412 [DEBUG] mod_sofia.c:344 Channel
>> sofia/external/14159927717 hanging up, cause: NORMAL_CLEARING
>> 2009-10-22 17:23:31.885412 [DEBUG] mod_sofia.c:382 Sending BYE to
>> sofia/external/14159927717
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:46
>> sofia/external/14159927717 Standard HANGUP, cause: NORMAL_CLEARING
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:464
>> (sofia/external/14159927717) State HANGUP going to sleep
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:325
>> (sofia/external/14159927717) State Change CS_HANGUP -> CS_REPORTING
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/14159927717 [BREAK]
>> subtracting 5 seconds for hh_call: 1095423
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/14159927717) Running State Change CS_REPORTING
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:555
>> (sofia/external/14159927717) State REPORTING
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:53
>> sofia/external/14159927717 Standard REPORTING, cause: NORMAL_CLEARING
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:555
>> (sofia/external/14159927717) State REPORTING going to sleep
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:319
>> (sofia/external/14159927717) State Change CS_REPORTING -> CS_DESTROY
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/14159927717 [BREAK]
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_session.c:1069 Session 7
>> (sofia/external/14159927717) Locked, Waiting on external entities
>> 2009-10-22 17:23:31.885412 [NOTICE] switch_core_session.c:1087 Session 7
>> (sofia/external/14159927717) Ended
>> 2009-10-22 17:23:31.885412 [NOTICE] switch_core_session.c:1089 Close
>> Channel sofia/external/14159927717 [CS_DESTROY]
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:401
>> (sofia/external/14159927717) Running State Change CS_DESTROY
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:412
>> (sofia/external/14159927717) State DESTROY
>> 2009-10-22 17:23:31.885412 [DEBUG] mod_sofia.c:261
>> sofia/external/14159927717 SOFIA DESTROY
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:60
>> sofia/external/14159927717 Standard DESTROY
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:412
>> (sofia/external/14159927717) State DESTROY going to sleep
>> 2009-10-22 17:23:31.885412 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>> teliax.com hc:NORMAL_CLEARING du:5 cn:sofia/external/14159927717
>> 2009-10-22 17:23:31.885412 [INFO] switch_cpp.cpp:1116 EVENTC_BILLING_CEIL,
>> subtracting 5 for hh_call:1095423
>> 2009-10-22 17:23:31.901620 [DEBUG] switch_core_state_machine.c:494 Hangup
>> Command luarun(hh_hangup.lua f8604694-4e44-4c4c-9805-722b3e9d6b68 1 1):
>> +OK
>>
>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:343
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
>> going to sleep
>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>> CS_HANGUP
>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:442 handler
>> already called, skipping state handler.
>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:325
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_HANGUP
>> -> CS_REPORTING
>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>> CS_REPORTING
>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:555
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State REPORTING
>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:53
>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard REPORTING,
>> cause: EXCHANGE_ROUTING_ERROR
>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:555
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State REPORTING going
>> to sleep
>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:319
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>> CS_REPORTING -> CS_DESTROY
>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_session.c:1069 Session 4
>> (sofia/external/+14158867717 at 199.173.100.16:5060) Locked, Waiting on
>> external entities
>> 2009-10-22 17:23:31.903742 [NOTICE] switch_core_session.c:1087 Session 4
>> (sofia/external/+14158867717 at 199.173.100.16:5060) Ended
>> 2009-10-22 17:23:31.903742 [NOTICE] switch_core_session.c:1089 Close
>> Channel sofia/external/+14158867717 at 199.173.100.16:5060 [CS_DESTROY]
>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:401
>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>> CS_DESTROY
>> 2009-10-22 17:23:31.904799 [DEBUG] switch_core_state_machine.c:412
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State DESTROY
>> 2009-10-22 17:23:31.904799 [DEBUG] mod_sofia.c:261 sofia/external/+
>> 14158867717 at 199.173.100.16:5060 SOFIA DESTROY
>> 2009-10-22 17:23:31.904799 [DEBUG] switch_core_state_machine.c:60
>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard DESTROY
>> 2009-10-22 17:23:31.904799 [DEBUG] switch_core_state_machine.c:412
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State DESTROY going to
>> sleep
>>
>>
>>
>> On Tue, Oct 20, 2009 at 1:44 AM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>>
>>>
>>> On Sun, Oct 18, 2009 at 9:48 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>
>>>> The debug level logs to the console, right? The pastebin, had log level
>>>> debug, sofia trace on for external and default, and sofia loglevel all 9. Is
>>>> there another log enable command I'm missing? It seems loglevel all 9
>>>> outputs enter and exit functions, but at least to my novice eye, it's not
>>>> too obvious why freeswitch is sending a BYE to my DID provider.
>>>>
>>>> Evidently the console loglevel debug did not take because there isn't a
>>> single DEBUG level line of output. Please try again. Just remember that
>>> debug level log messages will have A LOT of information so be patient while
>>> sifting through it. Of course, feel free to pastebin the output and add a
>>> link to it in this email thread.
>>>
>>> -MC
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mcampbellsmith at gmail.com  Thu Oct 22 21:00:49 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Fri, 23 Oct 2009 15:00:49 +1100
Subject: [Freeswitch-users] Core Dump question!
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71701E84203@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701E840EB@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221258o6936000dn6eb2e22189c45435@mail.gmail.com>
	<DF11CCC6-5C89-4489-908C-92A0FCC94AE4@jerris.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E841E8@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221725g6233ec20w49a73864400426d6@mail.gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E84200@EXMBXCLUS01.citservers.local>
	<86a32abc0910222004y46bf08b6jda75b6b8f1979420@mail.gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E84203@EXMBXCLUS01.citservers.local>
Message-ID: <33c87fa30910222100rae92b90uf94d7ae58a5cdc94@mail.gmail.com>

I think the (exported) means you don't have the latest svn, but probably the
officially released build 1.0.4 that can be downloaded from the FS page.

I think you should see something like (the latest trunk is 15203):

freeswitch at internal> version

FreeSWITCH Version 1.0.trunk (15126)

I guess you need to checkout the latest FS trunk

On Fri, Oct 23, 2009 at 2:25 PM, Ujjval Karihaloo
<ujjval at simplesignal.com>wrote:

>  freeswitch at internal> version
>
> FreeSWITCH Version 1.0.4 (exported)
>
>
>
> freeswitch at internal>
>
>
>
> Ujjval Karihaloo
>
> VP Voice Engineering
>
> IP Phone: +13032428610
>
> E-Fax: +17202391690
>
>
>
> SimpleSignal Inc.
>
> 88 Inverness Circle East
>
> Suite K105
>
> Englewood, CO  80112
>
> [image: bvoip] <http://www.simplesignal.com/>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Diego Viola
> *Sent:* Thursday, October 22, 2009 9:04 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Core Dump question!
>
>
>
> Type "version" on the CLI.
>
> On Fri, Oct 23, 2009 at 2:52 AM, Ujjval Karihaloo <ujjval at simplesignal.com>
> wrote:
>
> How do I tell if it?s the latest?I downloaded is yesterday..and installed
> it from freeswitch.org
>
>
>
> Ujjval Karihaloo
>
> VP Voice Engineering
>
> IP Phone: +13032428610
>
> E-Fax: +17202391690
>
>
>
> SimpleSignal Inc.
>
> 88 Inverness Circle East
>
> Suite K105
>
> Englewood, CO  80112
>
> [image: bvoip] <http://www.simplesignal.com/>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, October 22, 2009 6:26 PM
>
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Core Dump question!
>
>
>
> Yes, if this is latest SVN (after a make current) then open a jira.
> -MC
>
> On Thu, Oct 22, 2009 at 5:04 PM, Ujjval Karihaloo <ujjval at simplesignal.com>
> wrote:
>
> I do have the core dump, should I open a ticket.
>
> I am running latest Freeswitch 1.0.4 and had done a make current just
> before it happened.
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From diego.viola at gmail.com  Thu Oct 22 22:15:21 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 23 Oct 2009 05:15:21 +0000
Subject: [Freeswitch-users] Core Dump question!
In-Reply-To: <33c87fa30910222100rae92b90uf94d7ae58a5cdc94@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701E840EB@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221258o6936000dn6eb2e22189c45435@mail.gmail.com>
	<DF11CCC6-5C89-4489-908C-92A0FCC94AE4@jerris.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E841E8@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221725g6233ec20w49a73864400426d6@mail.gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E84200@EXMBXCLUS01.citservers.local>
	<86a32abc0910222004y46bf08b6jda75b6b8f1979420@mail.gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E84203@EXMBXCLUS01.citservers.local>
	<33c87fa30910222100rae92b90uf94d7ae58a5cdc94@mail.gmail.com>
Message-ID: <86a32abc0910222215p6eb79f66lbbef2b245f4d17fd@mail.gmail.com>

He's running 1.0.4, he needs to checkout from SVN trunk and build FS from
scratch.

Diego

On Fri, Oct 23, 2009 at 4:00 AM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> I think the (exported) means you don't have the latest svn, but probably
> the officially released build 1.0.4 that can be downloaded from the FS page.
>
> I think you should see something like (the latest trunk is 15203):
>
> freeswitch at internal> version
>
> FreeSWITCH Version 1.0.trunk (15126)
>
> I guess you need to checkout the latest FS trunk
>
>
> On Fri, Oct 23, 2009 at 2:25 PM, Ujjval Karihaloo <ujjval at simplesignal.com
> > wrote:
>
>>  freeswitch at internal> version
>>
>> FreeSWITCH Version 1.0.4 (exported)
>>
>>
>>
>> freeswitch at internal>
>>
>>
>>
>> Ujjval Karihaloo
>>
>> VP Voice Engineering
>>
>> IP Phone: +13032428610
>>
>> E-Fax: +17202391690
>>
>>
>>
>> SimpleSignal Inc.
>>
>> 88 Inverness Circle East
>>
>> Suite K105
>>
>> Englewood, CO  80112
>>
>> [image: bvoip] <http://www.simplesignal.com/>
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Diego Viola
>> *Sent:* Thursday, October 22, 2009 9:04 PM
>>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* Re: [Freeswitch-users] Core Dump question!
>>
>>
>>
>> Type "version" on the CLI.
>>
>> On Fri, Oct 23, 2009 at 2:52 AM, Ujjval Karihaloo <
>> ujjval at simplesignal.com> wrote:
>>
>> How do I tell if it?s the latest?I downloaded is yesterday..and installed
>> it from freeswitch.org
>>
>>
>>
>> Ujjval Karihaloo
>>
>> VP Voice Engineering
>>
>> IP Phone: +13032428610
>>
>> E-Fax: +17202391690
>>
>>
>>
>> SimpleSignal Inc.
>>
>> 88 Inverness Circle East
>>
>> Suite K105
>>
>> Englewood, CO  80112
>>
>> [image: bvoip] <http://www.simplesignal.com/>
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
>> Collins
>> *Sent:* Thursday, October 22, 2009 6:26 PM
>>
>>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* Re: [Freeswitch-users] Core Dump question!
>>
>>
>>
>> Yes, if this is latest SVN (after a make current) then open a jira.
>> -MC
>>
>> On Thu, Oct 22, 2009 at 5:04 PM, Ujjval Karihaloo <
>> ujjval at simplesignal.com> wrote:
>>
>> I do have the core dump, should I open a ticket.
>>
>> I am running latest Freeswitch 1.0.4 and had done a make current just
>> before it happened.
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From maciej.aniserowicz at gmail.com  Fri Oct 23 00:36:31 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Fri, 23 Oct 2009 00:36:31 -0700 (PDT)
Subject: [Freeswitch-users] "Can not record session. Media not enabled
 on channel."
In-Reply-To: <87f2f3b90910201746h43a31b32m5c163fd98373f915@mail.gmail.com>
References: <1256034659005-3857858.post@n2.nabble.com>
	<87f2f3b90910201746h43a31b32m5c163fd98373f915@mail.gmail.com>
Message-ID: <1256283391890-3877285.post@n2.nabble.com>


The dialplan is very simple:

	<extension name="Recording test">
		<condition field="destination_number" expression="^111111\d*$">
			<action application="sleep" data="3000" />
			<action application="answer"/>
			<action application="say" data="en name_spelled iterated
${destination_number}"/>
			<action application="sleep" data="1000" />
			<action application="playback" data="local_stream://my_music"/>
		</condition>
	</extension>

Before debugging I have another question: I start recording in event handler
for ChannelAnswer event. Is it possible that it's too soon to start
recording? Maybe I should start recording in some other event?

MAniserowicz



mercutioviz wrote:
> 
> What's in the dialplan for this channel? Is bypass-media or proxy-media
> set
> to true? Do a debug trace and post it in pastebin.
> -MC
> 
> On Tue, Oct 20, 2009 at 3:30 AM, Maciej Aniserowicz <
> maciej.aniserowicz at gmail.com> wrote:
> 
>>
>> Hello,
>> I am using the same set of extensions for testing the system during
>> development, they include XLite, Cisco sip phone and several extensions
>> that
>> just play some audio file.
>> Sometimes, very rarely, this message "Can not record session.  Media not
>> enabled on channel." appears on FS console.
>> Like I wrote before, I don't change any codec settings and always use the
>> same set of devices/emulators.
>> What can cause this message?
>>
>> Thanks,
>> Maciej Aniserowicz
>>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3877285.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From tculjaga at gmail.com  Fri Oct 23 01:16:14 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 23 Oct 2009 10:16:14 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>
	<65d96fc80910220753n6d2efa39l470791f75db9a640@mail.gmail.com>
	<Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>
	<65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
	<Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
Message-ID: <65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>

> TC>
> TC>it is gonna be easier to track.
> TC>
> TC>Tomorrow i will test on 1.0.4 but please lets move to trunk.
>
> i make it a bit later, to move tickets to jira and source to svn i
> need some time to undertand how this system is works, especially jira.
>
>
audio issue is better now :)

however i have a few questions:

1. can we control codec framing size via config file setting (e.g. PCMA:20,
PCMU:20)?
2. can we control tunneling via config file setting?
3. can we control mediaWaitForConnect flag within setup message via config
file setting?


Now, when i can test more and place outgoing calls to different equipment, i
found that there is an issue when we get h.225 progress without a fastStart
element.

Here is a tshark:

  5.242526    10.4.62.7 -> 10.4.4.254   H.225.0 CS: setup OpenLogicalChannel

  5.243982   10.4.4.254 -> 10.4.62.7    H.225.0 CS: callProceeding
 10.512617   10.4.4.254 -> 10.4.62.7    H.225.0 CS: progress
 10.983697   10.4.4.254 -> 10.4.62.7    H.225.0 CS: alerting
 20.002796   10.4.4.254 -> 10.4.62.7    H.225.0 CS: connect
 20.002981   10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
terminalCapabilitySet
 20.003210   10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
masterSlaveDetermination
 31.472362   10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: releaseComplete
endSessionCommand


the terminating GW didn't  include a faststart element within a returning
h.225 message we didn't match the capabilities (framing of them) in our
setup (and you are waiting an open LC to start pre_answer).... so now, the
terminator is waiting for the originator to start exchanging TCS/MSD. As
tunneling is true, this should be done using h.225 Facility messages.


your behavior should be like this:

  5.242526    10.4.62.7 -> 10.4.4.254   H.225.0 CS: setup
OpenLogicalChannel  g711A with 30 ms
  5.243982   10.4.4.254 -> 10.4.62.7    H.225.0 CS: callProceeding
 10.512617   10.4.4.254 -> 10.4.62.7    H.225.0 CS: progress

                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS: facility
terminalCapabilitySet
                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS: facility
masterSlaveDetermination

                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
terminalCapabilitySet
                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
masterSlaveDetermination

                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
terminalCapabilitySetAck
                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
masterSlaveDeterminationAck

                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS: facility
terminalCapabilitySetAck
                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS: facility
masterSlaveDeterminationAck

                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS: facility
openlogicalchannel (g711A)
                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
openlogicalchannel (g711A)

                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS: facility
openlogicalchannelAck
                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
openlogicalchannelAck

                       <now you can start pre_answer!>

 10.983697   10.4.4.254 -> 10.4.62.7    H.225.0 CS: alerting
...
...
...
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From tculjaga at gmail.com  Fri Oct 23 01:44:59 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 23 Oct 2009 10:44:59 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>
	<65d96fc80910220753n6d2efa39l470791f75db9a640@mail.gmail.com>
	<Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>
	<65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
	<Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
Message-ID: <65d96fc80910230144r49692f66w56f61085ac42cb44@mail.gmail.com>

btw you are back with an old issue:


static const char modulename[] = "h323";
static const char* h323_formats[] = {
    "G.711-ALaw-64k", "PCMU",
    "G.711-uLaw-64k", "PCMA",
    "GSM-06.10", "GSM",
    "MS-GSM", "msgsm",
    "SpeexNarrow", "speex",
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From red.rain.seven at gmail.com  Fri Oct 23 01:54:45 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Fri, 23 Oct 2009 16:54:45 +0800
Subject: [Freeswitch-users] sched_api doesn't get launched
In-Reply-To: <191c3a030910211309sb727b37g6324e1833f2a1e8c@mail.gmail.com>
References: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
	<3F1747F9-C543-4C6F-BB01-4DD5FA5B7BC5@jerris.com>
	<59ad9ca10910161153h5d531533vd844e49b9ca6d896@mail.gmail.com>
	<87f2f3b90910161237h356b7aa2q6ec12c6ecf78dc34@mail.gmail.com>
	<59ad9ca10910210244q5410f2bdq3b93c25f95673d11@mail.gmail.com>
	<191c3a030910211309sb727b37g6324e1833f2a1e8c@mail.gmail.com>
Message-ID: <59ad9ca10910230154i6d41ddfci8957b152c1884800@mail.gmail.com>

Thanks to c6burns on IRC channel for the tip to use execute_on_answer in
combination with eval, and of course everyone here that pointing me to the
right direction.
I was able to execute sched_api with eval, but not with the combination of
execute_on_answer. The argument just don't get parsed as the eval argument.
Here is the code:

setVariable("execute_on_answer", "eval sched_api(+%ld none uuid_displace
<some uuid> start
/usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux)");


Here is the log:
2009-10-23 16:17:53.633189 [DEBUG] switch_scheduler.c:214 Added task 2
sched_api_function (none) to run at 1256285888
EXECUTE sofia/internal/1688004 at 192.168.1.66 *eval(+OK *Added: 2
)

2009-10-23 16:18:04.52701 [DEBUG] switch_channel.c:1927 sofia/internal/
1688004 at 192.168.1.66 *execute on answer: eval(sched_api(+37 none
uuid_displace 84d74f41-8668-4138-943f-f076e94046ad start
/usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux))*
EXECUTE sofia/internal/1688004 at 192.168.1.66 eval(sched_api(+37 none
uuid_displace 84d74f41-8668-4138-943f-f076e94046ad start
/usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux))


The first part of the log is when I just do execute("eval",
"${sched_api(+%ld none uuid_displace <some uuid> start
/usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux)}");
And I simple get a OK, and then later on the sched_api gets executed.

However, on the excerpt of the second log. which I use the combination with
execute_on_answer like specified in the yellow hight code above. It doesn't
simply return a OK, instead it shows the whole blob of code. and does
nothing later.



On Thu, Oct 22, 2009 at 4:09 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Yes you need an API object as described in other email.
>
> Which line of code from java caused that segfault
> It looks like a simple NULL string issue that we may want to hunt down.
>
>
> On Wed, Oct 21, 2009 at 4:44 AM, Henry Huang <red.rain.seven at gmail.com>wrote:
>
>> I can't seem to find the right thing to use in mod_java to execute api
>> commands, only api_after_bridge
>>
>> 2009-10-21 17:42:46.593094 [NOTICE] mod_sofia.c:1509 Pre-Answer
>> sofia/internal/1688003 at 192.168.1.66!
>> #
>> # A fatal error has been detected by the Java Runtime Environment:
>> #
>> #  SIGSEGV (0xb) at pc=0x004e4480, pid=1927, tid=16116624
>> #
>> # JRE version: 6.0_16-b01
>> # Java VM: Java HotSpot(TM) Client VM (14.2-b01 mixed mode linux-x86 )
>> # Problematic frame:
>> # C  [libc.so.6+0x6f480]  strcpy+0x10
>> #
>> # An error report file with more information is saved as:
>> # /usr/local/freeswitch/bin/hs_err_pid1927.log
>> 2009-10-21 17:42:59.883729 [ERR] switch_core_session.c:1374 Invalid
>> Application sched_api
>> 2009-10-21 17:42:59.883729 [NOTICE] switch_core_session.c:1375 Hangup
>> sofia/internal/1688003 at 192.168.1.66 [CS_EXECUTE]
>> [DESTINATION_OUT_OF_ORDER]
>> #
>> # If you would like to submit a bug report, please visit:
>> #   http://java.sun.com/webapps/bugreport/crash.jsp
>> # The crash happened outside the Java Virtual Machine in native code.
>> # See problematic frame for where to report the bug.
>>
>>
>> On Sat, Oct 17, 2009 at 3:37 AM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>>
>>>
>>> On Fri, Oct 16, 2009 at 11:53 AM, Henry Huang <red.rain.seven at gmail.com>wrote:
>>>
>>>> So how would you trigger it from a script dialplan? The only time it
>>>> seemed to work is when I did setVariable("api_after_bridge", "sched_api blah
>>>> blah blah");
>>>> but then it gets executed after the channel's been teared down. I
>>>> thought api_after_bridge means right after the call gets connected.
>>>>
>>>> I need something to execute an api command right before or right after
>>>> the call gets bridged.
>>>>
>>>> api_after_bridge is a channel variable, so using setVariable works just
>>> fine. If you need to sched_api is an API only. Check these out:
>>> http://wiki.freeswitch.org/wiki/Mod_commands#Misc._Commands
>>>
>>> So you need an API object in order to use it. I don't know the syntax for
>>> creating an api obj in Java but in Lua it goes like this:
>>> api = freeswitch.API();
>>> res = api:execute("sched_api","+300 none my_api my_api_args")
>>>
>>> Remember, if the method you are using isn't found in the dial plan tools
>>> then it isn't a dial plan application. Make sure it's on the list:
>>> http://wiki.freeswitch.org/wiki/Mod_dptools
>>>
>>> On the other hand, API commands are listed here:
>>> http://wiki.freeswitch.org/wiki/Mod_commands
>>>
>>> dptools require a session object, api commands require an api object...
>>>
>>> -MC
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Henry Huang
>> UniC Solution - Communication Unified
>> VoIP & Open Source software Consultant
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From saeedahmad1981 at gmail.com  Fri Oct 23 03:07:40 2009
From: saeedahmad1981 at gmail.com (Saeed Ahmad)
Date: Fri, 23 Oct 2009 12:07:40 +0200
Subject: [Freeswitch-users] Heartbeat question
In-Reply-To: <87f2f3b90910221118n5d9fa0b9u92b45227c1d7ec97@mail.gmail.com>
References: <86a32abc0910081956k91b3600sb8c556e589f5b089@mail.gmail.com> 
	<86a32abc0910082003v3f5d4e6dua8e7eae18fd2f939@mail.gmail.com> 
	<6b65470d0910082036g204d5dcbw26e1e43ef80c7c84@mail.gmail.com> 
	<86a32abc0910082245j36b8b3e9ncaf1cdd358fce319@mail.gmail.com> 
	<191c3a030910090902t38282f72l6b6d1733a758836@mail.gmail.com> 
	<86a32abc0910091427i2395fd80t3bc557186ece3972@mail.gmail.com> 
	<86a32abc0910091430hc151ed8of085fcc3af2f1de4@mail.gmail.com> 
	<86a32abc0910091501y40dd34c4r1bca3eeb2b6fa693@mail.gmail.com> 
	<c96b95650910221047i545601a4m9e0a645341068d15@mail.gmail.com> 
	<87f2f3b90910221118n5d9fa0b9u92b45227c1d7ec97@mail.gmail.com>
Message-ID: <c96b95650910230307p79e71c6o9c46a004ecb72a78@mail.gmail.com>

Yeah the word **hearbeat** was confusing me, now its ok.

So one use case would be to count live calls with much detailed information.


is it *resource*-*intensive** operation? can i do it on wholesale traffic
(500+ calls).*

...any other interesting use cases?

Thanks.

On Thu, Oct 22, 2009 at 8:18 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Thu, Oct 22, 2009 at 10:47 AM, Saeed Ahmad <saeedahmad1981 at gmail.com>wrote:
>
>> What is heartbeat and what are the uses cases?
>>
>> Sorry i didn't find much information on wiki.
>>
>> Thanks.
>>
>> A session heartbeat is just an event that is sent to your script and gives
> you updated information about the call in progress. It is *VERY* different
> from the system heartbeat, which is sent out to all event socket
> subscribers, not just a particular session.
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bottleman at icf.org.ru  Fri Oct 23 04:03:53 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Fri, 23 Oct 2009 15:03:53 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>
	<65d96fc80910220753n6d2efa39l470791f75db9a640@mail.gmail.com>
	<Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>
	<65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
	<Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>

On 2009-10-23 10:16 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>> TC>
TC>> TC>it is gonna be easier to track.
TC>> TC>
TC>> TC>Tomorrow i will test on 1.0.4 but please lets move to trunk.
TC>>
TC>> i make it a bit later, to move tickets to jira and source to svn i
TC>> need some time to undertand how this system is works, especially jira.
TC>>
TC>>
TC>audio issue is better now :)
TC>
TC>however i have a few questions:
TC>
TC>1. can we control codec framing size via config file setting (e.g. PCMA:20,
TC>PCMU:20)?

at this time i think no, there is a number issues in codec part now.

TC>2. can we control tunneling via config file setting?

at this time no, i implement it later.

TC>3. can we control mediaWaitForConnect flag within setup message via config
TC>file setting?

what is mediaWaitForConnect flag, may be another trmin in h323?

TC>Now, when i can test more and place outgoing calls to different equipment, i
TC>found that there is an issue when we get h.225 progress without a fastStart
TC>element.
TC>
TC>Here is a tshark:
TC>
TC>  5.242526    10.4.62.7 -> 10.4.4.254   H.225.0 CS: setup OpenLogicalChannel
TC>
TC>  5.243982   10.4.4.254 -> 10.4.62.7    H.225.0 CS: callProceeding
TC> 10.512617   10.4.4.254 -> 10.4.62.7    H.225.0 CS: progress
TC> 10.983697   10.4.4.254 -> 10.4.62.7    H.225.0 CS: alerting
TC> 20.002796   10.4.4.254 -> 10.4.62.7    H.225.0 CS: connect
TC> 20.002981   10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>terminalCapabilitySet
TC> 20.003210   10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>masterSlaveDetermination
TC> 31.472362   10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: releaseComplete
TC>endSessionCommand
TC>
TC>
TC>the terminating GW didn't  include a faststart element within a returning
TC>h.225 message we didn't match the capabilities (framing of them) in our
TC>setup (and you are waiting an open LC to start pre_answer).... so now, the
TC>terminator is waiting for the originator to start exchanging TCS/MSD. As
TC>tunneling is true, this should be done using h.225 Facility messages.
TC>
TC>
TC>your behavior should be like this:
TC>
TC>  5.242526    10.4.62.7 -> 10.4.4.254   H.225.0 CS: setup
TC>OpenLogicalChannel  g711A with 30 ms
TC>  5.243982   10.4.4.254 -> 10.4.62.7    H.225.0 CS: callProceeding
TC> 10.512617   10.4.4.254 -> 10.4.62.7    H.225.0 CS: progress
TC>
TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS: facility
TC>terminalCapabilitySet
TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS: facility
TC>masterSlaveDetermination
TC>
TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>terminalCapabilitySet
TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>masterSlaveDetermination
TC>
TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>terminalCapabilitySetAck
TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>masterSlaveDeterminationAck
TC>
TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS: facility
TC>terminalCapabilitySetAck
TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS: facility
TC>masterSlaveDeterminationAck
TC>
TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS: facility
TC>openlogicalchannel (g711A)
TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>openlogicalchannel (g711A)
TC>
TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS: facility
TC>openlogicalchannelAck
TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>openlogicalchannelAck
TC>
TC>                       <now you can start pre_answer!>
TC>
TC> 10.983697   10.4.4.254 -> 10.4.62.7    H.225.0 CS: alerting
TC>...

may bee, while i in hospital i have a very limited ways for testing, 
especially for inbound calls throuce h323. i find one issues in signaling 
part in h323plus, src/h323.cxx grep "Very Frustrating - S.H." try uncomment 
fast start handling there, may be it help.

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From bottleman at icf.org.ru  Fri Oct 23 04:11:04 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Fri, 23 Oct 2009 15:11:04 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910230144r49692f66w56f61085ac42cb44@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>
	<65d96fc80910220753n6d2efa39l470791f75db9a640@mail.gmail.com>
	<Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>
	<65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
	<Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
	<65d96fc80910230144r49692f66w56f61085ac42cb44@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910231504060.5644@srv.icf.org.ru>

On 2009-10-23 10:44 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>btw you are back with an old issue:
TC>
TC>
TC>static const char modulename[] = "h323";
TC>static const char* h323_formats[] = {
TC>    "G.711-ALaw-64k", "PCMU",
TC>    "G.711-uLaw-64k", "PCMA",
TC>    "GSM-06.10", "GSM",
TC>    "MS-GSM", "msgsm",
TC>    "SpeexNarrow", "speex",

there is a isues with codecs, it is not enoth to put it there to work, 
at this time work only PCMU PCMA GSM and G729.
									    

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From tculjaga at gmail.com  Fri Oct 23 04:22:40 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 23 Oct 2009 13:22:40 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>
	<65d96fc80910220753n6d2efa39l470791f75db9a640@mail.gmail.com>
	<Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>
	<65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
	<Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
	<Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>
Message-ID: <65d96fc80910230422q334506c7v89664f2e61696a8a@mail.gmail.com>

i meant you switched PCMA and PCMU...

T.

2009/10/23 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-23 10:16 +0200, Tihomir Culjaga wrote
> freeswitch-users at lists.fre...:
>
> TC>> TC>
> TC>> TC>it is gonna be easier to track.
> TC>> TC>
> TC>> TC>Tomorrow i will test on 1.0.4 but please lets move to trunk.
> TC>>
> TC>> i make it a bit later, to move tickets to jira and source to svn i
> TC>> need some time to undertand how this system is works, especially jira.
> TC>>
> TC>>
> TC>audio issue is better now :)
> TC>
> TC>however i have a few questions:
> TC>
> TC>1. can we control codec framing size via config file setting (e.g.
> PCMA:20,
> TC>PCMU:20)?
>
> at this time i think no, there is a number issues in codec part now.
>
> TC>2. can we control tunneling via config file setting?
>
> at this time no, i implement it later.
>
> TC>3. can we control mediaWaitForConnect flag within setup message via
> config
> TC>file setting?
>
> what is mediaWaitForConnect flag, may be another trmin in h323?
>
> TC>Now, when i can test more and place outgoing calls to different
> equipment, i
> TC>found that there is an issue when we get h.225 progress without a
> fastStart
> TC>element.
> TC>
> TC>Here is a tshark:
> TC>
> TC>  5.242526    10.4.62.7 -> 10.4.4.254   H.225.0 CS: setup
> OpenLogicalChannel
> TC>
> TC>  5.243982   10.4.4.254 -> 10.4.62.7    H.225.0 CS: callProceeding
> TC> 10.512617   10.4.4.254 -> 10.4.62.7    H.225.0 CS: progress
> TC> 10.983697   10.4.4.254 -> 10.4.62.7    H.225.0 CS: alerting
> TC> 20.002796   10.4.4.254 -> 10.4.62.7    H.225.0 CS: connect
> TC> 20.002981   10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
> TC>terminalCapabilitySet
> TC> 20.003210   10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
> TC>masterSlaveDetermination
> TC> 31.472362   10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS:
> releaseComplete
> TC>endSessionCommand
> TC>
> TC>
> TC>the terminating GW didn't  include a faststart element within a
> returning
> TC>h.225 message we didn't match the capabilities (framing of them) in our
> TC>setup (and you are waiting an open LC to start pre_answer).... so now,
> the
> TC>terminator is waiting for the originator to start exchanging TCS/MSD. As
> TC>tunneling is true, this should be done using h.225 Facility messages.
> TC>
> TC>
> TC>your behavior should be like this:
> TC>
> TC>  5.242526    10.4.62.7 -> 10.4.4.254   H.225.0 CS: setup
> TC>OpenLogicalChannel  g711A with 30 ms
> TC>  5.243982   10.4.4.254 -> 10.4.62.7    H.225.0 CS: callProceeding
> TC> 10.512617   10.4.4.254 -> 10.4.62.7    H.225.0 CS: progress
> TC>
> TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS:
> facility
> TC>terminalCapabilitySet
> TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS:
> facility
> TC>masterSlaveDetermination
> TC>
> TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS:
> facility
> TC>terminalCapabilitySet
> TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS:
> facility
> TC>masterSlaveDetermination
> TC>
> TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS:
> facility
> TC>terminalCapabilitySetAck
> TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS:
> facility
> TC>masterSlaveDeterminationAck
> TC>
> TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS:
> facility
> TC>terminalCapabilitySetAck
> TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS:
> facility
> TC>masterSlaveDeterminationAck
> TC>
> TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS:
> facility
> TC>openlogicalchannel (g711A)
> TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS:
> facility
> TC>openlogicalchannel (g711A)
> TC>
> TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS:
> facility
> TC>openlogicalchannelAck
> TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS:
> facility
> TC>openlogicalchannelAck
> TC>
> TC>                       <now you can start pre_answer!>
> TC>
> TC> 10.983697   10.4.4.254 -> 10.4.62.7    H.225.0 CS: alerting
> TC>...
>
> may bee, while i in hospital i have a very limited ways for testing,
> especially for inbound calls throuce h323. i find one issues in signaling
> part in h323plus, src/h323.cxx grep "Very Frustrating - S.H." try uncomment
> fast start handling there, may be it help.
>
> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129-RIPE
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bottleman at icf.org.ru  Fri Oct 23 04:32:41 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Fri, 23 Oct 2009 15:32:41 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910230422q334506c7v89664f2e61696a8a@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>
	<65d96fc80910220753n6d2efa39l470791f75db9a640@mail.gmail.com>
	<Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>
	<65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
	<Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
	<Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>
	<65d96fc80910230422q334506c7v89664f2e61696a8a@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910231532050.16254@srv.icf.org.ru>

On 2009-10-23 13:22 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

my bad, thx.

TC>i meant you switched PCMA and PCMU...
TC>
TC>T.
TC>
TC>2009/10/23 Georgiewskiy Yuriy <bottleman at icf.org.ru>
TC>
TC>> On 2009-10-23 10:16 +0200, Tihomir Culjaga wrote
TC>> freeswitch-users at lists.fre...:
TC>>
TC>> TC>> TC>
TC>> TC>> TC>it is gonna be easier to track.
TC>> TC>> TC>
TC>> TC>> TC>Tomorrow i will test on 1.0.4 but please lets move to trunk.
TC>> TC>>
TC>> TC>> i make it a bit later, to move tickets to jira and source to svn i
TC>> TC>> need some time to undertand how this system is works, especially jira.
TC>> TC>>
TC>> TC>>
TC>> TC>audio issue is better now :)
TC>> TC>
TC>> TC>however i have a few questions:
TC>> TC>
TC>> TC>1. can we control codec framing size via config file setting (e.g.
TC>> PCMA:20,
TC>> TC>PCMU:20)?
TC>>
TC>> at this time i think no, there is a number issues in codec part now.
TC>>
TC>> TC>2. can we control tunneling via config file setting?
TC>>
TC>> at this time no, i implement it later.
TC>>
TC>> TC>3. can we control mediaWaitForConnect flag within setup message via
TC>> config
TC>> TC>file setting?
TC>>
TC>> what is mediaWaitForConnect flag, may be another trmin in h323?
TC>>
TC>> TC>Now, when i can test more and place outgoing calls to different
TC>> equipment, i
TC>> TC>found that there is an issue when we get h.225 progress without a
TC>> fastStart
TC>> TC>element.
TC>> TC>
TC>> TC>Here is a tshark:
TC>> TC>
TC>> TC>  5.242526    10.4.62.7 -> 10.4.4.254   H.225.0 CS: setup
TC>> OpenLogicalChannel
TC>> TC>
TC>> TC>  5.243982   10.4.4.254 -> 10.4.62.7    H.225.0 CS: callProceeding
TC>> TC> 10.512617   10.4.4.254 -> 10.4.62.7    H.225.0 CS: progress
TC>> TC> 10.983697   10.4.4.254 -> 10.4.62.7    H.225.0 CS: alerting
TC>> TC> 20.002796   10.4.4.254 -> 10.4.62.7    H.225.0 CS: connect
TC>> TC> 20.002981   10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>> TC>terminalCapabilitySet
TC>> TC> 20.003210   10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>> TC>masterSlaveDetermination
TC>> TC> 31.472362   10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS:
TC>> releaseComplete
TC>> TC>endSessionCommand
TC>> TC>
TC>> TC>
TC>> TC>the terminating GW didn't  include a faststart element within a
TC>> returning
TC>> TC>h.225 message we didn't match the capabilities (framing of them) in our
TC>> TC>setup (and you are waiting an open LC to start pre_answer).... so now,
TC>> the
TC>> TC>terminator is waiting for the originator to start exchanging TCS/MSD. As
TC>> TC>tunneling is true, this should be done using h.225 Facility messages.
TC>> TC>
TC>> TC>
TC>> TC>your behavior should be like this:
TC>> TC>
TC>> TC>  5.242526    10.4.62.7 -> 10.4.4.254   H.225.0 CS: setup
TC>> TC>OpenLogicalChannel  g711A with 30 ms
TC>> TC>  5.243982   10.4.4.254 -> 10.4.62.7    H.225.0 CS: callProceeding
TC>> TC> 10.512617   10.4.4.254 -> 10.4.62.7    H.225.0 CS: progress
TC>> TC>
TC>> TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS:
TC>> facility
TC>> TC>terminalCapabilitySet
TC>> TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS:
TC>> facility
TC>> TC>masterSlaveDetermination
TC>> TC>
TC>> TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS:
TC>> facility
TC>> TC>terminalCapabilitySet
TC>> TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS:
TC>> facility
TC>> TC>masterSlaveDetermination
TC>> TC>
TC>> TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS:
TC>> facility
TC>> TC>terminalCapabilitySetAck
TC>> TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS:
TC>> facility
TC>> TC>masterSlaveDeterminationAck
TC>> TC>
TC>> TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS:
TC>> facility
TC>> TC>terminalCapabilitySetAck
TC>> TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS:
TC>> facility
TC>> TC>masterSlaveDeterminationAck
TC>> TC>
TC>> TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS:
TC>> facility
TC>> TC>openlogicalchannel (g711A)
TC>> TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS:
TC>> facility
TC>> TC>openlogicalchannel (g711A)
TC>> TC>
TC>> TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS:
TC>> facility
TC>> TC>openlogicalchannelAck
TC>> TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS:
TC>> facility
TC>> TC>openlogicalchannelAck
TC>> TC>
TC>> TC>                       <now you can start pre_answer!>
TC>> TC>
TC>> TC> 10.983697   10.4.4.254 -> 10.4.62.7    H.225.0 CS: alerting
TC>> TC>...
TC>>
TC>> may bee, while i in hospital i have a very limited ways for testing,
TC>> especially for inbound calls throuce h323. i find one issues in signaling
TC>> part in h323plus, src/h323.cxx grep "Very Frustrating - S.H." try uncomment
TC>> fast start handling there, may be it help.
TC>>
TC>> C ?????????                       With Best Regards
TC>> ???????????? ????.                Georgiewskiy Yuriy
TC>> +7 4872 711666                    +7 4872 711666
TC>> ???? +7 4872 711143               fax +7 4872 711143
TC>> ???????? ??? "?? ?? ??????"       IT Service Ltd
TC>> http://nkoort.ru                  http://nkoort.ru
TC>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
TC>> YG129-RIPE                        YG129-RIPE
TC>>
TC>> _______________________________________________
TC>> FreeSWITCH-users mailing list
TC>> FreeSWITCH-users at lists.freeswitch.org
TC>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> http://www.freeswitch.org
TC>>
TC>>
TC>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From helmut.kuper at ewetel.de  Fri Oct 23 04:35:54 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Fri, 23 Oct 2009 13:35:54 +0200
Subject: [Freeswitch-users] Qustion about INFO messages
	after	Connect/Answer
In-Reply-To: <191c3a030910221008n60341314hc7a7e6a55a50ef15@mail.gmail.com>
References: <4AD859A4.5000101@ewetel.de>	<191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>	<4ADC85C0.7060908@ewetel.de>
	<4ADC8B6E.7070404@ewetel.de>	<191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>	<87f2f3b90910191435s7528e256r85df5f12fc634f24@mail.gmail.com>	<4ADF483B.4050807@ewetel.de>	<87f2f3b90910211048s208daf1eyfae501091ab75530@mail.gmail.com>	<4AE07DF4.5040803@ewetel.de>	<085B77EE-DA3F-4ADD-86DC-E59D1B6553EA@freeswitch.org>
	<191c3a030910221008n60341314hc7a7e6a55a50ef15@mail.gmail.com>
Message-ID: <4AE1951A.1030906@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Anthony,

thanks! The "unknown" thing is gone as long as you don't use
"originate_callee_id_name"


The "originate_callee_id_name" chvar causes caller's AND callee's
display ("caller name" line)to be updated with the content of that chvar
as soon as you set it.

Let's say you use "hubu" as originate_callee_id_name. As soon as callee
has picked up the phone it's display ("caller name" line) is updated
with "hubu" and right after that with caller's name. The caller's
display ("callee name" line) is updated to "hubu" (as expected).

So it's still not nice to see a name switch in callee's display for
caller's name after picking up.

In the following you see the SIP flows from

a) caller to FS
b) FS to callee


The more interesting one is b) I guess




#######################################################
This is the sip flow from caller to FS, which is OK:



INVITE sip:1000 at 85.16.246.12:5061;user=phone SIP/2.0
23/Oct/2009-13:26:44.730

INVITE sip:1000 at 85.16.246.12:5061;user=phone SIP/2.0
Via: SIP/2.0/UDP 85.16.245.206:1024;branch=z9hG4bK-zdcpisfjx11n;rport
From: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
To: <sip:1000 at 85.16.246.12:5061;user=phone>
Call-ID: 3c32462e758f-y19kuulwdknf
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:1001 at 85.16.245.206:1024;line=eg3wp69a>;reg-id=1
X-Serialnumber: 0004134002CB
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom820/8.2.16
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 734

v=0
o=root 1420196469 1420196469 IN IP4 85.16.245.206
s=call
c=IN IP4 85.16.245.206
t=0 0
m=audio 51286 RTP/SAVP 0 8 9 99 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:CjdPN1m2iLKAyXrQ2pWkQfCMtiISlF94ItTdsDis
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=audio 51286 RTP/AVP 0 8 9 99 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-ev

SIP/2.0 100 Trying
23/Oct/2009-13:26:44.731

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 85.16.245.206:1024;branch=z9hG4bK-zdcpisfjx11n;rport=1024
From: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
To: <sip:1000 at 85.16.246.12:5061;user=phone>
Call-ID: 3c32462e758f-y19kuulwdknf
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
Content-Length: 0


SIP/2.0 180 Ringing
23/Oct/2009-13:26:45.3

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 85.16.245.206:1024;branch=z9hG4bK-zdcpisfjx11n;rport=1024
From: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
To: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
Call-ID: 3c32462e758f-y19kuulwdknf
CSeq: 1 INVITE
Contact: <sip:1000 at 85.16.246.12:5061;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
Content-Length: 0
X-FS-Display-Name: hubu
X-FS-Display-Number:
X-FS-Support: update_display


SIP/2.0 200 OK
23/Oct/2009-13:26:47.367

SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.16.245.206:1024;branch=z9hG4bK-zdcpisfjx11n;rport=1024
From: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
To: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
Call-ID: 3c32462e758f-y19kuulwdknf
CSeq: 1 INVITE
Contact: <sip:1000 at 85.16.246.12:5061;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
Session-Expires: 3600;refresher=uas
Min-SE: 120
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 354
X-FS-Display-Name: hubu
X-FS-Display-Number:
X-FS-Support: update_display

v=0
o=FreeSWITCH 1256285701 1256285702 IN IP4 85.16.246.12
s=FreeSWITCH
c=IN IP4 85.16.246.12
t=0 0
m=audio 11506 RTP/SAVP 8 101
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:/BFm/LUmDBuao0wqg6MS/l4acfzOHoHRFQr3N0R9
m=audio 0 RTP/AVP 19

ACK sip:1000 at 85.16.246.12:5061;transport=udp SIP/2.0
23/Oct/2009-13:26:47.392

ACK sip:1000 at 85.16.246.12:5061;transport=udp SIP/2.0
Via: SIP/2.0/UDP 85.16.245.206:1024;branch=z9hG4bK-d474to81d7gh;rport
From: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
To: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
Call-ID: 3c32462e758f-y19kuulwdknf
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:1001 at 85.16.245.206:1024;line=eg3wp69a>;reg-id=1
Content-Length: 0


INFO sip:1001 at 85.16.245.206:1024;line=eg3wp69a SIP/2.0
23/Oct/2009-13:26:47.416

INFO sip:1001 at 85.16.245.206:1024;line=eg3wp69a SIP/2.0
Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKB244eKZ08Uj3H
Max-Forwards: 70
From: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
To: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
Call-ID: 3c32462e758f-y19kuulwdknf
CSeq: 122030267 INFO
Contact: <sip:1000 at 85.16.246.12:5061;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Type: message/sipfrag
Content-Length: 24

From:
To: "hubu" 1000

SIP/2.0 200 Ok
23/Oct/2009-13:26:47.619

SIP/2.0 200 Ok
Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKB244eKZ08Uj3H
From: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
To: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
Call-ID: 3c32462e758f-y19kuulwdknf
CSeq: 122030267 INFO
Contact: <sip:1001 at 85.16.245.206:1024;line=eg3wp69a>;reg-id=1
Content-Length: 0


BYE sip:1001 at 85.16.245.206:1024;line=eg3wp69a SIP/2.0
23/Oct/2009-13:26:49.18

BYE sip:1001 at 85.16.245.206:1024;line=eg3wp69a SIP/2.0
Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKDmQpj9072DZ8r
Max-Forwards: 70
From: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
To: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
Call-ID: 3c32462e758f-y19kuulwdknf
CSeq: 122030268 BYE
Contact: <sip:1000 at 85.16.246.12:5061;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0


SIP/2.0 200 OK
23/Oct/2009-13:26:49.52

SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKDmQpj9072DZ8r
From: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
To: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
Call-ID: 3c32462e758f-y19kuulwdknf
CSeq: 122030268 BYE
Contact: <sip:1001 at 85.16.245.206:1024;line=eg3wp69a>;reg-id=1
User-Agent: snom820/8.2.16
RTP-RxStat: Total_Rx_Pkts=65,Rx_Pkts=65,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0
RTP-TxStat: Total_Tx_Pkts=0,Tx_Pkts=94,Remote_Tx_Pkts=0
Content-Length: 0





####################################################################

This is the SIP flow from FS to callee.

The first INFO sent from FS to callee is the Problem. It contains "hubu"
as "To:" header. If there is s need for this message I guess the "From"
header should be changed, because from the callee-phone's point of view
the "To:" header is the caller and the "From" header is the callee. In
my opinion this INFO message shouldn't be sent at all, but I don't know
what you intend it to do.


INVITE sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
23/Oct/2009-13:26:44.978

INVITE sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bK86rt91cpH1FBF
Max-Forwards: 68
From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=rp27e1KSe969D
To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 122030266 INVITE
Contact: <sip:mod_sofia at 85.16.246.12:5061>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 325
X-Serialnumber: 0004134002CB
P-Key-Flags: resolution="31x13", keys="4"
X-FS-Support: update_display
Remote-Party-ID: "1001 an PBX1"
<sip:1001 at 85.16.246.12>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1256285400 1256285401 IN IP4 85.16.246.12
s=FreeSWITCH
c=IN IP4 85.16.246.12
t=0 0
m=audio 11804 RTP/AVP 95 103 9 8 101 13
a=rtpmap:95 CELT/48000
a=rtpmap:103 SPEEX/32000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:10

SIP/2.0 180 Ringing
23/Oct/2009-13:26:45.2

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bK86rt91cpH1FBF
From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=rp27e1KSe969D
To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 122030266 INVITE
Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Content-Length: 0


SIP/2.0 180 Ringing
23/Oct/2009-13:26:45.504

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bK86rt91cpH1FBF
From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=rp27e1KSe969D
To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 122030266 INVITE
Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Content-Length: 0


SIP/2.0 180 Ringing
23/Oct/2009-13:26:46.504

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bK86rt91cpH1FBF
From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=rp27e1KSe969D
To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 122030266 INVITE
Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Content-Length: 0


SIP/2.0 200 Ok
23/Oct/2009-13:26:47.361

SIP/2.0 200 Ok
Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bK86rt91cpH1FBF
From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=rp27e1KSe969D
To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 122030266 INVITE
Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
User-Agent: snom820/LAB_HD_HEAD_11_09_09_11_20
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Type: application/sdp
Content-Length: 244

v=0
o=root 353690302 353690303 IN IP4 85.16.245.228
s=call
c=IN IP4 85.16.245.228
t=0 0
m=audio 60878 RTP/AVP 9 8 101
a=rtpmap:9 g722/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:10
a=sendrecv

ACK sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
23/Oct/2009-13:26:47.362

ACK sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bK9FjKBXXSea6Xa
Max-Forwards: 70
From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=rp27e1KSe969D
To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 122030266 ACK
Contact: <sip:mod_sofia at 85.16.246.12:5061>
Content-Length: 0


INFO sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
23/Oct/2009-13:26:47.367

INFO sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKaSBcDreXBKvgp
Max-Forwards: 70
From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=rp27e1KSe969D
To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 122030267 INFO
Contact: <sip:mod_sofia at 85.16.246.12:5061>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Type: message/sipfrag
Content-Length: 24

From:
To: "hubu" 1000

SIP/2.0 200 Ok
23/Oct/2009-13:26:47.766

SIP/2.0 200 Ok
Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKaSBcDreXBKvgp
From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=rp27e1KSe969D
To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 122030267 INFO
Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
Content-Length: 0


INFO sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
23/Oct/2009-13:26:47.766

INFO sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKcByXgeg4548ND
Max-Forwards: 70
From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=rp27e1KSe969D
To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 122030268 INFO
Contact: <sip:mod_sofia at 85.16.246.12:5061>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Type: message/sipfrag
Content-Length: 32

From:
To: "1001 an PBX1" 1001

SIP/2.0 200 Ok
23/Oct/2009-13:26:47.860

SIP/2.0 200 Ok
Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKcByXgeg4548ND
From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=rp27e1KSe969D
To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 122030268 INFO
Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
Content-Length: 0


BYE sip:mod_sofia at 85.16.246.12:5061 SIP/2.0
23/Oct/2009-13:26:49.3

BYE sip:mod_sofia at 85.16.246.12:5061 SIP/2.0
Via: SIP/2.0/UDP 85.16.245.228:1024;branch=z9hG4bK-susgzpvc38s7;rport
From: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
To: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=rp27e1KSe969D
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 1 BYE
Max-Forwards: 70
Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
User-Agent: snom820/LAB_HD_HEAD_11_09_09_11_20
RTP-RxStat: Total_Rx_Pkts=0,Rx_Pkts=69,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0
RTP-TxStat: Total_Tx_Pkts=70,Tx_Pkts=65,Remote_Tx_Pkts=0
Content-Length: 0


SIP/2.0 200 OK
23/Oct/2009-13:26:49.4

SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.16.245.228:1024;branch=z9hG4bK-susgzpvc38s7;rport=1024
From: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
To: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=rp27e1KSe969D
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 1 BYE
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Length: 0










On 22.10.2009 19:08, Anthony Minessale wrote:
> I'm sure that problem is gone in svn trunk.
> 
> 
> On Thu, Oct 22, 2009 at 11:25 AM, Brian West <brian at freeswitch.org
> <mailto:brian at freeswitch.org>> wrote:
> 
>     I can't get what exactly you re talking about. Can you clarify ? Also
>     please include the packets of interest only not the full trace if its
>     not relevant to the bug.
> 
>     /b
> 
>     On Oct 22, 2009, at 10:44 AM, Helmut Kuper wrote:
> 
>     > -----BEGIN PGP SIGNED MESSAGE-----
>     > Hash: SHA1
>     >
>     > Hi Mike,
>     >
>     > here it is:
>     >
>     >
>     > Dialplan:
>     >
>     >
>     >    <extension name="Local_Extension">
>     >      <condition field="destination_number" expression="^(10[01][0-9])
>     > $">
>     >        <action application="set" data="dialed_extension=$1"/>
>     >        <action application="export" data="dialed_extension=$1"/>
>     >        <action application="set" data="transfer_ringback=$$
>     > {hold_music}"/>
>     >        <action application="set" data="hangup_after_bridge=true"/>
>     >        <action application="bridge"
>     > data="user/${dialed_extension}@${domain_name}"/>
>     >      </condition>
>     >    </extension>
>     >
> 
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

- -- 

Mit freundlichen Gr??en
Helmut Kuper
Gesch?ftseinheit FD - L?sungen f?r Finanzdienstleister
Telefax: (0441) 8000-2799
mailto:helmut.kuper at ewetel.de
___________________________________
EWE TEL GmbH
Cloppenburger Stra?e 310
26133 Oldenburg
EWE TEL GmbH

Handelsregister Amtsgericht Oldenburg HRB 3723
Vorsitzender des Aufsichtsrates: Heiko Harms
Gesch?ftsf?hrung: Hans-Joachim Iken (Vorsitzender),
Ulf Heggenberger, Dr. Norbert Schulz, Dirk Thole
Homepage: http://www.ewetel.de
___________________________________
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=cHej
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From tculjaga at gmail.com  Fri Oct 23 04:52:06 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 23 Oct 2009 13:52:06 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>
	<65d96fc80910220753n6d2efa39l470791f75db9a640@mail.gmail.com>
	<Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>
	<65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
	<Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
	<Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>
Message-ID: <65d96fc80910230452s6b4a179aqa5bc41dc34efc403@mail.gmail.com>

>
> TC>3. can we control mediaWaitForConnect flag within setup message via
> config
> TC>file setting?
>
> what is mediaWaitForConnect flag, may be another trmin in h323?
>
>
>
If the calling endpoint sets the mediaWaitForConnect element to TRUE in the
Setup message, then
the called endpoint shall not send any media until after the Connect message
is sent.
The calling endpoint may begin transmitting media (according to the channels
opened) immediately
upon receiving a Q.931 message containing fastStart. Thus, the called
endpoint must be prepared to
immediately receive media on the channels it accepted in the Q.931 message
containing fastStart.
Note that national requirements may prohibit calling endpoints from
transmitting media prior to
receipt of a Connect message; it is the responsibility of the endpoint to
comply with applicable
requirements.


check "H225_Setup_UUIE & H323SignalPDU::BuildSetup" within src/h323pdu.cxx
(H323plus)



> TC>...
>
> may bee, while i in hospital i have a very limited ways for testing,
> especially for inbound calls throuce h323. i find one issues in signaling
> part in h323plus, src/h323.cxx grep "Very Frustrating - S.H." try uncomment
> fast start handling there, may be it help.
>
>
I'm not sure it is gonna help. This is only for CallProceeding having a
faststart element... What i have is a progress message without a faststart
element but with h245 address... it should go to StartControlChannel but i
think it is stuck since you call pre_answer before it actually opens a LC.


PBoolean H323Connection::OnReceivedProgress(const H323SignalPDU & pdu)
{
  if (pdu.m_h323_uu_pdu.m_h323_message_body.GetTag() !=
H225_H323_UU_PDU_h323_message_body::e_progress)
    return FALSE;
  const H225_Progress_UUIE & progress =
pdu.m_h323_uu_pdu.m_h323_message_body;

  SetRemoteVersions(progress.m_protocolIdentifier);
  SetRemotePartyInfo(pdu);
  SetRemoteApplication(progress.m_destinationInfo);

  // Check for fastStart data and start fast
  if (progress.HasOptionalField(H225_Progress_UUIE::e_fastStart))
    HandleFastStartAcknowledge(progress.m_fastStart);

  // Check that it has the H.245 channel connection info
  if (progress.HasOptionalField(H225_Progress_UUIE::e_h245Address))
    return StartControlChannel(progress.m_h245Address);

  return TRUE;
}



you should handle this and postpone pre_answer until you get an open LC.


T.
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From helmut.kuper at ewetel.de  Fri Oct 23 05:09:50 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Fri, 23 Oct 2009 14:09:50 +0200
Subject: [Freeswitch-users] INFO-Messages: Send_display in pre_answer state
	doesnt work anymore
Message-ID: <4AE19D0E.9050301@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

when I use pre_answer app and right after that the send_display app,
there is no INFO message anymore.

My dialplan condition:

<condition field="${sip_hangup_phrase}" expression="^Do Not Disturb"
break="on-true">
                        <action application="pre_answer"/>
                        <action application="send_display" data="Bitte
nicht stoeren"/>
                        <action application="sleep" data="500"/>
                        <action application="playback"
data="ansagen/22.wav"/>
</condition>

changing "pre_answer" to "answer" works, but I don't want to answer the
call for error messages.

I use latest SVN trunk.

regards
helmut
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From bottleman at icf.org.ru  Fri Oct 23 05:13:48 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Fri, 23 Oct 2009 16:13:48 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910230452s6b4a179aqa5bc41dc34efc403@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>
	<65d96fc80910220753n6d2efa39l470791f75db9a640@mail.gmail.com>
	<Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>
	<65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
	<Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
	<Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>
	<65d96fc80910230452s6b4a179aqa5bc41dc34efc403@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910231610220.21492@srv.icf.org.ru>

On 2009-10-23 13:52 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>>
TC>> TC>3. can we control mediaWaitForConnect flag within setup message via
TC>> config
TC>> TC>file setting?
TC>>
TC>> what is mediaWaitForConnect flag, may be another trmin in h323?
TC>>
TC>>
TC>>
TC>If the calling endpoint sets the mediaWaitForConnect element to TRUE in the
TC>Setup message, then
TC>the called endpoint shall not send any media until after the Connect message
TC>is sent.
TC>The calling endpoint may begin transmitting media (according to the channels
TC>opened) immediately
TC>upon receiving a Q.931 message containing fastStart. Thus, the called
TC>endpoint must be prepared to
TC>immediately receive media on the channels it accepted in the Q.931 message
TC>containing fastStart.
TC>Note that national requirements may prohibit calling endpoints from
TC>transmitting media prior to
TC>receipt of a Connect message; it is the responsibility of the endpoint to
TC>comply with applicable
TC>requirements.
TC>
TC>
TC>check "H225_Setup_UUIE & H323SignalPDU::BuildSetup" within src/h323pdu.cxx
TC>(H323plus)

i think it can be implemented later, but, why it may be needed? can you explain some 
situation where it need?

TC>> TC>...
TC>>
TC>> may bee, while i in hospital i have a very limited ways for testing,
TC>> especially for inbound calls throuce h323. i find one issues in signaling
TC>> part in h323plus, src/h323.cxx grep "Very Frustrating - S.H." try uncomment
TC>> fast start handling there, may be it help.
TC>>
TC>>
TC>I'm not sure it is gonna help. This is only for CallProceeding having a
TC>faststart element... What i have is a progress message without a faststart
TC>element but with h245 address... it should go to StartControlChannel but i
TC>think it is stuck since you call pre_answer before it actually opens a LC.
TC>
TC>
TC>PBoolean H323Connection::OnReceivedProgress(const H323SignalPDU & pdu)
TC>{
TC>  if (pdu.m_h323_uu_pdu.m_h323_message_body.GetTag() !=
TC>H225_H323_UU_PDU_h323_message_body::e_progress)
TC>    return FALSE;
TC>  const H225_Progress_UUIE & progress =
TC>pdu.m_h323_uu_pdu.m_h323_message_body;
TC>
TC>  SetRemoteVersions(progress.m_protocolIdentifier);
TC>  SetRemotePartyInfo(pdu);
TC>  SetRemoteApplication(progress.m_destinationInfo);
TC>
TC>  // Check for fastStart data and start fast
TC>  if (progress.HasOptionalField(H225_Progress_UUIE::e_fastStart))
TC>    HandleFastStartAcknowledge(progress.m_fastStart);
TC>
TC>  // Check that it has the H.245 channel connection info
TC>  if (progress.HasOptionalField(H225_Progress_UUIE::e_h245Address))
TC>    return StartControlChannel(progress.m_h245Address);
TC>
TC>  return TRUE;
TC>}
TC>
TC>
TC>
TC>you should handle this and postpone pre_answer until you get an open LC.

pre_answer is not complete at this time, i say it a some kinde of hack, there is
another issues with it ans sofia in case proxy-media true.

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From tculjaga at gmail.com  Fri Oct 23 05:38:40 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 23 Oct 2009 14:38:40 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910231610220.21492@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>
	<65d96fc80910220753n6d2efa39l470791f75db9a640@mail.gmail.com>
	<Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>
	<65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
	<Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
	<Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>
	<65d96fc80910230452s6b4a179aqa5bc41dc34efc403@mail.gmail.com>
	<Pine.LNX.4.64.0910231610220.21492@srv.icf.org.ru>
Message-ID: <65d96fc80910230538w7105be6fve14956ab3b247bf6@mail.gmail.com>

>
> TC>check "H225_Setup_UUIE & H323SignalPDU::BuildSetup" within
> src/h323pdu.cxx
> TC>(H323plus)
>
> i think it can be implemented later, but, why it may be needed? can you
> explain some
> situation where it need?
>
>
> TC>
> TC>you should handle this and postpone pre_answer until you get an open LC.
>
> pre_answer is not complete at this time, i say it a some kinde of hack,
> there is
> another issues with it ans sofia in case proxy-media true.
>
>

bool FSH323Connection::OnReceivedProgress(const H323SignalPDU &pdu)
{
        PTRACE(4, "mod_h323\t======>FSH323Connection::OnReceivedProgress");
        m_txAudioOpened.Wait();
        switch_channel_mark_pre_answered(m_fsChannel);
        return true;
}



so for me the workaround for this was:




bool FSH323Connection::OnReceivedProgress(const H323SignalPDU &pdu)
{
        PTRACE(4, "mod_h323\t======>FSH323Connection::OnReceivedProgress");

        PTRACE(4, "mod_h323\t======>FSH323Connection::OnReceivedProgress -
disabled pre_answer!!!!");

        //m_txAudioOpened.Wait();
        //switch_channel_mark_pre_answered(m_fsChannel);
        return true;
}
-------------- next part --------------
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From bottleman at icf.org.ru  Fri Oct 23 05:53:32 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Fri, 23 Oct 2009 16:53:32 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910230538w7105be6fve14956ab3b247bf6@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>
	<65d96fc80910220753n6d2efa39l470791f75db9a640@mail.gmail.com>
	<Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>
	<65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
	<Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
	<Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>
	<65d96fc80910230452s6b4a179aqa5bc41dc34efc403@mail.gmail.com>
	<Pine.LNX.4.64.0910231610220.21492@srv.icf.org.ru>
	<65d96fc80910230538w7105be6fve14956ab3b247bf6@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910231646210.29582@srv.icf.org.ru>

On 2009-10-23 14:38 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>>
TC>> TC>check "H225_Setup_UUIE & H323SignalPDU::BuildSetup" within
TC>> src/h323pdu.cxx
TC>> TC>(H323plus)
TC>>
TC>> i think it can be implemented later, but, why it may be needed? can you
TC>> explain some
TC>> situation where it need?
TC>>
TC>>
TC>> TC>
TC>> TC>you should handle this and postpone pre_answer until you get an open LC.
TC>>
TC>> pre_answer is not complete at this time, i say it a some kinde of hack,
TC>> there is
TC>> another issues with it ans sofia in case proxy-media true.
TC>>
TC>>
TC>
TC>bool FSH323Connection::OnReceivedProgress(const H323SignalPDU &pdu)
TC>{
TC>        PTRACE(4, "mod_h323\t======>FSH323Connection::OnReceivedProgress");
TC>        m_txAudioOpened.Wait();
TC>        switch_channel_mark_pre_answered(m_fsChannel);
TC>        return true;
TC>}
TC>
TC>
TC>
TC>so for me the workaround for this was:
TC>
TC>
TC>
TC>
TC>bool FSH323Connection::OnReceivedProgress(const H323SignalPDU &pdu)
TC>{
TC>        PTRACE(4, "mod_h323\t======>FSH323Connection::OnReceivedProgress");
TC>
TC>        PTRACE(4, "mod_h323\t======>FSH323Connection::OnReceivedProgress -
TC>disabled pre_answer!!!!");
TC>
TC>        //m_txAudioOpened.Wait();
TC>        //switch_channel_mark_pre_answered(m_fsChannel);
TC>        return true;
TC>}
TC>

in that chase wee are not hear anything going inband if receive progress ind from called h323 endpoint, 
there will bee ringback, for exmaple mobule fone then it out of network. if you dont need
this make this changes until i fix it.

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From Prometheus001 at gmx.net  Fri Oct 23 06:08:40 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 23 Oct 2009 15:08:40 +0200
Subject: [Freeswitch-users] Force registering external gateways though
	OpenSIPS load balancer
Message-ID: <4AE1AAD8.9070908@gmx.net>

Hello,

in a freeswitch cluster (FS1 and FS2) behind an OpenSIPS I want
Freeswitch to register to external gateways through the OpenSIPS load
balancer, in order to later receive incoming calls through the load
balancer.
Is there a way to tell Freeswitch in it's Gateway definition to define
an additional path (e.g. fs_path) for it's registration, so that all
registrations go via the OpenSIPS load balancer?
Current flow
    FS1(register) => external_gateway (1st FS Machine)
    external_gateway(invite) =>FS1
    FS2(register) => external_gateway (2nd FS Machine)
    external_gateway(invite) =>FS2

Desired flow
    FS1(register) => OpenSIPS => external_gateway (1st FS Machine)
    external_gateway(invite) => OpenSIPS => FS1
    FS2(register) => OpenSIPS => external_gateway (2nd FS Machine)
    external_gateway(invite) => OpenSIPS =>FS2
I tried to add "fs_path" to the gateway definition, but had no success.
It still registered directly.

Maybe by adding via route information or modifying contact header?

Best regards
Peter.


From tculjaga at gmail.com  Fri Oct 23 06:14:55 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 23 Oct 2009 15:14:55 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910231646210.29582@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>
	<65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
	<Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
	<Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>
	<65d96fc80910230452s6b4a179aqa5bc41dc34efc403@mail.gmail.com>
	<Pine.LNX.4.64.0910231610220.21492@srv.icf.org.ru>
	<65d96fc80910230538w7105be6fve14956ab3b247bf6@mail.gmail.com>
	<Pine.LNX.4.64.0910231646210.29582@srv.icf.org.ru>
Message-ID: <65d96fc80910230614m2b2504d0r92006b78d267a399@mail.gmail.com>

>
>
> TC>
> TC>bool FSH323Connection::OnReceivedProgress(const H323SignalPDU &pdu)
> TC>{
> TC>        PTRACE(4,
> "mod_h323\t======>FSH323Connection::OnReceivedProgress");
> TC>
> TC>        PTRACE(4, "mod_h323\t======>FSH323Connection::OnReceivedProgress
> -
> TC>disabled pre_answer!!!!");
> TC>
> TC>        //m_txAudioOpened.Wait();
> TC>        //switch_channel_mark_pre_answered(m_fsChannel);
> TC>        return true;
> TC>}
> TC>
>
> in that chase wee are not hear anything going inband if receive progress
> ind from called h323 endpoint,
> there will bee ringback, for exmaple mobule fone then it out of network. if
> you dont need
> this make this changes until i fix it.
>
>

not true, because you have mediaWaitForConnect = false... the terminating
endpoint can send media before H.225 connect  message.... and this actually
works well :P



  7.317880   10.4.62.89 -> 10.4.62.7    SIP/SDP Request: INVITE
sip:00914392122 at singtel, with session description
  7.318319    10.4.62.7 -> 10.4.62.89   SIP Status: 100 Trying
  7.331430    10.4.62.7 -> 10.4.62.89   SIP Status: 407 Proxy Authentication
Required
  7.339420   10.4.62.89 -> 10.4.62.7    SIP Request: ACK
sip:00914392122 at singtel
  7.345078   10.4.62.89 -> 10.4.62.7    SIP/SDP Request: INVITE
sip:00914392122 at singtel, with session description
  7.345378    10.4.62.7 -> 10.4.62.89   SIP Status: 100 Trying
  7.387166    10.4.62.7 -> 10.4.4.254   H.225.0 CS: setup OpenLogicalChannel

  7.388636   10.4.4.254 -> 10.4.62.7    H.225.0 CS: callProceeding
  9.389852   10.4.4.254 -> 10.4.62.7    H.225.0 CS: progress
 10.639897   10.4.4.254 -> 10.4.62.7    H.225.0 CS: alerting
 10.651322    10.4.62.7 -> 10.4.62.89   SIP Status: 180 Ringing
 10.653932    10.4.62.7 -> 10.4.198.113 H.245 terminalCapabilitySet
 10.654565    10.4.62.7 -> 10.4.198.113 H.245 masterSlaveDetermination
 10.659757 10.4.198.113 -> 10.4.62.7    H.245 terminalCapabilitySet
 10.659814 10.4.198.113 -> 10.4.62.7    H.245 masterSlaveDetermination
 10.660161 10.4.198.113 -> 10.4.62.7    H.245 terminalCapabilitySetAck
 10.660238 10.4.198.113 -> 10.4.62.7    H.245 masterSlaveDeterminationAck
 10.666028    10.4.62.7 -> 10.4.198.113 H.245 terminalCapabilitySetAck
 10.670388    10.4.62.7 -> 10.4.198.113 H.245 masterSlaveDeterminationAck
 10.674693 10.4.198.113 -> 10.4.62.7    H.245 openLogicalChannel (g711A)
 10.682410    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 1
#678: OLC found 10.4.62.7/10.4.198.113/129
 10.683902    10.4.62.7 -> 10.4.198.113 H.245 openLogicalChannelAck
 10.687378    10.4.62.7 -> 10.4.198.113 H.245 openLogicalChannel (g711A)
#723: OLC found 10.4.198.113/10.4.62.7/108
 10.691579 10.4.198.113 -> 10.4.62.7    H.245 openLogicalChannelAck
 10.778413  10.4.142.38 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=0, Time=24640
 10.798476  10.4.142.38 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=1, Time=24800
 10.818432  10.4.142.38 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=2, Time=24960

<-------------- snip ------------->

 13.298358  10.4.142.38 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=126, Time=44800
 13.318460  10.4.142.38 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=127, Time=44960
 13.338405  10.4.142.38 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=128, Time=45120
 13.358353  10.4.142.38 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=129, Time=45280
 13.369984   10.4.4.254 -> 10.4.62.7    H.225.0 CS: connect
 13.378381  10.4.142.38 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=130, Time=45440
 13.382330    10.4.62.7 -> 10.4.62.89   SIP/SDP Status: 200 OK, with session
description
 13.388833    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 3
 13.389123    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 3
 13.396419   10.4.62.89 -> 10.4.62.7    SIP Request: ACK
sip:00914392122 at 10.4.62.7:5060;transport=udp
 13.398457  10.4.142.38 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=131, Time=45600
 13.405954   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMU,
SSRC=0xDEF4B36, Seq=27943, Time=991142687
 13.418401  10.4.142.38 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=132, Time=45760
 13.425864   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMU,
SSRC=0xDEF4B36, Seq=27944, Time=991142847
 13.438360  10.4.142.38 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=133, Time=45920
 13.438570    10.4.62.7 -> 10.4.62.89   RTP PT=ITU-T G.711 PCMA,
SSRC=0x172DD4B, Seq=46377, Time=640
 13.446202   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xDEF4B36, Seq=27945, Time=991143007
 13.458320  10.4.142.38 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=134, Time=46080
 13.458467    10.4.62.7 -> 10.4.62.89   RTP PT=ITU-T G.711 PCMA,
SSRC=0x172DD4B, Seq=46378, Time=800
 13.459008    10.4.62.7 -> 10.4.142.38  RTP PT=ITU-T G.711 PCMA,
SSRC=0xB9D8D8, Seq=1379, Time=991143007
 13.466010   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xDEF4B36, Seq=27946, Time=991143167
 13.478408  10.4.142.38 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=135, Time=46240
 13.478470    10.4.62.7 -> 10.4.142.38  RTP PT=ITU-T G.711 PCMA,
SSRC=0xB9D8D8, Seq=1380, Time=991143167
 13.478749    10.4.62.7 -> 10.4.62.89   RTP PT=ITU-T G.711 PCMA,
SSRC=0x172DD4B, Seq=46379, Time=960

 <------------ snip -------------->

 15.098561    10.4.62.7 -> 10.4.62.89   RTP PT=ITU-T G.711 PCMA,
SSRC=0x172DD4B, Seq=46460, Time=13920
 15.099011    10.4.62.7 -> 10.4.142.38  RTP PT=ITU-T G.711 PCMA,
SSRC=0xB9D8D8, Seq=1461, Time=991156127
 15.105847   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xDEF4B36, Seq=28028, Time=991156287
 15.118353  10.4.142.38 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=217, Time=59360
 15.118434    10.4.62.7 -> 10.4.62.89   RTP PT=ITU-T G.711 PCMA,
SSRC=0x172DD4B, Seq=46461, Time=14080
 15.119540    10.4.62.7 -> 10.4.142.38  RTP PT=ITU-T G.711 PCMA,
SSRC=0xB9D8D8, Seq=1462, Time=991156287
 15.122951 10.4.198.113 -> 10.4.62.7    H.245 closeLogicalChannel
 15.122986 10.4.198.113 -> 10.4.62.7    H.245 endSessionCommand
 15.125003    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 3
 15.125257    10.4.62.7 -> 10.4.198.113 H.245 closeLogicalChannelAck
 15.125857   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xDEF4B36, Seq=28029, Time=991156447
 15.127900    10.4.62.7 -> 10.4.198.113 H.245 endSessionCommand
 15.128461    10.4.62.7 -> 10.4.4.254   H.225.0 CS: releaseComplete
 15.138328  10.4.142.38 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=218, Time=59520
 15.139550    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 3
 15.142189    10.4.62.7 -> 10.4.62.89   SIP Request: BYE
sip:1001 at 10.4.62.89<sip%3A1001 at 10.4.62.89>
;transport=udp
 15.145990   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xDEF4B36, Seq=28030, Time=991156607
 15.146010    10.4.62.7 -> 10.4.62.89   ICMP Destination unreachable (Port
unreachable)
 15.150016   10.4.62.89 -> 10.4.62.7    SIP Status: 200 OK



 so the real solution is to implement a check for CallProceeding , Progress
and Facility message whether it has a faststart element included. It it is
true than you might start pre_answer.


also, i don't see any handling of Call Proceeding ... what if there is a
fastStart element in CallProceeding message? :)


T.

T.
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From zyyzj2 at gmail.com  Thu Oct 22 23:40:57 2009
From: zyyzj2 at gmail.com (zyyzj)
Date: Fri, 23 Oct 2009 14:40:57 +0800
Subject: [Freeswitch-users] dynamic gateway
Message-ID: <61ee6b990910222340r49ae8fa4oafa1bd4acfecd71c@mail.gmail.com>

hello, all,

    I want bridge call to gateway without register.

    Example:

    <action application="bridge" data="sofia/sipinterface_1/$
1 at 192.168.111.101:5060"/>

    Error is : sofia_reg.c:1568 No Matching gateway found

    My purpose is the gateway is realtime changed from script(etc lua,
php...).

    How can I do that?


-- 
We Create The World!
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From sharad at coraltele.com  Fri Oct 23 04:36:05 2009
From: sharad at coraltele.com (sharad)
Date: Fri, 23 Oct 2009 17:06:05 +0530
Subject: [Freeswitch-users] How to reset the personal greeting
Message-ID: <002001ca53d5$01b5c6b0$0c04a8c0@compaq77db609e>

Hi

I am new to Freeswitch so my question may be a stupid question.

I just want to know how to disable the personal greeting to the default one. One user has recorded his personal greeting now how can he make this default.

I could not find any option for the same.

Plz advice.

Thanks & regards
Sharad garg
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From mattdfong at gmail.com  Fri Oct 23 06:46:53 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Fri, 23 Oct 2009 20:46:53 +0700
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
In-Reply-To: <4256bf830910222019x4bd7929eoadcb757294cc37d2@mail.gmail.com>
References: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
	<4256bf830908041151l55980e3fifb85c75b87535426@mail.gmail.com>
	<3D0B855A-5B2C-4C6F-9306-64B0ECE97091@jerris.com>
	<4256bf830910180725y72539d98gb138934afc18d58b@mail.gmail.com>
	<A5ECDCD9-1A12-4A31-89F0-E6F4C79FE9E9@jerris.com>
	<4256bf830910182148mf7f2bf9vdcccd687e0caa1e7@mail.gmail.com>
	<87f2f3b90910191144q25750bayfd7a7448e82e78ff@mail.gmail.com>
	<4256bf830910221042l68ea769cg532bbd339bae0790@mail.gmail.com>
	<191c3a030910221226j167e0adw5f631f759cd7cfa0@mail.gmail.com>
	<4256bf830910222019x4bd7929eoadcb757294cc37d2@mail.gmail.com>
Message-ID: <4256bf830910230646j506e3629t7223e4fdc04026ce@mail.gmail.com>

Is there a way I can reset the forwardvars for a channel or execute some
action that would reset this value? I'm pretty certain there is no endless
looping going on in my dial plan.
Thanks.

--matt

On Fri, Oct 23, 2009 at 10:19 AM, Matthew Fong <mattdfong at gmail.com> wrote:

> Hi Anthm,
> Thanks for the response. I'll use pastebin from now on.
>
> How would I go about fixing this so my calls do not get hung-up?
>
> --matt
>
>
> On Fri, Oct 23, 2009 at 2:26 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> 1) please do not flood emails with traces reference pastebins
>> 2) the error is a routing loop being prevented so you are looping the call
>> into your same box too many times.
>>
>> You could have figured this out your self by looking at one of the lines
>> of code where it reported the hangup.
>>
>>  if (!switch_strlen_zero(forwardvar)) {
>>         forwardval = atoi(forwardvar) - 1;
>>     }
>>     if (forwardval <= 0) {
>>         switch_channel_hangup(channel,
>> SWITCH_CAUSE_EXCHANGE_ROUTING_ERROR);
>>         return SWITCH_STATUS_FALSE;
>>
>>     }
>>
>>
>> On Thu, Oct 22, 2009 at 12:42 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>>
>>> I figured out how to turn debug mode on. For some reason F8 didn't work,
>>> so I just set the default debug mode to DEBUG in switch.conf.xml
>>>
>>> http://pastebin.freeswitch.org/10805
>>> <http://pastebin.freeswitch.org/10805>So my program does more or less
>>> the same sequence for an Agent every time when connected to another party.
>>> They uuid_kill the bridged parties uuid, and then are transfered to an
>>> extension that park's the agent (using the transfer_after_bridge variable --
>>> in my dialplan extn 1998). In some instances the agent is then transfered to
>>> another extension, played a wav file, and then park'ed again
>>>
>>> Sometimes, FreeSWITCH unexpectidly hangsup the Agent channel citing an
>>> EXCHANGE_ROUTING_ERROR. It seems (to me to happen at random times)...and
>>> it's more prevelent with some did providers than others. With
>>> didforsale.org it happens about one ever 5 bridges. With icall.com it
>>> only happens about once every 30 bridges. With IPKall it happens rarely. I
>>> can't figure it out, and it's driving me crazy :)
>>>
>>>
>>> A GOOD UUID_KILL SEQUENCE:
>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_ivr_play_say.c:1432 done
>>> playing file
>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:340
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXECUTE going to
>>> sleep
>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>> CS_HIBERNATE
>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:355
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE
>>> 2009-10-22 17:22:31.741845 [DEBUG] mod_sofia.c:160 sofia/external/+
>>> 14158867717 at 199.173.100.16:5060 SOFIA HIBERNATE
>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_ivr_bridge.c:650
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>> CS_HIBERNATE -> CS_RESET
>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:355
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE going
>>> to sleep
>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>> CS_RESET
>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:336
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET
>>> 2009-10-22 17:22:31.741845 [DEBUG] mod_sofia.c:145 sofia/external/+
>>> 14158867717 at 199.173.100.16:5060 SOFIA RESET
>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_ivr_bridge.c:637
>>> sofia/external/+14158867717 at 199.173.100.16:5060 CUSTOM RESET
>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:67
>>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard RESET
>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:336
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET going to
>>> sleep
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:340
>>> (sofia/external/14159927717) State EXECUTE going to sleep
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/14159927717) Running State Change CS_HIBERNATE
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:355
>>> (sofia/external/14159927717) State HIBERNATE
>>> 2009-10-22 17:22:31.748086 [DEBUG] mod_sofia.c:160
>>> sofia/external/14159927717 SOFIA HIBERNATE
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:650
>>> (sofia/external/14159927717) State Change CS_HIBERNATE -> CS_RESET
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/14159927717 [BREAK]
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:355
>>> (sofia/external/14159927717) State HIBERNATE going to sleep
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/14159927717) Running State Change CS_RESET
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:336
>>> (sofia/external/14159927717) State RESET
>>> 2009-10-22 17:22:31.748086 [DEBUG] mod_sofia.c:145
>>> sofia/external/14159927717 SOFIA RESET
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:637
>>> sofia/external/14159927717 CUSTOM RESET
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:642
>>> (sofia/external/14159927717) State Change CS_RESET -> CS_SOFT_EXECUTE
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/14159927717 [BREAK]
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:336
>>> (sofia/external/14159927717) State RESET going to sleep
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/14159927717) Running State Change CS_SOFT_EXECUTE
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:346
>>> (sofia/external/14159927717) State SOFT_EXECUTE
>>> 2009-10-22 17:22:31.748086 [DEBUG] mod_sofia.c:442 SOFIA TRANSMIT
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:660
>>> sofia/external/14159927717 CUSTOM SOFT_EXECUTE
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_originate.c:674
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_RESET
>>> -> CS_SOFT_EXECUTE
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:120
>>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message
>>> [DISPLAY]
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:120
>>> sofia/external/14159927717 receive message [DISPLAY]
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:924
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>> CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:968
>>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message [BRIDGE]
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:630 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:975
>>> sofia/external/14159927717 receive message [BRIDGE]
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:630 Send signal
>>> sofia/external/14159927717 [BREAK]
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:1019
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:22:31.749553 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>> CS_EXCHANGE_MEDIA
>>> 2009-10-22 17:22:31.749553 [DEBUG] switch_core_state_machine.c:343
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
>>> 2009-10-22 17:22:31.749553 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK
>>> multipler is being overrided to1
>>> 2009-10-22 17:22:34.300662 [INFO] switch_cpp.cpp:1116 BG_AG_GEN_CALLS,
>>> Agents total:1 avail:0 OutChan:0 AR:0 +-Calls:0:Q Call NYNoc
>>> multipler is being overrided to1
>>> 2009-10-22 17:22:39.303041 [INFO] switch_cpp.cpp:1116 BG_AG_GEN_CALLS,
>>> Agents total:1 avail:0 OutChan:0 AR:0 +-Calls:0:Q Call NYNoc
>>> 2009-10-22 17:22:40.626379 [NOTICE] sofia.c:328 Hangup
>>> sofia/external/14159927717 [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
>>> 2009-10-22 17:22:40.626379 [DEBUG] switch_channel.c:1810 Send signal
>>> sofia/external/14159927717 [KILL]
>>> 2009-10-22 17:22:40.626379 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/14159927717 [BREAK]
>>> active_true=1
>>> --contact NOT in playmsg or conference, update AGENT
>>> 2009-10-22 17:22:40.626379 [DEBUG] switch_core_state_machine.c:437 thread
>>> mismatch skipping state handler.
>>> 2009-10-22 17:22:40.626379 [INFO] switch_cpp.cpp:1116
>>> EVENTC_HANGUP_CONTACT, Q_CONTACT fired for agent:1/1 for qcall:1
>>> contact:4801442/4
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:440
>>> sofia/external/14159927717 ending bridge by request from write function
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:495
>>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message
>>> [UNBRIDGE]
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:630 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
>>> DONE [sofia/external/+14158867717 at 199.173.100.16:5060]
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:523 Send signal
>>> sofia/external/14159927717 [BREAK]
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr.c:1361 (sofia/external/+
>>> 14158867717 at 199.173.100.16:5060) State Change CS_EXCHANGE_MEDIA ->
>>> CS_ROUTING
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr.c:1365 sofia/external/+
>>> 14158867717 at 199.173.100.16:5060 receive message [TRANSFER]
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:630 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:22:40.639760 [NOTICE] switch_ivr.c:1367 Transfer
>>> sofia/external/+14158867717 at 199.173.100.16:5060 to xml[1998 at default]
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:343
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
>>> going to sleep
>>>  2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>> CS_ROUTING
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:333
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State ROUTING
>>> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:130 sofia/external/+
>>> 14158867717 at 199.173.100.16:5060 SOFIA ROUTING
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:78
>>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard ROUTING
>>> 2009-10-22 17:22:40.639760 [INFO] mod_dialplan_xml.c:391 Processing
>>> +14158867717->1998 in context default
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->tod_example] continue=true
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (PASS)
>>> [tod_example] ${strftime(%w)}(4) =~ /^([1-5])$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (PASS)
>>> [tod_example] ${strftime(%H%M)}(1722) =~ /^((09|1[0-7])[0-5][0-9]|1800)$/
>>> break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Action
>>> set(open=true)
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->ivr_demo] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [ivr_demo] destination_number(1998) =~ /^5000$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1975] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1975] destination_number(1998) =~ /^1975$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1971] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1971] destination_number(1998) =~ /^1971$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1972] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1972] destination_number(1998) =~ /^1972$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1920] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1920] destination_number(1998) =~ /^1920$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1981] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1981] destination_number(1998) =~ /^1981$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1982] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1982] destination_number(1998) =~ /^1982$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1481] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1481] destination_number(1998) =~ /^1481$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1989] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1989] destination_number(1998) =~ /^1989$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1990] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1990] destination_number(1998) =~ /^1990$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1991] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1991] destination_number(1998) =~ /^1991$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1992] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1992] destination_number(1998) =~ /^1992$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1993] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1993] destination_number(1998) =~ /^1993$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1994] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1994] destination_number(1998) =~ /^1994$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1995] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1995] destination_number(1998) =~ /^1995$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1996] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1996] destination_number(1998) =~ /^1996$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1997] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1997] destination_number(1998) =~ /^1997$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1998] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (PASS)
>>> [1998] destination_number(1998) =~ /^1998$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Action
>>> set(transfer_after_bridge=1998)
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Action park()
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:114
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>> CS_ROUTING -> CS_EXECUTE
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:333
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State ROUTING going to
>>> sleep
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>> CS_EXECUTE
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:340
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXECUTE
>>> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:173 sofia/external/+
>>> 14158867717 at 199.173.100.16:5060 SOFIA EXECUTE
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:151
>>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard EXECUTE
>>> EXECUTE sofia/external/+14158867717 at 199.173.100.16:5060 set(open=true)
>>> 2009-10-22 17:22:40.639760 [DEBUG] mod_dptools.c:752 sofia/external/+
>>> 14158867717 at 199.173.100.16:5060 SET [open]=[true]
>>> EXECUTE sofia/external/+14158867717 at 199.173.100.16:5060set(transfer_after_bridge=1998)
>>> 2009-10-22 17:22:40.639760 [DEBUG] mod_dptools.c:752 sofia/external/+
>>> 14158867717 at 199.173.100.16:5060 SET [transfer_after_bridge]=[1998]
>>> EXECUTE sofia/external/+14158867717 at 199.173.100.16:5060 park()
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:446
>>> sofia/external/14159927717 ending bridge by request from read function
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
>>> DONE [sofia/external/14159927717]
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:523 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> subtracting 9 seconds for hh_call: 1095422
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:346
>>> (sofia/external/14159927717) State SOFT_EXECUTE going to sleep
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/14159927717) Running State Change CS_HANGUP
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:464
>>> (sofia/external/14159927717) State HANGUP
>>> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:344 Channel
>>> sofia/external/14159927717 hanging up, cause: NORMAL_CLEARING
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:46
>>> sofia/external/14159927717 Standard HANGUP, cause: NORMAL_CLEARING
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:464
>>> (sofia/external/14159927717) State HANGUP going to sleep
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:325
>>> (sofia/external/14159927717) State Change CS_HANGUP -> CS_REPORTING
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/14159927717 [BREAK]
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/14159927717) Running State Change CS_REPORTING
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:555
>>> (sofia/external/14159927717) State REPORTING
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:53
>>> sofia/external/14159927717 Standard REPORTING, cause: NORMAL_CLEARING
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:555
>>> (sofia/external/14159927717) State REPORTING going to sleep
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:319
>>> (sofia/external/14159927717) State Change CS_REPORTING -> CS_DESTROY
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/14159927717 [BREAK]
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:1069 Session 6
>>> (sofia/external/14159927717) Locked, Waiting on external entities
>>> 2009-10-22 17:22:40.639760 [NOTICE] switch_core_session.c:1087 Session 6
>>> (sofia/external/14159927717) Ended
>>> 2009-10-22 17:22:40.639760 [NOTICE] switch_core_session.c:1089 Close
>>> Channel sofia/external/14159927717 [CS_DESTROY]
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:401
>>> (sofia/external/14159927717) Running State Change CS_DESTROY
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:412
>>> (sofia/external/14159927717) State DESTROY
>>> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:261
>>> sofia/external/14159927717 SOFIA DESTROY
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:60
>>> sofia/external/14159927717 Standard DESTROY
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:412
>>> (sofia/external/14159927717) State DESTROY going to sleep
>>> 2009-10-22 17:22:40.639760 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>>> teliax.com hc:NORMAL_CLEARING du:9 cn:sofia/external/14159927717
>>>
>>>
>>>
>>> A BAD UUID_KILL SEQUENCE: (where the agent is hungup, unexpectedly)
>>>
>>> --bridged in: 0.005821 seconds
>>> 2009-10-22 17:23:26.718930 [INFO] switch_cpp.cpp:1116 AG_CBRIDGE,
>>> connected agent:1/1 contact:4801439/22/1/1
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_play_say.c:1432 done
>>> playing file
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:340
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXECUTE going to
>>> sleep
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>> CS_HIBERNATE
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE
>>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:160 sofia/external/+
>>> 14158867717 at 199.173.100.16:5060 SOFIA HIBERNATE
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:650
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>> CS_HIBERNATE -> CS_RESET
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE going
>>> to sleep
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>> CS_RESET
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET
>>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:145 sofia/external/+
>>> 14158867717 at 199.173.100.16:5060 SOFIA RESET
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:637
>>> sofia/external/+14158867717 at 199.173.100.16:5060 CUSTOM RESET
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:67
>>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard RESET
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET going to
>>> sleep
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:340
>>> (sofia/external/14159927717) State EXECUTE going to sleep
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/14159927717) Running State Change CS_HIBERNATE
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
>>> (sofia/external/14159927717) State HIBERNATE
>>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:160
>>> sofia/external/14159927717 SOFIA HIBERNATE
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:650
>>> (sofia/external/14159927717) State Change CS_HIBERNATE -> CS_RESET
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/14159927717 [BREAK]
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
>>> (sofia/external/14159927717) State HIBERNATE going to sleep
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/14159927717) Running State Change CS_RESET
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
>>> (sofia/external/14159927717) State RESET
>>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:145
>>> sofia/external/14159927717 SOFIA RESET
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:637
>>> sofia/external/14159927717 CUSTOM RESET
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:642
>>> (sofia/external/14159927717) State Change CS_RESET -> CS_SOFT_EXECUTE
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/14159927717 [BREAK]
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
>>> (sofia/external/14159927717) State RESET going to sleep
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/14159927717) Running State Change CS_SOFT_EXECUTE
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:346
>>> (sofia/external/14159927717) State SOFT_EXECUTE
>>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:442 SOFIA TRANSMIT
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:660
>>> sofia/external/14159927717 CUSTOM SOFT_EXECUTE
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_originate.c:674
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_RESET
>>> -> CS_SOFT_EXECUTE
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:120
>>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message
>>> [DISPLAY]
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:120
>>> sofia/external/14159927717 receive message [DISPLAY]
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:924
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>> CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:968
>>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message [BRIDGE]
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:630 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:975
>>> sofia/external/14159927717 receive message [BRIDGE]
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:630 Send signal
>>> sofia/external/14159927717 [BREAK]
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:1019
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>> CS_EXCHANGE_MEDIA
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:343
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
>>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK
>>> 2009-10-22 17:23:26.740065 [DEBUG] switch_core_io.c:234 sofia/external/+
>>> 14158867717 at 199.173.100.16:5060 receive message [TRANSCODING_NECESSARY]
>>> 2009-10-22 17:23:27.319618 [DEBUG] switch_core_media_bug.c:393 Removing
>>> BUG from sofia/external/+14158867717 at 199.173.100.16:5060
>>> multipler is being overrided to1
>>> 2009-10-22 17:23:29.406414 [INFO] switch_cpp.cpp:1116 BG_AG_GEN_CALLS,
>>> Agents total:1 avail:0 OutChan:0 AR:0 +-Calls:0:Q Call NYNoc
>>> 2009-10-22 17:23:31.873352 [NOTICE] mod_commands.c:1523 Hangup
>>> sofia/external/14159927717 [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
>>> 2009-10-22 17:23:31.874411 [INFO] switch_cpp.cpp:1116
>>> EVENTC_HANGUP_CONTACT, Q_CONTACT fired for agent:1/1 for qcall:1
>>> contact:4801439/1
>>> active_true=1
>>> --contact NOT in playmsg or conference, update AGENT
>>> 2009-10-22 17:23:31.875475 [DEBUG] switch_channel.c:1810 Send signal
>>> sofia/external/14159927717 [KILL]
>>> 2009-10-22 17:23:31.875475 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/14159927717 [BREAK]
>>> 2009-10-22 17:23:31.875475 [DEBUG] switch_core_state_machine.c:437 thread
>>> mismatch skipping state handler.
>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:446
>>> sofia/external/14159927717 ending bridge by request from read function
>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
>>> DONE [sofia/external/14159927717]
>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:523 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:495
>>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message
>>> [UNBRIDGE]
>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_session.c:630 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
>>> DONE [sofia/external/+14158867717 at 199.173.100.16:5060]
>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:523 Send signal
>>> sofia/external/14159927717 [BREAK]
>>> 2009-10-22 17:23:31.882153 [NOTICE] switch_ivr.c:1268 Hangup
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [CS_EXCHANGE_MEDIA]
>>> [EXCHANGE_ROUTING_ERROR]
>>>  2009-10-22 17:23:31.882153 [DEBUG] switch_channel.c:1810 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [KILL]
>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_state_machine.c:464
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HANGUP
>>> 2009-10-22 17:23:31.882153 [DEBUG] mod_sofia.c:344 Channel
>>> sofia/external/+14158867717 at 199.173.100.16:5060 hanging up, cause:
>>> EXCHANGE_ROUTING_ERROR
>>> 2009-10-22 17:23:31.882153 [DEBUG] mod_sofia.c:382 Sending BYE to
>>> sofia/external/+14158867717 at 199.173.100.16:5060
>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_state_machine.c:46
>>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard HANGUP, cause:
>>> EXCHANGE_ROUTING_ERROR
>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_state_machine.c:464
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HANGUP going to
>>> sleep
>>> 2009-10-22 17:23:31.884357 [INFO] switch_cpp.cpp:1116 HH_HANGUP,
>>> agent:1/1 uuid: f8604694-4e44-4c4c-9805-722b3e9d6b68
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:346
>>> (sofia/external/14159927717) State SOFT_EXECUTE going to sleep
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/14159927717) Running State Change CS_HANGUP
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:464
>>> (sofia/external/14159927717) State HANGUP
>>> 2009-10-22 17:23:31.885412 [DEBUG] mod_sofia.c:344 Channel
>>> sofia/external/14159927717 hanging up, cause: NORMAL_CLEARING
>>> 2009-10-22 17:23:31.885412 [DEBUG] mod_sofia.c:382 Sending BYE to
>>> sofia/external/14159927717
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:46
>>> sofia/external/14159927717 Standard HANGUP, cause: NORMAL_CLEARING
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:464
>>> (sofia/external/14159927717) State HANGUP going to sleep
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:325
>>> (sofia/external/14159927717) State Change CS_HANGUP -> CS_REPORTING
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/14159927717 [BREAK]
>>> subtracting 5 seconds for hh_call: 1095423
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/14159927717) Running State Change CS_REPORTING
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:555
>>> (sofia/external/14159927717) State REPORTING
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:53
>>> sofia/external/14159927717 Standard REPORTING, cause: NORMAL_CLEARING
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:555
>>> (sofia/external/14159927717) State REPORTING going to sleep
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:319
>>> (sofia/external/14159927717) State Change CS_REPORTING -> CS_DESTROY
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/14159927717 [BREAK]
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_session.c:1069 Session 7
>>> (sofia/external/14159927717) Locked, Waiting on external entities
>>> 2009-10-22 17:23:31.885412 [NOTICE] switch_core_session.c:1087 Session 7
>>> (sofia/external/14159927717) Ended
>>> 2009-10-22 17:23:31.885412 [NOTICE] switch_core_session.c:1089 Close
>>> Channel sofia/external/14159927717 [CS_DESTROY]
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:401
>>> (sofia/external/14159927717) Running State Change CS_DESTROY
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:412
>>> (sofia/external/14159927717) State DESTROY
>>> 2009-10-22 17:23:31.885412 [DEBUG] mod_sofia.c:261
>>> sofia/external/14159927717 SOFIA DESTROY
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:60
>>> sofia/external/14159927717 Standard DESTROY
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:412
>>> (sofia/external/14159927717) State DESTROY going to sleep
>>> 2009-10-22 17:23:31.885412 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>>> teliax.com hc:NORMAL_CLEARING du:5 cn:sofia/external/14159927717
>>> 2009-10-22 17:23:31.885412 [INFO] switch_cpp.cpp:1116
>>> EVENTC_BILLING_CEIL, subtracting 5 for hh_call:1095423
>>> 2009-10-22 17:23:31.901620 [DEBUG] switch_core_state_machine.c:494 Hangup
>>> Command luarun(hh_hangup.lua f8604694-4e44-4c4c-9805-722b3e9d6b68 1 1):
>>> +OK
>>>
>>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:343
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
>>> going to sleep
>>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>> CS_HANGUP
>>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:442
>>> handler already called, skipping state handler.
>>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:325
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_HANGUP
>>> -> CS_REPORTING
>>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>> CS_REPORTING
>>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:555
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State REPORTING
>>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:53
>>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard REPORTING,
>>> cause: EXCHANGE_ROUTING_ERROR
>>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:555
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State REPORTING going
>>> to sleep
>>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:319
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>> CS_REPORTING -> CS_DESTROY
>>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_session.c:1069 Session 4
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Locked, Waiting on
>>> external entities
>>> 2009-10-22 17:23:31.903742 [NOTICE] switch_core_session.c:1087 Session 4
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Ended
>>> 2009-10-22 17:23:31.903742 [NOTICE] switch_core_session.c:1089 Close
>>> Channel sofia/external/+14158867717 at 199.173.100.16:5060 [CS_DESTROY]
>>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:401
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>> CS_DESTROY
>>> 2009-10-22 17:23:31.904799 [DEBUG] switch_core_state_machine.c:412
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State DESTROY
>>> 2009-10-22 17:23:31.904799 [DEBUG] mod_sofia.c:261 sofia/external/+
>>> 14158867717 at 199.173.100.16:5060 SOFIA DESTROY
>>> 2009-10-22 17:23:31.904799 [DEBUG] switch_core_state_machine.c:60
>>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard DESTROY
>>> 2009-10-22 17:23:31.904799 [DEBUG] switch_core_state_machine.c:412
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State DESTROY going to
>>> sleep
>>>
>>>
>>>
>>> On Tue, Oct 20, 2009 at 1:44 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>>
>>>>
>>>> On Sun, Oct 18, 2009 at 9:48 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>>
>>>>> The debug level logs to the console, right? The pastebin, had log level
>>>>> debug, sofia trace on for external and default, and sofia loglevel all 9. Is
>>>>> there another log enable command I'm missing? It seems loglevel all 9
>>>>> outputs enter and exit functions, but at least to my novice eye, it's not
>>>>> too obvious why freeswitch is sending a BYE to my DID provider.
>>>>>
>>>>> Evidently the console loglevel debug did not take because there isn't a
>>>> single DEBUG level line of output. Please try again. Just remember that
>>>> debug level log messages will have A LOT of information so be patient while
>>>> sifting through it. Of course, feel free to pastebin the output and add a
>>>> link to it in this email thread.
>>>>
>>>> -MC
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From anthony.minessale at gmail.com  Fri Oct 23 07:33:01 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 23 Oct 2009 09:33:01 -0500
Subject: [Freeswitch-users] INFO-Messages: Send_display in pre_answer
	state doesnt work anymore
In-Reply-To: <4AE19D0E.9050301@ewetel.de>
References: <4AE19D0E.9050301@ewetel.de>
Message-ID: <191c3a030910230733i4fce1c55v892056f549f67ed0@mail.gmail.com>

There was a concern with letting this happen before answer.
I changed it in last revision, try it out.
The problem is that the way polycom does it, it violates the RFC.

Also know you can set sip_callee_id_name and sip_callee_id_number before you
send a ring_ready pre_answer or answer to send the display update in that
packet instead of sending a separate one.


On Fri, Oct 23, 2009 at 7:09 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> when I use pre_answer app and right after that the send_display app,
> there is no INFO message anymore.
>
> My dialplan condition:
>
> <condition field="${sip_hangup_phrase}" expression="^Do Not Disturb"
> break="on-true">
>                        <action application="pre_answer"/>
>                        <action application="send_display" data="Bitte
> nicht stoeren"/>
>                        <action application="sleep" data="500"/>
>                        <action application="playback"
> data="ansagen/22.wav"/>
> </condition>
>
> changing "pre_answer" to "answer" works, but I don't want to answer the
> call for error messages.
>
> I use latest SVN trunk.
>
> regards
> helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
>
> iD8DBQFK4Z0O4tZeNddg3dwRAkB/AKCLN9luR0KGunHVGso2t3wJIo4dqQCdGdjp
> CY+yWGyVlyDpDs//WFtCqiA=
> =eG4W
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Fri Oct 23 07:36:31 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 23 Oct 2009 09:36:31 -0500
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
In-Reply-To: <4256bf830910230646j506e3629t7223e4fdc04026ce@mail.gmail.com>
References: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
	<3D0B855A-5B2C-4C6F-9306-64B0ECE97091@jerris.com>
	<4256bf830910180725y72539d98gb138934afc18d58b@mail.gmail.com>
	<A5ECDCD9-1A12-4A31-89F0-E6F4C79FE9E9@jerris.com>
	<4256bf830910182148mf7f2bf9vdcccd687e0caa1e7@mail.gmail.com>
	<87f2f3b90910191144q25750bayfd7a7448e82e78ff@mail.gmail.com>
	<4256bf830910221042l68ea769cg532bbd339bae0790@mail.gmail.com>
	<191c3a030910221226j167e0adw5f631f759cd7cfa0@mail.gmail.com>
	<4256bf830910222019x4bd7929eoadcb757294cc37d2@mail.gmail.com>
	<4256bf830910230646j506e3629t7223e4fdc04026ce@mail.gmail.com>
Message-ID: <191c3a030910230736o7a4201e8s4addd39d9e1c83ae@mail.gmail.com>

you can set the variable max_forwards to a high number.
It does not magically set itself so you must be looping through many
transfers at least.


On Fri, Oct 23, 2009 at 8:46 AM, Matthew Fong <mattdfong at gmail.com> wrote:

> Is there a way I can reset the forwardvars for a channel or execute some
> action that would reset this value? I'm pretty certain there is no endless
> looping going on in my dial plan.
> Thanks.
>
> --matt
>
>
> On Fri, Oct 23, 2009 at 10:19 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>
>> Hi Anthm,
>> Thanks for the response. I'll use pastebin from now on.
>>
>> How would I go about fixing this so my calls do not get hung-up?
>>
>> --matt
>>
>>
>> On Fri, Oct 23, 2009 at 2:26 AM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> 1) please do not flood emails with traces reference pastebins
>>> 2) the error is a routing loop being prevented so you are looping the
>>> call into your same box too many times.
>>>
>>> You could have figured this out your self by looking at one of the lines
>>> of code where it reported the hangup.
>>>
>>>  if (!switch_strlen_zero(forwardvar)) {
>>>         forwardval = atoi(forwardvar) - 1;
>>>     }
>>>     if (forwardval <= 0) {
>>>         switch_channel_hangup(channel,
>>> SWITCH_CAUSE_EXCHANGE_ROUTING_ERROR);
>>>         return SWITCH_STATUS_FALSE;
>>>
>>>     }
>>>
>>>
>>> On Thu, Oct 22, 2009 at 12:42 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>
>>>> I figured out how to turn debug mode on. For some reason F8 didn't work,
>>>> so I just set the default debug mode to DEBUG in switch.conf.xml
>>>>
>>>> http://pastebin.freeswitch.org/10805
>>>> <http://pastebin.freeswitch.org/10805>So my program does more or less
>>>> the same sequence for an Agent every time when connected to another party.
>>>> They uuid_kill the bridged parties uuid, and then are transfered to an
>>>> extension that park's the agent (using the transfer_after_bridge variable --
>>>> in my dialplan extn 1998). In some instances the agent is then transfered to
>>>> another extension, played a wav file, and then park'ed again
>>>>
>>>> Sometimes, FreeSWITCH unexpectidly hangsup the Agent channel citing an
>>>> EXCHANGE_ROUTING_ERROR. It seems (to me to happen at random times)...and
>>>> it's more prevelent with some did providers than others. With
>>>> didforsale.org it happens about one ever 5 bridges. With icall.com it
>>>> only happens about once every 30 bridges. With IPKall it happens rarely. I
>>>> can't figure it out, and it's driving me crazy :)
>>>>
>>>>
>>>> A GOOD UUID_KILL SEQUENCE:
>>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_ivr_play_say.c:1432 done
>>>> playing file
>>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:340
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXECUTE going
>>>> to sleep
>>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>>> CS_HIBERNATE
>>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:355
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE
>>>> 2009-10-22 17:22:31.741845 [DEBUG] mod_sofia.c:160 sofia/external/+
>>>> 14158867717 at 199.173.100.16:5060 SOFIA HIBERNATE
>>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_ivr_bridge.c:650
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>>> CS_HIBERNATE -> CS_RESET
>>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:355
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE going
>>>> to sleep
>>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>>> CS_RESET
>>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:336
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET
>>>> 2009-10-22 17:22:31.741845 [DEBUG] mod_sofia.c:145 sofia/external/+
>>>> 14158867717 at 199.173.100.16:5060 SOFIA RESET
>>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_ivr_bridge.c:637
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 CUSTOM RESET
>>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:67
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard RESET
>>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:336
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET going to
>>>> sleep
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:340
>>>> (sofia/external/14159927717) State EXECUTE going to sleep
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/14159927717) Running State Change CS_HIBERNATE
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:355
>>>> (sofia/external/14159927717) State HIBERNATE
>>>> 2009-10-22 17:22:31.748086 [DEBUG] mod_sofia.c:160
>>>> sofia/external/14159927717 SOFIA HIBERNATE
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:650
>>>> (sofia/external/14159927717) State Change CS_HIBERNATE -> CS_RESET
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/14159927717 [BREAK]
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:355
>>>> (sofia/external/14159927717) State HIBERNATE going to sleep
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/14159927717) Running State Change CS_RESET
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:336
>>>> (sofia/external/14159927717) State RESET
>>>> 2009-10-22 17:22:31.748086 [DEBUG] mod_sofia.c:145
>>>> sofia/external/14159927717 SOFIA RESET
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:637
>>>> sofia/external/14159927717 CUSTOM RESET
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:642
>>>> (sofia/external/14159927717) State Change CS_RESET -> CS_SOFT_EXECUTE
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/14159927717 [BREAK]
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:336
>>>> (sofia/external/14159927717) State RESET going to sleep
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/14159927717) Running State Change CS_SOFT_EXECUTE
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:346
>>>> (sofia/external/14159927717) State SOFT_EXECUTE
>>>> 2009-10-22 17:22:31.748086 [DEBUG] mod_sofia.c:442 SOFIA TRANSMIT
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:660
>>>> sofia/external/14159927717 CUSTOM SOFT_EXECUTE
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_originate.c:674
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_RESET
>>>> -> CS_SOFT_EXECUTE
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:120
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message
>>>> [DISPLAY]
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:120
>>>> sofia/external/14159927717 receive message [DISPLAY]
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:924
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>>> CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:968
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message
>>>> [BRIDGE]
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:630 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:975
>>>> sofia/external/14159927717 receive message [BRIDGE]
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:630 Send signal
>>>> sofia/external/14159927717 [BREAK]
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:1019
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>>> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:22:31.749553 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>>> CS_EXCHANGE_MEDIA
>>>> 2009-10-22 17:22:31.749553 [DEBUG] switch_core_state_machine.c:343
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
>>>> 2009-10-22 17:22:31.749553 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK
>>>> multipler is being overrided to1
>>>> 2009-10-22 17:22:34.300662 [INFO] switch_cpp.cpp:1116 BG_AG_GEN_CALLS,
>>>> Agents total:1 avail:0 OutChan:0 AR:0 +-Calls:0:Q Call NYNoc
>>>> multipler is being overrided to1
>>>> 2009-10-22 17:22:39.303041 [INFO] switch_cpp.cpp:1116 BG_AG_GEN_CALLS,
>>>> Agents total:1 avail:0 OutChan:0 AR:0 +-Calls:0:Q Call NYNoc
>>>> 2009-10-22 17:22:40.626379 [NOTICE] sofia.c:328 Hangup
>>>> sofia/external/14159927717 [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
>>>> 2009-10-22 17:22:40.626379 [DEBUG] switch_channel.c:1810 Send signal
>>>> sofia/external/14159927717 [KILL]
>>>> 2009-10-22 17:22:40.626379 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/14159927717 [BREAK]
>>>> active_true=1
>>>> --contact NOT in playmsg or conference, update AGENT
>>>> 2009-10-22 17:22:40.626379 [DEBUG] switch_core_state_machine.c:437
>>>> thread mismatch skipping state handler.
>>>> 2009-10-22 17:22:40.626379 [INFO] switch_cpp.cpp:1116
>>>> EVENTC_HANGUP_CONTACT, Q_CONTACT fired for agent:1/1 for qcall:1
>>>> contact:4801442/4
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:440
>>>> sofia/external/14159927717 ending bridge by request from write function
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:495
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message
>>>> [UNBRIDGE]
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:630 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
>>>> DONE [sofia/external/+14158867717 at 199.173.100.16:5060]
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:523 Send signal
>>>> sofia/external/14159927717 [BREAK]
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr.c:1361 (sofia/external/+
>>>> 14158867717 at 199.173.100.16:5060) State Change CS_EXCHANGE_MEDIA ->
>>>> CS_ROUTING
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr.c:1365 sofia/external/+
>>>> 14158867717 at 199.173.100.16:5060 receive message [TRANSFER]
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:630 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:22:40.639760 [NOTICE] switch_ivr.c:1367 Transfer
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 to xml[1998 at default]
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:343
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
>>>> going to sleep
>>>>  2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>>> CS_ROUTING
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:333
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State ROUTING
>>>> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:130 sofia/external/+
>>>> 14158867717 at 199.173.100.16:5060 SOFIA ROUTING
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:78
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard ROUTING
>>>> 2009-10-22 17:22:40.639760 [INFO] mod_dialplan_xml.c:391 Processing
>>>> +14158867717->1998 in context default
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->tod_example] continue=true
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (PASS)
>>>> [tod_example] ${strftime(%w)}(4) =~ /^([1-5])$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (PASS)
>>>> [tod_example] ${strftime(%H%M)}(1722) =~ /^((09|1[0-7])[0-5][0-9]|1800)$/
>>>> break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Action
>>>> set(open=true)
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->ivr_demo] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [ivr_demo] destination_number(1998) =~ /^5000$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1975] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1975] destination_number(1998) =~ /^1975$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1971] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1971] destination_number(1998) =~ /^1971$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1972] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1972] destination_number(1998) =~ /^1972$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1920] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1920] destination_number(1998) =~ /^1920$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1981] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1981] destination_number(1998) =~ /^1981$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1982] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1982] destination_number(1998) =~ /^1982$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1481] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1481] destination_number(1998) =~ /^1481$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1989] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1989] destination_number(1998) =~ /^1989$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1990] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1990] destination_number(1998) =~ /^1990$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1991] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1991] destination_number(1998) =~ /^1991$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1992] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1992] destination_number(1998) =~ /^1992$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1993] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1993] destination_number(1998) =~ /^1993$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1994] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1994] destination_number(1998) =~ /^1994$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1995] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1995] destination_number(1998) =~ /^1995$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1996] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1996] destination_number(1998) =~ /^1996$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1997] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1997] destination_number(1998) =~ /^1997$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1998] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (PASS)
>>>> [1998] destination_number(1998) =~ /^1998$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Action
>>>> set(transfer_after_bridge=1998)
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Action park()
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:114
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>>> CS_ROUTING -> CS_EXECUTE
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:333
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State ROUTING going
>>>> to sleep
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>>> CS_EXECUTE
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:340
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXECUTE
>>>> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:173 sofia/external/+
>>>> 14158867717 at 199.173.100.16:5060 SOFIA EXECUTE
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:151
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard EXECUTE
>>>> EXECUTE sofia/external/+14158867717 at 199.173.100.16:5060 set(open=true)
>>>> 2009-10-22 17:22:40.639760 [DEBUG] mod_dptools.c:752 sofia/external/+
>>>> 14158867717 at 199.173.100.16:5060 SET [open]=[true]
>>>> EXECUTE sofia/external/+14158867717 at 199.173.100.16:5060set(transfer_after_bridge=1998)
>>>> 2009-10-22 17:22:40.639760 [DEBUG] mod_dptools.c:752 sofia/external/+
>>>> 14158867717 at 199.173.100.16:5060 SET [transfer_after_bridge]=[1998]
>>>> EXECUTE sofia/external/+14158867717 at 199.173.100.16:5060 park()
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:446
>>>> sofia/external/14159927717 ending bridge by request from read function
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
>>>> DONE [sofia/external/14159927717]
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:523 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> subtracting 9 seconds for hh_call: 1095422
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:346
>>>> (sofia/external/14159927717) State SOFT_EXECUTE going to sleep
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/14159927717) Running State Change CS_HANGUP
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:464
>>>> (sofia/external/14159927717) State HANGUP
>>>> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:344 Channel
>>>> sofia/external/14159927717 hanging up, cause: NORMAL_CLEARING
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:46
>>>> sofia/external/14159927717 Standard HANGUP, cause: NORMAL_CLEARING
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:464
>>>> (sofia/external/14159927717) State HANGUP going to sleep
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:325
>>>> (sofia/external/14159927717) State Change CS_HANGUP -> CS_REPORTING
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/14159927717 [BREAK]
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/14159927717) Running State Change CS_REPORTING
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:555
>>>> (sofia/external/14159927717) State REPORTING
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:53
>>>> sofia/external/14159927717 Standard REPORTING, cause: NORMAL_CLEARING
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:555
>>>> (sofia/external/14159927717) State REPORTING going to sleep
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:319
>>>> (sofia/external/14159927717) State Change CS_REPORTING -> CS_DESTROY
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/14159927717 [BREAK]
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:1069 Session 6
>>>> (sofia/external/14159927717) Locked, Waiting on external entities
>>>> 2009-10-22 17:22:40.639760 [NOTICE] switch_core_session.c:1087 Session 6
>>>> (sofia/external/14159927717) Ended
>>>> 2009-10-22 17:22:40.639760 [NOTICE] switch_core_session.c:1089 Close
>>>> Channel sofia/external/14159927717 [CS_DESTROY]
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:401
>>>> (sofia/external/14159927717) Running State Change CS_DESTROY
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:412
>>>> (sofia/external/14159927717) State DESTROY
>>>> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:261
>>>> sofia/external/14159927717 SOFIA DESTROY
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:60
>>>> sofia/external/14159927717 Standard DESTROY
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:412
>>>> (sofia/external/14159927717) State DESTROY going to sleep
>>>> 2009-10-22 17:22:40.639760 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>>>> teliax.com hc:NORMAL_CLEARING du:9 cn:sofia/external/14159927717
>>>>
>>>>
>>>>
>>>> A BAD UUID_KILL SEQUENCE: (where the agent is hungup, unexpectedly)
>>>>
>>>> --bridged in: 0.005821 seconds
>>>> 2009-10-22 17:23:26.718930 [INFO] switch_cpp.cpp:1116 AG_CBRIDGE,
>>>> connected agent:1/1 contact:4801439/22/1/1
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_play_say.c:1432 done
>>>> playing file
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:340
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXECUTE going
>>>> to sleep
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>>> CS_HIBERNATE
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE
>>>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:160 sofia/external/+
>>>> 14158867717 at 199.173.100.16:5060 SOFIA HIBERNATE
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:650
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>>> CS_HIBERNATE -> CS_RESET
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE going
>>>> to sleep
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>>> CS_RESET
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET
>>>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:145 sofia/external/+
>>>> 14158867717 at 199.173.100.16:5060 SOFIA RESET
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:637
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 CUSTOM RESET
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:67
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard RESET
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET going to
>>>> sleep
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:340
>>>> (sofia/external/14159927717) State EXECUTE going to sleep
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/14159927717) Running State Change CS_HIBERNATE
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
>>>> (sofia/external/14159927717) State HIBERNATE
>>>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:160
>>>> sofia/external/14159927717 SOFIA HIBERNATE
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:650
>>>> (sofia/external/14159927717) State Change CS_HIBERNATE -> CS_RESET
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/14159927717 [BREAK]
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
>>>> (sofia/external/14159927717) State HIBERNATE going to sleep
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/14159927717) Running State Change CS_RESET
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
>>>> (sofia/external/14159927717) State RESET
>>>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:145
>>>> sofia/external/14159927717 SOFIA RESET
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:637
>>>> sofia/external/14159927717 CUSTOM RESET
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:642
>>>> (sofia/external/14159927717) State Change CS_RESET -> CS_SOFT_EXECUTE
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/14159927717 [BREAK]
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
>>>> (sofia/external/14159927717) State RESET going to sleep
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/14159927717) Running State Change CS_SOFT_EXECUTE
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:346
>>>> (sofia/external/14159927717) State SOFT_EXECUTE
>>>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:442 SOFIA TRANSMIT
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:660
>>>> sofia/external/14159927717 CUSTOM SOFT_EXECUTE
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_originate.c:674
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_RESET
>>>> -> CS_SOFT_EXECUTE
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:120
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message
>>>> [DISPLAY]
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:120
>>>> sofia/external/14159927717 receive message [DISPLAY]
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:924
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>>> CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:968
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message
>>>> [BRIDGE]
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:630 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:975
>>>> sofia/external/14159927717 receive message [BRIDGE]
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:630 Send signal
>>>> sofia/external/14159927717 [BREAK]
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:1019
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>>> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>>> CS_EXCHANGE_MEDIA
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:343
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
>>>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK
>>>> 2009-10-22 17:23:26.740065 [DEBUG] switch_core_io.c:234 sofia/external/+
>>>> 14158867717 at 199.173.100.16:5060 receive message [TRANSCODING_NECESSARY]
>>>> 2009-10-22 17:23:27.319618 [DEBUG] switch_core_media_bug.c:393 Removing
>>>> BUG from sofia/external/+14158867717 at 199.173.100.16:5060
>>>> multipler is being overrided to1
>>>> 2009-10-22 17:23:29.406414 [INFO] switch_cpp.cpp:1116 BG_AG_GEN_CALLS,
>>>> Agents total:1 avail:0 OutChan:0 AR:0 +-Calls:0:Q Call NYNoc
>>>> 2009-10-22 17:23:31.873352 [NOTICE] mod_commands.c:1523 Hangup
>>>> sofia/external/14159927717 [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
>>>> 2009-10-22 17:23:31.874411 [INFO] switch_cpp.cpp:1116
>>>> EVENTC_HANGUP_CONTACT, Q_CONTACT fired for agent:1/1 for qcall:1
>>>> contact:4801439/1
>>>> active_true=1
>>>> --contact NOT in playmsg or conference, update AGENT
>>>> 2009-10-22 17:23:31.875475 [DEBUG] switch_channel.c:1810 Send signal
>>>> sofia/external/14159927717 [KILL]
>>>> 2009-10-22 17:23:31.875475 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/14159927717 [BREAK]
>>>> 2009-10-22 17:23:31.875475 [DEBUG] switch_core_state_machine.c:437
>>>> thread mismatch skipping state handler.
>>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:446
>>>> sofia/external/14159927717 ending bridge by request from read function
>>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
>>>> DONE [sofia/external/14159927717]
>>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:523 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:495
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message
>>>> [UNBRIDGE]
>>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_session.c:630 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
>>>> DONE [sofia/external/+14158867717 at 199.173.100.16:5060]
>>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:523 Send signal
>>>> sofia/external/14159927717 [BREAK]
>>>> 2009-10-22 17:23:31.882153 [NOTICE] switch_ivr.c:1268 Hangup
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [CS_EXCHANGE_MEDIA]
>>>> [EXCHANGE_ROUTING_ERROR]
>>>>  2009-10-22 17:23:31.882153 [DEBUG] switch_channel.c:1810 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [KILL]
>>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_state_machine.c:464
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HANGUP
>>>> 2009-10-22 17:23:31.882153 [DEBUG] mod_sofia.c:344 Channel
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 hanging up, cause:
>>>> EXCHANGE_ROUTING_ERROR
>>>> 2009-10-22 17:23:31.882153 [DEBUG] mod_sofia.c:382 Sending BYE to
>>>> sofia/external/+14158867717 at 199.173.100.16:5060
>>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_state_machine.c:46
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard HANGUP, cause:
>>>> EXCHANGE_ROUTING_ERROR
>>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_state_machine.c:464
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HANGUP going to
>>>> sleep
>>>> 2009-10-22 17:23:31.884357 [INFO] switch_cpp.cpp:1116 HH_HANGUP,
>>>> agent:1/1 uuid: f8604694-4e44-4c4c-9805-722b3e9d6b68
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:346
>>>> (sofia/external/14159927717) State SOFT_EXECUTE going to sleep
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/14159927717) Running State Change CS_HANGUP
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:464
>>>> (sofia/external/14159927717) State HANGUP
>>>> 2009-10-22 17:23:31.885412 [DEBUG] mod_sofia.c:344 Channel
>>>> sofia/external/14159927717 hanging up, cause: NORMAL_CLEARING
>>>> 2009-10-22 17:23:31.885412 [DEBUG] mod_sofia.c:382 Sending BYE to
>>>> sofia/external/14159927717
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:46
>>>> sofia/external/14159927717 Standard HANGUP, cause: NORMAL_CLEARING
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:464
>>>> (sofia/external/14159927717) State HANGUP going to sleep
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:325
>>>> (sofia/external/14159927717) State Change CS_HANGUP -> CS_REPORTING
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/14159927717 [BREAK]
>>>> subtracting 5 seconds for hh_call: 1095423
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/14159927717) Running State Change CS_REPORTING
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:555
>>>> (sofia/external/14159927717) State REPORTING
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:53
>>>> sofia/external/14159927717 Standard REPORTING, cause: NORMAL_CLEARING
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:555
>>>> (sofia/external/14159927717) State REPORTING going to sleep
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:319
>>>> (sofia/external/14159927717) State Change CS_REPORTING -> CS_DESTROY
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/14159927717 [BREAK]
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_session.c:1069 Session 7
>>>> (sofia/external/14159927717) Locked, Waiting on external entities
>>>> 2009-10-22 17:23:31.885412 [NOTICE] switch_core_session.c:1087 Session 7
>>>> (sofia/external/14159927717) Ended
>>>> 2009-10-22 17:23:31.885412 [NOTICE] switch_core_session.c:1089 Close
>>>> Channel sofia/external/14159927717 [CS_DESTROY]
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:401
>>>> (sofia/external/14159927717) Running State Change CS_DESTROY
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:412
>>>> (sofia/external/14159927717) State DESTROY
>>>> 2009-10-22 17:23:31.885412 [DEBUG] mod_sofia.c:261
>>>> sofia/external/14159927717 SOFIA DESTROY
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:60
>>>> sofia/external/14159927717 Standard DESTROY
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:412
>>>> (sofia/external/14159927717) State DESTROY going to sleep
>>>> 2009-10-22 17:23:31.885412 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>>>> teliax.com hc:NORMAL_CLEARING du:5 cn:sofia/external/14159927717
>>>> 2009-10-22 17:23:31.885412 [INFO] switch_cpp.cpp:1116
>>>> EVENTC_BILLING_CEIL, subtracting 5 for hh_call:1095423
>>>> 2009-10-22 17:23:31.901620 [DEBUG] switch_core_state_machine.c:494
>>>> Hangup Command luarun(hh_hangup.lua f8604694-4e44-4c4c-9805-722b3e9d6b68 1
>>>> 1):
>>>> +OK
>>>>
>>>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:343
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
>>>> going to sleep
>>>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>>> CS_HANGUP
>>>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:442
>>>> handler already called, skipping state handler.
>>>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:325
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>>> CS_HANGUP -> CS_REPORTING
>>>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>>> CS_REPORTING
>>>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:555
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State REPORTING
>>>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:53
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard REPORTING,
>>>> cause: EXCHANGE_ROUTING_ERROR
>>>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:555
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State REPORTING going
>>>> to sleep
>>>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:319
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>>> CS_REPORTING -> CS_DESTROY
>>>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_session.c:1069 Session 4
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Locked, Waiting on
>>>> external entities
>>>> 2009-10-22 17:23:31.903742 [NOTICE] switch_core_session.c:1087 Session 4
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Ended
>>>> 2009-10-22 17:23:31.903742 [NOTICE] switch_core_session.c:1089 Close
>>>> Channel sofia/external/+14158867717 at 199.173.100.16:5060 [CS_DESTROY]
>>>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:401
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>>> CS_DESTROY
>>>> 2009-10-22 17:23:31.904799 [DEBUG] switch_core_state_machine.c:412
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State DESTROY
>>>> 2009-10-22 17:23:31.904799 [DEBUG] mod_sofia.c:261 sofia/external/+
>>>> 14158867717 at 199.173.100.16:5060 SOFIA DESTROY
>>>> 2009-10-22 17:23:31.904799 [DEBUG] switch_core_state_machine.c:60
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard DESTROY
>>>> 2009-10-22 17:23:31.904799 [DEBUG] switch_core_state_machine.c:412
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State DESTROY going
>>>> to sleep
>>>>
>>>>
>>>>
>>>> On Tue, Oct 20, 2009 at 1:44 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>>
>>>>>
>>>>> On Sun, Oct 18, 2009 at 9:48 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>>>
>>>>>> The debug level logs to the console, right? The pastebin, had log
>>>>>> level debug, sofia trace on for external and default, and sofia loglevel all
>>>>>> 9. Is there another log enable command I'm missing? It seems loglevel all 9
>>>>>> outputs enter and exit functions, but at least to my novice eye, it's not
>>>>>> too obvious why freeswitch is sending a BYE to my DID provider.
>>>>>>
>>>>>> Evidently the console loglevel debug did not take because there isn't
>>>>> a single DEBUG level line of output. Please try again. Just remember that
>>>>> debug level log messages will have A LOT of information so be patient while
>>>>> sifting through it. Of course, feel free to pastebin the output and add a
>>>>> link to it in this email thread.
>>>>>
>>>>> -MC
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:213-799-1400
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From tculjaga at gmail.com  Fri Oct 23 07:57:30 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 23 Oct 2009 16:57:30 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <65d96fc80910230614m2b2504d0r92006b78d267a399@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
	<Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
	<Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>
	<65d96fc80910230452s6b4a179aqa5bc41dc34efc403@mail.gmail.com>
	<Pine.LNX.4.64.0910231610220.21492@srv.icf.org.ru>
	<65d96fc80910230538w7105be6fve14956ab3b247bf6@mail.gmail.com>
	<Pine.LNX.4.64.0910231646210.29582@srv.icf.org.ru>
	<65d96fc80910230614m2b2504d0r92006b78d267a399@mail.gmail.com>
Message-ID: <65d96fc80910230757h4595f396o2dc1a1bb781d3a8a@mail.gmail.com>

a solution to H323 endpoint => FS => SIP user no audio issue

is to disable a wait for tx Audio ... for  case
SWITCH_MESSAGE_INDICATE_ANSWER:{

//m_txAudioOpened.Wait();


                case SWITCH_MESSAGE_INDICATE_ANSWER:{

                        switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_CONSOLE, "ANSWER: we got Answer event\n");

                        if (switch_channel_test_flag(channel, CF_OUTBOUND))
{

                                switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_CONSOLE, "ANSWER: we got Answer event - CF_OUTBOUND
\n");
                                return SWITCH_STATUS_FALSE;
                        }
                        AnsweringCall(H323Connection::AnswerCallNow);

                        switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_CONSOLE, "ANSWER: suppose the call is Answered Now\n");
                        PTRACE(4, "mod_h323\tMedia started on connection "
<< *this);

                        // test
                        //switch_channel_mark_answered(m_fsChannel);

                        m_rxAudioOpened.Wait();
                        switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_CONSOLE, "ANSWER: wait for m_rxAudioOpened\n");
                        //m_txAudioOpened.Wait();
                        switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_CONSOLE, "ANSWER: we disable wait for m_txAudioOpened\n");

                        switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_CONSOLE, "ANSWER: were waiting for rx/tx AudioOpen\n");

                        if (!switch_channel_test_flag(m_fsChannel,
CF_EARLY_MEDIA)) {

                                switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_CONSOLE, "ANSWER: we have early media\n");

                                PTRACE(4,
"mod_h323\t-------------------->switch_channel_mark_answered(m_fsChannel) "
<< *this);
                                switch_channel_mark_answered(m_fsChannel);
                                switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_CONSOLE, "ANSWER: answered in early Media\n");
                        }
                        break;
                }


Now, I'm able to both originate and terminate cals with 2-way audio...
the signaling looks correct...



outgoing:

1369.425046    10.4.62.7 -> 10.4.62.89   SIP/SDP Request: INVITE
sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>;transport=udp, with session
description
1369.426255    10.4.62.7 -> 10.4.62.31   H.225.0 CS: alerting
1369.435950   10.4.62.89 -> 10.4.62.7    SIP Status: 100 Trying
1369.449065   10.4.62.89 -> 10.4.62.7    SIP Status: 180 Ringing
1369.605109    10.4.62.7 -> 10.4.62.31   H.225.0 CS: progress
OpenLogicalChannel
1369.609788   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
terminalCapabilitySet
1369.610489   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
masterSlaveDetermination
1369.619071    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
terminalCapabilitySet
1369.620349    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
terminalCapabilitySetAck
1369.623215   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
terminalCapabilitySetAck
1369.625591    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
masterSlaveDeterminationAck
1369.628174   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
masterSlaveDeterminationAck
1370.966958   10.4.62.89 -> 10.4.62.7    SIP/SDP Status: 200 OK, with
session description
1370.967431    10.4.62.7 -> 10.4.62.89   SIP Request: ACK
sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>;transport=udp
1370.975172    10.4.62.7 -> 10.4.62.31   H.225.0 CS: connect
1372.354383   10.4.62.89 -> 10.4.62.7    SIP Request: BYE
sip:mod_sofia at 10.4.62.7:5060
1372.355147    10.4.62.7 -> 10.4.62.89   SIP Status: 200 OK
1372.392904    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: releaseComplete
endSessionCommand
1372.397302   10.4.62.31 -> 10.4.62.7    H.225.0 CS: releaseComplete


incoming:


1502.817154   10.4.62.31 -> 10.4.62.7    H.225.0 CS: setup
OpenLogicalChannel
1502.833732    10.4.62.7 -> 10.4.62.31   H.225.0 CS: callProceeding
1502.850909    10.4.62.7 -> 10.4.62.89   SIP/SDP Request: INVITE
sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>;transport=udp, with session
description
1502.851758    10.4.62.7 -> 10.4.62.31   H.225.0 CS: alerting
1502.861828   10.4.62.89 -> 10.4.62.7    SIP Status: 100 Trying
1502.875127   10.4.62.89 -> 10.4.62.7    SIP Status: 180 Ringing
1503.033258    10.4.62.7 -> 10.4.62.31   H.225.0 CS: progress
OpenLogicalChannel
1503.037908   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
terminalCapabilitySet
1503.038608   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
masterSlaveDetermination
1503.050154    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
terminalCapabilitySet
1503.051381    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
terminalCapabilitySetAck
1503.054297   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
terminalCapabilitySetAck
1503.054917    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
masterSlaveDeterminationAck
1503.057933   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
masterSlaveDeterminationAck
1505.485493   10.4.62.89 -> 10.4.62.7    SIP/SDP Status: 200 OK, with
session description
1505.486018    10.4.62.7 -> 10.4.62.89   SIP Request: ACK
sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>;transport=udp
1505.493611    10.4.62.7 -> 10.4.62.31   H.225.0 CS: connect
1509.565959   10.4.62.89 -> 10.4.62.7    SIP Request: BYE
sip:mod_sofia at 10.4.62.7:5060
1509.566722    10.4.62.7 -> 10.4.62.89   SIP Status: 200 OK
1509.577435    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: releaseComplete
endSessionCommand
1509.582066   10.4.62.31 -> 10.4.62.7    H.225.0 CS: releaseComplete



... i still need to check the CDRs as well but here we are :)
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From sicfslist at gmail.com  Fri Oct 23 07:58:10 2009
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Fri, 23 Oct 2009 09:58:10 -0500
Subject: [Freeswitch-users] dynamic gateway
In-Reply-To: <61ee6b990910222340r49ae8fa4oafa1bd4acfecd71c@mail.gmail.com>
References: <61ee6b990910222340r49ae8fa4oafa1bd4acfecd71c@mail.gmail.com>
Message-ID: <35b355e90910230758u53a76a2fo875ac2d6182e63e3@mail.gmail.com>

Hello:

Your issue is that this portion of the above string "sipinterface_1" has to
be a valid sip profile.  To do this:
  -- determine the appropriate sip profile you want to use (usually
external)
  -- send the traffic to whatever ip you want via this portion of your
string $1 at 192.168.111.101:5060

You can't dynamically change the /sipinterface_1/ parameter.

SDR
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From anthony.minessale at gmail.com  Fri Oct 23 08:02:07 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 23 Oct 2009 10:02:07 -0500
Subject: [Freeswitch-users] sched_api doesn't get launched
In-Reply-To: <59ad9ca10910230154i6d41ddfci8957b152c1884800@mail.gmail.com>
References: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
	<3F1747F9-C543-4C6F-BB01-4DD5FA5B7BC5@jerris.com>
	<59ad9ca10910161153h5d531533vd844e49b9ca6d896@mail.gmail.com>
	<87f2f3b90910161237h356b7aa2q6ec12c6ecf78dc34@mail.gmail.com>
	<59ad9ca10910210244q5410f2bdq3b93c25f95673d11@mail.gmail.com>
	<191c3a030910211309sb727b37g6324e1833f2a1e8c@mail.gmail.com>
	<59ad9ca10910230154i6d41ddfci8957b152c1884800@mail.gmail.com>
Message-ID: <191c3a030910230802m1cb847agfdaf6e436296e748@mail.gmail.com>

it's probably related to escaping the data.
I was sick of watching you suffer so i added api_on_answer variable to
trunk.


On Fri, Oct 23, 2009 at 3:54 AM, Henry Huang <red.rain.seven at gmail.com>wrote:

> Thanks to c6burns on IRC channel for the tip to use execute_on_answer in
> combination with eval, and of course everyone here that pointing me to the
> right direction.
> I was able to execute sched_api with eval, but not with the combination of
> execute_on_answer. The argument just don't get parsed as the eval argument.
> Here is the code:
>
> setVariable("execute_on_answer", "eval sched_api(+%ld none uuid_displace
> <some uuid> start
> /usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux)");
>
>
> Here is the log:
> 2009-10-23 16:17:53.633189 [DEBUG] switch_scheduler.c:214 Added task 2
> sched_api_function (none) to run at 1256285888
> EXECUTE sofia/internal/1688004 at 192.168.1.66 *eval(+OK *Added: 2
> )
>
> 2009-10-23 16:18:04.52701 [DEBUG] switch_channel.c:1927 sofia/internal/
> 1688004 at 192.168.1.66 *execute on answer: eval(sched_api(+37 none
> uuid_displace 84d74f41-8668-4138-943f-f076e94046ad start
> /usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux))*
> EXECUTE sofia/internal/1688004 at 192.168.1.66 eval(sched_api(+37 none
> uuid_displace 84d74f41-8668-4138-943f-f076e94046ad start
> /usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux))
>
>
> The first part of the log is when I just do execute("eval",
> "${sched_api(+%ld none uuid_displace <some uuid> start
> /usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux)}");
> And I simple get a OK, and then later on the sched_api gets executed.
>
> However, on the excerpt of the second log. which I use the combination with
> execute_on_answer like specified in the yellow hight code above. It doesn't
> simply return a OK, instead it shows the whole blob of code. and does
> nothing later.
>
>
>
> On Thu, Oct 22, 2009 at 4:09 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Yes you need an API object as described in other email.
>>
>> Which line of code from java caused that segfault
>> It looks like a simple NULL string issue that we may want to hunt down.
>>
>>
>> On Wed, Oct 21, 2009 at 4:44 AM, Henry Huang <red.rain.seven at gmail.com>wrote:
>>
>>> I can't seem to find the right thing to use in mod_java to execute api
>>> commands, only api_after_bridge
>>>
>>> 2009-10-21 17:42:46.593094 [NOTICE] mod_sofia.c:1509 Pre-Answer
>>> sofia/internal/1688003 at 192.168.1.66!
>>> #
>>> # A fatal error has been detected by the Java Runtime Environment:
>>> #
>>> #  SIGSEGV (0xb) at pc=0x004e4480, pid=1927, tid=16116624
>>> #
>>> # JRE version: 6.0_16-b01
>>> # Java VM: Java HotSpot(TM) Client VM (14.2-b01 mixed mode linux-x86 )
>>> # Problematic frame:
>>> # C  [libc.so.6+0x6f480]  strcpy+0x10
>>> #
>>> # An error report file with more information is saved as:
>>> # /usr/local/freeswitch/bin/hs_err_pid1927.log
>>> 2009-10-21 17:42:59.883729 [ERR] switch_core_session.c:1374 Invalid
>>> Application sched_api
>>> 2009-10-21 17:42:59.883729 [NOTICE] switch_core_session.c:1375 Hangup
>>> sofia/internal/1688003 at 192.168.1.66 [CS_EXECUTE]
>>> [DESTINATION_OUT_OF_ORDER]
>>> #
>>> # If you would like to submit a bug report, please visit:
>>> #   http://java.sun.com/webapps/bugreport/crash.jsp
>>> # The crash happened outside the Java Virtual Machine in native code.
>>> # See problematic frame for where to report the bug.
>>>
>>>
>>> On Sat, Oct 17, 2009 at 3:37 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>>
>>>>
>>>> On Fri, Oct 16, 2009 at 11:53 AM, Henry Huang <red.rain.seven at gmail.com
>>>> > wrote:
>>>>
>>>>> So how would you trigger it from a script dialplan? The only time it
>>>>> seemed to work is when I did setVariable("api_after_bridge", "sched_api blah
>>>>> blah blah");
>>>>> but then it gets executed after the channel's been teared down. I
>>>>> thought api_after_bridge means right after the call gets connected.
>>>>>
>>>>> I need something to execute an api command right before or right after
>>>>> the call gets bridged.
>>>>>
>>>>> api_after_bridge is a channel variable, so using setVariable works just
>>>> fine. If you need to sched_api is an API only. Check these out:
>>>> http://wiki.freeswitch.org/wiki/Mod_commands#Misc._Commands
>>>>
>>>> So you need an API object in order to use it. I don't know the syntax
>>>> for creating an api obj in Java but in Lua it goes like this:
>>>> api = freeswitch.API();
>>>> res = api:execute("sched_api","+300 none my_api my_api_args")
>>>>
>>>> Remember, if the method you are using isn't found in the dial plan tools
>>>> then it isn't a dial plan application. Make sure it's on the list:
>>>> http://wiki.freeswitch.org/wiki/Mod_dptools
>>>>
>>>> On the other hand, API commands are listed here:
>>>> http://wiki.freeswitch.org/wiki/Mod_commands
>>>>
>>>> dptools require a session object, api commands require an api object...
>>>>
>>>> -MC
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Henry Huang
>>> UniC Solution - Communication Unified
>>> VoIP & Open Source software Consultant
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Henry Huang
> UniC Solution - Communication Unified
> VoIP & Open Source software Consultant
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From saeedahmad1981 at gmail.com  Fri Oct 23 08:08:38 2009
From: saeedahmad1981 at gmail.com (Saeed Ahmad)
Date: Fri, 23 Oct 2009 17:08:38 +0200
Subject: [Freeswitch-users] dynamic gateway
In-Reply-To: <35b355e90910230758u53a76a2fo875ac2d6182e63e3@mail.gmail.com>
References: <61ee6b990910222340r49ae8fa4oafa1bd4acfecd71c@mail.gmail.com> 
	<35b355e90910230758u53a76a2fo875ac2d6182e63e3@mail.gmail.com>
Message-ID: <c96b95650910230808l7b532529r106431d3f858b1ae@mail.gmail.com>

what you get if you run

sofia status profile sipinterface_1
- SAT

On Fri, Oct 23, 2009 at 4:58 PM, Shelby Ramsey <sicfslist at gmail.com> wrote:

> sipinterface_1
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From bottleman at icf.org.ru  Fri Oct 23 08:17:47 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Fri, 23 Oct 2009 19:17:47 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910230757h4595f396o2dc1a1bb781d3a8a@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
	<Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
	<Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>
	<65d96fc80910230452s6b4a179aqa5bc41dc34efc403@mail.gmail.com>
	<Pine.LNX.4.64.0910231610220.21492@srv.icf.org.ru>
	<65d96fc80910230538w7105be6fve14956ab3b247bf6@mail.gmail.com>
	<Pine.LNX.4.64.0910231646210.29582@srv.icf.org.ru>
	<65d96fc80910230614m2b2504d0r92006b78d267a399@mail.gmail.com>
	<65d96fc80910230757h4595f396o2dc1a1bb781d3a8a@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910231903510.17804@srv.icf.org.ru>

On 2009-10-23 16:57 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

i have question to developers about one proce in fs

src/switch_ivr_originate.c

static switch_status_t originate_on_consume_media_transmit(switch_core_session_t *session)
{
    switch_channel_t *channel = switch_core_session_get_channel(session);

		if (!switch_channel_test_flag(channel, CF_PROXY_MODE)) {
		while (switch_channel_get_state(channel) == CS_CONSUME_MEDIA && !switch_channel_test_flag(chann
			if (!switch_channel_media_ready(channel)) {
			    switch_yield(10000);
			} else {
				switch_ivr_sleep(session, 10, SWITCH_FALSE, NULL);
			}
		}
	    }

	switch_channel_clear_state_handler(channel, &originate_state_handlers);

	return SWITCH_STATUS_FALSE;
}

what exacly it do? 

call scheme like this sip->fs->h323->gk->h323(on same fs)->fs(same too) and there i have no audio issues.
if bridge connect while it sleep i have audio, if it not sleep while bridge connect i have no audio.

TC>a solution to H323 endpoint => FS => SIP user no audio issue
TC>
TC>is to disable a wait for tx Audio ... for  case
TC>SWITCH_MESSAGE_INDICATE_ANSWER:{
TC>
TC>//m_txAudioOpened.Wait();
TC>
TC>
TC>                case SWITCH_MESSAGE_INDICATE_ANSWER:{
TC>
TC>                        switch_log_printf(SWITCH_CHANNEL_LOG,
TC>SWITCH_LOG_CONSOLE, "ANSWER: we got Answer event\n");
TC>
TC>                        if (switch_channel_test_flag(channel, CF_OUTBOUND))
TC>{
TC>
TC>                                switch_log_printf(SWITCH_CHANNEL_LOG,
TC>SWITCH_LOG_CONSOLE, "ANSWER: we got Answer event - CF_OUTBOUND
TC>\n");
TC>                                return SWITCH_STATUS_FALSE;
TC>                        }
TC>                        AnsweringCall(H323Connection::AnswerCallNow);
TC>
TC>                        switch_log_printf(SWITCH_CHANNEL_LOG,
TC>SWITCH_LOG_CONSOLE, "ANSWER: suppose the call is Answered Now\n");
TC>                        PTRACE(4, "mod_h323\tMedia started on connection "
TC><< *this);
TC>
TC>                        // test
TC>                        //switch_channel_mark_answered(m_fsChannel);
TC>
TC>                        m_rxAudioOpened.Wait();
TC>                        switch_log_printf(SWITCH_CHANNEL_LOG,
TC>SWITCH_LOG_CONSOLE, "ANSWER: wait for m_rxAudioOpened\n");
TC>                        //m_txAudioOpened.Wait();
TC>                        switch_log_printf(SWITCH_CHANNEL_LOG,
TC>SWITCH_LOG_CONSOLE, "ANSWER: we disable wait for m_txAudioOpened\n");
TC>
TC>                        switch_log_printf(SWITCH_CHANNEL_LOG,
TC>SWITCH_LOG_CONSOLE, "ANSWER: were waiting for rx/tx AudioOpen\n");
TC>
TC>                        if (!switch_channel_test_flag(m_fsChannel,
TC>CF_EARLY_MEDIA)) {
TC>
TC>                                switch_log_printf(SWITCH_CHANNEL_LOG,
TC>SWITCH_LOG_CONSOLE, "ANSWER: we have early media\n");
TC>
TC>                                PTRACE(4,
TC>"mod_h323\t-------------------->switch_channel_mark_answered(m_fsChannel) "
TC><< *this);
TC>                                switch_channel_mark_answered(m_fsChannel);
TC>                                switch_log_printf(SWITCH_CHANNEL_LOG,
TC>SWITCH_LOG_CONSOLE, "ANSWER: answered in early Media\n");
TC>                        }
TC>                        break;
TC>                }
TC>
TC>
TC>Now, I'm able to both originate and terminate cals with 2-way audio...
TC>the signaling looks correct...
TC>
TC>
TC>
TC>outgoing:
TC>
TC>1369.425046    10.4.62.7 -> 10.4.62.89   SIP/SDP Request: INVITE
TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>;transport=udp, with session
TC>description
TC>1369.426255    10.4.62.7 -> 10.4.62.31   H.225.0 CS: alerting
TC>1369.435950   10.4.62.89 -> 10.4.62.7    SIP Status: 100 Trying
TC>1369.449065   10.4.62.89 -> 10.4.62.7    SIP Status: 180 Ringing
TC>1369.605109    10.4.62.7 -> 10.4.62.31   H.225.0 CS: progress
TC>OpenLogicalChannel
TC>1369.609788   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>terminalCapabilitySet
TC>1369.610489   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>masterSlaveDetermination
TC>1369.619071    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
TC>terminalCapabilitySet
TC>1369.620349    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
TC>terminalCapabilitySetAck
TC>1369.623215   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>terminalCapabilitySetAck
TC>1369.625591    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
TC>masterSlaveDeterminationAck
TC>1369.628174   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>masterSlaveDeterminationAck
TC>1370.966958   10.4.62.89 -> 10.4.62.7    SIP/SDP Status: 200 OK, with
TC>session description
TC>1370.967431    10.4.62.7 -> 10.4.62.89   SIP Request: ACK
TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>;transport=udp
TC>1370.975172    10.4.62.7 -> 10.4.62.31   H.225.0 CS: connect
TC>1372.354383   10.4.62.89 -> 10.4.62.7    SIP Request: BYE
TC>sip:mod_sofia at 10.4.62.7:5060
TC>1372.355147    10.4.62.7 -> 10.4.62.89   SIP Status: 200 OK
TC>1372.392904    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: releaseComplete
TC>endSessionCommand
TC>1372.397302   10.4.62.31 -> 10.4.62.7    H.225.0 CS: releaseComplete
TC>
TC>
TC>incoming:
TC>
TC>
TC>1502.817154   10.4.62.31 -> 10.4.62.7    H.225.0 CS: setup
TC>OpenLogicalChannel
TC>1502.833732    10.4.62.7 -> 10.4.62.31   H.225.0 CS: callProceeding
TC>1502.850909    10.4.62.7 -> 10.4.62.89   SIP/SDP Request: INVITE
TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>;transport=udp, with session
TC>description
TC>1502.851758    10.4.62.7 -> 10.4.62.31   H.225.0 CS: alerting
TC>1502.861828   10.4.62.89 -> 10.4.62.7    SIP Status: 100 Trying
TC>1502.875127   10.4.62.89 -> 10.4.62.7    SIP Status: 180 Ringing
TC>1503.033258    10.4.62.7 -> 10.4.62.31   H.225.0 CS: progress
TC>OpenLogicalChannel
TC>1503.037908   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>terminalCapabilitySet
TC>1503.038608   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>masterSlaveDetermination
TC>1503.050154    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
TC>terminalCapabilitySet
TC>1503.051381    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
TC>terminalCapabilitySetAck
TC>1503.054297   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>terminalCapabilitySetAck
TC>1503.054917    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
TC>masterSlaveDeterminationAck
TC>1503.057933   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>masterSlaveDeterminationAck
TC>1505.485493   10.4.62.89 -> 10.4.62.7    SIP/SDP Status: 200 OK, with
TC>session description
TC>1505.486018    10.4.62.7 -> 10.4.62.89   SIP Request: ACK
TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>;transport=udp
TC>1505.493611    10.4.62.7 -> 10.4.62.31   H.225.0 CS: connect
TC>1509.565959   10.4.62.89 -> 10.4.62.7    SIP Request: BYE
TC>sip:mod_sofia at 10.4.62.7:5060
TC>1509.566722    10.4.62.7 -> 10.4.62.89   SIP Status: 200 OK
TC>1509.577435    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: releaseComplete
TC>endSessionCommand
TC>1509.582066   10.4.62.31 -> 10.4.62.7    H.225.0 CS: releaseComplete
TC>
TC>
TC>
TC>... i still need to check the CDRs as well but here we are :)
TC>

can you send a diff? in you call scheme call from h323 endpoint to fs is not have RAS?, 
because i don't have no audio issues in transit from h323 to sip, but my calls a going
thorough GK and fs is regitered on them, my call scheme is h323ep-RAS->GK-RAS->fs.

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From anthony.minessale at gmail.com  Fri Oct 23 08:37:03 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 23 Oct 2009 10:37:03 -0500
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910231903510.17804@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
	<Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>
	<65d96fc80910230452s6b4a179aqa5bc41dc34efc403@mail.gmail.com>
	<Pine.LNX.4.64.0910231610220.21492@srv.icf.org.ru>
	<65d96fc80910230538w7105be6fve14956ab3b247bf6@mail.gmail.com>
	<Pine.LNX.4.64.0910231646210.29582@srv.icf.org.ru>
	<65d96fc80910230614m2b2504d0r92006b78d267a399@mail.gmail.com>
	<65d96fc80910230757h4595f396o2dc1a1bb781d3a8a@mail.gmail.com>
	<Pine.LNX.4.64.0910231903510.17804@srv.icf.org.ru>
Message-ID: <191c3a030910230837x78ea6093i387617a1adfdf958@mail.gmail.com>

if you were on trunk that line of code would be gone.
you really can't do development on 1.0.4 its 6 months old and it will cause
you more trouble than you think when you eventually upgrade if you do not do
it soon.


2009/10/23 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-23 16:57 +0200, Tihomir Culjaga wrote
> freeswitch-users at lists.fre...:
>
> i have question to developers about one proce in fs
>
> src/switch_ivr_originate.c
>
> static switch_status_t
> originate_on_consume_media_transmit(switch_core_session_t *session)
> {
>    switch_channel_t *channel = switch_core_session_get_channel(session);
>
>                if (!switch_channel_test_flag(channel, CF_PROXY_MODE)) {
>                while (switch_channel_get_state(channel) == CS_CONSUME_MEDIA
> && !switch_channel_test_flag(chann
>                        if (!switch_channel_media_ready(channel)) {
>                            switch_yield(10000);
>                        } else {
>                                switch_ivr_sleep(session, 10, SWITCH_FALSE,
> NULL);
>                        }
>                }
>            }
>
>        switch_channel_clear_state_handler(channel,
> &originate_state_handlers);
>
>        return SWITCH_STATUS_FALSE;
> }
>
> what exacly it do?
>
> call scheme like this sip->fs->h323->gk->h323(on same fs)->fs(same too) and
> there i have no audio issues.
> if bridge connect while it sleep i have audio, if it not sleep while bridge
> connect i have no audio.
>
> TC>a solution to H323 endpoint => FS => SIP user no audio issue
> TC>
> TC>is to disable a wait for tx Audio ... for  case
> TC>SWITCH_MESSAGE_INDICATE_ANSWER:{
> TC>
> TC>//m_txAudioOpened.Wait();
> TC>
> TC>
> TC>                case SWITCH_MESSAGE_INDICATE_ANSWER:{
> TC>
> TC>                        switch_log_printf(SWITCH_CHANNEL_LOG,
> TC>SWITCH_LOG_CONSOLE, "ANSWER: we got Answer event\n");
> TC>
> TC>                        if (switch_channel_test_flag(channel,
> CF_OUTBOUND))
> TC>{
> TC>
> TC>                                switch_log_printf(SWITCH_CHANNEL_LOG,
> TC>SWITCH_LOG_CONSOLE, "ANSWER: we got Answer event - CF_OUTBOUND
> TC>\n");
> TC>                                return SWITCH_STATUS_FALSE;
> TC>                        }
> TC>                        AnsweringCall(H323Connection::AnswerCallNow);
> TC>
> TC>                        switch_log_printf(SWITCH_CHANNEL_LOG,
> TC>SWITCH_LOG_CONSOLE, "ANSWER: suppose the call is Answered Now\n");
> TC>                        PTRACE(4, "mod_h323\tMedia started on connection
> "
> TC><< *this);
> TC>
> TC>                        // test
> TC>                        //switch_channel_mark_answered(m_fsChannel);
> TC>
> TC>                        m_rxAudioOpened.Wait();
> TC>                        switch_log_printf(SWITCH_CHANNEL_LOG,
> TC>SWITCH_LOG_CONSOLE, "ANSWER: wait for m_rxAudioOpened\n");
> TC>                        //m_txAudioOpened.Wait();
> TC>                        switch_log_printf(SWITCH_CHANNEL_LOG,
> TC>SWITCH_LOG_CONSOLE, "ANSWER: we disable wait for m_txAudioOpened\n");
> TC>
> TC>                        switch_log_printf(SWITCH_CHANNEL_LOG,
> TC>SWITCH_LOG_CONSOLE, "ANSWER: were waiting for rx/tx AudioOpen\n");
> TC>
> TC>                        if (!switch_channel_test_flag(m_fsChannel,
> TC>CF_EARLY_MEDIA)) {
> TC>
> TC>                                switch_log_printf(SWITCH_CHANNEL_LOG,
> TC>SWITCH_LOG_CONSOLE, "ANSWER: we have early media\n");
> TC>
> TC>                                PTRACE(4,
> TC>"mod_h323\t-------------------->switch_channel_mark_answered(m_fsChannel)
> "
> TC><< *this);
> TC>
>  switch_channel_mark_answered(m_fsChannel);
> TC>                                switch_log_printf(SWITCH_CHANNEL_LOG,
> TC>SWITCH_LOG_CONSOLE, "ANSWER: answered in early Media\n");
> TC>                        }
> TC>                        break;
> TC>                }
> TC>
> TC>
> TC>Now, I'm able to both originate and terminate cals with 2-way audio...
> TC>the signaling looks correct...
> TC>
> TC>
> TC>
> TC>outgoing:
> TC>
> TC>1369.425046    10.4.62.7 -> 10.4.62.89   SIP/SDP Request: INVITE
> TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> <sip%3A1001 at 10.4.62.89<sip%253A1001 at 10.4.62.89>>;transport=udp,
> with session
> TC>description
> TC>1369.426255    10.4.62.7 -> 10.4.62.31   H.225.0 CS: alerting
> TC>1369.435950   10.4.62.89 -> 10.4.62.7    SIP Status: 100 Trying
> TC>1369.449065   10.4.62.89 -> 10.4.62.7    SIP Status: 180 Ringing
> TC>1369.605109    10.4.62.7 -> 10.4.62.31   H.225.0 CS: progress
> TC>OpenLogicalChannel
> TC>1369.609788   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
> TC>terminalCapabilitySet
> TC>1369.610489   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
> TC>masterSlaveDetermination
> TC>1369.619071    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
> TC>terminalCapabilitySet
> TC>1369.620349    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
> TC>terminalCapabilitySetAck
> TC>1369.623215   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
> TC>terminalCapabilitySetAck
> TC>1369.625591    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
> TC>masterSlaveDeterminationAck
> TC>1369.628174   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
> TC>masterSlaveDeterminationAck
> TC>1370.966958   10.4.62.89 -> 10.4.62.7    SIP/SDP Status: 200 OK, with
> TC>session description
> TC>1370.967431    10.4.62.7 -> 10.4.62.89   SIP Request: ACK
> TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> <sip%3A1001 at 10.4.62.89<sip%253A1001 at 10.4.62.89>
> >;transport=udp
> TC>1370.975172    10.4.62.7 -> 10.4.62.31   H.225.0 CS: connect
> TC>1372.354383   10.4.62.89 -> 10.4.62.7    SIP Request: BYE
> TC>sip:mod_sofia at 10.4.62.7:5060
> TC>1372.355147    10.4.62.7 -> 10.4.62.89   SIP Status: 200 OK
> TC>1372.392904    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS:
> releaseComplete
> TC>endSessionCommand
> TC>1372.397302   10.4.62.31 -> 10.4.62.7    H.225.0 CS: releaseComplete
> TC>
> TC>
> TC>incoming:
> TC>
> TC>
> TC>1502.817154   10.4.62.31 -> 10.4.62.7    H.225.0 CS: setup
> TC>OpenLogicalChannel
> TC>1502.833732    10.4.62.7 -> 10.4.62.31   H.225.0 CS: callProceeding
> TC>1502.850909    10.4.62.7 -> 10.4.62.89   SIP/SDP Request: INVITE
> TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> <sip%3A1001 at 10.4.62.89<sip%253A1001 at 10.4.62.89>>;transport=udp,
> with session
> TC>description
> TC>1502.851758    10.4.62.7 -> 10.4.62.31   H.225.0 CS: alerting
> TC>1502.861828   10.4.62.89 -> 10.4.62.7    SIP Status: 100 Trying
> TC>1502.875127   10.4.62.89 -> 10.4.62.7    SIP Status: 180 Ringing
> TC>1503.033258    10.4.62.7 -> 10.4.62.31   H.225.0 CS: progress
> TC>OpenLogicalChannel
> TC>1503.037908   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
> TC>terminalCapabilitySet
> TC>1503.038608   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
> TC>masterSlaveDetermination
> TC>1503.050154    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
> TC>terminalCapabilitySet
> TC>1503.051381    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
> TC>terminalCapabilitySetAck
> TC>1503.054297   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
> TC>terminalCapabilitySetAck
> TC>1503.054917    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
> TC>masterSlaveDeterminationAck
> TC>1503.057933   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
> TC>masterSlaveDeterminationAck
> TC>1505.485493   10.4.62.89 -> 10.4.62.7    SIP/SDP Status: 200 OK, with
> TC>session description
> TC>1505.486018    10.4.62.7 -> 10.4.62.89   SIP Request: ACK
> TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> <sip%3A1001 at 10.4.62.89<sip%253A1001 at 10.4.62.89>
> >;transport=udp
> TC>1505.493611    10.4.62.7 -> 10.4.62.31   H.225.0 CS: connect
> TC>1509.565959   10.4.62.89 -> 10.4.62.7    SIP Request: BYE
> TC>sip:mod_sofia at 10.4.62.7:5060
> TC>1509.566722    10.4.62.7 -> 10.4.62.89   SIP Status: 200 OK
> TC>1509.577435    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS:
> releaseComplete
> TC>endSessionCommand
> TC>1509.582066   10.4.62.31 -> 10.4.62.7    H.225.0 CS: releaseComplete
> TC>
> TC>
> TC>
> TC>... i still need to check the CDRs as well but here we are :)
> TC>
>
> can you send a diff? in you call scheme call from h323 endpoint to fs is
> not have RAS?,
> because i don't have no audio issues in transit from h323 to sip, but my
> calls a going
> thorough GK and fs is regitered on them, my call scheme is
> h323ep-RAS->GK-RAS->fs.
>
> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129-RIPE
>
> _______________________________________________
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>
>


-- 
Anthony Minessale II

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From bottleman at icf.org.ru  Fri Oct 23 08:46:58 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Fri, 23 Oct 2009 19:46:58 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <191c3a030910230837x78ea6093i387617a1adfdf958@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
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	<65d96fc80910230452s6b4a179aqa5bc41dc34efc403@mail.gmail.com>
	<Pine.LNX.4.64.0910231610220.21492@srv.icf.org.ru>
	<65d96fc80910230538w7105be6fve14956ab3b247bf6@mail.gmail.com>
	<Pine.LNX.4.64.0910231646210.29582@srv.icf.org.ru>
	<65d96fc80910230614m2b2504d0r92006b78d267a399@mail.gmail.com>
	<65d96fc80910230757h4595f396o2dc1a1bb781d3a8a@mail.gmail.com>
	<Pine.LNX.4.64.0910231903510.17804@srv.icf.org.ru>
	<191c3a030910230837x78ea6093i387617a1adfdf958@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910231943250.22146@srv.icf.org.ru>

On 2009-10-23 10:37 -0500, Anthony Minessale wrote freeswitch-users at lists.f...:

i have no way to install trunk at this time, i will go out of hospital about one week later, after 
this i will can try it on trunk.

AM>if you were on trunk that line of code would be gone.
AM>you really can't do development on 1.0.4 its 6 months old and it will cause
AM>you more trouble than you think when you eventually upgrade if you do not do
AM>it soon.
AM>
AM>
AM>2009/10/23 Georgiewskiy Yuriy <bottleman at icf.org.ru>
AM>
AM>> On 2009-10-23 16:57 +0200, Tihomir Culjaga wrote
AM>> freeswitch-users at lists.fre...:
AM>>
AM>> i have question to developers about one proce in fs
AM>>
AM>> src/switch_ivr_originate.c
AM>>
AM>> static switch_status_t
AM>> originate_on_consume_media_transmit(switch_core_session_t *session)
AM>> {
AM>>    switch_channel_t *channel = switch_core_session_get_channel(session);
AM>>
AM>>                if (!switch_channel_test_flag(channel, CF_PROXY_MODE)) {
AM>>                while (switch_channel_get_state(channel) == CS_CONSUME_MEDIA
AM>> && !switch_channel_test_flag(chann
AM>>                        if (!switch_channel_media_ready(channel)) {
AM>>                            switch_yield(10000);
AM>>                        } else {
AM>>                                switch_ivr_sleep(session, 10, SWITCH_FALSE,
AM>> NULL);
AM>>                        }
AM>>                }
AM>>            }
AM>>
AM>>        switch_channel_clear_state_handler(channel,
AM>> &originate_state_handlers);
AM>>
AM>>        return SWITCH_STATUS_FALSE;
AM>> }
AM>>
AM>> what exacly it do?
AM>>
AM>> call scheme like this sip->fs->h323->gk->h323(on same fs)->fs(same too) and
AM>> there i have no audio issues.
AM>> if bridge connect while it sleep i have audio, if it not sleep while bridge
AM>> connect i have no audio.
AM>>
AM>> TC>a solution to H323 endpoint => FS => SIP user no audio issue
AM>> TC>
AM>> TC>is to disable a wait for tx Audio ... for  case
AM>> TC>SWITCH_MESSAGE_INDICATE_ANSWER:{
AM>> TC>
AM>> TC>//m_txAudioOpened.Wait();
AM>> TC>
AM>> TC>
AM>> TC>                case SWITCH_MESSAGE_INDICATE_ANSWER:{
AM>> TC>
AM>> TC>                        switch_log_printf(SWITCH_CHANNEL_LOG,
AM>> TC>SWITCH_LOG_CONSOLE, "ANSWER: we got Answer event\n");
AM>> TC>
AM>> TC>                        if (switch_channel_test_flag(channel,
AM>> CF_OUTBOUND))
AM>> TC>{
AM>> TC>
AM>> TC>                                switch_log_printf(SWITCH_CHANNEL_LOG,
AM>> TC>SWITCH_LOG_CONSOLE, "ANSWER: we got Answer event - CF_OUTBOUND
AM>> TC>\n");
AM>> TC>                                return SWITCH_STATUS_FALSE;
AM>> TC>                        }
AM>> TC>                        AnsweringCall(H323Connection::AnswerCallNow);
AM>> TC>
AM>> TC>                        switch_log_printf(SWITCH_CHANNEL_LOG,
AM>> TC>SWITCH_LOG_CONSOLE, "ANSWER: suppose the call is Answered Now\n");
AM>> TC>                        PTRACE(4, "mod_h323\tMedia started on connection
AM>> "
AM>> TC><< *this);
AM>> TC>
AM>> TC>                        // test
AM>> TC>                        //switch_channel_mark_answered(m_fsChannel);
AM>> TC>
AM>> TC>                        m_rxAudioOpened.Wait();
AM>> TC>                        switch_log_printf(SWITCH_CHANNEL_LOG,
AM>> TC>SWITCH_LOG_CONSOLE, "ANSWER: wait for m_rxAudioOpened\n");
AM>> TC>                        //m_txAudioOpened.Wait();
AM>> TC>                        switch_log_printf(SWITCH_CHANNEL_LOG,
AM>> TC>SWITCH_LOG_CONSOLE, "ANSWER: we disable wait for m_txAudioOpened\n");
AM>> TC>
AM>> TC>                        switch_log_printf(SWITCH_CHANNEL_LOG,
AM>> TC>SWITCH_LOG_CONSOLE, "ANSWER: were waiting for rx/tx AudioOpen\n");
AM>> TC>
AM>> TC>                        if (!switch_channel_test_flag(m_fsChannel,
AM>> TC>CF_EARLY_MEDIA)) {
AM>> TC>
AM>> TC>                                switch_log_printf(SWITCH_CHANNEL_LOG,
AM>> TC>SWITCH_LOG_CONSOLE, "ANSWER: we have early media\n");
AM>> TC>
AM>> TC>                                PTRACE(4,
AM>> TC>"mod_h323\t-------------------->switch_channel_mark_answered(m_fsChannel)
AM>> "
AM>> TC><< *this);
AM>> TC>
AM>>  switch_channel_mark_answered(m_fsChannel);
AM>> TC>                                switch_log_printf(SWITCH_CHANNEL_LOG,
AM>> TC>SWITCH_LOG_CONSOLE, "ANSWER: answered in early Media\n");
AM>> TC>                        }
AM>> TC>                        break;
AM>> TC>                }
AM>> TC>
AM>> TC>
AM>> TC>Now, I'm able to both originate and terminate cals with 2-way audio...
AM>> TC>the signaling looks correct...
AM>> TC>
AM>> TC>
AM>> TC>
AM>> TC>outgoing:
AM>> TC>
AM>> TC>1369.425046    10.4.62.7 -> 10.4.62.89   SIP/SDP Request: INVITE
AM>> TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> <sip%3A1001 at 10.4.62.89<sip%253A1001 at 10.4.62.89>>;transport=udp,
AM>> with session
AM>> TC>description
AM>> TC>1369.426255    10.4.62.7 -> 10.4.62.31   H.225.0 CS: alerting
AM>> TC>1369.435950   10.4.62.89 -> 10.4.62.7    SIP Status: 100 Trying
AM>> TC>1369.449065   10.4.62.89 -> 10.4.62.7    SIP Status: 180 Ringing
AM>> TC>1369.605109    10.4.62.7 -> 10.4.62.31   H.225.0 CS: progress
AM>> TC>OpenLogicalChannel
AM>> TC>1369.609788   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
AM>> TC>terminalCapabilitySet
AM>> TC>1369.610489   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
AM>> TC>masterSlaveDetermination
AM>> TC>1369.619071    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
AM>> TC>terminalCapabilitySet
AM>> TC>1369.620349    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
AM>> TC>terminalCapabilitySetAck
AM>> TC>1369.623215   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
AM>> TC>terminalCapabilitySetAck
AM>> TC>1369.625591    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
AM>> TC>masterSlaveDeterminationAck
AM>> TC>1369.628174   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
AM>> TC>masterSlaveDeterminationAck
AM>> TC>1370.966958   10.4.62.89 -> 10.4.62.7    SIP/SDP Status: 200 OK, with
AM>> TC>session description
AM>> TC>1370.967431    10.4.62.7 -> 10.4.62.89   SIP Request: ACK
AM>> TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> <sip%3A1001 at 10.4.62.89<sip%253A1001 at 10.4.62.89>
AM>> >;transport=udp
AM>> TC>1370.975172    10.4.62.7 -> 10.4.62.31   H.225.0 CS: connect
AM>> TC>1372.354383   10.4.62.89 -> 10.4.62.7    SIP Request: BYE
AM>> TC>sip:mod_sofia at 10.4.62.7:5060
AM>> TC>1372.355147    10.4.62.7 -> 10.4.62.89   SIP Status: 200 OK
AM>> TC>1372.392904    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS:
AM>> releaseComplete
AM>> TC>endSessionCommand
AM>> TC>1372.397302   10.4.62.31 -> 10.4.62.7    H.225.0 CS: releaseComplete
AM>> TC>
AM>> TC>
AM>> TC>incoming:
AM>> TC>
AM>> TC>
AM>> TC>1502.817154   10.4.62.31 -> 10.4.62.7    H.225.0 CS: setup
AM>> TC>OpenLogicalChannel
AM>> TC>1502.833732    10.4.62.7 -> 10.4.62.31   H.225.0 CS: callProceeding
AM>> TC>1502.850909    10.4.62.7 -> 10.4.62.89   SIP/SDP Request: INVITE
AM>> TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> <sip%3A1001 at 10.4.62.89<sip%253A1001 at 10.4.62.89>>;transport=udp,
AM>> with session
AM>> TC>description
AM>> TC>1502.851758    10.4.62.7 -> 10.4.62.31   H.225.0 CS: alerting
AM>> TC>1502.861828   10.4.62.89 -> 10.4.62.7    SIP Status: 100 Trying
AM>> TC>1502.875127   10.4.62.89 -> 10.4.62.7    SIP Status: 180 Ringing
AM>> TC>1503.033258    10.4.62.7 -> 10.4.62.31   H.225.0 CS: progress
AM>> TC>OpenLogicalChannel
AM>> TC>1503.037908   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
AM>> TC>terminalCapabilitySet
AM>> TC>1503.038608   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
AM>> TC>masterSlaveDetermination
AM>> TC>1503.050154    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
AM>> TC>terminalCapabilitySet
AM>> TC>1503.051381    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
AM>> TC>terminalCapabilitySetAck
AM>> TC>1503.054297   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
AM>> TC>terminalCapabilitySetAck
AM>> TC>1503.054917    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
AM>> TC>masterSlaveDeterminationAck
AM>> TC>1503.057933   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
AM>> TC>masterSlaveDeterminationAck
AM>> TC>1505.485493   10.4.62.89 -> 10.4.62.7    SIP/SDP Status: 200 OK, with
AM>> TC>session description
AM>> TC>1505.486018    10.4.62.7 -> 10.4.62.89   SIP Request: ACK
AM>> TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> <sip%3A1001 at 10.4.62.89<sip%253A1001 at 10.4.62.89>
AM>> >;transport=udp
AM>> TC>1505.493611    10.4.62.7 -> 10.4.62.31   H.225.0 CS: connect
AM>> TC>1509.565959   10.4.62.89 -> 10.4.62.7    SIP Request: BYE
AM>> TC>sip:mod_sofia at 10.4.62.7:5060
AM>> TC>1509.566722    10.4.62.7 -> 10.4.62.89   SIP Status: 200 OK
AM>> TC>1509.577435    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS:
AM>> releaseComplete
AM>> TC>endSessionCommand
AM>> TC>1509.582066   10.4.62.31 -> 10.4.62.7    H.225.0 CS: releaseComplete
AM>> TC>
AM>> TC>
AM>> TC>
AM>> TC>... i still need to check the CDRs as well but here we are :)
AM>> TC>
AM>>
AM>> can you send a diff? in you call scheme call from h323 endpoint to fs is
AM>> not have RAS?,
AM>> because i don't have no audio issues in transit from h323 to sip, but my
AM>> calls a going
AM>> thorough GK and fs is regitered on them, my call scheme is
AM>> h323ep-RAS->GK-RAS->fs.
AM>>
AM>> C ?????????                       With Best Regards
AM>> ???????????? ????.                Georgiewskiy Yuriy
AM>> +7 4872 711666                    +7 4872 711666
AM>> ???? +7 4872 711143               fax +7 4872 711143
AM>> ???????? ??? "?? ?? ??????"       IT Service Ltd
AM>> http://nkoort.ru                  http://nkoort.ru
AM>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
AM>> YG129-RIPE                        YG129-RIPE
AM>>
AM>> _______________________________________________
AM>> FreeSWITCH-users mailing list
AM>> FreeSWITCH-users at lists.freeswitch.org
AM>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
AM>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
AM>> http://www.freeswitch.org
AM>>
AM>>
AM>
AM>
AM>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From anthony.minessale at gmail.com  Fri Oct 23 08:51:57 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 23 Oct 2009 10:51:57 -0500
Subject: [Freeswitch-users] Qustion about INFO messages after
	Connect/Answer
In-Reply-To: <4AE1951A.1030906@ewetel.de>
References: <4AD859A4.5000101@ewetel.de> <4ADC8B6E.7070404@ewetel.de>
	<191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>
	<87f2f3b90910191435s7528e256r85df5f12fc634f24@mail.gmail.com>
	<4ADF483B.4050807@ewetel.de>
	<87f2f3b90910211048s208daf1eyfae501091ab75530@mail.gmail.com>
	<4AE07DF4.5040803@ewetel.de>
	<085B77EE-DA3F-4ADD-86DC-E59D1B6553EA@freeswitch.org>
	<191c3a030910221008n60341314hc7a7e6a55a50ef15@mail.gmail.com>
	<4AE1951A.1030906@ewetel.de>
Message-ID: <191c3a030910230851i6e0b081ajaf6c294a7f9cd3a8@mail.gmail.com>

should be even better in 15210


On Fri, Oct 23, 2009 at 6:35 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi Anthony,
>
> thanks! The "unknown" thing is gone as long as you don't use
> "originate_callee_id_name"
>
>
> The "originate_callee_id_name" chvar causes caller's AND callee's
> display ("caller name" line)to be updated with the content of that chvar
> as soon as you set it.
>
> Let's say you use "hubu" as originate_callee_id_name. As soon as callee
> has picked up the phone it's display ("caller name" line) is updated
> with "hubu" and right after that with caller's name. The caller's
> display ("callee name" line) is updated to "hubu" (as expected).
>
> So it's still not nice to see a name switch in callee's display for
> caller's name after picking up.
>
> In the following you see the SIP flows from
>
> a) caller to FS
> b) FS to callee
>
>
> The more interesting one is b) I guess
>
>
>
>
> #######################################################
> This is the sip flow from caller to FS, which is OK:
>
>
>
> INVITE sip:1000 at 85.16.246.12:5061;user=phone SIP/2.0
> 23/Oct/2009-13:26:44.730
>
> INVITE sip:1000 at 85.16.246.12:5061;user=phone SIP/2.0
> Via: SIP/2.0/UDP 85.16.245.206:1024;branch=z9hG4bK-zdcpisfjx11n;rport
> From: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
> To: <sip:1000 at 85.16.246.12:5061;user=phone>
> Call-ID: 3c32462e758f-y19kuulwdknf
> CSeq: 1 INVITE
> Max-Forwards: 70
> Contact: <sip:1001 at 85.16.245.206:1024;line=eg3wp69a>;reg-id=1
> X-Serialnumber: 0004134002CB
> P-Key-Flags: resolution="31x13", keys="4"
> User-Agent: snom820/8.2.16
> Accept: application/sdp
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
> PRACK, MESSAGE, INFO, UPDATE
> Allow-Events: talk, hold, refer, call-info
> Supported: timer, 100rel, replaces, from-change
> Session-Expires: 3600;refresher=uas
> Min-SE: 90
> Content-Type: application/sdp
> Content-Length: 734
>
> v=0
> o=root 1420196469 1420196469 IN IP4 85.16.245.206
> s=call
> c=IN IP4 85.16.245.206
> t=0 0
> m=audio 51286 RTP/SAVP 0 8 9 99 3 18 4 101
> a=crypto:1 AES_CM_128_HMAC_SHA1_32
> inline:CjdPN1m2iLKAyXrQ2pWkQfCMtiISlF94ItTdsDis
> a=rtpmap:0 pcmu/8000
> a=rtpmap:8 pcma/8000
> a=rtpmap:9 g722/8000
> a=rtpmap:99 g726-32/8000
> a=rtpmap:3 gsm/8000
> a=rtpmap:18 g729/8000
> a=rtpmap:4 g723/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
> m=audio 51286 RTP/AVP 0 8 9 99 3 18 4 101
> a=rtpmap:0 pcmu/8000
> a=rtpmap:8 pcma/8000
> a=rtpmap:9 g722/8000
> a=rtpmap:99 g726-32/8000
> a=rtpmap:3 gsm/8000
> a=rtpmap:18 g729/8000
> a=rtpmap:4 g723/8000
> a=rtpmap:101 telephone-ev
>
> SIP/2.0 100 Trying
> 23/Oct/2009-13:26:44.731
>
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 85.16.245.206:1024;branch=z9hG4bK-zdcpisfjx11n;rport=1024
> From: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
> To: <sip:1000 at 85.16.246.12:5061;user=phone>
> Call-ID: 3c32462e758f-y19kuulwdknf
> CSeq: 1 INVITE
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
> Content-Length: 0
>
>
> SIP/2.0 180 Ringing
> 23/Oct/2009-13:26:45.3
>
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 85.16.245.206:1024;branch=z9hG4bK-zdcpisfjx11n;rport=1024
> From: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
> To: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
> Call-ID: 3c32462e758f-y19kuulwdknf
> CSeq: 1 INVITE
> Contact: <sip:1000 at 85.16.246.12:5061;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
> Accept: application/sdp
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
> Content-Length: 0
> X-FS-Display-Name: hubu
> X-FS-Display-Number:
> X-FS-Support: update_display
>
>
> SIP/2.0 200 OK
> 23/Oct/2009-13:26:47.367
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 85.16.245.206:1024;branch=z9hG4bK-zdcpisfjx11n;rport=1024
> From: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
> To: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
> Call-ID: 3c32462e758f-y19kuulwdknf
> CSeq: 1 INVITE
> Contact: <sip:1000 at 85.16.246.12:5061;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
> Session-Expires: 3600;refresher=uas
> Min-SE: 120
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 354
> X-FS-Display-Name: hubu
> X-FS-Display-Number:
> X-FS-Support: update_display
>
> v=0
> o=FreeSWITCH 1256285701 1256285702 IN IP4 85.16.246.12
> s=FreeSWITCH
> c=IN IP4 85.16.246.12
> t=0 0
> m=audio 11506 RTP/SAVP 8 101
> a=rtpmap:8 pcma/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=crypto:1 AES_CM_128_HMAC_SHA1_32
> inline:/BFm/LUmDBuao0wqg6MS/l4acfzOHoHRFQr3N0R9
> m=audio 0 RTP/AVP 19
>
> ACK sip:1000 at 85.16.246.12:5061;transport=udp SIP/2.0
> 23/Oct/2009-13:26:47.392
>
> ACK sip:1000 at 85.16.246.12:5061;transport=udp SIP/2.0
> Via: SIP/2.0/UDP 85.16.245.206:1024;branch=z9hG4bK-d474to81d7gh;rport
> From: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
> To: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
> Call-ID: 3c32462e758f-y19kuulwdknf
> CSeq: 1 ACK
> Max-Forwards: 70
> Contact: <sip:1001 at 85.16.245.206:1024;line=eg3wp69a>;reg-id=1
> Content-Length: 0
>
>
> INFO sip:1001 at 85.16.245.206:1024;line=eg3wp69a SIP/2.0
> 23/Oct/2009-13:26:47.416
>
> INFO sip:1001 at 85.16.245.206:1024;line=eg3wp69a SIP/2.0
> Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKB244eKZ08Uj3H
> Max-Forwards: 70
> From: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
> To: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
> Call-ID: 3c32462e758f-y19kuulwdknf
> CSeq: 122030267 INFO
> Contact: <sip:1000 at 85.16.246.12:5061;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Content-Type: message/sipfrag
> Content-Length: 24
>
> From:
> To: "hubu" 1000
>
> SIP/2.0 200 Ok
> 23/Oct/2009-13:26:47.619
>
> SIP/2.0 200 Ok
> Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKB244eKZ08Uj3H
> From: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
> To: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
> Call-ID: 3c32462e758f-y19kuulwdknf
> CSeq: 122030267 INFO
> Contact: <sip:1001 at 85.16.245.206:1024;line=eg3wp69a>;reg-id=1
> Content-Length: 0
>
>
> BYE sip:1001 at 85.16.245.206:1024;line=eg3wp69a SIP/2.0
> 23/Oct/2009-13:26:49.18
>
> BYE sip:1001 at 85.16.245.206:1024;line=eg3wp69a SIP/2.0
> Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKDmQpj9072DZ8r
> Max-Forwards: 70
> From: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
> To: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
> Call-ID: 3c32462e758f-y19kuulwdknf
> CSeq: 122030268 BYE
> Contact: <sip:1000 at 85.16.246.12:5061;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Reason: Q.850;cause=16;text="NORMAL_CLEARING"
> Content-Length: 0
>
>
> SIP/2.0 200 OK
> 23/Oct/2009-13:26:49.52
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKDmQpj9072DZ8r
> From: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
> To: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
> Call-ID: 3c32462e758f-y19kuulwdknf
> CSeq: 122030268 BYE
> Contact: <sip:1001 at 85.16.245.206:1024;line=eg3wp69a>;reg-id=1
> User-Agent: snom820/8.2.16
> RTP-RxStat:
> Total_Rx_Pkts=65,Rx_Pkts=65,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0
> RTP-TxStat: Total_Tx_Pkts=0,Tx_Pkts=94,Remote_Tx_Pkts=0
> Content-Length: 0
>
>
>
>
>
> ####################################################################
>
> This is the SIP flow from FS to callee.
>
> The first INFO sent from FS to callee is the Problem. It contains "hubu"
> as "To:" header. If there is s need for this message I guess the "From"
> header should be changed, because from the callee-phone's point of view
> the "To:" header is the caller and the "From" header is the callee. In
> my opinion this INFO message shouldn't be sent at all, but I don't know
> what you intend it to do.
>
>
> INVITE sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
> 23/Oct/2009-13:26:44.978
>
> INVITE sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
> Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bK86rt91cpH1FBF
> Max-Forwards: 68
> From: "1001 an PBX1" <sip:1001 at 85.16.246.12 <sip%3A1001 at 85.16.246.12>
> >;tag=rp27e1KSe969D
> To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>
> Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
> CSeq: 122030266 INVITE
> Contact: <sip:mod_sofia at 85.16.246.12:5061>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 325
> X-Serialnumber: 0004134002CB
> P-Key-Flags: resolution="31x13", keys="4"
> X-FS-Support: update_display
> Remote-Party-ID: "1001 an PBX1"
> <sip:1001 at 85.16.246.12 <sip%3A1001 at 85.16.246.12>
> >;party=calling;screen=yes;privacy=off
>
> v=0
> o=FreeSWITCH 1256285400 1256285401 IN IP4 85.16.246.12
> s=FreeSWITCH
> c=IN IP4 85.16.246.12
> t=0 0
> m=audio 11804 RTP/AVP 95 103 9 8 101 13
> a=rtpmap:95 CELT/48000
> a=rtpmap:103 SPEEX/32000
> a=rtpmap:9 G722/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:10
>
> SIP/2.0 180 Ringing
> 23/Oct/2009-13:26:45.2
>
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bK86rt91cpH1FBF
> From: "1001 an PBX1" <sip:1001 at 85.16.246.12 <sip%3A1001 at 85.16.246.12>
> >;tag=rp27e1KSe969D
> To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
> Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
> CSeq: 122030266 INVITE
> Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
> PRACK, MESSAGE, INFO, UPDATE
> Allow-Events: talk, hold, refer, call-info
> Content-Length: 0
>
>
> SIP/2.0 180 Ringing
> 23/Oct/2009-13:26:45.504
>
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bK86rt91cpH1FBF
> From: "1001 an PBX1" <sip:1001 at 85.16.246.12 <sip%3A1001 at 85.16.246.12>
> >;tag=rp27e1KSe969D
> To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
> Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
> CSeq: 122030266 INVITE
> Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
> PRACK, MESSAGE, INFO, UPDATE
> Allow-Events: talk, hold, refer, call-info
> Content-Length: 0
>
>
> SIP/2.0 180 Ringing
> 23/Oct/2009-13:26:46.504
>
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bK86rt91cpH1FBF
> From: "1001 an PBX1" <sip:1001 at 85.16.246.12 <sip%3A1001 at 85.16.246.12>
> >;tag=rp27e1KSe969D
> To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
> Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
> CSeq: 122030266 INVITE
> Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
> PRACK, MESSAGE, INFO, UPDATE
> Allow-Events: talk, hold, refer, call-info
> Content-Length: 0
>
>
> SIP/2.0 200 Ok
> 23/Oct/2009-13:26:47.361
>
> SIP/2.0 200 Ok
> Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bK86rt91cpH1FBF
> From: "1001 an PBX1" <sip:1001 at 85.16.246.12 <sip%3A1001 at 85.16.246.12>
> >;tag=rp27e1KSe969D
> To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
> Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
> CSeq: 122030266 INVITE
> Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
> User-Agent: snom820/LAB_HD_HEAD_11_09_09_11_20
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
> PRACK, MESSAGE, INFO, UPDATE
> Allow-Events: talk, hold, refer, call-info
> Supported: timer, 100rel, replaces, from-change
> Content-Type: application/sdp
> Content-Length: 244
>
> v=0
> o=root 353690302 353690303 IN IP4 85.16.245.228
> s=call
> c=IN IP4 85.16.245.228
> t=0 0
> m=audio 60878 RTP/AVP 9 8 101
> a=rtpmap:9 g722/8000
> a=rtpmap:8 pcma/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:10
> a=sendrecv
>
> ACK sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
> 23/Oct/2009-13:26:47.362
>
> ACK sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
> Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bK9FjKBXXSea6Xa
> Max-Forwards: 70
> From: "1001 an PBX1" <sip:1001 at 85.16.246.12 <sip%3A1001 at 85.16.246.12>
> >;tag=rp27e1KSe969D
> To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
> Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
> CSeq: 122030266 ACK
> Contact: <sip:mod_sofia at 85.16.246.12:5061>
> Content-Length: 0
>
>
> INFO sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
> 23/Oct/2009-13:26:47.367
>
> INFO sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
> Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKaSBcDreXBKvgp
> Max-Forwards: 70
> From: "1001 an PBX1" <sip:1001 at 85.16.246.12 <sip%3A1001 at 85.16.246.12>
> >;tag=rp27e1KSe969D
> To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
> Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
> CSeq: 122030267 INFO
> Contact: <sip:mod_sofia at 85.16.246.12:5061>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Content-Type: message/sipfrag
> Content-Length: 24
>
> From:
> To: "hubu" 1000
>
> SIP/2.0 200 Ok
> 23/Oct/2009-13:26:47.766
>
> SIP/2.0 200 Ok
> Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKaSBcDreXBKvgp
> From: "1001 an PBX1" <sip:1001 at 85.16.246.12 <sip%3A1001 at 85.16.246.12>
> >;tag=rp27e1KSe969D
> To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
> Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
> CSeq: 122030267 INFO
> Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
> Content-Length: 0
>
>
> INFO sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
> 23/Oct/2009-13:26:47.766
>
> INFO sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
> Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKcByXgeg4548ND
> Max-Forwards: 70
> From: "1001 an PBX1" <sip:1001 at 85.16.246.12 <sip%3A1001 at 85.16.246.12>
> >;tag=rp27e1KSe969D
> To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
> Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
> CSeq: 122030268 INFO
> Contact: <sip:mod_sofia at 85.16.246.12:5061>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Content-Type: message/sipfrag
> Content-Length: 32
>
> From:
> To: "1001 an PBX1" 1001
>
> SIP/2.0 200 Ok
> 23/Oct/2009-13:26:47.860
>
> SIP/2.0 200 Ok
> Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKcByXgeg4548ND
> From: "1001 an PBX1" <sip:1001 at 85.16.246.12 <sip%3A1001 at 85.16.246.12>
> >;tag=rp27e1KSe969D
> To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
> Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
> CSeq: 122030268 INFO
> Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
> Content-Length: 0
>
>
> BYE sip:mod_sofia at 85.16.246.12:5061 SIP/2.0
> 23/Oct/2009-13:26:49.3
>
> BYE sip:mod_sofia at 85.16.246.12:5061 SIP/2.0
> Via: SIP/2.0/UDP 85.16.245.228:1024;branch=z9hG4bK-susgzpvc38s7;rport
> From: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
> To: "1001 an PBX1" <sip:1001 at 85.16.246.12 <sip%3A1001 at 85.16.246.12>
> >;tag=rp27e1KSe969D
> Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
> CSeq: 1 BYE
> Max-Forwards: 70
> Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
> User-Agent: snom820/LAB_HD_HEAD_11_09_09_11_20
> RTP-RxStat: Total_Rx_Pkts=0,Rx_Pkts=69,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0
> RTP-TxStat: Total_Tx_Pkts=70,Tx_Pkts=65,Remote_Tx_Pkts=0
> Content-Length: 0
>
>
> SIP/2.0 200 OK
> 23/Oct/2009-13:26:49.4
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 85.16.245.228:1024;branch=z9hG4bK-susgzpvc38s7;rport=1024
> From: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
> To: "1001 an PBX1" <sip:1001 at 85.16.246.12 <sip%3A1001 at 85.16.246.12>
> >;tag=rp27e1KSe969D
> Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
> CSeq: 1 BYE
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Content-Length: 0
>
>
>
>
>
>
>
>
>
>
> On 22.10.2009 19:08, Anthony Minessale wrote:
> > I'm sure that problem is gone in svn trunk.
> >
> >
> > On Thu, Oct 22, 2009 at 11:25 AM, Brian West <brian at freeswitch.org
> > <mailto:brian at freeswitch.org>> wrote:
> >
> >     I can't get what exactly you re talking about. Can you clarify ? Also
> >     please include the packets of interest only not the full trace if its
> >     not relevant to the bug.
> >
> >     /b
> >
> >     On Oct 22, 2009, at 10:44 AM, Helmut Kuper wrote:
> >
> >     > -----BEGIN PGP SIGNED MESSAGE-----
> >     > Hash: SHA1
> >     >
> >     > Hi Mike,
> >     >
> >     > here it is:
> >     >
> >     >
> >     > Dialplan:
> >     >
> >     >
> >     >    <extension name="Local_Extension">
> >     >      <condition field="destination_number"
> expression="^(10[01][0-9])
> >     > $">
> >     >        <action application="set" data="dialed_extension=$1"/>
> >     >        <action application="export" data="dialed_extension=$1"/>
> >     >        <action application="set" data="transfer_ringback=$$
> >     > {hold_music}"/>
> >     >        <action application="set" data="hangup_after_bridge=true"/>
> >     >        <action application="bridge"
> >     > data="user/${dialed_extension}@${domain_name}"/>
> >     >      </condition>
> >     >    </extension>
> >     >
> >
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> > iax:guest at conference.freeswitch.org/888
> > <http://iax:guest at conference.freeswitch.org/888>
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> > pstn:213-799-1400
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> - --
>
> Mit freundlichen Gr??en
> Helmut Kuper
> Gesch?ftseinheit FD - L?sungen f?r Finanzdienstleister
> Telefax: (0441) 8000-2799
> mailto:helmut.kuper at ewetel.de
> ___________________________________
> EWE TEL GmbH
> Cloppenburger Stra?e 310
> 26133 Oldenburg
> EWE TEL GmbH
>
> Handelsregister Amtsgericht Oldenburg HRB 3723
> Vorsitzender des Aufsichtsrates: Heiko Harms
> Gesch?ftsf?hrung: Hans-Joachim Iken (Vorsitzender),
> Ulf Heggenberger, Dr. Norbert Schulz, Dirk Thole
> Homepage: http://www.ewetel.de
> ___________________________________
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
>
> iD8DBQFK4ZUZ4tZeNddg3dwRAr+lAJ9b3geDcwT0bkz++YG9Sjq62AtHBQCfWALu
> jkUQIoNN6CYJFp10ebA5MIU=
> =cHej
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Fri Oct 23 09:00:48 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 23 Oct 2009 09:00:48 -0700
Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Starting
	Shortly, Please Start 	Calling In
Message-ID: <87f2f3b90910230900l2201894bjc7a57c8093617b8@mail.gmail.com>

FYI,

The weekly call will begin soon. The agenda is here:
http://bit.ly/O0oGB

Note that the wiki is giving me problems with editing. I will let you know
when Raymond and I get it squared away.
-MC
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From tculjaga at gmail.com  Fri Oct 23 09:26:40 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 23 Oct 2009 18:26:40 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910231943250.22146@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910230452s6b4a179aqa5bc41dc34efc403@mail.gmail.com>
	<Pine.LNX.4.64.0910231610220.21492@srv.icf.org.ru>
	<65d96fc80910230538w7105be6fve14956ab3b247bf6@mail.gmail.com>
	<Pine.LNX.4.64.0910231646210.29582@srv.icf.org.ru>
	<65d96fc80910230614m2b2504d0r92006b78d267a399@mail.gmail.com>
	<65d96fc80910230757h4595f396o2dc1a1bb781d3a8a@mail.gmail.com>
	<Pine.LNX.4.64.0910231903510.17804@srv.icf.org.ru>
	<191c3a030910230837x78ea6093i387617a1adfdf958@mail.gmail.com>
	<Pine.LNX.4.64.0910231943250.22146@srv.icf.org.ru>
Message-ID: <65d96fc80910230926y6c3e7ee3i9bb2182da1ba8757@mail.gmail.com>

2009/10/23 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-23 10:37 -0500, Anthony Minessale wrote
> freeswitch-users at lists.f...:
>
> i have no way to install trunk at this time, i will go out of hospital
> about one week later, after
> this i will can try it on trunk.
>
> AM>if you were on trunk that line of code would be gone.
> AM>you really can't do development on 1.0.4 its 6 months old and it will
> cause
> AM>you more trouble than you think when you eventually upgrade if you do
> not do
>

uh... you are still in there ... damn sorry to hear that.
I know what the feeling is... i was in the same position a couple of months
ago... anyhow at least i had plenty of time to do my private research.. hope
it is the case with you as well.

so, if you have that time i can offer you my development server with trunk
installed and all the hands on.


let me know...

BTW: it really doesn't have sense to develop on 1.0.4 ... the proof of
concept was done. I'm able to place calls in both directions so, lets move
to trunk now.

T.
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From ujjval at simplesignal.com  Fri Oct 23 09:54:48 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Fri, 23 Oct 2009 09:54:48 -0700
Subject: [Freeswitch-users] Setting up Conference with Moderator
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local>
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local>

Any ideas on this one. Look slike only way rite now is to have a different Dest Phone number for a moderator.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ujjval Karihaloo
Sent: Thursday, October 22, 2009 9:02 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Setting up Conference with Moderator

Hi

I have the Basic Conferencing working. Here is my Dial Plan.

 I want to be able to setup a Moderator PIN different from other participants, when I add the moderator flag it logs me in directly w/o asking for a PIN..
action application="conference" data="conference.conf at wideband+flags{moderator}+159753<mailto:conference.conf at wideband+flags%7bmoderator%7d+159753>"/>

DialPlan is below for the normal user, and it asks for the PIN with below settings.

       Ujj Inbound from SS - start
    -->
    <extension name="simplesignal">   <!-- your provider or any name you'd like to call it -->
     <condition field="destination_number" expression="2142349127">  <!-- your DID for this gateway-->
        <action application="conference" data="conference.conf at wideband+159753<mailto:conference.conf at wideband+159753>"/>
     </condition>
</extension>

    <!--
       Ujj Inbound from SS - end




And I am using the existing conference.conf.xml file in the auto_loads directory.


Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690

SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO  80112
[cid:image001.jpg at 01CA53CF.3D2B8B40]<http://www.simplesignal.com/>

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From saeedahmad1981 at gmail.com  Fri Oct 23 10:16:46 2009
From: saeedahmad1981 at gmail.com (Saeed Ahmad)
Date: Fri, 23 Oct 2009 19:16:46 +0200
Subject: [Freeswitch-users] CPU Usage Question
In-Reply-To: <4add4a7a.1408c00a.4283.ffffc11f@mx.google.com>
References: <4add4a7a.1408c00a.4283.ffffc11f@mx.google.com>
Message-ID: <c96b95650910231016me63190fr4641ed7737637a85@mail.gmail.com>

I think we can't create two profiles per single IP. So what is the
workaround to solve this issue?

SAT

On Tue, Oct 20, 2009 at 7:28 AM, brian at freeswitch.org
<brian at freeswitch.org>wrote:

> It's because Sofia used one thread per profile so depending on load you
> will see one CPU used more.
>
> /b
>
> Sent from my HTC on the Now Network from Sprint!
>
> ----- Reply message -----
> From: "DJB" <djbinter at yahoo.com>
> Date: Mon, Oct 19, 2009 9:13 PM
> Subject: [Freeswitch-users] CPU Usage Question
> To: "FREESWITCH-USERS MAILING LIST" <freeswitch-users at lists.freeswitch.org
> >
>
>
> We have about 1500 concurrent calls on the server.  And, the CPU usage is
> shown below:
>
> 02:06:46 AM  CPU   %user   %nice    %sys %iowait    %irq   %soft  %steal
> %idle    intr/s
> 02:06:46 AM  all    0.06    0.00    0.03    0.44    0.00    0.00    0.00
> 99.47   1028.25
> 02:06:46 AM    0    0.08    0.03    0.03    0.01    0.00    0.01    0.00
> 99.84   1007.07
> 02:06:46 AM    1    0.06    0.00    0.02    0.01    0.00    0.00    0.00
> 99.91      0.00
> 02:06:46 AM    2    0.07    0.00    0.08    3.49    0.02    0.00    0.00
> 96.34     21.18
> 02:06:46 AM    3    0.05    0.00    0.01    0.01    0.00    0.00    0.00
> 99.94      0.00
> 02:06:46 AM    4    0.04    0.00    0.02    0.00    0.00    0.00    0.00
> 99.93      0.00
> 02:06:46 AM    5    0.05    0.00    0.02    0.00    0.00    0.00    0.00
> 99.92      0.00
> 02:06:46 AM    6    0.05    0.00    0.02    0.02    0.00    0.00    0.00
> 99.91      0.00
> 02:06:46 AM    7    0.05    0.00    0.02    0.01    0.00    0.00    0.00
> 99.93      0.00
>
>
> 1) I am wondering whether it is normal that every time when i checked, only
> CPU 2 always have the highest %iowait (in this case 3.49%).
>
> 2) Shoudn't it spread out and balance them to utilize all CPUs evenly?
>
> Thank you.
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rob4manhere at gmail.com  Fri Oct 23 10:30:23 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Fri, 23 Oct 2009 12:30:23 -0500
Subject: [Freeswitch-users] Setting up Conference with Moderator
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local>
	<3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local>
Message-ID: <E97CD3D0-52FF-44D1-BB89-23816045CE21@gmail.com>

The first caller isn't challenged for the pin (don't really know why--  
maybe somebody else can elaborate on how the pin is designed to be  
used).  So to work around it, I validate the conference pin and  
moderator pin independently via an IVR (or dynamically with a script  
and odbc call) then put them in the conference with the right profile  
and flags.

Here is a simple example (I created these wavs so they aren't in your  
default setup but you can get the idea):

in ivr.conf:

     <menu name="conference_menu"
           greet-long="welcome_please_enter_conference_pin.wav"
           greet-short="check_and_try_again.wav"
	  invalid-sound="passcode_invalid.wav"
	  exit-sound="voicemail/vm-goodbye.wav"
           timeout="10000"
           inter-digit-timeout="5000"
           max-failures="3"
           max-timeouts="3"
           digit-len="7">
       <entry action="menu-sub" digits="123456"  
param="conference_123456_moderator_menu" /> <!-- conference moderator  
menu -->
     </menu>

     <menu name="conference_123456_moderator_menu"
           greet- 
long 
= 
"conference_confirmed_enter_moderator_pin_or_1_to_join_as_participant 
.wav"
           greet-short="check_moderator_pin_or_1_to_join.wav"
           invalid-sound="invalid_moderator_pin.wav"
           exit-sound="voicemail/vm-goodbye.wav"
           timeout="10000"
           inter-digit-timeout="5000"
           max-failures="3"
           max-timeouts="3"
           digit-len="5">
       <entry action="menu-exec-app" digits="1234" param="conference  
123456 at default+flags{moderator}" />
       <entry action="menu-exec-app" digits="1" param="conference  
123456 at default+flags{}" />
      </menu>


You also need to add the "wait-mod" conference-flag to the conference  
profile if you want to allow participants to join but not let them  
talk until the moderator arrives.

In conference.conf.xml add to the profile you're using (default in  
this example):

       <param name="conference-flags" value="wait-mod" />


 From your public dial plan call the IVR or script:
   <action application="ivr" data="conference_menu" />


Now you should be able to call, enter a conference number of 123456  
and sit listening to music until the moderator joins (using pin 1234)  
and starts the call.  Maybe I completely misunderstood the built-in  
pin system but since the first person isn't challenged I had to do it  
this way.

Good luck!

Rob


On Oct 23, 2009, at 11:54 AM, Ujjval Karihaloo wrote:

> Any ideas on this one. Look slike only way rite now is to have a  
> different Dest Phone number for a moderator.
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Ujjval Karihaloo
> Sent: Thursday, October 22, 2009 9:02 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Setting up Conference with Moderator
>
> Hi
>
> I have the Basic Conferencing working. Here is my Dial Plan.
>
>  I want to be able to setup a Moderator PIN different from other  
> participants, when I add the moderator flag it logs me in directly w/ 
> o asking for a PIN..
> action application="conference" data="conference.conf at wideband 
> +flags{moderator}+159753"/>
>
> DialPlan is below for the normal user, and it asks for the PIN with  
> below settings.
>
>        Ujj Inbound from SS - start
>     -->
>     <extension name="simplesignal">   <!-- your provider or any name  
> you'd like to call it -->
>      <condition field="destination_number" expression="2142349127">   
> <!-- your DID for this gateway-->
>         <action application="conference"  
> data="conference.conf at wideband+159753"/>
>      </condition>
> </extension>
>
>     <!--
>        Ujj Inbound from SS ? end
>
>
>
>
> And I am using the existing conference.conf.xml file in the  
> auto_loads directory.
>
>
> Ujjval Karihaloo
> VP Voice Engineering
> IP Phone: +13032428610
> E-Fax: +17202391690
>
> SimpleSignal Inc.
> 88 Inverness Circle East
> Suite K105
> Englewood, CO  80112
> <image001.jpg>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jerry.richards at teotech.com  Fri Oct 23 10:31:06 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 23 Oct 2009 10:31:06 -0700
Subject: [Freeswitch-users] Inbound DTMF Not Recognized By IVR
Message-ID: <573AFFBA055B4ECD83A910C12BE67129@greyhawk.tonecommander.com>


I installed FS on a machine with a Sangoma A101D (PRI) card and if I make an
inbound call to the FS IVR, it does not recognize DTMF digits from the PSTN
phone.  If I call IVR from an internal phone, then it does recognize the
DTMF digits.  I have mostly default configurations for everything.

Best Regards,
JErry



From brian at freeswitch.org  Fri Oct 23 10:43:01 2009
From: brian at freeswitch.org (Brian West)
Date: Fri, 23 Oct 2009 12:43:01 -0500
Subject: [Freeswitch-users] Setting up Conference with Moderator
In-Reply-To: <E97CD3D0-52FF-44D1-BB89-23816045CE21@gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local>
	<3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local>
	<E97CD3D0-52FF-44D1-BB89-23816045CE21@gmail.com>
Message-ID: <71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>

Well first off you're not defining a pine here...

confname at profilename+flags{mute|deaf|waste|moderator}+[conference pin  
number]

That might be why its not asking for a pin.

/b

On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:

>       <entry action="menu-exec-app" digits="1" param="conference  
> 123456 at default+flags{}" />



From chris at cloudtel.com  Fri Oct 23 10:47:43 2009
From: chris at cloudtel.com (Chris Burns)
Date: Fri, 23 Oct 2009 13:47:43 -0400
Subject: [Freeswitch-users] sched_api doesn't get launched
In-Reply-To: <191c3a030910230802m1cb847agfdaf6e436296e748@mail.gmail.com>
References: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
	<59ad9ca10910230154i6d41ddfci8957b152c1884800@mail.gmail.com>
	<191c3a030910230802m1cb847agfdaf6e436296e748@mail.gmail.com>
Message-ID: <200910231347.43883.chris@cloudtel.com>

poor bbhenry :)

Added in r15207 please test and update docu if necessary:
http://wiki.freeswitch.org/wiki/Variable_api_on_answer

On October 23, 2009 11:02:07 am Anthony Minessale wrote:
> it's probably related to escaping the data.
> I was sick of watching you suffer so i added api_on_answer variable to
> trunk.
>
> On Fri, Oct 23, 2009 at 3:54 AM, Henry Huang 
<red.rain.seven at gmail.com>wrote:
> > Thanks to c6burns on IRC channel for the tip to use execute_on_answer in
> > combination with eval, and of course everyone here that pointing me to
> > the right direction.
> > I was able to execute sched_api with eval, but not with the combination
> > of execute_on_answer. The argument just don't get parsed as the eval
> > argument. Here is the code:
> >
> > setVariable("execute_on_answer", "eval sched_api(+%ld none uuid_displace
> > <some uuid> start
> > /usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux)");
> >
> >
> > Here is the log:
> > 2009-10-23 16:17:53.633189 [DEBUG] switch_scheduler.c:214 Added task 2
> > sched_api_function (none) to run at 1256285888
> > EXECUTE sofia/internal/1688004 at 192.168.1.66 *eval(+OK *Added: 2
> > )
> >
> > 2009-10-23 16:18:04.52701 [DEBUG] switch_channel.c:1927 sofia/internal/
> > 1688004 at 192.168.1.66 *execute on answer: eval(sched_api(+37 none
> > uuid_displace 84d74f41-8668-4138-943f-f076e94046ad start
> > /usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux))*
> > EXECUTE sofia/internal/1688004 at 192.168.1.66 eval(sched_api(+37 none
> > uuid_displace 84d74f41-8668-4138-943f-f076e94046ad start
> > /usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux))
> >
> >
> > The first part of the log is when I just do execute("eval",
> > "${sched_api(+%ld none uuid_displace <some uuid> start
> > /usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux)}");
> > And I simple get a OK, and then later on the sched_api gets executed.
> >
> > However, on the excerpt of the second log. which I use the combination
> > with execute_on_answer like specified in the yellow hight code above. It
> > doesn't simply return a OK, instead it shows the whole blob of code. and
> > does nothing later.
> >
> >
> >
> > On Thu, Oct 22, 2009 at 4:09 AM, Anthony Minessale <
> >
> > anthony.minessale at gmail.com> wrote:
> >> Yes you need an API object as described in other email.
> >>
> >> Which line of code from java caused that segfault
> >> It looks like a simple NULL string issue that we may want to hunt down.
> >>
> >> On Wed, Oct 21, 2009 at 4:44 AM, Henry Huang 
<red.rain.seven at gmail.com>wrote:
> >>> I can't seem to find the right thing to use in mod_java to execute api
> >>> commands, only api_after_bridge
> >>>
> >>> 2009-10-21 17:42:46.593094 [NOTICE] mod_sofia.c:1509 Pre-Answer
> >>> sofia/internal/1688003 at 192.168.1.66!
> >>> #
> >>> # A fatal error has been detected by the Java Runtime Environment:
> >>> #
> >>> #  SIGSEGV (0xb) at pc=0x004e4480, pid=1927, tid=16116624
> >>> #
> >>> # JRE version: 6.0_16-b01
> >>> # Java VM: Java HotSpot(TM) Client VM (14.2-b01 mixed mode linux-x86 )
> >>> # Problematic frame:
> >>> # C  [libc.so.6+0x6f480]  strcpy+0x10
> >>> #
> >>> # An error report file with more information is saved as:
> >>> # /usr/local/freeswitch/bin/hs_err_pid1927.log
> >>> 2009-10-21 17:42:59.883729 [ERR] switch_core_session.c:1374 Invalid
> >>> Application sched_api
> >>> 2009-10-21 17:42:59.883729 [NOTICE] switch_core_session.c:1375 Hangup
> >>> sofia/internal/1688003 at 192.168.1.66 [CS_EXECUTE]
> >>> [DESTINATION_OUT_OF_ORDER]
> >>> #
> >>> # If you would like to submit a bug report, please visit:
> >>> #   http://java.sun.com/webapps/bugreport/crash.jsp
> >>> # The crash happened outside the Java Virtual Machine in native code.
> >>> # See problematic frame for where to report the bug.
> >>>
> >>> On Sat, Oct 17, 2009 at 3:37 AM, Michael Collins 
<msc at freeswitch.org>wrote:
> >>>> On Fri, Oct 16, 2009 at 11:53 AM, Henry Huang
> >>>> <red.rain.seven at gmail.com
> >>>>
> >>>> > wrote:
> >>>>>
> >>>>> So how would you trigger it from a script dialplan? The only time it
> >>>>> seemed to work is when I did setVariable("api_after_bridge",
> >>>>> "sched_api blah blah blah");
> >>>>> but then it gets executed after the channel's been teared down. I
> >>>>> thought api_after_bridge means right after the call gets connected.
> >>>>>
> >>>>> I need something to execute an api command right before or right
> >>>>> after the call gets bridged.
> >>>>>
> >>>>> api_after_bridge is a channel variable, so using setVariable works
> >>>>> just
> >>>>
> >>>> fine. If you need to sched_api is an API only. Check these out:
> >>>> http://wiki.freeswitch.org/wiki/Mod_commands#Misc._Commands
> >>>>
> >>>> So you need an API object in order to use it. I don't know the syntax
> >>>> for creating an api obj in Java but in Lua it goes like this:
> >>>> api = freeswitch.API();
> >>>> res = api:execute("sched_api","+300 none my_api my_api_args")
> >>>>
> >>>> Remember, if the method you are using isn't found in the dial plan
> >>>> tools then it isn't a dial plan application. Make sure it's on the
> >>>> list: http://wiki.freeswitch.org/wiki/Mod_dptools
> >>>>
> >>>> On the other hand, API commands are listed here:
> >>>> http://wiki.freeswitch.org/wiki/Mod_commands
> >>>>
> >>>> dptools require a session object, api commands require an api
> >>>> object...
> >>>>
> >>>> -MC
> >>>>
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>
> >>> --
> >>> Henry Huang
> >>> UniC Solution - Communication Unified
> >>> VoIP & Open Source software Consultant
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> >>>s http://www.freeswitch.org
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessa
> >>le at gmail.com> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> iax:guest at conference.freeswitch.org/888
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at co
> >>nference.freeswitch.org> pstn:213-799-1400
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > --
> > Henry Huang
> > UniC Solution - Communication Unified
> > VoIP & Open Source software Consultant
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org




From rob4manhere at gmail.com  Fri Oct 23 11:00:12 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Fri, 23 Oct 2009 13:00:12 -0500
Subject: [Freeswitch-users] Setting up Conference with Moderator
In-Reply-To: <71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local>
	<3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local>
	<E97CD3D0-52FF-44D1-BB89-23816045CE21@gmail.com>
	<71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>
Message-ID: <89C5467E-A7BC-4DA6-94C2-51FAF5793E68@gmail.com>

Right- this was an example without using the built-in pin but instead  
using IVR.


On Oct 23, 2009, at 12:43 PM, Brian West wrote:

> Well first off you're not defining a pine here...
>
> confname at profilename+flags{mute|deaf|waste|moderator}+[conference pin
> number]
>
> That might be why its not asking for a pin.
>
> /b
>
> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:
>
>>      <entry action="menu-exec-app" digits="1" param="conference
>> 123456 at default+flags{}" />
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From kristian.kielhofner at gmail.com  Fri Oct 23 11:14:07 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 23 Oct 2009 14:14:07 -0400
Subject: [Freeswitch-users] Issues with SIP+TCP?
Message-ID: <2d9149cd0910231114p7dd3c0cfo10ad30003929e4f8@mail.gmail.com>

Hello everyone,

  I'm having some issues with SIP and TCP.  I've used it before with
success but I'm seeing some strange behavior...

 Level 7 debugs with siptrace on both profiles.  UDP invite from
softphone comes in on port 5062, it's supposed to bridge to
10.70.0.62.  When configured to use UDP FS sends an INVITE (nothing
currently answers) while TCP doesn't send anything (confirmed with
siptrace and packet sniffer).  I confirmed this behavior with a
gateway configured for TCP and appending ;transport=tcp to a bridge
line.

  This is trunk rev 15211 on an Intel Mac running Snow Leopard.  I've
also confirmed this behavior on an Intel Linux machine running Ubuntu
(not sure of version ATM).

TCP:

http://pastebin.freeswitch.org/10825

UDP:

http://pastebin.freeswitch.org/10826

dialplan (UDP):

    <extension name="smhpbx">
      <condition field="destination_number" expression="^(7887)$">
        <action application="set" data="call_timeout=60"/>
        <action application="set" data="effective_caller_id_name=Voalte Test"/>
        <action application="set"
data="effective_caller_id_number=19412848354"/>
        <action application="bridge" data="sofia/avaya/7887 at 10.70.0.62"/>
      </condition>
    </extension>

dialplan (TCP):

    <extension name="smhpbx">
      <condition field="destination_number" expression="^(7887)$">
        <action application="set" data="call_timeout=60"/>
        <action application="set" data="effective_caller_id_name=Voalte Test"/>
        <action application="set"
data="effective_caller_id_number=19412848354"/>
        <action application="bridge"
data="sofia/avaya/7887 at 10.70.0.62;transport=tcp"/>
      </condition>
    </extension>

  Any thoughts?

Thanks!

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From bottleman at icf.org.ru  Fri Oct 23 11:17:05 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Fri, 23 Oct 2009 22:17:05 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910230926y6c3e7ee3i9bb2182da1ba8757@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910230452s6b4a179aqa5bc41dc34efc403@mail.gmail.com>
	<Pine.LNX.4.64.0910231610220.21492@srv.icf.org.ru>
	<65d96fc80910230538w7105be6fve14956ab3b247bf6@mail.gmail.com>
	<Pine.LNX.4.64.0910231646210.29582@srv.icf.org.ru>
	<65d96fc80910230614m2b2504d0r92006b78d267a399@mail.gmail.com>
	<65d96fc80910230757h4595f396o2dc1a1bb781d3a8a@mail.gmail.com>
	<Pine.LNX.4.64.0910231903510.17804@srv.icf.org.ru>
	<191c3a030910230837x78ea6093i387617a1adfdf958@mail.gmail.com>
	<Pine.LNX.4.64.0910231943250.22146@srv.icf.org.ru>
	<65d96fc80910230926y6c3e7ee3i9bb2182da1ba8757@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910232208030.29940@srv.icf.org.ru>

On 2009-10-23 18:26 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>2009/10/23 Georgiewskiy Yuriy <bottleman at icf.org.ru>
TC>
TC>> On 2009-10-23 10:37 -0500, Anthony Minessale wrote
TC>> freeswitch-users at lists.f...:
TC>>
TC>> i have no way to install trunk at this time, i will go out of hospital
TC>> about one week later, after
TC>> this i will can try it on trunk.
TC>>
TC>> AM>if you were on trunk that line of code would be gone.
TC>> AM>you really can't do development on 1.0.4 its 6 months old and it will
TC>> cause
TC>> AM>you more trouble than you think when you eventually upgrade if you do
TC>> not do
TC>>
TC>
TC>uh... you are still in there ... damn sorry to hear that.
TC>I know what the feeling is... i was in the same position a couple of months
TC>ago... anyhow at least i had plenty of time to do my private research.. hope
TC>it is the case with you as well.
TC>
TC>so, if you have that time i can offer you my development server with trunk
TC>installed and all the hands on.
TC>
TC>
TC>let me know...
TC>
TC>BTW: it really doesn't have sense to develop on 1.0.4 ... the proof of
TC>concept was done. I'm able to place calls in both directions so, lets move
TC>to trunk now.

i have my own voip infrastructure on my work, and it's better for me to use it for tests, 
and use my personal on work too, i have problem with various hardware and call making, 
there is i have dialup connection and all what i can - make one call whit only g729 codec, 
or receive one call, no more, it's is a bandwich limitation. from hardware i have only ip 
phone artdio ipf2000 and one addpac gateway. also ssh sesssion is very slow.

P.S. people from russian community report what current version of module work fine on fs 
trunk version.

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From rob4manhere at gmail.com  Fri Oct 23 11:22:53 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Fri, 23 Oct 2009 13:22:53 -0500
Subject: [Freeswitch-users] Setting up Conference with Moderator
In-Reply-To: <71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local>
	<3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local>
	<E97CD3D0-52FF-44D1-BB89-23816045CE21@gmail.com>
	<71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>
Message-ID: <114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com>

I just re-tested with the pin in my dial plan:

   <action application="conference" data="conference 123456 at default 
+flags{}+1234" />

And it doesn't challenge me for the pin.  I just drop right in.  I  
figured this is how it was intended, since the wiki says the pin is  
set initially and only challenged in later attempts [by future callers]:

"The first time a conference name (confname) is used, it will be  
created on demand, and the pin will be set to what ever is specified  
at that time: the pin in the data string if specified, or if not, the  
"pin" setting in the conference profile, and if that is also  
unspecified, then there is no pin protection. Any later attempt to  
join the conference must specify the same pin number, if one existed  
when it was created. "


The wiki also says that the wait-mod might be  "used in conjunction  
with an IVR where the moderators are authenticated with an extra pass- 
code", which is what I did.  I guess that's why I didn't understand  
the point of the +pin.

I'm sure there's a scenario where its used and useful, the wiki just  
doesn't explain it.

Rob

On Oct 23, 2009, at 12:43 PM, Brian West wrote:

> Well first off you're not defining a pine here...
>
> confname at profilename+flags{mute|deaf|waste|moderator}+[conference pin
> number]
>
> That might be why its not asking for a pin.
>
> /b
>
> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:
>
>>      <entry action="menu-exec-app" digits="1" param="conference
>> 123456 at default+flags{}" />
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From msc at freeswitch.org  Fri Oct 23 12:00:28 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 23 Oct 2009 12:00:28 -0700
Subject: [Freeswitch-users] "Can not record session. Media not enabled
	on channel."
In-Reply-To: <1256283391890-3877285.post@n2.nabble.com>
References: <1256034659005-3857858.post@n2.nabble.com>
	<87f2f3b90910201746h43a31b32m5c163fd98373f915@mail.gmail.com>
	<1256283391890-3877285.post@n2.nabble.com>
Message-ID: <87f2f3b90910231200v71e72661id4cc9f2b7fba5610@mail.gmail.com>

On Fri, Oct 23, 2009 at 12:36 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> The dialplan is very simple:
>
>        <extension name="Recording test">
>                <condition field="destination_number"
> expression="^111111\d*$">
>                        <action application="sleep" data="3000" />
>                        <action application="answer"/>
>                        <action application="say" data="en name_spelled
> iterated
> ${destination_number}"/>
>                        <action application="sleep" data="1000" />
>                        <action application="playback"
> data="local_stream://my_music"/>
>                </condition>
>        </extension>
>
> Before debugging I have another question: I start recording in event
> handler
> for ChannelAnswer event. Is it possible that it's too soon to start
> recording? Maybe I should start recording in some other event?
>

That would be an odd scenario but maybe. It would be best if you could catch
it in the act so that we could see exactly what is happening. The other
thing you could do is deliberately start recording on the channel prior to
answering and see if you always get the error. In other words, try to make
it fail under a certain set of circumstances to see if your theory is
correct.
-MC
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From msc at freeswitch.org  Fri Oct 23 12:01:44 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 23 Oct 2009 12:01:44 -0700
Subject: [Freeswitch-users] CPU Usage Question
In-Reply-To: <c96b95650910231016me63190fr4641ed7737637a85@mail.gmail.com>
References: <4add4a7a.1408c00a.4283.ffffc11f@mx.google.com>
	<c96b95650910231016me63190fr4641ed7737637a85@mail.gmail.com>
Message-ID: <87f2f3b90910231201x11644dfal60c4a8556404abe@mail.gmail.com>

On Fri, Oct 23, 2009 at 10:16 AM, Saeed Ahmad <saeedahmad1981 at gmail.com>wrote:

> I think we can't create two profiles per single IP. So what is the
> workaround to solve this issue?
>
> SAT
>
>
Two profiles can share the same IP address, just not the same IP and port.
The example internal and external profiles both use the same IP address but
they use different ports (5060 and 5080).
-MC
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From freeswitch.noob at gmail.com  Fri Oct 23 12:34:09 2009
From: freeswitch.noob at gmail.com (freeswitch noob)
Date: Fri, 23 Oct 2009 14:34:09 -0500
Subject: [Freeswitch-users] Hostname
Message-ID: <7f783b530910231234u5d0e2fedxea4884d9bcddc9d3@mail.gmail.com>

I have mod xml_curl installed and I am getting the following passed to my
script.

[hostname] => myhost.local
[section] => dialplan


I also have multiple versions of FS running on the same box.  Is there a way
to have each FS instance on my box have a unique hostname ?

Thanks in advance.
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From kristian.kielhofner at gmail.com  Fri Oct 23 12:46:11 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 23 Oct 2009 15:46:11 -0400
Subject: [Freeswitch-users] Hostname
In-Reply-To: <7f783b530910231234u5d0e2fedxea4884d9bcddc9d3@mail.gmail.com>
References: <7f783b530910231234u5d0e2fedxea4884d9bcddc9d3@mail.gmail.com>
Message-ID: <2d9149cd0910231246t58d2bbd3k3466d9c0fca18000@mail.gmail.com>

Can't you use different contexts or something else to tell them apart?

On Fri, Oct 23, 2009 at 3:34 PM, freeswitch noob
<freeswitch.noob at gmail.com> wrote:
> I have mod xml_curl installed and I am getting the following passed to my
> script.
>
> [hostname] => myhost.local
> [section] => dialplan
>
>
> I also have multiple versions of FS running on the same box.? Is there a way
> to have each FS instance on my box have a unique hostname ?
>
> Thanks in advance.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From chris at cloudtel.com  Fri Oct 23 12:58:09 2009
From: chris at cloudtel.com (Chris Burns)
Date: Fri, 23 Oct 2009 15:58:09 -0400
Subject: [Freeswitch-users] Hostname
In-Reply-To: <2d9149cd0910231246t58d2bbd3k3466d9c0fca18000@mail.gmail.com>
References: <7f783b530910231234u5d0e2fedxea4884d9bcddc9d3@mail.gmail.com>
	<2d9149cd0910231246t58d2bbd3k3466d9c0fca18000@mail.gmail.com>
Message-ID: <200910231558.09699.chris@cloudtel.com>

one real quick way would be put different GET var in each server's binding

On October 23, 2009 03:46:11 pm Kristian Kielhofner wrote:
> Can't you use different contexts or something else to tell them apart?
>
> On Fri, Oct 23, 2009 at 3:34 PM, freeswitch noob
>
> <freeswitch.noob at gmail.com> wrote:
> > I have mod xml_curl installed and I am getting the following passed to my
> > script.
> >
> > [hostname] => myhost.local
> > [section] => dialplan
> >
> >
> > I also have multiple versions of FS running on the same box.? Is there a
> > way to have each FS instance on my box have a unique hostname ?
> >
> > Thanks in advance.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org




From freeswitch.noob at gmail.com  Fri Oct 23 13:42:48 2009
From: freeswitch.noob at gmail.com (freeswitch noob)
Date: Fri, 23 Oct 2009 15:42:48 -0500
Subject: [Freeswitch-users] Hostname
In-Reply-To: <200910231558.09699.chris@cloudtel.com>
References: <7f783b530910231234u5d0e2fedxea4884d9bcddc9d3@mail.gmail.com>
	<2d9149cd0910231246t58d2bbd3k3466d9c0fca18000@mail.gmail.com>
	<200910231558.09699.chris@cloudtel.com>
Message-ID: <7f783b530910231342t5393763aueff6bbc679b3c55@mail.gmail.com>

Yeah, I was just trying to make it easier on myself.  I have scripts from a
friend that parse xml_curl requests based on the hostname, I was hoping to
not have to re-write them to read something else from the post that FS makes
from xml_curl.  But from what it sounds like I will have to.


On Fri, Oct 23, 2009 at 2:58 PM, Chris Burns <chris at cloudtel.com> wrote:

> one real quick way would be put different GET var in each server's binding
>
> On October 23, 2009 03:46:11 pm Kristian Kielhofner wrote:
> > Can't you use different contexts or something else to tell them apart?
> >
> > On Fri, Oct 23, 2009 at 3:34 PM, freeswitch noob
> >
> > <freeswitch.noob at gmail.com> wrote:
> > > I have mod xml_curl installed and I am getting the following passed to
> my
> > > script.
> > >
> > > [hostname] => myhost.local
> > > [section] => dialplan
> > >
> > >
> > > I also have multiple versions of FS running on the same box.  Is there
> a
> > > way to have each FS instance on my box have a unique hostname ?
> > >
> > > Thanks in advance.
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From freeswitch-users-list at metik.com  Fri Oct 23 14:37:35 2009
From: freeswitch-users-list at metik.com (Metik)
Date: Fri, 23 Oct 2009 17:37:35 -0400
Subject: [Freeswitch-users] Hostname
In-Reply-To: <7f783b530910231342t5393763aueff6bbc679b3c55@mail.gmail.com>
References: <7f783b530910231234u5d0e2fedxea4884d9bcddc9d3@mail.gmail.com><2d9149cd0910231246t58d2bbd3k3466d9c0fca18000@mail.gmail.com><200910231558.09699.chris@cloudtel.com>
	<7f783b530910231342t5393763aueff6bbc679b3c55@mail.gmail.com>
Message-ID: <65286EEC1128407F82A019CF0EFB9C38@chaos>

Why not simply overwrite the value of the variable used throughout the script... 

-- xml_curl.conf --
...
      <param name="gateway-url" value="http://localhost/index.php?xhostname=myhost" bindings="dialplan"/>
...

-- index.php --

<?

$_REQUEST['hostname'] = "$xhostname";

...
  ----- Original Message ----- 
  From: freeswitch noob 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, October 23, 2009 4:42 PM
  Subject: Re: [Freeswitch-users] Hostname


  Yeah, I was just trying to make it easier on myself.  I have scripts from a friend that parse xml_curl requests based on the hostname, I was hoping to not have to re-write them to read something else from the post that FS makes from xml_curl.  But from what it sounds like I will have to.



  On Fri, Oct 23, 2009 at 2:58 PM, Chris Burns <chris at cloudtel.com> wrote:

    one real quick way would be put different GET var in each server's binding


    On October 23, 2009 03:46:11 pm Kristian Kielhofner wrote:
    > Can't you use different contexts or something else to tell them apart?
    >
    > On Fri, Oct 23, 2009 at 3:34 PM, freeswitch noob
    >
    > <freeswitch.noob at gmail.com> wrote:
    > > I have mod xml_curl installed and I am getting the following passed to my
    > > script.
    > >
    > > [hostname] => myhost.local
    > > [section] => dialplan
    > >
    > >
    > > I also have multiple versions of FS running on the same box.  Is there a
    > > way to have each FS instance on my box have a unique hostname ?
    > >
    > > Thanks in advance.
    > >
    > > _______________________________________________
    > > FreeSWITCH-users mailing list
    > > FreeSWITCH-users at lists.freeswitch.org
    > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    > > http://www.freeswitch.org



    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From freeswitch-users-list at metik.com  Fri Oct 23 14:48:05 2009
From: freeswitch-users-list at metik.com (Metik)
Date: Fri, 23 Oct 2009 17:48:05 -0400
Subject: [Freeswitch-users] Hostname
Message-ID: <AEDF2544E2D24EC5ACEB0361357C16EE@chaos>

Please note that this would essentially be taking Chris' suggestion a little further but the effort involved would be minimal.  
  ----- Original Message ----- 
  From: Metik 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, October 23, 2009 5:37 PM
  Subject: Re: [Freeswitch-users] Hostname


  Why not simply overwrite the value of the variable used throughout the script... 

  -- xml_curl.conf --
  ...
        <param name="gateway-url" value="http://localhost/index.php?xhostname=myhost" bindings="dialplan"/>
  ...

  -- index.php --

  <?

  $_REQUEST['hostname'] = "$xhostname";

  ...
    ----- Original Message ----- 
    From: freeswitch noob 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Friday, October 23, 2009 4:42 PM
    Subject: Re: [Freeswitch-users] Hostname


    Yeah, I was just trying to make it easier on myself.  I have scripts from a friend that parse xml_curl requests based on the hostname, I was hoping to not have to re-write them to read something else from the post that FS makes from xml_curl.  But from what it sounds like I will have to.



    On Fri, Oct 23, 2009 at 2:58 PM, Chris Burns <chris at cloudtel.com> wrote:

      one real quick way would be put different GET var in each server's binding


      On October 23, 2009 03:46:11 pm Kristian Kielhofner wrote:
      > Can't you use different contexts or something else to tell them apart?
      >
      > On Fri, Oct 23, 2009 at 3:34 PM, freeswitch noob
      >
      > <freeswitch.noob at gmail.com> wrote:
      > > I have mod xml_curl installed and I am getting the following passed to my
      > > script.
      > >
      > > [hostname] => myhost.local
      > > [section] => dialplan
      > >
      > >
      > > I also have multiple versions of FS running on the same box.  Is there a
      > > way to have each FS instance on my box have a unique hostname ?
      > >
      > > Thanks in advance.
      > >
      > > _______________________________________________
      > > FreeSWITCH-users mailing list
      > > FreeSWITCH-users at lists.freeswitch.org
      > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      > > http://www.freeswitch.org



      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org





----------------------------------------------------------------------------


    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From freeswitch-users-list at metik.com  Fri Oct 23 15:18:41 2009
From: freeswitch-users-list at metik.com (Metik)
Date: Fri, 23 Oct 2009 18:18:41 -0400
Subject: [Freeswitch-users] Hostname
In-Reply-To: <AEDF2544E2D24EC5ACEB0361357C16EE@chaos>
References: <AEDF2544E2D24EC5ACEB0361357C16EE@chaos>
Message-ID: <A52BE10FDCCE4267966353EA5C5F02D7@chaos>

Forget it--this will not work because FS uses a POST (vs GET). Most likely it would attempt to actually POST to a file named "index.php?xhostname=myhost".

However, if there was some way to add an arbitrary POST variable to the HTTP transaction, it would work. I am sure the FS developers have already thought of this and its a case of RTFM.

-metik
  ----- Original Message ----- 
  From: Metik 
  To: Metik ; freeswitch-users at lists.freeswitch.org 
  Sent: Friday, October 23, 2009 5:48 PM
  Subject: Re: [Freeswitch-users] Hostname


  Please note that this would essentially be taking Chris' suggestion a little further but the effort involved would be minimal.  
    ----- Original Message ----- 
    From: Metik 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Friday, October 23, 2009 5:37 PM
    Subject: Re: [Freeswitch-users] Hostname


    Why not simply overwrite the value of the variable used throughout the script... 

    -- xml_curl.conf --
    ...
          <param name="gateway-url" value="http://localhost/index.php?xhostname=myhost" bindings="dialplan"/>
    ...

    -- index.php --

    <?

    $_REQUEST['hostname'] = "$xhostname";

    ...
      ----- Original Message ----- 
      From: freeswitch noob 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Friday, October 23, 2009 4:42 PM
      Subject: Re: [Freeswitch-users] Hostname


      Yeah, I was just trying to make it easier on myself.  I have scripts from a friend that parse xml_curl requests based on the hostname, I was hoping to not have to re-write them to read something else from the post that FS makes from xml_curl.  But from what it sounds like I will have to.



      On Fri, Oct 23, 2009 at 2:58 PM, Chris Burns <chris at cloudtel.com> wrote:

        one real quick way would be put different GET var in each server's binding


        On October 23, 2009 03:46:11 pm Kristian Kielhofner wrote:
        > Can't you use different contexts or something else to tell them apart?
        >
        > On Fri, Oct 23, 2009 at 3:34 PM, freeswitch noob
        >
        > <freeswitch.noob at gmail.com> wrote:
        > > I have mod xml_curl installed and I am getting the following passed to my
        > > script.
        > >
        > > [hostname] => myhost.local
        > > [section] => dialplan
        > >
        > >
        > > I also have multiple versions of FS running on the same box.  Is there a
        > > way to have each FS instance on my box have a unique hostname ?
        > >
        > > Thanks in advance.
        > >
        > > _______________________________________________
        > > FreeSWITCH-users mailing list
        > > FreeSWITCH-users at lists.freeswitch.org
        > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        > > http://www.freeswitch.org



        _______________________________________________
        FreeSWITCH-users mailing list
        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org





--------------------------------------------------------------------------


      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From chris at cloudtel.com  Fri Oct 23 16:23:03 2009
From: chris at cloudtel.com (Chris Burns)
Date: Fri, 23 Oct 2009 19:23:03 -0400
Subject: [Freeswitch-users] Hostname
In-Reply-To: <A52BE10FDCCE4267966353EA5C5F02D7@chaos>
References: <AEDF2544E2D24EC5ACEB0361357C16EE@chaos>
	<A52BE10FDCCE4267966353EA5C5F02D7@chaos>
Message-ID: <200910231923.03648.chris@cloudtel.com>

Fake the hostname in the query string:
http://test/dir.php?hostname=mmmpotatoes

Add this to the top of your buddy's script (if its even PHP):
$hostname = $_POST['hostname'] = $_REQUEST['hostname'] = $_GET['hostname'];


On October 23, 2009 06:18:41 pm Metik wrote:
> Forget it--this will not work because FS uses a POST (vs GET). Most likely
> it would attempt to actually POST to a file named
> "index.php?xhostname=myhost".
>
> However, if there was some way to add an arbitrary POST variable to the
> HTTP transaction, it would work. I am sure the FS developers have already
> thought of this and its a case of RTFM.
>
> -metik
>   ----- Original Message -----
>   From: Metik
>   To: Metik ; freeswitch-users at lists.freeswitch.org
>   Sent: Friday, October 23, 2009 5:48 PM
>   Subject: Re: [Freeswitch-users] Hostname
>
>
>   Please note that this would essentially be taking Chris' suggestion a
> little further but the effort involved would be minimal. ----- Original
> Message -----
>     From: Metik
>     To: freeswitch-users at lists.freeswitch.org
>     Sent: Friday, October 23, 2009 5:37 PM
>     Subject: Re: [Freeswitch-users] Hostname
>
>
>     Why not simply overwrite the value of the variable used throughout the
> script...
>
>     -- xml_curl.conf --
>     ...
>           <param name="gateway-url"
> value="http://localhost/index.php?xhostname=myhost" bindings="dialplan"/>
> ...
>
>     -- index.php --
>
>     <?
>
>     $_REQUEST['hostname'] = "$xhostname";
>
>     ...
>       ----- Original Message -----
>       From: freeswitch noob
>       To: freeswitch-users at lists.freeswitch.org
>       Sent: Friday, October 23, 2009 4:42 PM
>       Subject: Re: [Freeswitch-users] Hostname
>
>
>       Yeah, I was just trying to make it easier on myself.  I have scripts
> from a friend that parse xml_curl requests based on the hostname, I was
> hoping to not have to re-write them to read something else from the post
> that FS makes from xml_curl.  But from what it sounds like I will have to.
>
>
>
>       On Fri, Oct 23, 2009 at 2:58 PM, Chris Burns <chris at cloudtel.com>
> wrote:
>
>         one real quick way would be put different GET var in each server's
> binding
>
>         On October 23, 2009 03:46:11 pm Kristian Kielhofner wrote:
>         > Can't you use different contexts or something else to tell them
>         > apart?
>         >
>         > On Fri, Oct 23, 2009 at 3:34 PM, freeswitch noob
>         >
>         > <freeswitch.noob at gmail.com> wrote:
>         > > I have mod xml_curl installed and I am getting the following
>         > > passed to my script.
>         > >
>         > > [hostname] => myhost.local
>         > > [section] => dialplan
>         > >
>         > >
>         > > I also have multiple versions of FS running on the same box. 
>         > > Is there a way to have each FS instance on my box have a unique
>         > > hostname ?
>         > >
>         > > Thanks in advance.
>         > >
>         > > _______________________________________________
>         > > FreeSWITCH-users mailing list
>         > > FreeSWITCH-users at lists.freeswitch.org
>         > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswi
>         > >tch-users http://www.freeswitch.org
>
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>        
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
> --------------------------------------------------------------------------
>
>
>       _______________________________________________
>       FreeSWITCH-users mailing list
>       FreeSWITCH-users at lists.freeswitch.org
>       http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>      
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From nagalenoj at gmail.com  Fri Oct 23 23:11:36 2009
From: nagalenoj at gmail.com (Nagalenoj)
Date: Fri, 23 Oct 2009 23:11:36 -0700 (PDT)
Subject: [Freeswitch-users] UUID of the newly originated call?
In-Reply-To: <191c3a030910221012w125a15d4m24f5fbe72c8a859d@mail.gmail.com>
References: <fe67a3340910202306i55a0b9ceve92ab19e000edf11@mail.gmail.com>
	<191c3a030910211720l26bd0303i2cb11e9adb28037d@mail.gmail.com>
	<26006565.post@talk.nabble.com>
	<191c3a030910221012w125a15d4m24f5fbe72c8a859d@mail.gmail.com>
Message-ID: <26036438.post@talk.nabble.com>


It is working fine in the current version(15203) of SVN. Thanks.


Anthony Minessale-2 wrote:
> 
> please update to svn trunk with "make current" and try again.
> 
> 
> On Thu, Oct 22, 2009 at 4:08 AM, Nagalenoj <nagalenoj at gmail.com> wrote:
> 
>>
>> I'm using 'FreeSWITCH Version 1.0.trunk (15106)'.
>>
>>
>> Anthony Minessale-2 wrote:
>> >
>> > which revision of FS are you using?
>> >
>> > On Wed, Oct 21, 2009 at 1:06 AM, Nagalenoj H. <nagalenoj at gmail.com>
>> wrote:
>> >
>> >> I've tried with origination_uuid.
>> >>
>> >> First, I tried with SIP and my program executes successfully as what I
>> >> expected. This program initiates a new call when a call comes and let
>> the
>> >> new call to eavesdrop the landed call.
>> >> When, I experimented with PRI(openzap), I'm facing the following
>> error.
>> >> And, I am unable to make any calls(even from CLI). It is reporting the
>> >> same
>> >> error for the subsequent calls.
>> >> Error:        2009-10-21 11:08:31.661252 [ERR] mod_openzap.c:1760
>> Session
>> >> for channel 1:1 not found [UUID: 8364eb30-be03-11de-aa07-07e76b7651a9]
>> >>
>> >> Full log is in http://pastebin.freeswitch.org/10780
>> >> My script is here, http://pastebin.freeswitch.org/10781
>> >>
>> >> What is this error for and how to avoid this?
>> >>
>> >> Is there any other way to get the uuid of the originated call except
>> >> explicitly defining(origination_uuid).?!
>> >>
>> >> --
>> >> Regards,
>> >> Nagalenoj H.
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> <MSN%3Aanthony_minessale at hotmail.com><
>> MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
>> >
>> >
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
>> >
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> <sip%3A888 at conference.freeswitch.org><
>> sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
>> >
>> > iax:guest at conference.freeswitch.org/888
>> >
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
>> >
>> > pstn:213-799-1400
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> --
>> View this message in context:
>> http://www.nabble.com/Re%3A-UUID-of-the-newly-originated-call--tp25987024p26006565.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/Re%3A-UUID-of-the-newly-originated-call--tp25987024p26036438.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From tculjaga at gmail.com  Sat Oct 24 01:04:33 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Sat, 24 Oct 2009 10:04:33 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910232208030.29940@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910230538w7105be6fve14956ab3b247bf6@mail.gmail.com>
	<Pine.LNX.4.64.0910231646210.29582@srv.icf.org.ru>
	<65d96fc80910230614m2b2504d0r92006b78d267a399@mail.gmail.com>
	<65d96fc80910230757h4595f396o2dc1a1bb781d3a8a@mail.gmail.com>
	<Pine.LNX.4.64.0910231903510.17804@srv.icf.org.ru>
	<191c3a030910230837x78ea6093i387617a1adfdf958@mail.gmail.com>
	<Pine.LNX.4.64.0910231943250.22146@srv.icf.org.ru>
	<65d96fc80910230926y6c3e7ee3i9bb2182da1ba8757@mail.gmail.com>
	<Pine.LNX.4.64.0910232208030.29940@srv.icf.org.ru>
Message-ID: <65d96fc80910240104m458849c0xf7a17d3560496cbf@mail.gmail.com>

> TC>BTW: it really doesn't have sense to develop on 1.0.4 ... the proof of
> TC>concept was done. I'm able to place calls in both directions so, lets
> move
> TC>to trunk now.
>
> i have my own voip infrastructure on my work, and it's better for me to use
> it for tests,
> and use my personal on work too, i have problem with various hardware and
> call making,
> there is i have dialup connection and all what i can - make one call whit
> only g729 codec,
> or receive one call, no more, it's is a bandwich limitation. from hardware
> i have only ip
> phone artdio ipf2000 and one addpac gateway. also ssh sesssion is very
> slow.
>
>
I see... it is not the ideal development environment :)... Indeed, a dialup
is not a good thing... i guess your g729 is compressed to the maximum :)



> P.S. people from russian community report what current version of module
> work fine on fs
> trunk version.
>
>
that's strange that they report it working as "m_txAudioOpened" is never
gonna be ready in pre_answer :P... i had to comment it to make it working.

anyhow, i moved everything to trunk and will do some tests on Monday.

T
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From raffaele.p.guidi at gmail.com  Sat Oct 24 01:42:35 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Sat, 24 Oct 2009 10:42:35 +0200
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <7A73AEA0-9138-4E02-897A-85159405A4EE@jerris.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<86a32abc0910171513n35d7d0echff940a16685ae1a4@mail.gmail.com>
	<86a32abc0910171521n403094bfvf27aa7c2a1c1f798@mail.gmail.com>
	<86a32abc0910171535l606eb9c6id54f7b3b5ae1b670@mail.gmail.com>
	<86a32abc0910171545l17f3f539ldba432db550c3eaa@mail.gmail.com>
	<86a32abc0910171601u6ab06e42xcf303301c85c57b1@mail.gmail.com>
	<86a32abc0910171605y73ecb75cqfb13923ad19adc56@mail.gmail.com>
	<86a32abc0910171711je1e57bduf2bb672472365ce1@mail.gmail.com>
	<00E26046-328F-440A-81F4-671F197F114C@gmail.com>
	<7A73AEA0-9138-4E02-897A-85159405A4EE@jerris.com>
Message-ID: <b0a8792f0910240142j1807a9d9i5f8f9bf46b2bf090@mail.gmail.com>

+1 for Confluence

On Tue, Oct 20, 2009 at 00:55, Michael Jerris <mike at jerris.com> wrote:

> we do have a license for this, people didn't seem to like it last time
> we looked at it, I can't recall why.
>
> On Oct 19, 2009, at 4:24 PM, Roberto Martins wrote:
>
> > what about http://www.atlassian.com/software/confluence/ they give
> > free licenses to open source project, and FS is using JIRA.
> >
> > roberto
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From odermann at googlemail.com  Sat Oct 24 05:13:33 2009
From: odermann at googlemail.com (Dennis)
Date: Sat, 24 Oct 2009 14:13:33 +0200
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
	<191c3a030910131101p418e0961t373db68285248ff6@mail.gmail.com>
	<538F4CF43B294D0EACABA281E1913B4C@chaos>
	<65d96fc80910131224m73f35b87g355be818ad3c0caf@mail.gmail.com>
	<48B637E0506F4A809E4C0CAF39BD08BB@chaos>
	<65d96fc80910132348t202905fbub57cc4c814eb4e21@mail.gmail.com>
	<5e414ed0910140731w1c7ebedr150e69cda8073155@mail.gmail.com>
	<191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com>
	<5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com>
	<191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com>
Message-ID: <5e414ed0910240513q316905ai5cf8c2ef63b52f60@mail.gmail.com>

ok, as written, i come back after some tests with fs and a thomson cirpack.

it did not work - at least in our tests.

we are using socket outbound and when a call comes in, it starts the
socket of fs. the number may be 123456. fs sends the respond 484 and
our carrier receives this information. but fs ends the call with
hangup_cause = invalid_number_format.
the carrier has one more digit for the phone number and sends 1234567
and the above mentioned behavior repeats.

the behavior we want and expected is, that the call stays in the
socket after response 484, so that the carrier can send the 1234567
into the same socket.
the management, when fs should send response 484 and when fs should be
answered would be programmed by us.
it also important, that fs keeps the call in the socket, so we can
tell fs, to answer the call after x seconds anyway.

any ideas, what we could do?


From djbinter at yahoo.com  Sat Oct 24 10:35:14 2009
From: djbinter at yahoo.com (DJB)
Date: Sat, 24 Oct 2009 10:35:14 -0700 (PDT)
Subject: [Freeswitch-users] Core Dump question!
In-Reply-To: <86a32abc0910222215p6eb79f66lbbef2b245f4d17fd@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701E840EB@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221258o6936000dn6eb2e22189c45435@mail.gmail.com>
	<DF11CCC6-5C89-4489-908C-92A0FCC94AE4@jerris.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E841E8@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221725g6233ec20w49a73864400426d6@mail.gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E84200@EXMBXCLUS01.citservers.local>
	<86a32abc0910222004y46bf08b6jda75b6b8f1979420@mail.gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E84203@EXMBXCLUS01.citservers.local>
	<33c87fa30910222100rae92b90uf94d7ae58a5cdc94@mail.gmail.com>
	<86a32abc0910222215p6eb79f66lbbef2b245f4d17fd@mail.gmail.com>
Message-ID: <227986.79244.qm@web37507.mail.mud.yahoo.com>

I wonder if he run "make current" from his 1.0.4 source file, it should update to the current trunk, wasn't it?

Dorn.



________________________________
From: Diego Viola <diego.viola at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Thu, October 22, 2009 10:15:21 PM
Subject: Re: [Freeswitch-users] Core Dump question!

He's running 1.0.4, he needs to checkout from SVN trunk and build FS from scratch.

Diego


On Fri, Oct 23, 2009 at 4:00 AM, Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:

I think the (exported) means you don't have the latest svn, but probably the officially released build 1.0.4 that can be downloaded from the FS page.
>
>I think you should see something like (the latest trunk is 15203):
>
>freeswitch at internal> version
>
>FreeSWITCH Version 1.0.trunk (15126)
>
>I guess you need to checkout the latest FS trunk
>
>
>
>On Fri, Oct 23, 2009 at 2:25 PM, Ujjval Karihaloo <ujjval at simplesignal.com> wrote:
>
>>>
>>
>>
>>
>>
>>
>>
>>
>>
>>>>
>>>>
>>freeswitch at internal> version
>>FreeSWITCH Version 1.0.4 (exported)
>> 
>>freeswitch at internal>
>>>>
>> 
>>Ujjval Karihaloo
>>VP Voice Engineering
>>IP Phone: +13032428610
>>E-Fax: +17202391690
>> 
>>SimpleSignal Inc.
>>88 Inverness Circle East
>>Suite K105
>>Englewood, CO  80112
>> 
>>>>
>>From:freeswitch-users-bounces at lists.freeswitch.org >>[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Diego
>>Viola
>>Sent: Thursday, October 22, 2009 9:04 PM
>>
>>To: freeswitch-users at lists.freeswitch.org
>>Subject: Re: [Freeswitch-users] Core Dump question!
>>>>
>> 
>>Type "version" on the
>>CLI.
>>>>
>>On Fri, Oct 23, 2009 at 2:52 AM, Ujjval Karihaloo <ujjval at simplesignal.com> wrote:
>>>>
>>>>
>>How do I tell if it?s the latest?I
>>downloaded is yesterday..and installed it from freeswitch.org
>>>>
>> 
>>Ujjval Karihaloo
>>VP Voice Engineering
>>IP Phone: +13032428610
>>E-Fax: +17202391690
>> 
>>SimpleSignal Inc.
>>88 Inverness Circle East
>>Suite K105
>>Englewood, CO  80112
>> 
>>>>
>>From:freeswitch-users-bounces at lists.freeswitch.org >>[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf
>>Of Michael Collins
>>Sent: Thursday, October 22, 2009 6:26 PM
>>>>
>>
>>To: freeswitch-users at lists.freeswitch.org
>>Subject: Re: [Freeswitch-users] Core Dump question!
>> 
>>Yes, if
>>this is latest SVN (after a make current) then open a jira.
>>>>-MC
>>>>
>>>>
>>>>
>>On
>>Thu, Oct 22, 2009 at 5:04 PM, Ujjval Karihaloo <ujjval at simplesignal.com>
>>wrote:
>>>>
>>>>
>>I do have the core dump, should I open a
>>ticket.
>>I am running latest Freeswitch 1.0.4 and
>>had done a make current just before it happened.
>> 
>> 
>>
>>>>_______________________________________________
>>>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>> 
>>_______________________________________________
>>>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>
>_______________________________________________
>>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>



      
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From rob4manhere at gmail.com  Sat Oct 24 10:53:23 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Sat, 24 Oct 2009 12:53:23 -0500
Subject: [Freeswitch-users] Core Dump question!
In-Reply-To: <227986.79244.qm@web37507.mail.mud.yahoo.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701E840EB@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221258o6936000dn6eb2e22189c45435@mail.gmail.com>
	<DF11CCC6-5C89-4489-908C-92A0FCC94AE4@jerris.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E841E8@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221725g6233ec20w49a73864400426d6@mail.gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E84200@EXMBXCLUS01.citservers.local>
	<86a32abc0910222004y46bf08b6jda75b6b8f1979420@mail.gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E84203@EXMBXCLUS01.citservers.local>
	<33c87fa30910222100rae92b90uf94d7ae58a5cdc94@mail.gmail.com>
	<86a32abc0910222215p6eb79f66lbbef2b245f4d17fd@mail.gmail.com>
	<227986.79244.qm@web37507.mail.mud.yahoo.com>
Message-ID: <DCF3C3F0-99EC-4E50-8979-771F0083396F@gmail.com>

No- "make current" won't work because it will try to do an "svn  
update", which won't be applicable since its not a checked out working  
copy of the svn repository.  He'll have to checkout the trunk with:
svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch
from, http://wiki.freeswitch.org/wiki/Installation_Guide

Rob

On Oct 24, 2009, at 12:35 PM, DJB wrote:

> I wonder if he run "make current" from his 1.0.4 source file, it  
> should update to the current trunk, wasn't it?
>
> Dorn.
>
> From: Diego Viola <diego.viola at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Thu, October 22, 2009 10:15:21 PM
> Subject: Re: [Freeswitch-users] Core Dump question!
>
> He's running 1.0.4, he needs to checkout from SVN trunk and build FS  
> from scratch.
>
> Diego
>
> On Fri, Oct 23, 2009 at 4:00 AM, Mark Campbell-Smith <mcampbellsmith at gmail.com 
> > wrote:
> I think the (exported) means you don't have the latest svn, but  
> probably the officially released build 1.0.4 that can be downloaded  
> from the FS page.
>
> I think you should see something like (the latest trunk is 15203):
>
> freeswitch at internal> version
>
> FreeSWITCH Version 1.0.trunk (15126)
>
> I guess you need to checkout the latest FS trunk
>
>
> On Fri, Oct 23, 2009 at 2:25 PM, Ujjval Karihaloo <ujjval at simplesignal.com 
> > wrote:
> freeswitch at internal> version
>
> FreeSWITCH Version 1.0.4 (exported)
>
>
> freeswitch at internal>
>
>
> Ujjval Karihaloo
>
> VP Voice Engineering
>
> IP Phone: +13032428610
>
> E-Fax: +17202391690
>
>
> SimpleSignal Inc.
>
> 88 Inverness Circle East
>
> Suite K105
>
> Englewood, CO  80112
>
> <image001.jpg>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Diego Viola
> Sent: Thursday, October 22, 2009 9:04 PM
>
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Core Dump question!
>
> Type "version" on the CLI.
>
> On Fri, Oct 23, 2009 at 2:52 AM, Ujjval Karihaloo <ujjval at simplesignal.com 
> > wrote:
>
> How do I tell if it?s the latest?I downloaded is yesterday..and  
> installed it from freeswitch.org
>
>
> Ujjval Karihaloo
>
> VP Voice Engineering
>
> IP Phone: +13032428610
>
> E-Fax: +17202391690
>
>
> SimpleSignal Inc.
>
> 88 Inverness Circle East
>
> Suite K105
>
> Englewood, CO  80112
>
> <image001.jpg>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Collins
> Sent: Thursday, October 22, 2009 6:26 PM
>
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Core Dump question!
>
>
> Yes, if this is latest SVN (after a make current) then open a jira.
> -MC
>
> On Thu, Oct 22, 2009 at 5:04 PM, Ujjval Karihaloo <ujjval at simplesignal.com 
> > wrote:
>
> I do have the core dump, should I open a ticket.
>
> I am running latest Freeswitch 1.0.4 and had done a make current  
> just before it happened.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From diego.viola at gmail.com  Sat Oct 24 17:28:19 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 25 Oct 2009 00:28:19 +0000
Subject: [Freeswitch-users] Core Dump question!
In-Reply-To: <DCF3C3F0-99EC-4E50-8979-771F0083396F@gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701E840EB@EXMBXCLUS01.citservers.local>
	<3C04B27FC880044F8FCD735D0D952FF71701E841E8@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221725g6233ec20w49a73864400426d6@mail.gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E84200@EXMBXCLUS01.citservers.local>
	<86a32abc0910222004y46bf08b6jda75b6b8f1979420@mail.gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E84203@EXMBXCLUS01.citservers.local>
	<33c87fa30910222100rae92b90uf94d7ae58a5cdc94@mail.gmail.com>
	<86a32abc0910222215p6eb79f66lbbef2b245f4d17fd@mail.gmail.com>
	<227986.79244.qm@web37507.mail.mud.yahoo.com>
	<DCF3C3F0-99EC-4E50-8979-771F0083396F@gmail.com>
Message-ID: <86a32abc0910241728i23de0605h61d732fba6aeecd8@mail.gmail.com>

What Rob said.

On Sat, Oct 24, 2009 at 5:53 PM, Rob Forman <rob4manhere at gmail.com> wrote:

> No- "make current" won't work because it will try to do an "svn
> update", which won't be applicable since its not a checked out working
> copy of the svn repository.  He'll have to checkout the trunk with:
> svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch
> from, http://wiki.freeswitch.org/wiki/Installation_Guide
>
> Rob
>
> On Oct 24, 2009, at 12:35 PM, DJB wrote:
>
> > I wonder if he run "make current" from his 1.0.4 source file, it
> > should update to the current trunk, wasn't it?
> >
> > Dorn.
> >
> > From: Diego Viola <diego.viola at gmail.com>
> > To: freeswitch-users at lists.freeswitch.org
> > Sent: Thu, October 22, 2009 10:15:21 PM
> > Subject: Re: [Freeswitch-users] Core Dump question!
> >
> > He's running 1.0.4, he needs to checkout from SVN trunk and build FS
> > from scratch.
> >
> > Diego
> >
> > On Fri, Oct 23, 2009 at 4:00 AM, Mark Campbell-Smith <
> mcampbellsmith at gmail.com
> > > wrote:
> > I think the (exported) means you don't have the latest svn, but
> > probably the officially released build 1.0.4 that can be downloaded
> > from the FS page.
> >
> > I think you should see something like (the latest trunk is 15203):
> >
> > freeswitch at internal> version
> >
> > FreeSWITCH Version 1.0.trunk (15126)
> >
> > I guess you need to checkout the latest FS trunk
> >
> >
> > On Fri, Oct 23, 2009 at 2:25 PM, Ujjval Karihaloo <
> ujjval at simplesignal.com
> > > wrote:
> > freeswitch at internal> version
> >
> > FreeSWITCH Version 1.0.4 (exported)
> >
> >
> > freeswitch at internal>
> >
> >
> > Ujjval Karihaloo
> >
> > VP Voice Engineering
> >
> > IP Phone: +13032428610
> >
> > E-Fax: +17202391690
> >
> >
> > SimpleSignal Inc.
> >
> > 88 Inverness Circle East
> >
> > Suite K105
> >
> > Englewood, CO  80112
> >
> > <image001.jpg>
> >
> >
> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org
> > ] On Behalf Of Diego Viola
> > Sent: Thursday, October 22, 2009 9:04 PM
> >
> >
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] Core Dump question!
> >
> > Type "version" on the CLI.
> >
> > On Fri, Oct 23, 2009 at 2:52 AM, Ujjval Karihaloo <
> ujjval at simplesignal.com
> > > wrote:
> >
> > How do I tell if it?s the latest?I downloaded is yesterday..and
> > installed it from freeswitch.org
> >
> >
> > Ujjval Karihaloo
> >
> > VP Voice Engineering
> >
> > IP Phone: +13032428610
> >
> > E-Fax: +17202391690
> >
> >
> > SimpleSignal Inc.
> >
> > 88 Inverness Circle East
> >
> > Suite K105
> >
> > Englewood, CO  80112
> >
> > <image001.jpg>
> >
> >
> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org
> > ] On Behalf Of Michael Collins
> > Sent: Thursday, October 22, 2009 6:26 PM
> >
> >
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] Core Dump question!
> >
> >
> > Yes, if this is latest SVN (after a make current) then open a jira.
> > -MC
> >
> > On Thu, Oct 22, 2009 at 5:04 PM, Ujjval Karihaloo <
> ujjval at simplesignal.com
> > > wrote:
> >
> > I do have the core dump, should I open a ticket.
> >
> > I am running latest Freeswitch 1.0.4 and had done a make current
> > just before it happened.
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From djbinter at yahoo.com  Sat Oct 24 22:55:38 2009
From: djbinter at yahoo.com (DJB)
Date: Sat, 24 Oct 2009 22:55:38 -0700 (PDT)
Subject: [Freeswitch-users] SIP UPDATE Method
Message-ID: <528227.52311.qm@web37508.mail.mud.yahoo.com>

I am wondering why after I update to trunk-15225, the Allow: UPDATE method is no longer there.

User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15225
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE

Am I missing something here?

Thank you.


      


From juanbackson at gmail.com  Sun Oct 25 05:23:37 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Sun, 25 Oct 2009 20:23:37 +0800
Subject: [Freeswitch-users] sip chat question
Message-ID: <27c25bc40910250523g23d4bde6k30d6a88d75391785@mail.gmail.com>

Hi,

I am not able to send chat message from the console and the error I am
getting is a strange "Invalid Profile".

Does anyone know what is wrong?


freeswitch at server.com> chat sip|454047001854840 at 192.168.1.87|
454047001854840 at 192.168.1.87|testing
API CALL [chat(sip|454047001854840 at 192.168.1.87|454047001854840 at 192.168.1.87|testing)]
output:
Error! Message Not Sent
freeswitch at server.com> 2009-10-25 10:51:01.793313 [ERR] sofia_presence.c:93
Chat proto [dp]
from [454047001854840 at 192.168.1.87]
to [454047001854840 at 192.168.1.87]
testing
Invalid Profile 192.168.1.87

freeswitch at server.com>
freeswitch at server.com> sofia status
API CALL [sofia(status)] output:
                     Name          Type
Data     State
=================================================================================================
                 external       profile    sip:mod_sofia at 192.168.1.87:5080
RUNNING (0)
                      internal       profile
sip:mod_sofia at 192.168.1.87:5060     RUNNING (0)
=================================================================================================
2 profiles 0 aliases
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From pippyduck1127 at hotmail.com  Sun Oct 25 11:17:32 2009
From: pippyduck1127 at hotmail.com (Pajongjit Buntaokit)
Date: Sun, 25 Oct 2009 18:17:32 +0000
Subject: [Freeswitch-users] Address Rupa: Database for Audio Data
In-Reply-To: <b39273530910221231u93d3560nbb27036620c46e9b@mail.gmail.com>
References: <BAY119-W52BD912544FAA1DE624B88B3C70@phx.gbl>
	<b39273530910130857jb7dec2eidd71e950d3902b25@mail.gmail.com>
Message-ID: <BAY119-W472B92F30085695612C877B3BB0@phx.gbl>


Hi Rupa,

Thank you very much. It works now. 

----
PB

> Date: Thu, 22 Oct 2009 14:31:58 -0500
> From: rupa at rupa.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Address Rupa: Database for Audio Data
> 
> I use the dialplan app session_record:
> 
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session
> 
> I call that in the appropriate parts before bridging to call.  For
> incoming it is just before the bridge to user/username.  For outgoing
> it is before I bridge to the outgoing provider.
> 
> On Thu, Oct 22, 2009 at 11:30 AM, Pajongjit Buntaokit
> <pippyduck1127 at hotmail.com> wrote:
> > Hi Rupa,
> >
> > Thanks again for your advice.
> >
> > I have been searching for the method to record in the freeswitch
> > documentation but I'm still not sure which command and method to make the
> > record for every call automatically.
> >
> > Which command or method do you use?
> > And to make the recording start and stop automatically every time when the
> > calls is started and end, where should I insert this command?
> >
> > Did you use the Mod commands 'uuid_record'?
> > If so, where to place this commands?
> > Please show me some clues?
> >
> > Thank you very much!
> >
> > PB
> >
> >
> >
> >
> >> Date: Tue, 13 Oct 2009 09:57:05 -0600
> >> From: rupa at rupa.comajongjit Buntaokit
> >> To: freeswitch-users at lists.freeswitch.org
> >> Subject: Re: [Freeswitch-users] Database for Audio Data
> >>
> >> What I do is record all calls and store the call with the UUID as the
> >> filename. Then when the call is hung up a CDR entry is sent to my web
> >> server. This CDR contains callerid and other info I might want to
> >> query by. The service on the web server inserts appropriate record(s)
> >> into the database. The recordings are available to the webserver.
> >> When one clicks on the "listen" link, the web server serves up the
> >> recording by UUID in the recording directory. I have a process that
> >> periodically removes old recordings from that dir. I don't purge the
> >> CDRs, though that is certainly possible.
> >>
> >> On Tue, Oct 13, 2009 at 7:58 AM, Pajongjit Buntaokit
> >> <pippyduck1127 at hotmail.com> wrote:
> >> > Hi,
> >> >
> >> > Does anyone know whether FreeSWITCH has a function to automatically
> >> > record
> >> > every call as an audio file in a server
> >> > or forward them to be stored in a database with additional parameters
> >> > such
> >> > as caller ID, date, starting time and ending time?
> >> >
> >> > So that these recorded audio data can be queried and retrieved with the
> >> > caller ID, date and time.
> >> >
> >> > Any suggestion or guidance, please advise.
> >> >
> >> > Thank you very much!
> >> >
> >> > ________________________________
> >> > Hotmail: Trusted email with Microsoft?s powerful SPAM protection. Sign
> >> > up
> >> > now.
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> -Rupa
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > ________________________________
> > Windows 7: It works the way you want. Learn more.
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> 
> 
> -- 
> -Rupa
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
_________________________________________________________________
Windows 7: Simplify your PC. Learn more.
http://www.microsoft.com/Windows/windows-7/default.aspx?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_evergreen1:102009
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From ujjval at simplesignal.com  Sun Oct 25 13:15:51 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Sun, 25 Oct 2009 13:15:51 -0700
Subject: [Freeswitch-users] Setting up Conference with Moderator
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local>
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701B6DCDD@EXMBXCLUS01.citservers.local>

One more try for help for COnferencing using moderator,

Anyone successfully accomplished this? Help appreciated!
________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ujjval Karihaloo [ujjval at simplesignal.com]
Sent: Thursday, October 22, 2009 9:01 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Setting up Conference with Moderator

Hi

I have the Basic Conferencing working. Here is my Dial Plan.

 I want to be able to setup a Moderator PIN different from other participants, when I add the moderator flag it logs me in directly w/o asking for a PIN..
action application="conference" data="conference.conf at wideband+flags{moderator}+159753<mailto:conference.conf at wideband+flags%7bmoderator%7d+159753>"/>

DialPlan is below for the normal user, and it asks for the PIN with below settings.

       Ujj Inbound from SS - start
    -->
    <extension name="simplesignal">   <!-- your provider or any name you'd like to call it -->
     <condition field="destination_number" expression="xxxxxxxxxx">  <!-- your DID for this gateway-->
        <action application="conference" data="conference.conf at wideband+159753<mailto:conference.conf at wideband+159753>"/>
     </condition>
</extension>

    <!--
       Ujj Inbound from SS ? end




And I am using the existing conference.conf.xml file in the auto_loads directory.


Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690

SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO  80112
[cid:image001.jpg at 01CA535A.E43E9870]<http://www.simplesignal.com/>

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From red.rain.seven at gmail.com  Sun Oct 25 19:58:57 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Mon, 26 Oct 2009 10:58:57 +0800
Subject: [Freeswitch-users] sched_api doesn't get launched
In-Reply-To: <200910231347.43883.chris@cloudtel.com>
References: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
	<59ad9ca10910230154i6d41ddfci8957b152c1884800@mail.gmail.com>
	<191c3a030910230802m1cb847agfdaf6e436296e748@mail.gmail.com>
	<200910231347.43883.chris@cloudtel.com>
Message-ID: <59ad9ca10910251958k314d1145rbb816d508358e52f@mail.gmail.com>

Anthony:

You are da man. Thank you. It's very kind of you to release me from endless
trial :P

On Sat, Oct 24, 2009 at 1:47 AM, Chris Burns <chris at cloudtel.com> wrote:

> poor bbhenry :)
>
> Added in r15207 please test and update docu if necessary:
> http://wiki.freeswitch.org/wiki/Variable_api_on_answer
>
> On October 23, 2009 11:02:07 am Anthony Minessale wrote:
> > it's probably related to escaping the data.
> > I was sick of watching you suffer so i added api_on_answer variable to
> > trunk.
> >
> > On Fri, Oct 23, 2009 at 3:54 AM, Henry Huang
> <red.rain.seven at gmail.com>wrote:
> > > Thanks to c6burns on IRC channel for the tip to use execute_on_answer
> in
> > > combination with eval, and of course everyone here that pointing me to
> > > the right direction.
> > > I was able to execute sched_api with eval, but not with the combination
> > > of execute_on_answer. The argument just don't get parsed as the eval
> > > argument. Here is the code:
> > >
> > > setVariable("execute_on_answer", "eval sched_api(+%ld none
> uuid_displace
> > > <some uuid> start
> > > /usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20
> mux)");
> > >
> > >
> > > Here is the log:
> > > 2009-10-23 16:17:53.633189 [DEBUG] switch_scheduler.c:214 Added task 2
> > > sched_api_function (none) to run at 1256285888
> > > EXECUTE sofia/internal/1688004 at 192.168.1.66 *eval(+OK *Added: 2
> > > )
> > >
> > > 2009-10-23 16:18:04.52701 [DEBUG] switch_channel.c:1927 sofia/internal/
> > > 1688004 at 192.168.1.66 *execute on answer: eval(sched_api(+37 none
> > > uuid_displace 84d74f41-8668-4138-943f-f076e94046ad start
> > > /usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux))*
> > > EXECUTE sofia/internal/1688004 at 192.168.1.66 eval(sched_api(+37 none
> > > uuid_displace 84d74f41-8668-4138-943f-f076e94046ad start
> > > /usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux))
> > >
> > >
> > > The first part of the log is when I just do execute("eval",
> > > "${sched_api(+%ld none uuid_displace <some uuid> start
> > > /usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20
> mux)}");
> > > And I simple get a OK, and then later on the sched_api gets executed.
> > >
> > > However, on the excerpt of the second log. which I use the combination
> > > with execute_on_answer like specified in the yellow hight code above.
> It
> > > doesn't simply return a OK, instead it shows the whole blob of code.
> and
> > > does nothing later.
> > >
> > >
> > >
> > > On Thu, Oct 22, 2009 at 4:09 AM, Anthony Minessale <
> > >
> > > anthony.minessale at gmail.com> wrote:
> > >> Yes you need an API object as described in other email.
> > >>
> > >> Which line of code from java caused that segfault
> > >> It looks like a simple NULL string issue that we may want to hunt
> down.
> > >>
> > >> On Wed, Oct 21, 2009 at 4:44 AM, Henry Huang
> <red.rain.seven at gmail.com>wrote:
> > >>> I can't seem to find the right thing to use in mod_java to execute
> api
> > >>> commands, only api_after_bridge
> > >>>
> > >>> 2009-10-21 17:42:46.593094 [NOTICE] mod_sofia.c:1509 Pre-Answer
> > >>> sofia/internal/1688003 at 192.168.1.66!
> > >>> #
> > >>> # A fatal error has been detected by the Java Runtime Environment:
> > >>> #
> > >>> #  SIGSEGV (0xb) at pc=0x004e4480, pid=1927, tid=16116624
> > >>> #
> > >>> # JRE version: 6.0_16-b01
> > >>> # Java VM: Java HotSpot(TM) Client VM (14.2-b01 mixed mode linux-x86
> )
> > >>> # Problematic frame:
> > >>> # C  [libc.so.6+0x6f480]  strcpy+0x10
> > >>> #
> > >>> # An error report file with more information is saved as:
> > >>> # /usr/local/freeswitch/bin/hs_err_pid1927.log
> > >>> 2009-10-21 17:42:59.883729 [ERR] switch_core_session.c:1374 Invalid
> > >>> Application sched_api
> > >>> 2009-10-21 17:42:59.883729 [NOTICE] switch_core_session.c:1375 Hangup
> > >>> sofia/internal/1688003 at 192.168.1.66 [CS_EXECUTE]
> > >>> [DESTINATION_OUT_OF_ORDER]
> > >>> #
> > >>> # If you would like to submit a bug report, please visit:
> > >>> #   http://java.sun.com/webapps/bugreport/crash.jsp
> > >>> # The crash happened outside the Java Virtual Machine in native code.
> > >>> # See problematic frame for where to report the bug.
> > >>>
> > >>> On Sat, Oct 17, 2009 at 3:37 AM, Michael Collins
> <msc at freeswitch.org>wrote:
> > >>>> On Fri, Oct 16, 2009 at 11:53 AM, Henry Huang
> > >>>> <red.rain.seven at gmail.com
> > >>>>
> > >>>> > wrote:
> > >>>>>
> > >>>>> So how would you trigger it from a script dialplan? The only time
> it
> > >>>>> seemed to work is when I did setVariable("api_after_bridge",
> > >>>>> "sched_api blah blah blah");
> > >>>>> but then it gets executed after the channel's been teared down. I
> > >>>>> thought api_after_bridge means right after the call gets connected.
> > >>>>>
> > >>>>> I need something to execute an api command right before or right
> > >>>>> after the call gets bridged.
> > >>>>>
> > >>>>> api_after_bridge is a channel variable, so using setVariable works
> > >>>>> just
> > >>>>
> > >>>> fine. If you need to sched_api is an API only. Check these out:
> > >>>> http://wiki.freeswitch.org/wiki/Mod_commands#Misc._Commands
> > >>>>
> > >>>> So you need an API object in order to use it. I don't know the
> syntax
> > >>>> for creating an api obj in Java but in Lua it goes like this:
> > >>>> api = freeswitch.API();
> > >>>> res = api:execute("sched_api","+300 none my_api my_api_args")
> > >>>>
> > >>>> Remember, if the method you are using isn't found in the dial plan
> > >>>> tools then it isn't a dial plan application. Make sure it's on the
> > >>>> list: http://wiki.freeswitch.org/wiki/Mod_dptools
> > >>>>
> > >>>> On the other hand, API commands are listed here:
> > >>>> http://wiki.freeswitch.org/wiki/Mod_commands
> > >>>>
> > >>>> dptools require a session object, api commands require an api
> > >>>> object...
> > >>>>
> > >>>> -MC
> > >>>>
> > >>>>
> > >>>> _______________________________________________
> > >>>> FreeSWITCH-users mailing list
> > >>>> FreeSWITCH-users at lists.freeswitch.org
> > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >>>> UNSUBSCRIBE:
> > >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >>>> http://www.freeswitch.org
> > >>>
> > >>> --
> > >>> Henry Huang
> > >>> UniC Solution - Communication Unified
> > >>> VoIP & Open Source software Consultant
> > >>>
> > >>> _______________________________________________
> > >>> FreeSWITCH-users mailing list
> > >>> FreeSWITCH-users at lists.freeswitch.org
> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-user
> > >>>s http://www.freeswitch.org
> > >>
> > >> --
> > >> Anthony Minessale II
> > >>
> > >> FreeSWITCH http://www.freeswitch.org/
> > >> ClueCon http://www.cluecon.com/
> > >> Twitter: http://twitter.com/FreeSWITCH_wire
> > >>
> > >> AIM: anthm
> > >> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com><
> MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> <PAYPAL%3Aanthony.minessa
> > >>le at gmail.com> IRC: irc.freenode.net #freeswitch
> > >>
> > >> FreeSWITCH Developer Conference
> > >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org><
> sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> > >> iax:guest at conference.freeswitch.org/888
> > >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> <googletalk%3Aconf%2B888 at co
> > >>nference.freeswitch.org> pstn:213-799-1400
> > >>
> > >> _______________________________________________
> > >> FreeSWITCH-users mailing list
> > >> FreeSWITCH-users at lists.freeswitch.org
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >> http://www.freeswitch.org
> > >
> > > --
> > > Henry Huang
> > > UniC Solution - Communication Unified
> > > VoIP & Open Source software Consultant
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From ujjval at simplesignal.com  Sun Oct 25 22:05:02 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Sun, 25 Oct 2009 22:05:02 -0700
Subject: [Freeswitch-users] Downloaded tar vs latest SVN - 200 OK has more
	headers
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE2@EXMBXCLUS01.citservers.local>

Hi,

 I used the downlaoded TAR ball and my calls worked, however, when upgrading to the SVN release...my SBC is rejecting the 200 OK (when the FS answers the call - using Conferencing app)..

Here are teh bad and good 200 OK,,,,I see a lot of additional headers startin gwith X:FS , can I remove them?

Bad one
-----------------
   SIP/2.0 200 OK
   v: SIP/2.0/UDP x.x.x.210:5060;branch=z9hG4bK1662379c08490b63b0b9be6978d93ebe
   f: "SIMPLE SIGNAL  "<sip:2222220208 at x.x.x.20:5080;user=phone>;tag=945d1734da594444aac9d071edc798a7
   To: "trunkgroup user" <sip:1111114847 at simplesignal.net;ssig=33402ab16f12e359b1f871b80bf7ff1a>;tag=gX1Z34952QBpS
   i: 674babc2dbf2a90da97bd68642865eae-23f978 at x.x.x.210<mailto:674babc2dbf2a90da97bd68642865eae-23f978 at x.x.x.210>
   CSeq: 3240 INVITE
   Contact: <sip:1111114847 at x.x.x.20:5080;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15225
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 249
   X-FS-Display-Name: +11111114847
   X-FS-Display-Number: +11111114847
   X-FS-Support: update_display
   P-Asserted-Identity: "+11111114847" <+11111114847>
   v=0
   o=FreeSWITCH 1256498762 1256498763 IN IP4 x.x.x.20
   s=FreeSWITCH
   c=IN IP4 x.x.x.20
   t=0 0
   m=audio 29406 RTP/AVP 0 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20
------------------------------
Good One
-------------------------------
   SIP/2.0 200 OK
   v: SIP/2.0/UDP x.x.x.210:5060;branch=z9hG4bKffebde0efa06a6aa7772bae5ca085627
   f: "SIMPLE SIGNAL  "<sip:2222220208 at x.x.x.20:5080;user=phone>;tag=895fd35d4979e6e5b0cdcc795b8c8df2
   To: "trunkgroup user" <sip:1111114847 at simplesignal.net;ssig=33402ab16f12e359b1f871b80bf7ff1a>;tag=KBXKrBQU99c6D
   i: 28f0d47f1163e0d478247e6729762d98-23fedb at x.x.x.210<mailto:28f0d47f1163e0d478247e6729762d98-23fedb at x.x.x.210>
   CSeq: 3278 INVITE
   Contact: <sip:1111114847 at x.x.x.20:5080;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 249
   v=0
   o=FreeSWITCH 1256512215 1256512216 IN IP4 x.x.x.20
   s=FreeSWITCH
   c=IN IP4 x.x.x.20
   t=0 0
   m=audio 17376 RTP/AVP 0 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20

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From velu.technical at gmail.com  Mon Oct 26 00:20:05 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Mon, 26 Oct 2009 12:50:05 +0530
Subject: [Freeswitch-users] meaning of created_time channel variable.
Message-ID: <1452e2980910260020m624a6a1bk2c3a5c36cb1cb5cf@mail.gmail.com>

Dear All,
     What is the value of created_time channel variable? Is this epoch
seconds?

Thanks & Regards,
Velusamy.
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From maciej.aniserowicz at gmail.com  Mon Oct 26 00:21:17 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Mon, 26 Oct 2009 00:21:17 -0700 (PDT)
Subject: [Freeswitch-users] "Can not record session. Media not enabled
 on channel."
In-Reply-To: <87f2f3b90910231200v71e72661id4cc9f2b7fba5610@mail.gmail.com>
References: <1256034659005-3857858.post@n2.nabble.com>
	<87f2f3b90910201746h43a31b32m5c163fd98373f915@mail.gmail.com>
	<1256283391890-3877285.post@n2.nabble.com>
	<87f2f3b90910231200v71e72661id4cc9f2b7fba5610@mail.gmail.com>
Message-ID: <1256541677550-3890610.post@n2.nabble.com>


Yes, I can confirm - this exact error occurs each time when I start recording
before the call is answered (just after sending ORIGINATE command) - but I
think that's completely understandable that media is not ready for an
unanswered call.
But... is there any other event that guarantees media to be ready?




mercutioviz wrote:
> 
> On Fri, Oct 23, 2009 at 12:36 AM, Maciej Aniserowicz <
> maciej.aniserowicz at gmail.com> wrote:
> 
>>
>> The dialplan is very simple:
>>
>>        <extension name="Recording test">
>>                <condition field="destination_number"
>> expression="^111111\d*$">
>>                        <action application="sleep" data="3000" />
>>                        <action application="answer"/>
>>                        <action application="say" data="en name_spelled
>> iterated
>> ${destination_number}"/>
>>                        <action application="sleep" data="1000" />
>>                        <action application="playback"
>> data="local_stream://my_music"/>
>>                </condition>
>>        </extension>
>>
>> Before debugging I have another question: I start recording in event
>> handler
>> for ChannelAnswer event. Is it possible that it's too soon to start
>> recording? Maybe I should start recording in some other event?
>>
> 
> That would be an odd scenario but maybe. It would be best if you could
> catch
> it in the act so that we could see exactly what is happening. The other
> thing you could do is deliberately start recording on the channel prior to
> answering and see if you always get the error. In other words, try to make
> it fail under a certain set of circumstances to see if your theory is
> correct.
> -MC
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3890610.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From tculjaga at gmail.com  Mon Oct 26 03:16:20 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 26 Oct 2009 11:16:20 +0100
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <65d96fc80910240104m458849c0xf7a17d3560496cbf@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910231646210.29582@srv.icf.org.ru>
	<65d96fc80910230614m2b2504d0r92006b78d267a399@mail.gmail.com>
	<65d96fc80910230757h4595f396o2dc1a1bb781d3a8a@mail.gmail.com>
	<Pine.LNX.4.64.0910231903510.17804@srv.icf.org.ru>
	<191c3a030910230837x78ea6093i387617a1adfdf958@mail.gmail.com>
	<Pine.LNX.4.64.0910231943250.22146@srv.icf.org.ru>
	<65d96fc80910230926y6c3e7ee3i9bb2182da1ba8757@mail.gmail.com>
	<Pine.LNX.4.64.0910232208030.29940@srv.icf.org.ru>
	<65d96fc80910240104m458849c0xf7a17d3560496cbf@mail.gmail.com>
Message-ID: <65d96fc80910260316k3f040488wa029504bf972c56e@mail.gmail.com>

>
>
>
>> P.S. people from russian community report what current version of module
>> work fine on fs
>> trunk version.
>>
>>
> that's strange that they report it working as "m_txAudioOpened" is never
> gonna be ready in pre_answer :P... i had to comment it to make it working.
>
> anyhow, i moved everything to trunk and will do some tests on Monday.
>
> T
>


Hello Yuriy, I tried the trunk (FreeSWITCH Version 1.0.trunk (15216M)) and
i'm getting some nice coredumps...


FS crashes when placing outgoing calls.

    coredump on outbound call: FS log and backtrace
http://pastebin.freeswitch.org/10834


FS crashes on incoming calls.

    coredump on inbound call: FS log and backtrace
http://pastebin.freeswitch.org/10835


FS crashes when i try to load mod_h323. I need 3-4 attempts to load it
without a crash.

     coredump on mod_h323 load: FS log and backtrace:
http://pastebin.freeswitch.org/10833


FS crashes on shutdown procedure if mod_h323 was loaded previously.

    coredump on mod_h323 load: FS log and backtrace:
http://pastebin.freeswitch.org/10836



It is quite bad :)
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From helmut.kuper at ewetel.de  Mon Oct 26 04:00:14 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Mon, 26 Oct 2009 12:00:14 +0100
Subject: [Freeswitch-users] Qustion about INFO messages
	after	Connect/Answer
In-Reply-To: <191c3a030910230851i6e0b081ajaf6c294a7f9cd3a8@mail.gmail.com>
References: <4AD859A4.5000101@ewetel.de>
	<4ADC8B6E.7070404@ewetel.de>	<191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>	<87f2f3b90910191435s7528e256r85df5f12fc634f24@mail.gmail.com>	<4ADF483B.4050807@ewetel.de>	<87f2f3b90910211048s208daf1eyfae501091ab75530@mail.gmail.com>	<4AE07DF4.5040803@ewetel.de>	<085B77EE-DA3F-4ADD-86DC-E59D1B6553EA@freeswitch.org>	<191c3a030910221008n60341314hc7a7e6a55a50ef15@mail.gmail.com>	<4AE1951A.1030906@ewetel.de>
	<191c3a030910230851i6e0b081ajaf6c294a7f9cd3a8@mail.gmail.com>
Message-ID: <4AE5813E.2030506@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Anthony,

hm, no not really. There is no change in this behaviour in "FreeSWITCH
Version 1.0.trunk (15225M)"


There is still a caller name change in callee's display.

I'm not sure who is wrong here. Either FS or Snom ...


regards
Helmut


On 23.10.2009 17:51, Anthony Minessale wrote:
> should be even better in 15210
> 
> 
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u9RajDfF0rNdzkaUOVjULmk=
=697e
-----END PGP SIGNATURE-----


From dome at tel.co.th  Mon Oct 26 04:24:28 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Mon, 26 Oct 2009 18:24:28 +0700
Subject: [Freeswitch-users] mod_nibblebill and memory problem
Message-ID: <8ccbff060910260424y7fb9f6eakbb45900838e6fd6d@mail.gmail.com>

Dear All,
             I'm running mod_nibblebill for my prepaid solution. I
still have problem with memory. I have 4 GB RAM and runing debian
squeeze 64 bit and 200 calls concurrent
         Last time nibblebill running with 1 min heartbeat. when i
check memory by htop FS user memory 2% anf growth to 60-89% in 8-9 hr.
and then FS crash.
         Now i change hearbeat to 0 it's mean nibble update balance
when end of call. but everything are same FS start from 2% and growth
to 20% in 2 days.
        When i unload nibblebill FS running fine.
        My question is when concurrent calls drop to 1-2 calls why FS
(I think nibblebill) still use memory ? something wrong in nibblebill
?

BG
Dome C.


From freeswitch.noob at gmail.com  Mon Oct 26 06:16:53 2009
From: freeswitch.noob at gmail.com (freeswitch noob)
Date: Mon, 26 Oct 2009 08:16:53 -0500
Subject: [Freeswitch-users] Hostname
In-Reply-To: <65286EEC1128407F82A019CF0EFB9C38@chaos>
References: <7f783b530910231234u5d0e2fedxea4884d9bcddc9d3@mail.gmail.com>
	<2d9149cd0910231246t58d2bbd3k3466d9c0fca18000@mail.gmail.com>
	<200910231558.09699.chris@cloudtel.com>
	<7f783b530910231342t5393763aueff6bbc679b3c55@mail.gmail.com>
	<65286EEC1128407F82A019CF0EFB9C38@chaos>
Message-ID: <7f783b530910260616u35189aafleba089d1e41f17c@mail.gmail.com>

Thank you for the get idea,  It works perfectly.


On Fri, Oct 23, 2009 at 4:37 PM, Metik <freeswitch-users-list at metik.com>wrote:

>  Why not simply overwrite the value of the variable used throughout the
> script...
>
> -- xml_curl.conf --
> ...
>       <param name="gateway-url" value="
> http://localhost/index.php?xhostname=myhost" bindings="dialplan"/>
> ...
>
> -- index.php --
>
> <?
>
> $_REQUEST['hostname'] = "$xhostname";
>
> ...
>
> ----- Original Message -----
> *From:* freeswitch noob <freeswitch.noob at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Friday, October 23, 2009 4:42 PM
> *Subject:* Re: [Freeswitch-users] Hostname
>
> Yeah, I was just trying to make it easier on myself.  I have scripts from a
> friend that parse xml_curl requests based on the hostname, I was hoping to
> not have to re-write them to read something else from the post that FS makes
> from xml_curl.  But from what it sounds like I will have to.
>
>
> On Fri, Oct 23, 2009 at 2:58 PM, Chris Burns <chris at cloudtel.com> wrote:
>
>> one real quick way would be put different GET var in each server's binding
>>
>> On October 23, 2009 03:46:11 pm Kristian Kielhofner wrote:
>> > Can't you use different contexts or something else to tell them apart?
>> >
>> > On Fri, Oct 23, 2009 at 3:34 PM, freeswitch noob
>> >
>> > <freeswitch.noob at gmail.com> wrote:
>> > > I have mod xml_curl installed and I am getting the following passed to
>> my
>> > > script.
>> > >
>> > > [hostname] => myhost.local
>> > > [section] => dialplan
>> > >
>> > >
>> > > I also have multiple versions of FS running on the same box.  Is there
>> a
>> > > way to have each FS instance on my box have a unique hostname ?
>> > >
>> > > Thanks in advance.
>> > >
>> > > _______________________________________________
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From helmut.kuper at ewetel.de  Mon Oct 26 07:24:14 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Mon, 26 Oct 2009 15:24:14 +0100
Subject: [Freeswitch-users] INFO-Messages: Send_display in pre_answer
 state doesnt work anymore
In-Reply-To: <191c3a030910230733i4fce1c55v892056f549f67ed0@mail.gmail.com>
References: <4AE19D0E.9050301@ewetel.de>
	<191c3a030910230733i4fce1c55v892056f549f67ed0@mail.gmail.com>
Message-ID: <4AE5B10E.7050800@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Anthony,

thx, it works now again. But the only way get it to work is using
send_display after pre_answer. "(sip|originate)_callee_id_name" doesn't
work with playback.

regards
Helmut


On 23.10.2009 16:33, Anthony Minessale wrote:
> There was a concern with letting this happen before answer.
> I changed it in last revision, try it out.
> The problem is that the way polycom does it, it violates the RFC.
> 
> Also know you can set sip_callee_id_name and sip_callee_id_number before
> you send a ring_ready pre_answer or answer to send the display update in
> that packet instead of sending a separate one.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)

iD8DBQFK5bEO4tZeNddg3dwRAkycAKCg61dUy1jkmzauQUDliqYw81aA5ACgmeim
hayGIQk5KlRN8edZGLom1mQ=
=a5QJ
-----END PGP SIGNATURE-----


From brian at freeswitch.org  Mon Oct 26 07:35:07 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 26 Oct 2009 09:35:07 -0500
Subject: [Freeswitch-users] Downloaded tar vs latest SVN - 200 OK has
	more headers
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE2@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE2@EXMBXCLUS01.citservers.local>
Message-ID: <6444A50B-E5F2-4CB5-8AF9-C3D8EBE67BA4@freeswitch.org>

No you can't remove them... And they are 100% valid so your SBC is in  
the wrong.

/b

On Oct 26, 2009, at 12:05 AM, Ujjval Karihaloo wrote:

> Hi,
>
>  I used the downlaoded TAR ball and my calls worked, however, when  
> upgrading to the SVN release...my SBC is rejecting the 200 OK (when  
> the FS answers the call - using Conferencing app)..
>
> Here are teh bad and good 200 OK,,,,I see a lot of additional  
> headers startin gwith X:FS , can I remove them?
>

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From anthony.minessale at gmail.com  Mon Oct 26 07:41:55 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Oct 2009 09:41:55 -0500
Subject: [Freeswitch-users] Qustion about INFO messages after
	Connect/Answer
In-Reply-To: <4AE5813E.2030506@ewetel.de>
References: <4AD859A4.5000101@ewetel.de>
	<87f2f3b90910191435s7528e256r85df5f12fc634f24@mail.gmail.com>
	<4ADF483B.4050807@ewetel.de>
	<87f2f3b90910211048s208daf1eyfae501091ab75530@mail.gmail.com>
	<4AE07DF4.5040803@ewetel.de>
	<085B77EE-DA3F-4ADD-86DC-E59D1B6553EA@freeswitch.org>
	<191c3a030910221008n60341314hc7a7e6a55a50ef15@mail.gmail.com>
	<4AE1951A.1030906@ewetel.de>
	<191c3a030910230851i6e0b081ajaf6c294a7f9cd3a8@mail.gmail.com>
	<4AE5813E.2030506@ewetel.de>
Message-ID: <191c3a030910260741i32578e38qf504a496a4fba475@mail.gmail.com>

Could you maybe consolidate all of your problems into 1 thread.  I am
getting dizzy.  You have 2 on the same subject and you say it works on one
and does not on the other.

Last week we tested all of this with latest trunk and there is no longer any
problems of any sort with the display related stuff.


On Mon, Oct 26, 2009 at 6:00 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi Anthony,
>
> hm, no not really. There is no change in this behaviour in "FreeSWITCH
> Version 1.0.trunk (15225M)"
>
>
> There is still a caller name change in callee's display.
>
> I'm not sure who is wrong here. Either FS or Snom ...
>
>
> regards
> Helmut
>
>
> On 23.10.2009 17:51, Anthony Minessale wrote:
> > should be even better in 15210
> >
> >
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
>
> iD8DBQFK5YE94tZeNddg3dwRAigrAJ0chk/kiDQo2Z6yFbjpJovmPor46ACfValq
> u9RajDfF0rNdzkaUOVjULmk=
> =697e
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From lei.tlfly at gmail.com  Sun Oct 25 06:58:37 2009
From: lei.tlfly at gmail.com (Lei Tang)
Date: Sun, 25 Oct 2009 21:58:37 +0800
Subject: [Freeswitch-users] How to load user account from databse ?
Message-ID: <50c41b4e0910250658n74f92d01v56f1ab82087ad809@mail.gmail.com>

Hi All:
   I'm a newbie to FS.  I'm using  FS as a sbc and have about 20000 user
account . Does somebody can tell me how to  make FS load use account
information from a database such as mssql or mysql?  Could you give me a
sample configuration file?
   Thanks a lots.
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From kristian.kielhofner at gmail.com  Mon Oct 26 07:56:32 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Mon, 26 Oct 2009 10:56:32 -0400
Subject: [Freeswitch-users] Resend: Issues with SIP + TCP?
Message-ID: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>

I originally sent this last Friday but I've been unable to confirm it
ever made it to the list.

Hello everyone,

 I'm having some issues with SIP and TCP.  I've used it before with
success but I'm seeing some strange behavior...

 Level 7 debugs with siptrace on both profiles.  UDP invite from
softphone comes in on port 5062, it's supposed to bridge to
10.70.0.62.  When configured to use UDP FS sends an INVITE (nothing
currently answers) while TCP doesn't send anything (confirmed with
siptrace and packet sniffer).  I confirmed this behavior with a
gateway configured for TCP and appending ;transport=tcp to a bridge
line.

 This is trunk rev 15211 on an Intel Mac running Snow Leopard.  I've
also confirmed this behavior on an Intel Linux machine running Ubuntu
(not sure of version ATM).

TCP:

http://pastebin.freeswitch.org/10825

UDP:

http://pastebin.freeswitch.org/10826

dialplan (UDP):

   <extension name="smhpbx">
     <condition field="destination_number" expression="^(7887)$">
       <action application="set" data="call_timeout=60"/>
       <action application="set" data="effective_caller_id_name=Voalte Test"/>
       <action application="set"
data="effective_caller_id_number=19412848354"/>
       <action application="bridge" data="sofia/avaya/7887 at 10.70.0.62"/>
     </condition>
   </extension>

dialplan (TCP):

   <extension name="smhpbx">
     <condition field="destination_number" expression="^(7887)$">
       <action application="set" data="call_timeout=60"/>
       <action application="set" data="effective_caller_id_name=Voalte Test"/>
       <action application="set"
data="effective_caller_id_number=19412848354"/>
       <action application="bridge"
data="sofia/avaya/7887 at 10.70.0.62;transport=tcp"/>
     </condition>
   </extension>

 Any thoughts?

Thanks!

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From freeswitch.noob at gmail.com  Mon Oct 26 07:59:37 2009
From: freeswitch.noob at gmail.com (freeswitch noob)
Date: Mon, 26 Oct 2009 09:59:37 -0500
Subject: [Freeswitch-users] How to load user account from databse ?
In-Reply-To: <50c41b4e0910250658n74f92d01v56f1ab82087ad809@mail.gmail.com>
References: <50c41b4e0910250658n74f92d01v56f1ab82087ad809@mail.gmail.com>
Message-ID: <7f783b530910260759p6864e582y8a5d6a1db349e22c@mail.gmail.com>

Lei,

I am still learning myself, but I think I can help enough and others can
chime in where I am wrong.

If you have FS up and running you will need to install xml_curl (
http://wiki.freeswitch.org/wiki/Mod_xml_curl).



Then you will need to edit the config for it in your
conf/autoload_configs/xml_curl_conf.xml and add the following to the file.

<binding name="externalDir">
      <param name="gateway-url" value="http://my.webserver.com/index.php"
bindings="directory"/>
    </binding>


Then for every registered user call, a request will be posted to that script
location.

The posts vary but look something like:

[hostname] =>
test.local
[section] =>
directory

[tag_name] =>
domain

[key_name] =>
name

[key_value] =>
company1.test.local
[action] =>
sip_auth

[sip_profile] =>
internal
[sip_user_agent] => eyeBeam release 1100v stamp
47073
[sip_auth_username] =>
100
[sip_auth_realm] =>
company1.test.local
[sip_auth_nonce] =>
37bd639a-d181-4df3-b53d-ab4172ca3be9
[sip_auth_uri] =>
sip:company1.test.local;transport=udp
[sip_contact_user] =>
100
[sip_contact_host] =>
10.43.43.2
[sip_to_user] =>
100
[sip_to_host] =>
company1.test.local
[sip_from_user] =>
100
[sip_from_host] =>
company1.test.local
[sip_request_host] =>
company1.test.local
[sip_auth_qop] =>
auth
[sip_auth_cnonce] =>
207aae18a2af959346f87a2a3c2c7f8a
[sip_auth_nc] =>
00000001
[sip_auth_response] =>
770a1519789ba7606b3dcc6c4b7a99c5
[sip_auth_method] =>
REGISTER
[key] =>
id

[user] =>
100

[domain] =>
company1.test.local

[ip] => 10.43.43.2


Then your script can handle the connection to the DB and the verification of
the user information.  My example uses PHP but any server side language that
can be posted to can be used in the above scenario.


On Sun, Oct 25, 2009 at 8:58 AM, Lei Tang <lei.tlfly at gmail.com> wrote:

> Hi All:
>    I'm a newbie to FS.  I'm using  FS as a sbc and have about 20000 user
> account . Does somebody can tell me how to  make FS load use account
> information from a database such as mssql or mysql?  Could you give me a
> sample configuration file?
>    Thanks a lots.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Mon Oct 26 08:08:59 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Oct 2009 10:08:59 -0500
Subject: [Freeswitch-users] SIP UPDATE Method
In-Reply-To: <528227.52311.qm@web37508.mail.mud.yahoo.com>
References: <528227.52311.qm@web37508.mail.mud.yahoo.com>
Message-ID: <191c3a030910260808y45952bffu7118057caba4ea4@mail.gmail.com>

It was never there before and it caused extreme havoc once we added it so we
took it away again.


On Sun, Oct 25, 2009 at 12:55 AM, DJB <djbinter at yahoo.com> wrote:

> I am wondering why after I update to trunk-15225, the Allow: UPDATE method
> is no longer there.
>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15225
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER,
> NOTIFY, PUBLISH, SUBSCRIBE
>
> Am I missing something here?
>
> Thank you.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From kristian.kielhofner at gmail.com  Mon Oct 26 08:15:54 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Mon, 26 Oct 2009 11:15:54 -0400
Subject: [Freeswitch-users] Downloaded tar vs latest SVN - 200 OK has
	more headers
In-Reply-To: <6444A50B-E5F2-4CB5-8AF9-C3D8EBE67BA4@freeswitch.org>
References: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE2@EXMBXCLUS01.citservers.local>
	<6444A50B-E5F2-4CB5-8AF9-C3D8EBE67BA4@freeswitch.org>
Message-ID: <2d9149cd0910260815s5dc39ec2n869b0bdbfb0333ac@mail.gmail.com>

Brian,

  This bothers me a bit.

  Of course they are "valid".  Anything prefixed with X- should be
ignored by the remote end unless they are specifically looking for it.

  However:

1)  We all know that just because the spec says they are valid doesn't
make it so with every vendor.
2)  As long as these headers are being attached people are going to
keep coming to the list with issues because of them.
3)  Attaching (seemingly arbitrary) headers like this breaks "interop
agreements" that we (the users) have with carriers and other vendors.
Every header we add is another reason why $CARRIER will refuse to
support me because "my INVITE doesn't match what I had during interop
testing".
4)  Why do we /need/ these headers in the first place?  What is using
them?  If we don't really, really, really need them they shouldn't be
there.
5)  Depending on the reason for them being there, this should still be
configurable.  I just don't see why not.  Obviously we could always
just patch the code (Sonus log statements :) but that probably
shouldn't be necessary here.

  I'm not trying to be difficult; I'm just looking for a better
explanation and real discussion.

On Mon, Oct 26, 2009 at 10:35 AM, Brian West <brian at freeswitch.org> wrote:
> No you can't remove them... And they are 100% valid so your SBC is in the
> wrong.
> /b
> On Oct 26, 2009, at 12:05 AM, Ujjval Karihaloo wrote:
>
> Hi,
>
> ?I used the downlaoded TAR ball and my calls worked, however, when upgrading
> to the SVN release...my SBC is rejecting the 200 OK (when the FS answers the
> call - using?Conferencing app)..
>
> Here are teh bad and good 200 OK,,,,I see a lot of additional headers
> startin gwith X:FS , can I remove them?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From shiyanov at gmail.com  Mon Oct 26 08:20:48 2009
From: shiyanov at gmail.com (Artem Shiyanov)
Date: Mon, 26 Oct 2009 18:20:48 +0300
Subject: [Freeswitch-users] Mod_socket: custom caller sip domain in
	originate command
Message-ID: <d211f54a0910260820v5372c65exeba8a3d2bfca5da4@mail.gmail.com>

Hi there!

Please, suggest how to specify custom caller sip domain (logical) in
originate command.
I've been trying several alternatives but no one worked:
1) specify full sip address in
origination_caller_id_number=1000 at uat.pbx.mblagov.starpoundtech.net - FS
adds its IP address so the
result From address would be 1000 at uat.pbx.mblagov.starpoundtech.net@
172.26.200.250
2) specify sip_from_uri varuable like follows:
bgapi originate {origination_caller_id_number=1000,
sip_from_uri=1000 at uat.pbx.mblagov.starpoundtech.net,originate_timeout=10}[origination_uuid=bf8e9778-d410-4504-988d-2d405303183c]sofia/internal/1002%172.26.200.250
&park()
FS doesn't even dial the called endpoint.

Thanks,
Artem
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From ujjval at simplesignal.com  Mon Oct 26 08:21:17 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Mon, 26 Oct 2009 08:21:17 -0700
Subject: [Freeswitch-users] Setting up Conference with Moderator
In-Reply-To: <114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local>
	<3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local>
	<E97CD3D0-52FF-44D1-BB89-23816045CE21@gmail.com>
	<71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>,
	<114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com>
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local>

Thx a lot Rob, reading the wiki your way or using IVR seems correct..
===============
The wiki also says that the wait-mod might be  "used in conjunction 
with an IVR where the moderators are authenticated with an extra pass-
code", which is what I did.  I guess that's why I didn't understand 
the point of the +pin.
======================

I will try it out.

Again thx a lot for your help. Will keep everyone posted.

________________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Forman [rob4manhere at gmail.com]
Sent: Friday, October 23, 2009 12:22 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator

I just re-tested with the pin in my dial plan:

   <action application="conference" data="conference 123456 at default
+flags{}+1234" />

And it doesn't challenge me for the pin.  I just drop right in.  I
figured this is how it was intended, since the wiki says the pin is
set initially and only challenged in later attempts [by future callers]:

"The first time a conference name (confname) is used, it will be
created on demand, and the pin will be set to what ever is specified
at that time: the pin in the data string if specified, or if not, the
"pin" setting in the conference profile, and if that is also
unspecified, then there is no pin protection. Any later attempt to
join the conference must specify the same pin number, if one existed
when it was created. "


The wiki also says that the wait-mod might be  "used in conjunction
with an IVR where the moderators are authenticated with an extra pass-
code", which is what I did.  I guess that's why I didn't understand
the point of the +pin.

I'm sure there's a scenario where its used and useful, the wiki just
doesn't explain it.

Rob

On Oct 23, 2009, at 12:43 PM, Brian West wrote:

> Well first off you're not defining a pine here...
>
> confname at profilename+flags{mute|deaf|waste|moderator}+[conference pin
> number]
>
> That might be why its not asking for a pin.
>
> /b
>
> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:
>
>>      <entry action="menu-exec-app" digits="1" param="conference
>> 123456 at default+flags{}" />
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From anthony.minessale at gmail.com  Mon Oct 26 08:25:09 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Oct 2009 10:25:09 -0500
Subject: [Freeswitch-users] Resend: Issues with SIP + TCP?
In-Reply-To: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>
References: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>
Message-ID: <191c3a030910260825w25b1891g523d5e30d72ee94f@mail.gmail.com>

i cant seem to reproduce it.

originate sofia/internal/1235 at conference.freeswitch.org;transport=tcp 9998

I get a working call and trace.

Could you possibly have a dns error?  I know it's an ip but it may still
fail if it has no dns.

try

sofia loglevel all 9

and look for other errors.




On Mon, Oct 26, 2009 at 9:56 AM, Kristian Kielhofner <
kristian.kielhofner at gmail.com> wrote:

> I originally sent this last Friday but I've been unable to confirm it
> ever made it to the list.
>
> Hello everyone,
>
>  I'm having some issues with SIP and TCP.  I've used it before with
> success but I'm seeing some strange behavior...
>
>  Level 7 debugs with siptrace on both profiles.  UDP invite from
> softphone comes in on port 5062, it's supposed to bridge to
> 10.70.0.62.  When configured to use UDP FS sends an INVITE (nothing
> currently answers) while TCP doesn't send anything (confirmed with
> siptrace and packet sniffer).  I confirmed this behavior with a
> gateway configured for TCP and appending ;transport=tcp to a bridge
> line.
>
>  This is trunk rev 15211 on an Intel Mac running Snow Leopard.  I've
> also confirmed this behavior on an Intel Linux machine running Ubuntu
> (not sure of version ATM).
>
> TCP:
>
> http://pastebin.freeswitch.org/10825
>
> UDP:
>
> http://pastebin.freeswitch.org/10826
>
> dialplan (UDP):
>
>   <extension name="smhpbx">
>     <condition field="destination_number" expression="^(7887)$">
>       <action application="set" data="call_timeout=60"/>
>       <action application="set" data="effective_caller_id_name=Voalte
> Test"/>
>       <action application="set"
> data="effective_caller_id_number=19412848354"/>
>       <action application="bridge" data="sofia/avaya/7887 at 10.70.0.62"/>
>     </condition>
>   </extension>
>
> dialplan (TCP):
>
>   <extension name="smhpbx">
>     <condition field="destination_number" expression="^(7887)$">
>       <action application="set" data="call_timeout=60"/>
>       <action application="set" data="effective_caller_id_name=Voalte
> Test"/>
>       <action application="set"
> data="effective_caller_id_number=19412848354"/>
>       <action application="bridge"
> data="sofia/avaya/7887 at 10.70.0.62;transport=tcp"/>
>     </condition>
>   </extension>
>
>  Any thoughts?
>
> Thanks!
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mayamatakeshi at gmail.com  Mon Oct 26 08:32:51 2009
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Tue, 27 Oct 2009 00:32:51 +0900
Subject: [Freeswitch-users] Mod_socket: custom caller sip domain in
	originate command
In-Reply-To: <d211f54a0910260820v5372c65exeba8a3d2bfca5da4@mail.gmail.com>
References: <d211f54a0910260820v5372c65exeba8a3d2bfca5da4@mail.gmail.com>
Message-ID: <15b9404e0910260832o3a297df5y1dc2093000decf17@mail.gmail.com>

On Tue, Oct 27, 2009 at 12:20 AM, Artem Shiyanov <shiyanov at gmail.com> wrote:

> Hi there!
>
> Please, suggest how to specify custom caller sip domain (logical) in
> originate command.
> I've been trying several alternatives but no one worked:
> 1) specify full sip address in
> origination_caller_id_number=1000 at uat.pbx.mblagov.starpoundtech.net - FS
> adds its IP address so the
> result From address would be 1000 at uat.pbx.mblagov.starpoundtech.net@
> 172.26.200.250
> 2) specify sip_from_uri varuable like follows:
> bgapi originate {origination_caller_id_number=1000,
> sip_from_uri=1000 at uat.pbx.mblagov.starpoundtech.net,originate_timeout=10}[origination_uuid=bf8e9778-d410-4504-988d-2d405303183c]sofia/internal/1002%172.26.200.250
> &park()
> FS doesn't even dial the called endpoint.
>

I use
sip_invite_domain
on bridge. I suppose it works with originate too.
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From delianspam at gmail.com  Mon Oct 26 08:35:30 2009
From: delianspam at gmail.com (Delian Tashev)
Date: Mon, 26 Oct 2009 17:35:30 +0200
Subject: [Freeswitch-users] Reroute a Call Based on the Disconnect Cause
References: <mailman.3342.1253283103.2787.freeswitch-users@lists.freeswitch.org>
Message-ID: <12a101ca5651$f47839e0$dd68ada0$@com>

Hello VoIP Geeks!

I am looking for a way to reroute calls on specific disconnect reasons. My
application is in Python. I will provide my solves so far, which for some
reason I cannot use. Why I need rerouting? Well, I am using the LCR module
which provides multiple routes per destination, like this:
routes=
[lcr_carrier=fs-rocks.com,lcr_rate=0.15000]sofia/default/1174 at 194.135.34.200
:5062|
[lcr_carrier=fs-rocks.com,lcr_rate=0.16000]sofia/default/1174 at 194.135.34.200
:5064

Solve 1: Python solve
Split the routes string to multiple routes. Bridge the call. Check if the
call failed and bridge again.

ivr_reroute_on=(\

    "CALL_REJECTED",\
    "CRASH",\
    "INVALID_NUMBER_FORMAT",\
    "NORMAL_TEMPORARY_FAILURE",\
    "NO_ROUTE_DESTINATION",\
    "NO_USER_RESPONSE",\

    "NUMBER_CHANGED",\

    "RECOVERY_ON_TIMER_EXPIRE",\

    "SERVICE_NOT_IMPLEMENTED",\

)       
...
session.execute("bridge", route)
hangup_cause = str(session.getVariable("originate_disposition"))
if hangup_cause in ivr_reroute_on:
 #bridge again
 ...
else:
 ...
 #do not bridge again

This is tested and works. However my manager does not likes it. I am not
commenting on the side effects of this code (i.e. if you have some
timers/schedules/scheduled events etc).

Solve 2: Freeswitch Source Code solve
Modify the source code of "switch_ivr_originate.c":

                                                if (to &&
!oglobals.continue_on_timeout) {
                                                                goto
outer_for;
                                                }
                                                after line 2523:

                                                +if
(switch_channel_cause2str(*cause) == SWITCH_CAUSE_CALL_REJECTED ) {
                                                +             goto
outer_for;
                                                +}

(In the best case I will add a new channel variable to Freeswitch to control
all the disconnect reasons from here) However my manager again does not
likes this. He says that I am not good enough in C/C++ to use this solution.

Solve 3: continue_on_fail
I have tried using the channel variable "continue_on_fail" which I believe
works only in the dialplan with no luck. I was unable to get it running both
in the dialplan and in Python. By the way I saw that "continue_on_fail" is
indeed implemented elsewhere, while "continue_on_timeout" is exactly in
"switch_ivr_originate.c" as expected.

Solve 4:
So can you please people recommend me any other solution? Any function or
variable to do the trick?

Thank you for your time and help!

Delian Tashev



From anthony.minessale at gmail.com  Mon Oct 26 08:41:11 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Oct 2009 10:41:11 -0500
Subject: [Freeswitch-users] Downloaded tar vs latest SVN - 200 OK has
	more headers
In-Reply-To: <2d9149cd0910260815s5dc39ec2n869b0bdbfb0333ac@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE2@EXMBXCLUS01.citservers.local>
	<6444A50B-E5F2-4CB5-8AF9-C3D8EBE67BA4@freeswitch.org>
	<2d9149cd0910260815s5dc39ec2n869b0bdbfb0333ac@mail.gmail.com>
Message-ID: <191c3a030910260841i6f0f0390mc5bfeab6e5ec7ab6@mail.gmail.com>

The headers are used to pass the callee-id info back to the other side so
you have the id of who you called.
The standards have failed us in this case as everything does it differently
to the point that there is no standard thus we have invented our own way to
carry this across from one FreeSWITCH box to another, but of course we can
never make anybody happy. =/


On Mon, Oct 26, 2009 at 10:15 AM, Kristian Kielhofner <
kristian.kielhofner at gmail.com> wrote:

> Brian,
>
>  This bothers me a bit.
>
>  Of course they are "valid".  Anything prefixed with X- should be
> ignored by the remote end unless they are specifically looking for it.
>
>  However:
>
> 1)  We all know that just because the spec says they are valid doesn't
> make it so with every vendor.
> 2)  As long as these headers are being attached people are going to
> keep coming to the list with issues because of them.
> 3)  Attaching (seemingly arbitrary) headers like this breaks "interop
> agreements" that we (the users) have with carriers and other vendors.
> Every header we add is another reason why $CARRIER will refuse to
> support me because "my INVITE doesn't match what I had during interop
> testing".
> 4)  Why do we /need/ these headers in the first place?  What is using
> them?  If we don't really, really, really need them they shouldn't be
> there.
> 5)  Depending on the reason for them being there, this should still be
> configurable.  I just don't see why not.  Obviously we could always
> just patch the code (Sonus log statements :) but that probably
> shouldn't be necessary here.
>
>  I'm not trying to be difficult; I'm just looking for a better
> explanation and real discussion.
>
> On Mon, Oct 26, 2009 at 10:35 AM, Brian West <brian at freeswitch.org> wrote:
> > No you can't remove them... And they are 100% valid so your SBC is in the
> > wrong.
> > /b
> > On Oct 26, 2009, at 12:05 AM, Ujjval Karihaloo wrote:
> >
> > Hi,
> >
> >  I used the downlaoded TAR ball and my calls worked, however, when
> upgrading
> > to the SVN release...my SBC is rejecting the 200 OK (when the FS answers
> the
> > call - using Conferencing app)..
> >
> > Here are teh bad and good 200 OK,,,,I see a lot of additional headers
> > startin gwith X:FS , can I remove them?
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From ujjval at simplesignal.com  Mon Oct 26 08:39:53 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Mon, 26 Oct 2009 08:39:53 -0700
Subject: [Freeswitch-users] Estimating Call Capacity
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE7@EXMBXCLUS01.citservers.local>

With the following spec for CPU and Memory can someone help me guesstimating how many simultaneous calls and Calls/sec a FS server can handle - Used as a Conferencing Server.

cat /proc/cpuinfo
processor       : 0
vendor_id       : GenuineIntel
cpu family      : 15
model           : 4
model name      : Intel(R) Pentium(R) 4 CPU 2.80GHz
stepping        : 1
cpu MHz         : 2800.386
cache size      : 1024 KB
physical id     : 0
siblings        : 2
core id         : 0
cpu cores       : 1
fdiv_bug        : no
hlt_bug         : no
f00f_bug        : no
coma_bug        : no
fpu             : yes
fpu_exception   : yes
cpuid level     : 5
wp              : yes
flags           : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm constant_tsc pni monitor ds_cpl cid cx16 xtpr
bogomips        : 5604.12

processor       : 1
vendor_id       : GenuineIntel
cpu family      : 15
model           : 4
model name      : Intel(R) Pentium(R) 4 CPU 2.80GHz
stepping        : 1
cpu MHz         : 2800.386
cache size      : 1024 KB
physical id     : 0
siblings        : 2
core id         : 0
cpu cores       : 1
fdiv_bug        : no
hlt_bug         : no
f00f_bug        : no
coma_bug        : no
fpu             : yes
fpu_exception   : yes
cpuid level     : 5
wp              : yes
flags           : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm constant_tsc pni monitor ds_cpl cid cx16 xtpr
bogomips        : 5600.22


# cat /proc/meminfo
MemTotal:      1035080 kB
MemFree:        517972 kB
Buffers:         85812 kB
Cached:         271264 kB
SwapCached:          0 kB
Active:         224292 kB
Inactive:       223008 kB
HighTotal:      130816 kB
HighFree:        29484 kB
LowTotal:       904264 kB
LowFree:        488488 kB
SwapTotal:     2031608 kB
SwapFree:      2031520 kB
Dirty:              80 kB
Writeback:           0 kB
AnonPages:       90172 kB
Mapped:          39880 kB
Slab:            60060 kB
PageTables:       3232 kB
NFS_Unstable:        0 kB
Bounce:              0 kB
CommitLimit:   2549148 kB
Committed_AS:   345780 kB
VmallocTotal:   114680 kB
VmallocUsed:      3584 kB
VmallocChunk:   110888 kB
HugePages_Total:     0
HugePages_Free:      0
HugePages_Rsvd:      0
Hugepagesize:     4096 kB

From kristian.kielhofner at gmail.com  Mon Oct 26 08:47:20 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Mon, 26 Oct 2009 11:47:20 -0400
Subject: [Freeswitch-users] Resend: Issues with SIP + TCP?
In-Reply-To: <191c3a030910260825w25b1891g523d5e30d72ee94f@mail.gmail.com>
References: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>
	<191c3a030910260825w25b1891g523d5e30d72ee94f@mail.gmail.com>
Message-ID: <2d9149cd0910260847i718c420ek99d8d291aa37e143@mail.gmail.com>

Tony,

  It seemed strange to me too (I'm using TCP in other places).

  I'll take another look at this with your suggestions for debugging.

  Thanks!

On Mon, Oct 26, 2009 at 11:25 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> i cant seem to reproduce it.
>
> originate sofia/internal/1235 at conference.freeswitch.org;transport=tcp 9998
>
> I get a working call and trace.
>
> Could you possibly have a dns error?? I know it's an ip but it may still
> fail if it has no dns.
>
> try
>
> sofia loglevel all 9
>
> and look for other errors.
>
>
>
>
> On Mon, Oct 26, 2009 at 9:56 AM, Kristian Kielhofner
> <kristian.kielhofner at gmail.com> wrote:
>>
>> I originally sent this last Friday but I've been unable to confirm it
>> ever made it to the list.
>>
>> Hello everyone,
>>
>> ?I'm having some issues with SIP and TCP. ?I've used it before with
>> success but I'm seeing some strange behavior...
>>
>> ?Level 7 debugs with siptrace on both profiles. ?UDP invite from
>> softphone comes in on port 5062, it's supposed to bridge to
>> 10.70.0.62. ?When configured to use UDP FS sends an INVITE (nothing
>> currently answers) while TCP doesn't send anything (confirmed with
>> siptrace and packet sniffer). ?I confirmed this behavior with a
>> gateway configured for TCP and appending ;transport=tcp to a bridge
>> line.
>>
>> ?This is trunk rev 15211 on an Intel Mac running Snow Leopard. ?I've
>> also confirmed this behavior on an Intel Linux machine running Ubuntu
>> (not sure of version ATM).
>>
>> TCP:
>>
>> http://pastebin.freeswitch.org/10825
>>
>> UDP:
>>
>> http://pastebin.freeswitch.org/10826
>>
>> dialplan (UDP):
>>
>> ? <extension name="smhpbx">
>> ? ? <condition field="destination_number" expression="^(7887)$">
>> ? ? ? <action application="set" data="call_timeout=60"/>
>> ? ? ? <action application="set" data="effective_caller_id_name=Voalte
>> Test"/>
>> ? ? ? <action application="set"
>> data="effective_caller_id_number=19412848354"/>
>> ? ? ? <action application="bridge" data="sofia/avaya/7887 at 10.70.0.62"/>
>> ? ? </condition>
>> ? </extension>
>>
>> dialplan (TCP):
>>
>> ? <extension name="smhpbx">
>> ? ? <condition field="destination_number" expression="^(7887)$">
>> ? ? ? <action application="set" data="call_timeout=60"/>
>> ? ? ? <action application="set" data="effective_caller_id_name=Voalte
>> Test"/>
>> ? ? ? <action application="set"
>> data="effective_caller_id_number=19412848354"/>
>> ? ? ? <action application="bridge"
>> data="sofia/avaya/7887 at 10.70.0.62;transport=tcp"/>
>> ? ? </condition>
>> ? </extension>
>>
>> ?Any thoughts?
>>
>> Thanks!
>>
>> --
>> Kristian Kielhofner
>> http://www.astlinux.org
>> http://blog.krisk.org
>> http://www.star2star.com
>> http://www.submityoursip.com
>> http://www.voalte.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From helmut.kuper at ewetel.de  Mon Oct 26 08:51:15 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Mon, 26 Oct 2009 16:51:15 +0100
Subject: [Freeswitch-users] Qustion about INFO messages
	after	Connect/Answer
In-Reply-To: <191c3a030910260741i32578e38qf504a496a4fba475@mail.gmail.com>
References: <4AD859A4.5000101@ewetel.de>	<87f2f3b90910191435s7528e256r85df5f12fc634f24@mail.gmail.com>	<4ADF483B.4050807@ewetel.de>	<87f2f3b90910211048s208daf1eyfae501091ab75530@mail.gmail.com>	<4AE07DF4.5040803@ewetel.de>	<085B77EE-DA3F-4ADD-86DC-E59D1B6553EA@freeswitch.org>	<191c3a030910221008n60341314hc7a7e6a55a50ef15@mail.gmail.com>	<4AE1951A.1030906@ewetel.de>	<191c3a030910230851i6e0b081ajaf6c294a7f9cd3a8@mail.gmail.com>	<4AE5813E.2030506@ewetel.de>
	<191c3a030910260741i32578e38qf504a496a4fba475@mail.gmail.com>
Message-ID: <4AE5C573.6020601@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Anthony,

sorry for making you dizzy ... bun in fact in my point of view I have
two different problems.

1.
One concerns the way using send_display in pre_answer mode. Simply to
send error texts to caller's display. This works again with latest trunk.


2.
The other one (this thread) concerns the (in my eyes newly introduced)
two INFO messages which FS sends to callee after callee picked up his
phone. The first INFO switches callee's display to a name set in
"originaton_callee_id_name" (a) and immediately after that the second
INFO switches it back to callee's real name (b). You can see this in
display only if (a) and (b) are not the same.

I used Snom 370 phones with FW 8.2.16 as caller and callee. I did an
internal call (sip to sip).


Maybe this is the same code problem, but on my level they are two
different problems, so sorry for confusing you. I hope this clears
things up.


On 26.10.2009 15:41, Anthony Minessale wrote:
> Could you maybe consolidate all of your problems into 1 thread.  I am
> getting dizzy.  You have 2 on the same subject and you say it works on
> one and does not on the other.
> 
> Last week we tested all of this with latest trunk and there is no longer
> any problems of any sort with the display related stuff.

-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)

iD8DBQFK5cVz4tZeNddg3dwRAt2+AJsEQuRTEhE74+XvciTXOC0+kr0tbwCfRzUf
fgVIZwAV/IthjWwvXzRO3TA=
=n7Sr
-----END PGP SIGNATURE-----


From shiyanov at gmail.com  Mon Oct 26 08:56:57 2009
From: shiyanov at gmail.com (Artem Shiyanov)
Date: Mon, 26 Oct 2009 18:56:57 +0300
Subject: [Freeswitch-users] Mod_socket: custom caller sip domain in
	originate command
In-Reply-To: <15b9404e0910260832o3a297df5y1dc2093000decf17@mail.gmail.com>
References: <d211f54a0910260820v5372c65exeba8a3d2bfca5da4@mail.gmail.com>
	<15b9404e0910260832o3a297df5y1dc2093000decf17@mail.gmail.com>
Message-ID: <d211f54a0910260856l53e59888i87704b6ba2ef5310@mail.gmail.com>

Tested- it works!
Thanks a lot!!



On Mon, Oct 26, 2009 at 6:32 PM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:

>
>
> On Tue, Oct 27, 2009 at 12:20 AM, Artem Shiyanov <shiyanov at gmail.com>wrote:
>
>> Hi there!
>>
>> Please, suggest how to specify custom caller sip domain (logical) in
>> originate command.
>> I've been trying several alternatives but no one worked:
>> 1) specify full sip address in
>> origination_caller_id_number=1000 at uat.pbx.mblagov.starpoundtech.net - FS
>> adds its IP address so the
>> result From address would be 1000 at uat.pbx.mblagov.starpoundtech.net@
>> 172.26.200.250
>> 2) specify sip_from_uri varuable like follows:
>> bgapi originate {origination_caller_id_number=1000,
>> sip_from_uri=1000 at uat.pbx.mblagov.starpoundtech.net,originate_timeout=10}[origination_uuid=bf8e9778-d410-4504-988d-2d405303183c]sofia/internal/1002%172.26.200.250
>> &park()
>> FS doesn't even dial the called endpoint.
>>
>
> I use
> sip_invite_domain
> on bridge. I suppose it works with originate too.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kristian.kielhofner at gmail.com  Mon Oct 26 08:59:41 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Mon, 26 Oct 2009 11:59:41 -0400
Subject: [Freeswitch-users] Downloaded tar vs latest SVN - 200 OK has
	more headers
In-Reply-To: <191c3a030910260841i6f0f0390mc5bfeab6e5ec7ab6@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE2@EXMBXCLUS01.citservers.local>
	<6444A50B-E5F2-4CB5-8AF9-C3D8EBE67BA4@freeswitch.org>
	<2d9149cd0910260815s5dc39ec2n869b0bdbfb0333ac@mail.gmail.com>
	<191c3a030910260841i6f0f0390mc5bfeab6e5ec7ab6@mail.gmail.com>
Message-ID: <2d9149cd0910260859v21b70bcnd949ff24170858b0@mail.gmail.com>

Anthony,

  So if I'm understanding you correctly, if you are always using
FreeSWITCH as an edge to other systems you should be able to safely
disable these headers?

On Mon, Oct 26, 2009 at 11:41 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> The headers are used to pass the callee-id info back to the other side so
> you have the id of who you called.
> The standards have failed us in this case as everything does it differently
> to the point that there is no standard thus we have invented our own way to
> carry this across from one FreeSWITCH box to another, but of course we can
> never make anybody happy. =/
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From anthony.minessale at gmail.com  Mon Oct 26 09:04:38 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Oct 2009 11:04:38 -0500
Subject: [Freeswitch-users] Qustion about INFO messages after
	Connect/Answer
In-Reply-To: <4AE5C573.6020601@ewetel.de>
References: <4AD859A4.5000101@ewetel.de>
	<87f2f3b90910211048s208daf1eyfae501091ab75530@mail.gmail.com>
	<4AE07DF4.5040803@ewetel.de>
	<085B77EE-DA3F-4ADD-86DC-E59D1B6553EA@freeswitch.org>
	<191c3a030910221008n60341314hc7a7e6a55a50ef15@mail.gmail.com>
	<4AE1951A.1030906@ewetel.de>
	<191c3a030910230851i6e0b081ajaf6c294a7f9cd3a8@mail.gmail.com>
	<4AE5813E.2030506@ewetel.de>
	<191c3a030910260741i32578e38qf504a496a4fba475@mail.gmail.com>
	<4AE5C573.6020601@ewetel.de>
Message-ID: <191c3a030910260904j275b752fi1135338c2d29ed2b@mail.gmail.com>

depending on your dialplan every time you bridge to a channel it changes the
display to match who you are talking to.  if you tried to set it with the
variable then you call someone that is going to cause this.  Take away the
display app and/or any special variables and let it naturally work.


On Mon, Oct 26, 2009 at 10:51 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi Anthony,
>
> sorry for making you dizzy ... bun in fact in my point of view I have
> two different problems.
>
> 1.
> One concerns the way using send_display in pre_answer mode. Simply to
> send error texts to caller's display. This works again with latest trunk.
>
>
> 2.
> The other one (this thread) concerns the (in my eyes newly introduced)
> two INFO messages which FS sends to callee after callee picked up his
> phone. The first INFO switches callee's display to a name set in
> "originaton_callee_id_name" (a) and immediately after that the second
> INFO switches it back to callee's real name (b). You can see this in
> display only if (a) and (b) are not the same.
>
> I used Snom 370 phones with FW 8.2.16 as caller and callee. I did an
> internal call (sip to sip).
>
>
> Maybe this is the same code problem, but on my level they are two
> different problems, so sorry for confusing you. I hope this clears
> things up.
>
>
> On 26.10.2009 15:41, Anthony Minessale wrote:
> > Could you maybe consolidate all of your problems into 1 thread.  I am
> > getting dizzy.  You have 2 on the same subject and you say it works on
> > one and does not on the other.
> >
> > Last week we tested all of this with latest trunk and there is no longer
> > any problems of any sort with the display related stuff.
>
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
>
> iD8DBQFK5cVz4tZeNddg3dwRAt2+AJsEQuRTEhE74+XvciTXOC0+kr0tbwCfRzUf
> fgVIZwAV/IthjWwvXzRO3TA=
> =n7Sr
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From tculjaga at gmail.com  Mon Oct 26 09:05:15 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 26 Oct 2009 17:05:15 +0100
Subject: [Freeswitch-users] Downloaded tar vs latest SVN - 200 OK has
	more headers
In-Reply-To: <191c3a030910260841i6f0f0390mc5bfeab6e5ec7ab6@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE2@EXMBXCLUS01.citservers.local>
	<6444A50B-E5F2-4CB5-8AF9-C3D8EBE67BA4@freeswitch.org>
	<2d9149cd0910260815s5dc39ec2n869b0bdbfb0333ac@mail.gmail.com>
	<191c3a030910260841i6f0f0390mc5bfeab6e5ec7ab6@mail.gmail.com>
Message-ID: <65d96fc80910260905r3953ba23p346271cdbafe25a@mail.gmail.com>

On Mon, Oct 26, 2009 at 4:41 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> The headers are used to pass the callee-id info back to the other side so
> you have the id of who you called.
> The standards have failed us in this case as everything does it differently
> to the point that there is no standard thus we have invented our own way to
> carry this across from one FreeSWITCH box to another, but of course we can
> never make anybody happy. =/
>
>
I agree with you, X headers should be ignored by the equipment normally.
Anyhow Kristian has a point here; there will be a lot of complains because
of broken SIP stack on many vendor equipments

So, can you consider some customizable a config option for such headers?

T.
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From anthony.minessale at gmail.com  Mon Oct 26 09:16:58 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Oct 2009 11:16:58 -0500
Subject: [Freeswitch-users] Downloaded tar vs latest SVN - 200 OK has
	more headers
In-Reply-To: <65d96fc80910260905r3953ba23p346271cdbafe25a@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE2@EXMBXCLUS01.citservers.local>
	<6444A50B-E5F2-4CB5-8AF9-C3D8EBE67BA4@freeswitch.org>
	<2d9149cd0910260815s5dc39ec2n869b0bdbfb0333ac@mail.gmail.com>
	<191c3a030910260841i6f0f0390mc5bfeab6e5ec7ab6@mail.gmail.com>
	<65d96fc80910260905r3953ba23p346271cdbafe25a@mail.gmail.com>
Message-ID: <191c3a030910260916x47884d07t9abc9b779afbde68@mail.gmail.com>

Thus perpetuating the wild-west of sip where you can't do anything according
to spec because you have to worry about stupid things not keeping up.
Sounds like the education system where I live too.....

I'll see what I can do.  It's always the other end that ppl pay for that
drive the free stuff to change its code.


On Mon, Oct 26, 2009 at 11:05 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:

>
>
> On Mon, Oct 26, 2009 at 4:41 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> The headers are used to pass the callee-id info back to the other side so
>> you have the id of who you called.
>> The standards have failed us in this case as everything does it
>> differently to the point that there is no standard thus we have invented our
>> own way to carry this across from one FreeSWITCH box to another, but of
>> course we can never make anybody happy. =/
>>
>>
> I agree with you, X headers should be ignored by the equipment normally.
> Anyhow Kristian has a point here; there will be a lot of complains because
> of broken SIP stack on many vendor equipments
>
> So, can you consider some customizable a config option for such headers?
>
> T.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From peter.olsson at visionutveckling.se  Mon Oct 26 09:56:01 2009
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 26 Oct 2009 17:56:01 +0100
Subject: [Freeswitch-users] Resend: Issues with SIP + TCP?
In-Reply-To: <2d9149cd0910260847i718c420ek99d8d291aa37e143@mail.gmail.com>
References: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>
	<191c3a030910260825w25b1891g523d5e30d72ee94f@mail.gmail.com>
	<2d9149cd0910260847i718c420ek99d8d291aa37e143@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C550895310C@cooper>

I'm also having problems with this. When running FS compiled about 10 days ago it works fine (don't remember exact revision), but when using latest SVN it doesn't work anymore. I seems like it's trying to use UDP when it should use TCP.

My setup is this: Avaya Communication Manager PBX -> Talks TLS to Avaya SIP SES Server -> Talks TCP to FreeSwitch.

The replies from FS seems to be sent using UDP instead of TCP, and when I keep the config and revert to the 10 day old version it starts working again, so there is definately something wrong.

I'll try to do some more testing, and get back with some SIP-traces as well.

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Kristian Kielhofner
Skickat: den 26 oktober 2009 16:47
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Resend: Issues with SIP + TCP?

Tony,

  It seemed strange to me too (I'm using TCP in other places).

  I'll take another look at this with your suggestions for debugging.

  Thanks!

On Mon, Oct 26, 2009 at 11:25 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> i cant seem to reproduce it.
>
> originate sofia/internal/1235 at conference.freeswitch.org;transport=tcp 9998
>
> I get a working call and trace.
>
> Could you possibly have a dns error?? I know it's an ip but it may still
> fail if it has no dns.
>
> try
>
> sofia loglevel all 9
>
> and look for other errors.
>
>
>
>
> On Mon, Oct 26, 2009 at 9:56 AM, Kristian Kielhofner
> <kristian.kielhofner at gmail.com> wrote:
>>
>> I originally sent this last Friday but I've been unable to confirm it
>> ever made it to the list.
>>
>> Hello everyone,
>>
>> ?I'm having some issues with SIP and TCP. ?I've used it before with
>> success but I'm seeing some strange behavior...
>>
>> ?Level 7 debugs with siptrace on both profiles. ?UDP invite from
>> softphone comes in on port 5062, it's supposed to bridge to
>> 10.70.0.62. ?When configured to use UDP FS sends an INVITE (nothing
>> currently answers) while TCP doesn't send anything (confirmed with
>> siptrace and packet sniffer). ?I confirmed this behavior with a
>> gateway configured for TCP and appending ;transport=tcp to a bridge
>> line.
>>
>> ?This is trunk rev 15211 on an Intel Mac running Snow Leopard. ?I've
>> also confirmed this behavior on an Intel Linux machine running Ubuntu
>> (not sure of version ATM).
>>
>> TCP:
>>
>> http://pastebin.freeswitch.org/10825
>>
>> UDP:
>>
>> http://pastebin.freeswitch.org/10826
>>
>> dialplan (UDP):
>>
>> ? <extension name="smhpbx">
>> ? ? <condition field="destination_number" expression="^(7887)$">
>> ? ? ? <action application="set" data="call_timeout=60"/>
>> ? ? ? <action application="set" data="effective_caller_id_name=Voalte
>> Test"/>
>> ? ? ? <action application="set"
>> data="effective_caller_id_number=19412848354"/>
>> ? ? ? <action application="bridge" data="sofia/avaya/7887 at 10.70.0.62"/>
>> ? ? </condition>
>> ? </extension>
>>
>> dialplan (TCP):
>>
>> ? <extension name="smhpbx">
>> ? ? <condition field="destination_number" expression="^(7887)$">
>> ? ? ? <action application="set" data="call_timeout=60"/>
>> ? ? ? <action application="set" data="effective_caller_id_name=Voalte
>> Test"/>
>> ? ? ? <action application="set"
>> data="effective_caller_id_number=19412848354"/>
>> ? ? ? <action application="bridge"
>> data="sofia/avaya/7887 at 10.70.0.62;transport=tcp"/>
>> ? ? </condition>
>> ? </extension>
>>
>> ?Any thoughts?
>>
>> Thanks!
>>
>> --
>> Kristian Kielhofner
>> http://www.astlinux.org
>> http://blog.krisk.org
>> http://www.star2star.com
>> http://www.submityoursip.com
>> http://www.voalte.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4ae5c6c832935743011996!



From peter.olsson at visionutveckling.se  Mon Oct 26 10:06:47 2009
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 26 Oct 2009 18:06:47 +0100
Subject: [Freeswitch-users] Resend: Issues with SIP + TCP?
In-Reply-To: <2d9149cd0910260847i718c420ek99d8d291aa37e143@mail.gmail.com>
References: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>
	<191c3a030910260825w25b1891g523d5e30d72ee94f@mail.gmail.com>
	<2d9149cd0910260847i718c420ek99d8d291aa37e143@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5508953113@cooper>

Hmm... I remembered incorrectly about my setup :) The Avaya PBX talks TLS to the Avaya SES Server, and then UDP to FS, not TCP - sorry, my bad!

However, something that has changed the last 10 days seems to affect my setup so it doesn't work anymore. I'll do some more SIP tracing, and get back when I know more about it.

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Kristian Kielhofner
Skickat: den 26 oktober 2009 16:47
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Resend: Issues with SIP + TCP?

Tony,

  It seemed strange to me too (I'm using TCP in other places).

  I'll take another look at this with your suggestions for debugging.

  Thanks!

On Mon, Oct 26, 2009 at 11:25 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> i cant seem to reproduce it.
>
> originate sofia/internal/1235 at conference.freeswitch.org;transport=tcp 9998
>
> I get a working call and trace.
>
> Could you possibly have a dns error?? I know it's an ip but it may still
> fail if it has no dns.
>
> try
>
> sofia loglevel all 9
>
> and look for other errors.
>
>
>
>
> On Mon, Oct 26, 2009 at 9:56 AM, Kristian Kielhofner
> <kristian.kielhofner at gmail.com> wrote:
>>
>> I originally sent this last Friday but I've been unable to confirm it
>> ever made it to the list.
>>
>> Hello everyone,
>>
>> ?I'm having some issues with SIP and TCP. ?I've used it before with
>> success but I'm seeing some strange behavior...
>>
>> ?Level 7 debugs with siptrace on both profiles. ?UDP invite from
>> softphone comes in on port 5062, it's supposed to bridge to
>> 10.70.0.62. ?When configured to use UDP FS sends an INVITE (nothing
>> currently answers) while TCP doesn't send anything (confirmed with
>> siptrace and packet sniffer). ?I confirmed this behavior with a
>> gateway configured for TCP and appending ;transport=tcp to a bridge
>> line.
>>
>> ?This is trunk rev 15211 on an Intel Mac running Snow Leopard. ?I've
>> also confirmed this behavior on an Intel Linux machine running Ubuntu
>> (not sure of version ATM).
>>
>> TCP:
>>
>> http://pastebin.freeswitch.org/10825
>>
>> UDP:
>>
>> http://pastebin.freeswitch.org/10826
>>
>> dialplan (UDP):
>>
>> ? <extension name="smhpbx">
>> ? ? <condition field="destination_number" expression="^(7887)$">
>> ? ? ? <action application="set" data="call_timeout=60"/>
>> ? ? ? <action application="set" data="effective_caller_id_name=Voalte
>> Test"/>
>> ? ? ? <action application="set"
>> data="effective_caller_id_number=19412848354"/>
>> ? ? ? <action application="bridge" data="sofia/avaya/7887 at 10.70.0.62"/>
>> ? ? </condition>
>> ? </extension>
>>
>> dialplan (TCP):
>>
>> ? <extension name="smhpbx">
>> ? ? <condition field="destination_number" expression="^(7887)$">
>> ? ? ? <action application="set" data="call_timeout=60"/>
>> ? ? ? <action application="set" data="effective_caller_id_name=Voalte
>> Test"/>
>> ? ? ? <action application="set"
>> data="effective_caller_id_number=19412848354"/>
>> ? ? ? <action application="bridge"
>> data="sofia/avaya/7887 at 10.70.0.62;transport=tcp"/>
>> ? ? </condition>
>> ? </extension>
>>
>> ?Any thoughts?
>>
>> Thanks!
>>
>> --
>> Kristian Kielhofner
>> http://www.astlinux.org
>> http://blog.krisk.org
>> http://www.star2star.com
>> http://www.submityoursip.com
>> http://www.voalte.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4ae5c6c832935743011996!



From brian at freeswitch.org  Mon Oct 26 10:18:48 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 26 Oct 2009 12:18:48 -0500
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE7@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE7@EXMBXCLUS01.citservers.local>
Message-ID: <C49F867E-9993-42EB-9E48-BA442996A5C2@freeswitch.org>

You'll have to do your own load testing.  Nobody can really tell you  
exactly how many you'll get.

/b

On Oct 26, 2009, at 10:39 AM, Ujjval Karihaloo wrote:

> With the following spec for CPU and Memory can someone help me  
> guesstimating how many simultaneous calls and Calls/sec a FS server  
> can handle - Used as a Conferencing Server.



From djbinter at yahoo.com  Mon Oct 26 10:19:50 2009
From: djbinter at yahoo.com (DJB)
Date: Mon, 26 Oct 2009 10:19:50 -0700 (PDT)
Subject: [Freeswitch-users] SIP UPDATE Method
In-Reply-To: <191c3a030910260808y45952bffu7118057caba4ea4@mail.gmail.com>
References: <528227.52311.qm@web37508.mail.mud.yahoo.com>
	<191c3a030910260808y45952bffu7118057caba4ea4@mail.gmail.com>
Message-ID: <33829.77997.qm@web37506.mail.mud.yahoo.com>

I wonder whether you will consider to put it back on the next version 1.0.5 since 1.0.4 has it?  

Regards,
Dorn B.




________________________________
From: Anthony Minessale <anthony.minessale at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Mon, October 26, 2009 8:08:59 AM
Subject: Re: [Freeswitch-users] SIP UPDATE Method

It was never there before and it caused extreme havoc once we added it so we took it away again.



On Sun, Oct 25, 2009 at 12:55 AM, DJB <djbinter at yahoo.com> wrote:

I am wondering why after I update to trunk-15225, the Allow: UPDATE method is no longer there.
>
>>User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15225
>>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>
>>Am I missing something here?
>
>>Thank you.
>
>
>
>
>>_______________________________________________
>>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
pstn:213-799-1400



      
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From brian at freeswitch.org  Mon Oct 26 10:22:42 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 26 Oct 2009 12:22:42 -0500
Subject: [Freeswitch-users] Resend: Issues with SIP + TCP?
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5508953113@cooper>
References: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>
	<191c3a030910260825w25b1891g523d5e30d72ee94f@mail.gmail.com>
	<2d9149cd0910260847i718c420ek99d8d291aa37e143@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953113@cooper>
Message-ID: <DB49781A-1F3A-40CC-B357-6DDE08059FED@freeswitch.org>

Finding the exact rev that broke it would be helpful.

/b

On Oct 26, 2009, at 12:06 PM, Peter Olsson wrote:

> Hmm... I remembered incorrectly about my setup :) The Avaya PBX  
> talks TLS to the Avaya SES Server, and then UDP to FS, not TCP -  
> sorry, my bad!
>
> However, something that has changed the last 10 days seems to affect  
> my setup so it doesn't work anymore. I'll do some more SIP tracing,  
> and get back when I know more about it.
>
> /Peter



From larclap at yahoo.com  Mon Oct 26 10:37:23 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Mon, 26 Oct 2009 10:37:23 -0700
Subject: [Freeswitch-users] Setup advice on small LAN
Message-ID: <005501ca5662$fa594aa0$ef0bdfe0$@com>

I am currently running FreeSwitch very successfully, thanks to the help from
many on this list. I am new to Linux so it was a challenge.

 

FS runs on a small LAN with about 5 other computers. The connection to the
internet is DSL with 3M down and 768kb down via Covad. The ITSP is
Flowroute.

 

If one of the computers does a big download, it messes with FS in two ways.
If a connection is made, the voices are broken up, intermittent and
difficult to understand. If the download is long enough, the connection to
Flowroute is no longer usable due to registration failure.

 

Somehow I need to isolate the FS box from the rest of the LAN, or give its
traffic precedence. Covad's suggestion was to place the FS box in the DMZ.
If I have to, I'll get another DSL line and isolate it that way.

 

Is there a write-up anywhere that might help me with this problem, or
lacking that, can anyone offer advice?

 

581 2009-10-26 07:15:58.307011 [NOTICE] sofia_reg.c:333 Registering
flowroute

582 2009-10-26 07:15:59.275622 [DEBUG] sofia.c:707 nua_i_outbound: unknown
event 8: 101 NAT detected

583 2009-10-26 07:15:59.470887 [DEBUG] sofia_reg.c:1414 Changing expire time
to 2243 by request of proxy sip:sip.flowroute.com

 ...

1087 2009-10-26 07:16:00.623158 [DEBUG] mod_event_socket.c:2302 Socket up
listening on 127.0.0.1:8021

1088 2009-10-26 07:16:01.382171 [DEBUG] sofia.c:707 nua_i_outbound: unknown
event 8: 101 NAT detected

1090 2009-10-26 07:26:34.918724 [DEBUG] sofia_reg.c:1414 Changing expire
time to 1607 by request of proxy sip:sip.flowroute.c

1091 2009-10-26 07:44:56.715662 [DEBUG] sofia_reg.c:1414 Changing expire
time to 505 by request of proxy sip:sip.flowroute.co

1092 2009-10-26 07:49:52.278202 [DEBUG] sofia_reg.c:1414 Changing expire
time to 210 by request of proxy sip:sip.flowroute.co

1093 2009-10-26 07:52:30.468045 [DEBUG] sofia_reg.c:1414 Changing expire
time to 52 by request of proxy sip:sip.flowroute.com

1094 2009-10-26 07:52:49.234006 [DEBUG] sofia_reg.c:1414 Changing expire
time to 33 by request of proxy sip:sip.flowroute.com

1095 2009-10-26 07:53:00.605161 [DEBUG] sofia_reg.c:1414 Changing expire
time to 22 by request of proxy sip:sip.flowroute.com

1096 2009-10-26 07:53:12.379214 [DEBUG] sofia_reg.c:1414 Changing expire
time to 10 by request of proxy sip:sip.flowroute.com

1097 2009-10-26 07:53:19.627029 [DEBUG] sofia_reg.c:1414 Changing expire
time to 3 by request of proxy sip:sip.flowroute.com

1098 2009-10-26 07:53:20.106923 [NOTICE] sofia_reg.c:333 Registering
flowroute

1099 2009-10-26 07:53:20.454870 [DEBUG] sofia_reg.c:1414 Changing expire
time to 2 by request of proxy sip:sip.flowroute.com

1100 2009-10-26 07:53:20.705833 [NOTICE] sofia_reg.c:333 Registering
flowroute

1101 2009-10-26 07:53:20.952781 [DEBUG] sofia_reg.c:1414 Changing expire
time to 1 by request of proxy sip:sip.flowroute.com

1102 2009-10-26 07:53:21.205740 [NOTICE] sofia_reg.c:333 Registering
flowroute

1103 2009-10-26 07:53:21.705656 [NOTICE] sofia_reg.c:333 Registering
flowroute

1104 2009-10-26 07:53:22.002609 [DEBUG] sofia_reg.c:1414 Changing expire
time to 60 by request of proxy sip:sip.flowroute.com

1105 2009-10-26 09:06:17.554726 [ERR] sofia_reg.c:1425 flowroute
Registration Failed with status Operation has no matching challenge  [904].
failure #1

 

Thanks Lars

 

 

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From peter.olsson at visionutveckling.se  Mon Oct 26 10:39:01 2009
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 26 Oct 2009 18:39:01 +0100
Subject: [Freeswitch-users] Resend: Issues with SIP + TCP?
In-Reply-To: <DB49781A-1F3A-40CC-B357-6DDE08059FED@freeswitch.org>
References: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>
	<191c3a030910260825w25b1891g523d5e30d72ee94f@mail.gmail.com>
	<2d9149cd0910260847i718c420ek99d8d291aa37e143@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953113@cooper>
	<DB49781A-1F3A-40CC-B357-6DDE08059FED@freeswitch.org>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5508953116@cooper>

Yes, I know :) However, now I think this is related to the new headers introduced, it's probably not a TCP issue.

Everything seems to work just fine until the 200 OK is sent, the Avaya PBX doesn't seem to accept that reply anymore.

The only differences I've found between a working revision, and a non-working is this;

In the working 200-OK transaction method UPDATE is listed in the Allow-header, and there is a header called "X-Actually-Support: UPDATE".

In the non-working one I don't have these, and instead I have these headers;
   X-FS-Display-Name: 9099
   X-FS-Display-Number: 9099
   X-FS-Support: update_display
   P-Asserted-Identity: "9099" <9099>

So I guess this could be related to the other thread going on right now "Downloaded tar vs latest SVN - 200 OK has more headers", and not a TCP issue. It's probably the Avaya doing something wrong (not the first time), but still it seems these changes affect more systems than mine.

I'm just using it as a lab setup for now, so if anyone want me to test something, I can do it immediately.

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West
Skickat: den 26 oktober 2009 18:23
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Resend: Issues with SIP + TCP?

Finding the exact rev that broke it would be helpful.

/b

On Oct 26, 2009, at 12:06 PM, Peter Olsson wrote:

> Hmm... I remembered incorrectly about my setup :) The Avaya PBX  
> talks TLS to the Avaya SES Server, and then UDP to FS, not TCP -  
> sorry, my bad!
>
> However, something that has changed the last 10 days seems to affect  
> my setup so it doesn't work anymore. I'll do some more SIP tracing,  
> and get back when I know more about it.
>
> /Peter


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4ae5dcba32932131620271!



From brian at freeswitch.org  Mon Oct 26 10:46:24 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 26 Oct 2009 12:46:24 -0500
Subject: [Freeswitch-users] Resend: Issues with SIP + TCP?
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5508953116@cooper>
References: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>
	<191c3a030910260825w25b1891g523d5e30d72ee94f@mail.gmail.com>
	<2d9149cd0910260847i718c420ek99d8d291aa37e143@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953113@cooper>
	<DB49781A-1F3A-40CC-B357-6DDE08059FED@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953116@cooper>
Message-ID: <E5059624-512D-4FF8-8D7D-974FDAD89F04@freeswitch.org>

Bet your hardware just barfs on those like others have... I mean  
really I HATE SIP. This is stupid.

/b

On Oct 26, 2009, at 12:39 PM, Peter Olsson wrote:

> In the non-working one I don't have these, and instead I have these  
> headers;
>   X-FS-Display-Name: 9099
>   X-FS-Display-Number: 9099
>   X-FS-Support: update_display
>   P-Asserted-Identity: "9099" <9099>



From brian at freeswitch.org  Mon Oct 26 10:46:46 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 26 Oct 2009 12:46:46 -0500
Subject: [Freeswitch-users] Setup advice on small LAN
In-Reply-To: <005501ca5662$fa594aa0$ef0bdfe0$@com>
References: <005501ca5662$fa594aa0$ef0bdfe0$@com>
Message-ID: <5CB08703-5F97-451C-93C6-E41AF61A0027@freeswitch.org>

Get a dedicated DSL line.  They aren't that expensive... I have four  
of them at my house!

/b

On Oct 26, 2009, at 12:37 PM, Lars Zeb wrote:

>
> Is there a write-up anywhere that might help me with this problem,  
> or lacking that, can anyone offer advice?

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From rob4manhere at gmail.com  Mon Oct 26 10:52:44 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Mon, 26 Oct 2009 12:52:44 -0500
Subject: [Freeswitch-users] Setup advice on small LAN
In-Reply-To: <005501ca5662$fa594aa0$ef0bdfe0$@com>
References: <005501ca5662$fa594aa0$ef0bdfe0$@com>
Message-ID: <A11DEDF1-BDC0-4E84-93CE-79F00E2E18BE@gmail.com>

Get a dedicated DSL.  That'll work better than any sort of traffic  
prioritization or shaping (I've tried).

Depending on your average channel use and codec, you could probably go  
with the smallest package and be fine.

Rob

On Oct 26, 2009, at 12:37 PM, Lars Zeb wrote:

> I am currently running FreeSwitch very successfully, thanks to the  
> help from many on this list. I am new to Linux so it was a challenge.
>
> FS runs on a small LAN with about 5 other computers. The connection  
> to the internet is DSL with 3M down and 768kb down via Covad. The  
> ITSP is Flowroute.
>
> If one of the computers does a big download, it messes with FS in  
> two ways. If a connection is made, the voices are broken up,  
> intermittent and difficult to understand. If the download is long  
> enough, the connection to Flowroute is no longer usable due to  
> registration failure.
>
> Somehow I need to isolate the FS box from the rest of the LAN, or  
> give its traffic precedence. Covad?s suggestion was to place the FS  
> box in the DMZ. If I have to, I?ll get another DSL line and isolate  
> it that way.
>
> Is there a write-up anywhere that might help me with this problem,  
> or lacking that, can anyone offer advice?
>
> 581 2009-10-26 07:15:58.307011 [NOTICE] sofia_reg.c:333 Registering  
> flowroute
> 582 2009-10-26 07:15:59.275622 [DEBUG] sofia.c:707 nua_i_outbound:  
> unknown event 8: 101 NAT detected
> 583 2009-10-26 07:15:59.470887 [DEBUG] sofia_reg.c:1414 Changing  
> expire time to 2243 by request of proxy sip:sip.flowroute.com
>  ...
> 1087 2009-10-26 07:16:00.623158 [DEBUG] mod_event_socket.c:2302  
> Socket up listening on 127.0.0.1:8021
> 1088 2009-10-26 07:16:01.382171 [DEBUG] sofia.c:707 nua_i_outbound:  
> unknown event 8: 101 NAT detected
> 1090 2009-10-26 07:26:34.918724 [DEBUG] sofia_reg.c:1414 Changing  
> expire time to 1607 by request of proxy sip:sip.flowroute.c
> 1091 2009-10-26 07:44:56.715662 [DEBUG] sofia_reg.c:1414 Changing  
> expire time to 505 by request of proxy sip:sip.flowroute.co
> 1092 2009-10-26 07:49:52.278202 [DEBUG] sofia_reg.c:1414 Changing  
> expire time to 210 by request of proxy sip:sip.flowroute.co
> 1093 2009-10-26 07:52:30.468045 [DEBUG] sofia_reg.c:1414 Changing  
> expire time to 52 by request of proxy sip:sip.flowroute.com
> 1094 2009-10-26 07:52:49.234006 [DEBUG] sofia_reg.c:1414 Changing  
> expire time to 33 by request of proxy sip:sip.flowroute.com
> 1095 2009-10-26 07:53:00.605161 [DEBUG] sofia_reg.c:1414 Changing  
> expire time to 22 by request of proxy sip:sip.flowroute.com
> 1096 2009-10-26 07:53:12.379214 [DEBUG] sofia_reg.c:1414 Changing  
> expire time to 10 by request of proxy sip:sip.flowroute.com
> 1097 2009-10-26 07:53:19.627029 [DEBUG] sofia_reg.c:1414 Changing  
> expire time to 3 by request of proxy sip:sip.flowroute.com
> 1098 2009-10-26 07:53:20.106923 [NOTICE] sofia_reg.c:333 Registering  
> flowroute
> 1099 2009-10-26 07:53:20.454870 [DEBUG] sofia_reg.c:1414 Changing  
> expire time to 2 by request of proxy sip:sip.flowroute.com
> 1100 2009-10-26 07:53:20.705833 [NOTICE] sofia_reg.c:333 Registering  
> flowroute
> 1101 2009-10-26 07:53:20.952781 [DEBUG] sofia_reg.c:1414 Changing  
> expire time to 1 by request of proxy sip:sip.flowroute.com
> 1102 2009-10-26 07:53:21.205740 [NOTICE] sofia_reg.c:333 Registering  
> flowroute
> 1103 2009-10-26 07:53:21.705656 [NOTICE] sofia_reg.c:333 Registering  
> flowroute
> 1104 2009-10-26 07:53:22.002609 [DEBUG] sofia_reg.c:1414 Changing  
> expire time to 60 by request of proxy sip:sip.flowroute.com
> 1105 2009-10-26 09:06:17.554726 [ERR] sofia_reg.c:1425 flowroute  
> Registration Failed with status Operation has no matching challenge   
> [904]. failure #1
>
> Thanks Lars
>
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus  
> signature database 4545 (20091026) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From william.suffill at gmail.com  Mon Oct 26 10:57:25 2009
From: william.suffill at gmail.com (William Suffill)
Date: Mon, 26 Oct 2009 13:57:25 -0400
Subject: [Freeswitch-users] Setup advice on small LAN
In-Reply-To: <5CB08703-5F97-451C-93C6-E41AF61A0027@freeswitch.org>
References: <005501ca5662$fa594aa0$ef0bdfe0$@com>
	<5CB08703-5F97-451C-93C6-E41AF61A0027@freeswitch.org>
Message-ID: <6b65470d0910261057k21655368w13b958d31345be3d@mail.gmail.com>

4 sounds a bit overkill. Hope you got a good deal. But yes isolating
it from general internet traffic is ideal. There are other methods to
reach that goal but they get rather complex once you realize all the
things that can effect calls. =/

-- W


From shouldbeq931 at googlemail.com  Mon Oct 26 10:56:50 2009
From: shouldbeq931 at googlemail.com (shouldbe q931)
Date: Mon, 26 Oct 2009 17:56:50 +0000
Subject: [Freeswitch-users] Setup advice on small LAN
In-Reply-To: <5CB08703-5F97-451C-93C6-E41AF61A0027@freeswitch.org>
References: <005501ca5662$fa594aa0$ef0bdfe0$@com>
	<5CB08703-5F97-451C-93C6-E41AF61A0027@freeswitch.org>
Message-ID: <649eaa470910261056j7ead284fq3290d5b3ea08fc7b@mail.gmail.com>

I agree, it might be a "dirty" solution, but its so much easier than
trying to get QoS running on a DSL line, or learning how to run a
traffic classifier...

On Mon, Oct 26, 2009 at 5:46 PM, Brian West <brian at freeswitch.org> wrote:
> Get a dedicated DSL line. ?They aren't that expensive... I have four of them
> at my house!
> /b
> On Oct 26, 2009, at 12:37 PM, Lars Zeb wrote:
>
>
> Is there a write-up anywhere that might help me with this problem, or
> lacking that, can anyone offer advice?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From anthony.minessale at gmail.com  Mon Oct 26 11:24:36 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Oct 2009 13:24:36 -0500
Subject: [Freeswitch-users] Resend: Issues with SIP + TCP?
In-Reply-To: <E5059624-512D-4FF8-8D7D-974FDAD89F04@freeswitch.org>
References: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>
	<191c3a030910260825w25b1891g523d5e30d72ee94f@mail.gmail.com>
	<2d9149cd0910260847i718c420ek99d8d291aa37e143@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953113@cooper>
	<DB49781A-1F3A-40CC-B357-6DDE08059FED@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953116@cooper>
	<E5059624-512D-4FF8-8D7D-974FDAD89F04@freeswitch.org>
Message-ID: <191c3a030910261124g188d6bdene89c54cbe65a0648@mail.gmail.com>

try r15230

add the profile param

<param name="pass-callee-id" value="false"/>


On Mon, Oct 26, 2009 at 12:46 PM, Brian West <brian at freeswitch.org> wrote:

> Bet your hardware just barfs on those like others have... I mean
> really I HATE SIP. This is stupid.
>
> /b
>
> On Oct 26, 2009, at 12:39 PM, Peter Olsson wrote:
>
> > In the non-working one I don't have these, and instead I have these
> > headers;
> >   X-FS-Display-Name: 9099
> >   X-FS-Display-Number: 9099
> >   X-FS-Support: update_display
> >   P-Asserted-Identity: "9099" <9099>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Mon Oct 26 11:26:48 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Oct 2009 13:26:48 -0500
Subject: [Freeswitch-users] Downloaded tar vs latest SVN - 200 OK has
	more headers
In-Reply-To: <191c3a030910260916x47884d07t9abc9b779afbde68@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE2@EXMBXCLUS01.citservers.local>
	<6444A50B-E5F2-4CB5-8AF9-C3D8EBE67BA4@freeswitch.org>
	<2d9149cd0910260815s5dc39ec2n869b0bdbfb0333ac@mail.gmail.com>
	<191c3a030910260841i6f0f0390mc5bfeab6e5ec7ab6@mail.gmail.com>
	<65d96fc80910260905r3953ba23p346271cdbafe25a@mail.gmail.com>
	<191c3a030910260916x47884d07t9abc9b779afbde68@mail.gmail.com>
Message-ID: <191c3a030910261126l7cf12dd1n369acb79f4fc35a4@mail.gmail.com>

This is ridiculous but here it is


try r15230

add the profile param

<param name="pass-callee-id" value="false"/>



On Mon, Oct 26, 2009 at 11:16 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Thus perpetuating the wild-west of sip where you can't do anything
> according to spec because you have to worry about stupid things not keeping
> up.  Sounds like the education system where I live too.....
>
> I'll see what I can do.  It's always the other end that ppl pay for that
> drive the free stuff to change its code.
>
>
> On Mon, Oct 26, 2009 at 11:05 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>>
>>
>> On Mon, Oct 26, 2009 at 4:41 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> The headers are used to pass the callee-id info back to the other side so
>>> you have the id of who you called.
>>> The standards have failed us in this case as everything does it
>>> differently to the point that there is no standard thus we have invented our
>>> own way to carry this across from one FreeSWITCH box to another, but of
>>> course we can never make anybody happy. =/
>>>
>>>
>> I agree with you, X headers should be ignored by the equipment normally.
>> Anyhow Kristian has a point here; there will be a lot of complains because
>> of broken SIP stack on many vendor equipments
>>
>> So, can you consider some customizable a config option for such headers?
>>
>> T.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From peter.olsson at visionutveckling.se  Mon Oct 26 11:27:24 2009
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 26 Oct 2009 19:27:24 +0100
Subject: [Freeswitch-users] Resend: Issues with SIP + TCP?
In-Reply-To: <E5059624-512D-4FF8-8D7D-974FDAD89F04@freeswitch.org>
References: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>
	<191c3a030910260825w25b1891g523d5e30d72ee94f@mail.gmail.com>
	<2d9149cd0910260847i718c420ek99d8d291aa37e143@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953113@cooper>
	<DB49781A-1F3A-40CC-B357-6DDE08059FED@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953116@cooper>
	<E5059624-512D-4FF8-8D7D-974FDAD89F04@freeswitch.org>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C550895311E@cooper>

I understand your frustration :) We deal with SIP integration with about 10 different PBX vendors today, And it's always something that doesn't work as it should. Right now I don't have anything more connected to FS though.

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West
Skickat: den 26 oktober 2009 18:46
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Resend: Issues with SIP + TCP?

Bet your hardware just barfs on those like others have... I mean  
really I HATE SIP. This is stupid.

/b

On Oct 26, 2009, at 12:39 PM, Peter Olsson wrote:

> In the non-working one I don't have these, and instead I have these  
> headers;
>   X-FS-Display-Name: 9099
>   X-FS-Display-Number: 9099
>   X-FS-Support: update_display
>   P-Asserted-Identity: "9099" <9099>


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4ae5e23832938073513968!



From mike at jerris.com  Mon Oct 26 11:29:13 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 26 Oct 2009 14:29:13 -0400
Subject: [Freeswitch-users] Downloaded tar vs latest SVN - 200 OK has
	more headers
In-Reply-To: <191c3a030910260916x47884d07t9abc9b779afbde68@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE2@EXMBXCLUS01.citservers.local>
	<6444A50B-E5F2-4CB5-8AF9-C3D8EBE67BA4@freeswitch.org>
	<2d9149cd0910260815s5dc39ec2n869b0bdbfb0333ac@mail.gmail.com>
	<191c3a030910260841i6f0f0390mc5bfeab6e5ec7ab6@mail.gmail.com>
	<65d96fc80910260905r3953ba23p346271cdbafe25a@mail.gmail.com>
	<191c3a030910260916x47884d07t9abc9b779afbde68@mail.gmail.com>
Message-ID: <5EA77040-5701-4AC8-99A3-DC8AB3FBBDBB@jerris.com>

New sofia profile param as follows:

     <!-- set this param to false if your gateway for some reason  
hates X- headers that is is supposed to ignore-->
     <!--<param name="pass-callee-id" value="false"/>-->


On Oct 26, 2009, at 12:16 PM, Anthony Minessale wrote:

> Thus perpetuating the wild-west of sip where you can't do anything  
> according to spec because you have to worry about stupid things not  
> keeping up.  Sounds like the education system where I live too.....
>
> I'll see what I can do.  It's always the other end that ppl pay for  
> that drive the free stuff to change its code.
>
>
> On Mon, Oct 26, 2009 at 11:05 AM, Tihomir Culjaga  
> <tculjaga at gmail.com> wrote:
>
>
> On Mon, Oct 26, 2009 at 4:41 PM, Anthony Minessale <anthony.minessale at gmail.com 
> > wrote:
> The headers are used to pass the callee-id info back to the other  
> side so you have the id of who you called.
> The standards have failed us in this case as everything does it  
> differently to the point that there is no standard thus we have  
> invented our own way to carry this across from one FreeSWITCH box to  
> another, but of course we can never make anybody happy. =/
>
>
> I agree with you, X headers should be ignored by the equipment  
> normally. Anyhow Kristian has a point here; there will be a lot of  
> complains because of broken SIP stack on many vendor equipments
>
> So, can you consider some customizable a config option for such  
> headers?
>
> T.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From brian at freeswitch.org  Mon Oct 26 11:40:27 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 26 Oct 2009 13:40:27 -0500
Subject: [Freeswitch-users] Resend: Issues with SIP + TCP?
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C550895311E@cooper>
References: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>
	<191c3a030910260825w25b1891g523d5e30d72ee94f@mail.gmail.com>
	<2d9149cd0910260847i718c420ek99d8d291aa37e143@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953113@cooper>
	<DB49781A-1F3A-40CC-B357-6DDE08059FED@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953116@cooper>
	<E5059624-512D-4FF8-8D7D-974FDAD89F04@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C550895311E@cooper>
Message-ID: <2C225D94-CD78-4683-A886-F767566033BF@freeswitch.org>

At some point we'll have to NO NO NO fix your broken crap.  :P  The  
reason we have sip so broken now is NOBODY CAN SAY NO!... Say it with  
me... "NO!"

/b

On Oct 26, 2009, at 1:27 PM, Peter Olsson wrote:

> I understand your frustration :) We deal with SIP integration with  
> about 10 different PBX vendors today, And it's always something that  
> doesn't work as it should. Right now I don't have anything more  
> connected to FS though.
>
> /Peter



From anthony.minessale at gmail.com  Mon Oct 26 11:45:31 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Oct 2009 13:45:31 -0500
Subject: [Freeswitch-users] SIP UPDATE Method
In-Reply-To: <33829.77997.qm@web37506.mail.mud.yahoo.com>
References: <528227.52311.qm@web37508.mail.mud.yahoo.com>
	<191c3a030910260808y45952bffu7118057caba4ea4@mail.gmail.com>
	<33829.77997.qm@web37506.mail.mud.yahoo.com>
Message-ID: <191c3a030910261145q46128a92w75d73158a7f73572@mail.gmail.com>

what exactly are you expecting to use it for?
We never really supported it anyway.


On Mon, Oct 26, 2009 at 12:19 PM, DJB <djbinter at yahoo.com> wrote:

> I wonder whether you will consider to put it back on the next version 1.0.5
> since 1.0.4 has it?
>
> Regards,
> Dorn B.
>
>
> ------------------------------
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Mon, October 26, 2009 8:08:59 AM
> *Subject:* Re: [Freeswitch-users] SIP UPDATE Method
>
> It was never there before and it caused extreme havoc once we added it so
> we took it away again.
>
>
> On Sun, Oct 25, 2009 at 12:55 AM, DJB <djbinter at yahoo.com> wrote:
>
>> I am wondering why after I update to trunk-15225, the Allow: UPDATE method
>> is no longer there.
>>
>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15225
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER,
>> NOTIFY, PUBLISH, SUBSCRIBE
>>
>> Am I missing something here?
>>
>> Thank you.
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From dwhite at olp.net  Mon Oct 26 11:48:04 2009
From: dwhite at olp.net (Dan White)
Date: Mon, 26 Oct 2009 13:48:04 -0500
Subject: [Freeswitch-users] Setup advice on small LAN
In-Reply-To: <005501ca5662$fa594aa0$ef0bdfe0$@com>
References: <005501ca5662$fa594aa0$ef0bdfe0$@com>
Message-ID: <20091026184804.GA5031@dan.olp.net>

On 26/10/09?10:37?-0700, Lars Zeb wrote:
>If one of the computers does a big download, it messes with FS in two ways.
>If a connection is made, the voices are broken up, intermittent and
>difficult to understand. If the download is long enough, the connection to
>Flowroute is no longer usable due to registration failure.

QoS is a two way street. You could spend a lot of time getting your egress
traffic properly prioritized, but if your ISP does not do the same
prioritization on your ingress traffic (toward you), you'll still have
problems during downloads.

If your friendly neighborhood ISP will work with you on prioritization,
that's another matter.

-- 
Dan White


From freeswitch-users-list at metik.com  Mon Oct 26 12:25:31 2009
From: freeswitch-users-list at metik.com (Metik)
Date: Mon, 26 Oct 2009 15:25:31 -0400
Subject: [Freeswitch-users] Setup advice on small LAN
In-Reply-To: <005501ca5662$fa594aa0$ef0bdfe0$@com>
References: <005501ca5662$fa594aa0$ef0bdfe0$@com>
Message-ID: <A52116049AB24FAFB9A2E1B9A0B069D4@chaos>

Lars,

If your ISP is a COVAD partner, ask them to reprovision the line for VOA (voice optimized access).  They will provide you with two seperate VLANs (one for best effort data and the other for real-time/voice traffic).  If they are unable to do so or do not understand your request, feel free to email my off list and I can help you. The cost difference should be minimal unless you move to "second line" (a/k/a "naked", "unbundled", or "dedicated") DSL.

-metik
  ----- Original Message ----- 
  From: Lars Zeb 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, October 26, 2009 1:37 PM
  Subject: [Freeswitch-users] Setup advice on small LAN


  I am currently running FreeSwitch very successfully, thanks to the help from many on this list. I am new to Linux so it was a challenge.

   

  FS runs on a small LAN with about 5 other computers. The connection to the internet is DSL with 3M down and 768kb down via Covad. The ITSP is Flowroute.

   

  If one of the computers does a big download, it messes with FS in two ways. If a connection is made, the voices are broken up, intermittent and difficult to understand. If the download is long enough, the connection to Flowroute is no longer usable due to registration failure.

   

  Somehow I need to isolate the FS box from the rest of the LAN, or give its traffic precedence. Covad's suggestion was to place the FS box in the DMZ. If I have to, I'll get another DSL line and isolate it that way.

   

  Is there a write-up anywhere that might help me with this problem, or lacking that, can anyone offer advice?

   

  581 2009-10-26 07:15:58.307011 [NOTICE] sofia_reg.c:333 Registering flowroute

  582 2009-10-26 07:15:59.275622 [DEBUG] sofia.c:707 nua_i_outbound: unknown event 8: 101 NAT detected

  583 2009-10-26 07:15:59.470887 [DEBUG] sofia_reg.c:1414 Changing expire time to 2243 by request of proxy sip:sip.flowroute.com

   ...

  1087 2009-10-26 07:16:00.623158 [DEBUG] mod_event_socket.c:2302 Socket up listening on 127.0.0.1:8021

  1088 2009-10-26 07:16:01.382171 [DEBUG] sofia.c:707 nua_i_outbound: unknown event 8: 101 NAT detected

  1090 2009-10-26 07:26:34.918724 [DEBUG] sofia_reg.c:1414 Changing expire time to 1607 by request of proxy sip:sip.flowroute.c

  1091 2009-10-26 07:44:56.715662 [DEBUG] sofia_reg.c:1414 Changing expire time to 505 by request of proxy sip:sip.flowroute.co

  1092 2009-10-26 07:49:52.278202 [DEBUG] sofia_reg.c:1414 Changing expire time to 210 by request of proxy sip:sip.flowroute.co

  1093 2009-10-26 07:52:30.468045 [DEBUG] sofia_reg.c:1414 Changing expire time to 52 by request of proxy sip:sip.flowroute.com

  1094 2009-10-26 07:52:49.234006 [DEBUG] sofia_reg.c:1414 Changing expire time to 33 by request of proxy sip:sip.flowroute.com

  1095 2009-10-26 07:53:00.605161 [DEBUG] sofia_reg.c:1414 Changing expire time to 22 by request of proxy sip:sip.flowroute.com

  1096 2009-10-26 07:53:12.379214 [DEBUG] sofia_reg.c:1414 Changing expire time to 10 by request of proxy sip:sip.flowroute.com

  1097 2009-10-26 07:53:19.627029 [DEBUG] sofia_reg.c:1414 Changing expire time to 3 by request of proxy sip:sip.flowroute.com

  1098 2009-10-26 07:53:20.106923 [NOTICE] sofia_reg.c:333 Registering flowroute

  1099 2009-10-26 07:53:20.454870 [DEBUG] sofia_reg.c:1414 Changing expire time to 2 by request of proxy sip:sip.flowroute.com

  1100 2009-10-26 07:53:20.705833 [NOTICE] sofia_reg.c:333 Registering flowroute

  1101 2009-10-26 07:53:20.952781 [DEBUG] sofia_reg.c:1414 Changing expire time to 1 by request of proxy sip:sip.flowroute.com

  1102 2009-10-26 07:53:21.205740 [NOTICE] sofia_reg.c:333 Registering flowroute

  1103 2009-10-26 07:53:21.705656 [NOTICE] sofia_reg.c:333 Registering flowroute

  1104 2009-10-26 07:53:22.002609 [DEBUG] sofia_reg.c:1414 Changing expire time to 60 by request of proxy sip:sip.flowroute.com

  1105 2009-10-26 09:06:17.554726 [ERR] sofia_reg.c:1425 flowroute Registration Failed with status Operation has no matching challenge  [904]. failure #1

   

  Thanks Lars

   

   



  __________ Information from ESET NOD32 Antivirus, version of virus signature database 4545 (20091026) __________

  The message was checked by ESET NOD32 Antivirus.

  http://www.eset.com



------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From djbinter at yahoo.com  Mon Oct 26 12:29:30 2009
From: djbinter at yahoo.com (DJB)
Date: Mon, 26 Oct 2009 12:29:30 -0700 (PDT)
Subject: [Freeswitch-users] SIP UPDATE Method
In-Reply-To: <191c3a030910261145q46128a92w75d73158a7f73572@mail.gmail.com>
References: <528227.52311.qm@web37508.mail.mud.yahoo.com>
	<191c3a030910260808y45952bffu7118057caba4ea4@mail.gmail.com>
	<33829.77997.qm@web37506.mail.mud.yahoo.com>
	<191c3a030910261145q46128a92w75d73158a7f73572@mail.gmail.com>
Message-ID: <514887.94094.qm@web37502.mail.mud.yahoo.com>

We were trying to see whether we can adjust call duration on Session timers.  It was a question from the application developers.  I am not sure what they are trying to do exactly.  

Thank you.



________________________________
From: Anthony Minessale <anthony.minessale at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Mon, October 26, 2009 11:45:31 AM
Subject: Re: [Freeswitch-users] SIP UPDATE Method

what exactly are you expecting to use it for?
We never really supported it anyway.



On Mon, Oct 26, 2009 at 12:19 PM, DJB <djbinter at yahoo.com> wrote:

I wonder whether you will consider to put it back on the next version 1.0.5 since 1.0.4 has it?  
>
>
>Regards,
>Dorn B.
>
>
>
>
>
________________________________
From: Anthony Minessale <anthony.minessale at gmail.com>
>To: freeswitch-users at lists.freeswitch.org
>Sent: Mon, October 26, 2009 8:08:59 AM
>Subject: Re: [Freeswitch-users] SIP UPDATE Method
>
>
>It was never there before and it
> caused extreme havoc once we added it so we took it away again.
>
>
>
>On Sun, Oct 25, 2009 at 12:55 AM, DJB <djbinter at yahoo.com> wrote:
>
>I am wondering why after I update to trunk-15225, the Allow: UPDATE method is no longer there.
>>
>>>>User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15225
>>>>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>
>>>>Am I missing something here?
>>
>>>>Thank you.
>>
>>
>>
>>
>>>>_______________________________________________
>>>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>
>
>-- 
>Anthony Minessale II
>
>FreeSWITCH http://www.freeswitch.org/
>ClueCon http://www.cluecon.com/
>>
>Twitter: http://twitter.com/FreeSWITCH_wire
>
>AIM: anthm
>MSN:anthony_minessale at hotmail.com
>>GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>IRC: irc.freenode.net #freeswitch
>
>FreeSWITCH Developer Conference
>sip:888 at conference.freeswitch.org
>iax:guest at conference.freeswitch.org/888
>googletalk:conf+888 at conference.freeswitch.org
>pstn:213-799-1400
>
>
>_______________________________________________
>>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
pstn:213-799-1400



      
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From anthony.minessale at gmail.com  Mon Oct 26 12:47:08 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Oct 2009 14:47:08 -0500
Subject: [Freeswitch-users] SIP UPDATE Method
In-Reply-To: <514887.94094.qm@web37502.mail.mud.yahoo.com>
References: <528227.52311.qm@web37508.mail.mud.yahoo.com>
	<191c3a030910260808y45952bffu7118057caba4ea4@mail.gmail.com>
	<33829.77997.qm@web37506.mail.mud.yahoo.com>
	<191c3a030910261145q46128a92w75d73158a7f73572@mail.gmail.com>
	<514887.94094.qm@web37502.mail.mud.yahoo.com>
Message-ID: <191c3a030910261247n12f1917al8110cf2636f41e20@mail.gmail.com>

I think session timers will use invite if there is no update.

the session-timeout profile param should control that but you have to double
the number you actually want because it sends the new invite at the halfway
point.



On Mon, Oct 26, 2009 at 2:29 PM, DJB <djbinter at yahoo.com> wrote:

> We were trying to see whether we can adjust call duration on Session
> timers.  It was a question from the application developers.  I am not sure
> what they are trying to do exactly.
>
> Thank you.
>
> ------------------------------
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Mon, October 26, 2009 11:45:31 AM
>
> *Subject:* Re: [Freeswitch-users] SIP UPDATE Method
>
> what exactly are you expecting to use it for?
> We never really supported it anyway.
>
>
> On Mon, Oct 26, 2009 at 12:19 PM, DJB <djbinter at yahoo.com> wrote:
>
>> I wonder whether you will consider to put it back on the next version
>> 1.0.5 since 1.0.4 has it?
>>
>> Regards,
>> Dorn B.
>>
>>
>> ------------------------------
>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Sent:* Mon, October 26, 2009 8:08:59 AM
>> *Subject:* Re: [Freeswitch-users] SIP UPDATE Method
>>
>> It was never there before and it caused extreme havoc once we added it so
>> we took it away again.
>>
>>
>> On Sun, Oct 25, 2009 at 12:55 AM, DJB <djbinter at yahoo.com> wrote:
>>
>>> I am wondering why after I update to trunk-15225, the Allow: UPDATE
>>> method is no longer there.
>>>
>>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15225
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER,
>>> NOTIFY, PUBLISH, SUBSCRIBE
>>>
>>> Am I missing something here?
>>>
>>> Thank you.
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Mon Oct 26 12:50:04 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Oct 2009 14:50:04 -0500
Subject: [Freeswitch-users] SIP UPDATE Method
In-Reply-To: <191c3a030910261247n12f1917al8110cf2636f41e20@mail.gmail.com>
References: <528227.52311.qm@web37508.mail.mud.yahoo.com>
	<191c3a030910260808y45952bffu7118057caba4ea4@mail.gmail.com>
	<33829.77997.qm@web37506.mail.mud.yahoo.com>
	<191c3a030910261145q46128a92w75d73158a7f73572@mail.gmail.com>
	<514887.94094.qm@web37502.mail.mud.yahoo.com>
	<191c3a030910261247n12f1917al8110cf2636f41e20@mail.gmail.com>
Message-ID: <191c3a030910261250q3b933692jd028f2067ae33252@mail.gmail.com>

but i think the minimum value you can set is 120

On Mon, Oct 26, 2009 at 2:47 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> I think session timers will use invite if there is no update.
>
> the session-timeout profile param should control that but you have to
> double the number you actually want because it sends the new invite at the
> halfway point.
>
>
>
>
> On Mon, Oct 26, 2009 at 2:29 PM, DJB <djbinter at yahoo.com> wrote:
>
>> We were trying to see whether we can adjust call duration on Session
>> timers.  It was a question from the application developers.  I am not sure
>> what they are trying to do exactly.
>>
>> Thank you.
>>
>> ------------------------------
>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Sent:* Mon, October 26, 2009 11:45:31 AM
>>
>> *Subject:* Re: [Freeswitch-users] SIP UPDATE Method
>>
>> what exactly are you expecting to use it for?
>> We never really supported it anyway.
>>
>>
>> On Mon, Oct 26, 2009 at 12:19 PM, DJB <djbinter at yahoo.com> wrote:
>>
>>> I wonder whether you will consider to put it back on the next version
>>> 1.0.5 since 1.0.4 has it?
>>>
>>> Regards,
>>> Dorn B.
>>>
>>>
>>> ------------------------------
>>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>>> *To:* freeswitch-users at lists.freeswitch.org
>>> *Sent:* Mon, October 26, 2009 8:08:59 AM
>>> *Subject:* Re: [Freeswitch-users] SIP UPDATE Method
>>>
>>> It was never there before and it caused extreme havoc once we added it so
>>> we took it away again.
>>>
>>>
>>> On Sun, Oct 25, 2009 at 12:55 AM, DJB <djbinter at yahoo.com> wrote:
>>>
>>>> I am wondering why after I update to trunk-15225, the Allow: UPDATE
>>>> method is no longer there.
>>>>
>>>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15225
>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
>>>> REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>>
>>>> Am I missing something here?
>>>>
>>>> Thank you.
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:213-799-1400
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From ujjval at simplesignal.com  Mon Oct 26 12:50:50 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Mon, 26 Oct 2009 12:50:50 -0700
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <C49F867E-9993-42EB-9E48-BA442996A5C2@freeswitch.org>
References: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE7@EXMBXCLUS01.citservers.local>,
	<C49F867E-9993-42EB-9E48-BA442996A5C2@freeswitch.org>
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701B6DCEE@EXMBXCLUS01.citservers.local>

Are there any benchmarking test results available publicly?
________________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West [brian at freeswitch.org]
Sent: Monday, October 26, 2009 11:18 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Estimating Call Capacity

You'll have to do your own load testing.  Nobody can really tell you
exactly how many you'll get.

/b

On Oct 26, 2009, at 10:39 AM, Ujjval Karihaloo wrote:

> With the following spec for CPU and Memory can someone help me
> guesstimating how many simultaneous calls and Calls/sec a FS server
> can handle - Used as a Conferencing Server.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From anthony.minessale at gmail.com  Mon Oct 26 12:59:01 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Oct 2009 14:59:01 -0500
Subject: [Freeswitch-users] SIP UPDATE Method
In-Reply-To: <191c3a030910261250q3b933692jd028f2067ae33252@mail.gmail.com>
References: <528227.52311.qm@web37508.mail.mud.yahoo.com>
	<191c3a030910260808y45952bffu7118057caba4ea4@mail.gmail.com>
	<33829.77997.qm@web37506.mail.mud.yahoo.com>
	<191c3a030910261145q46128a92w75d73158a7f73572@mail.gmail.com>
	<514887.94094.qm@web37502.mail.mud.yahoo.com>
	<191c3a030910261247n12f1917al8110cf2636f41e20@mail.gmail.com>
	<191c3a030910261250q3b933692jd028f2067ae33252@mail.gmail.com>
Message-ID: <191c3a030910261259p3b1fcfecvbbaed9af0af4d206@mail.gmail.com>

in r15233 i put it back to the way it originally was but I may have to
remove that if it causes more problems.
We tried to handle update for display updating which was not working but the
handler for it was still in place which may have broken some automatic
behavior regarding update so I added it back to how it was originally to
determine that.


On Mon, Oct 26, 2009 at 2:50 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> but i think the minimum value you can set is 120
>
>
> On Mon, Oct 26, 2009 at 2:47 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> I think session timers will use invite if there is no update.
>>
>> the session-timeout profile param should control that but you have to
>> double the number you actually want because it sends the new invite at the
>> halfway point.
>>
>>
>>
>>
>> On Mon, Oct 26, 2009 at 2:29 PM, DJB <djbinter at yahoo.com> wrote:
>>
>>> We were trying to see whether we can adjust call duration on Session
>>> timers.  It was a question from the application developers.  I am not sure
>>> what they are trying to do exactly.
>>>
>>> Thank you.
>>>
>>> ------------------------------
>>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>>> *To:* freeswitch-users at lists.freeswitch.org
>>> *Sent:* Mon, October 26, 2009 11:45:31 AM
>>>
>>> *Subject:* Re: [Freeswitch-users] SIP UPDATE Method
>>>
>>> what exactly are you expecting to use it for?
>>> We never really supported it anyway.
>>>
>>>
>>> On Mon, Oct 26, 2009 at 12:19 PM, DJB <djbinter at yahoo.com> wrote:
>>>
>>>> I wonder whether you will consider to put it back on the next version
>>>> 1.0.5 since 1.0.4 has it?
>>>>
>>>> Regards,
>>>> Dorn B.
>>>>
>>>>
>>>> ------------------------------
>>>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>>>> *To:* freeswitch-users at lists.freeswitch.org
>>>> *Sent:* Mon, October 26, 2009 8:08:59 AM
>>>> *Subject:* Re: [Freeswitch-users] SIP UPDATE Method
>>>>
>>>> It was never there before and it caused extreme havoc once we added it
>>>> so we took it away again.
>>>>
>>>>
>>>> On Sun, Oct 25, 2009 at 12:55 AM, DJB <djbinter at yahoo.com> wrote:
>>>>
>>>>> I am wondering why after I update to trunk-15225, the Allow: UPDATE
>>>>> method is no longer there.
>>>>>
>>>>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15225
>>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
>>>>> REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>>>
>>>>> Am I missing something here?
>>>>>
>>>>> Thank you.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:213-799-1400
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:213-799-1400
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Mon Oct 26 13:03:59 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 26 Oct 2009 13:03:59 -0700
Subject: [Freeswitch-users] Resend: Issues with SIP + TCP?
In-Reply-To: <2C225D94-CD78-4683-A886-F767566033BF@freeswitch.org>
References: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>
	<191c3a030910260825w25b1891g523d5e30d72ee94f@mail.gmail.com>
	<2d9149cd0910260847i718c420ek99d8d291aa37e143@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953113@cooper>
	<DB49781A-1F3A-40CC-B357-6DDE08059FED@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953116@cooper>
	<E5059624-512D-4FF8-8D7D-974FDAD89F04@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C550895311E@cooper>
	<2C225D94-CD78-4683-A886-F767566033BF@freeswitch.org>
Message-ID: <87f2f3b90910261303i71d0c4afx3b154a2035d6fdd0@mail.gmail.com>

On Mon, Oct 26, 2009 at 11:40 AM, Brian West <brian at freeswitch.org> wrote:

> At some point we'll have to NO NO NO fix your broken crap.  :P  The
> reason we have sip so broken now is NOBODY CAN SAY NO!... Say it with
> me... "NO!"
>
> /b
>
> I was wondering... does anyone make a "SIP certification" program kinda
like a pen-tester except to find all the ways your SIP setup is broken? Just
curious.
-MC
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From brian at freeswitch.org  Mon Oct 26 13:13:00 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 26 Oct 2009 15:13:00 -0500
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71701B6DCEE@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE7@EXMBXCLUS01.citservers.local>,
	<C49F867E-9993-42EB-9E48-BA442996A5C2@freeswitch.org>
	<3C04B27FC880044F8FCD735D0D952FF71701B6DCEE@EXMBXCLUS01.citservers.local>
Message-ID: <2726B936-348C-40E2-B8F0-D3EB6E732C48@freeswitch.org>

I highly doubt it... You can wait for someone to post their results  
but in the end you'll have to do your own load testing because not  
everyone's numbers will jive with your use case.  Which is the reason  
the project never posts or endorses a set call count.

/b

On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote:

> Are there any benchmarking test results available publicly?
> ________________________________________



From brian at freeswitch.org  Mon Oct 26 13:18:37 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 26 Oct 2009 15:18:37 -0500
Subject: [Freeswitch-users] Resend: Issues with SIP + TCP?
In-Reply-To: <87f2f3b90910261303i71d0c4afx3b154a2035d6fdd0@mail.gmail.com>
References: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>
	<191c3a030910260825w25b1891g523d5e30d72ee94f@mail.gmail.com>
	<2d9149cd0910260847i718c420ek99d8d291aa37e143@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953113@cooper>
	<DB49781A-1F3A-40CC-B357-6DDE08059FED@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953116@cooper>
	<E5059624-512D-4FF8-8D7D-974FDAD89F04@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C550895311E@cooper>
	<2C225D94-CD78-4683-A886-F767566033BF@freeswitch.org>
	<87f2f3b90910261303i71d0c4afx3b154a2035d6fdd0@mail.gmail.com>
Message-ID: <214AD29A-7DEB-4D42-BFF7-7FDDB0901125@freeswitch.org>

You have SIPit, which was the SIP Backoff till Pillsbury got their  
panties in a wad.

/b

On Oct 26, 2009, at 3:03 PM, Michael Collins wrote:

> I was wondering... does anyone make a "SIP certification" program  
> kinda like a pen-tester except to find all the ways your SIP setup  
> is broken? Just curious.
> -MC



From msc at freeswitch.org  Mon Oct 26 13:20:00 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 26 Oct 2009 13:20:00 -0700
Subject: [Freeswitch-users] How to load user account from databse ?
In-Reply-To: <50c41b4e0910250658n74f92d01v56f1ab82087ad809@mail.gmail.com>
References: <50c41b4e0910250658n74f92d01v56f1ab82087ad809@mail.gmail.com>
Message-ID: <87f2f3b90910261320t7cce76eam167da2dcf0dbc38d@mail.gmail.com>

On Sun, Oct 25, 2009 at 6:58 AM, Lei Tang <lei.tlfly at gmail.com> wrote:

> Hi All:
>    I'm a newbie to FS.  I'm using  FS as a sbc and have about 20000 user
> account . Does somebody can tell me how to  make FS load use account
> information from a database such as mssql or mysql?  Could you give me a
> sample configuration file?
>    Thanks a lots.
>
>
Lei,

The feature that you want is mod_xml_curl - it allows you to pull config
information from a web server, and that web server will do the db lookup.
For more information check out these resources:

Wiki docs: http://wiki.freeswitch.org/wiki/Mod_xml_curl
Example:
http://fisheye.freeswitch.org/browse/FreeSWITCH/contrib/trixter/xml-curl

We don't have a soup-to-nuts how-to on mod_xml_curl because it can be done
in so many different ways. Someone could write a book on implementing
xml_curl techniques. Your best bet is to figure out how you want to store
your user information, then decide what's the best way to pull that
information from the database, then setup a web server to handle the
request/return process. Between the docs and the examples you should be able
to get up and running.

-MC
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From vinuth.madinur at gmail.com  Mon Oct 26 13:28:54 2009
From: vinuth.madinur at gmail.com (Vinuth Madinur)
Date: Tue, 27 Oct 2009 01:58:54 +0530
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <2726B936-348C-40E2-B8F0-D3EB6E732C48@freeswitch.org>
References: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE7@EXMBXCLUS01.citservers.local>
	<C49F867E-9993-42EB-9E48-BA442996A5C2@freeswitch.org>
	<3C04B27FC880044F8FCD735D0D952FF71701B6DCEE@EXMBXCLUS01.citservers.local>
	<2726B936-348C-40E2-B8F0-D3EB6E732C48@freeswitch.org>
Message-ID: <910309030910261328g4e2f51balaf73c033892c6147@mail.gmail.com>

Here are a few benchmarks that I had stumbled upon.

http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2

<http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2>
Thanks,
Vinuth.


On Tue, Oct 27, 2009 at 1:43 AM, Brian West <brian at freeswitch.org> wrote:

> I highly doubt it... You can wait for someone to post their results
> but in the end you'll have to do your own load testing because not
> everyone's numbers will jive with your use case.  Which is the reason
> the project never posts or endorses a set call count.
>
> /b
>
> On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote:
>
> > Are there any benchmarking test results available publicly?
> > ________________________________________
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kristian.kielhofner at gmail.com  Mon Oct 26 13:33:07 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Mon, 26 Oct 2009 16:33:07 -0400
Subject: [Freeswitch-users] Downloaded tar vs latest SVN - 200 OK has
	more headers
In-Reply-To: <191c3a030910261126l7cf12dd1n369acb79f4fc35a4@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE2@EXMBXCLUS01.citservers.local>
	<6444A50B-E5F2-4CB5-8AF9-C3D8EBE67BA4@freeswitch.org>
	<2d9149cd0910260815s5dc39ec2n869b0bdbfb0333ac@mail.gmail.com>
	<191c3a030910260841i6f0f0390mc5bfeab6e5ec7ab6@mail.gmail.com>
	<65d96fc80910260905r3953ba23p346271cdbafe25a@mail.gmail.com>
	<191c3a030910260916x47884d07t9abc9b779afbde68@mail.gmail.com>
	<191c3a030910261126l7cf12dd1n369acb79f4fc35a4@mail.gmail.com>
Message-ID: <2d9149cd0910261333w6129a272lfca3e6495aa9305e@mail.gmail.com>

It is ridiculous but thank you very much!

On Mon, Oct 26, 2009 at 2:26 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> This is ridiculous but here it is
>
>
> try r15230
>
> add the profile param
>
> <param name="pass-callee-id" value="false"/>
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From woof at iwoof.org  Mon Oct 26 13:47:54 2009
From: woof at iwoof.org (Andy Spitzer)
Date: Mon, 26 Oct 2009 16:47:54 -0400
Subject: [Freeswitch-users] Resend: Issues with SIP + TCP?
In-Reply-To: <87f2f3b90910261303i71d0c4afx3b154a2035d6fdd0@mail.gmail.com>
References: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>
	<191c3a030910260825w25b1891g523d5e30d72ee94f@mail.gmail.com>
	<2d9149cd0910260847i718c420ek99d8d291aa37e143@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953113@cooper>
	<DB49781A-1F3A-40CC-B357-6DDE08059FED@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953116@cooper>
	<E5059624-512D-4FF8-8D7D-974FDAD89F04@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C550895311E@cooper>
	<2C225D94-CD78-4683-A886-F767566033BF@freeswitch.org>
	<87f2f3b90910261303i71d0c4afx3b154a2035d6fdd0@mail.gmail.com>
Message-ID: <op.u2fdd2srg0dksq@woof3.fuzzy>

Woof!

On Mon, 26 Oct 2009 16:03:59 -0400, Michael Collins <msc at freeswitch.org>  
wrote:
>> I was wondering... does anyone make a "SIP certification" program kinda
> like a pen-tester except to find all the ways your SIP setup is broken?  
> Just curious.

Here is a start:
    http://interop.sipxecs.org/

It's best for testing phone implementations, but it can handle other UA's  
as well.

--Woof!


From gmaruzz at celliax.org  Mon Oct 26 13:55:45 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 26 Oct 2009 21:55:45 +0100
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <910309030910261328g4e2f51balaf73c033892c6147@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE7@EXMBXCLUS01.citservers.local>
	<C49F867E-9993-42EB-9E48-BA442996A5C2@freeswitch.org>
	<3C04B27FC880044F8FCD735D0D952FF71701B6DCEE@EXMBXCLUS01.citservers.local>
	<2726B936-348C-40E2-B8F0-D3EB6E732C48@freeswitch.org>
	<910309030910261328g4e2f51balaf73c033892c6147@mail.gmail.com>
Message-ID: <7b197bef0910261355v22840195s8cc71bd28ebd57ca@mail.gmail.com>

On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur
<vinuth.madinur at gmail.com> wrote:
> Here are a few benchmarks that I had stumbled upon.
> http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2

Please remember NO benchmarks are endorsed by the FS community or
developers, because there are just too many variables, and a simple
figure is just useful for marketing hype, not for real dimensioning.

You MUST do your own benchmarking, so you get an idea about how to
dimension for your own use case and hardware.


> Thanks,
> Vinuth.
>
> On Tue, Oct 27, 2009 at 1:43 AM, Brian West <brian at freeswitch.org> wrote:
>>
>> I highly doubt it... You can wait for someone to post their results
>> but in the end you'll have to do your own load testing because not
>> everyone's numbers will jive with your use case. ?Which is the reason
>> the project never posts or endorses a set call count.
>>
>> /b
>>
>> On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote:
>>
>> > Are there any benchmarking test results available publicly?
>> > ________________________________________
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From msc at freeswitch.org  Mon Oct 26 14:22:51 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 26 Oct 2009 14:22:51 -0700
Subject: [Freeswitch-users] "Can not record session. Media not enabled
	on channel."
In-Reply-To: <1256541677550-3890610.post@n2.nabble.com>
References: <1256034659005-3857858.post@n2.nabble.com>
	<87f2f3b90910201746h43a31b32m5c163fd98373f915@mail.gmail.com>
	<1256283391890-3877285.post@n2.nabble.com>
	<87f2f3b90910231200v71e72661id4cc9f2b7fba5610@mail.gmail.com>
	<1256541677550-3890610.post@n2.nabble.com>
Message-ID: <87f2f3b90910261422n6952fa26ya6a2a66452365146@mail.gmail.com>

On Mon, Oct 26, 2009 at 12:21 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> Yes, I can confirm - this exact error occurs each time when I start
> recording
> before the call is answered (just after sending ORIGINATE command) - but I
> think that's completely understandable that media is not ready for an
> unanswered call.
> But... is there any other event that guarantees media to be ready?
>
> Update to latest SVN and try again.
-MC
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From sjmudd at pobox.com  Mon Oct 26 14:23:42 2009
From: sjmudd at pobox.com (Simon J Mudd)
Date: Mon, 26 Oct 2009 22:23:42 +0100
Subject: [Freeswitch-users] problems registering on www.freeswitch.org
Message-ID: <20091026212342.GA28484@mad06.wl0.org>

Hi,

I tried to register on freeswitch.org using username sjmudd. It said the
username already existed so I requested having my password resent. Nothing
recieved.  I've also tried to register a new user simon.mudd with the
same results, no mail received.

I also see the following when trying to register which looks worrying:
http://merlin.wl0.org/20091026/e46494be187a53b3c27740aaff360910abd4facb.png

I'm pretty sure that I'm not filtering out the emails but can't be sure.
Could whoever maintains the web page contact me off list to help me
determine where the problem is?

Thanks,

Simon Mudd


From msc at freeswitch.org  Mon Oct 26 14:24:34 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 26 Oct 2009 14:24:34 -0700
Subject: [Freeswitch-users] Inbound DTMF Not Recognized By IVR
In-Reply-To: <573AFFBA055B4ECD83A910C12BE67129@greyhawk.tonecommander.com>
References: <573AFFBA055B4ECD83A910C12BE67129@greyhawk.tonecommander.com>
Message-ID: <87f2f3b90910261424s52b37a81ge79dc148fa9c181@mail.gmail.com>

On Fri, Oct 23, 2009 at 10:31 AM, Jerry Richards <jerry.richards at teotech.com
> wrote:

>
> I installed FS on a machine with a Sangoma A101D (PRI) card and if I make
> an
> inbound call to the FS IVR, it does not recognize DTMF digits from the PSTN
> phone.  If I call IVR from an internal phone, then it does recognize the
> DTMF digits.  I have mostly default configurations for everything.
>
> Best Regards,
> JErry
>
> I'm assuming that the DTMFs are coming in-band? If so make sure that you
issue the start_dtmf dialplan app.
-MC

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf
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From msc at freeswitch.org  Mon Oct 26 14:27:43 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 26 Oct 2009 14:27:43 -0700
Subject: [Freeswitch-users] problems registering on www.freeswitch.org
In-Reply-To: <20091026212342.GA28484@mad06.wl0.org>
References: <20091026212342.GA28484@mad06.wl0.org>
Message-ID: <87f2f3b90910261427w11a4f747r897894ea69e5fcd1@mail.gmail.com>

On Mon, Oct 26, 2009 at 2:23 PM, Simon J Mudd <sjmudd at pobox.com> wrote:

> Hi,
>
> I tried to register on freeswitch.org using username sjmudd. It said the
> username already existed so I requested having my password resent. Nothing
> recieved.  I've also tried to register a new user simon.mudd with the
> same results, no mail received.
>
> I also see the following when trying to register which looks worrying:
> http://merlin.wl0.org/20091026/e46494be187a53b3c27740aaff360910abd4facb.png
>
> I'm pretty sure that I'm not filtering out the emails but can't be sure.
> Could whoever maintains the web page contact me off list to help me
> determine where the problem is?
>
> Thanks,
>
> Simon Mudd
>
Thanks for letting us know. I'll check it out.
-MC
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From tculjaga at gmail.com  Mon Oct 26 14:28:44 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 26 Oct 2009 22:28:44 +0100
Subject: [Freeswitch-users] Downloaded tar vs latest SVN - 200 OK has
	more headers
In-Reply-To: <191c3a030910261126l7cf12dd1n369acb79f4fc35a4@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE2@EXMBXCLUS01.citservers.local>
	<6444A50B-E5F2-4CB5-8AF9-C3D8EBE67BA4@freeswitch.org>
	<2d9149cd0910260815s5dc39ec2n869b0bdbfb0333ac@mail.gmail.com>
	<191c3a030910260841i6f0f0390mc5bfeab6e5ec7ab6@mail.gmail.com>
	<65d96fc80910260905r3953ba23p346271cdbafe25a@mail.gmail.com>
	<191c3a030910260916x47884d07t9abc9b779afbde68@mail.gmail.com>
	<191c3a030910261126l7cf12dd1n369acb79f4fc35a4@mail.gmail.com>
Message-ID: <65d96fc80910261428v764eba7w5c58f5f99714c5f1@mail.gmail.com>

On Mon, Oct 26, 2009 at 7:26 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> This is ridiculous but here it is
>
>
> try r15230
>
> add the profile param
>
> <param name="pass-callee-id" value="false"/>
>
>
sorry for that but, this will save you a lot of e-mail explaining why calls
are not going through...

thanks man!

T.
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From tculjaga at gmail.com  Mon Oct 26 14:34:19 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 26 Oct 2009 22:34:19 +0100
Subject: [Freeswitch-users] Setup advice on small LAN
In-Reply-To: <A52116049AB24FAFB9A2E1B9A0B069D4@chaos>
References: <005501ca5662$fa594aa0$ef0bdfe0$@com>
	<A52116049AB24FAFB9A2E1B9A0B069D4@chaos>
Message-ID: <65d96fc80910261434n2938e812pfd62b3144d5a0559@mail.gmail.com>

>
>
> If one of the computers does a big download, it messes with FS in two ways.
> If a connection is made, the voices are broken up, intermittent and
> difficult to understand. If the download is long enough, the connection to
> Flowroute is no longer usable due to registration failure.
>
>

In any case, regardless if you are using a dedicated or mixed dsl line you
should flag your voice traffic properly.

signaling AF41, RTP EF... your voice traffic must never be flagged as pure
date when sending it through open internet!



T.
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From costa.zikalala at gmail.com  Mon Oct 26 14:39:31 2009
From: costa.zikalala at gmail.com (Costa Zikalala)
Date: Mon, 26 Oct 2009 23:39:31 +0200
Subject: [Freeswitch-users] How to load user account from databse ?
In-Reply-To: <87f2f3b90910261320t7cce76eam167da2dcf0dbc38d@mail.gmail.com>
References: <50c41b4e0910250658n74f92d01v56f1ab82087ad809@mail.gmail.com> 
	<87f2f3b90910261320t7cce76eam167da2dcf0dbc38d@mail.gmail.com>
Message-ID: <59daa2cd0910261439v414129edwbf8c98564c50cc13@mail.gmail.com>

Thanks for your response Michael,

Both the resources you've referred to don't explicitly say much about
databases. Could you elaborate just a bit on how on information would
retrieved from a database (MySQL) and presented to mod_xml_curl.

Thanks again.



2009/10/26 Michael Collins <msc at freeswitch.org>

>
>
> On Sun, Oct 25, 2009 at 6:58 AM, Lei Tang <lei.tlfly at gmail.com> wrote:
>
>> Hi All:
>>    I'm a newbie to FS.  I'm using  FS as a sbc and have about 20000 user
>> account . Does somebody can tell me how to  make FS load use account
>> information from a database such as mssql or mysql?  Could you give me a
>> sample configuration file?
>>    Thanks a lots.
>>
>>
> Lei,
>
> The feature that you want is mod_xml_curl - it allows you to pull config
> information from a web server, and that web server will do the db lookup.
> For more information check out these resources:
>
> Wiki docs: http://wiki.freeswitch.org/wiki/Mod_xml_curl
> Example:
> http://fisheye.freeswitch.org/browse/FreeSWITCH/contrib/trixter/xml-curl
>
> We don't have a soup-to-nuts how-to on mod_xml_curl because it can be done
> in so many different ways. Someone could write a book on implementing
> xml_curl techniques. Your best bet is to figure out how you want to store
> your user information, then decide what's the best way to pull that
> information from the database, then setup a web server to handle the
> request/return process. Between the docs and the examples you should be able
> to get up and running.
>
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From freeswitch-users-list at metik.com  Mon Oct 26 15:41:54 2009
From: freeswitch-users-list at metik.com (Metik)
Date: Mon, 26 Oct 2009 18:41:54 -0400
Subject: [Freeswitch-users] Setup advice on small LAN
In-Reply-To: <65d96fc80910261434n2938e812pfd62b3144d5a0559@mail.gmail.com>
References: <005501ca5662$fa594aa0$ef0bdfe0$@com><A52116049AB24FAFB9A2E1B9A0B069D4@chaos>
	<65d96fc80910261434n2938e812pfd62b3144d5a0559@mail.gmail.com>
Message-ID: <360485E934C6406FB0DE6B4ACCA2D021@chaos>

Unfortunately, most North American providers ignore (and in most cases reclassify) it before it reaches their border routers and it will be treated as best effort. Typically, the problems are introduced within the first mile and its simply a matter of getting the packet safely pass the edge unless your ISP is grossly oversubscribed (in terms of servicable traffic limited by their particular hardware and bandwidth).

If it is too costly to move to "second line" or "dedicated" DSL, he should be able to improve audio quality by acquiring a broadband router that has minimal QoS capabilities and adequate CPU (since the majority of them use software based queuing and packet fragmentation).  The only caveat is that the degree of success can vary between firmware versions.  Some of the consumer (gaming) or small business (VPN) grade routers work well (Linksys, DLINK, etc.).

-metik  


  From: Tihomir Culjaga 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, October 26, 2009 5:34 PM
  Subject: Re: [Freeswitch-users] Setup advice on small LAN




      If one of the computers does a big download, it messes with FS in two ways. If a connection is made, the voices are broken up, intermittent and difficult to understand. If the download is long enough, the connection to Flowroute is no longer usable due to registration failure.



  In any case, regardless if you are using a dedicated or mixed dsl line you should flag your voice traffic properly.

  signaling AF41, RTP EF... your voice traffic must never be flagged as pure date when sending it through open internet!



  T.



------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From egable+freeswitch at gmail.com  Mon Oct 26 15:01:29 2009
From: egable+freeswitch at gmail.com (Eliot Gable)
Date: Mon, 26 Oct 2009 18:01:29 -0400
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <bf6923ed0910261446x6a184ecdnee2e73e904406246@mail.gmail.com>
References: <bf6923ed0910261446x6a184ecdnee2e73e904406246@mail.gmail.com>
Message-ID: <bf6923ed0910261501k1ff8c892rfc198c31e30ab286@mail.gmail.com>

Although, FYI, I just benchmarked mod_xml_curl on a separate web app
server from FS with FS on a Dell R710 with their current best
processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and 32
GB memory. The web app server is less than half the power of the R710.
I maxed the web app server at 300 calls per second (both setting up
and tearing down) and the R710 running FS was 65% idle. No audio was
being proxied through FS, though. If I were running the web app server
on an equivalent R710, they probably would have been on-par with each
other in performance. Extrapolating, I expect that in such a case I
should be able to get at least 650 CPS out of FS, though for
production I would probably limit it to 400 CPS or less so I leave
room for miscellaneous tasks. I maxed out the R710 at over 16,000
simultaneous calls (again, no audio proxying) but the only reason I
couldn't do more was because I hit some sort of thread creation limit
in Linux. There was about 17 GB of memory used for this many calls.
This should give you some ballpark idea of what you can accomplish
with FS.

At some point, I will track down and resolve the thread creation
issue, at which time I believe call limits will be limited either by a
complex combination of available memory, the speed of the processor,
the cost of thread context switching, calls per second setup rate, and
call duration.

--
Eliot Gable

> -----Original Message-----
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli
>
> Sent: Monday, October 26, 2009 4:56 PM
>
> To: freeswitch-users at lists.freeswitch.org
>
> Subject: Re: [Freeswitch-users] Estimating Call Capacity
>
>
>
> On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur
>
> <vinuth.madinur at gmail.com> wrote:
>
> > Here are a few benchmarks that I had stumbled upon.
>
> > http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2
>
>
>
> Please remember NO benchmarks are endorsed by the FS community or
>
> developers, because there are just too many variables, and a simple
>
> figure is just useful for marketing hype, not for real dimensioning.
>
>
>
> You MUST do your own benchmarking, so you get an idea about how to
>
> dimension for your own use case and hardware.
>
>
>
>
>
> > Thanks,
>
> > Vinuth.
>
> >
>
> > On Tue, Oct 27, 2009 at 1:43 AM, Brian West <brian at freeswitch.org> wrote:
>
> >>
>
> >> I highly doubt it... You can wait for someone to post their results
>
> >> but in the end you'll have to do your own load testing because not
>
> >> everyone's numbers will jive with your use case.? Which is the reason
>
> >> the project never posts or endorses a set call count.
>
> >>
>
> >> /b
>
> >>
>
> >> On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote:
>
> >>
>
> >> > Are there any benchmarking test results available publicly?
>
> >> > ________________________________________
>
> >>
>
> >>
>
> >> _______________________________________________
>
> >> FreeSWITCH-users mailing list
>
> >> FreeSWITCH-users at lists.freeswitch.org
>
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> >> http://www.freeswitch.org
>
> >
>
> >
>
> > _______________________________________________
>
> > FreeSWITCH-users mailing list
>
> > FreeSWITCH-users at lists.freeswitch.org
>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> > http://www.freeswitch.org
>
> >
>
> >
>
>
>
>
>
>
>
> --
>
> Sincerely,
>
>
>
> Giovanni Maruzzelli
>
> Cell : +39-347-2665618
>
>
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org


From anthony.minessale at gmail.com  Mon Oct 26 16:03:37 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Oct 2009 18:03:37 -0500
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <bf6923ed0910261501k1ff8c892rfc198c31e30ab286@mail.gmail.com>
References: <bf6923ed0910261446x6a184ecdnee2e73e904406246@mail.gmail.com>
	<bf6923ed0910261501k1ff8c892rfc198c31e30ab286@mail.gmail.com>
Message-ID: <191c3a030910261603t6b539101p10ba7327ae9584e2@mail.gmail.com>

i wonder if we can at least get a taco-bell steak burrito for that if we
can't win the s-prize


On Mon, Oct 26, 2009 at 5:01 PM, Eliot Gable
<egable+freeswitch at gmail.com<egable%2Bfreeswitch at gmail.com>
> wrote:

> Although, FYI, I just benchmarked mod_xml_curl on a separate web app
> server from FS with FS on a Dell R710 with their current best
> processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and 32
> GB memory. The web app server is less than half the power of the R710.
> I maxed the web app server at 300 calls per second (both setting up
> and tearing down) and the R710 running FS was 65% idle. No audio was
> being proxied through FS, though. If I were running the web app server
> on an equivalent R710, they probably would have been on-par with each
> other in performance. Extrapolating, I expect that in such a case I
> should be able to get at least 650 CPS out of FS, though for
> production I would probably limit it to 400 CPS or less so I leave
> room for miscellaneous tasks. I maxed out the R710 at over 16,000
> simultaneous calls (again, no audio proxying) but the only reason I
> couldn't do more was because I hit some sort of thread creation limit
> in Linux. There was about 17 GB of memory used for this many calls.
> This should give you some ballpark idea of what you can accomplish
> with FS.
>
> At some point, I will track down and resolve the thread creation
> issue, at which time I believe call limits will be limited either by a
> complex combination of available memory, the speed of the processor,
> the cost of thread context switching, calls per second setup rate, and
> call duration.
>
> --
> Eliot Gable
>
> > -----Original Message-----
> >
> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Giovanni
> Maruzzelli
> >
> > Sent: Monday, October 26, 2009 4:56 PM
> >
> > To: freeswitch-users at lists.freeswitch.org
> >
> > Subject: Re: [Freeswitch-users] Estimating Call Capacity
> >
> >
> >
> > On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur
> >
> > <vinuth.madinur at gmail.com> wrote:
> >
> > > Here are a few benchmarks that I had stumbled upon.
> >
> > >
> http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2<http://wiki.voiceworks.pl/display/%7Epawel/FreeSwitch+performance+on+SUN+x2200+M2>
> >
> >
> >
> > Please remember NO benchmarks are endorsed by the FS community or
> >
> > developers, because there are just too many variables, and a simple
> >
> > figure is just useful for marketing hype, not for real dimensioning.
> >
> >
> >
> > You MUST do your own benchmarking, so you get an idea about how to
> >
> > dimension for your own use case and hardware.
> >
> >
> >
> >
> >
> > > Thanks,
> >
> > > Vinuth.
> >
> > >
> >
> > > On Tue, Oct 27, 2009 at 1:43 AM, Brian West <brian at freeswitch.org>
> wrote:
> >
> > >>
> >
> > >> I highly doubt it... You can wait for someone to post their results
> >
> > >> but in the end you'll have to do your own load testing because not
> >
> > >> everyone's numbers will jive with your use case.  Which is the reason
> >
> > >> the project never posts or endorses a set call count.
> >
> > >>
> >
> > >> /b
> >
> > >>
> >
> > >> On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote:
> >
> > >>
> >
> > >> > Are there any benchmarking test results available publicly?
> >
> > >> > ________________________________________
> >
> > >>
> >
> > >>
> >
> > >> _______________________________________________
> >
> > >> FreeSWITCH-users mailing list
> >
> > >> FreeSWITCH-users at lists.freeswitch.org
> >
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > >> http://www.freeswitch.org
> >
> > >
> >
> > >
> >
> > > _______________________________________________
> >
> > > FreeSWITCH-users mailing list
> >
> > > FreeSWITCH-users at lists.freeswitch.org
> >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > > http://www.freeswitch.org
> >
> > >
> >
> > >
> >
> >
> >
> >
> >
> >
> >
> > --
> >
> > Sincerely,
> >
> >
> >
> > Giovanni Maruzzelli
> >
> > Cell : +39-347-2665618
> >
> >
> >
> > _______________________________________________
> >
> > FreeSWITCH-users mailing list
> >
> > FreeSWITCH-users at lists.freeswitch.org
> >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Mon Oct 26 17:04:57 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 26 Oct 2009 17:04:57 -0700
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <191c3a030910261603t6b539101p10ba7327ae9584e2@mail.gmail.com>
References: <bf6923ed0910261446x6a184ecdnee2e73e904406246@mail.gmail.com>
	<bf6923ed0910261501k1ff8c892rfc198c31e30ab286@mail.gmail.com>
	<191c3a030910261603t6b539101p10ba7327ae9584e2@mail.gmail.com>
Message-ID: <87f2f3b90910261704n2708c573wc3ecbf80571b39f6@mail.gmail.com>

On Mon, Oct 26, 2009 at 4:03 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> i wonder if we can at least get a taco-bell steak burrito for that if we
> can't win the s-prize
>
>
>
Or at least a chalupa.
-MC
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From jmesquita at freeswitch.org  Mon Oct 26 17:08:47 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 26 Oct 2009 22:08:47 -0200
Subject: [Freeswitch-users] mod_nibblebill and memory problem
In-Reply-To: <8ccbff060910260424y7fb9f6eakbb45900838e6fd6d@mail.gmail.com>
References: <8ccbff060910260424y7fb9f6eakbb45900838e6fd6d@mail.gmail.com>
Message-ID: <ebbc2d550910261708rad012f0rddb8401e744a309f@mail.gmail.com>

Why don't you get us more information for debugging? We could use some vg
output, maybe?

JM

On Mon, Oct 26, 2009 at 9:24 AM, Dome Charoenyost <dome at tel.co.th> wrote:

> Dear All,
>             I'm running mod_nibblebill for my prepaid solution. I
> still have problem with memory. I have 4 GB RAM and runing debian
> squeeze 64 bit and 200 calls concurrent
>         Last time nibblebill running with 1 min heartbeat. when i
> check memory by htop FS user memory 2% anf growth to 60-89% in 8-9 hr.
> and then FS crash.
>         Now i change hearbeat to 0 it's mean nibble update balance
> when end of call. but everything are same FS start from 2% and growth
> to 20% in 2 days.
>        When i unload nibblebill FS running fine.
>        My question is when concurrent calls drop to 1-2 calls why FS
> (I think nibblebill) still use memory ? something wrong in nibblebill
> ?
>
> BG
> Dome C.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Mon Oct 26 17:38:04 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 26 Oct 2009 19:38:04 -0500
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <87f2f3b90910261704n2708c573wc3ecbf80571b39f6@mail.gmail.com>
References: <bf6923ed0910261446x6a184ecdnee2e73e904406246@mail.gmail.com>
	<bf6923ed0910261501k1ff8c892rfc198c31e30ab286@mail.gmail.com>
	<191c3a030910261603t6b539101p10ba7327ae9584e2@mail.gmail.com>
	<87f2f3b90910261704n2708c573wc3ecbf80571b39f6@mail.gmail.com>
Message-ID: <11489FD5-12F9-4EF3-BA2E-11ED25E02902@freeswitch.org>

yo quiero taco bell

/b

On Oct 26, 2009, at 7:04 PM, Michael Collins wrote:

> Or at least a chalupa.
> -MC



From larclap at yahoo.com  Mon Oct 26 19:29:40 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Mon, 26 Oct 2009 19:29:40 -0700
Subject: [Freeswitch-users] Problem with inbound call answered but no sound
Message-ID: <016e01ca56ad$56362380$02a26a80$@com>

I have tried to update (make current) twice since 15183. All inbound calls
are picked up but the caller hears nothing but a couple of clicks. The most
recent version I've tried is 15241.

 

Any ideas on what may be causing this? 

 

http://pastebin.freeswitch.org/10843

 

Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686
i386 GNU/Linux

 

Thanks Lars

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From brian at freeswitch.org  Mon Oct 26 19:45:42 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 26 Oct 2009 21:45:42 -0500
Subject: [Freeswitch-users] Problem with inbound call answered but no
	sound
In-Reply-To: <016e01ca56ad$56362380$02a26a80$@com>
References: <016e01ca56ad$56362380$02a26a80$@com>
Message-ID: <168575F8-6507-4AD2-B448-2D6FDA0152B6@freeswitch.org>

you're behind nat and you didn't set the ext-rtp-ip or ext-sip-ip  
correctly?

/b

On Oct 26, 2009, at 9:29 PM, Lars Zeb wrote:

> I have tried to update (make current) twice since 15183. All inbound  
> calls are picked up but the caller hears nothing but a couple of  
> clicks. The most recent version I?ve tried is 15241.
>
> Any ideas on what may be causing this?
>
> http://pastebin.freeswitch.org/10843
>
> Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686  
> i686 i386 GNU/Linux
>
> Thanks Lars

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From egable+freeswitch at gmail.com  Mon Oct 26 20:02:15 2009
From: egable+freeswitch at gmail.com (Eliot Gable)
Date: Mon, 26 Oct 2009 23:02:15 -0400
Subject: [Freeswitch-users] Problem with inbound call answered but no
	sound
In-Reply-To: <bf6923ed0910262001p69074ebk699ce7af802e0c9@mail.gmail.com>
References: <016e01ca56ad$56362380$02a26a80$@com>
	<bf6923ed0910262001p69074ebk699ce7af802e0c9@mail.gmail.com>
Message-ID: <bf6923ed0910262002s5afa3c44mc20780ac468f217c@mail.gmail.com>

FYI, it generally makes debugging easier if you do this:

sofia profile external siptrace on
sofia profile internal siptrace on

That way you can see the actual signaling and it is usually more clear
what is going on. In most cases, you will probably be able to figure
it out yourself just looking at the signaling.


 On Mon, Oct 26, 2009 at 10:29 PM, Lars Zeb <larclap at yahoo.com> wrote:
> I have tried to update (make current) twice since 15183. All inbound calls
> are picked up but the caller hears nothing but a couple of clicks. The most
> recent version I?ve tried is 15241.
>
>
>
> Any ideas on what may be causing this?
>
>
>
> http://pastebin.freeswitch.org/10843
>
>
>
> Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686
> i386 GNU/Linux
>
>
>
> Thanks Lars
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero


From larclap at yahoo.com  Mon Oct 26 20:03:53 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Mon, 26 Oct 2009 20:03:53 -0700
Subject: [Freeswitch-users] Problem with inbound call answered but
	no	sound
In-Reply-To: <168575F8-6507-4AD2-B448-2D6FDA0152B6@freeswitch.org>
References: <016e01ca56ad$56362380$02a26a80$@com>
	<168575F8-6507-4AD2-B448-2D6FDA0152B6@freeswitch.org>
Message-ID: <018801ca56b2$1e63e0f0$5b2ba2d0$@com>

I haven't changed anything since v15183, where it worked OK.

 

In conf/sip_profiles/external:

 

     <param name="ext-rtp-ip" value="auto-nat"/>

     <param name="ext-sip-ip" value="auto-nat"/>

 

And

 

 

Name                    external

Domain Name             N/A

DBName                  sofia_reg_external

Pres Hosts

Dialplan                XML

Context                 public

Challenge Realm         auto_to

RTP-IP                  192.168.10.29

Ext-RTP-IP              192.168.10.29

SIP-IP                  192.168.10.29

Ext-SIP-IP              192.168.10.29

URL                     sip:mod_sofia at 192.168.10.29:5090

BIND-URL                sip:mod_sofia at 192.168.10.29:5090

HOLD-MUSIC              local_stream://moh

OUTBOUND-PROXY          N/A

CODECS                  PCMU,PCMA,GSM

TEL-EVENT               101

DTMF-MODE               rfc2833

CNG                     13

SESSION-TO              0

MAX-DIALOG              0

NOMEDIA                 false

LATE-NEG                false

PROXY-MEDIA             false

AGGRESSIVENAT           false

STUN-ENABLED            true

STUN-AUTO-DISABLE       false

CALLS-IN                1

FAILED-CALLS-IN         1

CALLS-OUT               0

FAILED-CALLS-OUT        0

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Monday, October 26, 2009 7:46 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Problem with inbound call answered but no
sound

 

you're behind nat and you didn't set the ext-rtp-ip or ext-sip-ip correctly?

 

/b

 

On Oct 26, 2009, at 9:29 PM, Lars Zeb wrote:





I have tried to update (make current) twice since 15183. All inbound calls
are picked up but the caller hears nothing but a couple of clicks. The most
recent version I've tried is 15241.

 

Any ideas on what may be causing this?

 

http://pastebin.freeswitch.org/10843

 

Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686
i386 GNU/Linux

 

Thanks Lars

 

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From red.rain.seven at gmail.com  Mon Oct 26 20:26:55 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Tue, 27 Oct 2009 11:26:55 +0800
Subject: [Freeswitch-users] How to load user account from databse ?
In-Reply-To: <59daa2cd0910261439v414129edwbf8c98564c50cc13@mail.gmail.com>
References: <50c41b4e0910250658n74f92d01v56f1ab82087ad809@mail.gmail.com>
	<87f2f3b90910261320t7cce76eam167da2dcf0dbc38d@mail.gmail.com>
	<59daa2cd0910261439v414129edwbf8c98564c50cc13@mail.gmail.com>
Message-ID: <59ad9ca10910262026y193f1c70q1d1b9c6b82e2ee8@mail.gmail.com>

We have used xml_odbc to get user data from database.

One question though, xml_odbc will load all the user info in database to
memory just like it is loading from plain xml config files when FS starts. I
was wondering if mod_xml_curl is doing this differently by only requesting
user information on demand (real time)? and not loading all the user
information from the beginning of FS start.



On Tue, Oct 27, 2009 at 5:39 AM, Costa Zikalala <costa.zikalala at gmail.com>wrote:

> Thanks for your response Michael,
>
> Both the resources you've referred to don't explicitly say much about
> databases. Could you elaborate just a bit on how on information would
> retrieved from a database (MySQL) and presented to mod_xml_curl.
>
> Thanks again.
>
>
>
> 2009/10/26 Michael Collins <msc at freeswitch.org>
>
>>
>>
>> On Sun, Oct 25, 2009 at 6:58 AM, Lei Tang <lei.tlfly at gmail.com> wrote:
>>
>>> Hi All:
>>>    I'm a newbie to FS.  I'm using  FS as a sbc and have about 20000 user
>>> account . Does somebody can tell me how to  make FS load use account
>>> information from a database such as mssql or mysql?  Could you give me a
>>> sample configuration file?
>>>    Thanks a lots.
>>>
>>>
>> Lei,
>>
>> The feature that you want is mod_xml_curl - it allows you to pull config
>> information from a web server, and that web server will do the db lookup.
>> For more information check out these resources:
>>
>> Wiki docs: http://wiki.freeswitch.org/wiki/Mod_xml_curl
>> Example:
>> http://fisheye.freeswitch.org/browse/FreeSWITCH/contrib/trixter/xml-curl
>>
>> We don't have a soup-to-nuts how-to on mod_xml_curl because it can be done
>> in so many different ways. Someone could write a book on implementing
>> xml_curl techniques. Your best bet is to figure out how you want to store
>> your user information, then decide what's the best way to pull that
>> information from the database, then setup a web server to handle the
>> request/return process. Between the docs and the examples you should be able
>> to get up and running.
>>
>> -MC
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From red.rain.seven at gmail.com  Mon Oct 26 20:34:33 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Tue, 27 Oct 2009 11:34:33 +0800
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <11489FD5-12F9-4EF3-BA2E-11ED25E02902@freeswitch.org>
References: <bf6923ed0910261446x6a184ecdnee2e73e904406246@mail.gmail.com>
	<bf6923ed0910261501k1ff8c892rfc198c31e30ab286@mail.gmail.com>
	<191c3a030910261603t6b539101p10ba7327ae9584e2@mail.gmail.com>
	<87f2f3b90910261704n2708c573wc3ecbf80571b39f6@mail.gmail.com>
	<11489FD5-12F9-4EF3-BA2E-11ED25E02902@freeswitch.org>
Message-ID: <59ad9ca10910262034v654f84bev5690057ce7dbeb98@mail.gmail.com>

ah..... do you have to mention about taco bell that now I have moved away
from US.
~~~drooling~~~

On Tue, Oct 27, 2009 at 8:38 AM, Brian West <brian at freeswitch.org> wrote:

> yo quiero taco bell
>
> /b
>
> On Oct 26, 2009, at 7:04 PM, Michael Collins wrote:
>
> > Or at least a chalupa.
> > -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From lakindia89 at gmail.com  Mon Oct 26 22:43:05 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Tue, 27 Oct 2009 11:13:05 +0530
Subject: [Freeswitch-users] Playing Background music as well as a file
Message-ID: <7d79b3930910262243h4d7bbbb9j83717b6f4d2d766a@mail.gmail.com>

Hi all,
    I've done experimenting with the uuid_displace and mux. What mux does is
playing a file when the conversation is also happening. But I've a different
requirement.
I need to play a background music to a UUID, that will get played
continuously and also I need to play some other voice message to that uuid.

Is it possible in freeswitch? If so please guide me on how to do that!
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From lei.tlfly at gmail.com  Mon Oct 26 23:41:17 2009
From: lei.tlfly at gmail.com (Lei Tang)
Date: Tue, 27 Oct 2009 14:41:17 +0800
Subject: [Freeswitch-users] How to load user account from databse ?
In-Reply-To: <59ad9ca10910262026y193f1c70q1d1b9c6b82e2ee8@mail.gmail.com>
References: <50c41b4e0910250658n74f92d01v56f1ab82087ad809@mail.gmail.com>
	<87f2f3b90910261320t7cce76eam167da2dcf0dbc38d@mail.gmail.com>
	<59daa2cd0910261439v414129edwbf8c98564c50cc13@mail.gmail.com>
	<59ad9ca10910262026y193f1c70q1d1b9c6b82e2ee8@mail.gmail.com>
Message-ID: <50c41b4e0910262341n5c4ae984ra2b96d47c20db6a1@mail.gmail.com>

Hi  noob and Michael, thanks for your answers,  I'll try to use
mod_xml_curl.
Hi  Henry,  http://wiki.freeswitch.org/wiki/Mod_xml_curl has mentioned, FS
will post a request to webserver when it get a registration request. you can
refer to the doc for more detail.




Lei.Tang
lei.tlfly at gmail.com
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From ivan at myrvold.org  Mon Oct 26 23:46:40 2009
From: ivan at myrvold.org (Ivan C Myrvold)
Date: Tue, 27 Oct 2009 07:46:40 +0100
Subject: [Freeswitch-users] SIP provider with extern rtp server
Message-ID: <A2D2E918-B494-4179-96BA-CA3A7717846C@myrvold.org>

I have used a SIP provider for more than a year. A few days ago, he  
said he was moving to a new server, and asked me to reconfigure. I  
did, and everything seemed to work fine, until I did an outgoing call  
to an external telephone. I found out I had no audio, in neither  
direction. Incoming calls was working fine.

My provider said that the rtp is not going through the sip server, as  
it did earlier, but now through several other IP's.

Do I have to do some special configuration to handle that?

Ivan


From dome at tel.co.th  Tue Oct 27 00:07:14 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Tue, 27 Oct 2009 14:07:14 +0700
Subject: [Freeswitch-users] Playing Background music as well as a file
In-Reply-To: <7d79b3930910262243h4d7bbbb9j83717b6f4d2d766a@mail.gmail.com>
References: <7d79b3930910262243h4d7bbbb9j83717b6f4d2d766a@mail.gmail.com>
Message-ID: <8ccbff060910270007x48b9ecdahf919a4f0e266f4e5@mail.gmail.com>

2009/10/27 lakshmanan ganapathy <lakindia89 at gmail.com>:
> Hi all,
> ??? I've done experimenting with the uuid_displace and mux. What mux does is
> playing a file when the conversation is also happening. But I've a different
> requirement.

Is posible to use uuid_displace in dialplan ?  i want to do music
background base on called id

> I need to play a background music to a UUID, that will get played
> continuously and also I need to play some other voice message to that uuid.
>
> Is it possible in freeswitch? If so please guide me on how to do that!
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From red.rain.seven at gmail.com  Tue Oct 27 00:51:01 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Tue, 27 Oct 2009 15:51:01 +0800
Subject: [Freeswitch-users] How to load user account from databse ?
In-Reply-To: <50c41b4e0910262341n5c4ae984ra2b96d47c20db6a1@mail.gmail.com>
References: <50c41b4e0910250658n74f92d01v56f1ab82087ad809@mail.gmail.com>
	<87f2f3b90910261320t7cce76eam167da2dcf0dbc38d@mail.gmail.com>
	<59daa2cd0910261439v414129edwbf8c98564c50cc13@mail.gmail.com>
	<59ad9ca10910262026y193f1c70q1d1b9c6b82e2ee8@mail.gmail.com>
	<50c41b4e0910262341n5c4ae984ra2b96d47c20db6a1@mail.gmail.com>
Message-ID: <59ad9ca10910270051n256fbb2g684dc7e920b4999e@mail.gmail.com>

Sorry, guys, I need to fix my earlier statement about xml_odbc. I just
talked to my developer and xml_odbc IS loading registered user on real time,
not from the beginning of the FS start.

So Lei Tang, you can use xml_odbc as well

On Tue, Oct 27, 2009 at 2:41 PM, Lei Tang <lei.tlfly at gmail.com> wrote:

> Hi  noob and Michael, thanks for your answers,  I'll try to use
> mod_xml_curl.
> Hi  Henry,  http://wiki.freeswitch.org/wiki/Mod_xml_curl has mentioned, FS
> will post a request to webserver when it get a registration request. you can
> refer to the doc for more detail.
>
>
>
>
>
> Lei.Tang
> lei.tlfly at gmail.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From peter.olsson at visionutveckling.se  Tue Oct 27 01:20:29 2009
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 27 Oct 2009 09:20:29 +0100
Subject: [Freeswitch-users] Resend: Issues with SIP + TCP?
In-Reply-To: <2C225D94-CD78-4683-A886-F767566033BF@freeswitch.org>
References: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>
	<191c3a030910260825w25b1891g523d5e30d72ee94f@mail.gmail.com>
	<2d9149cd0910260847i718c420ek99d8d291aa37e143@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953113@cooper>
	<DB49781A-1F3A-40CC-B357-6DDE08059FED@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953116@cooper>
	<E5059624-512D-4FF8-8D7D-974FDAD89F04@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C550895311E@cooper>
	<2C225D94-CD78-4683-A886-F767566033BF@freeswitch.org>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C55089531CB@cooper>

Thanks Brian, it works now. I'll try to learn to say NO next time :)

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West
Skickat: den 26 oktober 2009 19:40
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Resend: Issues with SIP + TCP?

At some point we'll have to NO NO NO fix your broken crap.  :P  The  
reason we have sip so broken now is NOBODY CAN SAY NO!... Say it with  
me... "NO!"

/b

On Oct 26, 2009, at 1:27 PM, Peter Olsson wrote:

> I understand your frustration :) We deal with SIP integration with  
> about 10 different PBX vendors today, And it's always something that  
> doesn't work as it should. Right now I don't have anything more  
> connected to FS though.
>
> /Peter


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4ae5ef4a32936548940180!



From lei.tlfly at gmail.com  Tue Oct 27 01:49:32 2009
From: lei.tlfly at gmail.com (Lei Tang)
Date: Tue, 27 Oct 2009 16:49:32 +0800
Subject: [Freeswitch-users] how to config FS with two net interface?
Message-ID: <50c41b4e0910270149k2eaf8ee8v78ef24776947f270@mail.gmail.com>

Hi all, I run FS on a machine with two net interface, each interface has a
ip addr, one of the them connect to public network(has ip addr A), the
other  connect to a private network(has ip addr B), FS server as a SIP
server for public through A, all outbound call will bridge to a softswitch
in private network through B. here is my sofia config file and diaplan
config:

sofia internal.xml
....
<param name="rtp-ip" value="A"/>
<param name="sip-ip" value="A"/>
 ....

sofia external.xml
....
<param name="rtp-ip" value="B"/>
<param name="sip-ip" value="B"/>
....

dialplan
......
<extension name="OUTBOUND">
    <condition field="destination_number" expression="^(\d+)$">
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION"/>
        <action application="set"
data="effective_caller_id_number=xxxxxxx"/>  <!--here change the caller
number -->
        <action application="bridge"
data="sofia/external/${destination_number}@xxxxx"/>
      </condition>
    </extension>
.....

then call seq is
sipAgent --> [internal -->(bridge)-->external] -->softswith
                          FREESWITCH

the question is, when sipAgent make a outbound call, FS can't recevie the
caller's up audio stream, I traced the SIP packets, found that FS has return
addr B in SDP when ack the invite request from sipAgent, the ack packet is
===============
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
xxxxx:12208;branch=z9hG4bK-d8754z-dc750d57652c7c51-1---d8754z-;rport=12208
From: "1000" <sip:xxxx at A>;tag=cb4d3c4e
To: "65960581" <sip:xxxx at A>;tag=DtvSc0QX01yKN
Call-ID: ZTI2NmIwZGZiYzlhOGNkNTdiYWUzMzkzZTMwYzgxZjI.
CSeq: 2 INVITE
Contact: <sip:xxxxxx at B:5060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 245

v=0
o=FreeSWITCH 1256598185 1256598186 IN IP4 B   ;>>>>wrong this is the ip addr
of the adapter connect to the private network
s=FreeSWITCH
c=IN IP4 B ;>>>>wrong this is the ip addr of the adapter connect to the
private network
t=0 0
m=audio 31066 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
================
I think FS should return A in SDP, not the external binding addr (B), does
somebody known how to solve this problem?

-- 
Lei.Tang
lei.tlfly at gmail.com
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From srinivas.ksvreddy at gmail.com  Tue Oct 27 02:32:48 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 27 Oct 2009 15:02:48 +0530
Subject: [Freeswitch-users] reload from managed
Message-ID: <f8af5740910270232k689e8856sbc0dc8b668d9eacf@mail.gmail.com>

Hi,

how can i call reload from managed code? and i called
Csharp_switch_xml_open_root, this is working first time, then onwards it is
not working.
is this the correct function to call reload?

-- 
Srinivasula Reddy K
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From srinivas.ksvreddy at gmail.com  Tue Oct 27 05:48:32 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 27 Oct 2009 18:18:32 +0530
Subject: [Freeswitch-users] switch_xml_open_root
Message-ID: <f8af5740910270548n3fe7883fp18f126afd421d89e@mail.gmail.com>

**

Hi,

when i am calling  switch_xml_open_root(1,err) . i am getting this warning
message.
HEAP[FreeSwitch.exe]: Invalid Address specified to RtlFreeHeap( 00160000,
100E9FD0 ). can any know. please help me.

Thanks
Srinivasula Reddy K
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From lakindia89 at gmail.com  Tue Oct 27 06:05:49 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Tue, 27 Oct 2009 18:35:49 +0530
Subject: [Freeswitch-users] Playing Background music as well as a file
In-Reply-To: <8ccbff060910270007x48b9ecdahf919a4f0e266f4e5@mail.gmail.com>
References: <7d79b3930910262243h4d7bbbb9j83717b6f4d2d766a@mail.gmail.com>
	<8ccbff060910270007x48b9ecdahf919a4f0e266f4e5@mail.gmail.com>
Message-ID: <7d79b3930910270605s675ac0fcn3e03e08f5f9d0584@mail.gmail.com>

I tried the uuid_displace by using the Event Socket Outbound.
Not tried within dialplan, and I'm not sure whether that could be done in
dialplan itself.


On Tue, Oct 27, 2009 at 12:37 PM, Dome Charoenyost <dome at tel.co.th> wrote:

> 2009/10/27 lakshmanan ganapathy <lakindia89 at gmail.com>:
> > Hi all,
> >     I've done experimenting with the uuid_displace and mux. What mux does
> is
> > playing a file when the conversation is also happening. But I've a
> different
> > requirement.
>
> Is posible to use uuid_displace in dialplan ?  i want to do music
> background base on called id
>
> > I need to play a background music to a UUID, that will get played
> > continuously and also I need to play some other voice message to that
> uuid.
> >
> > Is it possible in freeswitch? If so please guide me on how to do that!
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From egable+freeswitch at gmail.com  Tue Oct 27 06:57:23 2009
From: egable+freeswitch at gmail.com (Eliot Gable)
Date: Tue, 27 Oct 2009 09:57:23 -0400
Subject: [Freeswitch-users] how to config FS with two net interface?
In-Reply-To: <50c41b4e0910270149k2eaf8ee8v78ef24776947f270@mail.gmail.com>
References: <50c41b4e0910270149k2eaf8ee8v78ef24776947f270@mail.gmail.com>
Message-ID: <bf6923ed0910270657n22fb6115n6314398006612e25@mail.gmail.com>

Try setting ext-rtp-ip and ext-sip-ip on both profiles.

On Tue, Oct 27, 2009 at 4:49 AM, Lei Tang <lei.tlfly at gmail.com> wrote:
> Hi all, I run FS on a machine with two net interface, each interface has a
> ip addr, one of the them connect to public network(has ip addr A), the
> other? connect to a private network(has ip addr B), FS server as a SIP
> server for public through A, all outbound call will bridge to a softswitch
> in private network through B. here is my sofia config file and diaplan
> config:
>
> sofia internal.xml
> ....
> <param name="rtp-ip" value="A"/>
> <param name="sip-ip" value="A"/>
> ?....
>
> sofia external.xml
> ....
> <param name="rtp-ip" value="B"/>
> <param name="sip-ip" value="B"/>
> ....
>
> dialplan
> ......
> <extension name="OUTBOUND">
> ??? <condition field="destination_number" expression="^(\d+)$">
> ??????? <action application="set" data="hangup_after_bridge=true"/>
> ??????? <action application="set"
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION"/>
> ??????? <action application="set"
> data="effective_caller_id_number=xxxxxxx"/>? <!--here change the caller
> number -->
> ??? ??? <action application="bridge"
> data="sofia/external/${destination_number}@xxxxx"/>
> ????? </condition>
> ??? </extension>
> .....
>
> then call seq is
> sipAgent --> [internal -->(bridge)-->external] -->softswith
> ????????????????????????? FREESWITCH
>
> the question is, when sipAgent make a outbound call, FS can't recevie the
> caller's up audio stream, I traced the SIP packets, found that FS has return
> addr B in SDP when ack the invite request from sipAgent, the ack packet is
> ===============
> SIP/2.0 183 Session Progress
> Via: SIP/2.0/UDP
> xxxxx:12208;branch=z9hG4bK-d8754z-dc750d57652c7c51-1---d8754z-;rport=12208
> From: "1000" <sip:xxxx at A>;tag=cb4d3c4e
> To: "65960581" <sip:xxxx at A>;tag=DtvSc0QX01yKN
> Call-ID: ZTI2NmIwZGZiYzlhOGNkNTdiYWUzMzkzZTMwYzgxZjI.
> CSeq: 2 INVITE
> Contact: <sip:xxxxxx at B:5060;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460
> Accept: application/sdp
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
> REFER, UPDATE, REGISTER, INFO, PUBLISH
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 245
>
> v=0
> o=FreeSWITCH 1256598185 1256598186 IN IP4 B?? ;>>>>wrong this is the ip addr
> of the adapter connect to the private network
> s=FreeSWITCH
> c=IN IP4 B ;>>>>wrong this is the ip addr of the adapter connect to the
> private network
> t=0 0
> m=audio 31066 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> ================
> I think FS should return A in SDP, not the external binding addr (B), does
> somebody known how to solve this problem?
>
> --
> Lei.Tang
> lei.tlfly at gmail.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero


From egable+freeswitch at gmail.com  Tue Oct 27 06:58:45 2009
From: egable+freeswitch at gmail.com (Eliot Gable)
Date: Tue, 27 Oct 2009 09:58:45 -0400
Subject: [Freeswitch-users] SIP provider with extern rtp server
In-Reply-To: <A2D2E918-B494-4179-96BA-CA3A7717846C@myrvold.org>
References: <A2D2E918-B494-4179-96BA-CA3A7717846C@myrvold.org>
Message-ID: <bf6923ed0910270658t61bb6687ne0b7d3ec5090e2ff@mail.gmail.com>

Make sure you let their media IPs through your firewall. Also, if you
are behind a NAT, check you have things passing to the correct
internal address.

On Tue, Oct 27, 2009 at 2:46 AM, Ivan C Myrvold <ivan at myrvold.org> wrote:
> I have used a SIP provider for more than a year. A few days ago, he
> said he was moving to a new server, and asked me to reconfigure. I
> did, and everything seemed to work fine, until I did an outgoing call
> to an external telephone. I found out I had no audio, in neither
> direction. Incoming calls was working fine.
>
> My provider said that the rtp is not going through the sip server, as
> it did earlier, but now through several other IP's.
>
> Do I have to do some special configuration to handle that?
>
> Ivan
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero


From maciej.aniserowicz at gmail.com  Tue Oct 27 07:55:34 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Tue, 27 Oct 2009 07:55:34 -0700 (PDT)
Subject: [Freeswitch-users] "Can not record session. Media not enabled
 on channel."
In-Reply-To: <87f2f3b90910261422n6952fa26ya6a2a66452365146@mail.gmail.com>
References: <1256034659005-3857858.post@n2.nabble.com>
	<87f2f3b90910201746h43a31b32m5c163fd98373f915@mail.gmail.com>
	<1256283391890-3877285.post@n2.nabble.com>
	<87f2f3b90910231200v71e72661id4cc9f2b7fba5610@mail.gmail.com>
	<1256541677550-3890610.post@n2.nabble.com>
	<87f2f3b90910261422n6952fa26ya6a2a66452365146@mail.gmail.com>
Message-ID: <1256655334215-3899478.post@n2.nabble.com>


Sorry, trunk does not compile on win7, here are the details:


rev.15247

---------------------------
Microsoft Visual C++ Debug Library
---------------------------
Debug Assertion Failed!

Program: ...ev\Projects\External\FreeSWITCH\Original\Debug\FreeSwitch.exe
File: f:\dd\vctools\crt_bld\self_x86\crt\src\dbgheap.c
Line: 1317

Expression: _CrtIsValidHeapPointer(pUserData)

For information on how your program can cause an assertion
failure, see the Visual C++ documentation on asserts.

(Press Retry to debug the application)
---------------------------
Abort   Retry   Ignore   
---------------------------

VS Call stack:

  ntdll.dll!77ccfadc()  
  [Frames below may be incorrect and/or missing, no symbols loaded for ntdll.dll] 
  ntdll.dll!77c9272c()  
  ntdll.dll!77c5e1ef()  
  msvcr90d.dll!_free_dbg_nolock(void * pUserData=0x00664b88, int nBlockUse=1)  Line 1317 + 0x9 bytes C++
  msvcr90d.dll!_free_dbg(void * pUserData=0x00664b88, int nBlockUse=1)  Line 1258 + 0xd bytes C++
  msvcr90d.dll!free(void * pUserData=0x00664b88)  Line 49 + 0xb bytes C++
> FreeSwitch.dll!switch_core_mime_add_type(const char * type=0x003bcd1c, const char * ext=0x003bcd37)  Line 748 + 0xc bytes C
  FreeSwitch.dll!load_mime_types()  Line 791 C
  FreeSwitch.dll!switch_core_init(unsigned int flags=65, switch_bool_t console=SWITCH_TRUE, const char * * err=0x003bf68c)  Line 1244 C
  FreeSwitch.dll!switch_core_init_and_modload(unsigned int flags=65, switch_bool_t console=SWITCH_TRUE, const char * * err=0x003bf68c)  Line 1454 + 0x11 bytes C
  FreeSwitch.exe!main(int argc=1, char * * argv=0x02144c40)  Line 764 + 0x23 bytes C
  FreeSwitch.exe!__tmainCRTStartup()  Line 586 + 0x19 bytes C
  FreeSwitch.exe!mainCRTStartup()  Line 403 C
  kernel32.dll!77713677()  
  ntdll.dll!77c39d72()  
  ntdll.dll!77c39d45()  


Error occurs in :

SWITCH_DECLARE(switch_status_t) switch_core_mime_add_type(const char *type, const char *ext)
{
 const char *check;
 switch_status_t status = SWITCH_STATUS_FALSE;

 switch_assert(type);
 switch_assert(ext);

 check = (const char *) switch_core_hash_find(runtime.mime_types, ext);

 if (!check) {
  char *ptype = switch_core_permanent_strdup(type);
  char *ext_list = strdup(ext);
  int argc = 0;
  char *argv[20] = { 0 };
  int x;

  switch_assert(ext_list);

  if ((argc = switch_separate_string(ext_list, ' ', argv, (sizeof(argv) / sizeof(argv[0]))))) {

   for (x = 0; x < argc; x++) {
    if (argv[x] && ptype) {
     switch_core_hash_insert(runtime.mime_types, argv[x], ptype);
    }
   }

   status = SWITCH_STATUS_SUCCESS;
  }

  free(ext_list);  // <--- HERE
 }

 return status;
}


  ----- Original Message ----- 
  From: mercutioviz [via freeswitch-users] 
  To: Maciej Aniserowicz 
  Sent: Monday, October 26, 2009 10:32 PM
  Subject: Re: [Freeswitch-users] "Can not record session. Media not enabled on channel."





  On Mon, Oct 26, 2009 at 12:21 AM, Maciej Aniserowicz <[hidden email]> wrote:


    Yes, I can confirm - this exact error occurs each time when I start recording
    before the call is answered (just after sending ORIGINATE command) - but I
    think that's completely understandable that media is not ready for an
    unanswered call.
    But... is there any other event that guarantees media to be ready?



  Update to latest SVN and try again.
  -MC



  _______________________________________________ 
  FreeSWITCH-users mailing list 
  [hidden email] 
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org



------------------------------------------------------------------------------

  View message @ http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3895104.html 
  To unsubscribe from Re: "Can not record session. Media not enabled on channel.", click here. 


-- 
View this message in context: http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3899478.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
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From rupa at rupa.com  Tue Oct 27 08:18:21 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 27 Oct 2009 10:18:21 -0500
Subject: [Freeswitch-users] reload from managed
In-Reply-To: <f8af5740910270232k689e8856sbc0dc8b668d9eacf@mail.gmail.com>
References: <f8af5740910270232k689e8856sbc0dc8b668d9eacf@mail.gmail.com>
Message-ID: <b39273530910270818j1e7f8c2eqdce44c3abdc72cc7@mail.gmail.com>

Is it necessary for you to call the reload function directly?  Why not
make an api call using switch_api_execute and the command "reloadxml".


On Tue, Oct 27, 2009 at 4:32 AM, srinivasula reddy
<srinivas.ksvreddy at gmail.com> wrote:
> Hi,
>
> how can i call reload from managed code? and i called
> Csharp_switch_xml_open_root, this is working first time, then onwards it is
> not working.
> is this the correct function to call reload?
>
> --
> Srinivasula Reddy K
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa


From anthony.minessale at gmail.com  Tue Oct 27 08:18:22 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 27 Oct 2009 10:18:22 -0500
Subject: [Freeswitch-users] "Can not record session. Media not enabled
	on channel."
In-Reply-To: <1256655334215-3899478.post@n2.nabble.com>
References: <1256034659005-3857858.post@n2.nabble.com>
	<87f2f3b90910201746h43a31b32m5c163fd98373f915@mail.gmail.com>
	<1256283391890-3877285.post@n2.nabble.com>
	<87f2f3b90910231200v71e72661id4cc9f2b7fba5610@mail.gmail.com>
	<1256541677550-3890610.post@n2.nabble.com>
	<87f2f3b90910261422n6952fa26ya6a2a66452365146@mail.gmail.com>
	<1256655334215-3899478.post@n2.nabble.com>
Message-ID: <191c3a030910270818y79f436d9h8f6311c4671502c2@mail.gmail.com>

won't compile or won't run?
maybe you should try rebuilding it.


On Tue, Oct 27, 2009 at 9:55 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

> Sorry, trunk does not compile on win7, here are the details:
>
>
> rev.15247
>
> ---------------------------
> Microsoft Visual C++ Debug Library
> ---------------------------
> Debug Assertion Failed!
>
> Program: ...ev\Projects\External\FreeSWITCH\Original\Debug\FreeSwitch.exe
> File: f:\dd\vctools\crt_bld\self_x86\crt\src\dbgheap.c
> Line: 1317
>
> Expression: _CrtIsValidHeapPointer(pUserData)
>
> For information on how your program can cause an assertion
> failure, see the Visual C++ documentation on asserts.
>
> (Press Retry to debug the application)
> ---------------------------
> Abort   Retry   Ignore
> ---------------------------
>
> VS Call stack:
>
>   ntdll.dll!77ccfadc()
>   [Frames below may be incorrect and/or missing, no symbols loaded for
> ntdll.dll]
>   ntdll.dll!77c9272c()
>   ntdll.dll!77c5e1ef()
>   msvcr90d.dll!_free_dbg_nolock(void * pUserData=0x00664b88, int
> nBlockUse=1)  Line 1317 + 0x9 bytes C++
>   msvcr90d.dll!_free_dbg(void * pUserData=0x00664b88, int nBlockUse=1)
> Line 1258 + 0xd bytes C++
>   msvcr90d.dll!free(void * pUserData=0x00664b88)  Line 49 + 0xb bytes C++
> > FreeSwitch.dll!switch_core_mime_add_type(const char * type=0x003bcd1c,
> const char * ext=0x003bcd37)  Line 748 + 0xc bytes C
>   FreeSwitch.dll!load_mime_types()  Line 791 C
>   FreeSwitch.dll!switch_core_init(unsigned int flags=65, switch_bool_t
> console=SWITCH_TRUE, const char * * err=0x003bf68c)  Line 1244 C
>   FreeSwitch.dll!switch_core_init_and_modload(unsigned int flags=65,
> switch_bool_t console=SWITCH_TRUE, const char * * err=0x003bf68c)  Line 1454
> + 0x11 bytes C
>   FreeSwitch.exe!main(int argc=1, char * * argv=0x02144c40)  Line 764 +
> 0x23 bytes C
>   FreeSwitch.exe!__tmainCRTStartup()  Line 586 + 0x19 bytes C
>   FreeSwitch.exe!mainCRTStartup()  Line 403 C
>   kernel32.dll!77713677()
>   ntdll.dll!77c39d72()
>   ntdll.dll!77c39d45()
>
>
> Error occurs in :
>
> SWITCH_DECLARE(switch_status_t) switch_core_mime_add_type(const char *type,
> const char *ext)
> {
>  const char *check;
>  switch_status_t status = SWITCH_STATUS_FALSE;
>
>  switch_assert(type);
>  switch_assert(ext);
>
>  check = (const char *) switch_core_hash_find(runtime.mime_types, ext);
>
>  if (!check) {
>   char *ptype = switch_core_permanent_strdup(type);
>   char *ext_list = strdup(ext);
>   int argc = 0;
>   char *argv[20] = { 0 };
>   int x;
>
>   switch_assert(ext_list);
>
>   if ((argc = switch_separate_string(ext_list, ' ', argv, (sizeof(argv) /
> sizeof(argv[0]))))) {
>
>    for (x = 0; x < argc; x++) {
>     if (argv[x] && ptype) {
>      switch_core_hash_insert(runtime.mime_types, argv[x], ptype);
>     }
>    }
>
>    status = SWITCH_STATUS_SUCCESS;
>   }
>
>   free(ext_list);  // <--- HERE
>  }
>
>  return status;
> }
>
>
>
> ----- Original Message -----
> *From:* [hidden email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3899478&i=0>
> *To:* [hidden email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3899478&i=1>
> *Sent:* Monday, October 26, 2009 10:32 PM
> *Subject:* Re: [Freeswitch-users] "Can not record session. Media not
> enabled on channel."
>
>
>
> On Mon, Oct 26, 2009 at 12:21 AM, Maciej Aniserowicz <[hidden email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3895104&i=0>
> > wrote:
>
>>
>> Yes, I can confirm - this exact error occurs each time when I start
>> recording
>> before the call is answered (just after sending ORIGINATE command) - but I
>> think that's completely understandable that media is not ready for an
>> unanswered call.
>> But... is there any other event that guarantees media to be ready?
>>
>> Update to latest SVN and try again.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> [hidden email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3895104&i=1>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> ------------------------------
> View this message in context: Re: [Freeswitch-users] "Can not record
> session. Media not enabled on channel."<http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3899478.html>
>
> Sent from the freeswitch-users mailing list archive<http://n2.nabble.com/freeswitch-users-f2379917.html>at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From brian at freeswitch.org  Tue Oct 27 08:26:27 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 27 Oct 2009 10:26:27 -0500
Subject: [Freeswitch-users] Problem with inbound call answered but no
	sound
In-Reply-To: <018801ca56b2$1e63e0f0$5b2ba2d0$@com>
References: <016e01ca56ad$56362380$02a26a80$@com>
	<168575F8-6507-4AD2-B448-2D6FDA0152B6@freeswitch.org>
	<018801ca56b2$1e63e0f0$5b2ba2d0$@com>
Message-ID: <BACDC899-CE4F-42E1-B2A0-C0C0E31F8787@freeswitch.org>

The issue is you're not behind nat-pmp or upnp so it can't figure out  
what your public IP is... you'll have to fill that in manually or  
enable those options on your router if possible.

/b

On Oct 26, 2009, at 10:03 PM, Lars Zeb wrote:

> I haven?t changed anything since v15183, where it worked OK.
>
> In conf/sip_profiles/external:
>
>      <param name="ext-rtp-ip" value="auto-nat"/>
>      <param name="ext-sip-ip" value="auto-nat"/>
>
> And
>

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From ivan at myrvold.org  Tue Oct 27 08:52:10 2009
From: ivan at myrvold.org (Ivan C Myrvold)
Date: Tue, 27 Oct 2009 16:52:10 +0100
Subject: [Freeswitch-users] SIP provider with extern rtp server
In-Reply-To: <bf6923ed0910270658t61bb6687ne0b7d3ec5090e2ff@mail.gmail.com>
References: <A2D2E918-B494-4179-96BA-CA3A7717846C@myrvold.org>
	<bf6923ed0910270658t61bb6687ne0b7d3ec5090e2ff@mail.gmail.com>
Message-ID: <CB8008BD-4066-4B5B-9043-214CFE938CD8@myrvold.org>

The server is on a public IP, so there is no nat issue here.

I can also see the rtp messages on wireshark starting just after the  
183 Session Progress message on the server, but just in one direction,  
coming in to the server.
So it looks like Freeswitch is stopping the rtp.
Is this because the rtp originates from another ip than the  sip  
provider ip?

Ivan

Den 27. okt. 2009 kl. 14:58 skrev Eliot Gable:

> Make sure you let their media IPs through your firewall. Also, if you
> are behind a NAT, check you have things passing to the correct
> internal address.
>
> On Tue, Oct 27, 2009 at 2:46 AM, Ivan C Myrvold <ivan at myrvold.org>  
> wrote:
>> I have used a SIP provider for more than a year. A few days ago, he
>> said he was moving to a new server, and asked me to reconfigure. I
>> did, and everything seemed to work fine, until I did an outgoing call
>> to an external telephone. I found out I had no audio, in neither
>> direction. Incoming calls was working fine.
>>
>> My provider said that the rtp is not going through the sip server, as
>> it did earlier, but now through several other IP's.
>>
>> Do I have to do some special configuration to handle that?
>>
>> Ivan
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>
>
>
> -- 
> Eliot Gable
>
> "We do not inherit the Earth from our ancestors: we borrow it from our
> children." ~David Brower
>
> "I decided the words were too conservative for me. We're not borrowing
> from our children, we're stealing from them--and it's not even
> considered to be a crime." ~David Brower
>
> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
> live; not live to eat.) ~Marcus Tullius Cicero
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>



From larclap at yahoo.com  Tue Oct 27 08:58:25 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Tue, 27 Oct 2009 08:58:25 -0700
Subject: [Freeswitch-users] Problem with inbound call answered but
	no	sound
In-Reply-To: <BACDC899-CE4F-42E1-B2A0-C0C0E31F8787@freeswitch.org>
References: <016e01ca56ad$56362380$02a26a80$@com>	<168575F8-6507-4AD2-B448-2D6FDA0152B6@freeswitch.org>	<018801ca56b2$1e63e0f0$5b2ba2d0$@com>
	<BACDC899-CE4F-42E1-B2A0-C0C0E31F8787@freeswitch.org>
Message-ID: <00d601ca571e$519f7300$f4de5900$@com>

Thanks for the reply, Brian.

 

Did something in FS change between v15183 and v15225 to make this occur? I
ask because this same configuration worked OK in the earlier version.

 

Lars

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Tuesday, October 27, 2009 8:26 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Problem with inbound call answered but no
sound

 

The issue is you're not behind nat-pmp or upnp so it can't figure out what
your public IP is... you'll have to fill that in manually or enable those
options on your router if possible.

 

/b

 

On Oct 26, 2009, at 10:03 PM, Lars Zeb wrote:





I haven't changed anything since v15183, where it worked OK.

 

In conf/sip_profiles/external:

 

     <param name="ext-rtp-ip" value="auto-nat"/>

     <param name="ext-sip-ip" value="auto-nat"/>

 

And

 

 

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From brian at freeswitch.org  Tue Oct 27 09:20:40 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 27 Oct 2009 11:20:40 -0500
Subject: [Freeswitch-users] Problem with inbound call answered but no
	sound
In-Reply-To: <00d601ca571e$519f7300$f4de5900$@com>
References: <016e01ca56ad$56362380$02a26a80$@com>	<168575F8-6507-4AD2-B448-2D6FDA0152B6@freeswitch.org>	<018801ca56b2$1e63e0f0$5b2ba2d0$@com>
	<BACDC899-CE4F-42E1-B2A0-C0C0E31F8787@freeswitch.org>
	<00d601ca571e$519f7300$f4de5900$@com>
Message-ID: <6D6D0E8C-84D0-4160-8457-3B6045DFDBC6@freeswitch.org>

Let me review that and see I can't off hand think of anything that  
would cause that.
/b

On Oct 27, 2009, at 10:58 AM, Lars Zeb wrote:

> Thanks for the reply, Brian.
>
> Did something in FS change between v15183 and v15225 to make this  
> occur? I ask because this same configuration worked OK in the  
> earlier version.
>
> Lars

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From egable+freeswitch at gmail.com  Tue Oct 27 10:34:29 2009
From: egable+freeswitch at gmail.com (Eliot Gable)
Date: Tue, 27 Oct 2009 13:34:29 -0400
Subject: [Freeswitch-users] SIP provider with extern rtp server
In-Reply-To: <CB8008BD-4066-4B5B-9043-214CFE938CD8@myrvold.org>
References: <A2D2E918-B494-4179-96BA-CA3A7717846C@myrvold.org>
	<bf6923ed0910270658t61bb6687ne0b7d3ec5090e2ff@mail.gmail.com>
	<CB8008BD-4066-4B5B-9043-214CFE938CD8@myrvold.org>
Message-ID: <bf6923ed0910271034w6c7e0cc5t685a5377764755e6@mail.gmail.com>

No, the IP address the media originates from does not need to be tied
to the SIP IP address. Can you send a Wireshark capture taken on the
FreeSWITCH server of both call legs? Or, if you can, pastebin a debug
log from FreeSWITCH console with sofia loglevel set to 9 and siptrace
on for any Sofia SIP profiles involved.

On Tue, Oct 27, 2009 at 11:52 AM, Ivan C Myrvold <ivan at myrvold.org> wrote:
> The server is on a public IP, so there is no nat issue here.
>
> I can also see the rtp messages on wireshark starting just after the
> 183 Session Progress message on the server, but just in one direction,
> coming in to the server.
> So it looks like Freeswitch is stopping the rtp.
> Is this because the rtp originates from another ip than the ?sip
> provider ip?
>
> Ivan
>
> Den 27. okt. 2009 kl. 14:58 skrev Eliot Gable:
>
>> Make sure you let their media IPs through your firewall. Also, if you
>> are behind a NAT, check you have things passing to the correct
>> internal address.
>>
>> On Tue, Oct 27, 2009 at 2:46 AM, Ivan C Myrvold <ivan at myrvold.org>
>> wrote:
>>> I have used a SIP provider for more than a year. A few days ago, he
>>> said he was moving to a new server, and asked me to reconfigure. I
>>> did, and everything seemed to work fine, until I did an outgoing call
>>> to an external telephone. I found out I had no audio, in neither
>>> direction. Incoming calls was working fine.
>>>
>>> My provider said that the rtp is not going through the sip server, as
>>> it did earlier, but now through several other IP's.
>>>
>>> Do I have to do some special configuration to handle that?
>>>
>>> Ivan
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Eliot Gable
>>
>> "We do not inherit the Earth from our ancestors: we borrow it from our
>> children." ~David Brower
>>
>> "I decided the words were too conservative for me. We're not borrowing
>> from our children, we're stealing from them--and it's not even
>> considered to be a crime." ~David Brower
>>
>> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
>> live; not live to eat.) ~Marcus Tullius Cicero
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero


From larclap at yahoo.com  Tue Oct 27 10:33:18 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Tue, 27 Oct 2009 10:33:18 -0700
Subject: [Freeswitch-users] Forcing endpoint registration
Message-ID: <010201ca572b$984e02a0$c8ea07e0$@com>

Is the following the correct command to force registration of an endpoint
below which is not showing up in 'sofia status profile internal' currently,
but when it does it looks like the following?

 

Call-ID:        mjkg8c3zlqwjzfd.hrt5

User:           1019 at 192.168.10.29

Contact:        "user" <sip:1019 at 192.168.10.103;line=3852>

Agent:          snom-m3-SIP/02.02 (MAC=0004132A31C7; HW=1)

Status:         Registered(UDP)(unknown) EXP(2009-10-27 10:46:07)

Host:           fs

IP:             192.168.10.103

Port:           5060

Auth-User:      1019

Auth-Realm:     192.168.10.29

MWI-Account:    1019 at 192.168.10.29

 

sofia profile internal  flush_inbound_reg  <mailto:1019 at 192.168.10.29>
1019 at 192.168.10.29

 

Thanks Lars

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From msc at freeswitch.org  Tue Oct 27 10:56:09 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 27 Oct 2009 10:56:09 -0700
Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Agenda - Oct
	30th
Message-ID: <87f2f3b90910271056v672b60ebybc34cb040760dfc2@mail.gmail.com>

Greetings all!

I just wanted to say thank you to those who have been contributing to the
documentation and cleanup efforts. Please keep up the good work. FYI, we
still have lots of janitorial stuff to do. Just keep checking the latest
conf call agenda for updates. This week's agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_30

We are always looking for people to help out with documentation and other
little projects so please feel free to ask me if you would like to assist.
Also, we are working on having members of the community give brief
presentations during the weekly conference calls. We want people to join the
conference and share with the group information about how they're using
FreeSWITCH in production, or they can give little tutorials on how to use
various parts of FS. We are also working on getting a web-cast enabled so
that we can have a little video to go with the audio. If you have something
you'd like to discuss with group please let me know.

Thanks for being such a great community!
-Michael
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From brian at freeswitch.org  Tue Oct 27 11:06:09 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 27 Oct 2009 13:06:09 -0500
Subject: [Freeswitch-users] Forcing endpoint registration
In-Reply-To: <010201ca572b$984e02a0$c8ea07e0$@com>
References: <010201ca572b$984e02a0$c8ea07e0$@com>
Message-ID: <D87B950A-EB02-4CA1-A6E7-E8570BAE49C4@freeswitch.org>

 From FreeSWITCH you can't force and endpoint to register... thats the  
endpoints job... what you're doing is just flushing the reg out of  
FreeSWITCH so now it can't find the endpoint...  Try adding a reboot  
on there maybe i'll reboot the device and cause it to register again...

/b

On Oct 27, 2009, at 12:33 PM, Lars Zeb wrote:

> Is the following the correct command to force registration of an  
> endpoint below which is not showing up in ?sofia status profile  
> internal? currently, but when it does it looks like the following?
>
> Call-ID:        mjkg8c3zlqwjzfd.hrt5
> User:           1019 at 192.168.10.29
> Contact:        "user" <sip:1019 at 192.168.10.103;line=3852>
> Agent:          snom-m3-SIP/02.02 (MAC=0004132A31C7; HW=1)
> Status:         Registered(UDP)(unknown) EXP(2009-10-27 10:46:07)
> Host:           fs
> IP:             192.168.10.103
> Port:           5060
> Auth-User:      1019
> Auth-Realm:     192.168.10.29
> MWI-Account:    1019 at 192.168.10.29
>
> sofia profile internal  flush_inbound_reg 1019 at 192.168.10.29
>
> Thanks Lars

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From jerry.richards at teotech.com  Tue Oct 27 12:17:07 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 27 Oct 2009 12:17:07 -0700
Subject: [Freeswitch-users] FS Training
Message-ID: <5B56DAA56F1F43EC973C226E8CD9BCFC@greyhawk.tonecommander.com>


Did the voting booth close?  I was unable to vote.  I'm not sure what link
to click and I have had some strange issues with my FS account today.

I would be interested in paid training.  Do you have plans for offering a
training session at your locale?  Or would you travel onsite to provide
training?

Best Regards,
Jerry



From cliff at develix.com  Tue Oct 27 12:51:50 2009
From: cliff at develix.com (Cliff Wells)
Date: Tue, 27 Oct 2009 12:51:50 -0700
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <bf6923ed0910261501k1ff8c892rfc198c31e30ab286@mail.gmail.com>
References: <bf6923ed0910261446x6a184ecdnee2e73e904406246@mail.gmail.com>
	<bf6923ed0910261501k1ff8c892rfc198c31e30ab286@mail.gmail.com>
Message-ID: <1256673110.9746.29.camel@portable-evil>

A little off-topic, but since call-capacity is the subject, what are
people using to analyze their CDR's to discover this?   I'm handling
about 30k calls per day but have only a bandwidth-based guesstimate of
the peak number of concurrent calls I'm handling.

If there's an open source solution, I'd appreciate a pointer.

Regards,
Cliff

On Mon, 2009-10-26 at 18:01 -0400, Eliot Gable wrote:
> Although, FYI, I just benchmarked mod_xml_curl on a separate web app
> server from FS with FS on a Dell R710 with their current best
> processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and 32
> GB memory. The web app server is less than half the power of the R710.
> I maxed the web app server at 300 calls per second (both setting up
> and tearing down) and the R710 running FS was 65% idle. No audio was
> being proxied through FS, though. If I were running the web app server
> on an equivalent R710, they probably would have been on-par with each
> other in performance. Extrapolating, I expect that in such a case I
> should be able to get at least 650 CPS out of FS, though for
> production I would probably limit it to 400 CPS or less so I leave
> room for miscellaneous tasks. I maxed out the R710 at over 16,000
> simultaneous calls (again, no audio proxying) but the only reason I
> couldn't do more was because I hit some sort of thread creation limit
> in Linux. There was about 17 GB of memory used for this many calls.
> This should give you some ballpark idea of what you can accomplish
> with FS.
> 
> At some point, I will track down and resolve the thread creation
> issue, at which time I believe call limits will be limited either by a
> complex combination of available memory, the speed of the processor,
> the cost of thread context switching, calls per second setup rate, and
> call duration.
> 
> --
> Eliot Gable
> 
> > -----Original Message-----
> >
> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli
> >
> > Sent: Monday, October 26, 2009 4:56 PM
> >
> > To: freeswitch-users at lists.freeswitch.org
> >
> > Subject: Re: [Freeswitch-users] Estimating Call Capacity
> >
> >
> >
> > On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur
> >
> > <vinuth.madinur at gmail.com> wrote:
> >
> > > Here are a few benchmarks that I had stumbled upon.
> >
> > > http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2
> >
> >
> >
> > Please remember NO benchmarks are endorsed by the FS community or
> >
> > developers, because there are just too many variables, and a simple
> >
> > figure is just useful for marketing hype, not for real dimensioning.
> >
> >
> >
> > You MUST do your own benchmarking, so you get an idea about how to
> >
> > dimension for your own use case and hardware.
> >
> >
> >
> >
> >
> > > Thanks,
> >
> > > Vinuth.
> >
> > >
> >
> > > On Tue, Oct 27, 2009 at 1:43 AM, Brian West <brian at freeswitch.org> wrote:
> >
> > >>
> >
> > >> I highly doubt it... You can wait for someone to post their results
> >
> > >> but in the end you'll have to do your own load testing because not
> >
> > >> everyone's numbers will jive with your use case.  Which is the reason
> >
> > >> the project never posts or endorses a set call count.
> >
> > >>
> >
> > >> /b
> >
> > >>
> >
> > >> On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote:
> >
> > >>
> >
> > >> > Are there any benchmarking test results available publicly?
> >
> > >> > ________________________________________
> >
> > >>
> >
> > >>
> >
> > >> _______________________________________________
> >
> > >> FreeSWITCH-users mailing list
> >
> > >> FreeSWITCH-users at lists.freeswitch.org
> >
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > >> http://www.freeswitch.org
> >
> > >
> >
> > >
> >
> > > _______________________________________________
> >
> > > FreeSWITCH-users mailing list
> >
> > > FreeSWITCH-users at lists.freeswitch.org
> >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > > http://www.freeswitch.org
> >
> > >
> >
> > >
> >
> >
> >
> >
> >
> >
> >
> > --
> >
> > Sincerely,
> >
> >
> >
> > Giovanni Maruzzelli
> >
> > Cell : +39-347-2665618
> >
> >
> >
> > _______________________________________________
> >
> > FreeSWITCH-users mailing list
> >
> > FreeSWITCH-users at lists.freeswitch.org
> >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 
http://www.google.com/search?q=vonage+sucks



From damin at nacs.net  Tue Oct 27 13:06:32 2009
From: damin at nacs.net (Gregory Boehnlein)
Date: Tue, 27 Oct 2009 16:06:32 -0400
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <1256673110.9746.29.camel@portable-evil>
References: <bf6923ed0910261446x6a184ecdnee2e73e904406246@mail.gmail.com>	<bf6923ed0910261501k1ff8c892rfc198c31e30ab286@mail.gmail.com>
	<1256673110.9746.29.camel@portable-evil>
Message-ID: <000f01ca5740$fd9d7a40$f8d86ec0$@net>

I'm fond of Vqmanager from ManagEngine. It is a passive SIP monitor. I.E.
you mirror the ports that your FS or Asterisk boxes, and VQmanager sniffs
the mirrors, tracking all sorts of good data.

You can install it on a Centos box, and get a free trial.

http://www.manageengine.com/products/vqmanager/index.html

What is really cool is that it actually monitors the RTP/RTCP as well as all
of the SIP headers and archives the calls, so you can look at calls from
several days ago and see EXACTLY what happened on them. I have used this
extensively to pinpoint bad Level 3 and X/O media gateways.. Much better
than trying to sniff packets in real-time and MAYBE catch a problem..

I've also used it to find/fix several SIP issues w/ odd endpoints.. Very
easy to see..

> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> users-bounces at lists.freeswitch.org] On Behalf Of Cliff Wells
> Sent: Tuesday, October 27, 2009 3:52 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Estimating Call Capacity
> 
> A little off-topic, but since call-capacity is the subject, what are
> people using to analyze their CDR's to discover this?   I'm handling
> about 30k calls per day but have only a bandwidth-based guesstimate of
> the peak number of concurrent calls I'm handling.
> 
> If there's an open source solution, I'd appreciate a pointer.
> 
> Regards,
> Cliff
> 
> On Mon, 2009-10-26 at 18:01 -0400, Eliot Gable wrote:
> > Although, FYI, I just benchmarked mod_xml_curl on a separate web app
> > server from FS with FS on a Dell R710 with their current best
> > processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and
> 32
> > GB memory. The web app server is less than half the power of the
> R710.
> > I maxed the web app server at 300 calls per second (both setting up
> > and tearing down) and the R710 running FS was 65% idle. No audio was
> > being proxied through FS, though. If I were running the web app
> server
> > on an equivalent R710, they probably would have been on-par with each
> > other in performance. Extrapolating, I expect that in such a case I
> > should be able to get at least 650 CPS out of FS, though for
> > production I would probably limit it to 400 CPS or less so I leave
> > room for miscellaneous tasks. I maxed out the R710 at over 16,000
> > simultaneous calls (again, no audio proxying) but the only reason I
> > couldn't do more was because I hit some sort of thread creation limit
> > in Linux. There was about 17 GB of memory used for this many calls.
> > This should give you some ballpark idea of what you can accomplish
> > with FS.
> >
> > At some point, I will track down and resolve the thread creation
> > issue, at which time I believe call limits will be limited either by
> a
> > complex combination of available memory, the speed of the processor,
> > the cost of thread context switching, calls per second setup rate,
> and
> > call duration.
> >
> > --
> > Eliot Gable
> >
> > > -----Original Message-----
> > >
> > > From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Giovanni Maruzzelli
> > >
> > > Sent: Monday, October 26, 2009 4:56 PM
> > >
> > > To: freeswitch-users at lists.freeswitch.org
> > >
> > > Subject: Re: [Freeswitch-users] Estimating Call Capacity
> > >
> > >
> > >
> > > On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur
> > >
> > > <vinuth.madinur at gmail.com> wrote:
> > >
> > > > Here are a few benchmarks that I had stumbled upon.
> > >
> > > >
> http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+
> x2200+M2
> > >
> > >
> > >
> > > Please remember NO benchmarks are endorsed by the FS community or
> > >
> > > developers, because there are just too many variables, and a simple
> > >
> > > figure is just useful for marketing hype, not for real
> dimensioning.
> > >
> > >
> > >
> > > You MUST do your own benchmarking, so you get an idea about how to
> > >
> > > dimension for your own use case and hardware.
> > >
> > >
> > >
> > >
> > >
> > > > Thanks,
> > >
> > > > Vinuth.
> > >
> > > >
> > >
> > > > On Tue, Oct 27, 2009 at 1:43 AM, Brian West
> <brian at freeswitch.org> wrote:
> > >
> > > >>
> > >
> > > >> I highly doubt it... You can wait for someone to post their
> results
> > >
> > > >> but in the end you'll have to do your own load testing because
> not
> > >
> > > >> everyone's numbers will jive with your use case.  Which is the
> reason
> > >
> > > >> the project never posts or endorses a set call count.
> > >
> > > >>
> > >
> > > >> /b
> > >
> > > >>
> > >
> > > >> On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote:
> > >
> > > >>
> > >
> > > >> > Are there any benchmarking test results available publicly?
> > >
> > > >> > ________________________________________
> > >
> > > >>
> > >
> > > >>
> > >
> > > >> _______________________________________________
> > >
> > > >> FreeSWITCH-users mailing list
> > >
> > > >> FreeSWITCH-users at lists.freeswitch.org
> > >
> > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > > >>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> users
> > >
> > > >> http://www.freeswitch.org
> > >
> > > >
> > >
> > > >
> > >
> > > > _______________________________________________
> > >
> > > > FreeSWITCH-users mailing list
> > >
> > > > FreeSWITCH-users at lists.freeswitch.org
> > >
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > > >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> users
> > >
> > > > http://www.freeswitch.org
> > >
> > > >
> > >
> > > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > > --
> > >
> > > Sincerely,
> > >
> > >
> > >
> > > Giovanni Maruzzelli
> > >
> > > Cell : +39-347-2665618
> > >
> > >
> > >
> > > _______________________________________________
> > >
> > > FreeSWITCH-users mailing list
> > >
> > > FreeSWITCH-users at lists.freeswitch.org
> > >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> users
> > >
> > > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> users
> > http://www.freeswitch.org
> --
> http://www.google.com/search?q=vonage+sucks
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> users
> http://www.freeswitch.org
> 
> --
> This message has been scanned for viruses and
> dangerous content by N2Net Mailshield, and is
> believed to be clean.



From sicfslist at gmail.com  Tue Oct 27 13:19:15 2009
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Tue, 27 Oct 2009 15:19:15 -0500
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <1256673110.9746.29.camel@portable-evil>
References: <bf6923ed0910261446x6a184ecdnee2e73e904406246@mail.gmail.com>	<bf6923ed0910261501k1ff8c892rfc198c31e30ab286@mail.gmail.com>
	<1256673110.9746.29.camel@portable-evil>
Message-ID: <4AE755C3.6060307@gmail.com>

Cliff,

Try using xml_rpc ... status or show channels will give you what you need.

SDR

Cliff Wells wrote:
> A little off-topic, but since call-capacity is the subject, what are
> people using to analyze their CDR's to discover this?   I'm handling
> about 30k calls per day but have only a bandwidth-based guesstimate of
> the peak number of concurrent calls I'm handling.
>
> If there's an open source solution, I'd appreciate a pointer.
>
> Regards,
> Cliff
>
> On Mon, 2009-10-26 at 18:01 -0400, Eliot Gable wrote:
>   
>> Although, FYI, I just benchmarked mod_xml_curl on a separate web app
>> server from FS with FS on a Dell R710 with their current best
>> processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and 32
>> GB memory. The web app server is less than half the power of the R710.
>> I maxed the web app server at 300 calls per second (both setting up
>> and tearing down) and the R710 running FS was 65% idle. No audio was
>> being proxied through FS, though. If I were running the web app server
>> on an equivalent R710, they probably would have been on-par with each
>> other in performance. Extrapolating, I expect that in such a case I
>> should be able to get at least 650 CPS out of FS, though for
>> production I would probably limit it to 400 CPS or less so I leave
>> room for miscellaneous tasks. I maxed out the R710 at over 16,000
>> simultaneous calls (again, no audio proxying) but the only reason I
>> couldn't do more was because I hit some sort of thread creation limit
>> in Linux. There was about 17 GB of memory used for this many calls.
>> This should give you some ballpark idea of what you can accomplish
>> with FS.
>>
>> At some point, I will track down and resolve the thread creation
>> issue, at which time I believe call limits will be limited either by a
>> complex combination of available memory, the speed of the processor,
>> the cost of thread context switching, calls per second setup rate, and
>> call duration.
>>
>> --
>> Eliot Gable
>>
>>     
>>> -----Original Message-----
>>>
>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli
>>>
>>> Sent: Monday, October 26, 2009 4:56 PM
>>>
>>> To: freeswitch-users at lists.freeswitch.org
>>>
>>> Subject: Re: [Freeswitch-users] Estimating Call Capacity
>>>
>>>
>>>
>>> On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur
>>>
>>> <vinuth.madinur at gmail.com> wrote:
>>>
>>>       
>>>> Here are a few benchmarks that I had stumbled upon.
>>>>         
>>>> http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2
>>>>         
>>>
>>> Please remember NO benchmarks are endorsed by the FS community or
>>>
>>> developers, because there are just too many variables, and a simple
>>>
>>> figure is just useful for marketing hype, not for real dimensioning.
>>>
>>>
>>>
>>> You MUST do your own benchmarking, so you get an idea about how to
>>>
>>> dimension for your own use case and hardware.
>>>
>>>
>>>
>>>
>>>
>>>       
>>>> Thanks,
>>>>         
>>>> Vinuth.
>>>>         
>>>> On Tue, Oct 27, 2009 at 1:43 AM, Brian West <brian at freeswitch.org> wrote:
>>>>         
>>>>> I highly doubt it... You can wait for someone to post their results
>>>>>           
>>>>> but in the end you'll have to do your own load testing because not
>>>>>           
>>>>> everyone's numbers will jive with your use case.  Which is the reason
>>>>>           
>>>>> the project never posts or endorses a set call count.
>>>>>           
>>>>> /b
>>>>>           
>>>>> On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote:
>>>>>           
>>>>>> Are there any benchmarking test results available publicly?
>>>>>>             
>>>>>> ________________________________________
>>>>>>             
>>>>> _______________________________________________
>>>>>           
>>>>> FreeSWITCH-users mailing list
>>>>>           
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>           
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>           
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>           
>>>>> http://www.freeswitch.org
>>>>>           
>>>> _______________________________________________
>>>>         
>>>> FreeSWITCH-users mailing list
>>>>         
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>         
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>         
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>         
>>>> http://www.freeswitch.org
>>>>         
>>>
>>>
>>>
>>>
>>>
>>> --
>>>
>>> Sincerely,
>>>
>>>
>>>
>>> Giovanni Maruzzelli
>>>
>>> Cell : +39-347-2665618
>>>
>>>
>>>
>>> _______________________________________________
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> http://www.freeswitch.org
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     



From msc at freeswitch.org  Tue Oct 27 13:39:30 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 27 Oct 2009 13:39:30 -0700
Subject: [Freeswitch-users] FS Training
In-Reply-To: <5B56DAA56F1F43EC973C226E8CD9BCFC@greyhawk.tonecommander.com>
References: <5B56DAA56F1F43EC973C226E8CD9BCFC@greyhawk.tonecommander.com>
Message-ID: <87f2f3b90910271339p5153fb00r59a53fb753b6cdbc@mail.gmail.com>

On Tue, Oct 27, 2009 at 12:17 PM, Jerry Richards <jerry.richards at teotech.com
> wrote:

>
> Did the voting booth close?  I was unable to vote.  I'm not sure what link
> to click and I have had some strange issues with my FS account today.
>
> I would be interested in paid training.  Do you have plans for offering a
> training session at your locale?  Or would you travel onsite to provide
> training?
>
> Jerry,

Yes the poll closed last Thursday. I'm glad to hear that you are interested
in paid training. We are still evaluating all the options. I will be
contacting people off list to discuss specifics. This is still in the
incubation stage, so no ETA on anything just yet.

-MC
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From mike at jerris.com  Tue Oct 27 14:31:26 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 27 Oct 2009 17:31:26 -0400
Subject: [Freeswitch-users] switch_xml_open_root
In-Reply-To: <f8af5740910270548n3fe7883fp18f126afd421d89e@mail.gmail.com>
References: <f8af5740910270548n3fe7883fp18f126afd421d89e@mail.gmail.com>
Message-ID: <86D18B9D-CE41-44A2-9CE1-B0EE00BE141C@jerris.com>

You probably should not be calling that function, what are you trying  
to do?

Mike

On Oct 27, 2009, at 8:48 AM, srinivasula reddy wrote:

>
>
> Hi,
>
> when i am calling  switch_xml_open_root(1,err) . i am getting this  
> warning message.
> HEAP[FreeSwitch.exe]: Invalid Address specified to RtlFreeHeap 
> ( 00160000, 100E9FD0 ). can any know. please help me.

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From msc at freeswitch.org  Tue Oct 27 16:34:43 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 27 Oct 2009 16:34:43 -0700
Subject: [Freeswitch-users] switch_xml_open_root
In-Reply-To: <86D18B9D-CE41-44A2-9CE1-B0EE00BE141C@jerris.com>
References: <f8af5740910270548n3fe7883fp18f126afd421d89e@mail.gmail.com>
	<86D18B9D-CE41-44A2-9CE1-B0EE00BE141C@jerris.com>
Message-ID: <87f2f3b90910271634i639ee11bta144542ff72a4718@mail.gmail.com>

On Tue, Oct 27, 2009 at 2:31 PM, Michael Jerris <mike at jerris.com> wrote:

> You probably should not be calling that function, what are you trying to
> do?
>
> Mike
>
> On Oct 27, 2009, at 8:48 AM, srinivasula reddy wrote:
>

You might also want to move this conversation to the -dev list because it's
a little intense for the -users list.
-MC
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From bottleman at icf.org.ru  Tue Oct 27 17:16:57 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Wed, 28 Oct 2009 03:16:57 +0300 (MSK)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910230614m2b2504d0r92006b78d267a399@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>
	<65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
	<Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
	<Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>
	<65d96fc80910230452s6b4a179aqa5bc41dc34efc403@mail.gmail.com>
	<Pine.LNX.4.64.0910231610220.21492@srv.icf.org.ru>
	<65d96fc80910230538w7105be6fve14956ab3b247bf6@mail.gmail.com>
	<Pine.LNX.4.64.0910231646210.29582@srv.icf.org.ru>
	<65d96fc80910230614m2b2504d0r92006b78d267a399@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910280313001.31425@srv.icf.org.ru>

On 2009-10-23 15:14 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:


TC>
TC> so the real solution is to implement a check for CallProceeding , Progress
TC>and Facility message whether it has a faststart element included. It it is
TC>true than you might start pre_answer.
TC>
TC>
TC>also, i don't see any handling of Call Proceeding ... what if there is a
TC>fastStart element in CallProceeding message? :)

Handling of fastStart in CallProceeding is commented out in h323plus library,
this is exploration from h323plus developers about this:


Yes that should be mera.

The problem is that Callproceeding does not always come from the remote it
may be generated by the gatekeeper. MERA where sending fast start elements
in the Call proceeding and connect. The call proceeding where not valid and
causing the media to fail. Normally (although valid) EP's do not set Fast
Start in Call proceeding so the code was disabled to resolve the MERA issue.

if you wont read "bugs" file in mod_h323, there is explaned how to enable it.

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From lei.tlfly at gmail.com  Tue Oct 27 18:14:54 2009
From: lei.tlfly at gmail.com (Lei Tang)
Date: Wed, 28 Oct 2009 09:14:54 +0800
Subject: [Freeswitch-users] how to config FS with two net interface?
In-Reply-To: <bf6923ed0910270657n22fb6115n6314398006612e25@mail.gmail.com>
References: <50c41b4e0910270149k2eaf8ee8v78ef24776947f270@mail.gmail.com>
	<bf6923ed0910270657n22fb6115n6314398006612e25@mail.gmail.com>
Message-ID: <50c41b4e0910271814h162d7878lac592351d0eefc7a@mail.gmail.com>

Thanks Eliot, It works.

2009/10/27 Eliot Gable
<egable+freeswitch at gmail.com<egable%2Bfreeswitch at gmail.com>
>

> Try setting ext-rtp-ip and ext-sip-ip on both profiles.
>
> On Tue, Oct 27, 2009 at 4:49 AM, Lei Tang <lei.tlfly at gmail.com> wrote:
> > Hi all, I run FS on a machine with two net interface, each interface has
> a
> > ip addr, one of the them connect to public network(has ip addr A), the
> > other  connect to a private network(has ip addr B), FS server as a SIP
> > server for public through A, all outbound call will bridge to a
> softswitch
> > in private network through B. here is my sofia config file and diaplan
> > config:
> >
> > sofia internal.xml
> > ....
> > <param name="rtp-ip" value="A"/>
> > <param name="sip-ip" value="A"/>
> >  ....
> >
> > sofia external.xml
> > ....
> > <param name="rtp-ip" value="B"/>
> > <param name="sip-ip" value="B"/>
> > ....
> >
> > dialplan
> > ......
> > <extension name="OUTBOUND">
> >     <condition field="destination_number" expression="^(\d+)$">
> >         <action application="set" data="hangup_after_bridge=true"/>
> >         <action application="set"
> >
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION"/>
> >         <action application="set"
> > data="effective_caller_id_number=xxxxxxx"/>  <!--here change the caller
> > number -->
> >         <action application="bridge"
> > data="sofia/external/${destination_number}@xxxxx"/>
> >       </condition>
> >     </extension>
> > .....
> >
> > then call seq is
> > sipAgent --> [internal -->(bridge)-->external] -->softswith
> >                           FREESWITCH
> >
> > the question is, when sipAgent make a outbound call, FS can't recevie the
> > caller's up audio stream, I traced the SIP packets, found that FS has
> return
> > addr B in SDP when ack the invite request from sipAgent, the ack packet
> is
> > ===============
> > SIP/2.0 183 Session Progress
> > Via: SIP/2.0/UDP
> >
> xxxxx:12208;branch=z9hG4bK-d8754z-dc750d57652c7c51-1---d8754z-;rport=12208
> > From: "1000" <sip:xxxx at A>;tag=cb4d3c4e
> > To: "65960581" <sip:xxxx at A>;tag=DtvSc0QX01yKN
> > Call-ID: ZTI2NmIwZGZiYzlhOGNkNTdiYWUzMzkzZTMwYzgxZjI.
> > CSeq: 2 INVITE
> > Contact: <sip:xxxxxx at B:5060;transport=udp>
> > User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460
> > Accept: application/sdp
> > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY,
> > REFER, UPDATE, REGISTER, INFO, PUBLISH
> > Supported: timer, precondition, path, replaces
> > Allow-Events: talk, presence, dialog, call-info, sla,
> > include-session-description, presence.winfo, message-summary, refer
> > Content-Type: application/sdp
> > Content-Disposition: session
> > Content-Length: 245
> >
> > v=0
> > o=FreeSWITCH 1256598185 1256598186 IN IP4 B   ;>>>>wrong this is the ip
> addr
> > of the adapter connect to the private network
> > s=FreeSWITCH
> > c=IN IP4 B ;>>>>wrong this is the ip addr of the adapter connect to the
> > private network
> > t=0 0
> > m=audio 31066 RTP/AVP 0 101
> > a=rtpmap:0 PCMU/8000
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-16
> > a=silenceSupp:off - - - -
> > a=ptime:20
> > ================
> > I think FS should return A in SDP, not the external binding addr (B),
> does
> > somebody known how to solve this problem?
> >
> > --
> > Lei.Tang
> > lei.tlfly at gmail.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Eliot Gable
>
> "We do not inherit the Earth from our ancestors: we borrow it from our
> children." ~David Brower
>
> "I decided the words were too conservative for me. We're not borrowing
> from our children, we're stealing from them--and it's not even
> considered to be a crime." ~David Brower
>
> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
> live; not live to eat.) ~Marcus Tullius Cicero
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Lei.Tang
lei.tlfly at gmail.com
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From brian at freeswitch.org  Tue Oct 27 18:25:00 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 27 Oct 2009 20:25:00 -0500
Subject: [Freeswitch-users] how to config FS with two net interface?
In-Reply-To: <50c41b4e0910271814h162d7878lac592351d0eefc7a@mail.gmail.com>
References: <50c41b4e0910270149k2eaf8ee8v78ef24776947f270@mail.gmail.com>
	<bf6923ed0910270657n22fb6115n6314398006612e25@mail.gmail.com>
	<50c41b4e0910271814h162d7878lac592351d0eefc7a@mail.gmail.com>
Message-ID: <B1381E00-7962-4498-8ACC-3369F310A6E6@freeswitch.org>

you should have just needed to set the rtp-ip and the sip-ip on both  
profiles to their values and it would have worked fine... their would  
have been no need to set the ext-*-ip equiv.

/b

On Oct 27, 2009, at 8:14 PM, Lei Tang wrote:

> Thanks Eliot, It works.



From lei.tlfly at gmail.com  Tue Oct 27 18:34:56 2009
From: lei.tlfly at gmail.com (Lei Tang)
Date: Wed, 28 Oct 2009 09:34:56 +0800
Subject: [Freeswitch-users] how to config FS with two net interface?
In-Reply-To: <B1381E00-7962-4498-8ACC-3369F310A6E6@freeswitch.org>
References: <50c41b4e0910270149k2eaf8ee8v78ef24776947f270@mail.gmail.com>
	<bf6923ed0910270657n22fb6115n6314398006612e25@mail.gmail.com>
	<50c41b4e0910271814h162d7878lac592351d0eefc7a@mail.gmail.com>
	<B1381E00-7962-4498-8ACC-3369F310A6E6@freeswitch.org>
Message-ID: <50c41b4e0910271834w1e0bf668yaf03f2b6179266ef@mail.gmail.com>

Hi Brian, It doesn't work if I only set rtp-ip and sip-ip. when  I set the
ext-rtp-ip, it works fine.

2009/10/28 Brian West <brian at freeswitch.org>

> you should have just needed to set the rtp-ip and the sip-ip on both
> profiles to their values and it would have worked fine... their would
> have been no need to set the ext-*-ip equiv.
>
> /b
>
> On Oct 27, 2009, at 8:14 PM, Lei Tang wrote:
>
> > Thanks Eliot, It works.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Lei.Tang
lei.tlfly at gmail.com
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From brian at freeswitch.org  Tue Oct 27 19:07:00 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 27 Oct 2009 21:07:00 -0500
Subject: [Freeswitch-users] how to config FS with two net interface?
In-Reply-To: <50c41b4e0910271834w1e0bf668yaf03f2b6179266ef@mail.gmail.com>
References: <50c41b4e0910270149k2eaf8ee8v78ef24776947f270@mail.gmail.com>
	<bf6923ed0910270657n22fb6115n6314398006612e25@mail.gmail.com>
	<50c41b4e0910271814h162d7878lac592351d0eefc7a@mail.gmail.com>
	<B1381E00-7962-4498-8ACC-3369F310A6E6@freeswitch.org>
	<50c41b4e0910271834w1e0bf668yaf03f2b6179266ef@mail.gmail.com>
Message-ID: <DE3384C4-FB19-47E7-A784-CB0D8F58D376@freeswitch.org>

Ok then something is broken badly... and that makes NO sense.  Because  
it works on my box.

/b

On Oct 27, 2009, at 8:34 PM, Lei Tang wrote:

> Hi Brian, It doesn't work if I only set rtp-ip and sip-ip. when  I  
> set the ext-rtp-ip, it works fine.
>
> 2009/10/28 Brian West <brian at freeswitch.org>
> you should have just needed to set the rtp-ip and the sip-ip on both
> profiles to their values and it would have worked fine... their would
> have been no need to set the ext-*-ip equiv.
>
> /b
>
> On Oct 27, 2009, at 8:14 PM, Lei Tang wrote:
>
> > Thanks Eliot, It works.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
> -- 
> Lei.Tang
> lei.tlfly at gmail.com
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From srinivas.ksvreddy at gmail.com  Tue Oct 27 21:37:05 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Wed, 28 Oct 2009 10:07:05 +0530
Subject: [Freeswitch-users] switch_xml_open_root
In-Reply-To: <86D18B9D-CE41-44A2-9CE1-B0EE00BE141C@jerris.com>
References: <f8af5740910270548n3fe7883fp18f126afd421d89e@mail.gmail.com>
	<86D18B9D-CE41-44A2-9CE1-B0EE00BE141C@jerris.com>
Message-ID: <f8af5740910272137t22fa559arc4b562a232634627@mail.gmail.com>

Hi mike,

thank for your reply.
i am trying to call that function from swig.cs.   its working fine first
time with the warning information, then onwards it is not working.
is there other way can i call reload function from freeswitch.managed code.
any help

srinivas



On Wed, Oct 28, 2009 at 3:01 AM, Michael Jerris <mike at jerris.com> wrote:

> You probably should not be calling that function, what are you trying to
> do?
>
> Mike
>
> On Oct 27, 2009, at 8:48 AM, srinivasula reddy wrote:
>
> *
> *
> Hi,
>
> when i am calling  switch_xml_open_root(1,err) . i am getting this warning
> message.
> HEAP[FreeSwitch.exe]: Invalid Address specified to RtlFreeHeap( 00160000,
> 100E9FD0 ). can any know. please help me.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Srinivasula Reddy K
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From lon at kickasspixels.com  Tue Oct 27 21:51:40 2009
From: lon at kickasspixels.com (Lon Baker)
Date: Tue, 27 Oct 2009 21:51:40 -0700
Subject: [Freeswitch-users] Retrieve conference member state using cli?
Message-ID: <5d3e0dc60910272151o13f7fad9k35c4f9f61cb1b287@mail.gmail.com>

Just like you can call conference list for conferences, is there a way to
retrieve the profile state of a conference member using the cli or xml/rpc?

Lon
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From mike at jerris.com  Tue Oct 27 22:21:32 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 28 Oct 2009 01:21:32 -0400
Subject: [Freeswitch-users] switch_xml_open_root
In-Reply-To: <f8af5740910272137t22fa559arc4b562a232634627@mail.gmail.com>
References: <f8af5740910270548n3fe7883fp18f126afd421d89e@mail.gmail.com>
	<86D18B9D-CE41-44A2-9CE1-B0EE00BE141C@jerris.com>
	<f8af5740910272137t22fa559arc4b562a232634627@mail.gmail.com>
Message-ID: <2B91DB5F-A48C-4ECC-A13F-FABF3CAED1BB@jerris.com>

just api execute reloadxml

Mike

On Oct 28, 2009, at 12:37 AM, srinivasula reddy wrote:

> Hi mike,
>
> thank for your reply.
> i am trying to call that function from swig.cs.   its working fine  
> first time with the warning information, then onwards it is not  
> working.
> is there other way can i call reload function from  
> freeswitch.managed code. any help
>


From srinivas.ksvreddy at gmail.com  Tue Oct 27 22:34:49 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Wed, 28 Oct 2009 11:04:49 +0530
Subject: [Freeswitch-users] reload from managed
In-Reply-To: <b39273530910270818j1e7f8c2eqdce44c3abdc72cc7@mail.gmail.com>
References: <f8af5740910270232k689e8856sbc0dc8b668d9eacf@mail.gmail.com>
	<b39273530910270818j1e7f8c2eqdce44c3abdc72cc7@mail.gmail.com>
Message-ID: <f8af5740910272234i5e7a7f94vff1d6a00a24352f3@mail.gmail.com>

i tried switch_api_execute its working fine, thanks rupa

On Tue, Oct 27, 2009 at 8:48 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> Is it necessary for you to call the reload function directly?  Why not
> make an api call using switch_api_execute and the command "reloadxml".
>
>
> On Tue, Oct 27, 2009 at 4:32 AM, srinivasula reddy
> <srinivas.ksvreddy at gmail.com> wrote:
> > Hi,
> >
> > how can i call reload from managed code? and i called
> > Csharp_switch_xml_open_root, this is working first time, then onwards it
> is
> > not working.
> > is this the correct function to call reload?
> >
> > --
> > Srinivasula Reddy K
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Srinivasula Reddy K
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From velu.technical at gmail.com  Wed Oct 28 01:29:38 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Wed, 28 Oct 2009 13:59:38 +0530
Subject: [Freeswitch-users] How to stop the playback files
Message-ID: <1452e2980910280129wab717eam4ae9da2554ab2e1@mail.gmail.com>

Dear All,
     I have played the list of voice files in playback like the following by
using ESL perl module,

      $conn->execute("set","playback_delimiter=!");
      $conn->execute("set","playback_sleep_val=100");
 $conn->playback($sound_path."ivr/ivr-welcome_to_freeswitch.wav!ivr/ivr-this_ivr_will_let_you_test_features.wav!ivr/ivr-you_may_exit_by_hanging_up.wav");

 In a loop I am checking the DTMF event, if that event comes I should stop
the above palyback. How can I do it?

Regards,
Velusamy.
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From mayamatakeshi at gmail.com  Wed Oct 28 01:33:18 2009
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Wed, 28 Oct 2009 17:33:18 +0900
Subject: [Freeswitch-users] Channel variables not being set when FS calls
	user
Message-ID: <15b9404e0910280133x27563aa4jc578a2092450c702@mail.gmail.com>

Hello,
in this page
http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide
we can read:

=============================
Variables:
Any variables defined in the domain or user will be defined as channel
variables when there is a call to user or when there is an inbound calls
from that user.
=============================

I can see the channel variables are set when there is an incoming call from
the user, but not when FS sends a call to the user.
Can someone confirm what is the expected behavior?

br,
takeshi
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From brian at freeswitch.org  Wed Oct 28 01:55:13 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 28 Oct 2009 03:55:13 -0500
Subject: [Freeswitch-users] Retrieve conference member state using cli?
In-Reply-To: <5d3e0dc60910272151o13f7fad9k35c4f9f61cb1b287@mail.gmail.com>
References: <5d3e0dc60910272151o13f7fad9k35c4f9f61cb1b287@mail.gmail.com>
Message-ID: <41933E56-8B6A-4D54-9989-D69BA6566515@freeswitch.org>

Type "conference" at the cli and the help will be displayed... same  
can be called over XML RPC.

/b

On Oct 27, 2009, at 11:51 PM, Lon Baker wrote:

> Just like you can call conference list for conferences, is there a  
> way to retrieve the profile state of a conference member using the  
> cli or xml/rpc?
>
> Lon



From brian at freeswitch.org  Wed Oct 28 01:56:43 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 28 Oct 2009 03:56:43 -0500
Subject: [Freeswitch-users] How to stop the playback files
In-Reply-To: <1452e2980910280129wab717eam4ae9da2554ab2e1@mail.gmail.com>
References: <1452e2980910280129wab717eam4ae9da2554ab2e1@mail.gmail.com>
Message-ID: <98BA0749-A312-4565-80A7-49B0BA48C1F0@freeswitch.org>

http://wiki.freeswitch.org/wiki/Channel_Variables#playback_terminators

Are you using sync or async socket?

/b

On Oct 28, 2009, at 3:29 AM, velusamy velu wrote:

> Dear All,
>      I have played the list of voice files in playback like the  
> following by using ESL perl module,
>
>       $conn->execute("set","playback_delimiter=!");
>       $conn->execute("set","playback_sleep_val=100");
>  $conn->playback($sound_path."ivr/ivr-welcome_to_freeswitch.wav!ivr/ 
> ivr-this_ivr_will_let_you_test_features.wav!ivr/ivr- 
> you_may_exit_by_hanging_up.wav");
>
>  In a loop I am checking the DTMF event, if that event comes I  
> should stop the above palyback. How can I do it?
>
> Regards,
> Velusamy.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From brian at freeswitch.org  Wed Oct 28 01:57:39 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 28 Oct 2009 03:57:39 -0500
Subject: [Freeswitch-users] Channel variables not being set when FS
	calls user
In-Reply-To: <15b9404e0910280133x27563aa4jc578a2092450c702@mail.gmail.com>
References: <15b9404e0910280133x27563aa4jc578a2092450c702@mail.gmail.com>
Message-ID: <D3006963-53C3-45E2-B67A-5B06B4AC4EBB@freeswitch.org>

Yes it is the expected behavior... if you wish to set them on an  
inbound call TO the user you'll use the set_user api to do so.  Its  
kinda like sudo in FreeSWITCH it'll load up all the variables for that  
user.

/b

On Oct 28, 2009, at 3:33 AM, mayamatakeshi wrote:

> I can see the channel variables are set when there is an incoming  
> call from the user, but not when FS sends a call to the user.
> Can someone confirm what is the expected behavior?



From srinivas.ksvreddy at gmail.com  Wed Oct 28 02:07:27 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Wed, 28 Oct 2009 14:37:27 +0530
Subject: [Freeswitch-users] db
Message-ID: <f8af5740910280207o4fdcaed2i7403d3f4bdff2fb2@mail.gmail.com>

Hi,

can i use sqlserver instead of sqllite. and can two freeswitch servers can
share same database(sqllite or sqlserver). any help would be great.

Thanks
Srinivasula Reddy K
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From mayamatakeshi at gmail.com  Wed Oct 28 02:23:22 2009
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Wed, 28 Oct 2009 18:23:22 +0900
Subject: [Freeswitch-users] Channel variables not being set when FS
	calls user
In-Reply-To: <D3006963-53C3-45E2-B67A-5B06B4AC4EBB@freeswitch.org>
References: <15b9404e0910280133x27563aa4jc578a2092450c702@mail.gmail.com>
	<D3006963-53C3-45E2-B67A-5B06B4AC4EBB@freeswitch.org>
Message-ID: <15b9404e0910280223s59183346me4783584b32148f8@mail.gmail.com>

Got it. Thanks.

On Wed, Oct 28, 2009 at 5:57 PM, Brian West <brian at freeswitch.org> wrote:

> Yes it is the expected behavior... if you wish to set them on an
> inbound call TO the user you'll use the set_user api to do so.  Its
> kinda like sudo in FreeSWITCH it'll load up all the variables for that
> user.
>
> /b
>
> On Oct 28, 2009, at 3:33 AM, mayamatakeshi wrote:
>
> > I can see the channel variables are set when there is an incoming
> > call from the user, but not when FS sends a call to the user.
> > Can someone confirm what is the expected behavior?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tculjaga at gmail.com  Wed Oct 28 02:24:25 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 28 Oct 2009 10:24:25 +0100
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910280313001.31425@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
	<Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>
	<65d96fc80910230452s6b4a179aqa5bc41dc34efc403@mail.gmail.com>
	<Pine.LNX.4.64.0910231610220.21492@srv.icf.org.ru>
	<65d96fc80910230538w7105be6fve14956ab3b247bf6@mail.gmail.com>
	<Pine.LNX.4.64.0910231646210.29582@srv.icf.org.ru>
	<65d96fc80910230614m2b2504d0r92006b78d267a399@mail.gmail.com>
	<Pine.LNX.4.64.0910280313001.31425@srv.icf.org.ru>
Message-ID: <65d96fc80910280224w55dd0acasfaac3ffe603e4bd4@mail.gmail.com>

>
>
>
> Handling of fastStart in CallProceeding is commented out in h323plus
> library,
> this is exploration from h323plus developers about this:
>
>
> Yes that should be mera.
>
> The problem is that Callproceeding does not always come from the remote it
> may be generated by the gatekeeper.


this is a feature .. called force_callproceeding. It means MERA will send a
provisional CallProceeding in order not to timeout on calls that don't
respond with that message on time. If this message contains a faststart
element it is certanly a bug and it has to be reported to them.


> MERA where sending fast start elements
> in the Call proceeding and connect. The call proceeding where not valid and
> causing the media to fail.


well if there is a correct faststart element within a connect message (or
alerting or facility or progress), the originator should adjust the media
resources accordingly. Here what could went wrong is just the media before
the next faststart element in the row.


> Normally (although valid) EP's do not set Fast
> Start in Call proceeding so the code was disabled to resolve the MERA
> issue.
>
>
well, this is unlikely as fast start element can be included in call
proceeding message. The developer's task is to determine whether a call
proceeding message is to be trusted or not.
Also, provisional call proceeding messages don't have faststart element
included! There are equipment (Cisco PGW / HSI) that are sending call
proceeding with faststart element and h245Controll (OLC + TCS/MSD) that has
to be treated correctly. Unfortunately, just disabling handling of
callproceeding faststart element is not a real option...



> if you wont read "bugs" file in mod_h323, there is explaned how to enable
> it.
>


of course i can enable it during build time but this will not solve interop
issues later we can encounter...




Do you maybe have some sniffs/traces of the wrong call proceeding message ?



...anyhow this is the expected behaviour when a GK/Proxy sends a provisional
Call Proceding to the terminator and later it receives the real Call
Proceeding carring faststart and h245Controll element within.

Entities in the signalling path shall also use the Facility message or the
Progress message to convey
any new information (such as Q.931 information elements, CallProceeding-UUIE
fields, tunnelled
non-H.323 protocols, and encapsulated H.245 messages) received in a Call
Proceeding message to
the other endpoint if the entity has already sent a Call Proceeding message.
This will allow the
entity, for example, to transmit the fastStart element to facilitate proper
establishment of a Fast
Connect call and/or a Progress Indicator to indicate the presence of in-band
tones and
announcements. When using the Facility message to carrying such information
extracted from the
Call Proceeding message, the reason in the Facility should be set to
forwardedElements.



in other words:

ORIG                          GK                                TERM
-------------------------------------------------------------------------------------------------------------
Setup OLC                     =>
Call proceeding (prov)        <=
                              Setup OLC                         =>
                              Call proceeding (OLC+TCS/MSD)     <=
Facility (OLC+TCS/MSD)        <=


<----------- normal call establishment scenario follows ----------->
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From velu.technical at gmail.com  Wed Oct 28 02:29:45 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Wed, 28 Oct 2009 14:59:45 +0530
Subject: [Freeswitch-users] How to stop the playback files
In-Reply-To: <98BA0749-A312-4565-80A7-49B0BA48C1F0@freeswitch.org>
References: <1452e2980910280129wab717eam4ae9da2554ab2e1@mail.gmail.com>
	<98BA0749-A312-4565-80A7-49B0BA48C1F0@freeswitch.org>
Message-ID: <1452e2980910280229h7d30bf84lb5143dc46eea95@mail.gmail.com>

I am using async socket. Shall I set the regular expression for
palyback_terminators?

On Wed, Oct 28, 2009 at 2:26 PM, Brian West <brian at freeswitch.org> wrote:

> http://wiki.freeswitch.org/wiki/Channel_Variables#playback_terminators
>
> Are you using sync or async socket?
>
> /b
>
> On Oct 28, 2009, at 3:29 AM, velusamy velu wrote:
>
> > Dear All,
> >      I have played the list of voice files in playback like the
> > following by using ESL perl module,
> >
> >       $conn->execute("set","playback_delimiter=!");
> >       $conn->execute("set","playback_sleep_val=100");
> >  $conn->playback($sound_path."ivr/ivr-welcome_to_freeswitch.wav!ivr/
> > ivr-this_ivr_will_let_you_test_features.wav!ivr/ivr-
> > you_may_exit_by_hanging_up.wav");
> >
> >  In a loop I am checking the DTMF event, if that event comes I
> > should stop the above palyback. How can I do it?
> >
> > Regards,
> > Velusamy.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Wed Oct 28 02:53:27 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 28 Oct 2009 04:53:27 -0500
Subject: [Freeswitch-users] db
In-Reply-To: <f8af5740910280207o4fdcaed2i7403d3f4bdff2fb2@mail.gmail.com>
References: <f8af5740910280207o4fdcaed2i7403d3f4bdff2fb2@mail.gmail.com>
Message-ID: <F760E917-11D5-428B-A1F0-31E9F3351C82@freeswitch.org>

http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core

Please don't cross post if possible.

/b

On Oct 28, 2009, at 4:07 AM, srinivasula reddy wrote:

> Hi,
>
> can i use sqlserver instead of sqllite. and can two freeswitch  
> servers can share same database(sqllite or sqlserver). any help  
> would be great.
>
> Thanks
> Srinivasula Reddy K
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From delianspam at gmail.com  Wed Oct 28 03:54:39 2009
From: delianspam at gmail.com (Delian Tashev)
Date: Wed, 28 Oct 2009 12:54:39 +0200
Subject: [Freeswitch-users] How to stop the playback files
In-Reply-To: <004801ca57b6$bd14c5c0$373e5140$@net>
References: <1452e2980910280129wab717eam4ae9da2554ab2e1@mail.gmail.com>	<98BA0749-A312-4565-80A7-49B0BA48C1F0@freeswitch.org>
	<004801ca57b6$bd14c5c0$373e5140$@net>
Message-ID: <005301ca57bd$0cce1bb0$266a5310$@com>

Hello Velusamy!

You may also be interested in using the method:
http://wiki.freeswitch.org/wiki/Mod_lua#session:playAndGetDigits
It play a file, gets DTMFs and can be terminated if the maximum number of
digits is reached (i.e. 1 digit).

Delian Tashev

-----Original Message-----
From: Brian West [mailto:brian at freeswitch.org] 
Sent: Wednesday, October 28, 2009 10:57
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] How to stop the playback files

http://wiki.freeswitch.org/wiki/Channel_Variables#playback_terminators

Are you using sync or async socket?

/b

On Oct 28, 2009, at 3:29 AM, velusamy velu wrote:

> Dear All,
>      I have played the list of voice files in playback like the  
> following by using ESL perl module,
>
>       $conn->execute("set","playback_delimiter=!");
>       $conn->execute("set","playback_sleep_val=100");
>  $conn->playback($sound_path."ivr/ivr-welcome_to_freeswitch.wav!ivr/ 
> ivr-this_ivr_will_let_you_test_features.wav!ivr/ivr- 
> you_may_exit_by_hanging_up.wav");
>
>  In a loop I am checking the DTMF event, if that event comes I  
> should stop the above palyback. How can I do it?
>
> Regards,
> Velusamy.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org








From brian at freeswitch.org  Wed Oct 28 07:13:29 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 28 Oct 2009 09:13:29 -0500
Subject: [Freeswitch-users] Problem with inbound call answered but no
	sound
In-Reply-To: <00d601ca571e$519f7300$f4de5900$@com>
References: <016e01ca56ad$56362380$02a26a80$@com>	<168575F8-6507-4AD2-B448-2D6FDA0152B6@freeswitch.org>	<018801ca56b2$1e63e0f0$5b2ba2d0$@com>
	<BACDC899-CE4F-42E1-B2A0-C0C0E31F8787@freeswitch.org>
	<00d601ca571e$519f7300$f4de5900$@com>
Message-ID: <8D1A994D-2DCD-4016-9BEF-D6FBD3FBAA1E@freeswitch.org>

What kind of router are you behind?

/b

On Oct 27, 2009, at 10:58 AM, Lars Zeb wrote:

> Thanks for the reply, Brian.
>
> Did something in FS change between v15183 and v15225 to make this  
> occur? I ask because this same configuration worked OK in the  
> earlier version.
>
> Lars
>

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From ivan at myrvold.org  Wed Oct 28 07:37:30 2009
From: ivan at myrvold.org (Ivan C Myrvold)
Date: Wed, 28 Oct 2009 15:37:30 +0100
Subject: [Freeswitch-users] SIP provider with extern rtp server
In-Reply-To: <bf6923ed0910271034w6c7e0cc5t685a5377764755e6@mail.gmail.com>
References: <A2D2E918-B494-4179-96BA-CA3A7717846C@myrvold.org>
	<bf6923ed0910270658t61bb6687ne0b7d3ec5090e2ff@mail.gmail.com>
	<CB8008BD-4066-4B5B-9043-214CFE938CD8@myrvold.org>
	<bf6923ed0910271034w6c7e0cc5t685a5377764755e6@mail.gmail.com>
Message-ID: <FA2C693B-FCE1-4BDD-BCE5-302AC699CEDB@myrvold.org>

Here is a debug log from a call from an internal phone out to an  
external (my iPhone with nbr 91316356):
http://pastebin.freeswitch.org/108578

Ivan

Den 27. okt. 2009 kl. 18:34 skrev Eliot Gable:

> No, the IP address the media originates from does not need to be tied
> to the SIP IP address. Can you send a Wireshark capture taken on the
> FreeSWITCH server of both call legs? Or, if you can, pastebin a debug
> log from FreeSWITCH console with sofia loglevel set to 9 and siptrace
> on for any Sofia SIP profiles involved.
>
> On Tue, Oct 27, 2009 at 11:52 AM, Ivan C Myrvold <ivan at myrvold.org>  
> wrote:
>> The server is on a public IP, so there is no nat issue here.
>>
>> I can also see the rtp messages on wireshark starting just after the
>> 183 Session Progress message on the server, but just in one  
>> direction,
>> coming in to the server.
>> So it looks like Freeswitch is stopping the rtp.
>> Is this because the rtp originates from another ip than the  sip
>> provider ip?
>>
>> Ivan
>>
>> Den 27. okt. 2009 kl. 14:58 skrev Eliot Gable:
>>
>>> Make sure you let their media IPs through your firewall. Also, if  
>>> you
>>> are behind a NAT, check you have things passing to the correct
>>> internal address.
>>>
>>> On Tue, Oct 27, 2009 at 2:46 AM, Ivan C Myrvold <ivan at myrvold.org>
>>> wrote:
>>>> I have used a SIP provider for more than a year. A few days ago, he
>>>> said he was moving to a new server, and asked me to reconfigure. I
>>>> did, and everything seemed to work fine, until I did an outgoing  
>>>> call
>>>> to an external telephone. I found out I had no audio, in neither
>>>> direction. Incoming calls was working fine.
>>>>
>>>> My provider said that the rtp is not going through the sip  
>>>> server, as
>>>> it did earlier, but now through several other IP's.
>>>>
>>>> Do I have to do some special configuration to handle that?
>>>>
>>>> Ivan
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>> users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Eliot Gable
>>>
>>> "We do not inherit the Earth from our ancestors: we borrow it from  
>>> our
>>> children." ~David Brower
>>>
>>> "I decided the words were too conservative for me. We're not  
>>> borrowing
>>> from our children, we're stealing from them--and it's not even
>>> considered to be a crime." ~David Brower
>>>
>>> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
>>> live; not live to eat.) ~Marcus Tullius Cicero
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>
>
>
> -- 
> Eliot Gable
>
> "We do not inherit the Earth from our ancestors: we borrow it from our
> children." ~David Brower
>
> "I decided the words were too conservative for me. We're not borrowing
> from our children, we're stealing from them--and it's not even
> considered to be a crime." ~David Brower
>
> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
> live; not live to eat.) ~Marcus Tullius Cicero
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>



From brian at freeswitch.org  Wed Oct 28 07:52:08 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 28 Oct 2009 09:52:08 -0500
Subject: [Freeswitch-users] [Freeswitch-dev] Build Issue
In-Reply-To: <3181A30B8C35AB4AA8577B78DDF4613805FCA7DE@nickel.mettonigroup.com>
References: <3181A30B8C35AB4AA8577B78DDF4613805FCA7DE@nickel.mettonigroup.com>
Message-ID: <8B9A5FFA-6940-4A43-983D-B8C8E954EAFE@freeswitch.org>

Clean and build again... the issue was files.freeswitch.org was having  
a little drama so I suspect it didn't download everything properly?

/b

On Oct 28, 2009, at 9:40 AM, Suneel Papineni wrote:

> Hi,
>
> Just now I have downloaded SVN version of FreeSwitch from http://svn.freeswitch.org/svn/freeswitch/trunk/ 
> .
>
> Tried to build the solution (Freeswitch.2008.sln ) using, but it  
> thrown many fatal errors(C1083)  and some warnings. Please find  
> below the list of errors and warnings.
>
> Could anyone let me know where I made mistake and source to get  
> these missing files.
>
> Thanks & Regards
> Suneel
>
>
> Warning               1              warning C4018: '<' : signed/ 
> unsigned mismatch                d:\Office\Projects\Freeswitch\libs 
> \pcre\pcre_ucp_searchfuncs.c           158         libpcre
> Warning               2              warning C4018: '<=' : signed/ 
> unsigned mismatch                d:\Office\Projects\Freeswitch\libs 
> \pcre\pcre_ucp_searchfuncs.c           163         libpcre
> Warning               3              Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> libteletone
> Error      4              fatal error C1083: Cannot open source  
> file: '..\..\pthreads-w32-2-7-0-release\pthread.c': No such file or  
> directory             c1            pthread
> Warning               5              Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_spidermonkey
> Error      6              fatal error C1083: Cannot open source  
> file: '..\..\mpg123\src\libmpg123\dct64.c': No such file or  
> directory                c1            libmpg123
> Error      7              fatal error C1083: Cannot open source  
> file: '..\..\mpg123\src\libmpg123\decode.c': No such file or  
> directory             c1            libmpg123
> Error      8              fatal error C1083: Cannot open source  
> file: '..\..\mpg123\src\libmpg123\decode_2to1.c': No such file or  
> directory             c1            libmpg123
> Error      9              fatal error C1083: Cannot open source  
> file: '..\..\mpg123\src\libmpg123\decode_4to1.c': No such file or  
> directory             c1            libmpg123
> Error      10           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\decode_ntom.c': No such file or  
> directory             c1            libmpg123
> Error      11           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\equalizer.c': No such file or  
> directory             c1            libmpg123
> Error      12           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\format.c': No such file or  
> directory                c1            libmpg123
> Error      13           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\frame.c': No such file or  
> directory                c1            libmpg123
> Error      14           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\icy.c': No such file or  
> directory                c1            libmpg123
> Error      15           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\icy2utf8.c': No such file or  
> directory             c1            libmpg123
> Error      16           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\id3.c': No such file or  
> directory                c1            libmpg123
> Error      17           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\layer1.c': No such file or  
> directory                c1            libmpg123
> Error      18           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\layer2.c': No such file or  
> directory                c1            libmpg123
> Error      19           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\layer3.c': No such file or  
> directory                c1            libmpg123
> Error      20           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\libmpg123.c': No such file or  
> directory             c1            libmpg123
> Error      21           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\optimize.c': No such file or  
> directory             c1            libmpg123
> Error      22           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\parse.c': No such file or  
> directory                c1            libmpg123
> Error      23           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\readers.c': No such file or  
> directory             c1            libmpg123
> Error      24           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\stringbuf.c': No such file or  
> directory             c1            libmpg123
> Error      25           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\tabinit.c': No such file or  
> directory                c1            libmpg123
> Error      26           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\compat.c': No such file or  
> directory             c1            libmpg123
> Error      27           fatal error C1083: Cannot open source file:  
> '..\..\libshout-2.2.2\src\httpp\httpp.c': No such file or  
> directory                c1            libshout
> Error      28           fatal error C1083: Cannot open source file:  
> '..\..\libshout-2.2.2\src\mp3.c': No such file or directory        
> c1                libshout
> Error      29           fatal error C1083: Cannot open source file:  
> '..\..\libshout-2.2.2\src\ogg.c': No such file or directory          
> c1                libshout
> Error      30           fatal error C1083: Cannot open source file:  
> '..\..\libshout-2.2.2\src\net\resolver.c': No such file or  
> directory             c1            libshout
> Error      31           fatal error C1083: Cannot open source file:  
> '..\..\libshout-2.2.2\src\shout.c': No such file or directory      
> c1                libshout
> Error      32           fatal error C1083: Cannot open source file:  
> '..\..\libshout-2.2.2\src\net\sock.c': No such file or  
> directory                c1            libshout
> Error      33           fatal error C1083: Cannot open source file:  
> '..\..\libshout-2.2.2\src\thread\thread.c': No such file or  
> directory             c1            libshout
> Error      34           fatal error C1083: Cannot open source file:  
> '..\..\libshout-2.2.2\src\timing\timing.c': No such file or  
> directory             c1            libshout
> Error      35           fatal error C1083: Cannot open source file:  
> '..\..\libshout-2.2.2\src\util.c': No such file or directory          
> c1                libshout
> Error      36           fatal error C1083: Cannot open source file:  
> '..\..\libshout-2.2.2\src\avl\avl.c': No such file or directory   
> c1                libshout
> Error      37           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\bitstream.c': No such file or  
> directory             c1            libmp3lame
> Error      38           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\version.c': No such file or  
> directory             c1            libmp3lame
> Error      39           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\VbrTag.c': No such file or  
> directory                c1            libmp3lame
> Error      40           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\vbrquantize.c': No such file or  
> directory             c1            libmp3lame
> Error      41           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\util.c': No such file or  
> directory                c1            libmp3lame
> Error      42           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\takehiro.c': No such file or  
> directory             c1            libmp3lame
> Error      43           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\set_get.c': No such file or  
> directory             c1            libmp3lame
> Error      44           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\reservoir.c': No such file or  
> directory             c1            libmp3lame
> Error      45           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\quantize_pvt.c': No such file or  
> directory             c1            libmp3lame
> Error      46           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\quantize.c': No such file or  
> directory             c1            libmp3lame
> Error      47           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\psymodel.c': No such file or  
> directory             c1            libmp3lame
> Error      48           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\presets.c': No such file or  
> directory             c1            libmp3lame
> Error      49           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\newmdct.c': No such file or  
> directory             c1            libmp3lame
> Error      50           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\mpglib_interface.c': No such file or  
> directory        c1            libmp3lame
> Error      51           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\lame.c': No such file or  
> directory                c1            libmp3lame
> Error      52           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\id3tag.c': No such file or  
> directory                c1            libmp3lame
> Error      53           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\gain_analysis.c': No such file or  
> directory             c1            libmp3lame
> Error      54           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\fft.c': No such file or  
> directory                c1            libmp3lame
> Error      55           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\encoder.c': No such file or  
> directory             c1            libmp3lame
> Error      56           fatal error C1083: Cannot open include file:  
> 'pthread.h': No such file or directory                d:\Office 
> \Projects\Freeswitch\libs\sofia-sip\libsofia-sip-ua\su 
> \su_alloc_lock.c              40                libsofia_sip_ua_static
> Error      57           fatal error C1083: Cannot open include file:  
> 'pthread.h': No such file or directory                d:\office 
> \projects\freeswitch\libs\sofia-sip\libsofia-sip-ua\su 
> \su_port.h          547         libsofia_sip_ua_static
> Error      58           fatal error C1083: Cannot open include file:  
> 'pthread.h': No such file or directory                d:\Office 
> \Projects\Freeswitch\libs\sofia-sip\libsofia-sip-ua\su 
> \su_log.c            46           libsofia_sip_ua_static
> Error      59           fatal error C1083: Cannot open include file:  
> 'pthread.h': No such file or directory                d:\office 
> \projects\freeswitch\libs\sofia-sip\libsofia-sip-ua\su 
> \su_port.h          547         libsofia_sip_ua_static
> Error      60           fatal error C1083: Cannot open include file:  
> 'pthread.h': No such file or directory                d:\office 
> \projects\freeswitch\libs\sofia-sip\libsofia-sip-ua\su 
> \su_port.h          547         libsofia_sip_ua_static
> Error      61           fatal error C1083: Cannot open include file:  
> 'pthread.h': No such file or directory                d:\Office 
> \Projects\Freeswitch\libs\sofia-sip\libsofia-sip-ua\su 
> \su_root.c          46           libsofia_sip_ua_static
> Error      62           fatal error C1083: Cannot open include file:  
> 'pthread.h': No such file or directory                d:\office 
> \projects\freeswitch\libs\sofia-sip\libsofia-sip-ua\su 
> \su_port.h          547         libsofia_sip_ua_static
> Error      63           fatal error C1083: Cannot open include file:  
> 'pthread.h': No such file or directory                d:\office 
> \projects\freeswitch\libs\sofia-sip\libsofia-sip-ua\su 
> \su_port.h          547         libsofia_sip_ua_static
> Error      64           fatal error C1083: Cannot open include file:  
> 'pthread.h': No such file or directory                d:\Office 
> \Projects\Freeswitch\libs\sofia-sip\libsofia-sip-ua\su 
> \su_uniqueid.c                85                libsofia_sip_ua_static
> Error      65           fatal error C1083: Cannot open include file:  
> 'pthread.h': No such file or directory                d:\office 
> \projects\freeswitch\libs\sofia-sip\libsofia-sip-ua\su 
> \su_port.h          547         libsofia_sip_ua_static
> Error      66           error BK1506 : cannot open file '.\Debug 
> \su_alloc_lock.sbr': No such file or directory      
> BSCMAKE                libsofia_sip_ua_static
> Warning               67           Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             esl
> Error      68           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\audio\au_none.c': No such file or  
> directory                c1            flite
> Error      69           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\audio\au_streaming.c': No such file or  
> directory             c1            flite
> Error      70           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\audio\au_wince.c': No such file or  
> directory             c1            flite
> Error      71           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\audio\audio.c': No such file or  
> directory                c1            flite
> Error      72           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\audio\auserver.c': No such file or  
> directory                c1            flite
> Error      73           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmulex\cmu_lex.c': No such file or  
> directory             c1            flite
> Error      74           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmulex\cmu_lex_data.c': No such file or  
> directory             c1            flite
> Error      75           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmulex\cmu_lex_entries.c': No such file or  
> directory        c1            flite
> Error      76           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmulex\cmu_lts_model.c': No such file or  
> directory        c1            flite
> Error      77           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmulex\cmu_lts_rules.c': No such file or  
> directory             c1            flite
> Error      78           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmulex\cmu_postlex.c': No such file or  
> directory             c1            flite
> Error      79           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_time_awb\cmu_time_awb_cart.c': No such  
> file or directory               c1            flite
> Error      80           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_time_awb\cmu_time_awb_clunits.c': No  
> such file or directory              c1            flite
> Error      81           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_time_awb\cmu_time_awb_lex_entry.c': No  
> such file or directory        c1            flite
> Error      82           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_time_awb\cmu_time_awb_lpc.c': No such  
> file or directory               c1            flite
> Error      83           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_time_awb\cmu_time_awb_mcep.c': No such  
> file or directory               c1            flite
> Error      84           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_kal\cmu_us_kal.c': No such file or  
> directory        c1            flite
> Error      85           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_kal\cmu_us_kal_diphone.c': No such  
> file or directory      c1            flite
> Error      86           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_kal\cmu_us_kal_lpc.c': No such file  
> or directory                c1            flite
> Error      87           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_kal\cmu_us_kal_res.c': No such file  
> or directory                c1            flite
> Error      88           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_kal\cmu_us_kal_residx.c': No such  
> file or directory      c1            flite
> Error      89           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\utils\cst_alloc.c': No such file or  
> directory                c1            flite
> Error      90           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\utils\cst_args.c': No such file or  
> directory                c1            flite
> Error      91           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\stats\cst_cart.c': No such file or  
> directory                c1            flite
> Error      92           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\cg\cst_cg.c': No such file or directory     
> c1                flite
> Error      93           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\wavesynth\cst_clunits.c': No such file or  
> directory             c1            flite
> Error      94           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\wavesynth\cst_diphone.c': No such file or  
> directory             c1            flite
> Error      95           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\utils\cst_endian.c': No such file or  
> directory             c1            flite
> Error      96           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\utils\cst_error.c': No such file or  
> directory                c1            flite
> Error      97           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\utils\cst_features.c': No such file or  
> directory             c1            flite
> Error      98           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\hrg\cst_ffeature.c': No such file or  
> directory             c1            flite
> Error      99           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\synth\cst_ffeatures.c': No such file or  
> directory             c1            flite
> Error      100         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\utils\cst_file_stdio.c': No such file or  
> directory             c1            flite
> Error      101         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\hrg\cst_item.c': No such file or  
> directory                c1            flite
> Error      102         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\lexicon\cst_lexicon.c': No such file or  
> directory             c1            flite
> Error      103         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\speech\cst_lpcres.c': No such file or  
> directory             c1            flite
> Error      104         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\lexicon\cst_lts.c': No such file or  
> directory                c1            flite
> Error      105         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\lexicon\cst_lts_rewrites.c': No such file or  
> directory             c1            flite
> Error      106         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\cg\cst_mlpg.c': No such file or  
> directory                c1            flite
> Error      107         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\cg\cst_mlsa.c': No such file or  
> directory                c1            flite
> Error      108         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\utils\cst_mmap_win32.c': No such file or  
> directory             c1            flite
> Error      109         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\synth\cst_phoneset.c': No such file or  
> directory             c1            flite
> Error      110         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\wavesynth\cst_reflpc.c': No such file or  
> directory             c1            flite
> Error      111         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\regex\cst_regex.c': No such file or  
> directory             c1            flite
> Error      112         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\hrg\cst_rel_io.c': No such file or  
> directory                c1            flite
> Error      113         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\hrg\cst_relation.c': No such file or  
> directory             c1            flite
> Error      114         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\wavesynth\cst_sigpr.c': No such file or  
> directory             c1            flite
> Error      115         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\utils\cst_socket.c': No such file or  
> directory             c1            flite
> Error      116         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\stats\cst_ss.c': No such file or  
> directory                c1            flite
> Error      117         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\utils\cst_string.c': No such file or  
> directory                c1            flite
> Error      118         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\wavesynth\cst_sts.c': No such file or  
> directory             c1            flite
> Error      119         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\synth\cst_synth.c': No such file or  
> directory             c1            flite
> Error      120         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\utils\cst_tokenstream.c': No such file or  
> directory             c1            flite
> Error      121         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\speech\cst_track.c': No such file or  
> directory             c1            flite
> Error      122         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\speech\cst_track_io.c': No such file or  
> directory             c1            flite
> Error      123         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\wavesynth\cst_units.c': No such file or  
> directory             c1            flite
> Error      124         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\synth\cst_utt_utils.c': No such file or  
> directory             c1            flite
> Error      125         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\hrg\cst_utterance.c': No such file or  
> directory             c1            flite
> Error      126         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\utils\cst_val.c': No such file or  
> directory                c1            flite
> Error      127         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\utils\cst_val_const.c': No such file or  
> directory             c1            flite
> Error      128         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\utils\cst_val_user.c': No such file or  
> directory             c1            flite
> Error      129         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\cg\cst_vc.c': No such file or directory     
> c1                flite
> Error      130         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\stats\cst_viterbi.c': No such file or  
> directory             c1            flite
> Error      131         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\synth\cst_voice.c': No such file or  
> directory             c1            flite
> Error      132         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\speech\cst_wave.c': No such file or  
> directory             c1            flite
> Error      133         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\speech\cst_wave_io.c': No such file or  
> directory             c1            flite
> Error      134         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\speech\cst_wave_utils.c': No such file or  
> directory             c1            flite
> Error      135         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\synth\flite.c': No such file or  
> directory                c1            flite
> Error      136         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\speech\rateconv.c': No such file or  
> directory             c1            flite
> Error      137         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\regex\regexp.c': No such file or  
> directory                c1            flite
> Error      138         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\regex\regsub.c': No such file or  
> directory                c1            flite
> Error      139         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\usenglish\us_aswd.c': No such file or  
> directory             c1            flite
> Error      140         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\usenglish\us_dur_stats.c': No such file or  
> directory             c1            flite
> Error      141         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\usenglish\us_durz_cart.c': No such file or  
> directory             c1            flite
> Error      142         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\usenglish\us_expand.c': No such file or  
> directory             c1            flite
> Error      143         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\usenglish\us_f0_model.c': No such file or  
> directory             c1            flite
> Error      144         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\usenglish\us_f0lr.c': No such file or  
> directory             c1            flite
> Error      145         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\usenglish\us_ffeatures.c': No such file or  
> directory             c1            flite
> Error      146         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\usenglish\us_gpos.c': No such file or  
> directory             c1            flite
> Error      147         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\usenglish\us_int_accent_cart.c': No such  
> file or directory                c1            flite
> Error      148         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\usenglish\us_int_tone_cart.c': No such file  
> or directory                c1            flite
> Error      149         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\usenglish\us_nums_cart.c': No such file or  
> directory        c1            flite
> Error      150         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\usenglish\us_phoneset.c': No such file or  
> directory             c1            flite
> Error      151         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\usenglish\us_phrasing_cart.c': No such file  
> or directory                c1            flite
> Error      152         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\usenglish\us_text.c': No such file or  
> directory             c1            flite
> Error      153         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\usenglish\usenglish.c': No such file or  
> directory             c1            flite
> Error      154         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_slt\cmu_us_slt.c': No such file or  
> directory             c1            flite
> Error      155         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_slt\cmu_us_slt_cg.c': No such file  
> or directory                c1            flite
> Error      156         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_slt\cmu_us_slt_cg_durmodel.c': No  
> such file or directory               c1            flite
> Error      157         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_slt\cmu_us_slt_cg_f0_trees.c': No  
> such file or directory               c1            flite
> Error      158         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_slt\cmu_us_slt_cg_mcep_trees.c': No  
> such file or directory              c1            flite
> Error      159         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_slt\cmu_us_slt_cg_params.c': No such  
> file or directory      c1            flite
> Error      160         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_slt\cmu_us_slt_cg_phonestate.c': No  
> such file or directory               c1            flite
> Error      161         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_awb\cmu_us_awb.c': No such file or  
> directory                c1            flite
> Error      162         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_awb\cmu_us_awb_cg.c': No such file  
> or directory      c1            flite
> Error      163         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_awb\cmu_us_awb_cg_durmodel.c': No  
> such file or directory           c1            flite
> Error      164         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_awb\cmu_us_awb_cg_f0_trees.c': No  
> such file or directory              c1            flite
> Error      165         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_awb\cmu_us_awb_cg_mcep_trees.c': No  
> such file or directory       c1            flite
> Error      166         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_awb\cmu_us_awb_cg_params.c': No such  
> file or directory                c1            flite
> Error      167         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_awb\cmu_us_awb_cg_phonestate.c': No  
> such file or directory       c1            flite
> Error      168         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_rms\cmu_us_rms.c': No such file or  
> directory        c1            flite
> Error      169         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_rms\cmu_us_rms_cg.c': No such file  
> or directory                c1            flite
> Error      170         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_rms\cmu_us_rms_cg_durmodel.c': No  
> such file or directory             c1            flite
> Error      171         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_rms\cmu_us_rms_cg_f0_trees.c': No  
> such file or directory                c1            flite
> Error      172         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_rms\cmu_us_rms_cg_mcep_trees.c': No  
> such file or directory         c1            flite
> Error      173         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_rms\cmu_us_rms_cg_params.c': No such  
> file or directory               c1            flite
> Error      174         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_rms\cmu_us_rms_cg_phonestate.c': No  
> such file or directory         c1            flite
> Error      175         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\audio\au_command.c': No such file or  
> directory             c1            flite
> Warning               176         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_fax
> Warning               177         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_lcr
> Warning               178         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_pocketsphinx
> Warning               179         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_spidermonkey_curl
> Warning               180         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_fsv
> Warning               181         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_voipcodecs
> Warning               182         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_tone_stream
> Warning               183         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_cdr_csv
> Warning               184         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_logfile
> Warning               185         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_dialplan_asterisk
> Warning               186         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_expr
> Warning               187         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_fifo
> Warning               188         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_say_nl
> Warning               189         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_say_it
> Warning               190         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_say_fr
> Warning               191         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_say_es
> Warning               192         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_say_de
> Warning               193         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_voicemail
> Warning               194         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_spidermonkey_socket
> Warning               195         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_local_stream
> Warning               196         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_esf
> Warning               197         warning C4244: 'function' :  
> conversion from 'int' to 'uint8_t', possible loss of  
> data                d:\Office\Projects\Freeswitch\src\mod 
> \applications\mod_esf\mod_esf.c         139         mod_esf
> Warning               198         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_h26x
> Warning               199         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_amr
> Warning               200         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_xml_cdr
> Warning               201         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_say_en
> Warning               202         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_xml_curl
> Warning               203         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_spidermonkey_odbc
> Warning               204         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_enum
> Warning               205         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_easyroute
> Warning               206         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_spidermonkey_teletone
> Warning               207         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_spidermonkey_core_db
> Warning               208         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_native_file
> Warning               209         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_g723_1
> Warning               210         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             fs_cli
> Warning               211         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_siren
> Warning               212         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_sofia
> Warning               213         warning C4018: '>' : signed/ 
> unsigned mismatch                d:\Office\Projects\Freeswitch\src 
> \mod\endpoints\mod_sofia\sofia.c 2145       mod_sofia
> Warning               214         warning C4018: '<' : signed/ 
> unsigned mismatch                d:\Office\Projects\Freeswitch\src 
> \mod\endpoints\mod_sofia\sofia.c 2145       mod_sofia
> Warning               215         warning C4018: '>' : signed/ 
> unsigned mismatch                d:\Office\Projects\Freeswitch\src 
> \mod\endpoints\mod_sofia\sofia.c 2832       mod_sofia
> Warning               216         warning C4018: '<' : signed/ 
> unsigned mismatch                d:\Office\Projects\Freeswitch\src 
> \mod\endpoints\mod_sofia\sofia.c 2832       mod_sofia
> Warning               217         warning C4702: unreachable  
> code                d:\office\projects\freeswitch\src\mod\endpoints 
> \mod_sofia\sofia.c  4005       mod_sofia
> Error      218         fatal error LNK1181: cannot open input file  
> '..\..\..\..\libs\win32\sofia\debug 
> \libsofia_sip_ua_static.lib'                mod_sofia           
> mod_sofia
> Error      219         fatal error C1083: Cannot open include file:  
> 'os.h': No such file or directory                d:\Office\Projects 
> \Freeswitch\libs\win32\libshout\shout\shout.h        26            
> mod_shout
> Warning               220         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_snom
> Warning               221         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_say_zh
> Warning               222         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_managed
> Warning               223         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_vmd
> Warning               224         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_file_string
> Warning               225         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_nibblebill
> Warning               226         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_say_ru
> Warning               227         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_limit
> Warning               228         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_event_socket
> Warning               229         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_dptools
> Warning               230         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_conference
> Warning               231         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_rss
> Warning               232         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_xml_rpc
> Warning               233         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_console
> Warning               234         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_commands
> Warning               235         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_dingaling
> Warning               236         warning C4018: '<' : signed/ 
> unsigned mismatch                d:\Office\Projects\Freeswitch\src 
> \mod\endpoints\mod_dingaling\mod_dingaling.c   2326        
> mod_dingaling
> Warning               237         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_ilbc
> Warning               238         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_loopback
> Warning               239         warning C4189: 'code' : local  
> variable is initialized but not referenced                d:\Office 
> \Projects\Freeswitch\src\mod\languages\mod_lua\mod_lua.cpp         
> 393         mod_lua
> Warning               240         warning C4101: 'error' :  
> unreferenced local variable                d:\Office\Projects 
> \Freeswitch\src\mod\languages\mod_lua\mod_lua.cpp        395          
> mod_lua
> Warning               241         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_event_multicast
> Warning               242         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_zeroconf
> Warning               243         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_dialplan_directory
> Warning               244         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_ldap
> Warning               245         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_dialplan_xml
> Warning               246         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_speex
> Warning               247         warning LNK4075: ignoring '/ 
> EDITANDCONTINUE' due to '/INCREMENTAL:NO'  
> specification                libspeexdsp.lib                mod_speex
> Warning               248         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_PortAudio
> Warning               249         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_iax
> Warning               250         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_sndfile
> Warning               251         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_g729
> Warning               252         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_valet_parking
> Warning               253         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> FreeSwitchConsole
> *************************************************************************
> Please consider the environment before printing this e-mail
> *************************************************************************
> This email and any files transmitted with it are confidential and
> intended solely for the use of the individual or entity to whom they
> are addressed. If you have received this email in error please notify
> the system manager.  http://www.mettoni.com
>
> Mettoni Ltd
> Registered in England and Wales: 4485956
> 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
> *************************************************************************
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org

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From maciej.aniserowicz at gmail.com  Wed Oct 28 09:21:43 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Wed, 28 Oct 2009 09:21:43 -0700 (PDT)
Subject: [Freeswitch-users] "Can not record session. Media not enabled
 on channel."
In-Reply-To: <191c3a030910270818y79f436d9h8f6311c4671502c2@mail.gmail.com>
References: <1256034659005-3857858.post@n2.nabble.com>
	<87f2f3b90910201746h43a31b32m5c163fd98373f915@mail.gmail.com>
	<1256283391890-3877285.post@n2.nabble.com>
	<87f2f3b90910231200v71e72661id4cc9f2b7fba5610@mail.gmail.com>
	<1256541677550-3890610.post@n2.nabble.com>
	<87f2f3b90910261422n6952fa26ya6a2a66452365146@mail.gmail.com>
	<1256655334215-3899478.post@n2.nabble.com>
	<191c3a030910270818y79f436d9h8f6311c4671502c2@mail.gmail.com>
Message-ID: <1256746903296-3906568.post@n2.nabble.com>


Correct - compiled but did not run. Works fine now.

I'll see if the error shows up again and let you know if it does.
Thanks,
MA



Anthony Minessale wrote:
> 
> won't compile or won't run?
> maybe you should try rebuilding it.
> 
> 
> On Tue, Oct 27, 2009 at 9:55 AM, Maciej Aniserowicz <
> maciej.aniserowicz at gmail.com> wrote:
> 
>> Sorry, trunk does not compile on win7, here are the details:
>>
>>
>> rev.15247
>>
>> ---------------------------
>> Microsoft Visual C++ Debug Library
>> ---------------------------
>> Debug Assertion Failed!
>>
>> 
>>
>>
>>
>> ----- Original Message -----
>> *From:* [hidden
>> email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3899478&i=0>
>> *To:* [hidden
>> email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3899478&i=1>
>> *Sent:* Monday, October 26, 2009 10:32 PM
>> *Subject:* Re: [Freeswitch-users] "Can not record session. Media not
>> enabled on channel."
>>
>>
>>
>> On Mon, Oct 26, 2009 at 12:21 AM, Maciej Aniserowicz <[hidden
>> email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3895104&i=0>
>> > wrote:
>>
>>>
>>> Yes, I can confirm - this exact error occurs each time when I start
>>> recording
>>> before the call is answered (just after sending ORIGINATE command) - but
>>> I
>>> think that's completely understandable that media is not ready for an
>>> unanswered call.
>>> But... is there any other event that guarantees media to be ready?
>>>
>>> Update to latest SVN and try again.
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> [hidden
>> email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3895104&i=1>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> ------------------------------
>> View this message in context: Re: [Freeswitch-users] "Can not record
>> session. Media not enabled on
>> channel."<http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3899478.html>
>>
>> Sent from the freeswitch-users mailing list
>> archive<http://n2.nabble.com/freeswitch-users-f2379917.html>at
>> Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3906568.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From mayamatakeshi at gmail.com  Wed Oct 28 09:35:35 2009
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 29 Oct 2009 01:35:35 +0900
Subject: [Freeswitch-users] No RTP destination auto-correction
Message-ID: <15b9404e0910280935u16f08ff6nd8527d8c0b9f32a8@mail.gmail.com>

I'm seeing a strange issue with eyebeam behind NAT.
If I make a call from eyebeam to FS, I can see FS receives packets from
eyebeam's nat address (confirmed with tcpdump and mod_event_socket DTMF) but
it doesn't change the RTP destination to the source of those packets and
sends the packets to the private address advertised in the SDP.
However, if I do hold/unhold and send some digits, FS logs "Auto Changing
port from X to Y" and starts to send the packets to the correct address.
This always work after a hold/unhold + digits but never happen just after
the first INVITE transaction.
Also, I don't see the problem if eyebeam receives a call from FS.
Has anyone seen this?

br,
takeshi
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From mattdfong at gmail.com  Wed Oct 28 09:37:56 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Wed, 28 Oct 2009 23:37:56 +0700
Subject: [Freeswitch-users] Determining Frame Size
Message-ID: <4256bf830910280937n1f946518w423597fa6584c26f@mail.gmail.com>

I noticed that the wait_for_silence's <silence_hits> <listen_hits> use
number of frames as base values (rather than seconds or milliseconds). I'd
like to wait for 5 seconds of consistent silence, but I'm wondering how I
can determine the sample rate (and thus frames per second) for an existing
call. Is there a variable I can read? Thanks.

--matt
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From anthony.minessale at gmail.com  Wed Oct 28 09:43:31 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 28 Oct 2009 11:43:31 -0500
Subject: [Freeswitch-users] No RTP destination auto-correction
In-Reply-To: <15b9404e0910280935u16f08ff6nd8527d8c0b9f32a8@mail.gmail.com>
References: <15b9404e0910280935u16f08ff6nd8527d8c0b9f32a8@mail.gmail.com>
Message-ID: <191c3a030910280943h2998c349kd22843247106490@mail.gmail.com>

are you >= r15256 because an issue with this was fixed in that revision.



On Wed, Oct 28, 2009 at 11:35 AM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:

> I'm seeing a strange issue with eyebeam behind NAT.
> If I make a call from eyebeam to FS, I can see FS receives packets from
> eyebeam's nat address (confirmed with tcpdump and mod_event_socket DTMF) but
> it doesn't change the RTP destination to the source of those packets and
> sends the packets to the private address advertised in the SDP.
> However, if I do hold/unhold and send some digits, FS logs "Auto Changing
> port from X to Y" and starts to send the packets to the correct address.
> This always work after a hold/unhold + digits but never happen just after
> the first INVITE transaction.
> Also, I don't see the problem if eyebeam receives a call from FS.
> Has anyone seen this?
>
> br,
> takeshi
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From brian at freeswitch.org  Wed Oct 28 09:51:54 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 28 Oct 2009 11:51:54 -0500
Subject: [Freeswitch-users] No RTP destination auto-correction
In-Reply-To: <15b9404e0910280935u16f08ff6nd8527d8c0b9f32a8@mail.gmail.com>
References: <15b9404e0910280935u16f08ff6nd8527d8c0b9f32a8@mail.gmail.com>
Message-ID: <A08ECEE6-46AF-497A-81CC-D65CB6BC0D95@freeswitch.org>

have you updated to the latest SVN?

/b

On Oct 28, 2009, at 11:35 AM, mayamatakeshi wrote:

> I'm seeing a strange issue with eyebeam behind NAT.
> If I make a call from eyebeam to FS, I can see FS receives packets  
> from eyebeam's nat address (confirmed with tcpdump and  
> mod_event_socket DTMF) but it doesn't change the RTP destination to  
> the source of those packets and sends the packets to the private  
> address advertised in the SDP.
> However, if I do hold/unhold and send some digits, FS logs "Auto  
> Changing port from X to Y" and starts to send the packets to the  
> correct address. This always work after a hold/unhold + digits but  
> never happen just after the first INVITE transaction.
> Also, I don't see the problem if eyebeam receives a call from FS.
> Has anyone seen this?
>
> br,
> takeshi
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From djbinter at yahoo.com  Wed Oct 28 09:56:07 2009
From: djbinter at yahoo.com (DJB)
Date: Wed, 28 Oct 2009 09:56:07 -0700 (PDT)
Subject: [Freeswitch-users] SIP UPDATE Method
In-Reply-To: <191c3a030910261259p3b1fcfecvbbaed9af0af4d206@mail.gmail.com>
References: <528227.52311.qm@web37508.mail.mud.yahoo.com>
	<191c3a030910260808y45952bffu7118057caba4ea4@mail.gmail.com>
	<33829.77997.qm@web37506.mail.mud.yahoo.com>
	<191c3a030910261145q46128a92w75d73158a7f73572@mail.gmail.com>
	<514887.94094.qm@web37502.mail.mud.yahoo.com>
	<191c3a030910261247n12f1917al8110cf2636f41e20@mail.gmail.com>
	<191c3a030910261250q3b933692jd028f2067ae33252@mail.gmail.com>
	<191c3a030910261259p3b1fcfecvbbaed9af0af4d206@mail.gmail.com>
Message-ID: <938120.75712.qm@web37508.mail.mud.yahoo.com>

Thank you very much, Anthony.  I really appreciate your help and excellent work of you and your team.

Regards,
Dorn B.



________________________________
From: Anthony Minessale <anthony.minessale at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Mon, October 26, 2009 12:59:01 PM
Subject: Re: [Freeswitch-users] SIP UPDATE Method

in r15233 i put it back to the way it originally was but I may have to remove that if it causes more problems.
We tried to handle update for display updating which was not working but the handler for it was still in place which may have broken some automatic behavior regarding update so I added it back to how it was originally to determine that.



On Mon, Oct 26, 2009 at 2:50 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

>but i think the minimum value you can set is 120
>
>
>
>On Mon, Oct 26, 2009 at 2:47 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>
>I think session timers will use invite if there is no update.
>>
>>the session-timeout profile param should control that but you have to double the number you actually want because it sends the new invite at the halfway point.
>>>>
>>
>>
>>
>>
>>
>>On Mon, Oct 26, 2009 at 2:29 PM, DJB <djbinter at yahoo.com> wrote:
>>
>>>>>
>>>
>>>We were trying to see whether we can adjust call duration on Session timers.  It was a question from the application developers.  I am not sure what they are trying to do exactly.  
>>>
>>>
>>>Thank you.
>>>
>>>
>>>>>>
>>>
________________________________
From: Anthony Minessale <anthony.minessale at gmail.com>
>>>To: freeswitch-users at lists.freeswitch.org
>>>Sent: Mon, October 26, 2009 11:45:31 AM
>>>>>>
>>>
>>>
>>>Subject: Re: [Freeswitch-users] SIP UPDATE Method
>>>
>>>
>>>what exactly are you expecting to use it for?
>>>>>>We never really supported it anyway.
>>>
>>>
>>>
>>>On Mon, Oct 26, 2009 at 12:19 PM, DJB <djbinter at yahoo.com> wrote:
>>>
>>>I wonder whether you will consider to put it back on the next version 1.0.5 since 1.0.4 has it?  
>>>>
>>>>
>>>>Regards,
>>>>Dorn B.
>>>>
>>>>
>>>>
>>>>
>>>>>>>>
>>>>
>>>>
________________________________
From: Anthony Minessale <anthony.minessale at gmail.com>
>>>>To: freeswitch-users at lists.freeswitch.org
>>>>Sent: Mon, October 26, 2009 8:08:59 AM
>>>>Subject: Re: [Freeswitch-users] SIP UPDATE Method
>>>>
>>>>
>>>>It was never there before and it
>>>> caused extreme havoc once we added it so we took it away again.
>>>>
>>>>
>>>>
>>>>On Sun, Oct 25, 2009 at 12:55 AM, DJB <djbinter at yahoo.com> wrote:
>>>>
>>>>I am wondering why after I update to trunk-15225, the Allow: UPDATE method is no longer there.
>>>>>
>>>>>>>>>>User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15225
>>>>>>>>>>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>>>
>>>>>>>>>>Am I missing something here?
>>>>>
>>>>>>>>>>Thank you.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>>>>>>_______________________________________________
>>>>>>>>>>FreeSWITCH-users mailing list
>>>>>FreeSWITCH-users at lists.freeswitch.org
>>>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>-- 
>>>>Anthony Minessale II
>>>>
>>>>FreeSWITCH http://www.freeswitch.org/
>>>>ClueCon http://www.cluecon.com/
>>>>>>>>
>>>>
>>>>
>>>>
>>>>Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>>AIM: anthm
>>>>MSN:anthony_minessale at hotmail.com
>>>>>>>>
>>>>
>>>>
>>>>GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>>>>>IRC: irc.freenode.net #freeswitch
>>>>
>>>>FreeSWITCH Developer Conference
>>>>sip:888 at conference.freeswitch.org
>>>>iax:guest at conference.freeswitch.org/888
>>>>googletalk:conf+888 at conference.freeswitch.org
>>>>pstn:213-799-1400
>>>>
>>>>
>>>>_______________________________________________
>>>>>>>>FreeSWITCH-users mailing list
>>>>FreeSWITCH-users at lists.freeswitch.org
>>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>>-- 
>>>Anthony Minessale II
>>>
>>>FreeSWITCH http://www.freeswitch.org/
>>>ClueCon http://www.cluecon.com/
>>>>>>
>>>
>>>
>>>Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>>AIM: anthm
>>>MSN:anthony_minessale at hotmail.com
>>>>>>
>>>
>>>GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>>>IRC: irc.freenode.net #freeswitch
>>>
>>>FreeSWITCH Developer Conference
>>>sip:888 at conference.freeswitch.org
>>>iax:guest at conference.freeswitch.org/888
>>>googletalk:conf+888 at conference.freeswitch.org
>>>pstn:213-799-1400
>>>
>>>
>>>_______________________________________________
>>>>>>FreeSWITCH-users mailing list
>>>FreeSWITCH-users at lists.freeswitch.org
>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>http://www.freeswitch.org
>>>
>>>
>>
>>
>>-- 
>>Anthony Minessale II
>>
>>FreeSWITCH http://www.freeswitch.org/
>>ClueCon http://www.cluecon.com/
>>>>
>>
>>Twitter: http://twitter.com/FreeSWITCH_wire
>>
>>AIM: anthm
>>MSN:anthony_minessale at hotmail.com
>>>>
>>GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>IRC: irc.freenode.net #freeswitch
>>
>>FreeSWITCH Developer Conference
>>sip:888 at conference.freeswitch.org
>>iax:guest at conference.freeswitch.org/888
>>googletalk:conf+888 at conference.freeswitch.org
>>pstn:213-799-1400
>>
>
>
>-- 
>Anthony Minessale II
>
>FreeSWITCH http://www.freeswitch.org/
>ClueCon http://www.cluecon.com/
>>
>Twitter: http://twitter.com/FreeSWITCH_wire
>
>AIM: anthm
>MSN:anthony_minessale at hotmail.com
>>GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>IRC: irc.freenode.net #freeswitch
>
>FreeSWITCH Developer Conference
>sip:888 at conference.freeswitch.org
>iax:guest at conference.freeswitch.org/888
>googletalk:conf+888 at conference.freeswitch.org
>pstn:213-799-1400
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
pstn:213-799-1400



      
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From msc at freeswitch.org  Wed Oct 28 10:00:18 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 28 Oct 2009 10:00:18 -0700
Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.5pre3 is available
	for testing!
Message-ID: <87f2f3b90910281000r4c2021b8u7232320614737e30@mail.gmail.com>

Hello FreeSWITCHers,

The latest FreeSWITCH version is now available for download on the
files<http://files.freeswitch.org/freeswitch-1.0.5pre3.tar.gz>site.
The announcement story is in the main
FreeSWITCH <http://www.freeswitch.org/node/212> site. Please download and
test, and then test some more. We need your feedback. The sooner we get your
feedback, the more quickly we can roll the official 1.0.5.

Thanks!
-Michael
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From jakub at pawlinski.pl  Wed Oct 28 09:46:21 2009
From: jakub at pawlinski.pl (=?ISO-8859-2?Q?Jakub_Pawli=F1ski?=)
Date: Wed, 28 Oct 2009 17:46:21 +0100
Subject: [Freeswitch-users] Problem with hangin bri
Message-ID: <69a9ce230910280946h5eae8c58m72f9b9492f08a329@mail.gmail.com>

Hi,
I have some problems with bri status. I have 3 chanel isdn modem, and zaptel
compatible quad bri card.  I can invoke calls from my voip phone to cell
phone, and vice versa, but when i make inbound and outbound connection in
nearly same time something goes wrong with chanells and after few calls all
of them has hangup status.

There is log about that in attachement, see "is already in use waiting for
it to become available." phrase. Time of this event is about 14:43:35.
Unload and Load open_zap module helped, but its not an solution because of
lost connections.
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From jakub at pawlinski.pl  Wed Oct 28 09:49:22 2009
From: jakub at pawlinski.pl (=?ISO-8859-2?Q?Jakub_Pawli=F1ski?=)
Date: Wed, 28 Oct 2009 17:49:22 +0100
Subject: [Freeswitch-users] Problem with hangin bri
In-Reply-To: <69a9ce230910280946h5eae8c58m72f9b9492f08a329@mail.gmail.com>
References: <69a9ce230910280946h5eae8c58m72f9b9492f08a329@mail.gmail.com>
Message-ID: <69a9ce230910280949j6b48d367q6bbbe4f6e445938@mail.gmail.com>

Hi,
I have some problems with bri status. I have 3 chanel isdn modem, and zaptel
compatible quad bri card.  I can invoke calls from my voip phone to cell
phone, and vice versa, but when i make inbound and outbound connection in
nearly same time something goes wrong with chanells and after few calls all
of them has hangup status.

There is log about that in attachement, see "is already in use waiting for
it to become available." phrase. Time of this event is about 14:43:35.
Unload and Load open_zap module helped, but its not an solution because of
lost connections.
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From mayamatakeshi at gmail.com  Wed Oct 28 10:23:51 2009
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 29 Oct 2009 02:23:51 +0900
Subject: [Freeswitch-users] No RTP destination auto-correction
In-Reply-To: <A08ECEE6-46AF-497A-81CC-D65CB6BC0D95@freeswitch.org>
References: <15b9404e0910280935u16f08ff6nd8527d8c0b9f32a8@mail.gmail.com>
	<A08ECEE6-46AF-497A-81CC-D65CB6BC0D95@freeswitch.org>
Message-ID: <15b9404e0910281023v6a84bba0s1ea00621f7eb51ed@mail.gmail.com>

I'm really sorry, guys.
I was just some revisions behind and forgot to make current and test before
posting.
It's fine now.
Thank you.

On Thu, Oct 29, 2009 at 1:51 AM, Brian West <brian at freeswitch.org> wrote:

> have you updated to the latest SVN?
>
> /b
>
> On Oct 28, 2009, at 11:35 AM, mayamatakeshi wrote:
>
> > I'm seeing a strange issue with eyebeam behind NAT.
> > If I make a call from eyebeam to FS, I can see FS receives packets
> > from eyebeam's nat address (confirmed with tcpdump and
> > mod_event_socket DTMF) but it doesn't change the RTP destination to
> > the source of those packets and sends the packets to the private
> > address advertised in the SDP.
> > However, if I do hold/unhold and send some digits, FS logs "Auto
> > Changing port from X to Y" and starts to send the packets to the
> > correct address. This always work after a hold/unhold + digits but
> > never happen just after the first INVITE transaction.
> > Also, I don't see the problem if eyebeam receives a call from FS.
> > Has anyone seen this?
> >
> > br,
> > takeshi
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From davis.erwin at gmail.com  Wed Oct 28 10:39:07 2009
From: davis.erwin at gmail.com (Erwin Davis)
Date: Wed, 28 Oct 2009 13:39:07 -0400
Subject: [Freeswitch-users] error in loading spidermonkey
Message-ID: <a38a91d50910281039pdf28a4fh536f44d238b9a954@mail.gmail.com>

Hi, I got an error in loading mod_spidermonkey. my fs in ver 1.04 runs on
fedora 8 VM. Any clue? thanks,

[CRIT] switch_loadable_module.c:871 Error Loading module
/usr/local/freeswitch/mod/mod_spidermonkey.so
**/usr/lib/libnss3.so: undefined symbol: PR_UnloadLibrary**
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From william.suffill at gmail.com  Wed Oct 28 10:48:04 2009
From: william.suffill at gmail.com (William Suffill)
Date: Wed, 28 Oct 2009 13:48:04 -0400
Subject: [Freeswitch-users] error in loading spidermonkey
In-Reply-To: <a38a91d50910281039pdf28a4fh536f44d238b9a954@mail.gmail.com>
References: <a38a91d50910281039pdf28a4fh536f44d238b9a954@mail.gmail.com>
Message-ID: <6b65470d0910281048l6bd3d6exfa8504149fd78a31@mail.gmail.com>

Per http://jira.freeswitch.org/browse/MDXMLINT-23

Try to configure it with --without-libcurl.


-- W


From msc at freeswitch.org  Wed Oct 28 11:12:53 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 28 Oct 2009 11:12:53 -0700
Subject: [Freeswitch-users] SIP provider with extern rtp server
In-Reply-To: <FA2C693B-FCE1-4BDD-BCE5-302AC699CEDB@myrvold.org>
References: <A2D2E918-B494-4179-96BA-CA3A7717846C@myrvold.org>
	<bf6923ed0910270658t61bb6687ne0b7d3ec5090e2ff@mail.gmail.com>
	<CB8008BD-4066-4B5B-9043-214CFE938CD8@myrvold.org>
	<bf6923ed0910271034w6c7e0cc5t685a5377764755e6@mail.gmail.com>
	<FA2C693B-FCE1-4BDD-BCE5-302AC699CEDB@myrvold.org>
Message-ID: <87f2f3b90910281112g6e72d22elcfd653991ecd50cc@mail.gmail.com>

On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold <ivan at myrvold.org> wrote:

> Here is a debug log from a call from an internal phone out to an
> external (my iPhone with nbr 91316356):
> http://pastebin.freeswitch.org/108578
>
> Ivan
>
> Uh... you wanna try that PB number again?
-MC
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From msc at freeswitch.org  Wed Oct 28 11:17:54 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 28 Oct 2009 11:17:54 -0700
Subject: [Freeswitch-users] Problem with hangin bri
In-Reply-To: <69a9ce230910280946h5eae8c58m72f9b9492f08a329@mail.gmail.com>
References: <69a9ce230910280946h5eae8c58m72f9b9492f08a329@mail.gmail.com>
Message-ID: <87f2f3b90910281117i38eabfaalbf412adcc7d608fe@mail.gmail.com>

Okay, obligatory questions:
Which version of FreeSWITCH are you running?
Which PRI library are you using?
Which BRI card are you using?

-MC

On Wed, Oct 28, 2009 at 9:46 AM, Jakub Pawli?ski <jakub at pawlinski.pl> wrote:

> Hi,
> I have some problems with bri status. I have 3 chanel isdn modem, and
> zaptel compatible quad bri card.  I can invoke calls from my voip phone to
> cell phone, and vice versa, but when i make inbound and outbound connection
> in nearly same time something goes wrong with chanells and after few calls
> all of them has hangup status.
>
> There is log about that in attachement, see "is already in use waiting for
> it to become available." phrase. Time of this event is about 14:43:35.
> Unload and Load open_zap module helped, but its not an solution because of
> lost connections.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Oct 28 11:23:54 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 28 Oct 2009 11:23:54 -0700
Subject: [Freeswitch-users] meaning of created_time channel variable.
In-Reply-To: <1452e2980910260020m624a6a1bk2c3a5c36cb1cb5cf@mail.gmail.com>
References: <1452e2980910260020m624a6a1bk2c3a5c36cb1cb5cf@mail.gmail.com>
Message-ID: <87f2f3b90910281123j6b5b49f6xbc369d9a1bb77e5a@mail.gmail.com>

On Mon, Oct 26, 2009 at 12:20 AM, velusamy velu <velu.technical at gmail.com>wrote:

> Dear All,
>      What is the value of created_time channel variable? Is this epoch
> seconds?
>

It's epoch microseconds. Divide by 1,000,000 to get epoch seconds...
-MC
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From larclap at yahoo.com  Wed Oct 28 11:42:01 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Wed, 28 Oct 2009 11:42:01 -0700
Subject: [Freeswitch-users] Problem with inbound call answered but
	no	sound
In-Reply-To: <8D1A994D-2DCD-4016-9BEF-D6FBD3FBAA1E@freeswitch.org>
References: <016e01ca56ad$56362380$02a26a80$@com>	<168575F8-6507-4AD2-B448-2D6FDA0152B6@freeswitch.org>	<018801ca56b2$1e63e0f0$5b2ba2d0$@com>	<BACDC899-CE4F-42E1-B2A0-C0C0E31F8787@freeswitch.org>	<00d601ca571e$519f7300$f4de5900$@com>
	<8D1A994D-2DCD-4016-9BEF-D6FBD3FBAA1E@freeswitch.org>
Message-ID: <004201ca57fe$56f92a40$04eb7ec0$@com>

BroadXent ADSL 8120 ? Netscreen 5XP

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Wednesday, October 28, 2009 7:13 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Problem with inbound call answered but no
sound

 

What kind of router are you behind?

 

/b

 

On Oct 27, 2009, at 10:58 AM, Lars Zeb wrote:





Thanks for the reply, Brian.

 

Did something in FS change between v15183 and v15225 to make this occur? I
ask because this same configuration worked OK in the earlier version.

 

Lars

 

 

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From mike at jerris.com  Wed Oct 28 11:51:48 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 28 Oct 2009 14:51:48 -0400
Subject: [Freeswitch-users] error in loading spidermonkey
In-Reply-To: <a38a91d50910281039pdf28a4fh536f44d238b9a954@mail.gmail.com>
References: <a38a91d50910281039pdf28a4fh536f44d238b9a954@mail.gmail.com>
Message-ID: <8F404160-9658-4D67-A074-53022C2F5560@jerris.com>

The libcurl is broken on your distro.  You can fix this by configuring  
with --without-libcurl which will use our working in tree copy instead  
of the broken one from your distro.

Mike

On Oct 28, 2009, at 1:39 PM, Erwin Davis wrote:

> Hi, I got an error in loading mod_spidermonkey. my fs in ver 1.04  
> runs on fedora 8 VM. Any clue? thanks,
>
> [CRIT] switch_loadable_module.c:871 Error Loading module /usr/local/ 
> freeswitch/mod/mod_spidermonkey.so
> **/usr/lib/libnss3.so: undefined symbol: PR_UnloadLibrary**
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From mi+thun at aldan.algebra.com  Wed Oct 28 10:30:39 2009
From: mi+thun at aldan.algebra.com (Mikhail T.)
Date: Wed, 28 Oct 2009 13:30:39 -0400
Subject: [Freeswitch-users] FreeSWITCH on FreeBSD
Message-ID: <4AE87FBF.1020701@aldan.algebra.com>

Hello!

I'm researching a phone system for an organization with 8-12 members.
The place has an Internet connection via Verizon FIOS. My first choice
of the operating system is FreeBSD (preferably -- on amd64 platform).
FreeBSD's port (net/freeswitch) seems nice and currently installs
version 1.0.4.3. Although fairly proficient with Unix and FreeBSD in
particular, I'm very new to the field of computer telephony, so my
questions are most likely to be rather naive. I apologize...

    * What hardware is known to work well on FreeBSD?
    * What else -- other than a computer with a phone card -- will I need?
    * Do end-user phones have to be regular analog phones (all
      plugged-in to the computer), or IP-phones, such as made by Avaya
      (connecting via an Ethernet switch)?
    * How is the computer connecting to the world? Do I plug-in the
      incoming analog phone line(s) somewhere, or is it going to go
      through the Internet?
    * If the connection to the world is over the Internet, don't we need
      an account with someone else, or is Verizon known to work with
      FreeSWITCH-based PBX-systems directly?
    * Should I not simply buy a used Avaya-system like this:

http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=110450339950&_trksid=p2759.l1259
      What will I gain and what will I lose (other than personal
      satisfaction, of course) by going that route instead of building
      my own box?

Thanks a lot! Yours,

    -mi




From Suneel.Papineni at mettoni.com  Wed Oct 28 10:42:08 2009
From: Suneel.Papineni at mettoni.com (Suneel Papineni)
Date: Wed, 28 Oct 2009 17:42:08 -0000
Subject: [Freeswitch-users] Network cable disconnection
Message-ID: <3181A30B8C35AB4AA8577B78DDF4613805FCA8F7@nickel.mettonigroup.com>

Hi,

 

I have configured FreeSwitch and played it by making calls. Every thing
went fine. Then I tried to see what happens if network cable is
unplugged. Freeswitch shown an error saying ping failed to the gateway
and IP address is changed to local system IP. After sometime network
cable plugged back in and in another 30 seconds freeswitch has shown it
got the IP of gateway and changed back from local IP to gateway IP.
After that when I tried to make calls, calls are not going through. Even
there is no log displayed in Freeswitch. Please find below the logs...

 

freeswitch at QA360BASExp> 2009-10-28 16:46:13.156250 [WARNING]
sofia.c:2814 Ping failed ihub-trunk

2009-10-28 16:46:13.234375 [INFO] mod_sofia.c:3160 IP change detected
[192.168.12.10]->[192.168.20.138] []->[]

2009-10-28 16:46:13.265625 [NOTICE] sofia_glue.c:3444 Reload XML
[Success]

2009-10-28 16:46:13.265625 [INFO] switch_time.c:661 Timezone reloaded 0
definitions

2009-10-28 16:46:14.156250 [NOTICE] sofia.c:921 Waiting for worker
thread

2009-10-28 16:46:14.156250 [NOTICE] sofia_glue.c:3505 deleted gateway
ihub-trunk

2009-10-28 16:46:14.156250 [NOTICE] sofia_reg.c:2122 Added gateway
'ihub-trunk' to profile 'TrunkExternal'

2009-10-28 16:46:14.156250 [NOTICE] sofia.c:2767 Started Profile
TrunkExternal [sofia_reg_TrunkExternal]

2009-10-28 16:46:14.156250 [ERR] sofia.c:805 Error Creating SIP UA for
profile: TrunkExternal

2009-10-28 16:46:14.156250 [NOTICE] sofia_glue.c:3505 deleted gateway
ihub-trunk

2009-10-28 16:46:53.234375 [INFO] mod_sofia.c:3160 IP change detected
[192.168.20.138]->[192.168.12.10] []->[]

2009-10-28 16:46:53.281250 [NOTICE] sofia_glue.c:3444 Reload XML
[Success]

2009-10-2816:46:53.281250 [INFO] switch_time.c:661 Timezone reloaded 0
definitions

 

--After this there were no messages

 

Calls went through if I reload freeswitch. 

 

Another surprise thing is I checked for the active ports in the system
before and after cable unplugged. Port 5060 disappeared after cable is
plugged out. This doesn't display even after cable is plugged back in. 

 

Thanks & Regards

Suneel

 

 


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From marty at makolink.com  Wed Oct 28 11:17:14 2009
From: marty at makolink.com (Martin Hickman)
Date: Wed, 28 Oct 2009 18:17:14 +0000
Subject: [Freeswitch-users] Multi-tenancy context manipulation
Message-ID: <1256753834.7175.21.camel@kombu>

Hello, it's my first post, and I'm quite new to Freeswitch...

I'm looking for a bit of direction / advice on Multi-tenant set-ups. I
have configured Freeswitch as per the fine Multi-tenant guide on the
Wiki, but this leaves some aspects such as transfers and IVR still in
the default context.

For example, transfering a call that is passed to features.xml, or
connecting to an extension thorough demo_ivr will fail if the call is
left in the default context.

Would you recommend a separate features.xml for each tenants context? If
the features provided by features.xml and the ivr are the same for all
tenants, could you advise on the best way to implement one features.xml
for all tenants?

Kind regards,
Marty Hickman



From jerry.richards at teotech.com  Wed Oct 28 13:37:25 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 28 Oct 2009 13:37:25 -0700
Subject: [Freeswitch-users] IVR Intro Clipped
Message-ID: <93CC79CC678F4CAEAC236BEEFBA955C9@greyhawk.tonecommander.com>


I notice that when I call IVR from the PSTN, the "Welcome to Freeswitch..."
introduction is clipped at the beginning, so it sounds like "come to
Freeswitch".  If I call 5000 internally, then I always hear the full
introduction.  What can I do to resolve this?

My XML config looks like:

<extension name="ivr_demo">
   <condition field="destination_number" expression="5000">
      <action application="answer"/>
      <action application="start_dtmf"/>
      <action application="ivr" data="demo_ivr"/>
   </condition>
</extension>

Best Regards,
Jerry



From ivan at myrvold.org  Wed Oct 28 13:51:15 2009
From: ivan at myrvold.org (Ivan C Myrvold)
Date: Wed, 28 Oct 2009 21:51:15 +0100
Subject: [Freeswitch-users] SIP provider with extern rtp server
In-Reply-To: <87f2f3b90910281112g6e72d22elcfd653991ecd50cc@mail.gmail.com>
References: <A2D2E918-B494-4179-96BA-CA3A7717846C@myrvold.org>
	<bf6923ed0910270658t61bb6687ne0b7d3ec5090e2ff@mail.gmail.com>
	<CB8008BD-4066-4B5B-9043-214CFE938CD8@myrvold.org>
	<bf6923ed0910271034w6c7e0cc5t685a5377764755e6@mail.gmail.com>
	<FA2C693B-FCE1-4BDD-BCE5-302AC699CEDB@myrvold.org>
	<87f2f3b90910281112g6e72d22elcfd653991ecd50cc@mail.gmail.com>
Message-ID: <8AC09649-2585-4BE7-A959-A7AC41650789@myrvold.org>

Oh, what happened to it?

Anyway, here is a new pb:

http://pastebin.freeswitch.org/10867

Ivan
Den 28. okt. 2009 kl. 19:12 skrev Michael Collins:

>
>
> On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold <ivan at myrvold.org>  
> wrote:
> Here is a debug log from a call from an internal phone out to an
> external (my iPhone with nbr 91316356):
> http://pastebin.freeswitch.org/108578
>
> Ivan
>
> Uh... you wanna try that PB number again?
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From brian at freeswitch.org  Wed Oct 28 13:51:16 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 28 Oct 2009 15:51:16 -0500
Subject: [Freeswitch-users] IVR Intro Clipped
In-Reply-To: <93CC79CC678F4CAEAC236BEEFBA955C9@greyhawk.tonecommander.com>
References: <93CC79CC678F4CAEAC236BEEFBA955C9@greyhawk.tonecommander.com>
Message-ID: <3A585780-F132-4D4B-AD97-66904A9C3446@freeswitch.org>

Sleep 1000 ms... we usually bring up media too fast before the other  
end is ready.

/b

On Oct 28, 2009, at 3:37 PM, Jerry Richards wrote:

>
> I notice that when I call IVR from the PSTN, the "Welcome to  
> Freeswitch..."
> introduction is clipped at the beginning, so it sounds like "come to
> Freeswitch".  If I call 5000 internally, then I always hear the full
> introduction.  What can I do to resolve this?
>
> My XML config looks like:
>
> <extension name="ivr_demo">
>   <condition field="destination_number" expression="5000">
>      <action application="answer"/>
>      <action application="start_dtmf"/>
>      <action application="ivr" data="demo_ivr"/>
>   </condition>
> </extension>
>
> Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From mariusz_kolo at wp.pl  Wed Oct 28 14:12:22 2009
From: mariusz_kolo at wp.pl (=?UTF-8?B?TWFyaXVzeiBLb8WCb2R6aWVqY3p5aw==?=)
Date: Wed, 28 Oct 2009 22:12:22 +0100
Subject: [Freeswitch-users] Problem with hangin bri
In-Reply-To: <87f2f3b90910281117i38eabfaalbf412adcc7d608fe@mail.gmail.com>
References: <69a9ce230910280946h5eae8c58m72f9b9492f08a329@mail.gmail.com>
	<87f2f3b90910281117i38eabfaalbf412adcc7d608fe@mail.gmail.com>
Message-ID: <4AE8B3B6.9000106@wp.pl>

Hi

I'm also working on this project, so i can answer your questions

Which version of FreeSWITCH are you running?

FreeSWITCH Version 1.0.trunk (15246)

Which PRI library are you using?
openzap Native stack

openzap.conf

[span zt BRI1]
trunk_type => bri
b-channel => 1-2
d-channel=> 3

openzap.conf

<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    <param name="debug" value="0"/>
  </settings>

   <pri_spans>
    <span id="1">
     <param name="mode" value="user"/>
     <param name="dialect" value="q931"/>
     <param name="dialplan" value="XML"/>
     <param name="context" value="default"/>
    </span>
  </pri_spans>
</configuration>

Which BRI card are you using?

Producer: http://www.phoniceq.com/
card model: http://quadbri.phoniceq.com/

Card instalation process (instruction from producer)

1) download bristuff staff from

http://junghanns.net/downloads/bristuff-0.4.0-RC3h.tar.gz
or
http://junghanns.net/downloads/bristuff-0.3.0-PRE-1y-z.tar.gz

unpack it and go to bristuff-*

2) download patcher from
http://quadbri.phoniceq.com/driver/bristuff/qozap-bristuff-0.3.0-PRE-1y-j-enableLEDS.patch

patch it using 

patch -p0 < qozap-bristuff-0.3.0-PRE-1y-j-enableLEDS.patch

3) you can check card using zttest (result should be 99.x)

Producer has said, that we are first client, it wants to use this card in freeswitch 

we are using 1 port (S/T interface). Our NT is "NT1 plus 2b1q" 


Thanks

Michael Collins pisze:
> Okay, obligatory questions:
> Which version of FreeSWITCH are you running?
> Which PRI library are you using?
> Which BRI card are you using?
>
> -MC
>
> On Wed, Oct 28, 2009 at 9:46 AM, Jakub Pawli?ski <jakub at pawlinski.pl 
> <mailto:jakub at pawlinski.pl>> wrote:
>
>     Hi,
>     I have some problems with bri status. I have 3 chanel isdn modem,
>     and zaptel compatible quad bri card.  I can invoke calls from my
>     voip phone to cell phone, and vice versa, but when i make inbound
>     and outbound connection in nearly same time something goes wrong
>     with chanells and after few calls all of them has hangup status.
>
>     There is log about that in attachement, see "is already in use
>     waiting for it to become available." phrase. Time of this event is
>     about 14:43:35. Unload and Load open_zap module helped, but its
>     not an solution because of lost connections.
>
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


-- 
Mariusz Ko?odziejczyk

Advanced Developing Architecture S.C.

tel.   : +48 609 381 316
e-mail : mariusz_kolo at wp.pl



From jerry.richards at teotech.com  Wed Oct 28 14:36:17 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 28 Oct 2009 14:36:17 -0700
Subject: [Freeswitch-users] IVR Intro Clipped
In-Reply-To: <3A585780-F132-4D4B-AD97-66904A9C3446@freeswitch.org>
References: <93CC79CC678F4CAEAC236BEEFBA955C9@greyhawk.tonecommander.com>
	<3A585780-F132-4D4B-AD97-66904A9C3446@freeswitch.org>
Message-ID: <5B061E041F8E4D8B81464B910D32F47E@greyhawk.tonecommander.com>


I modified my dialplan as shown, but the clipping persists.  Should the
sleep be placed somewhere else?

<extension name="ivr_demo">
  <condition field="destination_number" expression="5000">
     <action application="sleep" data="1000"\>
     <action application="answer"/>
     <action application="start_dtmf"/>
     <action application="ivr" data="demo_ivr"/>
  </condition>
</extension>

Best Regards,
Jerry
 

-----Original Message-----
From: Brian West [mailto:brian at freeswitch.org] 
Sent: Wednesday, October 28, 2009 1:51 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] IVR Intro Clipped

Sleep 1000 ms... we usually bring up media too fast before the other end is
ready.

/b

On Oct 28, 2009, at 3:37 PM, Jerry Richards wrote:

>
> I notice that when I call IVR from the PSTN, the "Welcome to 
> Freeswitch..."
> introduction is clipped at the beginning, so it sounds like "come to 
> Freeswitch".  If I call 5000 internally, then I always hear the full 
> introduction.  What can I do to resolve this?
>
> My XML config looks like:
>
> <extension name="ivr_demo">
>   <condition field="destination_number" expression="5000">
>      <action application="answer"/>
>      <action application="start_dtmf"/>
>      <action application="ivr" data="demo_ivr"/>
>   </condition>
> </extension>
>
> Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> users
> http://www.freeswitch.org






From frank at carmickle.com  Wed Oct 28 14:45:31 2009
From: frank at carmickle.com (Frank Carmickle)
Date: Wed, 28 Oct 2009 17:45:31 -0400
Subject: [Freeswitch-users] IVR Intro Clipped
In-Reply-To: <5B061E041F8E4D8B81464B910D32F47E@greyhawk.tonecommander.com>
References: <93CC79CC678F4CAEAC236BEEFBA955C9@greyhawk.tonecommander.com>
	<3A585780-F132-4D4B-AD97-66904A9C3446@freeswitch.org>
	<5B061E041F8E4D8B81464B910D32F47E@greyhawk.tonecommander.com>
Message-ID: <20091028214531.GM13895@base.carmickle.com>

Hello

Yes.  Answer first.

--FC


On Wed, Oct 28, Jerry Richards wrote:
> 
> I modified my dialplan as shown, but the clipping persists.  Should the
> sleep be placed somewhere else?
> 
> <extension name="ivr_demo">
>   <condition field="destination_number" expression="5000">
>      <action application="sleep" data="1000"\>
>      <action application="answer"/>
>      <action application="start_dtmf"/>
>      <action application="ivr" data="demo_ivr"/>
>   </condition>
> </extension>
> 
> Best Regards,
> Jerry
>  
> 
> -----Original Message-----
> From: Brian West [mailto:brian at freeswitch.org] 
> Sent: Wednesday, October 28, 2009 1:51 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] IVR Intro Clipped
> 
> Sleep 1000 ms... we usually bring up media too fast before the other end is
> ready.
> 
> /b
> 
> On Oct 28, 2009, at 3:37 PM, Jerry Richards wrote:
> 
> >
> > I notice that when I call IVR from the PSTN, the "Welcome to 
> > Freeswitch..."
> > introduction is clipped at the beginning, so it sounds like "come to 
> > Freeswitch".  If I call 5000 internally, then I always hear the full 
> > introduction.  What can I do to resolve this?
> >
> > My XML config looks like:
> >
> > <extension name="ivr_demo">
> >   <condition field="destination_number" expression="5000">
> >      <action application="answer"/>
> >      <action application="start_dtmf"/>
> >      <action application="ivr" data="demo_ivr"/>
> >   </condition>
> > </extension>
> >
> > Best Regards,
> > Jerry
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
> 
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From sicfslist at gmail.com  Wed Oct 28 14:47:53 2009
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Wed, 28 Oct 2009 16:47:53 -0500
Subject: [Freeswitch-users] IVR Intro Clipped
In-Reply-To: <5B061E041F8E4D8B81464B910D32F47E@greyhawk.tonecommander.com>
References: <93CC79CC678F4CAEAC236BEEFBA955C9@greyhawk.tonecommander.com>	<3A585780-F132-4D4B-AD97-66904A9C3446@freeswitch.org>
	<5B061E041F8E4D8B81464B910D32F47E@greyhawk.tonecommander.com>
Message-ID: <4AE8BC09.1060602@gmail.com>

Jerry,

Put the sleep after the answer.  That should fix it.

Shelby

Jerry Richards wrote:
> I modified my dialplan as shown, but the clipping persists.  Should the
> sleep be placed somewhere else?
>
> <extension name="ivr_demo">
>   <condition field="destination_number" expression="5000">
>      <action application="sleep" data="1000"\>
>      <action application="answer"/>
>      <action application="start_dtmf"/>
>      <action application="ivr" data="demo_ivr"/>
>   </condition>
> </extension>
>
> Best Regards,
> Jerry
>  
>
> -----Original Message-----
> From: Brian West [mailto:brian at freeswitch.org] 
> Sent: Wednesday, October 28, 2009 1:51 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] IVR Intro Clipped
>
> Sleep 1000 ms... we usually bring up media too fast before the other end is
> ready.
>
> /b
>
> On Oct 28, 2009, at 3:37 PM, Jerry Richards wrote:
>
>   
>> I notice that when I call IVR from the PSTN, the "Welcome to 
>> Freeswitch..."
>> introduction is clipped at the beginning, so it sounds like "come to 
>> Freeswitch".  If I call 5000 internally, then I always hear the full 
>> introduction.  What can I do to resolve this?
>>
>> My XML config looks like:
>>
>> <extension name="ivr_demo">
>>   <condition field="destination_number" expression="5000">
>>      <action application="answer"/>
>>      <action application="start_dtmf"/>
>>      <action application="ivr" data="demo_ivr"/>
>>   </condition>
>> </extension>
>>
>> Best Regards,
>> Jerry
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>>     
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   



From cliff at develix.com  Wed Oct 28 14:55:17 2009
From: cliff at develix.com (Cliff Wells)
Date: Wed, 28 Oct 2009 14:55:17 -0700
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <000f01ca5740$fd9d7a40$f8d86ec0$@net>
References: <bf6923ed0910261446x6a184ecdnee2e73e904406246@mail.gmail.com>
	<bf6923ed0910261501k1ff8c892rfc198c31e30ab286@mail.gmail.com>
	<1256673110.9746.29.camel@portable-evil>
	<000f01ca5740$fd9d7a40$f8d86ec0$@net>
Message-ID: <1256766917.28102.29.camel@portable-evil>

That's really nice looking software.   Thanks for the pointer.

Cliff


On Tue, 2009-10-27 at 16:06 -0400, Gregory Boehnlein wrote:
> I'm fond of Vqmanager from ManagEngine. It is a passive SIP monitor. I.E.
> you mirror the ports that your FS or Asterisk boxes, and VQmanager sniffs
> the mirrors, tracking all sorts of good data.
> 
> You can install it on a Centos box, and get a free trial.
> 
> http://www.manageengine.com/products/vqmanager/index.html
> 
> What is really cool is that it actually monitors the RTP/RTCP as well as all
> of the SIP headers and archives the calls, so you can look at calls from
> several days ago and see EXACTLY what happened on them. I have used this
> extensively to pinpoint bad Level 3 and X/O media gateways.. Much better
> than trying to sniff packets in real-time and MAYBE catch a problem..
> 
> I've also used it to find/fix several SIP issues w/ odd endpoints.. Very
> easy to see..
> 
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> > users-bounces at lists.freeswitch.org] On Behalf Of Cliff Wells
> > Sent: Tuesday, October 27, 2009 3:52 PM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] Estimating Call Capacity
> > 
> > A little off-topic, but since call-capacity is the subject, what are
> > people using to analyze their CDR's to discover this?   I'm handling
> > about 30k calls per day but have only a bandwidth-based guesstimate of
> > the peak number of concurrent calls I'm handling.
> > 
> > If there's an open source solution, I'd appreciate a pointer.
> > 
> > Regards,
> > Cliff
> > 
> > On Mon, 2009-10-26 at 18:01 -0400, Eliot Gable wrote:
> > > Although, FYI, I just benchmarked mod_xml_curl on a separate web app
> > > server from FS with FS on a Dell R710 with their current best
> > > processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and
> > 32
> > > GB memory. The web app server is less than half the power of the
> > R710.
> > > I maxed the web app server at 300 calls per second (both setting up
> > > and tearing down) and the R710 running FS was 65% idle. No audio was
> > > being proxied through FS, though. If I were running the web app
> > server
> > > on an equivalent R710, they probably would have been on-par with each
> > > other in performance. Extrapolating, I expect that in such a case I
> > > should be able to get at least 650 CPS out of FS, though for
> > > production I would probably limit it to 400 CPS or less so I leave
> > > room for miscellaneous tasks. I maxed out the R710 at over 16,000
> > > simultaneous calls (again, no audio proxying) but the only reason I
> > > couldn't do more was because I hit some sort of thread creation limit
> > > in Linux. There was about 17 GB of memory used for this many calls.
> > > This should give you some ballpark idea of what you can accomplish
> > > with FS.
> > >
> > > At some point, I will track down and resolve the thread creation
> > > issue, at which time I believe call limits will be limited either by
> > a
> > > complex combination of available memory, the speed of the processor,
> > > the cost of thread context switching, calls per second setup rate,
> > and
> > > call duration.
> > >
> > > --
> > > Eliot Gable
> > >
> > > > -----Original Message-----
> > > >
> > > > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> > Giovanni Maruzzelli
> > > >
> > > > Sent: Monday, October 26, 2009 4:56 PM
> > > >
> > > > To: freeswitch-users at lists.freeswitch.org
> > > >
> > > > Subject: Re: [Freeswitch-users] Estimating Call Capacity
> > > >
> > > >
> > > >
> > > > On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur
> > > >
> > > > <vinuth.madinur at gmail.com> wrote:
> > > >
> > > > > Here are a few benchmarks that I had stumbled upon.
> > > >
> > > > >
> > http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+
> > x2200+M2
> > > >
> > > >
> > > >
> > > > Please remember NO benchmarks are endorsed by the FS community or
> > > >
> > > > developers, because there are just too many variables, and a simple
> > > >
> > > > figure is just useful for marketing hype, not for real
> > dimensioning.
> > > >
> > > >
> > > >
> > > > You MUST do your own benchmarking, so you get an idea about how to
> > > >
> > > > dimension for your own use case and hardware.
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > > Thanks,
> > > >
> > > > > Vinuth.
> > > >
> > > > >
> > > >
> > > > > On Tue, Oct 27, 2009 at 1:43 AM, Brian West
> > <brian at freeswitch.org> wrote:
> > > >
> > > > >>
> > > >
> > > > >> I highly doubt it... You can wait for someone to post their
> > results
> > > >
> > > > >> but in the end you'll have to do your own load testing because
> > not
> > > >
> > > > >> everyone's numbers will jive with your use case.  Which is the
> > reason
> > > >
> > > > >> the project never posts or endorses a set call count.
> > > >
> > > > >>
> > > >
> > > > >> /b
> > > >
> > > > >>
> > > >
> > > > >> On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote:
> > > >
> > > > >>
> > > >
> > > > >> > Are there any benchmarking test results available publicly?
> > > >
> > > > >> > ________________________________________
> > > >
> > > > >>
> > > >
> > > > >>
> > > >
> > > > >> _______________________________________________
> > > >
> > > > >> FreeSWITCH-users mailing list
> > > >
> > > > >> FreeSWITCH-users at lists.freeswitch.org
> > > >
> > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >
> > > > >>
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > > >
> > > > >> http://www.freeswitch.org
> > > >
> > > > >
> > > >
> > > > >
> > > >
> > > > > _______________________________________________
> > > >
> > > > > FreeSWITCH-users mailing list
> > > >
> > > > > FreeSWITCH-users at lists.freeswitch.org
> > > >
> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >
> > > > >
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > > >
> > > > > http://www.freeswitch.org
> > > >
> > > > >
> > > >
> > > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > --
> > > >
> > > > Sincerely,
> > > >
> > > >
> > > >
> > > > Giovanni Maruzzelli
> > > >
> > > > Cell : +39-347-2665618
> > > >
> > > >
> > > >
> > > > _______________________________________________
> > > >
> > > > FreeSWITCH-users mailing list
> > > >
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > >
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >
> > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > > >
> > > > http://www.freeswitch.org
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > > http://www.freeswitch.org
> > --
> > http://www.google.com/search?q=vonage+sucks
> > 
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
> > 
> > --
> > This message has been scanned for viruses and
> > dangerous content by N2Net Mailshield, and is
> > believed to be clean.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 
http://www.google.com/search?q=vonage+sucks



From cliff at develix.com  Wed Oct 28 14:55:17 2009
From: cliff at develix.com (Cliff Wells)
Date: Wed, 28 Oct 2009 14:55:17 -0700
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <000f01ca5740$fd9d7a40$f8d86ec0$@net>
References: <bf6923ed0910261446x6a184ecdnee2e73e904406246@mail.gmail.com>
	<bf6923ed0910261501k1ff8c892rfc198c31e30ab286@mail.gmail.com>
	<1256673110.9746.29.camel@portable-evil>
	<000f01ca5740$fd9d7a40$f8d86ec0$@net>
Message-ID: <1256766917.28102.29.camel@portable-evil>

That's really nice looking software.   Thanks for the pointer.

Cliff


On Tue, 2009-10-27 at 16:06 -0400, Gregory Boehnlein wrote:
> I'm fond of Vqmanager from ManagEngine. It is a passive SIP monitor. I.E.
> you mirror the ports that your FS or Asterisk boxes, and VQmanager sniffs
> the mirrors, tracking all sorts of good data.
> 
> You can install it on a Centos box, and get a free trial.
> 
> http://www.manageengine.com/products/vqmanager/index.html
> 
> What is really cool is that it actually monitors the RTP/RTCP as well as all
> of the SIP headers and archives the calls, so you can look at calls from
> several days ago and see EXACTLY what happened on them. I have used this
> extensively to pinpoint bad Level 3 and X/O media gateways.. Much better
> than trying to sniff packets in real-time and MAYBE catch a problem..
> 
> I've also used it to find/fix several SIP issues w/ odd endpoints.. Very
> easy to see..
> 
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> > users-bounces at lists.freeswitch.org] On Behalf Of Cliff Wells
> > Sent: Tuesday, October 27, 2009 3:52 PM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] Estimating Call Capacity
> > 
> > A little off-topic, but since call-capacity is the subject, what are
> > people using to analyze their CDR's to discover this?   I'm handling
> > about 30k calls per day but have only a bandwidth-based guesstimate of
> > the peak number of concurrent calls I'm handling.
> > 
> > If there's an open source solution, I'd appreciate a pointer.
> > 
> > Regards,
> > Cliff
> > 
> > On Mon, 2009-10-26 at 18:01 -0400, Eliot Gable wrote:
> > > Although, FYI, I just benchmarked mod_xml_curl on a separate web app
> > > server from FS with FS on a Dell R710 with their current best
> > > processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and
> > 32
> > > GB memory. The web app server is less than half the power of the
> > R710.
> > > I maxed the web app server at 300 calls per second (both setting up
> > > and tearing down) and the R710 running FS was 65% idle. No audio was
> > > being proxied through FS, though. If I were running the web app
> > server
> > > on an equivalent R710, they probably would have been on-par with each
> > > other in performance. Extrapolating, I expect that in such a case I
> > > should be able to get at least 650 CPS out of FS, though for
> > > production I would probably limit it to 400 CPS or less so I leave
> > > room for miscellaneous tasks. I maxed out the R710 at over 16,000
> > > simultaneous calls (again, no audio proxying) but the only reason I
> > > couldn't do more was because I hit some sort of thread creation limit
> > > in Linux. There was about 17 GB of memory used for this many calls.
> > > This should give you some ballpark idea of what you can accomplish
> > > with FS.
> > >
> > > At some point, I will track down and resolve the thread creation
> > > issue, at which time I believe call limits will be limited either by
> > a
> > > complex combination of available memory, the speed of the processor,
> > > the cost of thread context switching, calls per second setup rate,
> > and
> > > call duration.
> > >
> > > --
> > > Eliot Gable
> > >
> > > > -----Original Message-----
> > > >
> > > > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> > Giovanni Maruzzelli
> > > >
> > > > Sent: Monday, October 26, 2009 4:56 PM
> > > >
> > > > To: freeswitch-users at lists.freeswitch.org
> > > >
> > > > Subject: Re: [Freeswitch-users] Estimating Call Capacity
> > > >
> > > >
> > > >
> > > > On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur
> > > >
> > > > <vinuth.madinur at gmail.com> wrote:
> > > >
> > > > > Here are a few benchmarks that I had stumbled upon.
> > > >
> > > > >
> > http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+
> > x2200+M2
> > > >
> > > >
> > > >
> > > > Please remember NO benchmarks are endorsed by the FS community or
> > > >
> > > > developers, because there are just too many variables, and a simple
> > > >
> > > > figure is just useful for marketing hype, not for real
> > dimensioning.
> > > >
> > > >
> > > >
> > > > You MUST do your own benchmarking, so you get an idea about how to
> > > >
> > > > dimension for your own use case and hardware.
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > > Thanks,
> > > >
> > > > > Vinuth.
> > > >
> > > > >
> > > >
> > > > > On Tue, Oct 27, 2009 at 1:43 AM, Brian West
> > <brian at freeswitch.org> wrote:
> > > >
> > > > >>
> > > >
> > > > >> I highly doubt it... You can wait for someone to post their
> > results
> > > >
> > > > >> but in the end you'll have to do your own load testing because
> > not
> > > >
> > > > >> everyone's numbers will jive with your use case.  Which is the
> > reason
> > > >
> > > > >> the project never posts or endorses a set call count.
> > > >
> > > > >>
> > > >
> > > > >> /b
> > > >
> > > > >>
> > > >
> > > > >> On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote:
> > > >
> > > > >>
> > > >
> > > > >> > Are there any benchmarking test results available publicly?
> > > >
> > > > >> > ________________________________________
> > > >
> > > > >>
> > > >
> > > > >>
> > > >
> > > > >> _______________________________________________
> > > >
> > > > >> FreeSWITCH-users mailing list
> > > >
> > > > >> FreeSWITCH-users at lists.freeswitch.org
> > > >
> > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >
> > > > >>
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > > >
> > > > >> http://www.freeswitch.org
> > > >
> > > > >
> > > >
> > > > >
> > > >
> > > > > _______________________________________________
> > > >
> > > > > FreeSWITCH-users mailing list
> > > >
> > > > > FreeSWITCH-users at lists.freeswitch.org
> > > >
> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >
> > > > >
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > > >
> > > > > http://www.freeswitch.org
> > > >
> > > > >
> > > >
> > > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > --
> > > >
> > > > Sincerely,
> > > >
> > > >
> > > >
> > > > Giovanni Maruzzelli
> > > >
> > > > Cell : +39-347-2665618
> > > >
> > > >
> > > >
> > > > _______________________________________________
> > > >
> > > > FreeSWITCH-users mailing list
> > > >
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > >
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >
> > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > > >
> > > > http://www.freeswitch.org
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > > http://www.freeswitch.org
> > --
> > http://www.google.com/search?q=vonage+sucks
> > 
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
> > 
> > --
> > This message has been scanned for viruses and
> > dangerous content by N2Net Mailshield, and is
> > believed to be clean.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 
http://www.google.com/search?q=vonage+sucks



From cliff at develix.com  Wed Oct 28 14:58:09 2009
From: cliff at develix.com (Cliff Wells)
Date: Wed, 28 Oct 2009 14:58:09 -0700
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <4AE755C3.6060307@gmail.com>
References: <bf6923ed0910261446x6a184ecdnee2e73e904406246@mail.gmail.com>
	<bf6923ed0910261501k1ff8c892rfc198c31e30ab286@mail.gmail.com>
	<1256673110.9746.29.camel@portable-evil> <4AE755C3.6060307@gmail.com>
Message-ID: <1256767089.17442.2.camel@portable-evil>

Hi Shelby,

Thanks!  That's pretty useful.   I also note that this same info is
available from the CLI, although I was curious as to what some of the
numbers indicated:

UP 0 years, 13 days, 19 hours, 25 minutes, 48 seconds, 887 milliseconds, 975 microseconds
529509 session(s) since startup
26 session(s) 0/30
1000 session(s) max

Specifically, 26 sessions 0/30... I take it this means there are 26
current sessions, but I'm unsure of what the "0/30" means.

Cliff


On Tue, 2009-10-27 at 15:19 -0500, Shelby Ramsey wrote:
> Cliff,
> 
> Try using xml_rpc ... status or show channels will give you what you need.
> 
> SDR
> 
> Cliff Wells wrote:
> > A little off-topic, but since call-capacity is the subject, what are
> > people using to analyze their CDR's to discover this?   I'm handling
> > about 30k calls per day but have only a bandwidth-based guesstimate of
> > the peak number of concurrent calls I'm handling.
> >
> > If there's an open source solution, I'd appreciate a pointer.
> >
> > Regards,
> > Cliff
> >
> > On Mon, 2009-10-26 at 18:01 -0400, Eliot Gable wrote:
> >   
> >> Although, FYI, I just benchmarked mod_xml_curl on a separate web app
> >> server from FS with FS on a Dell R710 with their current best
> >> processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and 32
> >> GB memory. The web app server is less than half the power of the R710.
> >> I maxed the web app server at 300 calls per second (both setting up
> >> and tearing down) and the R710 running FS was 65% idle. No audio was
> >> being proxied through FS, though. If I were running the web app server
> >> on an equivalent R710, they probably would have been on-par with each
> >> other in performance. Extrapolating, I expect that in such a case I
> >> should be able to get at least 650 CPS out of FS, though for
> >> production I would probably limit it to 400 CPS or less so I leave
> >> room for miscellaneous tasks. I maxed out the R710 at over 16,000
> >> simultaneous calls (again, no audio proxying) but the only reason I
> >> couldn't do more was because I hit some sort of thread creation limit
> >> in Linux. There was about 17 GB of memory used for this many calls.
> >> This should give you some ballpark idea of what you can accomplish
> >> with FS.
> >>
> >> At some point, I will track down and resolve the thread creation
> >> issue, at which time I believe call limits will be limited either by a
> >> complex combination of available memory, the speed of the processor,
> >> the cost of thread context switching, calls per second setup rate, and
> >> call duration.
> >>
> >> --
> >> Eliot Gable
> >>
> >>     
> >>> -----Original Message-----
> >>>
> >>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli
> >>>
> >>> Sent: Monday, October 26, 2009 4:56 PM
> >>>
> >>> To: freeswitch-users at lists.freeswitch.org
> >>>
> >>> Subject: Re: [Freeswitch-users] Estimating Call Capacity
> >>>
> >>>
> >>>
> >>> On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur
> >>>
> >>> <vinuth.madinur at gmail.com> wrote:
> >>>
> >>>       
> >>>> Here are a few benchmarks that I had stumbled upon.
> >>>>         
> >>>> http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2
> >>>>         
> >>>
> >>> Please remember NO benchmarks are endorsed by the FS community or
> >>>
> >>> developers, because there are just too many variables, and a simple
> >>>
> >>> figure is just useful for marketing hype, not for real dimensioning.
> >>>
> >>>
> >>>
> >>> You MUST do your own benchmarking, so you get an idea about how to
> >>>
> >>> dimension for your own use case and hardware.
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>       
> >>>> Thanks,
> >>>>         
> >>>> Vinuth.
> >>>>         
> >>>> On Tue, Oct 27, 2009 at 1:43 AM, Brian West <brian at freeswitch.org> wrote:
> >>>>         
> >>>>> I highly doubt it... You can wait for someone to post their results
> >>>>>           
> >>>>> but in the end you'll have to do your own load testing because not
> >>>>>           
> >>>>> everyone's numbers will jive with your use case.  Which is the reason
> >>>>>           
> >>>>> the project never posts or endorses a set call count.
> >>>>>           
> >>>>> /b
> >>>>>           
> >>>>> On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote:
> >>>>>           
> >>>>>> Are there any benchmarking test results available publicly?
> >>>>>>             
> >>>>>> ________________________________________
> >>>>>>             
> >>>>> _______________________________________________
> >>>>>           
> >>>>> FreeSWITCH-users mailing list
> >>>>>           
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>           
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>           
> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>           
> >>>>> http://www.freeswitch.org
> >>>>>           
> >>>> _______________________________________________
> >>>>         
> >>>> FreeSWITCH-users mailing list
> >>>>         
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>         
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>         
> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>         
> >>>> http://www.freeswitch.org
> >>>>         
> >>>
> >>>
> >>>
> >>>
> >>>
> >>> --
> >>>
> >>> Sincerely,
> >>>
> >>>
> >>>
> >>> Giovanni Maruzzelli
> >>>
> >>> Cell : +39-347-2665618
> >>>
> >>>
> >>>
> >>> _______________________________________________
> >>>
> >>> FreeSWITCH-users mailing list
> >>>
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>>
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>
> >>> http://www.freeswitch.org
> >>>       
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>     
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 
http://www.google.com/search?q=vonage+sucks



From brian at freeswitch.org  Wed Oct 28 15:02:16 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 28 Oct 2009 17:02:16 -0500
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <1256767089.17442.2.camel@portable-evil>
References: <bf6923ed0910261446x6a184ecdnee2e73e904406246@mail.gmail.com>
	<bf6923ed0910261501k1ff8c892rfc198c31e30ab286@mail.gmail.com>
	<1256673110.9746.29.camel@portable-evil>
	<4AE755C3.6060307@gmail.com>
	<1256767089.17442.2.camel@portable-evil>
Message-ID: <499FBF53-6D73-4C3D-8059-EBDD5C4E3C25@freeswitch.org>

sessions per second now/max

/b

On Oct 28, 2009, at 4:58 PM, Cliff Wells wrote:

> current sessions, but I'm unsure of what the "0/30" means.



From msc at freeswitch.org  Wed Oct 28 15:28:24 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 28 Oct 2009 15:28:24 -0700
Subject: [Freeswitch-users] Problem with hangin bri
In-Reply-To: <4AE8B3B6.9000106@wp.pl>
References: <69a9ce230910280946h5eae8c58m72f9b9492f08a329@mail.gmail.com>
	<87f2f3b90910281117i38eabfaalbf412adcc7d608fe@mail.gmail.com>
	<4AE8B3B6.9000106@wp.pl>
Message-ID: <87f2f3b90910281528t633b765x7c2bb858f41ce154@mail.gmail.com>

Thanks. Can you collect debug logs of this happening? See
http://wiki.freeswitch.org/wiki/Reporting_Bugs for helpful tips on
collecting debug information. Use pastebin to dump all the log info and
reply here with the link. We don't have too many BRI users but I believe
there are a few so hopefully we can help you get up and running.
-MC

2009/10/28 Mariusz Ko?odziejczyk <mariusz_kolo at wp.pl>

> Hi
>
> I'm also working on this project, so i can answer your questions
>
> Which version of FreeSWITCH are you running?
>
> FreeSWITCH Version 1.0.trunk (15246)
>
> Which PRI library are you using?
> openzap Native stack
>
> openzap.conf
>
> [span zt BRI1]
> trunk_type => bri
> b-channel => 1-2
> d-channel=> 3
>
> openzap.conf
>
> <configuration name="openzap.conf" description="OpenZAP Configuration">
>  <settings>
>    <param name="debug" value="0"/>
>  </settings>
>
>   <pri_spans>
>    <span id="1">
>     <param name="mode" value="user"/>
>     <param name="dialect" value="q931"/>
>     <param name="dialplan" value="XML"/>
>     <param name="context" value="default"/>
>    </span>
>  </pri_spans>
> </configuration>
>
> Which BRI card are you using?
>
> Producer: http://www.phoniceq.com/
> card model: http://quadbri.phoniceq.com/
>
> Card instalation process (instruction from producer)
>
> 1) download bristuff staff from
>
> http://junghanns.net/downloads/bristuff-0.4.0-RC3h.tar.gz
> or
> http://junghanns.net/downloads/bristuff-0.3.0-PRE-1y-z.tar.gz
>
> unpack it and go to bristuff-*
>
> 2) download patcher from
>
> http://quadbri.phoniceq.com/driver/bristuff/qozap-bristuff-0.3.0-PRE-1y-j-enableLEDS.patch
>
> patch it using
>
> patch -p0 < qozap-bristuff-0.3.0-PRE-1y-j-enableLEDS.patch
>
> 3) you can check card using zttest (result should be 99.x)
>
> Producer has said, that we are first client, it wants to use this card in
> freeswitch
>
> we are using 1 port (S/T interface). Our NT is "NT1 plus 2b1q"
>
>
> Thanks
>
> Michael Collins pisze:
> > Okay, obligatory questions:
> > Which version of FreeSWITCH are you running?
> > Which PRI library are you using?
> > Which BRI card are you using?
> >
> > -MC
> >
> > On Wed, Oct 28, 2009 at 9:46 AM, Jakub Pawli?ski <jakub at pawlinski.pl
> > <mailto:jakub at pawlinski.pl>> wrote:
> >
> >     Hi,
> >     I have some problems with bri status. I have 3 chanel isdn modem,
> >     and zaptel compatible quad bri card.  I can invoke calls from my
> >     voip phone to cell phone, and vice versa, but when i make inbound
> >     and outbound connection in nearly same time something goes wrong
> >     with chanells and after few calls all of them has hangup status.
> >
> >     There is log about that in attachement, see "is already in use
> >     waiting for it to become available." phrase. Time of this event is
> >     about 14:43:35. Unload and Load open_zap module helped, but its
> >     not an solution because of lost connections.
> >
> >
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> --
> Mariusz Ko?odziejczyk
>
> Advanced Developing Architecture S.C.
>
> tel.   : +48 609 381 316
> e-mail : mariusz_kolo at wp.pl
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Wed Oct 28 15:33:28 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 28 Oct 2009 15:33:28 -0700
Subject: [Freeswitch-users] Multi-tenancy context manipulation
In-Reply-To: <1256753834.7175.21.camel@kombu>
References: <1256753834.7175.21.camel@kombu>
Message-ID: <87f2f3b90910281533l36647887u85375f005824228b@mail.gmail.com>

On Wed, Oct 28, 2009 at 11:17 AM, Martin Hickman <marty at makolink.com> wrote:

> Hello, it's my first post, and I'm quite new to Freeswitch...
>
> I'm looking for a bit of direction / advice on Multi-tenant set-ups. I
> have configured Freeswitch as per the fine Multi-tenant guide on the
> Wiki, but this leaves some aspects such as transfers and IVR still in
> the default context.
>
> For example, transfering a call that is passed to features.xml, or
> connecting to an extension thorough demo_ivr will fail if the call is
> left in the default context.
>
> Would you recommend a separate features.xml for each tenants context? If
> the features provided by features.xml and the ivr are the same for all
> tenants, could you advise on the best way to implement one features.xml
> for all tenants?
>
> Kind regards,
> Marty Hickman
>
> How many contexts and tenants are you looking at? Also, will that change in
the future, that is, will you be adding tenants? Keep that in mind because
you might be looking at a good case for using mod_xml_curl...
-MC
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From bottleman at icf.org.ru  Wed Oct 28 17:26:34 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Thu, 29 Oct 2009 03:26:34 +0300 (MSK)
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910290314590.32176@srv.icf.org.ru>

On 2009-10-06 22:47 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:


Hi, now i have another questions about fs and not sip protocols,
call scheme like this - user1-sip->fs-sip->user2, 
as i see if in sip profile enable inbound-proxy-media SDP is goes
end-to-end if sip is on both call legs, if call goes throuch another 
protocol module, for example my mod_h323 (call scheme like 
user1-sip->fs-h323->someH323Ep) there is no way to negotiate 
codec because i have no SDP from called party for example and have no 
way to insert it on sip leg, or i don't see this way?

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From craig at overthewire.com.au  Wed Oct 28 17:40:05 2009
From: craig at overthewire.com.au (Craig Askings)
Date: Thu, 29 Oct 2009 10:40:05 +1000
Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.5pre3 is
	available for testing!
In-Reply-To: <87f2f3b90910281000r4c2021b8u7232320614737e30@mail.gmail.com>
References: <87f2f3b90910281000r4c2021b8u7232320614737e30@mail.gmail.com>
Message-ID: <8cc991dd0910281740w2c7a9d10q93dd5b5f0b4dbd6@mail.gmail.com>

Is the plan to run 1.0.5 as a stable branch with bug fix updates or will it
be the case of just follow the trunk like it is with any problems you
encounter in 1.0.4?

2009/10/29 Michael Collins <msc at freeswitch.org>

> Hello FreeSWITCHers,
>
> The latest FreeSWITCH version is now available for download on the files<http://files.freeswitch.org/freeswitch-1.0.5pre3.tar.gz>site. The announcement story is in the main
> FreeSWITCH <http://www.freeswitch.org/node/212> site. Please download and
> test, and then test some more. We need your feedback. The sooner we get your
> feedback, the more quickly we can roll the official 1.0.5.
>
> --
Craig Askings

Network Engineer | Over the Wire Pty Ltd
craig at overthewire.com.au | www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365
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From jason at jasonjgw.net  Wed Oct 28 18:26:23 2009
From: jason at jasonjgw.net (Jason White)
Date: Thu, 29 Oct 2009 12:26:23 +1100
Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.5pre3 is
 available for testing!
In-Reply-To: <8cc991dd0910281740w2c7a9d10q93dd5b5f0b4dbd6@mail.gmail.com>
References: <87f2f3b90910281000r4c2021b8u7232320614737e30@mail.gmail.com>
	<8cc991dd0910281740w2c7a9d10q93dd5b5f0b4dbd6@mail.gmail.com>
Message-ID: <20091029012622.GA3778@jdc.jasonjgw.net>

Craig Askings <craig at overthewire.com.au> wrote:
> Is the plan to run 1.0.5 as a stable branch with bug fix updates or will it
> be the case of just follow the trunk like it is with any problems you
> encounter in 1.0.4?

I can't speak for the developers, but I would expect the current policy to
continue: bug fixes are placed in trunk, and bug reports should be submitted
against whatever the current trunk revision is at the time of lodging the bug
reports.

The maintainers of FreeSWITCH are very effective, in general, at keeping SVN
trunk relatively stable.



From info at daccii.it  Wed Oct 28 14:05:17 2009
From: info at daccii.it (Albano Daniele Salvatore - Lavoro)
Date: Wed, 28 Oct 2009 22:05:17 +0100
Subject: [Freeswitch-users] Freeswitch seems to doesn't reknow dial tone
 after the first call using OpenZAP (analog spans)
Message-ID: <4AE8B20D.4040608@daccii.it>

Hi to all,

this is my first message here!

I'm trying to setup a freeswitch box using 1.0.4 version compiled from 
sources, ubuntu 8.04.03 lts with lastest updates and lastest zaptel 
modules/tools compiled from sources. As hardware i'm using an OpenVox 
A800P.

My problem is the following: if i do a call trought openzap it works, 
but when i hang down and try a second call it doesn't work!

Looking with ztmonitor and hearing recorded audio file it seems that 
freeswitch do the job on the first outgoing call but, on the second, it 
does nothing because it doesn't reknow dial tone (but it is there). If i 
call the freeswitch mananged number (i can answer or not, nothing 
change) i can do another call, but, in any case, the successive call 
doesn't start.

Looking with ztmonitor 1 -v i see that before the first call i get the 
special dial tone (a continue 425 frequency to state that the line is 
free to dial sended by the provider) but after the first call is done, 
before the second, i didn't see any special dial tone ... that appears 
for less than a sec when freeswitch starts to compose the number.

Here logs for first call
------
2009-10-28 21:38:39.616112 [INFO] ozmod_zt.c:636 Setting echo cancel to 
64 taps for 1:1
2009-10-28 21:38:39.616112 [DEBUG] ozmod_analog.c:450 Executing state 
handler on 1:1 for DIALING
2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:398 
(OpenZAP/1:1/xyz) Running State Change CS_INIT
2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:481 
(OpenZAP/1:1/xyz) State INIT
2009-10-28 21:38:39.616112 [DEBUG] mod_openzap.c:368 (OpenZAP/1:1/xyz) 
State Change CS_INIT -> CS_ROUTING
2009-10-28 21:38:39.616112 [DEBUG] switch_core_session.c:932 Send signal 
OpenZAP/1:1/xyz [BREAK]
2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:481 
(OpenZAP/1:1/xyz) State INIT going to sleep
2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:398 
(OpenZAP/1:1/xyz) Running State Change CS_ROUTING
2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:484 
(OpenZAP/1:1/xyz) State ROUTING
2009-10-28 21:38:39.616112 [DEBUG] mod_openzap.c:391 OpenZAP/1:1/xyz 
CHANNEL ROUTING
2009-10-28 21:38:39.616112 [DEBUG] switch_ivr_originate.c:63 
(OpenZAP/1:1/xyz) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2009-10-28 21:38:39.616112 [DEBUG] switch_core_session.c:932 Send signal 
OpenZAP/1:1/xyz [BREAK]
2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:484 
(OpenZAP/1:1/xyz) State ROUTING going to sleep
2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:398 
(OpenZAP/1:1/xyz) Running State Change CS_CONSUME_MEDIA
2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:503 
(OpenZAP/1:1/xyz) State CONSUME_MEDIA
2009-10-28 21:38:40.475897 [DEBUG] ozmod_analog.c:655 Detected tone DIAL 
on 1:1
2009-10-28 21:38:40.475897 [DEBUG] mod_openzap.c:1216 got FXO sig 1:1 
[TONE_DETECTED]
------

Here logs for second call
------
2009-10-28 21:39:07.105916 [INFO] ozmod_zt.c:636 Setting echo cancel to 
64 taps for 1:1
2009-10-28 21:39:07.105916 [DEBUG] ozmod_analog.c:450 Executing state 
handler on 1:1 for DIALING
2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:398 
(OpenZAP/1:1/xyz) Running State Change CS_INIT
2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:481 
(OpenZAP/1:1/xyz) State INIT
2009-10-28 21:39:07.118644 [DEBUG] mod_openzap.c:368 (OpenZAP/1:1/xyz) 
State Change CS_INIT -> CS_ROUTING
2009-10-28 21:39:07.118644 [DEBUG] switch_core_session.c:932 Send signal 
OpenZAP/1:1/xyz [BREAK]
2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:481 
(OpenZAP/1:1/xyz) State INIT going to sleep
2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:398 
(OpenZAP/1:1/xyz) Running State Change CS_ROUTING
2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:484 
(OpenZAP/1:1/xyz) State ROUTING
2009-10-28 21:39:07.118644 [DEBUG] mod_openzap.c:391 OpenZAP/1:1/xyz 
CHANNEL ROUTING
2009-10-28 21:39:07.118644 [DEBUG] switch_ivr_originate.c:63 
(OpenZAP/1:1/xyz) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2009-10-28 21:39:07.118644 [DEBUG] switch_core_session.c:932 Send signal 
OpenZAP/1:1/xyz [BREAK]
2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:484 
(OpenZAP/1:1/xyz) State ROUTING going to sleep
2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:398 
(OpenZAP/1:1/xyz) Running State Change CS_CONSUME_MEDIA
2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:503 
(OpenZAP/1:1/xyz) State CONSUME_MEDIA
------

Following my configs!

Really thanks for the help,


Best Regards,
Daniele

---
ZAPTEL.CONF
---
# Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#

# It must be in the module loading order


# Span 1: OPVXA1200/0 "OpenVox A1200P/A800P Board 1" (MASTER)
fxsks=1
fxsks=2
fxsks=3
fxsks=4
fxoks=5

# Global data

loadzone        = it
defaultzone     = it
---

ZT.CONF
---
[defaults]
codec_ms => 20
wink_ms => 150
flash_ms => 750
echo_cancel_level => 64
rxgain => 0.0
txgain => 0.0
---

OPENZAP.CONF
---
[span zt]
number => xyz
fxo-channel => 1
number => xyz
fxs-channel => 2
number => xyz
fxs-channel => 3
number => xyz
fxs-channel => 4

[span zt]
fxs-channel => 5
---

TONES.CONF
---
[it]
generate-dial => %(200, 200, 425);%(600, 1000, 425)
detect-dial => 425

generate-ring => %(1000, 4000, 425)
detect-ring => 425

generate-busy => %(500, 500, 425)
detect-busy => 425

generate-callwaiting-sas => %(400,100,425);%(250,100,425);%(150,0,425);
detect-callwaiting-sas => 425
---

OPENZAP.CONF.XML
---
<configuration name="openzap.conf" description="OpenZAP Configuration">
   <settings>
       <param name="debug" value="0"/>
       <param name="hold-music" value="$${moh_uri}"/>
   </settings>

   <analog_spans>

       <span id="1">
           <param name="tonegroup" value="it"/>
           <param name="digit-timeout" value="2000"/>
           <param name="max-digits" value="11"/>
           <param name="dialplan" value="XML"/>
           <param name="context" value="default"/>
           <param name="enable-callerid" value="true"/>
       </span>

       <span id="2">
           <param name="tonegroup" value="it"/>
           <param name="digit-timeout" value="2000"/>
           <param name="max-digits" value="11"/>
           <param name="dialplan" value="XML"/>
           <param name="context" value="default"/>
           <param name="enable-callerid" value="true"/>
       </span>

   </analog_spans>
</configuration>
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From davis.erwin at gmail.com  Wed Oct 28 19:00:14 2009
From: davis.erwin at gmail.com (Erwin Davis)
Date: Wed, 28 Oct 2009 19:00:14 -0700
Subject: [Freeswitch-users] error in loading spidermonkey
In-Reply-To: <8F404160-9658-4D67-A074-53022C2F5560@jerris.com>
References: <a38a91d50910281039pdf28a4fh536f44d238b9a954@mail.gmail.com>
	<8F404160-9658-4D67-A074-53022C2F5560@jerris.com>
Message-ID: <a38a91d50910281900s79dd1675se5f8bac4bc722ec6@mail.gmail.com>

fixed. Thanks,

On Wed, Oct 28, 2009 at 11:51 AM, Michael Jerris <mike at jerris.com> wrote:

> The libcurl is broken on your distro.  You can fix this by configuring
> with --without-libcurl which will use our working in tree copy instead
> of the broken one from your distro.
>
> Mike
>
> On Oct 28, 2009, at 1:39 PM, Erwin Davis wrote:
>
> > Hi, I got an error in loading mod_spidermonkey. my fs in ver 1.04
> > runs on fedora 8 VM. Any clue? thanks,
> >
> > [CRIT] switch_loadable_module.c:871 Error Loading module /usr/local/
> > freeswitch/mod/mod_spidermonkey.so
> > **/usr/lib/libnss3.so: undefined symbol: PR_UnloadLibrary**
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From egable+freeswitch at gmail.com  Wed Oct 28 19:21:05 2009
From: egable+freeswitch at gmail.com (Eliot Gable)
Date: Wed, 28 Oct 2009 22:21:05 -0400
Subject: [Freeswitch-users] SIP provider with extern rtp server
In-Reply-To: <8AC09649-2585-4BE7-A959-A7AC41650789@myrvold.org>
References: <A2D2E918-B494-4179-96BA-CA3A7717846C@myrvold.org>
	<bf6923ed0910270658t61bb6687ne0b7d3ec5090e2ff@mail.gmail.com>
	<CB8008BD-4066-4B5B-9043-214CFE938CD8@myrvold.org>
	<bf6923ed0910271034w6c7e0cc5t685a5377764755e6@mail.gmail.com>
	<FA2C693B-FCE1-4BDD-BCE5-302AC699CEDB@myrvold.org>
	<87f2f3b90910281112g6e72d22elcfd653991ecd50cc@mail.gmail.com>
	<8AC09649-2585-4BE7-A959-A7AC41650789@myrvold.org>
Message-ID: <bf6923ed0910281921uad8c525i850c08361b21538d@mail.gmail.com>

See that 200 OK that keeps coming in over and over and over and over
again? That's because they never received your ACK. If you can turn on
sofia loglevel to 9 and then watch where you send the ACK, you will
probably have your answer to why the other system did not receive it.
If you're still not sure what's going on, post another pastebin with
sofia loglevel set to 9.


On Wed, Oct 28, 2009 at 4:51 PM, Ivan C Myrvold <ivan at myrvold.org> wrote:
> Oh, what happened to it?
> Anyway, here is a new pb:
> http://pastebin.freeswitch.org/10867
> Ivan
> Den 28. okt. 2009 kl. 19:12 skrev Michael Collins:
>
>
> On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold <ivan at myrvold.org> wrote:
>>
>> Here is a debug log from a call from an internal phone out to an
>> external (my iPhone with nbr 91316356):
>> http://pastebin.freeswitch.org/108578
>>
>> Ivan
>>
> Uh... you wanna try that PB number again?
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero


From davis.erwin at gmail.com  Wed Oct 28 19:22:33 2009
From: davis.erwin at gmail.com (Erwin Davis)
Date: Wed, 28 Oct 2009 19:22:33 -0700
Subject: [Freeswitch-users] pass arguments into javascript
Message-ID: <a38a91d50910281922q59aad78s223851b06a6617ec@mail.gmail.com>

Hi, new to javascript. I tried to pass two arguments into javascript,

<action application="javascript" data="test.js $1 $2"/>


In test.js, I tried to use argv[1]  to retrieve $1 and argv[2] to retrieve
$2, however, the javascript test.js complained about argv[] as undefined
variables. How to retrieve the passing arguments in a javascript same as the
case above. Thanks,
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From egable+freeswitch at gmail.com  Wed Oct 28 20:23:09 2009
From: egable+freeswitch at gmail.com (Eliot Gable)
Date: Wed, 28 Oct 2009 23:23:09 -0400
Subject: [Freeswitch-users] pass arguments into javascript
In-Reply-To: <a38a91d50910281922q59aad78s223851b06a6617ec@mail.gmail.com>
References: <a38a91d50910281922q59aad78s223851b06a6617ec@mail.gmail.com>
Message-ID: <bf6923ed0910282023w5fbf8197uf1fd7fef7475f8ce@mail.gmail.com>

Should work fine. I use this:

var calling_num = argv[0];
var called_num = argv[1];

Are you sure you actually had valid data in $1 and $2? Try to call it
from the CLI:

jsrun test.js testvar1 testvar2



On Wed, Oct 28, 2009 at 10:22 PM, Erwin Davis <davis.erwin at gmail.com> wrote:
> Hi, new to javascript. I tried to pass two arguments into javascript,
>
> <action application="javascript" data="test.js $1 $2"/>
>
>
> In test.js, I tried to use argv[1]? to retrieve $1 and argv[2] to retrieve
> $2, however, the javascript test.js complained about argv[] as undefined
> variables. How to retrieve the passing arguments in a javascript same as the
> case above. Thanks,
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero


From davis.erwin at gmail.com  Thu Oct 29 07:48:32 2009
From: davis.erwin at gmail.com (Erwin Davis)
Date: Thu, 29 Oct 2009 10:48:32 -0400
Subject: [Freeswitch-users] pass arguments into javascript
In-Reply-To: <bf6923ed0910282023w5fbf8197uf1fd7fef7475f8ce@mail.gmail.com>
References: <a38a91d50910281922q59aad78s223851b06a6617ec@mail.gmail.com>
	<bf6923ed0910282023w5fbf8197uf1fd7fef7475f8ce@mail.gmail.com>
Message-ID: <a38a91d50910290748l2edb6df7xb1971a833bc7fabf@mail.gmail.com>

yes, I forgot the single quota in javascript statement. Now I saw the
aruments. Thanks,

On 10/28/09, Eliot Gable
<egable+freeswitch at gmail.com<egable%2Bfreeswitch at gmail.com>>
wrote:
>
> Should work fine. I use this:
>
> var calling_num = argv[0];
> var called_num = argv[1];
>
> Are you sure you actually had valid data in $1 and $2? Try to call it
> from the CLI:
>
> jsrun test.js testvar1 testvar2
>
>
>
>
> On Wed, Oct 28, 2009 at 10:22 PM, Erwin Davis <davis.erwin at gmail.com>
> wrote:
> > Hi, new to javascript. I tried to pass two arguments into javascript,
> >
> > <action application="javascript" data="test.js $1 $2"/>
> >
> >
> > In test.js, I tried to use argv[1]  to retrieve $1 and argv[2] to
> retrieve
> > $2, however, the javascript test.js complained about argv[] as undefined
> > variables. How to retrieve the passing arguments in a javascript same as
> the
> > case above. Thanks,
> >
>
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Eliot Gable
>
> "We do not inherit the Earth from our ancestors: we borrow it from our
> children." ~David Brower
>
> "I decided the words were too conservative for me. We're not borrowing
> from our children, we're stealing from them--and it's not even
> considered to be a crime." ~David Brower
>
> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
> live; not live to eat.) ~Marcus Tullius Cicero
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mi+thun at aldan.algebra.com  Thu Oct 29 07:56:33 2009
From: mi+thun at aldan.algebra.com (Mikhail T.)
Date: Thu, 29 Oct 2009 10:56:33 -0400
Subject: [Freeswitch-users] FreeSWITCH on FreeBSD
In-Reply-To: <4AE9BF36.3090201@gmail.com>
References: <4AE87EDB.90306@aldan.algebra.com> <4AE9BF36.3090201@gmail.com>
Message-ID: <4AE9AD21.9000300@aldan.algebra.com>

Richard E Neese ???????(??):
>>      * What hardware is known to work well on FreeBSD?
>>      * What else -- other than a computer with a phone card -- will I
>> need?
>>      * Do end-user phones have to be regular analog phones (all
>>        plugged-in to the computer), or IP-phones, such as made by Avaya
>>        (connecting via an Ethernet switch)?
>>      * How is the computer connecting to the world? Do I plug-in the
>>        incoming analog phone line(s) somewhere, or is it going to go
>>        through the Internet?
>>      * If the connection to the world is over the Internet, don't we
>> need
>>        an account with someone else, or is Verizon known to work with
>>        FreeSWITCH-based PBX-systems directly?
>>      * Should I not simply buy a used Avaya-system like this:
>>       
>> http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=110450339950&_trksid=p2759.l1259
>>
>>        What will I gain and what will I lose (other than personal
>>        satisfaction, of course) by going that route instead of building
>>        my own box?
Thank you very much, Richard, for you answering some of my questions. A
few remain, however:
> using freeswitch on bsd is free. nd all you kneed is phones like
> polycom soundpoint or aastra  or snome or cisco . Sip phones.
So, the phones connect -- via CAT6 cables -- to my network switch, and
talk to my FreeBSD server running FreeSWITCH. Ok. But how does the
server connect to the world? Somewhere between me talking into a Polycom
handset in New Jersey and my friend in hearing me in Ukraine, the audio
has to hit the telephone network... Where does this happen?

Do I plug-in a regular phone line into the same server? If so, what
hardware do I need inside it?
Or do I use a VOIP-provider -- connecting my server to them via the
Internet? And if so, which providers should I consider -- would Verizon
themselves qualify?

I ask, because the phone-service provided with Verizon's FIOS is already
a VOIP of some kind -- there is no copper line any more, it is all done
through optic cable... Verizon-provided device connects to that fiber on
one end, and gives you a phone-line on the other...

So it would be rather silly for my SIP-phones to convert the analog
speech into packets, then for the FreeSWITCH server to convert them back
into analog audio, and then for Verizon's equipment right in my house to
digitize back...

Thanks! Yours,

    -mi



From chris at cloudtel.com  Thu Oct 29 08:25:37 2009
From: chris at cloudtel.com (Chris Burns)
Date: Thu, 29 Oct 2009 11:25:37 -0400
Subject: [Freeswitch-users] Determining Frame Size
In-Reply-To: <4256bf830910280937n1f946518w423597fa6584c26f@mail.gmail.com>
References: <4256bf830910280937n1f946518w423597fa6584c26f@mail.gmail.com>
Message-ID: <200910291125.38079.chris@cloudtel.com>

Depends on your codec. Frame size = ptime. So it can vary from call to call, 
but most you will find it to be 20ms.

So if we are using speex-wb at 20ms, each frame is 0.02 seconds long. 50 
frames = 1 second of sound.

On October 28, 2009 12:37:56 pm Matthew Fong wrote:
> I noticed that the wait_for_silence's <silence_hits> <listen_hits> use
> number of frames as base values (rather than seconds or milliseconds). I'd
> like to wait for 5 seconds of consistent silence, but I'm wondering how I
> can determine the sample rate (and thus frames per second) for an existing
> call. Is there a variable I can read? Thanks.
>
> --matt




From anthony.minessale at gmail.com  Thu Oct 29 09:12:03 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 29 Oct 2009 11:12:03 -0500
Subject: [Freeswitch-users] Freeswitch seems to doesn't reknow dial tone
	after the first call using OpenZAP (analog spans)
In-Reply-To: <4AE8B20D.4040608@daccii.it>
References: <4AE8B20D.4040608@daccii.it>
Message-ID: <191c3a030910290912x9489784s4cc8e224363f3796@mail.gmail.com>

You may want to work with FreeSWITCH svn trunk for the time being.
There have been some changes and if you have a problem to fix it's easier to
debug with that version.


2009/10/28 Albano Daniele Salvatore - Lavoro <info at daccii.it>

> Hi to all,
>
> this is my first message here!
>
> I'm trying to setup a freeswitch box using 1.0.4 version compiled from
> sources, ubuntu 8.04.03 lts with lastest updates and lastest zaptel
> modules/tools compiled from sources. As hardware i'm using an OpenVox A800P.
>
> My problem is the following: if i do a call trought openzap it works, but
> when i hang down and try a second call it doesn't work!
>
> Looking with ztmonitor and hearing recorded audio file it seems that
> freeswitch do the job on the first outgoing call but, on the second, it does
> nothing because it doesn't reknow dial tone (but it is there). If i call the
> freeswitch mananged number (i can answer or not, nothing change) i can do
> another call, but, in any case, the successive call doesn't start.
>
> Looking with ztmonitor 1 -v i see that before the first call i get the
> special dial tone (a continue 425 frequency to state that the line is free
> to dial sended by the provider) but after the first call is done, before the
> second, i didn't see any special dial tone ... that appears for less than a
> sec when freeswitch starts to compose the number.
>
> Here logs for first call
> ------
> 2009-10-28 21:38:39.616112 [INFO] ozmod_zt.c:636 Setting echo cancel to 64
> taps for 1:1
> 2009-10-28 21:38:39.616112 [DEBUG] ozmod_analog.c:450 Executing state
> handler on 1:1 for DIALING
> 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:398
> (OpenZAP/1:1/xyz) Running State Change CS_INIT
> 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:481
> (OpenZAP/1:1/xyz) State INIT
> 2009-10-28 21:38:39.616112 [DEBUG] mod_openzap.c:368 (OpenZAP/1:1/xyz)
> State Change CS_INIT -> CS_ROUTING
> 2009-10-28 21:38:39.616112 [DEBUG] switch_core_session.c:932 Send signal
> OpenZAP/1:1/xyz [BREAK]
> 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:481
> (OpenZAP/1:1/xyz) State INIT going to sleep
> 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:398
> (OpenZAP/1:1/xyz) Running State Change CS_ROUTING
> 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:484
> (OpenZAP/1:1/xyz) State ROUTING
> 2009-10-28 21:38:39.616112 [DEBUG] mod_openzap.c:391 OpenZAP/1:1/xyz
> CHANNEL ROUTING
> 2009-10-28 21:38:39.616112 [DEBUG] switch_ivr_originate.c:63
> (OpenZAP/1:1/xyz) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2009-10-28 21:38:39.616112 [DEBUG] switch_core_session.c:932 Send signal
> OpenZAP/1:1/xyz [BREAK]
> 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:484
> (OpenZAP/1:1/xyz) State ROUTING going to sleep
> 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:398
> (OpenZAP/1:1/xyz) Running State Change CS_CONSUME_MEDIA
> 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:503
> (OpenZAP/1:1/xyz) State CONSUME_MEDIA
> 2009-10-28 21:38:40.475897 [DEBUG] ozmod_analog.c:655 Detected tone DIAL on
> 1:1
> 2009-10-28 21:38:40.475897 [DEBUG] mod_openzap.c:1216 got FXO sig 1:1
> [TONE_DETECTED]
> ------
>
> Here logs for second call
> ------
> 2009-10-28 21:39:07.105916 [INFO] ozmod_zt.c:636 Setting echo cancel to 64
> taps for 1:1
> 2009-10-28 21:39:07.105916 [DEBUG] ozmod_analog.c:450 Executing state
> handler on 1:1 for DIALING
> 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:398
> (OpenZAP/1:1/xyz) Running State Change CS_INIT
> 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:481
> (OpenZAP/1:1/xyz) State INIT
> 2009-10-28 21:39:07.118644 [DEBUG] mod_openzap.c:368 (OpenZAP/1:1/xyz)
> State Change CS_INIT -> CS_ROUTING
> 2009-10-28 21:39:07.118644 [DEBUG] switch_core_session.c:932 Send signal
> OpenZAP/1:1/xyz [BREAK]
> 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:481
> (OpenZAP/1:1/xyz) State INIT going to sleep
> 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:398
> (OpenZAP/1:1/xyz) Running State Change CS_ROUTING
> 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:484
> (OpenZAP/1:1/xyz) State ROUTING
> 2009-10-28 21:39:07.118644 [DEBUG] mod_openzap.c:391 OpenZAP/1:1/xyz
> CHANNEL ROUTING
> 2009-10-28 21:39:07.118644 [DEBUG] switch_ivr_originate.c:63
> (OpenZAP/1:1/xyz) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2009-10-28 21:39:07.118644 [DEBUG] switch_core_session.c:932 Send signal
> OpenZAP/1:1/xyz [BREAK]
> 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:484
> (OpenZAP/1:1/xyz) State ROUTING going to sleep
> 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:398
> (OpenZAP/1:1/xyz) Running State Change CS_CONSUME_MEDIA
> 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:503
> (OpenZAP/1:1/xyz) State CONSUME_MEDIA
> ------
>
> Following my configs!
>
> Really thanks for the help,
>
>
> Best Regards,
> Daniele
>
> ---
> ZAPTEL.CONF
> ---
> # Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
> # Zaptel Configuration File
> #
> # This file is parsed by the Zaptel Configurator, ztcfg
> #
>
> # It must be in the module loading order
>
>
> # Span 1: OPVXA1200/0 "OpenVox A1200P/A800P Board 1" (MASTER)
> fxsks=1
> fxsks=2
> fxsks=3
> fxsks=4
> fxoks=5
>
> # Global data
>
> loadzone        = it
> defaultzone     = it
> ---
>
> ZT.CONF
> ---
> [defaults]
> codec_ms => 20
> wink_ms => 150
> flash_ms => 750
> echo_cancel_level => 64
> rxgain => 0.0
> txgain => 0.0
> ---
>
> OPENZAP.CONF
> ---
> [span zt]
> number => xyz
> fxo-channel => 1
> number => xyz
> fxs-channel => 2
> number => xyz
> fxs-channel => 3
> number => xyz
> fxs-channel => 4
>
> [span zt]
> fxs-channel => 5
> ---
>
> TONES.CONF
> ---
> [it]
> generate-dial => %(200, 200, 425);%(600, 1000, 425)
> detect-dial => 425
>
> generate-ring => %(1000, 4000, 425)
> detect-ring => 425
>
> generate-busy => %(500, 500, 425)
> detect-busy => 425
>
> generate-callwaiting-sas => %(400,100,425);%(250,100,425);%(150,0,425);
> detect-callwaiting-sas => 425
> ---
>
> OPENZAP.CONF.XML
> ---
> <configuration name="openzap.conf" description="OpenZAP Configuration">
>  <settings>
>      <param name="debug" value="0"/>
>      <param name="hold-music" value="$${moh_uri}"/>
>  </settings>
>
>  <analog_spans>
>
>      <span id="1">
>          <param name="tonegroup" value="it"/>
>          <param name="digit-timeout" value="2000"/>
>          <param name="max-digits" value="11"/>
>          <param name="dialplan" value="XML"/>
>          <param name="context" value="default"/>
>          <param name="enable-callerid" value="true"/>
>      </span>
>
>      <span id="2">
>          <param name="tonegroup" value="it"/>
>          <param name="digit-timeout" value="2000"/>
>          <param name="max-digits" value="11"/>
>          <param name="dialplan" value="XML"/>
>          <param name="context" value="default"/>
>          <param name="enable-callerid" value="true"/>
>      </span>
>
>  </analog_spans>
> </configuration>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Thu Oct 29 10:14:05 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 29 Oct 2009 12:14:05 -0500
Subject: [Freeswitch-users] Employment Opportunities
Message-ID: <191c3a030910291014i741ec928p6ee001ac584c495d@mail.gmail.com>

We're looking for a few individuals to work with us on the CudaTel Product
powered by FreeSWITCH.

AJAX Developer:
Must have strong skill in HTTP/CGI/Javascript preferably some exp in jquery.
We would prefer if are not overly obsessed with any standards but respect
them enough to try to follow them in a well-balanced way.
Best if you live near Silicon Valley, Ann Arbor Michigan, or Milwaukee WI
but not mandatory.
Must be self-motivated. OCD a plus.  Willing to work late if necessary to
get things done right.
Working knowledge of Perl, DBI-X Catalyst and TT2 would make your life
easier.
Knowing a little about FS and Telephony would be nice.
Must be able to work diligently from a ssh terminal.

Back-end Developer:
Good knowledge of 2 or more Perl, DBI-X Catalyst and TT2.
Attention to detail, good debugging skills.
Best if you live near Silicon Valley, Ann Arbor Michigan, or Milwaukee WI
but not mandatory.
Must be self-motivated. OCD a plus.  Willing to work late if necessary to
get things done right.
Ability to work well with the guy in the other job description above.
Knowing a little about FS and Telephony would be nice.
Must be able to work diligently from a ssh terminal.


Both positions involve a hands-on "extreme programming" type environment
with shared screens and a permanent line of communication to your peers.
Send CR to jobs at freeswitch.org

-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Thu Oct 29 10:26:52 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 29 Oct 2009 10:26:52 -0700
Subject: [Freeswitch-users] Freeswitch seems to doesn't reknow dial tone
	after the first call using OpenZAP (analog spans)
In-Reply-To: <191c3a030910290912x9489784s4cc8e224363f3796@mail.gmail.com>
References: <4AE8B20D.4040608@daccii.it>
	<191c3a030910290912x9489784s4cc8e224363f3796@mail.gmail.com>
Message-ID: <87f2f3b90910291026u2c62ded6v5b9458a9c2e85877@mail.gmail.com>

On Thu, Oct 29, 2009 at 9:12 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> You may want to work with FreeSWITCH svn trunk for the time being.
> There have been some changes and if you have a problem to fix it's easier
> to debug with that version.
>
>
> 2009/10/28 Albano Daniele Salvatore - Lavoro <info at daccii.it>
>
>> Hi to all,
>>
>> this is my first message here!
>>
>> Welcome!

Like Tony said, the latest SVN trunk is probably what you want. We are
already doing pre-releases of 1.0.5. SVN trunk is extremely stable. Also,
you can get realtime help via IRC at irc.freenode.net, channels #freeswitch
and #openzap.

-MC
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From r.wilczynski at gmail.com  Thu Oct 29 02:46:06 2009
From: r.wilczynski at gmail.com (=?ISO-8859-2?Q?Robert_Wilczy=F1ski?=)
Date: Thu, 29 Oct 2009 10:46:06 +0100
Subject: [Freeswitch-users] Gateway registration fails: IP used for realm in
	place of configured realm value
Message-ID: <7e07d6d80910290246o2daea1e7o2b8be91b137ad2c8@mail.gmail.com>

Hi,

We have two FS instances production (prod) and gateway (gw)

when prod starts it tries to register a gateway:

<include>
??? <gateway name="fs_gateway">
??? ??? <param name="realm" value="24ccdomain"/>
??? ??? <param name="proxy" value="$${local_ip_v4}:15060" />
??? ??? <param name="username" value="24cc"/>
??? ??? <param name="password" value="24ccpass"/>
??? ??? <param name="expire-seconds" value="60"/>
??? ??? <param name="register" value="true"/>
??? </gateway>
</include>

and fails:

2009-10-29 10:19:45.449220 [NOTICE] sofia_glue.c:3534 Reload XML [Success]
2009-10-29 10:19:45.450221 [INFO] mod_enum.c:808 ENUM Reloaded
2009-10-29 10:19:45.450221 [NOTICE] sofia_reg.c:2117 Added gateway
'fs_gateway' to profile 'external'
2009-10-29 10:19:45.450221 [NOTICE] sofia.c:3093 Adding Alias
[outbound] for profile [external]
2009-10-29 10:19:45.450221 [NOTICE] sofia.c:3093 Adding Alias [nat]
for profile [external]
2009-10-29 10:19:45.451221 [NOTICE] sofia.c:3108 Started Profile
external [sofia_reg_external]
2009-10-29 10:19:45.452221 [INFO] switch_time.c:661 Timezone reloaded
530 definitions
2009-10-29 10:19:45.628743 [NOTICE] sofia_reg.c:333 Registering fs_gateway
2009-10-29 10:19:45.996790 [NOTICE] sofia.c:1202 Waiting for worker thread
2009-10-29 10:19:45.998790 [NOTICE] sofia.c:1729 Adding Alias
[192.168.1.20] for profile [internal]
2009-10-29 10:19:45.998790 [NOTICE] sofia.c:3093 Adding Alias
[default] for profile [internal]
2009-10-29 10:19:45.998790 [NOTICE] sofia.c:3108 Started Profile
internal [sofia_reg_internal]
2009-10-29 10:19:53.354724 [ERR] sofia_reg.c:1425 fs_gateway
Registration Failed with status Forbidden [403]. failure #1
2009-10-29 10:19:53.440735 [WARNING] sofia_reg.c:364 fs_gateway Failed
Registration, setting retry to 60 seconds.
2009-10-29 10:20:53.108812 [NOTICE] sofia_reg.c:333 Registering fs_gateway
2009-10-29 10:20:53.116813 [ERR] sofia_reg.c:1425 fs_gateway
Registration Failed with status Forbidden [403]. failure #2
2009-10-29 10:20:53.208825 [WARNING] sofia_reg.c:364 fs_gateway Failed
Registration, setting retry to 90 seconds.

gw is configured like that:

<include>
??? <domain name="24ccdomain">
??? ??? <params>
??? ??? ??? <param name="dial-string"
value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
??? ??? </params>

??? ??? <groups>
??? ??? ??? <group name="default">
??? ??? ??? ??? <users>
??? ??? ??? ??? ??? <user id="24cc">
??? ??? ??? ??? ??? ??? <params>
??? ??? ??? ??? ??? ??? ??? <param name="password" value="24ccpass"/>
??? ??? ??? ??? ??? ??? </params>
??? ??? ??? ??? ??? ??? <variables>
??? ??? ??? ??? ??? ??? ??? <variable name="user_context" value="default"/>
??? ??? ??? ??? ??? ??? ??? <variable name="effective_caller_id_name"
value="24 CallCenter"/>
??? ??? ??? ??? ??? ??? </variables>
??? ??? ??? ??? ??? </user>
??? ??? ??? ??? </users>
??? ??? ??? </group>
??? ??? </groups>
??? </domain>
</include>

and outputs:

2009-10-29 10:11:29.165700 [WARNING] sofia_reg.c:1737 Can't find user
[24cc at 192.168.1.20]
You must define a domain called '192.168.1.20' in your directory and
add a user with the id="24cc" attribute
and you must configure your device to use the proper domain in it's
authentication credentials.

on each registration attempt from prod.

This worked perfectly fine with one of previous revisions but started
failing on 15257. Using siptrace we were able to narrow down the issue
to prod sending an IP address as realm instead of gateway configured
realm (as it did in previous revision):

Authorization: Digest username="24cc", realm="192.168.1.20",
nonce="4d82d624-c884-6340-8f7f-a4575f2583bd",
cnonce="Es23bj8MEi29T7UQ80i7sw", algorithm=MD5,
uri="sip:192.168.1.20:15060",
response="afea161876aad137657bf1f923d18659", qop=auth, nc=00000004

Is there any configuration setting we are missing that would bring
back previous behavior?

--
Best regards,
Robert Wilczynski


From brian at freeswitch.org  Thu Oct 29 11:31:26 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 29 Oct 2009 13:31:26 -0500
Subject: [Freeswitch-users] Gateway registration fails: IP used for
	realm in place of configured realm value
In-Reply-To: <7e07d6d80910290246o2daea1e7o2b8be91b137ad2c8@mail.gmail.com>
References: <7e07d6d80910290246o2daea1e7o2b8be91b137ad2c8@mail.gmail.com>
Message-ID: <6E1B3448-2BD7-492C-B129-59B2D813B8EE@freeswitch.org>

It's because you need to set the from-user and from-domain because  
FreeSWITCH don't care about the realm because WE provide you the realm  
in our challenge... you only provide realm for broken providers such  
as BroadVoice.

remove realm from your setting.  Add from-user and from-domain

/b

On Oct 29, 2009, at 4:46 AM, Robert Wilczy?ski wrote:

> Hi,
>
> We have two FS instances production (prod) and gateway (gw)
>
> when prod starts it tries to register a gateway:
>
> <include>
>     <gateway name="fs_gateway">
>         <param name="realm" value="24ccdomain"/>
>         <param name="proxy" value="$${local_ip_v4}:15060" />
>         <param name="username" value="24cc"/>
>         <param name="password" value="24ccpass"/>
>         <param name="expire-seconds" value="60"/>
>         <param name="register" value="true"/>
>     </gateway>
> </include>
>
> and fails:
>
> 2009-10-29 10:19:45.449220 [NOTICE] sofia_glue.c:3534 Reload XML  
> [Success]
> 2009-10-29 10:19:45.450221 [INFO] mod_enum.c:808 ENUM Reloaded
> 2009-10-29 10:19:45.450221 [NOTICE] sofia_reg.c:2117 Added gateway
> 'fs_gateway' to profile 'external'
> 2009-10-29 10:19:45.450221 [NOTICE] sofia.c:3093 Adding Alias
> [outbound] for profile [external]
> 2009-10-29 10:19:45.450221 [NOTICE] sofia.c:3093 Adding Alias [nat]
> for profile [external]
> 2009-10-29 10:19:45.451221 [NOTICE] sofia.c:3108 Started Profile
> external [sofia_reg_external]
> 2009-10-29 10:19:45.452221 [INFO] switch_time.c:661 Timezone reloaded
> 530 definitions
> 2009-10-29 10:19:45.628743 [NOTICE] sofia_reg.c:333 Registering  
> fs_gateway
> 2009-10-29 10:19:45.996790 [NOTICE] sofia.c:1202 Waiting for worker  
> thread
> 2009-10-29 10:19:45.998790 [NOTICE] sofia.c:1729 Adding Alias
> [192.168.1.20] for profile [internal]
> 2009-10-29 10:19:45.998790 [NOTICE] sofia.c:3093 Adding Alias
> [default] for profile [internal]
> 2009-10-29 10:19:45.998790 [NOTICE] sofia.c:3108 Started Profile
> internal [sofia_reg_internal]
> 2009-10-29 10:19:53.354724 [ERR] sofia_reg.c:1425 fs_gateway
> Registration Failed with status Forbidden [403]. failure #1
> 2009-10-29 10:19:53.440735 [WARNING] sofia_reg.c:364 fs_gateway Failed
> Registration, setting retry to 60 seconds.
> 2009-10-29 10:20:53.108812 [NOTICE] sofia_reg.c:333 Registering  
> fs_gateway
> 2009-10-29 10:20:53.116813 [ERR] sofia_reg.c:1425 fs_gateway
> Registration Failed with status Forbidden [403]. failure #2
> 2009-10-29 10:20:53.208825 [WARNING] sofia_reg.c:364 fs_gateway Failed
> Registration, setting retry to 90 seconds.
>
> gw is configured like that:
>
> <include>
>     <domain name="24ccdomain">
>         <params>
>             <param name="dial-string"
> value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact($ 
> {dialed_user}@${dialed_domain})}"/>
>         </params>
>
>         <groups>
>             <group name="default">
>                 <users>
>                     <user id="24cc">
>                         <params>
>                             <param name="password" value="24ccpass"/>
>                         </params>
>                         <variables>
>                             <variable name="user_context"  
> value="default"/>
>                             <variable name="effective_caller_id_name"
> value="24 CallCenter"/>
>                         </variables>
>                     </user>
>                 </users>
>             </group>
>         </groups>
>     </domain>
> </include>
>
> and outputs:
>
> 2009-10-29 10:11:29.165700 [WARNING] sofia_reg.c:1737 Can't find user
> [24cc at 192.168.1.20]
> You must define a domain called '192.168.1.20' in your directory and
> add a user with the id="24cc" attribute
> and you must configure your device to use the proper domain in it's
> authentication credentials.
>
> on each registration attempt from prod.
>
> This worked perfectly fine with one of previous revisions but started
> failing on 15257. Using siptrace we were able to narrow down the issue
> to prod sending an IP address as realm instead of gateway configured
> realm (as it did in previous revision):
>
> Authorization: Digest username="24cc", realm="192.168.1.20",
> nonce="4d82d624-c884-6340-8f7f-a4575f2583bd",
> cnonce="Es23bj8MEi29T7UQ80i7sw", algorithm=MD5,
> uri="sip:192.168.1.20:15060",
> response="afea161876aad137657bf1f923d18659", qop=auth, nc=00000004
>
> Is there any configuration setting we are missing that would bring
> back previous behavior?
>
> --
> Best regards,
> Robert Wilczynski
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From larclap at yahoo.com  Thu Oct 29 16:20:19 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Thu, 29 Oct 2009 16:20:19 -0700
Subject: [Freeswitch-users] Error 488- Incompatible Destination
Message-ID: <010a01ca58ee$622ae700$2680b500$@com>

I must have been playing with the settings on a SNOM 320 because suddenly it
cannot connect to FreeSwitch. 

 

I have looked at similar posts and it seems like it must be a codec problem
on the SNOM, but I cannot figure it out. 

 

Can someone point me in the right direction?

http://pastebin.freeswitch.org/10887

 

Thanks, Lars

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From brian at freeswitch.org  Thu Oct 29 16:37:29 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 29 Oct 2009 18:37:29 -0500
Subject: [Freeswitch-users] Error 488- Incompatible Destination
In-Reply-To: <010a01ca58ee$622ae700$2680b500$@com>
References: <010a01ca58ee$622ae700$2680b500$@com>
Message-ID: <849DE6E6-1191-47C2-9151-7A97C8B35A57@freeswitch.org>

No its not ALL of the sudden... Please check the SRTP settings on the  
identity.  Thats why you're getting that... the phone defaults to  
sending crypto in AVP which is invalid... we reject it.

This has been this way for some time I suspect you have updated your  
snom recently or aren't on 7.1.35 or higher which has the options to  
fix this on the phone.  set it to SAVP

RTP/SAVP: to off will fix it. Unless you're doing SRTP then set it to  
mandatory.

/b

On Oct 29, 2009, at 6:20 PM, Lars Zeb wrote:

> I must have been playing with the settings on a SNOM 320 because  
> suddenly it cannot connect to FreeSwitch.
>
> I have looked at similar posts and it seems like it must be a codec  
> problem on the SNOM, but I cannot figure it out.
>
> Can someone point me in the right direction?
>
> http://pastebin.freeswitch.org/10887
>
> Thanks, Lars

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From anthony.minessale at gmail.com  Thu Oct 29 16:55:42 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 29 Oct 2009 18:55:42 -0500
Subject: [Freeswitch-users] Error 488- Incompatible Destination
In-Reply-To: <849DE6E6-1191-47C2-9151-7A97C8B35A57@freeswitch.org>
References: <010a01ca58ee$622ae700$2680b500$@com>
	<849DE6E6-1191-47C2-9151-7A97C8B35A57@freeswitch.org>
Message-ID: <191c3a030910291655i2ee1472r642352bdeea5dfbf@mail.gmail.com>

ironically, snom reported this as a bug to us when we used to tolerate it so
they made us fix it thus breaking it on their own phones.


On Thu, Oct 29, 2009 at 6:37 PM, Brian West <brian at freeswitch.org> wrote:

> No its not ALL of the sudden... Please check the SRTP settings on the
> identity.  Thats why you're getting that... the phone defaults to sending
> crypto in AVP which is invalid... we reject it.
>
> This has been this way for some time I suspect you have updated your snom
> recently or aren't on 7.1.35 or higher which has the options to fix this on
> the phone.  set it to SAVP
>
> RTP/SAVP: to off will fix it. Unless you're doing SRTP then set it to
> mandatory.
>
> /b
>
> On Oct 29, 2009, at 6:20 PM, Lars Zeb wrote:
>
> I must have been playing with the settings on a SNOM 320 because suddenly
> it cannot connect to FreeSwitch.
>
> I have looked at similar posts and it seems like it must be a codec problem
> on the SNOM, but I cannot figure it out.
>
> Can someone point me in the right direction?
>
> http://pastebin.freeswitch.org/10887
>
> Thanks, Lars
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From larclap at yahoo.com  Thu Oct 29 17:30:26 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Thu, 29 Oct 2009 17:30:26 -0700
Subject: [Freeswitch-users] Problem with inbound call answered but
	no	sound
In-Reply-To: <8D1A994D-2DCD-4016-9BEF-D6FBD3FBAA1E@freeswitch.org>
References: <016e01ca56ad$56362380$02a26a80$@com>	<168575F8-6507-4AD2-B448-2D6FDA0152B6@freeswitch.org>	<018801ca56b2$1e63e0f0$5b2ba2d0$@com>	<BACDC899-CE4F-42E1-B2A0-C0C0E31F8787@freeswitch.org>	<00d601ca571e$519f7300$f4de5900$@com>
	<8D1A994D-2DCD-4016-9BEF-D6FBD3FBAA1E@freeswitch.org>
Message-ID: <012101ca58f8$3d1b3f50$b751bdf0$@com>

Updated to v15279 and all is OK now.

 

Thanks, Brian

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Wednesday, October 28, 2009 7:13 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Problem with inbound call answered but no
sound

 

What kind of router are you behind?

 

/b

 

On Oct 27, 2009, at 10:58 AM, Lars Zeb wrote:





Thanks for the reply, Brian.

 

Did something in FS change between v15183 and v15225 to make this occur? I
ask because this same configuration worked OK in the earlier version.

 

Lars

 

 

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From larclap at yahoo.com  Thu Oct 29 17:40:57 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Thu, 29 Oct 2009 17:40:57 -0700
Subject: [Freeswitch-users] Error 488- Incompatible Destination
In-Reply-To: <849DE6E6-1191-47C2-9151-7A97C8B35A57@freeswitch.org>
References: <010a01ca58ee$622ae700$2680b500$@com>
	<849DE6E6-1191-47C2-9151-7A97C8B35A57@freeswitch.org>
Message-ID: <012601ca58f9$a5b1e2c0$f115a840$@com>

I meant "suddenly" after I had been messing with the SNOM's settings. The
firmware is 7.3.14. The SRTP stuff is:

 

Symmetrical RTP:       off 

RTP Encryption:        on 

Dynamic G.726 payload: on 

SRTP Auth-tag:         AES-32 

RTP/SAVP:             off

 

Lars

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, October 29, 2009 4:37 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Error 488- Incompatible Destination

 

No its not ALL of the sudden... Please check the SRTP settings on the
identity.  Thats why you're getting that... the phone defaults to sending
crypto in AVP which is invalid... we reject it.

 

This has been this way for some time I suspect you have updated your snom
recently or aren't on 7.1.35 or higher which has the options to fix this on
the phone.  set it to SAVP

 

RTP/SAVP: to off will fix it. Unless you're doing SRTP then set it to
mandatory.

 

/b

 

On Oct 29, 2009, at 6:20 PM, Lars Zeb wrote:





I must have been playing with the settings on a SNOM 320 because suddenly it
cannot connect to FreeSwitch.

 

I have looked at similar posts and it seems like it must be a codec problem
on the SNOM, but I cannot figure it out.

 

Can someone point me in the right direction?

http://pastebin.freeswitch.org/10887

 

Thanks, Lars

 

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From brian at freeswitch.org  Thu Oct 29 18:06:07 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 29 Oct 2009 20:06:07 -0500
Subject: [Freeswitch-users] Error 488- Incompatible Destination
In-Reply-To: <012601ca58f9$a5b1e2c0$f115a840$@com>
References: <010a01ca58ee$622ae700$2680b500$@com>
	<849DE6E6-1191-47C2-9151-7A97C8B35A57@freeswitch.org>
	<012601ca58f9$a5b1e2c0$f115a840$@com>
Message-ID: <CB02F8C3-FC81-41C5-9BD7-DEE5C126F1F0@freeswitch.org>

SEE THIS TAG?  If you have SRTP on you MUST put this to Optional if  
you care for it to work... read the debug a little better it'll tell  
you to reference an RFC on this also.

/b

On Oct 29, 2009, at 7:40 PM, Lars Zeb wrote:

> RTP/SAVP:             off

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From JCasale at activenetwerx.com  Thu Oct 29 19:22:48 2009
From: JCasale at activenetwerx.com (Joseph L. Casale)
Date: Fri, 30 Oct 2009 02:22:48 +0000
Subject: [Freeswitch-users] Error building rpm's from trunk
Message-ID: <ABF9510930E1374BA4B4C61A01104FBDC89637@Monterossa.activenetwerx.local>

I fetched a copy of trunk, tar and gzipped it up and placed it in my
SOURCES dir, I edited the bzip extension in the spec file and tried to
build rpms but got this error:


Checking for unpackaged file(s): /usr/lib/rpm/check-files /var/tmp/freeswitch-1.0.4-1-root-rpm-builder
error: Installed (but unpackaged) file(s) found:
   /opt/freeswitch/mod/ozmod_sangoma_boost.so

RPM build errors:
    Installed (but unpackaged) file(s) found:
   /opt/freeswitch/mod/ozmod_sangoma_boost.so


Anyone know if trunk successfully builds into rpms on CentOSx86 or is
this an issue on my end?

Thanks!
jlc


From brian at freeswitch.org  Thu Oct 29 19:39:57 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 29 Oct 2009 21:39:57 -0500
Subject: [Freeswitch-users] Error building rpm's from trunk
In-Reply-To: <ABF9510930E1374BA4B4C61A01104FBDC89637@Monterossa.activenetwerx.local>
References: <ABF9510930E1374BA4B4C61A01104FBDC89637@Monterossa.activenetwerx.local>
Message-ID: <67478BE8-E29B-4A35-B0E4-2746DF09BB29@freeswitch.org>

perl -i -pe 's/ss7_boost/sangoma_boost/g' freeswitch.spec

or svn up.

/b

On Oct 29, 2009, at 9:22 PM, Joseph L. Casale wrote:

> I fetched a copy of trunk, tar and gzipped it up and placed it in my
> SOURCES dir, I edited the bzip extension in the spec file and tried to
> build rpms but got this error:
>
>
> Checking for unpackaged file(s): /usr/lib/rpm/check-files /var/tmp/ 
> freeswitch-1.0.4-1-root-rpm-builder
> error: Installed (but unpackaged) file(s) found:
>   /opt/freeswitch/mod/ozmod_sangoma_boost.so
>
> RPM build errors:
>    Installed (but unpackaged) file(s) found:
>   /opt/freeswitch/mod/ozmod_sangoma_boost.so
>
>
> Anyone know if trunk successfully builds into rpms on CentOSx86 or is
> this an issue on my end?
>
> Thanks!
> jlc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From brian at freeswitch.org  Thu Oct 29 19:41:24 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 29 Oct 2009 21:41:24 -0500
Subject: [Freeswitch-users] CELT Codec on Friday's meeting
Message-ID: <26F3FA5E-ED72-4EAD-ACA4-C3D037F1D655@freeswitch.org>

Please update to 1.0.5pre4 if you plan on using the CELT codec on the  
conference tomorrow.  I'm rolling the pre4 tarball right now.

Thanks,
Brian



From ujjval at simplesignal.com  Thu Oct 29 21:34:30 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Thu, 29 Oct 2009 21:34:30 -0700
Subject: [Freeswitch-users] Setting up Conference with Moderator
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local>
	<3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local>
	<E97CD3D0-52FF-44D1-BB89-23816045CE21@gmail.com>
	<71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>,
	<114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local>
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local>

Rob:

 For some reason, I don't see the DTMF appear on the fs_CLI when using the below configuration....so it basically timesout.

UK



-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ujjval Karihaloo
Sent: Monday, October 26, 2009 9:21 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator

Thx a lot Rob, reading the wiki your way or using IVR seems correct..
===============
The wiki also says that the wait-mod might be  "used in conjunction 
with an IVR where the moderators are authenticated with an extra pass-
code", which is what I did.  I guess that's why I didn't understand 
the point of the +pin.
======================

I will try it out.

Again thx a lot for your help. Will keep everyone posted.

________________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Forman [rob4manhere at gmail.com]
Sent: Friday, October 23, 2009 12:22 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator

I just re-tested with the pin in my dial plan:

   <action application="conference" data="conference 123456 at default
+flags{}+1234" />

And it doesn't challenge me for the pin.  I just drop right in.  I
figured this is how it was intended, since the wiki says the pin is
set initially and only challenged in later attempts [by future callers]:

"The first time a conference name (confname) is used, it will be
created on demand, and the pin will be set to what ever is specified
at that time: the pin in the data string if specified, or if not, the
"pin" setting in the conference profile, and if that is also
unspecified, then there is no pin protection. Any later attempt to
join the conference must specify the same pin number, if one existed
when it was created. "


The wiki also says that the wait-mod might be  "used in conjunction
with an IVR where the moderators are authenticated with an extra pass-
code", which is what I did.  I guess that's why I didn't understand
the point of the +pin.

I'm sure there's a scenario where its used and useful, the wiki just
doesn't explain it.

Rob

On Oct 23, 2009, at 12:43 PM, Brian West wrote:

> Well first off you're not defining a pine here...
>
> confname at profilename+flags{mute|deaf|waste|moderator}+[conference pin
> number]
>
> That might be why its not asking for a pin.
>
> /b
>
> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:
>
>>      <entry action="menu-exec-app" digits="1" param="conference
>> 123456 at default+flags{}" />
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From velu.technical at gmail.com  Thu Oct 29 23:57:41 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Fri, 30 Oct 2009 12:27:41 +0530
Subject: [Freeswitch-users] Flushing the Event buffer in Perl Event Socket
Message-ID: <1452e2980910292357i38379319ib4283f7189d05abe@mail.gmail.com>

Dear All,
      I receiving the events in while loop by using recvEventTimed method in
ESL.pm. I have to flush that Event buffer after some particular time. How
can I do it?

Thanks,
Velusamy
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From codecomplete at free.fr  Fri Oct 30 05:06:17 2009
From: codecomplete at free.fr (Fred-145)
Date: Fri, 30 Oct 2009 05:06:17 -0700 (PDT)
Subject: [Freeswitch-users] PBX vs. SIP proxy?
In-Reply-To: <4AD914FE.4020709@gmail.com>
References: <25920788.post@talk.nabble.com> <4AD914FE.4020709@gmail.com>
Message-ID: <26128938.post@talk.nabble.com>


Thanks Meftah for the help. So it looks like an SIP proxy is just to handle
the registration process, while a PBX offers services like ringing phones,
setting up calls, handling voice mail, etc.
-- 
View this message in context: http://old.nabble.com/PBX-vs.-SIP-proxy--tp25920788p26128938.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From rob4manhere at gmail.com  Fri Oct 30 06:22:55 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Fri, 30 Oct 2009 08:22:55 -0500
Subject: [Freeswitch-users] Setting up Conference with Moderator
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local>
	<3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local>
	<E97CD3D0-52FF-44D1-BB89-23816045CE21@gmail.com>
	<71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>,
	<114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local>
	<3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local>
Message-ID: <F88E7B78-4CB8-45BD-98CE-81195862A891@gmail.com>

I've never had any problem with that.  Is your logging at debug level  
so you can see the RECV DTFM in the log/fs_cli?  Are you calling from  
a SIP phone on the pbx, or via a PSTN provider?  Maybe your provider  
isn't passing them through.

Make sure your logging is turned up then try something simpler, like  
calling the echo application, and see if DTFM comes through.

Rob

On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote:

> Rob:
>
> For some reason, I don't see the DTMF appear on the fs_CLI when  
> using the below configuration....so it basically timesout.
>
> UK
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Ujjval Karihaloo
> Sent: Monday, October 26, 2009 9:21 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
>
> Thx a lot Rob, reading the wiki your way or using IVR seems correct..
> ===============
> The wiki also says that the wait-mod might be  "used in conjunction
> with an IVR where the moderators are authenticated with an extra pass-
> code", which is what I did.  I guess that's why I didn't understand
> the point of the +pin.
> ======================
>
> I will try it out.
>
> Again thx a lot for your help. Will keep everyone posted.
>
> ________________________________________
> From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Rob Forman [rob4manhere at gmail.com]
> Sent: Friday, October 23, 2009 12:22 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
>
> I just re-tested with the pin in my dial plan:
>
>   <action application="conference" data="conference 123456 at default
> +flags{}+1234" />
>
> And it doesn't challenge me for the pin.  I just drop right in.  I
> figured this is how it was intended, since the wiki says the pin is
> set initially and only challenged in later attempts [by future  
> callers]:
>
> "The first time a conference name (confname) is used, it will be
> created on demand, and the pin will be set to what ever is specified
> at that time: the pin in the data string if specified, or if not, the
> "pin" setting in the conference profile, and if that is also
> unspecified, then there is no pin protection. Any later attempt to
> join the conference must specify the same pin number, if one existed
> when it was created. "
>
>
> The wiki also says that the wait-mod might be  "used in conjunction
> with an IVR where the moderators are authenticated with an extra pass-
> code", which is what I did.  I guess that's why I didn't understand
> the point of the +pin.
>
> I'm sure there's a scenario where its used and useful, the wiki just
> doesn't explain it.
>
> Rob
>
> On Oct 23, 2009, at 12:43 PM, Brian West wrote:
>
>> Well first off you're not defining a pine here...
>>
>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference pin
>> number]
>>
>> That might be why its not asking for a pin.
>>
>> /b
>>
>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:
>>
>>>     <entry action="menu-exec-app" digits="1" param="conference
>>> 123456 at default+flags{}" />
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From anthony.minessale at gmail.com  Fri Oct 30 06:50:02 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 30 Oct 2009 08:50:02 -0500
Subject: [Freeswitch-users] Flushing the Event buffer in Perl Event
	Socket
In-Reply-To: <1452e2980910292357i38379319ib4283f7189d05abe@mail.gmail.com>
References: <1452e2980910292357i38379319ib4283f7189d05abe@mail.gmail.com>
Message-ID: <191c3a030910300650w7b80568eu4c41c805b9372acc@mail.gmail.com>

read them in a timed loop of some small number of MS until you get a timeout
meaning you have flushed them all.


On Fri, Oct 30, 2009 at 1:57 AM, velusamy velu <velu.technical at gmail.com>wrote:

> Dear All,
>       I receiving the events in while loop by using recvEventTimed method
> in ESL.pm. I have to flush that Event buffer after some particular time. How
> can I do it?
>
> Thanks,
> Velusamy
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From ujjval at simplesignal.com  Fri Oct 30 07:43:07 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Fri, 30 Oct 2009 07:43:07 -0700
Subject: [Freeswitch-users] Setting up Conference with Moderator
In-Reply-To: <F88E7B78-4CB8-45BD-98CE-81195862A891@gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local>
	<3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local>
	<E97CD3D0-52FF-44D1-BB89-23816045CE21@gmail.com>
	<71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>,
	<114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local>
	<3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local>
	<F88E7B78-4CB8-45BD-98CE-81195862A891@gmail.com>
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local>

It's strange... a tcpdump tells me that there is no DTMF from my provider when using IVR, but when I call into a TN that goes directly into the Conference App, I see DTMF from the provider.



-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Forman
Sent: Friday, October 30, 2009 7:23 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator

I've never had any problem with that.  Is your logging at debug level  
so you can see the RECV DTFM in the log/fs_cli?  Are you calling from  
a SIP phone on the pbx, or via a PSTN provider?  Maybe your provider  
isn't passing them through.

Make sure your logging is turned up then try something simpler, like  
calling the echo application, and see if DTFM comes through.

Rob

On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote:

> Rob:
>
> For some reason, I don't see the DTMF appear on the fs_CLI when  
> using the below configuration....so it basically timesout.
>
> UK
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Ujjval Karihaloo
> Sent: Monday, October 26, 2009 9:21 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
>
> Thx a lot Rob, reading the wiki your way or using IVR seems correct..
> ===============
> The wiki also says that the wait-mod might be  "used in conjunction
> with an IVR where the moderators are authenticated with an extra pass-
> code", which is what I did.  I guess that's why I didn't understand
> the point of the +pin.
> ======================
>
> I will try it out.
>
> Again thx a lot for your help. Will keep everyone posted.
>
> ________________________________________
> From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Rob Forman [rob4manhere at gmail.com]
> Sent: Friday, October 23, 2009 12:22 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
>
> I just re-tested with the pin in my dial plan:
>
>   <action application="conference" data="conference 123456 at default
> +flags{}+1234" />
>
> And it doesn't challenge me for the pin.  I just drop right in.  I
> figured this is how it was intended, since the wiki says the pin is
> set initially and only challenged in later attempts [by future  
> callers]:
>
> "The first time a conference name (confname) is used, it will be
> created on demand, and the pin will be set to what ever is specified
> at that time: the pin in the data string if specified, or if not, the
> "pin" setting in the conference profile, and if that is also
> unspecified, then there is no pin protection. Any later attempt to
> join the conference must specify the same pin number, if one existed
> when it was created. "
>
>
> The wiki also says that the wait-mod might be  "used in conjunction
> with an IVR where the moderators are authenticated with an extra pass-
> code", which is what I did.  I guess that's why I didn't understand
> the point of the +pin.
>
> I'm sure there's a scenario where its used and useful, the wiki just
> doesn't explain it.
>
> Rob
>
> On Oct 23, 2009, at 12:43 PM, Brian West wrote:
>
>> Well first off you're not defining a pine here...
>>
>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference pin
>> number]
>>
>> That might be why its not asking for a pin.
>>
>> /b
>>
>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:
>>
>>>     <entry action="menu-exec-app" digits="1" param="conference
>>> 123456 at default+flags{}" />
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From rob4manhere at gmail.com  Fri Oct 30 08:34:28 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Fri, 30 Oct 2009 10:34:28 -0500
Subject: [Freeswitch-users] Setting up Conference with Moderator
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local>
	<3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local>
	<E97CD3D0-52FF-44D1-BB89-23816045CE21@gmail.com>
	<71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>,
	<114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local>
	<3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local>
	<F88E7B78-4CB8-45BD-98CE-81195862A891@gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local>
Message-ID: <D7D48F51-0917-4D9A-B8B4-DE8F552BB254@gmail.com>

Hm, strange.  I haven't seen that before.  Can you pastebin your logs  
at debug level?

On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote:

> It's strange... a tcpdump tells me that there is no DTMF from my  
> provider when using IVR, but when I call into a TN that goes  
> directly into the Conference App, I see DTMF from the provider.
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Rob Forman
> Sent: Friday, October 30, 2009 7:23 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
>
> I've never had any problem with that.  Is your logging at debug level
> so you can see the RECV DTFM in the log/fs_cli?  Are you calling from
> a SIP phone on the pbx, or via a PSTN provider?  Maybe your provider
> isn't passing them through.
>
> Make sure your logging is turned up then try something simpler, like
> calling the echo application, and see if DTFM comes through.
>
> Rob
>
> On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote:
>
>> Rob:
>>
>> For some reason, I don't see the DTMF appear on the fs_CLI when
>> using the below configuration....so it basically timesout.
>>
>> UK
>>
>>
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org
>> ] On Behalf Of Ujjval Karihaloo
>> Sent: Monday, October 26, 2009 9:21 AM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
>>
>> Thx a lot Rob, reading the wiki your way or using IVR seems correct..
>> ===============
>> The wiki also says that the wait-mod might be  "used in conjunction
>> with an IVR where the moderators are authenticated with an extra  
>> pass-
>> code", which is what I did.  I guess that's why I didn't understand
>> the point of the +pin.
>> ======================
>>
>> I will try it out.
>>
>> Again thx a lot for your help. Will keep everyone posted.
>>
>> ________________________________________
>> From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org
>> ] On Behalf Of Rob Forman [rob4manhere at gmail.com]
>> Sent: Friday, October 23, 2009 12:22 PM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
>>
>> I just re-tested with the pin in my dial plan:
>>
>>  <action application="conference" data="conference 123456 at default
>> +flags{}+1234" />
>>
>> And it doesn't challenge me for the pin.  I just drop right in.  I
>> figured this is how it was intended, since the wiki says the pin is
>> set initially and only challenged in later attempts [by future
>> callers]:
>>
>> "The first time a conference name (confname) is used, it will be
>> created on demand, and the pin will be set to what ever is specified
>> at that time: the pin in the data string if specified, or if not, the
>> "pin" setting in the conference profile, and if that is also
>> unspecified, then there is no pin protection. Any later attempt to
>> join the conference must specify the same pin number, if one existed
>> when it was created. "
>>
>>
>> The wiki also says that the wait-mod might be  "used in conjunction
>> with an IVR where the moderators are authenticated with an extra  
>> pass-
>> code", which is what I did.  I guess that's why I didn't understand
>> the point of the +pin.
>>
>> I'm sure there's a scenario where its used and useful, the wiki just
>> doesn't explain it.
>>
>> Rob
>>
>> On Oct 23, 2009, at 12:43 PM, Brian West wrote:
>>
>>> Well first off you're not defining a pine here...
>>>
>>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference  
>>> pin
>>> number]
>>>
>>> That might be why its not asking for a pin.
>>>
>>> /b
>>>
>>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:
>>>
>>>>    <entry action="menu-exec-app" digits="1" param="conference
>>>> 123456 at default+flags{}" />
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian.stafford at lattice-voice.com  Fri Oct 30 08:47:40 2009
From: brian.stafford at lattice-voice.com (Brian Stafford)
Date: Fri, 30 Oct 2009 15:47:40 +0000
Subject: [Freeswitch-users] mod_valet_parking: auto reports on wrong leg of
	call
Message-ID: <4AEB0A9C.7010907@lattice-voice.com>

Hi all

I did a 'make update' to 15289 and I found the auto feature added to 
mod_valet_parking so I figured I'd try it out.  Everything works fine 
except that the extension number for the parked call is reported on the 
wrong leg of the call.  It'd be good to sort this out as this is a 
useful feature for us.

Regards
Brian Stafford


From msc at freeswitch.org  Fri Oct 30 08:53:44 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 30 Oct 2009 08:53:44 -0700
Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Starting
	Shortly, Please Start 	Calling In
Message-ID: <87f2f3b90910300853s7ff6c38bt67f660e9d8581bdf@mail.gmail.com>

Hey everyone! The weekly meeting will start in a few minutes. Here's the
agenda:

http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_30

Please feel free to add other subjects. If you have something to share with
the community that would be great. Also, if you have questions for the dev
team by all means bring them too.

-Michael
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From brian at freeswitch.org  Fri Oct 30 09:00:29 2009
From: brian at freeswitch.org (Brian West)
Date: Fri, 30 Oct 2009 11:00:29 -0500
Subject: [Freeswitch-users] mod_valet_parking: auto reports on wrong leg
	of call
In-Reply-To: <4AEB0A9C.7010907@lattice-voice.com>
References: <4AEB0A9C.7010907@lattice-voice.com>
Message-ID: <BC287CE9-67CA-4E6A-A6D1-38ABBC2C51A7@freeswitch.org>

You have to be doing it wrong then.

Can you show us your dialplan you should have two extensions one for  
the lot range and one to attended transfer someone into the lot.

/b




On Oct 30, 2009, at 10:47 AM, Brian Stafford wrote:

> Hi all
>
> I did a 'make update' to 15289 and I found the auto feature added to
> mod_valet_parking so I figured I'd try it out.  Everything works fine
> except that the extension number for the parked call is reported on  
> the
> wrong leg of the call.  It'd be good to sort this out as this is a
> useful feature for us.
>
> Regards
> Brian Stafford



From djbinter at yahoo.com  Fri Oct 30 09:03:29 2009
From: djbinter at yahoo.com (DJB)
Date: Fri, 30 Oct 2009 09:03:29 -0700 (PDT)
Subject: [Freeswitch-users] Freeswitch in signaling path only
Message-ID: <429925.60305.qm@web37508.mail.mud.yahoo.com>

I am wondering why I cannot do as condition#2.

For Lua in dialplan, when I have the followings:


--WORKING--
(Condition#1)
.
.
session:execute("set","bypass_media=true")
session:execute("set","hangup_after_bridge=true")
session:execute("set","continue_on_fail=true")
.
.
 session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.1|sofia/external/" .. called_num .. "@1.1.1.2")
.
.

--NOT WORKING--
(Condition#2)
Note:  FS tries to be in media path and send re-invite.
.
.
session:execute("set","bypass_media=true")
session:execute("set","hangup_after_bridge=true")
session:execute("set","continue_on_fail=true")
.
.
 session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.1")
 session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.2")
.
.

Thank you,
Dorn B.


      


From djbinter at yahoo.com  Fri Oct 30 09:04:05 2009
From: djbinter at yahoo.com (DJB)
Date: Fri, 30 Oct 2009 09:04:05 -0700 (PDT)
Subject: [Freeswitch-users] CDR CSV variables
Message-ID: <609781.88322.qm@web37502.mail.mud.yahoo.com>

I wonder if I don't want to have b-leg in cdr csv, is there any variables that can give me the actual gateway ip address that is actually went out.

For instance, if I have this in my dialplan:
<action application="bridge" data="sofia/default/$1 at 192.168.1.4|sofia/default/$1 at 192.168.1.5"/>
the only value that I can think of from cdr csv is to get remote_ip_last_arg, but it would contains the whole line of both ip addresses.

Thank you,
Dorn B.



      


From brian.stafford at lattice-voice.com  Fri Oct 30 09:16:38 2009
From: brian.stafford at lattice-voice.com (Brian Stafford)
Date: Fri, 30 Oct 2009 16:16:38 +0000
Subject: [Freeswitch-users] mod_valet_parking: auto reports on wrong leg
 of call
In-Reply-To: <BC287CE9-67CA-4E6A-A6D1-38ABBC2C51A7@freeswitch.org>
References: <4AEB0A9C.7010907@lattice-voice.com>
	<BC287CE9-67CA-4E6A-A6D1-38ABBC2C51A7@freeswitch.org>
Message-ID: <4AEB1166.80002@lattice-voice.com>

Brian West wrote:
> You have to be doing it wrong then.
>
> Can you show us your dialplan you should have two extensions one for  
> the lot range and one to attended transfer someone into the lot.
>
> /b
>   
The relevant excerpt from the dialplan is

<extension name="valet_unpark">
<condition field="destination_number" expression="^(41[0-9])$">
<action application="answer"/>
<action application="valet_park" data="valet_lot $1"/>
</condition>
</extension>

<extension name="valet_park">
<condition field="destination_number" expression="^(420)$">
<action application="answer"/>
<action application="valet_park" data="valet_lot auto in 410 419"/>
</condition>
</extension>

x410-419 are the slots and 420 parks a call. Parking by picking one of 
410-419 works fine and subsequently dialling them from another works 
fine, I added x420 for the auto feature.

Regards
Brian


From ivan at myrvold.org  Fri Oct 30 09:16:37 2009
From: ivan at myrvold.org (Ivan C Myrvold)
Date: Fri, 30 Oct 2009 17:16:37 +0100
Subject: [Freeswitch-users] SIP provider with extern rtp server
In-Reply-To: <bf6923ed0910281921uad8c525i850c08361b21538d@mail.gmail.com>
References: <A2D2E918-B494-4179-96BA-CA3A7717846C@myrvold.org>
	<bf6923ed0910270658t61bb6687ne0b7d3ec5090e2ff@mail.gmail.com>
	<CB8008BD-4066-4B5B-9043-214CFE938CD8@myrvold.org>
	<bf6923ed0910271034w6c7e0cc5t685a5377764755e6@mail.gmail.com>
	<FA2C693B-FCE1-4BDD-BCE5-302AC699CEDB@myrvold.org>
	<87f2f3b90910281112g6e72d22elcfd653991ecd50cc@mail.gmail.com>
	<8AC09649-2585-4BE7-A959-A7AC41650789@myrvold.org>
	<bf6923ed0910281921uad8c525i850c08361b21538d@mail.gmail.com>
Message-ID: <544D39F2-40AB-41B4-BF18-89D7492B17EE@myrvold.org>

I have already set debug to 9, on both profiles.

Ivan


Den 29. okt. 2009 kl. 03:21 skrev Eliot Gable:

> See that 200 OK that keeps coming in over and over and over and over
> again? That's because they never received your ACK. If you can turn on
> sofia loglevel to 9 and then watch where you send the ACK, you will
> probably have your answer to why the other system did not receive it.
> If you're still not sure what's going on, post another pastebin with
> sofia loglevel set to 9.
>
>
> On Wed, Oct 28, 2009 at 4:51 PM, Ivan C Myrvold <ivan at myrvold.org>  
> wrote:
>> Oh, what happened to it?
>> Anyway, here is a new pb:
>> http://pastebin.freeswitch.org/10867
>> Ivan
>> Den 28. okt. 2009 kl. 19:12 skrev Michael Collins:
>>
>>
>> On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold <ivan at myrvold.org>  
>> wrote:
>>>
>>> Here is a debug log from a call from an internal phone out to an
>>> external (my iPhone with nbr 91316356):
>>> http://pastebin.freeswitch.org/108578
>>>
>>> Ivan
>>>
>> Uh... you wanna try that PB number again?
>> -MC
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>>
>
>
>
> -- 
> Eliot Gable
>
> "We do not inherit the Earth from our ancestors: we borrow it from our
> children." ~David Brower
>
> "I decided the words were too conservative for me. We're not borrowing
> from our children, we're stealing from them--and it's not even
> considered to be a crime." ~David Brower
>
> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
> live; not live to eat.) ~Marcus Tullius Cicero
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>



From chris at cloudtel.com  Fri Oct 30 10:00:01 2009
From: chris at cloudtel.com (Chris Burns)
Date: Fri, 30 Oct 2009 13:00:01 -0400
Subject: [Freeswitch-users] Freeswitch in signaling path only
In-Reply-To: <429925.60305.qm@web37508.mail.mud.yahoo.com>
References: <429925.60305.qm@web37508.mail.mud.yahoo.com>
Message-ID: <200910301300.01766.chris@cloudtel.com>

Do you have a debug level message from switch_ivr_originate.c in your log?
"Channel is already up, delaying proxy mode 'till both legs are answered."

Set bypass_media b4 each bridge. It is unsetting on you and setting 
bypass_media_after_bridge because you already answered the channel running 
the lua script.

On October 30, 2009 12:03:29 pm DJB wrote:
> I am wondering why I cannot do as condition#2.
>
> For Lua in dialplan, when I have the followings:
>
>
> --WORKING--
> (Condition#1)
> .
> .
> session:execute("set","bypass_media=true")
> session:execute("set","hangup_after_bridge=true")
> session:execute("set","continue_on_fail=true")
> .
> .
>  session:execute("bridge","sofia/external/" .. called_num ..
> "@1.1.1.1|sofia/external/" .. called_num .. "@1.1.1.2") .
> .
>
> --NOT WORKING--
> (Condition#2)
> Note:  FS tries to be in media path and send re-invite.
> .
> .
> session:execute("set","bypass_media=true")
> session:execute("set","hangup_after_bridge=true")
> session:execute("set","continue_on_fail=true")
> .
> .
>  session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.1")
>  session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.2")
> .
> .
>
> Thank you,
> Dorn B.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From egable+freeswitch at gmail.com  Fri Oct 30 10:30:54 2009
From: egable+freeswitch at gmail.com (Eliot Gable)
Date: Fri, 30 Oct 2009 13:30:54 -0400
Subject: [Freeswitch-users] SIP provider with extern rtp server
In-Reply-To: <544D39F2-40AB-41B4-BF18-89D7492B17EE@myrvold.org>
References: <A2D2E918-B494-4179-96BA-CA3A7717846C@myrvold.org>
	<bf6923ed0910270658t61bb6687ne0b7d3ec5090e2ff@mail.gmail.com>
	<CB8008BD-4066-4B5B-9043-214CFE938CD8@myrvold.org>
	<bf6923ed0910271034w6c7e0cc5t685a5377764755e6@mail.gmail.com>
	<FA2C693B-FCE1-4BDD-BCE5-302AC699CEDB@myrvold.org>
	<87f2f3b90910281112g6e72d22elcfd653991ecd50cc@mail.gmail.com>
	<8AC09649-2585-4BE7-A959-A7AC41650789@myrvold.org>
	<bf6923ed0910281921uad8c525i850c08361b21538d@mail.gmail.com>
	<544D39F2-40AB-41B4-BF18-89D7492B17EE@myrvold.org>
Message-ID: <bf6923ed0910301030x375fd6b9m75328c0e3a58b70f@mail.gmail.com>

fsctl loglevel debug
console loglevel debug
sofia profile internal siptrace on
sofia profile external siptrace on
sofia loglevel all 9
^^^^^^^^^^^^^^^^^^^^^

Then run your call, then do this:

sofia loglevel all 0
sofia profile external siptrace off
sofia profile internal siptrace off
fsctl loglevel warning
console loglevel warning

On Fri, Oct 30, 2009 at 12:16 PM, Ivan C Myrvold <ivan at myrvold.org> wrote:
> I have already set debug to 9, on both profiles.
>
> Ivan
>
>
> Den 29. okt. 2009 kl. 03:21 skrev Eliot Gable:
>
>> See that 200 OK that keeps coming in over and over and over and over
>> again? That's because they never received your ACK. If you can turn on
>> sofia loglevel to 9 and then watch where you send the ACK, you will
>> probably have your answer to why the other system did not receive it.
>> If you're still not sure what's going on, post another pastebin with
>> sofia loglevel set to 9.
>>
>>
>> On Wed, Oct 28, 2009 at 4:51 PM, Ivan C Myrvold <ivan at myrvold.org>
>> wrote:
>>> Oh, what happened to it?
>>> Anyway, here is a new pb:
>>> http://pastebin.freeswitch.org/10867
>>> Ivan
>>> Den 28. okt. 2009 kl. 19:12 skrev Michael Collins:
>>>
>>>
>>> On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold <ivan at myrvold.org>
>>> wrote:
>>>>
>>>> Here is a debug log from a call from an internal phone out to an
>>>> external (my iPhone with nbr 91316356):
>>>> http://pastebin.freeswitch.org/108578
>>>>
>>>> Ivan
>>>>
>>> Uh... you wanna try that PB number again?
>>> -MC
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>> --
>> Eliot Gable
>>
>> "We do not inherit the Earth from our ancestors: we borrow it from our
>> children." ~David Brower
>>
>> "I decided the words were too conservative for me. We're not borrowing
>> from our children, we're stealing from them--and it's not even
>> considered to be a crime." ~David Brower
>>
>> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
>> live; not live to eat.) ~Marcus Tullius Cicero
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero


From r.wilczynski at gmail.com  Fri Oct 30 11:28:06 2009
From: r.wilczynski at gmail.com (=?ISO-8859-2?Q?Robert_Wilczy=F1ski?=)
Date: Fri, 30 Oct 2009 19:28:06 +0100
Subject: [Freeswitch-users] Gateway registration fails: IP used for
	realm in place of configured realm value
Message-ID: <56bfa1be0910301128n6169ee34h540f959042b9ad26@mail.gmail.com>

Hi Brian,

Thanks for the explanation. This seems to have solved the issue.

Robert.

> ---------- Forwarded message ----------
> From:?Brian West <brian at freeswitch.org>
> To:?freeswitch-users at lists.freeswitch.org
> Date:?Thu, 29 Oct 2009 13:31:26 -0500
> Subject:?Re: [Freeswitch-users] Gateway registration fails: IP used for realm in place of configured realm value
> It's because you need to set the from-user and from-domain because FreeSWITCH don't care about the realm because WE provide you the realm in our challenge... you only provide realm for broken providers such as BroadVoice.
>
> remove realm from your setting. ?Add from-user and from-domain
>
> /b
>
> On Oct 29, 2009, at 4:46 AM, Robert Wilczy?ski wrote:
>
>> Hi,
>>
>> We have two FS instances production (prod) and gateway (gw)
>>
>> when prod starts it tries to register a gateway:
>>
>> <include>
>> ? ?<gateway name="fs_gateway">
>> ? ? ? ?<param name="realm" value="24ccdomain"/>
>> ? ? ? ?<param name="proxy" value="$${local_ip_v4}:15060" />
>> ? ? ? ?<param name="username" value="24cc"/>
>> ? ? ? ?<param name="password" value="24ccpass"/>
>> ? ? ? ?<param name="expire-seconds" value="60"/>
>> ? ? ? ?<param name="register" value="true"/>
>> ? ?</gateway>
>> </include>
>>
>> and fails:
>>
>> 2009-10-29 10:19:45.449220 [NOTICE] sofia_glue.c:3534 Reload XML [Success]
>> 2009-10-29 10:19:45.450221 [INFO] mod_enum.c:808 ENUM Reloaded
>> 2009-10-29 10:19:45.450221 [NOTICE] sofia_reg.c:2117 Added gateway
>> 'fs_gateway' to profile 'external'
>> 2009-10-29 10:19:45.450221 [NOTICE] sofia.c:3093 Adding Alias
>> [outbound] for profile [external]
>> 2009-10-29 10:19:45.450221 [NOTICE] sofia.c:3093 Adding Alias [nat]
>> for profile [external]
>> 2009-10-29 10:19:45.451221 [NOTICE] sofia.c:3108 Started Profile
>> external [sofia_reg_external]
>> 2009-10-29 10:19:45.452221 [INFO] switch_time.c:661 Timezone reloaded
>> 530 definitions
>> 2009-10-29 10:19:45.628743 [NOTICE] sofia_reg.c:333 Registering fs_gateway
>> 2009-10-29 10:19:45.996790 [NOTICE] sofia.c:1202 Waiting for worker thread
>> 2009-10-29 10:19:45.998790 [NOTICE] sofia.c:1729 Adding Alias
>> [192.168.1.20] for profile [internal]
>> 2009-10-29 10:19:45.998790 [NOTICE] sofia.c:3093 Adding Alias
>> [default] for profile [internal]
>> 2009-10-29 10:19:45.998790 [NOTICE] sofia.c:3108 Started Profile
>> internal [sofia_reg_internal]
>> 2009-10-29 10:19:53.354724 [ERR] sofia_reg.c:1425 fs_gateway
>> Registration Failed with status Forbidden [403]. failure #1
>> 2009-10-29 10:19:53.440735 [WARNING] sofia_reg.c:364 fs_gateway Failed
>> Registration, setting retry to 60 seconds.
>> 2009-10-29 10:20:53.108812 [NOTICE] sofia_reg.c:333 Registering fs_gateway
>> 2009-10-29 10:20:53.116813 [ERR] sofia_reg.c:1425 fs_gateway
>> Registration Failed with status Forbidden [403]. failure #2
>> 2009-10-29 10:20:53.208825 [WARNING] sofia_reg.c:364 fs_gateway Failed
>> Registration, setting retry to 90 seconds.
>>
>> gw is configured like that:
>>
>> <include>
>> ? ?<domain name="24ccdomain">
>> ? ? ? ?<params>
>> ? ? ? ? ? ?<param name="dial-string"
>> value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>> ? ? ? ?</params>
>>
>> ? ? ? ?<groups>
>> ? ? ? ? ? ?<group name="default">
>> ? ? ? ? ? ? ? ?<users>
>> ? ? ? ? ? ? ? ? ? ?<user id="24cc">
>> ? ? ? ? ? ? ? ? ? ? ? ?<params>
>> ? ? ? ? ? ? ? ? ? ? ? ? ? ?<param name="password" value="24ccpass"/>
>> ? ? ? ? ? ? ? ? ? ? ? ?</params>
>> ? ? ? ? ? ? ? ? ? ? ? ?<variables>
>> ? ? ? ? ? ? ? ? ? ? ? ? ? ?<variable name="user_context" value="default"/>
>> ? ? ? ? ? ? ? ? ? ? ? ? ? ?<variable name="effective_caller_id_name"
>> value="24 CallCenter"/>
>> ? ? ? ? ? ? ? ? ? ? ? ?</variables>
>> ? ? ? ? ? ? ? ? ? ?</user>
>> ? ? ? ? ? ? ? ?</users>
>> ? ? ? ? ? ?</group>
>> ? ? ? ?</groups>
>> ? ?</domain>
>> </include>
>>
>> and outputs:
>>
>> 2009-10-29 10:11:29.165700 [WARNING] sofia_reg.c:1737 Can't find user
>> [24cc at 192.168.1.20]
>> You must define a domain called '192.168.1.20' in your directory and
>> add a user with the id="24cc" attribute
>> and you must configure your device to use the proper domain in it's
>> authentication credentials.
>>
>> on each registration attempt from prod.
>>
>> This worked perfectly fine with one of previous revisions but started
>> failing on 15257. Using siptrace we were able to narrow down the issue
>> to prod sending an IP address as realm instead of gateway configured
>> realm (as it did in previous revision):
>>
>> Authorization: Digest username="24cc", realm="192.168.1.20",
>> nonce="4d82d624-c884-6340-8f7f-a4575f2583bd",
>> cnonce="Es23bj8MEi29T7UQ80i7sw", algorithm=MD5,
>> uri="sip:192.168.1.20:15060",
>> response="afea161876aad137657bf1f923d18659", qop=auth, nc=00000004
>>
>> Is there any configuration setting we are missing that would bring
>> back previous behavior?
>>
>> --
>> Best regards,
>> Robert Wilczynski
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org


From djbinter at yahoo.com  Fri Oct 30 11:31:12 2009
From: djbinter at yahoo.com (DJB)
Date: Fri, 30 Oct 2009 11:31:12 -0700 (PDT)
Subject: [Freeswitch-users] Freeswitch in signaling path only
In-Reply-To: <200910301300.01766.chris@cloudtel.com>
References: <429925.60305.qm@web37508.mail.mud.yahoo.com>
	<200910301300.01766.chris@cloudtel.com>
Message-ID: <312533.76921.qm@web37505.mail.mud.yahoo.com>

Now i have as follows, but it's still the same result.  By the way, I am running:  FreeSWITCH Version 1.0.4 (exported)
.
.
.
session:execute("set","hangup_after_bridge=true")
session:execute("set","continue_on_fail=true")
session:execute("set","originate_timeout=2")
session:execute("set","originate_retries=3")
session:execute("set","progress_timeout=15")
.
.
.
while row do
 local gw_ip_address = row.gw_ip_address
 local cust_name = row.cust_name
 session:execute("set", "accountcode=" ..cust_name .. "")
 session:execute("set","bypass_media=true")
 session:execute("bridge","sofia/external/" .. called_num .. "@XX.XX.XX.XX.146")
 session:execute("set","bypass_media=true")
 session:execute("bridge","sofia/external/" .. called_num .. "@XX.XX.XX.XX.105")
 -- Block for testing -- session:execute("bridge","sofia/external/" .. called_num .. "@" .. gw_ip_address .."")
 row = cur:fetch (row, "a")
end

Here is the debug for switch_ivr_originate.c:

2009-10-30 11:09:52.877832 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C
S_CONSUME_MEDIA
2009-10-30 11:09:53.17811 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX.1
46]
2009-10-30 11:09:54.285453 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.105) State Change CS_ROUTING -> C
S_CONSUME_MEDIA
2009-10-30 11:09:54.422426 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX.
105]
2009-10-30 11:09:55.694761 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C
S_CONSUME_MEDIA
2009-10-30 11:09:55.836036 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX.
146]
2009-10-30 11:09:57.107697 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.105) State Change CS_ROUTING -> C
S_CONSUME_MEDIA
2009-10-30 11:09:57.254664 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX.
105]
2009-10-30 11:12:03.129097 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C
S_CONSUME_MEDIA
2009-10-30 11:12:03.273055 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX.
146]
2009-10-30 11:12:04.546410 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.105) State Change CS_ROUTING -> C
S_CONSUME_MEDIA
2009-10-30 11:12:04.682661 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX.
105]
2009-10-30 11:12:15.781701 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 16 [NORMAL_CLEARING]
2009-10-30 11:12:33.349162 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C
S_CONSUME_MEDIA
2009-10-30 11:12:33.470989 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX.
146]
2009-10-30 11:12:34.724641 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL]
2009-10-30 11:12:34.730634 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL]
2009-10-30 11:12:34.750637 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL]


FIRST ROUTE: XX.XX.XX.146 
and I tried to failed the first route and it gave 500 back, then it goes to the next one.
SECOND ROUTE:  XX.XX.XX.105


Thank you,
Dorn B.



----- Original Message ----
From: Chris Burns <chris at cloudtel.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Fri, October 30, 2009 10:00:01 AM
Subject: Re: [Freeswitch-users] Freeswitch in signaling path only

Do you have a debug level message from switch_ivr_originate.c in your log?
"Channel is already up, delaying proxy mode 'till both legs are answered."

Set bypass_media b4 each bridge. It is unsetting on you and setting 
bypass_media_after_bridge because you already answered the channel running 
the lua script.

On October 30, 2009 12:03:29 pm DJB wrote:
> I am wondering why I cannot do as condition#2.
>
> For Lua in dialplan, when I have the followings:
>
>
> --WORKING--
> (Condition#1)
> .
> .
> session:execute("set","bypass_media=true")
> session:execute("set","hangup_after_bridge=true")
> session:execute("set","continue_on_fail=true")
> .
> .
>  session:execute("bridge","sofia/external/" .. called_num ..
> "@1.1.1.1|sofia/external/" .. called_num .. "@1.1.1.2") .
> .
>
> --NOT WORKING--
> (Condition#2)
> Note:  FS tries to be in media path and send re-invite.
> .
> .
> session:execute("set","bypass_media=true")
> session:execute("set","hangup_after_bridge=true")
> session:execute("set","continue_on_fail=true")
> .
> .
>  session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.1")
>  session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.2")
> .
> .
>
> Thank you,
> Dorn B.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



      


From pedroprado at msn.com  Fri Oct 30 11:53:00 2009
From: pedroprado at msn.com (Pedro Prado)
Date: Fri, 30 Oct 2009 16:53:00 -0200
Subject: [Freeswitch-users] Control Panel
Message-ID: <SNT102-W5248A31FD39933F270318BC1B60@phx.gbl>


Hi,

Which control panel are you using with FreeSwitch? WikiPbx, FreePbx or other?

Thanks,
Pedro Prado
 		 	   		  
_________________________________________________________________
Voc? sabia que com o Hotmail voc? tem espa?o ilimitado para guardar seus e-mails? Come?e a usar j?!
http://www.microsoft.com/brasil/windows/windowslive/products/hotmail.aspx
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From tleyden at branchcut.com  Fri Oct 30 12:16:34 2009
From: tleyden at branchcut.com (Traun Leyden)
Date: Fri, 30 Oct 2009 23:46:34 +0430
Subject: [Freeswitch-users] Control Panel
In-Reply-To: <SNT102-W5248A31FD39933F270318BC1B60@phx.gbl>
References: <SNT102-W5248A31FD39933F270318BC1B60@phx.gbl>
Message-ID: <f9f80d540910301216u669f19c2tb3845b93d261247c@mail.gmail.com>

>From watching the mailing list, I've seen:

95% are using straight XML combined with their home-cooked solutions based
on mod_xml_curl, which makes it pretty easy to write your own custom GUI for
freeswitch.

Out of the rest, the most are probably using pfsense, freepbx, and a small
handful of people using wikipbx (based on forum activity, guess would be
50ish).  There's also TCAPI (php).

Domain-based multi-tenancy is on the horizon for wikipbx, which has been
chugging along slowly, but has not died.  BroadTone (broadtone.net) has
kindly donated some test servers and a web developer, so kudos goes out to
them.

2009/10/30 Pedro Prado <pedroprado at msn.com>

>  Hi,
>
> Which control panel are you using with FreeSwitch? WikiPbx, FreePbx or
> other?
>
> Thanks,
> Pedro Prado
>
> ------------------------------
> Navegue com seguran?a com o Novo Internet Explorer 8. Baixe agora, ?
> gratis!<http://brasil.microsoft.com.br/IE8/mergulhe/?utm_source=MSN%3BHotmail&utm_medium=Tagline&utm_campaign=IE8>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Fri Oct 30 12:19:39 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 30 Oct 2009 12:19:39 -0700
Subject: [Freeswitch-users] Control Panel
In-Reply-To: <SNT102-W5248A31FD39933F270318BC1B60@phx.gbl>
References: <SNT102-W5248A31FD39933F270318BC1B60@phx.gbl>
Message-ID: <87f2f3b90910301219j45531d15y70482aaa4fb923a2@mail.gmail.com>

2009/10/30 Pedro Prado <pedroprado at msn.com>

>  Hi,
>
> Which control panel are you using with FreeSwitch? WikiPbx, FreePbx or
> other?
>
These are all at various stages of production/usability:
Most of the people here just use emacs/vim/etc. and then use the built-in
webserver for simple stuff. (see http://wiki.freeswitch.org/wiki/Webapi)

For the others:
WikiPBX is an enigma. I haven't heard of anything new in quite some time. I
don't recommend it.
FreePBX V3 - very promising, in early development, don't expect anything
usable right away
FusionPBX - Farther along, simpler than FreePBX, closest to being usable in
production
FSGUI - Donated to contrib by jmesquita; simple, work in progress,
interesting to play with

CudaTel - It's the bomb! ;)

-MC
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From msc at freeswitch.org  Fri Oct 30 12:32:45 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 30 Oct 2009 12:32:45 -0700
Subject: [Freeswitch-users] PBX vs. SIP proxy?
In-Reply-To: <26128938.post@talk.nabble.com>
References: <25920788.post@talk.nabble.com> <4AD914FE.4020709@gmail.com>
	<26128938.post@talk.nabble.com>
Message-ID: <87f2f3b90910301232v16191998occa33e0485598da1@mail.gmail.com>

On Fri, Oct 30, 2009 at 5:06 AM, Fred-145 <codecomplete at free.fr> wrote:

>
> Thanks Meftah for the help. So it looks like an SIP proxy is just to handle
> the registration process, while a PBX offers services like ringing phones,
> setting up calls, handling voice mail, etc.
>

Of course, as usual, the lines are getting blurred. Look at opensips.org and
see all the crazy things that a good SIP proxy can actually do.
-MC
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From codecomplete at free.fr  Fri Oct 30 13:22:45 2009
From: codecomplete at free.fr (Fred-145)
Date: Fri, 30 Oct 2009 13:22:45 -0700 (PDT)
Subject: [Freeswitch-users] PBX vs. SIP proxy?
In-Reply-To: <87f2f3b90910301232v16191998occa33e0485598da1@mail.gmail.com>
References: <25920788.post@talk.nabble.com> <4AD914FE.4020709@gmail.com>
	<26128938.post@talk.nabble.com>
	<87f2f3b90910301232v16191998occa33e0485598da1@mail.gmail.com>
Message-ID: <26136363.post@talk.nabble.com>


Thanks for the link. But then, since OpenSIPs doesn't run on Windows, I'll
stick to FreeSwitch :)
-- 
View this message in context: http://old.nabble.com/PBX-vs.-SIP-proxy--tp25920788p26136363.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From ivan at myrvold.org  Fri Oct 30 13:26:23 2009
From: ivan at myrvold.org (Ivan C Myrvold)
Date: Fri, 30 Oct 2009 21:26:23 +0100
Subject: [Freeswitch-users] SIP provider with extern rtp server
In-Reply-To: <bf6923ed0910301030x375fd6b9m75328c0e3a58b70f@mail.gmail.com>
References: <A2D2E918-B494-4179-96BA-CA3A7717846C@myrvold.org>
	<bf6923ed0910270658t61bb6687ne0b7d3ec5090e2ff@mail.gmail.com>
	<CB8008BD-4066-4B5B-9043-214CFE938CD8@myrvold.org>
	<bf6923ed0910271034w6c7e0cc5t685a5377764755e6@mail.gmail.com>
	<FA2C693B-FCE1-4BDD-BCE5-302AC699CEDB@myrvold.org>
	<87f2f3b90910281112g6e72d22elcfd653991ecd50cc@mail.gmail.com>
	<8AC09649-2585-4BE7-A959-A7AC41650789@myrvold.org>
	<bf6923ed0910281921uad8c525i850c08361b21538d@mail.gmail.com>
	<544D39F2-40AB-41B4-BF18-89D7492B17EE@myrvold.org>
	<bf6923ed0910301030x375fd6b9m75328c0e3a58b70f@mail.gmail.com>
Message-ID: <8BB98561-BAA3-46B4-939F-FBA5EF79BD06@myrvold.org>

Yes, now I got a more detailed trace. Thank you for helping me with  
this.

A new pastebin at http://pastebin.freeswitch.org/10905

Ivan

Den 30. okt. 2009 kl. 18:30 skrev Eliot Gable:

> fsctl loglevel debug
> console loglevel debug
> sofia profile internal siptrace on
> sofia profile external siptrace on
> sofia loglevel all 9
> ^^^^^^^^^^^^^^^^^^^^^
>
> Then run your call, then do this:
>
> sofia loglevel all 0
> sofia profile external siptrace off
> sofia profile internal siptrace off
> fsctl loglevel warning
> console loglevel warning
>
> On Fri, Oct 30, 2009 at 12:16 PM, Ivan C Myrvold <ivan at myrvold.org>  
> wrote:
>> I have already set debug to 9, on both profiles.
>>
>> Ivan
>>
>>
>> Den 29. okt. 2009 kl. 03:21 skrev Eliot Gable:
>>
>>> See that 200 OK that keeps coming in over and over and over and over
>>> again? That's because they never received your ACK. If you can  
>>> turn on
>>> sofia loglevel to 9 and then watch where you send the ACK, you will
>>> probably have your answer to why the other system did not receive  
>>> it.
>>> If you're still not sure what's going on, post another pastebin with
>>> sofia loglevel set to 9.
>>>
>>>
>>> On Wed, Oct 28, 2009 at 4:51 PM, Ivan C Myrvold <ivan at myrvold.org>
>>> wrote:
>>>> Oh, what happened to it?
>>>> Anyway, here is a new pb:
>>>> http://pastebin.freeswitch.org/10867
>>>> Ivan
>>>> Den 28. okt. 2009 kl. 19:12 skrev Michael Collins:
>>>>
>>>>
>>>> On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold <ivan at myrvold.org>
>>>> wrote:
>>>>>
>>>>> Here is a debug log from a call from an internal phone out to an
>>>>> external (my iPhone with nbr 91316356):
>>>>> http://pastebin.freeswitch.org/108578
>>>>>
>>>>> Ivan
>>>>>
>>>> Uh... you wanna try that PB number again?
>>>> -MC
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>> users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>> users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>>
>>> --
>>> Eliot Gable
>>>
>>> "We do not inherit the Earth from our ancestors: we borrow it from  
>>> our
>>> children." ~David Brower
>>>
>>> "I decided the words were too conservative for me. We're not  
>>> borrowing
>>> from our children, we're stealing from them--and it's not even
>>> considered to be a crime." ~David Brower
>>>
>>> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
>>> live; not live to eat.) ~Marcus Tullius Cicero
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>
>
>
> -- 
> Eliot Gable
>
> "We do not inherit the Earth from our ancestors: we borrow it from our
> children." ~David Brower
>
> "I decided the words were too conservative for me. We're not borrowing
> from our children, we're stealing from them--and it's not even
> considered to be a crime." ~David Brower
>
> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
> live; not live to eat.) ~Marcus Tullius Cicero
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>



From diego.viola at gmail.com  Fri Oct 30 13:28:48 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 30 Oct 2009 20:28:48 +0000
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <86a32abc0910162312le6c08b8jb53ae83bcafadab1@mail.gmail.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
	<6b65470d0910131009y1cf136c3q8c0932375c19f272@mail.gmail.com>
	<86a32abc0910161510ma323718g410bcf1029156077@mail.gmail.com>
	<a87d43fd0910161729u42db5ea7qafb5604c0680954@mail.gmail.com>
	<86a32abc0910161811h15eac05dwc3fe0c2533b7397a@mail.gmail.com>
	<86a32abc0910161847k415b9722ubc5439ea1b9f6e78@mail.gmail.com>
	<86a32abc0910161854k7fd3fca6pc0686e928c772d2c@mail.gmail.com>
	<86a32abc0910162050m6bad9a9do171b268e083b3d0d@mail.gmail.com>
	<87f2f3b90910162254q23893e4cx160fe4a5ae64cc91@mail.gmail.com>
	<86a32abc0910162312le6c08b8jb53ae83bcafadab1@mail.gmail.com>
Message-ID: <86a32abc0910301328h642726et2bae30997f8f37ae@mail.gmail.com>

This needs to be converted to documentation form also.

http://wiki.freeswitch.org/wiki/Proxy_Media#How_to_detect_when_it_should_be_used

On Sat, Oct 17, 2009 at 6:12 AM, Diego Viola <diego.viola at gmail.com> wrote:

> Thanks, our wiki is getting a lot better :).
>
> Diego
>
> On Sat, Oct 17, 2009 at 5:54 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Diego, when I get some time I will help convert these from IRC to wiki.
>> Thanks,
>> MC
>>
>>
>> On Fri, Oct 16, 2009 at 8:50 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>
>>> This needs to be converted to documentation form too.
>>>
>>> http://wiki.freeswitch.org/wiki/Mod_vmd#Related_conversations
>>>
>>> Diego
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From msc at freeswitch.org  Fri Oct 30 14:05:03 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 30 Oct 2009 14:05:03 -0700
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <86a32abc0910301328h642726et2bae30997f8f37ae@mail.gmail.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
	<86a32abc0910161510ma323718g410bcf1029156077@mail.gmail.com>
	<a87d43fd0910161729u42db5ea7qafb5604c0680954@mail.gmail.com>
	<86a32abc0910161811h15eac05dwc3fe0c2533b7397a@mail.gmail.com>
	<86a32abc0910161847k415b9722ubc5439ea1b9f6e78@mail.gmail.com>
	<86a32abc0910161854k7fd3fca6pc0686e928c772d2c@mail.gmail.com>
	<86a32abc0910162050m6bad9a9do171b268e083b3d0d@mail.gmail.com>
	<87f2f3b90910162254q23893e4cx160fe4a5ae64cc91@mail.gmail.com>
	<86a32abc0910162312le6c08b8jb53ae83bcafadab1@mail.gmail.com>
	<86a32abc0910301328h642726et2bae30997f8f37ae@mail.gmail.com>
Message-ID: <87f2f3b90910301405j4fdb3f48we864e386fdc878ad@mail.gmail.com>

thanks for the heads up! I'll ask someone in the know to fill that out.
-MC

On Fri, Oct 30, 2009 at 1:28 PM, Diego Viola <diego.viola at gmail.com> wrote:

> This needs to be converted to documentation form also.
>
>
> http://wiki.freeswitch.org/wiki/Proxy_Media#How_to_detect_when_it_should_be_used
>
>
> On Sat, Oct 17, 2009 at 6:12 AM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> Thanks, our wiki is getting a lot better :).
>>
>> Diego
>>
>> On Sat, Oct 17, 2009 at 5:54 AM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Diego, when I get some time I will help convert these from IRC to wiki.
>>> Thanks,
>>> MC
>>>
>>>
>>> On Fri, Oct 16, 2009 at 8:50 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>
>>>> This needs to be converted to documentation form too.
>>>>
>>>> http://wiki.freeswitch.org/wiki/Mod_vmd#Related_conversations
>>>>
>>>> Diego
>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From diego.viola at gmail.com  Fri Oct 30 14:36:27 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 30 Oct 2009 21:36:27 +0000
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <87f2f3b90910301405j4fdb3f48we864e386fdc878ad@mail.gmail.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
	<a87d43fd0910161729u42db5ea7qafb5604c0680954@mail.gmail.com>
	<86a32abc0910161811h15eac05dwc3fe0c2533b7397a@mail.gmail.com>
	<86a32abc0910161847k415b9722ubc5439ea1b9f6e78@mail.gmail.com>
	<86a32abc0910161854k7fd3fca6pc0686e928c772d2c@mail.gmail.com>
	<86a32abc0910162050m6bad9a9do171b268e083b3d0d@mail.gmail.com>
	<87f2f3b90910162254q23893e4cx160fe4a5ae64cc91@mail.gmail.com>
	<86a32abc0910162312le6c08b8jb53ae83bcafadab1@mail.gmail.com>
	<86a32abc0910301328h642726et2bae30997f8f37ae@mail.gmail.com>
	<87f2f3b90910301405j4fdb3f48we864e386fdc878ad@mail.gmail.com>
Message-ID: <86a32abc0910301436q25cbbdc8h182c586819ccad34@mail.gmail.com>

No problem, when I have some time I need to merge these two pages also.

http://wiki.freeswitch.org/wiki/Multi-tenant
http://wiki.freeswitch.org/wiki/Multiple_Companies

=D

Diego


On Fri, Oct 30, 2009 at 9:05 PM, Michael Collins <msc at freeswitch.org> wrote:

> thanks for the heads up! I'll ask someone in the know to fill that out.
> -MC
>
>
> On Fri, Oct 30, 2009 at 1:28 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> This needs to be converted to documentation form also.
>>
>>
>> http://wiki.freeswitch.org/wiki/Proxy_Media#How_to_detect_when_it_should_be_used
>>
>>
>> On Sat, Oct 17, 2009 at 6:12 AM, Diego Viola <diego.viola at gmail.com>wrote:
>>
>>> Thanks, our wiki is getting a lot better :).
>>>
>>> Diego
>>>
>>> On Sat, Oct 17, 2009 at 5:54 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> Diego, when I get some time I will help convert these from IRC to wiki.
>>>> Thanks,
>>>> MC
>>>>
>>>>
>>>> On Fri, Oct 16, 2009 at 8:50 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>>
>>>>> This needs to be converted to documentation form too.
>>>>>
>>>>> http://wiki.freeswitch.org/wiki/Mod_vmd#Related_conversations
>>>>>
>>>>> Diego
>>>>>
>>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Fri Oct 30 14:52:51 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 30 Oct 2009 22:52:51 +0100
Subject: [Freeswitch-users] mod_t38gateway
Message-ID: <65d96fc80910301452x2a831733h92b05861d6c94123@mail.gmail.com>

does anybody know how does it work and how to use it in a dialplan?


freeswitch at nemesis>
freeswitch at nemesis>
freeswitch at nemesis> load mod_t38gateway
API CALL [load(mod_t38gateway)] output:
+OK

2009-10-30 22:44:38.204268 [NOTICE] mod_t38gateway.c:147 T.38 gateway
enabled
2009-10-30 22:44:38.204268 [CONSOLE] switch_loadable_module.c:889
Successfully Loaded [mod_t38gateway]
2009-10-30 22:44:38.204268 [NOTICE] switch_loadable_module.c:248 Adding
Application 't38gateway'
2009-10-30 22:44:38.205374 [NOTICE] switch_loadable_module.c:270 Adding API
Function 't38gateway'
freeswitch at nemesis>
freeswitch at nemesis>
freeswitch at nemesis>
freeswitch at nemesis> t38gateway
API CALL [t38gateway()] output:
-USAGE: <uuid> <command>

freeswitch at nemesis>

T.
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From hjqlopez at hotmail.com  Fri Oct 30 15:07:04 2009
From: hjqlopez at hotmail.com (Humberto Quintana)
Date: Fri, 30 Oct 2009 18:07:04 -0400
Subject: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media
Message-ID: <COL104-W179212DC517C7BD2DB189D0B60@phx.gbl>


Hi everybody,

I'd like some help with this situation that is 'haunting' me :-) 

My scenario is as follows:

inbound-bypass-media is set in the profile because we dont want FS handling the media.

1. A calls B
2. FS sends to B the A's SDP
3. B answers
4. FS sends to A the B's SDP
5. Media going directly between A and B
6. B REFERs the call to C (blind transfer with no reINVITE for Hold)
7. FS accepts(202) the REFER and sends the NOTIFY
7a. B and FS send the BYE
8. FS sends an INVITE  to C with A's SDP
9. C answers
10. FS doesn't send a reINVITE to A to let it know about C's SDP


Is that the expected FS behavior or is this a bug?


I'd appreciate any hints. Thanks,


Humberto


 		 	   		  
_________________________________________________________________
Lots of fantastic Windows 7 offers, in one convenient place. Get the perfect deal for you now.
http://go.microsoft.com/?linkid=9691633
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From diego.viola at gmail.com  Fri Oct 30 15:09:34 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 30 Oct 2009 22:09:34 +0000
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <86a32abc0910301436q25cbbdc8h182c586819ccad34@mail.gmail.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
	<86a32abc0910161811h15eac05dwc3fe0c2533b7397a@mail.gmail.com>
	<86a32abc0910161847k415b9722ubc5439ea1b9f6e78@mail.gmail.com>
	<86a32abc0910161854k7fd3fca6pc0686e928c772d2c@mail.gmail.com>
	<86a32abc0910162050m6bad9a9do171b268e083b3d0d@mail.gmail.com>
	<87f2f3b90910162254q23893e4cx160fe4a5ae64cc91@mail.gmail.com>
	<86a32abc0910162312le6c08b8jb53ae83bcafadab1@mail.gmail.com>
	<86a32abc0910301328h642726et2bae30997f8f37ae@mail.gmail.com>
	<87f2f3b90910301405j4fdb3f48we864e386fdc878ad@mail.gmail.com>
	<86a32abc0910301436q25cbbdc8h182c586819ccad34@mail.gmail.com>
Message-ID: <86a32abc0910301509i10ac4f21rc90d4f6130e0759e@mail.gmail.com>

Another one to convert:

http://wiki.freeswitch.org/wiki/User:Agx#CAMP_ON

On Fri, Oct 30, 2009 at 9:36 PM, Diego Viola <diego.viola at gmail.com> wrote:

> No problem, when I have some time I need to merge these two pages also.
>
> http://wiki.freeswitch.org/wiki/Multi-tenant
> http://wiki.freeswitch.org/wiki/Multiple_Companies
>
> =D
>
> Diego
>
>
>
> On Fri, Oct 30, 2009 at 9:05 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> thanks for the heads up! I'll ask someone in the know to fill that out.
>> -MC
>>
>>
>> On Fri, Oct 30, 2009 at 1:28 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>
>>> This needs to be converted to documentation form also.
>>>
>>>
>>> http://wiki.freeswitch.org/wiki/Proxy_Media#How_to_detect_when_it_should_be_used
>>>
>>>
>>> On Sat, Oct 17, 2009 at 6:12 AM, Diego Viola <diego.viola at gmail.com>wrote:
>>>
>>>> Thanks, our wiki is getting a lot better :).
>>>>
>>>> Diego
>>>>
>>>> On Sat, Oct 17, 2009 at 5:54 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>> Diego, when I get some time I will help convert these from IRC to wiki.
>>>>> Thanks,
>>>>> MC
>>>>>
>>>>>
>>>>> On Fri, Oct 16, 2009 at 8:50 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>>>
>>>>>> This needs to be converted to documentation form too.
>>>>>>
>>>>>> http://wiki.freeswitch.org/wiki/Mod_vmd#Related_conversations
>>>>>>
>>>>>> Diego
>>>>>>
>>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From msc at freeswitch.org  Fri Oct 30 15:19:32 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 30 Oct 2009 15:19:32 -0700
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <86a32abc0910301509i10ac4f21rc90d4f6130e0759e@mail.gmail.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
	<86a32abc0910161847k415b9722ubc5439ea1b9f6e78@mail.gmail.com>
	<86a32abc0910161854k7fd3fca6pc0686e928c772d2c@mail.gmail.com>
	<86a32abc0910162050m6bad9a9do171b268e083b3d0d@mail.gmail.com>
	<87f2f3b90910162254q23893e4cx160fe4a5ae64cc91@mail.gmail.com>
	<86a32abc0910162312le6c08b8jb53ae83bcafadab1@mail.gmail.com>
	<86a32abc0910301328h642726et2bae30997f8f37ae@mail.gmail.com>
	<87f2f3b90910301405j4fdb3f48we864e386fdc878ad@mail.gmail.com>
	<86a32abc0910301436q25cbbdc8h182c586819ccad34@mail.gmail.com>
	<86a32abc0910301509i10ac4f21rc90d4f6130e0759e@mail.gmail.com>
Message-ID: <87f2f3b90910301519x54ac6c6fr13d546b19e493090@mail.gmail.com>

actually that's a user's personal page so you don't have to worry about that
one. :)
-MC

On Fri, Oct 30, 2009 at 3:09 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Another one to convert:
>
> http://wiki.freeswitch.org/wiki/User:Agx#CAMP_ON
>
>
> On Fri, Oct 30, 2009 at 9:36 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> No problem, when I have some time I need to merge these two pages also.
>>
>> http://wiki.freeswitch.org/wiki/Multi-tenant
>> http://wiki.freeswitch.org/wiki/Multiple_Companies
>>
>> =D
>>
>> Diego
>>
>>
>>
>> On Fri, Oct 30, 2009 at 9:05 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> thanks for the heads up! I'll ask someone in the know to fill that out.
>>> -MC
>>>
>>>
>>> On Fri, Oct 30, 2009 at 1:28 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>
>>>> This needs to be converted to documentation form also.
>>>>
>>>>
>>>> http://wiki.freeswitch.org/wiki/Proxy_Media#How_to_detect_when_it_should_be_used
>>>>
>>>>
>>>> On Sat, Oct 17, 2009 at 6:12 AM, Diego Viola <diego.viola at gmail.com>wrote:
>>>>
>>>>> Thanks, our wiki is getting a lot better :).
>>>>>
>>>>> Diego
>>>>>
>>>>> On Sat, Oct 17, 2009 at 5:54 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>
>>>>>> Diego, when I get some time I will help convert these from IRC to
>>>>>> wiki.
>>>>>> Thanks,
>>>>>> MC
>>>>>>
>>>>>>
>>>>>> On Fri, Oct 16, 2009 at 8:50 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>>>>
>>>>>>> This needs to be converted to documentation form too.
>>>>>>>
>>>>>>> http://wiki.freeswitch.org/wiki/Mod_vmd#Related_conversations
>>>>>>>
>>>>>>> Diego
>>>>>>>
>>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Fri Oct 30 15:25:50 2009
From: brian at freeswitch.org (Brian West)
Date: Fri, 30 Oct 2009 17:25:50 -0500
Subject: [Freeswitch-users] no REINVITE on Blind Transfer with
	bypass_media
In-Reply-To: <COL104-W179212DC517C7BD2DB189D0B60@phx.gbl>
References: <COL104-W179212DC517C7BD2DB189D0B60@phx.gbl>
Message-ID: <41A5BD86-AC6C-4E0F-BD93-229DD9AF9622@freeswitch.org>

First you forgot to mention what SVN rev you're on...

/b

On Oct 30, 2009, at 5:07 PM, Humberto Quintana wrote:

> Hi everybody,
>
> I'd like some help with this situation that is 'haunting' me :-)
>
> My scenario is as follows:
>
> inbound-bypass-media is set in the profile because we dont want FS  
> handling the media.
>
> 1. A calls B
> 2. FS sends to B the A's SDP
> 3. B answers
> 4. FS sends to A the B's SDP
> 5. Media going directly between A and B
> 6. B REFERs the call to C (blind transfer with no reINVITE for Hold)
> 7. FS accepts(202) the REFER and sends the NOTIFY
> 7a. B and FS send the BYE
> 8. FS sends an INVITE  to C with A's SDP
> 9. C answers
> 10. FS doesn't send a reINVITE to A to let it know about C's SDP
>
>
> Is that the expected FS behavior or is this a bug?
>
>
> I'd appreciate any hints. Thanks,
>
>
> Humberto
>
>
>
> Lots of fantastic offers on Windows 7, in one convenient place. Get  
> a deal on Windows 7 now  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Fri Oct 30 15:58:40 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 30 Oct 2009 17:58:40 -0500
Subject: [Freeswitch-users] no REINVITE on Blind Transfer with
	bypass_media
In-Reply-To: <COL104-W179212DC517C7BD2DB189D0B60@phx.gbl>
References: <COL104-W179212DC517C7BD2DB189D0B60@phx.gbl>
Message-ID: <191c3a030910301558t505ce395u9ccbd36ce3df3033@mail.gmail.com>

profile options

<param name="media-option" value="resume-media-on-hold"/>
<param name="media-option" value="bypass-media-after-att-xfer"/>


On Fri, Oct 30, 2009 at 5:07 PM, Humberto Quintana <hjqlopez at hotmail.com>wrote:

>  Hi everybody,
>
> I'd like some help with this situation that is 'haunting' me :-)
>
> My scenario is as follows:
>
> inbound-bypass-media is set in the profile because we dont want FS handling
> the media.
>
> 1. A calls B
> 2. FS sends to B the A's SDP
> 3. B answers
> 4. FS sends to A the B's SDP
> 5. Media going directly between A and B
> 6. B REFERs the call to C (blind transfer with no reINVITE for Hold)
> 7. FS accepts(202) the REFER and sends the NOTIFY
> 7a. B and FS send the BYE
> 8. FS sends an INVITE  to C with A's SDP
> 9. C answers
> 10. FS doesn't send a reINVITE to A to let it know about C's SDP
>
>
> Is that the expected FS behavior or is this a bug?
>
>
> I'd appreciate any hints. Thanks,
>
>
> Humberto
>
>
>
> ------------------------------
> Lots of fantastic offers on Windows 7, in one convenient place. Get a deal
> on Windows 7 now <http://go.microsoft.com/?linkid=9691628>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From dome at tel.co.th  Sat Oct 31 04:08:24 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Sat, 31 Oct 2009 18:08:24 +0700
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
Message-ID: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>

Dear All,
          I'm disable all CDR and use nibble_bill (modify) and
odbc_query for create CDR and dedcut balance. but i found many zombie
channel and can't kill
Like a
51555b02-c5ba-11de-af77-774e15238dbe,outbound,2009-10-30
18:11:38,1256951498,sofia/external/66848087977 at xxx.xxx.xx.191:7050,CS_REPORTING,FreeSWITCH,66850304266,xxx.xxx.xxx.191,7050,,,XML,public,,,,,

I'm using FreeSWITCH Version 1.0.5pre3 (hacked)

How to fix ?

Dome C.


From info at daccii.it  Sat Oct 31 05:40:31 2009
From: info at daccii.it (Albano Daniele Salvatore - Lavoro)
Date: Sat, 31 Oct 2009 13:40:31 +0100
Subject: [Freeswitch-users] Freeswitch seems to doesn't reknow dial tone
 after the first call using OpenZAP (analog spans)
In-Reply-To: <87f2f3b90910291026u2c62ded6v5b9458a9c2e85877@mail.gmail.com>
References: <4AE8B20D.4040608@daccii.it>	<191c3a030910290912x9489784s4cc8e224363f3796@mail.gmail.com>
	<87f2f3b90910291026u2c62ded6v5b9458a9c2e85877@mail.gmail.com>
Message-ID: <4AEC303F.4020207@daccii.it>

Hi,

i've done more tests, with svn too, but nothing to do. The strange thing 
is when freeswitch shutdown and close zaptel channel i get back the dial 
tone from my provider (i'm checking trought ztmonitor 1 -vv). Can be 
something is missing when span get closed?

PS1: with svn i can do only one call, but i can receive calls without 
problems
PS2: i joined yesterday on irc but i had few time ... hope to join monday


Best Regards,
Daniele

Michael Collins ha scritto:
> 
> 
> On Thu, Oct 29, 2009 at 9:12 AM, Anthony Minessale 
> <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>> wrote:
> 
>     You may want to work with FreeSWITCH svn trunk for the time being.
>     There have been some changes and if you have a problem to fix it's
>     easier to debug with that version.
> 
> 
>     2009/10/28 Albano Daniele Salvatore - Lavoro <info at daccii.it
>     <mailto:info at daccii.it>>
> 
>         Hi to all,
> 
>         this is my first message here!
> 
> Welcome!
> 
> Like Tony said, the latest SVN trunk is probably what you want. We are 
> already doing pre-releases of 1.0.5. SVN trunk is extremely stable. 
> Also, you can get realtime help via IRC at irc.freenode.net 
> <http://irc.freenode.net>, channels #freeswitch and #openzap.
> 
> -MC
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From mike at jerris.com  Sat Oct 31 08:25:23 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 31 Oct 2009 11:25:23 -0400
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <5e414ed0910240513q316905ai5cf8c2ef63b52f60@mail.gmail.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
	<191c3a030910131101p418e0961t373db68285248ff6@mail.gmail.com>
	<538F4CF43B294D0EACABA281E1913B4C@chaos>
	<65d96fc80910131224m73f35b87g355be818ad3c0caf@mail.gmail.com>
	<48B637E0506F4A809E4C0CAF39BD08BB@chaos>
	<65d96fc80910132348t202905fbub57cc4c814eb4e21@mail.gmail.com>
	<5e414ed0910140731w1c7ebedr150e69cda8073155@mail.gmail.com>
	<191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com>
	<5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com>
	<191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com>
	<5e414ed0910240513q316905ai5cf8c2ef63b52f60@mail.gmail.com>
Message-ID: <ACB65510-C38F-4345-B555-BD2DCF61B444@jerris.com>

This may be possible with tcp, how could this work on udp?  Can you  
provide an rfc reference on this?

Mike

On Oct 24, 2009, at 8:13 AM, Dennis wrote:

> ok, as written, i come back after some tests with fs and a thomson  
> cirpack.
>
> it did not work - at least in our tests.
>
> we are using socket outbound and when a call comes in, it starts the
> socket of fs. the number may be 123456. fs sends the respond 484 and
> our carrier receives this information. but fs ends the call with
> hangup_cause = invalid_number_format.
> the carrier has one more digit for the phone number and sends 1234567
> and the above mentioned behavior repeats.
>
> the behavior we want and expected is, that the call stays in the
> socket after response 484, so that the carrier can send the 1234567
> into the same socket.
> the management, when fs should send response 484 and when fs should be
> answered would be programmed by us.
> it also important, that fs keeps the call in the socket, so we can
> tell fs, to answer the call after x seconds anyway.
>
> any ideas, what we could do?



From freeswitch-list at puzzled.xs4all.nl  Sat Oct 31 08:48:53 2009
From: freeswitch-list at puzzled.xs4all.nl (Patrick List)
Date: Sat, 31 Oct 2009 16:48:53 +0100
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <ACB65510-C38F-4345-B555-BD2DCF61B444@jerris.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>	<191c3a030910131101p418e0961t373db68285248ff6@mail.gmail.com>	<538F4CF43B294D0EACABA281E1913B4C@chaos>	<65d96fc80910131224m73f35b87g355be818ad3c0caf@mail.gmail.com>	<48B637E0506F4A809E4C0CAF39BD08BB@chaos>	<65d96fc80910132348t202905fbub57cc4c814eb4e21@mail.gmail.com>	<5e414ed0910140731w1c7ebedr150e69cda8073155@mail.gmail.com>	<191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com>	<5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com>	<191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com>	<5e414ed0910240513q316905ai5cf8c2ef63b52f60@mail.gmail.com>
	<ACB65510-C38F-4345-B555-BD2DCF61B444@jerris.com>
Message-ID: <4AEC5C65.6050800@puzzled.xs4all.nl>

> On Oct 24, 2009, at 8:13 AM, Dennis wrote:
>
>> ok, as written, i come back after some tests with fs and a thomson
>> cirpack.

No idea if this is useful as I'm a noob with fs. If not please excuse 
the noise. In the past Asterisk to work properly with Cirpack needed the 
following patch:

diff -uNr asterisk-1.4.19.org/main/rtp.c asterisk-1.4.19/main/rtp.c
--- asterisk-1.4.19.org/main/rtp.c	2007-10-08 22:06:33.000000000 +0200
+++ asterisk-1.4.19/main/rtp.c	2007-11-11 13:12:28.000000000 +0100
@@ -1383,6 +1383,7 @@
  	[34] = {1, AST_FORMAT_H263},
  	[103] = {1, AST_FORMAT_H263_PLUS},
  	[97] = {1, AST_FORMAT_ILBC},
+	[96] = {0, AST_RTP_DTMF},
  	[99] = {1, AST_FORMAT_H264},
  	[101] = {0, AST_RTP_DTMF},
  	[110] = {1, AST_FORMAT_SPEEX},

Maybe this helps.

Regards,
Patrick


From brian at freeswitch.org  Sat Oct 31 08:58:37 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 31 Oct 2009 10:58:37 -0500
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <4AEC5C65.6050800@puzzled.xs4all.nl>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>	<191c3a030910131101p418e0961t373db68285248ff6@mail.gmail.com>	<538F4CF43B294D0EACABA281E1913B4C@chaos>	<65d96fc80910131224m73f35b87g355be818ad3c0caf@mail.gmail.com>	<48B637E0506F4A809E4C0CAF39BD08BB@chaos>	<65d96fc80910132348t202905fbub57cc4c814eb4e21@mail.gmail.com>	<5e414ed0910140731w1c7ebedr150e69cda8073155@mail.gmail.com>	<191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com>	<5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com>	<191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com>	<5e414ed0910240513q316905ai5cf8c2ef63b52f60@mail.gmail.com>
	<ACB65510-C38F-4345-B555-BD2DCF61B444@jerris.com>
	<4AEC5C65.6050800@puzzled.xs4all.nl>
Message-ID: <188D171E-C1E9-439B-BCCB-EE5E80BD21B7@freeswitch.org>

We have <param name="rfc2833-pt" value="101"/> on the profile and you  
to set this.

So you can set it to 96 if needed.  But you shouldn't have to do that  
if they say 96 and we say 101 they should be listening on 101 from us  
and we should be listening on 96 from them... thats why its called an  
RTP map.

/b

On Oct 31, 2009, at 10:48 AM, Patrick List wrote:

>> On Oct 24, 2009, at 8:13 AM, Dennis wrote:
>>
>>> ok, as written, i come back after some tests with fs and a thomson
>>> cirpack.
>
> No idea if this is useful as I'm a noob with fs. If not please excuse
> the noise. In the past Asterisk to work properly with Cirpack needed  
> the
> following patch:
>
> diff -uNr asterisk-1.4.19.org/main/rtp.c asterisk-1.4.19/main/rtp.c
> --- asterisk-1.4.19.org/main/rtp.c	2007-10-08 22:06:33.000000000 +0200
> +++ asterisk-1.4.19/main/rtp.c	2007-11-11 13:12:28.000000000 +0100
> @@ -1383,6 +1383,7 @@
>  	[34] = {1, AST_FORMAT_H263},
>  	[103] = {1, AST_FORMAT_H263_PLUS},
>  	[97] = {1, AST_FORMAT_ILBC},
> +	[96] = {0, AST_RTP_DTMF},
>  	[99] = {1, AST_FORMAT_H264},
>  	[101] = {0, AST_RTP_DTMF},
>  	[110] = {1, AST_FORMAT_SPEEX},
>
> Maybe this helps.
>
> Regards,
> Patrick
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From rupa at rupa.com  Sat Oct 31 08:59:18 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 31 Oct 2009 10:59:18 -0500
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
Message-ID: <b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>

I believe this means you are hung up in CDR reporting.  You can get a
core of the running system and look at the stack traces to see where
exactly it is hung up.

Try using:

support-d/fscore_pb gcore yourname

This will put a copy of the stacktrace on the pastebin for poeple (or
you) to look at.

On Sat, Oct 31, 2009 at 6:08 AM, Dome Charoenyost <dome at tel.co.th> wrote:
> Dear All,
> ? ? ? ? ?I'm disable all CDR and use nibble_bill (modify) and
> odbc_query for create CDR and dedcut balance. but i found many zombie
> channel and can't kill
> Like a
> 51555b02-c5ba-11de-af77-774e15238dbe,outbound,2009-10-30
> 18:11:38,1256951498,sofia/external/66848087977 at xxx.xxx.xx.191:7050,CS_REPORTING,FreeSWITCH,66850304266,xxx.xxx.xxx.191,7050,,,XML,public,,,,,
>
> I'm using FreeSWITCH Version 1.0.5pre3 (hacked)
>
> How to fix ?
>
> Dome C.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa


From brian at freeswitch.org  Sat Oct 31 09:11:49 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 31 Oct 2009 11:11:49 -0500
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
Message-ID: <140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>

So you're doing your billing INLINE with the session using the OSBC  
Query app?  Not the most optimal way!

/b

On Oct 31, 2009, at 10:59 AM, Rupa Schomaker wrote:

> I believe this means you are hung up in CDR reporting.  You can get a
> core of the running system and look at the stack traces to see where
> exactly it is hung up.
>
> Try using:
>
> support-d/fscore_pb gcore yourname
>
> This will put a copy of the stacktrace on the pastebin for poeple (or
> you) to look at.
>
> On Sat, Oct 31, 2009 at 6:08 AM, Dome Charoenyost <dome at tel.co.th>  
> wrote:
>> Dear All,
>>          I'm disable all CDR and use nibble_bill (modify) and
>> odbc_query for create CDR and dedcut balance. but i found many zombie
>> channel and can't kill
>> Like a
>> 51555b02-c5ba-11de-af77-774e15238dbe,outbound,2009-10-30
>> 18:11:38,1256951498,sofia/external/66848087977 at xxx.xxx.xx. 
>> 191:7050,CS_REPORTING,FreeSWITCH,66850304266,xxx.xxx.xxx. 
>> 191,7050,,,XML,public,,,,,
>>
>> I'm using FreeSWITCH Version 1.0.5pre3 (hacked)
>>
>> How to fix ?
>>
>> Dome C.

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From dome at tel.co.th  Sat Oct 31 09:32:35 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Sat, 31 Oct 2009 23:32:35 +0700
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
	<140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
Message-ID: <8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>

I use odbc_query for retrive balance and get LCR from my billing DB.
and use nibble_bill

Dome C.


2009/10/31 Brian West <brian at freeswitch.org>:
> So you're doing your billing INLINE with the session using the OSBC Query
> app? ?Not the most optimal way!
> /b
> On Oct 31, 2009, at 10:59 AM, Rupa Schomaker wrote:
>
> I believe this means you are hung up in CDR reporting. ?You can get a
> core of the running system and look at the stack traces to see where
> exactly it is hung up.
>
> Try using:
>
> support-d/fscore_pb gcore yourname
>
> This will put a copy of the stacktrace on the pastebin for poeple (or
> you) to look at.
>
> On Sat, Oct 31, 2009 at 6:08 AM, Dome Charoenyost <dome at tel.co.th> wrote:
>
> Dear All,
>
> ? ? ? ? ?I'm disable all CDR and use nibble_bill (modify) and
>
> odbc_query for create CDR and dedcut balance. but i found many zombie
>
> channel and can't kill
>
> Like a
>
> 51555b02-c5ba-11de-af77-774e15238dbe,outbound,2009-10-30
>
> 18:11:38,1256951498,sofia/external/66848087977 at xxx.xxx.xx.191:7050,CS_REPORTING,FreeSWITCH,66850304266,xxx.xxx.xxx.191,7050,,,XML,public,,,,,
>
> I'm using FreeSWITCH Version 1.0.5pre3 (hacked)
>
> How to fix ?
>
> Dome C.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From brian at freeswitch.org  Sat Oct 31 09:44:39 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 31 Oct 2009 11:44:39 -0500
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
	<140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
Message-ID: <AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>

You should never do billing inline with the session thread is all I'm  
saying.

/b

On Oct 31, 2009, at 11:32 AM, Dome Charoenyost wrote:

> I use odbc_query for retrive balance and get LCR from my billing DB.
> and use nibble_bill
>
> Dome C.



From jmesquita at freeswitch.org  Sat Oct 31 09:56:17 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 31 Oct 2009 14:56:17 -0200
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
	<140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
	<AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
Message-ID: <ebbc2d550910310956x3ce32c53n44efed28a54ef36c@mail.gmail.com>

Dome, just to explain what Brian is saying:

Doing billing inline in this case means that the session thread (what you
see in show channels). If, for whatever reason, something goes wrong on the
DB connection or task for the billing, this session thread will be stuck
with it leaving it dangling around the system. This is what most likely is
happening to you. The right way to do it is to let the session thread go
(which mod_cdr_csv does) and then process the billing.

None of the methods you have described do that. You might consider
post-processing your CDR/billing information to avoid coming up with this
kind of problem that are very hard to predict.

Hope that helps,

JM

PS: Nonetheless, I still think it is valid to get a core dump like Rupa
metioned.

On Sat, Oct 31, 2009 at 2:44 PM, Brian West <brian at freeswitch.org> wrote:

> You should never do billing inline with the session thread is all I'm
> saying.
>
> /b
>
> On Oct 31, 2009, at 11:32 AM, Dome Charoenyost wrote:
>
> > I use odbc_query for retrive balance and get LCR from my billing DB.
> > and use nibble_bill
> >
> > Dome C.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dome at tel.co.th  Sat Oct 31 10:05:27 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Sun, 1 Nov 2009 00:05:27 +0700
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
	<140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
	<AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
Message-ID: <8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com>

2009/10/31 Brian West <brian at freeswitch.org>:
> You should never do billing inline with the session thread is all I'm
> saying.
Sorry I don't understand.  This is my example dialplan
       <action application="set"
data="dialoption=nibble_account=${caller_id_number},nibble_rate=0.001,ignore_early_media=false,originate_timeout=20/>
        <action application="odbc_query" data="select carrier_name As
provider_id,balance::text As lcr_balance,(balance/rate)::int as
lcr_timeout,(balance/rate*60)::int as lcr_timeout_sec  from
lcr('${caller_id_number}','$1','external');"/>
        <action application="log" data="ALERT Balance ${lcr_balance}"/>
        <action application="log" data="ALERT timeout ${lcr_timeout}"/>
        <action application="execute_extension"
data="ck_zerobalance_${lcr_balance} XML public"/>
        <action application="execute_extension"
data="tz_say_min_${lcr_timeout} XML public"/>
        <action application="sched_hangup" data="+${lcr_timeout_sec}
alloted_timeout"/>
        <action application="log" data="ALERT timeout_sec ${lcr_timeout_sec}"/>
        <action application="odbc_query" data="select
'route_'||route_id::text As name,gw_prefix||prefix||gw_suffix As value
from lcr('${caller_id_number}','$1','external') ;"/>
        <action application="bridge" data="{${dialoption}},${route_1}"/>
        <action application="bridge" data="{${dialoption}},${route_2}"/>
        <action application="bridge" data="{${dialoption}},${route_3}"/>


Dome C.

>
> /b
>
> On Oct 31, 2009, at 11:32 AM, Dome Charoenyost wrote:
>
>> I use odbc_query for retrive balance and get LCR from my billing DB.
>> and use nibble_bill
>>
>> Dome C.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From dome at tel.co.th  Sat Oct 31 10:08:26 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Sun, 1 Nov 2009 00:08:26 +0700
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <ebbc2d550910310956x3ce32c53n44efed28a54ef36c@mail.gmail.com>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
	<140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
	<AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
	<ebbc2d550910310956x3ce32c53n44efed28a54ef36c@mail.gmail.com>
Message-ID: <8ccbff060910311008k51b89440i8c210a09ba82b9ab@mail.gmail.com>

Thanks JM.
   So mod_nibblebill not good for running with high traffic. so we
can't do realtime billing ?

Dome C.


2009/10/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
> Dome, just to explain what Brian is saying:
>
> Doing billing inline in this case means that the session thread (what you
> see in show channels). If, for whatever reason, something goes wrong on the
> DB connection or task for the billing, this session thread will be stuck
> with it leaving it dangling around the system. This is what most likely is
> happening to you. The right way to do it is to let the session thread go
> (which mod_cdr_csv does) and then process the billing.
>
> None of the methods you have described do that. You might consider
> post-processing your CDR/billing information to avoid coming up with this
> kind of problem that are very hard to predict.
>
> Hope that helps,
>
> JM
>
> PS: Nonetheless, I still think it is valid to get a core dump like Rupa
> metioned.
>
> On Sat, Oct 31, 2009 at 2:44 PM, Brian West <brian at freeswitch.org> wrote:
>>
>> You should never do billing inline with the session thread is all I'm
>> saying.
>>
>> /b
>>
>> On Oct 31, 2009, at 11:32 AM, Dome Charoenyost wrote:
>>
>> > I use odbc_query for retrive balance and get LCR from my billing DB.
>> > and use nibble_bill
>> >
>> > Dome C.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From brian at freeswitch.org  Sat Oct 31 10:11:08 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 31 Oct 2009 12:11:08 -0500
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
	<140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
	<AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
	<8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com>
Message-ID: <EEDE7B6D-F722-4557-BBFE-AC0031C0BA86@freeswitch.org>

I think once you get the backtrace like rupa said we can see that  
maybe odbc_query is really hanging or something similar.

/b

On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:

> 2009/10/31 Brian West <brian at freeswitch.org>:
>> You should never do billing inline with the session thread is all I'm
>> saying.

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From jmesquita at freeswitch.org  Sat Oct 31 10:18:13 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 31 Oct 2009 15:18:13 -0200
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <EEDE7B6D-F722-4557-BBFE-AC0031C0BA86@freeswitch.org>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
	<140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
	<AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
	<8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com>
	<EEDE7B6D-F722-4557-BBFE-AC0031C0BA86@freeswitch.org>
Message-ID: <ebbc2d550910311018m7bc9b86bi120649c36dbf3413@mail.gmail.com>

No, mod_nibblebill definetely needs to be enhanced but it is not the problem
and it can be used with high load traffic.

The one I am not sure about is odbc_query since it was not developed for
that.

Do what Rupa said, please.

Regards,

JM

On Sat, Oct 31, 2009 at 3:11 PM, Brian West <brian at freeswitch.org> wrote:

> I think once you get the backtrace like rupa said we can see that maybe
> odbc_query is really hanging or something similar.
>
> /b
>
> On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:
>
> 2009/10/31 Brian West <brian at freeswitch.org>:
>
> You should never do billing inline with the session thread is all I'm
>
> saying.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dome at tel.co.th  Sat Oct 31 10:21:26 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Sun, 1 Nov 2009 00:21:26 +0700
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <EEDE7B6D-F722-4557-BBFE-AC0031C0BA86@freeswitch.org>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
	<140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
	<AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
	<8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com>
	<EEDE7B6D-F722-4557-BBFE-AC0031C0BA86@freeswitch.org>
Message-ID: <8ccbff060910311021k45ceeemf2a56c2dd65d2a6c@mail.gmail.com>

i try
support-d/fscore_pb gcore root
but only get

[New Thread 0xb7656b90 (LWP 28445)]
[New Thread 0xb77bbb90 (LWP 28444)]
0xb7a5e007 in select () from /lib/libc.so.6
Saved corefile core.28443

i got core file but no more file

Dome C.

2009/11/1 Brian West <brian at freeswitch.org>:
> I think once you get the backtrace like rupa said we can see that maybe
> odbc_query is really hanging or something similar.
> /b
> On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:
>
> 2009/10/31 Brian West <brian at freeswitch.org>:
>
> You should never do billing inline with the session thread is all I'm
>
> saying.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From dome at tel.co.th  Sat Oct 31 10:34:07 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Sun, 1 Nov 2009 00:34:07 +0700
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <ebbc2d550910311018m7bc9b86bi120649c36dbf3413@mail.gmail.com>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
	<140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
	<AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
	<8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com>
	<EEDE7B6D-F722-4557-BBFE-AC0031C0BA86@freeswitch.org>
	<ebbc2d550910311018m7bc9b86bi120649c36dbf3413@mail.gmail.com>
Message-ID: <8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com>

I found bug in fscore_pb   $pwd should be $mypwd

Now i post already. please help me check.

Dome C.


2009/11/1 Jo?o Mesquita <jmesquita at freeswitch.org>:
> No, mod_nibblebill definetely needs to be enhanced but it is not the problem
> and it can be used with high load traffic.
>
> The one I am not sure about is odbc_query since it was not developed for
> that.
>
> Do what Rupa said, please.
>
> Regards,
>
> JM
>
> On Sat, Oct 31, 2009 at 3:11 PM, Brian West <brian at freeswitch.org> wrote:
>>
>> I think once you get the backtrace like rupa said we can see that maybe
>> odbc_query is really hanging or something similar.
>> /b
>> On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:
>>
>> 2009/10/31 Brian West <brian at freeswitch.org>:
>>
>> You should never do billing inline with the session thread is all I'm
>>
>> saying.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From mike at jerris.com  Sat Oct 31 10:42:52 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 31 Oct 2009 13:42:52 -0400
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179
In-Reply-To: <87f2f3b90910221054l8abc3far9a18a66f0959f950@mail.gmail.com>
References: <mailman.1516.1256123625.2495.freeswitch-users@lists.freeswitch.org>
	<a60fed5e0910212108v4755203y693b8bdec5501f5f@mail.gmail.com>
	<87f2f3b90910212141w72b8f5a2oce032adb974d71e2@mail.gmail.com>
	<b39273530910220551q14d22150ldf8d4bff0b6fd011@mail.gmail.com>
	<87f2f3b90910221054l8abc3far9a18a66f0959f950@mail.gmail.com>
Message-ID: <509453DC-57BF-43D0-B704-FF0F2BB58EC0@jerris.com>

You actually can use these in conditions.  Just need to be careful  
that the var you are conditioning on is already set.

Mike

On Oct 22, 2009, at 1:54 PM, Michael Collins wrote:

>
>
> On Thu, Oct 22, 2009 at 5:51 AM, Rupa Schomaker <rupa at rupa.com> wrote:
> cond would be helpful here?  I updated the wiki on this one just now
> with a bit more detail.  It is a api call. so, you'd use it like:
>
> ${cond(eval ? trueval : falseval)}
>
> so to get a value of ERR if the var my myvar is > 15 you could:
>
> ${cond(${myvar} > 15 ? ERR : OK)}
>
> If both sides of the comparison operator are numeric then it does
> numeric comparison otherwise it does lexical string comparison.
>
> Rupa,
>
> Yes, you can do the set/cond API trick but you can only do it in the  
> action or anti-action tags, not in the condition tags. I'm sure you  
> know that but I want all those reading this thread to make the  
> connection.
> -MC
>

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From stevendt at primrosebank.net  Sat Oct 31 10:37:19 2009
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Sat, 31 Oct 2009 17:37:19 -0000
Subject: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question
Message-ID: <C4ADBF041AB44AE399847AF51ED740A1@bp1.ad.bp.com>

Hi,

I am new to FreeSwitch  - and VOIP too (as the following questions will indicate). 

I am trying to setup a VOIP system at home and am just getting started, I have been trawling through the Docs/Wiki and realise that I have a lot to learn !

Anyway, here's the first (of what I'm sure will be many !) questions, hopefully someone will take the type to help me out......

OK, I have already worked out that FreeSwitch seems to be much better supported under Linux than Windows (is that a fair comment ?), but while I have a long term intention to give Linux a go, it's something that is on the back burner (and has been for some time). At the moment, the current plan is to use FreeSwitch under WindowsXP and I have loaded the latest Windows Install. Everything seems to be working fine so far - well, in test mode anyway, my VOIP phones and teleco interface won't be here until next week, so at the moment, I'm just trying out the system with X-Lite.

The first question that I have is about WAPSERVER. This was installed as part of the FreeSwitch install. At the moment, I am manually starting FreeSwitch and WAPSERVER does not seem to start automatically. When I manually start it, it says that it is OFFLINE until I manually put it ONLINE. So :-

1. What does WAPSERVER actually do ?
2. How should it be started ?
3. My next step is to configure the Server PC to auto-logon to my Windows 2003 Server domain and put FreeSwitch in the Startup Group - do I need to add WAPSERVER there too ?

The next question is about the ServerIP address. Not thinking far enough ahead, the machine that I installed FreeSwitch on uses DHCP to get its IP Address. I guess that the FreeSwitch server should really have a static IP Address. If I change the PC to have a static IP Address, then 

1. Do I need to re-install FreeSwitch
2. If not, is there a or is there command to get FreeSwitch to updated itself with a new IP Address ?
3. Maybe the Server does not need to know the change - maybe if only affect the clients (Phones/SoftPhones ?)

Thanks & Best Regards
Dave
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From sicfslist at gmail.com  Sat Oct 31 11:13:13 2009
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Sat, 31 Oct 2009 13:13:13 -0500
Subject: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question
In-Reply-To: <C4ADBF041AB44AE399847AF51ED740A1@bp1.ad.bp.com>
References: <C4ADBF041AB44AE399847AF51ED740A1@bp1.ad.bp.com>
Message-ID: <4AEC7E39.5030206@gmail.com>

Dave,

You should check out the settings in $FS_DIR/conf/sofia_profiles/ and 
$FS_DIR/conf/vars.xml.  To make the changes take effect just restart FS 
or use the API commands for reloading the sofia profiles.

You'll definitely want to set a static IP address on the FS server (or 
make sure the DHCP server always assigns the same IP address to the FS 
server).  Otherwise you'll constantly be changing the clients that have 
to connect to FS.

SDR


Dave Stevenson wrote:
> Hi,
>  
> I am new to FreeSwitch  - and VOIP too (as the following questions 
> will indicate).
>  
> I am trying to setup a VOIP system at home and am just getting 
> started, I have been trawling through the Docs/Wiki and realise that I 
> have a lot to learn !
>  
> Anyway, here's the first (of what I'm sure will be many !) questions, 
> hopefully someone will take the type to help me out......
>  
> OK, I have already worked out that FreeSwitch seems to be much better 
> supported under Linux than Windows (is that a fair comment ?), but 
> while I have a long term intention to give Linux a go, it's something 
> that is on the back burner (and has been for some time). At the 
> moment, the current plan is to use FreeSwitch under WindowsXP and I 
> have loaded the latest Windows Install. Everything seems to be working 
> fine so far - well, in test mode anyway, my VOIP phones and teleco 
> interface won't be here until next week, so at the moment, I'm just 
> trying out the system with X-Lite.
>  
> The first question that I have is about WAPSERVER. This was installed 
> as part of the FreeSwitch install. At the moment, I am manually 
> starting FreeSwitch and WAPSERVER does not seem to start 
> automatically. When I manually start it, it says that it is OFFLINE 
> until I manually put it ONLINE. So :-
>  
> 1. What does WAPSERVER actually do ?
> 2. How should it be started ?
> 3. My next step is to configure the Server PC to auto-logon to my 
> Windows 2003 Server domain and put FreeSwitch in the Startup Group - 
> do I need to add WAPSERVER there too ?
>  
> The next question is about the ServerIP address. Not thinking far 
> enough ahead, the machine that I installed FreeSwitch on uses DHCP to 
> get its IP Address. I guess that the FreeSwitch server should really 
> have a static IP Address. If I change the PC to have a static IP 
> Address, then
>  
> 1. Do I need to re-install FreeSwitch
> 2. If not, is there a or is there command to get FreeSwitch to updated 
> itself with a new IP Address ?
> 3. Maybe the Server does not need to know the change - maybe if only 
> affect the clients (Phones/SoftPhones ?)
>  
> Thanks & Best Regards
> Dave
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From rupa at rupa.com  Sat Oct 31 11:17:52 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 31 Oct 2009 13:17:52 -0500
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
	<140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
	<AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
	<8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com>
	<EEDE7B6D-F722-4557-BBFE-AC0031C0BA86@freeswitch.org>
	<ebbc2d550910311018m7bc9b86bi120649c36dbf3413@mail.gmail.com>
	<8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com>
Message-ID: <b39273530910311117k3b4bcfa5s98a5b720b1775e20@mail.gmail.com>

fscore_pb has been updated.

Next time put in the pastebin url.  This one was 10911.

Ok, a suspicious thread looks below.  If I read it right, you are
doing an originate from within the hangup handler.  That doesn't look
normal or sane to me.

Thread 41 (process 30732):
#0  0xb7a5e007 in select () from /lib/libc.so.6
#1  0xb7dde39f in apr_sleep (t=1000) at time/unix/time.c:246
#2  0xb7db12c0 in do_sleep (t=1000) at src/switch_time.c:109
#3  0xb7db15ac in switch_cond_next () at src/switch_time.c:196
#4  0xb7d862b8 in switch_ivr_originate (session=0x0, bleg=0xb48c2f00,
cause=0xb48c2ed4, bridgeto=0xb1133340
"{origination_caller_id_number=0857151465,callback_uuid=c32f54cc-c6b8-11de-af77-774e15238dbe,ignore_early_media=true}sofia/external/66857151465 at 115.178.62.191:7050|sofia/external/66857151465 at 115.178.62"...,
timelimit_sec=60, table=0x0, cid_name_override=0xb7e8c976
"FreeSWITCH", cid_num_override=0x90ec0800 "0857151465",
caller_profile_override=0x0, ovars=0x0, flags=0) at
src/switch_ivr_originate.c:2132
#5  0xb6f0def1 in originate_function (cmd=0x90164ce2
"{origination_caller_id_number=0857151465,callback_uuid=c32f54cc-c6b8-11de-af77-774e15238dbe,ignore_early_media=true}sofia/external/66857151465 at 115.178.62.191:7050|sofia/external/66857151465 at 115.178.62"...,
session=0x0, stream=0xb48c3004) at mod_commands.c:2413
#6  0xb7d66fb1 in switch_api_execute (cmd=0x90164cd8 "originate",
arg=0x90164ce2 "{origination_caller_id_number=0857151465,callback_uuid=c32f54cc-c6b8-11de-af77-774e15238dbe,ignore_early_media=true}sofia/external/66857151465 at 115.178.62.191:7050|sofia/external/66857151465 at 115.178.62"...,
session=0x0, stream=0xb48c3004) at src/switch_loadable_module.c:1567
#7  0xb7d53c4a in switch_core_session_hangup_state
(session=0x958ab678) at src/switch_core_state_machine.c:506
#8  0xb7d385e6 in switch_channel_perform_hangup (channel=0x958afd70,
file=0xb6af8788 "mod_dptools.c", func=0xb6af8ba8 "hangup_function",
line=637, hangup_cause=SWITCH_CAUSE_NORMAL_CLEARING) at
src/switch_channel.c:1898
#9  0xb6aee906 in hangup_function (session=0x958ab678, data=0xb7e82ea5
"") at mod_dptools.c:637
#10 0xb7d4fe1f in switch_core_session_exec (session=0x958ab678,
application_interface=0xb6e10ec0, arg=0xb7e82ea5 "") at
src/switch_core_session.c:1552
#11 0xb7d4f73e in switch_core_session_execute_application
(session=0x958ab678, app=0x90163a80 "hangup", arg=0xb7e82ea5 "") at
src/switch_core_session.c:1474
#12 0xb7d51131 in switch_core_standard_on_execute (session=0x958ab678)
at src/switch_core_state_machine.c:166
#13 0xb7d5215a in switch_core_session_run (session=0x958ab678) at
src/switch_core_state_machine.c:340
#14 0xb7d4e199 in switch_core_session_thread (thread=0x9020bd98,
obj=0x958ab678) at src/switch_core_session.c:1135
#15 0xb7ddc2fc in dummy_worker (opaque=0x9020bd98) at
threadproc/unix/thread.c:138
#16 0xb7caaf3b in start_thread () from /lib/libpthread.so.0
#17 0xb7a64bee in clone () from /lib/libc.so.6


On Sat, Oct 31, 2009 at 12:34 PM, Dome Charoenyost <dome at tel.co.th> wrote:
> I found bug in fscore_pb ? $pwd should be $mypwd
>
> Now i post already. please help me check.
>
> Dome C.
>
>
> 2009/11/1 Jo?o Mesquita <jmesquita at freeswitch.org>:
>> No, mod_nibblebill definetely needs to be enhanced but it is not the problem
>> and it can be used with high load traffic.
>>
>> The one I am not sure about is odbc_query since it was not developed for
>> that.
>>
>> Do what Rupa said, please.
>>
>> Regards,
>>
>> JM
>>
>> On Sat, Oct 31, 2009 at 3:11 PM, Brian West <brian at freeswitch.org> wrote:
>>>
>>> I think once you get the backtrace like rupa said we can see that maybe
>>> odbc_query is really hanging or something similar.
>>> /b
>>> On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:
>>>
>>> 2009/10/31 Brian West <brian at freeswitch.org>:
>>>
>>> You should never do billing inline with the session thread is all I'm
>>>
>>> saying.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa


From dujinfang at gmail.com  Sat Oct 31 11:24:31 2009
From: dujinfang at gmail.com (Seven Du)
Date: Sun, 1 Nov 2009 02:24:31 +0800
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
	<140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
	<AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
	<8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com>
	<EEDE7B6D-F722-4557-BBFE-AC0031C0BA86@freeswitch.org>
	<ebbc2d550910311018m7bc9b86bi120649c36dbf3413@mail.gmail.com>
	<8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com>
Message-ID: <23f91030910311124s1f8844ddw63da23c2ca7ab8a9@mail.gmail.com>

I also got some zombie channels, if someone can help me take a look that's
really nice.

http://pastebin.freeswitch.org/10912

I only loaded mod_cdr_csv
Is it ok to use mod_xml_cdr?

Thanks.


2009/11/1 Dome Charoenyost <dome at tel.co.th>

> I found bug in fscore_pb   $pwd should be $mypwd
>
> Now i post already. please help me check.
>
> Dome C.
>
>
> 2009/11/1 Jo?o Mesquita <jmesquita at freeswitch.org>:
> > No, mod_nibblebill definetely needs to be enhanced but it is not the
> problem
> > and it can be used with high load traffic.
> >
> > The one I am not sure about is odbc_query since it was not developed for
> > that.
> >
> > Do what Rupa said, please.
> >
> > Regards,
> >
> > JM
> >
> > On Sat, Oct 31, 2009 at 3:11 PM, Brian West <brian at freeswitch.org>
> wrote:
> >>
> >> I think once you get the backtrace like rupa said we can see that maybe
> >> odbc_query is really hanging or something similar.
> >> /b
> >> On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:
> >>
> >> 2009/10/31 Brian West <brian at freeswitch.org>:
> >>
> >> You should never do billing inline with the session thread is all I'm
> >>
> >> saying.
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From stevendt at primrosebank.net  Sat Oct 31 11:37:28 2009
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Sat, 31 Oct 2009 18:37:28 -0000
Subject: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question
References: <C4ADBF041AB44AE399847AF51ED740A1@bp1.ad.bp.com>
	<4AEC7E39.5030206@gmail.com>
Message-ID: <FEF9804494E84F9CB430906C465A67F7@bp1.ad.bp.com>

Hi Shelby,

thanks for the reply on the IP Address Question.

I do not have a $FS_DIR/conf/sofia_profiles/ directory ?

$FS_DIR/conf/vars.xml has a line :-
<X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>

I'd guess that this means that the Server uses the machines IP address 
automatically, so, I can change the IP address to fixed and don't need to 
make any changes at the Server end - just need to update the Clients.

regards
Dave


----- Original Message ----- 
From: "Shelby Ramsey" <sicfslist at gmail.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, October 31, 2009 6:13 PM
Subject: Re: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question


> Dave,
>
> You should check out the settings in $FS_DIR/conf/sofia_profiles/ and
> $FS_DIR/conf/vars.xml.  To make the changes take effect just restart FS
> or use the API commands for reloading the sofia profiles.
>
> You'll definitely want to set a static IP address on the FS server (or
> make sure the DHCP server always assigns the same IP address to the FS
> server).  Otherwise you'll constantly be changing the clients that have
> to connect to FS.
>
> SDR
>
>
> Dave Stevenson wrote:
>> Hi,
>>
>> I am new to FreeSwitch  - and VOIP too (as the following questions
>> will indicate).
>>
>> I am trying to setup a VOIP system at home and am just getting
>> started, I have been trawling through the Docs/Wiki and realise that I
>> have a lot to learn !
>>
>> Anyway, here's the first (of what I'm sure will be many !) questions,
>> hopefully someone will take the type to help me out......
>>
>> OK, I have already worked out that FreeSwitch seems to be much better
>> supported under Linux than Windows (is that a fair comment ?), but
>> while I have a long term intention to give Linux a go, it's something
>> that is on the back burner (and has been for some time). At the
>> moment, the current plan is to use FreeSwitch under WindowsXP and I
>> have loaded the latest Windows Install. Everything seems to be working
>> fine so far - well, in test mode anyway, my VOIP phones and teleco
>> interface won't be here until next week, so at the moment, I'm just
>> trying out the system with X-Lite.
>>
>> The first question that I have is about WAPSERVER. This was installed
>> as part of the FreeSwitch install. At the moment, I am manually
>> starting FreeSwitch and WAPSERVER does not seem to start
>> automatically. When I manually start it, it says that it is OFFLINE
>> until I manually put it ONLINE. So :-
>>
>> 1. What does WAPSERVER actually do ?
>> 2. How should it be started ?
>> 3. My next step is to configure the Server PC to auto-logon to my
>> Windows 2003 Server domain and put FreeSwitch in the Startup Group -
>> do I need to add WAPSERVER there too ?
>>
>> The next question is about the ServerIP address. Not thinking far
>> enough ahead, the machine that I installed FreeSwitch on uses DHCP to
>> get its IP Address. I guess that the FreeSwitch server should really
>> have a static IP Address. If I change the PC to have a static IP
>> Address, then
>>
>> 1. Do I need to re-install FreeSwitch
>> 2. If not, is there a or is there command to get FreeSwitch to updated
>> itself with a new IP Address ?
>> 3. Maybe the Server does not need to know the change - maybe if only
>> affect the clients (Phones/SoftPhones ?)
>>
>> Thanks & Best Regards
>> Dave
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From sicfslist at gmail.com  Sat Oct 31 12:07:39 2009
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Sat, 31 Oct 2009 14:07:39 -0500
Subject: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question
In-Reply-To: <FEF9804494E84F9CB430906C465A67F7@bp1.ad.bp.com>
References: <C4ADBF041AB44AE399847AF51ED740A1@bp1.ad.bp.com>	<4AEC7E39.5030206@gmail.com>
	<FEF9804494E84F9CB430906C465A67F7@bp1.ad.bp.com>
Message-ID: <4AEC8AFB.9040306@gmail.com>

Dave,

I'm not sure how the file structure works on Windows.  But sofia is the 
SIP endpoint (and what you're wanting to confgure).

The bind_server setting only relates to mod_dingaling (I think).  Yes 
you can change the IP address ... but you'll need to restart FS or 
reload the sofia profiles.  The conf settings for external_sip_ip, etc. 
are the ones that matter. 

Good luck!  You can also hop on freenode @ #freeswitch and everyone 
there is quite helpful.

SDR




Dave Stevenson wrote:
> Hi Shelby,
>
> thanks for the reply on the IP Address Question.
>
> I do not have a $FS_DIR/conf/sofia_profiles/ directory ?
>
> $FS_DIR/conf/vars.xml has a line :-
> <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>
>
> I'd guess that this means that the Server uses the machines IP address 
> automatically, so, I can change the IP address to fixed and don't need to 
> make any changes at the Server end - just need to update the Clients.
>
> regards
> Dave
>
>
> ----- Original Message ----- 
> From: "Shelby Ramsey" <sicfslist at gmail.com>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Saturday, October 31, 2009 6:13 PM
> Subject: Re: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question
>
>
>   
>> Dave,
>>
>> You should check out the settings in $FS_DIR/conf/sofia_profiles/ and
>> $FS_DIR/conf/vars.xml.  To make the changes take effect just restart FS
>> or use the API commands for reloading the sofia profiles.
>>
>> You'll definitely want to set a static IP address on the FS server (or
>> make sure the DHCP server always assigns the same IP address to the FS
>> server).  Otherwise you'll constantly be changing the clients that have
>> to connect to FS.
>>
>> SDR
>>
>>
>> Dave Stevenson wrote:
>>     
>>> Hi,
>>>
>>> I am new to FreeSwitch  - and VOIP too (as the following questions
>>> will indicate).
>>>
>>> I am trying to setup a VOIP system at home and am just getting
>>> started, I have been trawling through the Docs/Wiki and realise that I
>>> have a lot to learn !
>>>
>>> Anyway, here's the first (of what I'm sure will be many !) questions,
>>> hopefully someone will take the type to help me out......
>>>
>>> OK, I have already worked out that FreeSwitch seems to be much better
>>> supported under Linux than Windows (is that a fair comment ?), but
>>> while I have a long term intention to give Linux a go, it's something
>>> that is on the back burner (and has been for some time). At the
>>> moment, the current plan is to use FreeSwitch under WindowsXP and I
>>> have loaded the latest Windows Install. Everything seems to be working
>>> fine so far - well, in test mode anyway, my VOIP phones and teleco
>>> interface won't be here until next week, so at the moment, I'm just
>>> trying out the system with X-Lite.
>>>
>>> The first question that I have is about WAPSERVER. This was installed
>>> as part of the FreeSwitch install. At the moment, I am manually
>>> starting FreeSwitch and WAPSERVER does not seem to start
>>> automatically. When I manually start it, it says that it is OFFLINE
>>> until I manually put it ONLINE. So :-
>>>
>>> 1. What does WAPSERVER actually do ?
>>> 2. How should it be started ?
>>> 3. My next step is to configure the Server PC to auto-logon to my
>>> Windows 2003 Server domain and put FreeSwitch in the Startup Group -
>>> do I need to add WAPSERVER there too ?
>>>
>>> The next question is about the ServerIP address. Not thinking far
>>> enough ahead, the machine that I installed FreeSwitch on uses DHCP to
>>> get its IP Address. I guess that the FreeSwitch server should really
>>> have a static IP Address. If I change the PC to have a static IP
>>> Address, then
>>>
>>> 1. Do I need to re-install FreeSwitch
>>> 2. If not, is there a or is there command to get FreeSwitch to updated
>>> itself with a new IP Address ?
>>> 3. Maybe the Server does not need to know the change - maybe if only
>>> affect the clients (Phones/SoftPhones ?)
>>>
>>> Thanks & Best Regards
>>> Dave
>>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   



From tayeb.meftah at gmail.com  Sat Oct 31 13:21:20 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sat, 31 Oct 2009 20:21:20 +0000
Subject: [Freeswitch-users] Mod_pjsip
Message-ID: <4AEC9C40.502@gmail.com>

		Hi my FreeSWITCH friends
Today we talked about FreeSWITCH libraries and we found pjsip, which is 
a lot better than sofia, it can do everything sofia can do and more, 
it's thread-safe and performs well, I would love to see a mod_pjsip.
The library is GPL but they also offer alternative licensing, don't get 
me wrong, Sofia is great but I think FreeSWITCH will benefit from a 
thread-safe SIP library.
don't say that i need some stuf for free, but we need to make fs a very
also we love freeswitch to run in a Embedded PC like the blackfin 
CPU/boards, Pjsip is very good for Embedded SIP Devicesawesome Open 
Source project
With much love,
DelphiWorld____ Information from ESET NOD32 Antivirus, version of virus 
signature database 4539 (20091024) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com




__________ Information from ESET NOD32 Antivirus, version of virus 
signature database 4539 (20091024) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com




__________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com




From rupa at rupa.com  Sat Oct 31 12:28:29 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 31 Oct 2009 14:28:29 -0500
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <23f91030910311124s1f8844ddw63da23c2ca7ab8a9@mail.gmail.com>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
	<140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
	<AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
	<8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com>
	<EEDE7B6D-F722-4557-BBFE-AC0031C0BA86@freeswitch.org>
	<ebbc2d550910311018m7bc9b86bi120649c36dbf3413@mail.gmail.com>
	<8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com>
	<23f91030910311124s1f8844ddw63da23c2ca7ab8a9@mail.gmail.com>
Message-ID: <b39273530910311228w2cc72ec1le65d6ffc79ae5b95@mail.gmail.com>

Nothing jumps out at me on this one.

mod_cdr_csv and/or mod_xml_cdr should be fine.

On Sat, Oct 31, 2009 at 1:24 PM, Seven Du <dujinfang at gmail.com> wrote:
> I also got some zombie channels, if someone can help me take a look that's
> really nice.
>
> http://pastebin.freeswitch.org/10912
>
> I only loaded mod_cdr_csv
> Is it ok to use mod_xml_cdr?
>
> Thanks.
>
>
> 2009/11/1 Dome Charoenyost <dome at tel.co.th>
>>
>> I found bug in fscore_pb ? $pwd should be $mypwd
>>
>> Now i post already. please help me check.
>>
>> Dome C.
>>
>>
>> 2009/11/1 Jo?o Mesquita <jmesquita at freeswitch.org>:
>> > No, mod_nibblebill definetely needs to be enhanced but it is not the
>> > problem
>> > and it can be used with high load traffic.
>> >
>> > The one I am not sure about is odbc_query since it was not developed for
>> > that.
>> >
>> > Do what Rupa said, please.
>> >
>> > Regards,
>> >
>> > JM
>> >
>> > On Sat, Oct 31, 2009 at 3:11 PM, Brian West <brian at freeswitch.org>
>> > wrote:
>> >>
>> >> I think once you get the backtrace like rupa said we can see that maybe
>> >> odbc_query is really hanging or something similar.
>> >> /b
>> >> On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:
>> >>
>> >> 2009/10/31 Brian West <brian at freeswitch.org>:
>> >>
>> >> You should never do billing inline with the session thread is all I'm
>> >>
>> >> saying.
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa


From stevendt at primrosebank.net  Sat Oct 31 12:32:53 2009
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Sat, 31 Oct 2009 19:32:53 -0000
Subject: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question
References: <C4ADBF041AB44AE399847AF51ED740A1@bp1.ad.bp.com>	<4AEC7E39.5030206@gmail.com><FEF9804494E84F9CB430906C465A67F7@bp1.ad.bp.com>
	<4AEC8AFB.9040306@gmail.com>
Message-ID: <7DA26A496A784BE398B9356B8D5A27B6@bp1.ad.bp.com>

Thanks Shelby,

the vars.xml file tells me that if external_sip_ip is unspecified (which I 
think it is), then the bind_server_ip is used, i.e., "auto"

Getting off topic, I stumbled on freenode today, I wasn't sure what I needed 
to do to get IRC running on Windows - yet, another "to do" !

(The link on http://wiki.freeswitch.org/wiki/Main_Page to IRC Chat 
(irc://irc.freenode.net/freeswitch) appears to be an invalid link)


regards
Dave


----- Original Message ----- 
From: "Shelby Ramsey" <sicfslist at gmail.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, October 31, 2009 7:07 PM
Subject: Re: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question


> Dave,
>
> I'm not sure how the file structure works on Windows.  But sofia is the
> SIP endpoint (and what you're wanting to confgure).
>
> The bind_server setting only relates to mod_dingaling (I think).  Yes
> you can change the IP address ... but you'll need to restart FS or
> reload the sofia profiles.  The conf settings for external_sip_ip, etc.
> are the ones that matter.
>
> Good luck!  You can also hop on freenode @ #freeswitch and everyone
> there is quite helpful.
>
> SDR
>
>
>
>
> Dave Stevenson wrote:
>> Hi Shelby,
>>
>> thanks for the reply on the IP Address Question.
>>
>> I do not have a $FS_DIR/conf/sofia_profiles/ directory ?
>>
>> $FS_DIR/conf/vars.xml has a line :-
>> <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>
>>
>> I'd guess that this means that the Server uses the machines IP address
>> automatically, so, I can change the IP address to fixed and don't need to
>> make any changes at the Server end - just need to update the Clients.
>>
>> regards
>> Dave
>>
>>
>> ----- Original Message ----- 
>> From: "Shelby Ramsey" <sicfslist at gmail.com>
>> To: <freeswitch-users at lists.freeswitch.org>
>> Sent: Saturday, October 31, 2009 6:13 PM
>> Subject: Re: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question
>>
>>
>>
>>> Dave,
>>>
>>> You should check out the settings in $FS_DIR/conf/sofia_profiles/ and
>>> $FS_DIR/conf/vars.xml.  To make the changes take effect just restart FS
>>> or use the API commands for reloading the sofia profiles.
>>>
>>> You'll definitely want to set a static IP address on the FS server (or
>>> make sure the DHCP server always assigns the same IP address to the FS
>>> server).  Otherwise you'll constantly be changing the clients that have
>>> to connect to FS.
>>>
>>> SDR
>>>
>>>
>>> Dave Stevenson wrote:
>>>
>>>> Hi,
>>>>
>>>> I am new to FreeSwitch  - and VOIP too (as the following questions
>>>> will indicate).
>>>>
>>>> I am trying to setup a VOIP system at home and am just getting
>>>> started, I have been trawling through the Docs/Wiki and realise that I
>>>> have a lot to learn !
>>>>
>>>> Anyway, here's the first (of what I'm sure will be many !) questions,
>>>> hopefully someone will take the type to help me out......
>>>>
>>>> OK, I have already worked out that FreeSwitch seems to be much better
>>>> supported under Linux than Windows (is that a fair comment ?), but
>>>> while I have a long term intention to give Linux a go, it's something
>>>> that is on the back burner (and has been for some time). At the
>>>> moment, the current plan is to use FreeSwitch under WindowsXP and I
>>>> have loaded the latest Windows Install. Everything seems to be working
>>>> fine so far - well, in test mode anyway, my VOIP phones and teleco
>>>> interface won't be here until next week, so at the moment, I'm just
>>>> trying out the system with X-Lite.
>>>>
>>>> The first question that I have is about WAPSERVER. This was installed
>>>> as part of the FreeSwitch install. At the moment, I am manually
>>>> starting FreeSwitch and WAPSERVER does not seem to start
>>>> automatically. When I manually start it, it says that it is OFFLINE
>>>> until I manually put it ONLINE. So :-
>>>>
>>>> 1. What does WAPSERVER actually do ?
>>>> 2. How should it be started ?
>>>> 3. My next step is to configure the Server PC to auto-logon to my
>>>> Windows 2003 Server domain and put FreeSwitch in the Startup Group -
>>>> do I need to add WAPSERVER there too ?
>>>>
>>>> The next question is about the ServerIP address. Not thinking far
>>>> enough ahead, the machine that I installed FreeSwitch on uses DHCP to
>>>> get its IP Address. I guess that the FreeSwitch server should really
>>>> have a static IP Address. If I change the PC to have a static IP
>>>> Address, then
>>>>
>>>> 1. Do I need to re-install FreeSwitch
>>>> 2. If not, is there a or is there command to get FreeSwitch to updated
>>>> itself with a new IP Address ?
>>>> 3. Maybe the Server does not need to know the change - maybe if only
>>>> affect the clients (Phones/SoftPhones ?)
>>>>
>>>> Thanks & Best Regards
>>>> Dave
>>>> ------------------------------------------------------------------------
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From mgg at giagnocavo.net  Sat Oct 31 13:00:52 2009
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Sat, 31 Oct 2009 16:00:52 -0400
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <b39273530910311228w2cc72ec1le65d6ffc79ae5b95@mail.gmail.com>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
	<140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
	<AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
	<8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com>
	<EEDE7B6D-F722-4557-BBFE-AC0031C0BA86@freeswitch.org>
	<ebbc2d550910311018m7bc9b86bi120649c36dbf3413@mail.gmail.com>
	<8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com>
	<23f91030910311124s1f8844ddw63da23c2ca7ab8a9@mail.gmail.com>
	<b39273530910311228w2cc72ec1le65d6ffc79ae5b95@mail.gmail.com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C6703243E2960@mse17be1.mse17.exchange.ms>

Which revision? Wasn't there some read-write lock issues on sessions checked in during the last week?

FWIW, we do inline billing, but the message queue timeouts are very low. Upgrading to current SVN kills us (~15025 works fine) -- some channels stay in CS_EXECUTE, some in CS_REPORTING. We know it's not our code blocking, because there'd be transaction timeouts within 30 seconds, and these are hanging around for a long time.

I suspect it's something with how we do our originates and state handlers. If we _overwrite_ the default originate handlers AND patch originate to blank out dialplan/context after duplicating the caller profile, it works fine. If we "do the right thing" and let the default handler set CS_CONSUME_MEDIA, then we get the calls hanging around forever. As I said, I suspect this is mismatch in what FS does and what our app assumes it does.

-Michael

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker
Sent: Saturday, October 31, 2009 1:28 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Many CS_REPORTING state Zombie session

Nothing jumps out at me on this one.

mod_cdr_csv and/or mod_xml_cdr should be fine.

On Sat, Oct 31, 2009 at 1:24 PM, Seven Du <dujinfang at gmail.com> wrote:
> I also got some zombie channels, if someone can help me take a look that's
> really nice.
>
> http://pastebin.freeswitch.org/10912
>
> I only loaded mod_cdr_csv
> Is it ok to use mod_xml_cdr?
>
> Thanks.
>
>
> 2009/11/1 Dome Charoenyost <dome at tel.co.th>
>>
>> I found bug in fscore_pb ? $pwd should be $mypwd
>>
>> Now i post already. please help me check.
>>
>> Dome C.
>>
>>
>> 2009/11/1 Jo?o Mesquita <jmesquita at freeswitch.org>:
>> > No, mod_nibblebill definetely needs to be enhanced but it is not the
>> > problem
>> > and it can be used with high load traffic.
>> >
>> > The one I am not sure about is odbc_query since it was not developed for
>> > that.
>> >
>> > Do what Rupa said, please.
>> >
>> > Regards,
>> >
>> > JM
>> >
>> > On Sat, Oct 31, 2009 at 3:11 PM, Brian West <brian at freeswitch.org>
>> > wrote:
>> >>
>> >> I think once you get the backtrace like rupa said we can see that maybe
>> >> odbc_query is really hanging or something similar.
>> >> /b
>> >> On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:
>> >>
>> >> 2009/10/31 Brian West <brian at freeswitch.org>:
>> >>
>> >> You should never do billing inline with the session thread is all I'm
>> >>
>> >> saying.
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From sicfslist at gmail.com  Sat Oct 31 13:09:04 2009
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Sat, 31 Oct 2009 15:09:04 -0500
Subject: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question
In-Reply-To: <7DA26A496A784BE398B9356B8D5A27B6@bp1.ad.bp.com>
References: <C4ADBF041AB44AE399847AF51ED740A1@bp1.ad.bp.com>	<4AEC7E39.5030206@gmail.com><FEF9804494E84F9CB430906C465A67F7@bp1.ad.bp.com>	<4AEC8AFB.9040306@gmail.com>
	<7DA26A496A784BE398B9356B8D5A27B6@bp1.ad.bp.com>
Message-ID: <4AEC9960.8010204@gmail.com>

Dave,

As I mentioned before ... the bind_server_ip directive is only for 
mod_dingaling.  I don't think sofia will use it.

As for IRC ... I have no knowledge of Windows machines or apps ... but I 
know there is an IRC client for FireFox called chatzilla.

SDR



Dave Stevenson wrote:
> Thanks Shelby,
>
> the vars.xml file tells me that if external_sip_ip is unspecified (which I 
> think it is), then the bind_server_ip is used, i.e., "auto"
>
> Getting off topic, I stumbled on freenode today, I wasn't sure what I needed 
> to do to get IRC running on Windows - yet, another "to do" !
>
> (The link on http://wiki.freeswitch.org/wiki/Main_Page to IRC Chat 
> (irc://irc.freenode.net/freeswitch) appears to be an invalid link)
>
>
> regards
> Dave
>
>
> ----- Original Message ----- 
> From: "Shelby Ramsey" <sicfslist at gmail.com>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Saturday, October 31, 2009 7:07 PM
> Subject: Re: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question
>
>
>   
>> Dave,
>>
>> I'm not sure how the file structure works on Windows.  But sofia is the
>> SIP endpoint (and what you're wanting to confgure).
>>
>> The bind_server setting only relates to mod_dingaling (I think).  Yes
>> you can change the IP address ... but you'll need to restart FS or
>> reload the sofia profiles.  The conf settings for external_sip_ip, etc.
>> are the ones that matter.
>>
>> Good luck!  You can also hop on freenode @ #freeswitch and everyone
>> there is quite helpful.
>>
>> SDR
>>
>>
>>
>>
>> Dave Stevenson wrote:
>>     
>>> Hi Shelby,
>>>
>>> thanks for the reply on the IP Address Question.
>>>
>>> I do not have a $FS_DIR/conf/sofia_profiles/ directory ?
>>>
>>> $FS_DIR/conf/vars.xml has a line :-
>>> <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>
>>>
>>> I'd guess that this means that the Server uses the machines IP address
>>> automatically, so, I can change the IP address to fixed and don't need to
>>> make any changes at the Server end - just need to update the Clients.
>>>
>>> regards
>>> Dave
>>>
>>>
>>> ----- Original Message ----- 
>>> From: "Shelby Ramsey" <sicfslist at gmail.com>
>>> To: <freeswitch-users at lists.freeswitch.org>
>>> Sent: Saturday, October 31, 2009 6:13 PM
>>> Subject: Re: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question
>>>
>>>
>>>
>>>       
>>>> Dave,
>>>>
>>>> You should check out the settings in $FS_DIR/conf/sofia_profiles/ and
>>>> $FS_DIR/conf/vars.xml.  To make the changes take effect just restart FS
>>>> or use the API commands for reloading the sofia profiles.
>>>>
>>>> You'll definitely want to set a static IP address on the FS server (or
>>>> make sure the DHCP server always assigns the same IP address to the FS
>>>> server).  Otherwise you'll constantly be changing the clients that have
>>>> to connect to FS.
>>>>
>>>> SDR
>>>>
>>>>
>>>> Dave Stevenson wrote:
>>>>
>>>>         
>>>>> Hi,
>>>>>
>>>>> I am new to FreeSwitch  - and VOIP too (as the following questions
>>>>> will indicate).
>>>>>
>>>>> I am trying to setup a VOIP system at home and am just getting
>>>>> started, I have been trawling through the Docs/Wiki and realise that I
>>>>> have a lot to learn !
>>>>>
>>>>> Anyway, here's the first (of what I'm sure will be many !) questions,
>>>>> hopefully someone will take the type to help me out......
>>>>>
>>>>> OK, I have already worked out that FreeSwitch seems to be much better
>>>>> supported under Linux than Windows (is that a fair comment ?), but
>>>>> while I have a long term intention to give Linux a go, it's something
>>>>> that is on the back burner (and has been for some time). At the
>>>>> moment, the current plan is to use FreeSwitch under WindowsXP and I
>>>>> have loaded the latest Windows Install. Everything seems to be working
>>>>> fine so far - well, in test mode anyway, my VOIP phones and teleco
>>>>> interface won't be here until next week, so at the moment, I'm just
>>>>> trying out the system with X-Lite.
>>>>>
>>>>> The first question that I have is about WAPSERVER. This was installed
>>>>> as part of the FreeSwitch install. At the moment, I am manually
>>>>> starting FreeSwitch and WAPSERVER does not seem to start
>>>>> automatically. When I manually start it, it says that it is OFFLINE
>>>>> until I manually put it ONLINE. So :-
>>>>>
>>>>> 1. What does WAPSERVER actually do ?
>>>>> 2. How should it be started ?
>>>>> 3. My next step is to configure the Server PC to auto-logon to my
>>>>> Windows 2003 Server domain and put FreeSwitch in the Startup Group -
>>>>> do I need to add WAPSERVER there too ?
>>>>>
>>>>> The next question is about the ServerIP address. Not thinking far
>>>>> enough ahead, the machine that I installed FreeSwitch on uses DHCP to
>>>>> get its IP Address. I guess that the FreeSwitch server should really
>>>>> have a static IP Address. If I change the PC to have a static IP
>>>>> Address, then
>>>>>
>>>>> 1. Do I need to re-install FreeSwitch
>>>>> 2. If not, is there a or is there command to get FreeSwitch to updated
>>>>> itself with a new IP Address ?
>>>>> 3. Maybe the Server does not need to know the change - maybe if only
>>>>> affect the clients (Phones/SoftPhones ?)
>>>>>
>>>>> Thanks & Best Regards
>>>>> Dave
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>           
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>         
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   



From diego.viola at gmail.com  Sat Oct 31 13:14:10 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 31 Oct 2009 20:14:10 +0000
Subject: [Freeswitch-users] Mod_pjsip
In-Reply-To: <4AEC9C40.502@gmail.com>
References: <4AEC9C40.502@gmail.com>
Message-ID: <86a32abc0910311314m21345e7fh1f3c673bee0e5667@mail.gmail.com>

This is pretty interesting, is sofia thread safe as well?

Diego

On Sat, Oct 31, 2009 at 8:21 PM, Meftah Tayeb <tayeb.meftah at gmail.com> wrote:
> ? ? ? ? ? ? ? ?Hi my FreeSWITCH friends
> Today we talked about FreeSWITCH libraries and we found pjsip, which is
> a lot better than sofia, it can do everything sofia can do and more,
> it's thread-safe and performs well, I would love to see a mod_pjsip.
> The library is GPL but they also offer alternative licensing, don't get
> me wrong, Sofia is great but I think FreeSWITCH will benefit from a
> thread-safe SIP library.
> don't say that i need some stuf for free, but we need to make fs a very
> also we love freeswitch to run in a Embedded PC like the blackfin
> CPU/boards, Pjsip is very good for Embedded SIP Devicesawesome Open
> Source project
> With much love,
> DelphiWorld____ Information from ESET NOD32 Antivirus, version of virus
> signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus
> signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From msc at freeswitch.org  Sat Oct 31 13:16:07 2009
From: msc at freeswitch.org (Michael S Collins)
Date: Sat, 31 Oct 2009 13:16:07 -0700
Subject: [Freeswitch-users] Mod_pjsip
In-Reply-To: <4AEC9C40.502@gmail.com>
References: <4AEC9C40.502@gmail.com>
Message-ID: <14A50F45-F0E7-4FEF-8C1C-66EA24CA64A7@freeswitch.org>

I can guarantee that the FS devs are well aware of pj-sip. If it was/ 
is a viable alternative then it would be considered. The fact that it  
isn't being used is a pretty good indication that it isn't suitable  
for FS at this time.

-MV

Sent from my iPhone

On Oct 31, 2009, at 1:21 PM, Meftah Tayeb <tayeb.meftah at gmail.com>  
wrote:

>        Hi my FreeSWITCH friends
> Today we talked about FreeSWITCH libraries and we found pjsip, which  
> is
> a lot better than sofia, it can do everything sofia can do and more,
> it's thread-safe and performs well, I would love to see a mod_pjsip.
> The library is GPL but they also offer alternative licensing, don't  
> get
> me wrong, Sofia is great but I think FreeSWITCH will benefit from a
> thread-safe SIP library.
> don't say that i need some stuf for free, but we need to make fs a  
> very
> also we love freeswitch to run in a Embedded PC like the blackfin
> CPU/boards, Pjsip is very good for Embedded SIP Devicesawesome Open
> Source project
> With much love,
> DelphiWorld____ Information from ESET NOD32 Antivirus, version of  
> virus
> signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus
> signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus  
> signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From stkn at freeswitch.org  Sat Oct 31 14:04:58 2009
From: stkn at freeswitch.org (Stefan Knoblich)
Date: Sat, 31 Oct 2009 22:04:58 +0100
Subject: [Freeswitch-users] Mod_pjsip
In-Reply-To: <14A50F45-F0E7-4FEF-8C1C-66EA24CA64A7@freeswitch.org>
References: <4AEC9C40.502@gmail.com>
	<14A50F45-F0E7-4FEF-8C1C-66EA24CA64A7@freeswitch.org>
Message-ID: <4AECA67A.7030708@freeswitch.org>

Michael S Collins wrote:
> I can guarantee that the FS devs are well aware of pj-sip. If it was/ 
> is a viable alternative then it would be considered. The fact that it  
> isn't being used is a pretty good indication that it isn't suitable  
> for FS at this time.
> 
> -MV
> 
> Sent from my iPhone

We already mentioned some of the reasons why it did get dropped 3 years ago
(first two points from memory, last two from old IRC logs):


[License incompatible (GPL), but i think tony tried to negotiate on alternate license terms]

Not possible to have multiple SIP profiles (due to global variables being used in the lib).

A race-condition under high load, that couldn't be resolved back then (with the help of the pjsip developers).

And the sofia module "just working" and surviving the scalability tests, so all efforts were focussed on
mod_sofia and pjsip got dropped.



stkn


From shiyanov at gmail.com  Sat Oct 31 14:09:56 2009
From: shiyanov at gmail.com (Artem Shiyanov)
Date: Sun, 1 Nov 2009 00:09:56 +0300
Subject: [Freeswitch-users] [mod_socket] Can't set channel variable SOMETIMES
Message-ID: <d211f54a0910311409n530c4049re16243aa524334a4@mail.gmail.com>

Hello!

I've got strange problem:
In my app which talks to FreeSwitch via mod_socket there is such logic:
<pre conditions: channel1 - newly created, parked channell - with help of
'uuid_create' and 'originate ... &park'; channel2 is talking with channel3>
for (channel in {array of channel1, channel2, channel3}) {
 set value for the channel variable X
}

uuid_transfer channel1 extension Y
uuid_transfer -both channel2 extension Y


In extension Y there is a logic based on the channel variable X.
The problem itself: every second launching falls, the reason - channel
variable X hasn't been set for one of channel1..3.
Sniffer shows that commands are sent (and confirmated by FreeSwitch)
consiquently, FreeSwitch sends OK for each command in time.
Temporary sollution that I found is to wait 0.5 sec after 'set' commands
have been sent and before transfer any channel to extension Y.

Maybe some already faced with similar problem?


Thanks,
Artem
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From anthony.minessale at gmail.com  Sat Oct 31 14:26:09 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 31 Oct 2009 16:26:09 -0500
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6703243E2960@mse17be1.mse17.exchange.ms>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
	<AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
	<8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com>
	<EEDE7B6D-F722-4557-BBFE-AC0031C0BA86@freeswitch.org>
	<ebbc2d550910311018m7bc9b86bi120649c36dbf3413@mail.gmail.com>
	<8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com>
	<23f91030910311124s1f8844ddw63da23c2ca7ab8a9@mail.gmail.com>
	<b39273530910311228w2cc72ec1le65d6ffc79ae5b95@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C6703243E2960@mse17be1.mse17.exchange.ms>
Message-ID: <191c3a030910311426of22a4fdyb6be7cabfb992bd6@mail.gmail.com>

it something doing a read/write lock then not clearing it or its bad build
and you have to rebuild clean.
the trace had no session threads present in it.
look in the code of any non-standard mods you are using for something that
did either switch_ivr_originate or switch_core_session_locate and never
called switch_core_session_rwunlock

#define SWITCH_DEBUG_RWLOCKS at the top of switch.h
and make clean ; make install to get rwlock debugging.


On Sat, Oct 31, 2009 at 3:00 PM, Michael Giagnocavo <mgg at giagnocavo.net>wrote:

> Which revision? Wasn't there some read-write lock issues on sessions
> checked in during the last week?
>
> FWIW, we do inline billing, but the message queue timeouts are very low.
> Upgrading to current SVN kills us (~15025 works fine) -- some channels stay
> in CS_EXECUTE, some in CS_REPORTING. We know it's not our code blocking,
> because there'd be transaction timeouts within 30 seconds, and these are
> hanging around for a long time.
>
> I suspect it's something with how we do our originates and state handlers.
> If we _overwrite_ the default originate handlers AND patch originate to
> blank out dialplan/context after duplicating the caller profile, it works
> fine. If we "do the right thing" and let the default handler set
> CS_CONSUME_MEDIA, then we get the calls hanging around forever. As I said, I
> suspect this is mismatch in what FS does and what our app assumes it does.
>
> -Michael
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker
> Sent: Saturday, October 31, 2009 1:28 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Many CS_REPORTING state Zombie session
>
> Nothing jumps out at me on this one.
>
> mod_cdr_csv and/or mod_xml_cdr should be fine.
>
> On Sat, Oct 31, 2009 at 1:24 PM, Seven Du <dujinfang at gmail.com> wrote:
> > I also got some zombie channels, if someone can help me take a look
> that's
> > really nice.
> >
> > http://pastebin.freeswitch.org/10912
> >
> > I only loaded mod_cdr_csv
> > Is it ok to use mod_xml_cdr?
> >
> > Thanks.
> >
> >
> > 2009/11/1 Dome Charoenyost <dome at tel.co.th>
> >>
> >> I found bug in fscore_pb   $pwd should be $mypwd
> >>
> >> Now i post already. please help me check.
> >>
> >> Dome C.
> >>
> >>
> >> 2009/11/1 Jo?o Mesquita <jmesquita at freeswitch.org>:
> >> > No, mod_nibblebill definetely needs to be enhanced but it is not the
> >> > problem
> >> > and it can be used with high load traffic.
> >> >
> >> > The one I am not sure about is odbc_query since it was not developed
> for
> >> > that.
> >> >
> >> > Do what Rupa said, please.
> >> >
> >> > Regards,
> >> >
> >> > JM
> >> >
> >> > On Sat, Oct 31, 2009 at 3:11 PM, Brian West <brian at freeswitch.org>
> >> > wrote:
> >> >>
> >> >> I think once you get the backtrace like rupa said we can see that
> maybe
> >> >> odbc_query is really hanging or something similar.
> >> >> /b
> >> >> On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:
> >> >>
> >> >> 2009/10/31 Brian West <brian at freeswitch.org>:
> >> >>
> >> >> You should never do billing inline with the session thread is all I'm
> >> >>
> >> >> saying.
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >>
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From anthony.minessale at gmail.com  Sat Oct 31 14:31:55 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 31 Oct 2009 16:31:55 -0500
Subject: [Freeswitch-users] [mod_socket] Can't set channel variable
	SOMETIMES
In-Reply-To: <d211f54a0910311409n530c4049re16243aa524334a4@mail.gmail.com>
References: <d211f54a0910311409n530c4049re16243aa524334a4@mail.gmail.com>
Message-ID: <191c3a030910311431r7f73c725g3f0834243e6c4a0f@mail.gmail.com>

wait for execute_complete events to be sure you application execute request
was received and processed or use uuid_setvar instead which is instant.



On Sat, Oct 31, 2009 at 4:09 PM, Artem Shiyanov <shiyanov at gmail.com> wrote:

> Hello!
>
> I've got strange problem:
> In my app which talks to FreeSwitch via mod_socket there is such logic:
> <pre conditions: channel1 - newly created, parked channell - with help of
> 'uuid_create' and 'originate ... &park'; channel2 is talking with channel3>
> for (channel in {array of channel1, channel2, channel3}) {
>  set value for the channel variable X
> }
>
> uuid_transfer channel1 extension Y
> uuid_transfer -both channel2 extension Y
>
>
> In extension Y there is a logic based on the channel variable X.
> The problem itself: every second launching falls, the reason - channel
> variable X hasn't been set for one of channel1..3.
> Sniffer shows that commands are sent (and confirmated by FreeSwitch)
> consiquently, FreeSwitch sends OK for each command in time.
> Temporary sollution that I found is to wait 0.5 sec after 'set' commands
> have been sent and before transfer any channel to extension Y.
>
> Maybe some already faced with similar problem?
>
>
> Thanks,
> Artem
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From tayeb.meftah at gmail.com  Sat Oct 31 15:44:28 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sat, 31 Oct 2009 22:44:28 +0000
Subject: [Freeswitch-users] Mod_pjsip
In-Reply-To: <4AECA67A.7030708@freeswitch.org>
References: <4AEC9C40.502@gmail.com>	<14A50F45-F0E7-4FEF-8C1C-66EA24CA64A7@freeswitch.org>
	<4AECA67A.7030708@freeswitch.org>
Message-ID: <4AECBDCC.9000105@gmail.com>

hi,
Pjsip support ICE, STUN and TURN!
to STKN:
if you don't pjsip, please stop talking or exit the discution
we want to kype Freeswitch Clean and universal
Stefan Knoblich a ?crit :
> Michael S Collins wrote:
>   
>> I can guarantee that the FS devs are well aware of pj-sip. If it was/ 
>> is a viable alternative then it would be considered. The fact that it  
>> isn't being used is a pretty good indication that it isn't suitable  
>> for FS at this time.
>>
>> -MV
>>
>> Sent from my iPhone
>>     
>
> We already mentioned some of the reasons why it did get dropped 3 years ago
> (first two points from memory, last two from old IRC logs):
>
>
> [License incompatible (GPL), but i think tony tried to negotiate on alternate license terms]
>
> Not possible to have multiple SIP profiles (due to global variables being used in the lib).
>
> A race-condition under high load, that couldn't be resolved back then (with the help of the pjsip developers).
>
> And the sofia module "just working" and surviving the scalability tests, so all efforts were focussed on
> mod_sofia and pjsip got dropped.
>
>
>
> stkn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
>   





__________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From stkn at freeswitch.org  Sat Oct 31 14:53:48 2009
From: stkn at freeswitch.org (Stefan Knoblich)
Date: Sat, 31 Oct 2009 22:53:48 +0100
Subject: [Freeswitch-users] Mod_pjsip
In-Reply-To: <4AECBDCC.9000105@gmail.com>
References: <4AEC9C40.502@gmail.com>	<14A50F45-F0E7-4FEF-8C1C-66EA24CA64A7@freeswitch.org>	<4AECA67A.7030708@freeswitch.org>
	<4AECBDCC.9000105@gmail.com>
Message-ID: <4AECB1EC.6050301@freeswitch.org>

Meftah Tayeb wrote:
> hi,
> Pjsip support ICE, STUN and TURN!
> to STKN:
> if you don't pjsip, please stop talking or exit the discution
> we want to kype Freeswitch Clean and universal

Huh?

In which part did i tell people to stop doing whatever they want with pjsip?

In my last email or on IRC?

I'm merely pointing out the reasons why mod_pjsip was abandoned by the core team 3 years ago
(so some of the points may have been fixed in the meantime).

If somebody feels the need to resurrect it, well, please do.
But don't expect Tony or anyone else from the core team to do that for you.

This is OpenSource after all...


stkn


From anthony.minessale at gmail.com  Sat Oct 31 14:55:04 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 31 Oct 2009 16:55:04 -0500
Subject: [Freeswitch-users] Mod_pjsip
In-Reply-To: <4AECBDCC.9000105@gmail.com>
References: <4AEC9C40.502@gmail.com>
	<14A50F45-F0E7-4FEF-8C1C-66EA24CA64A7@freeswitch.org>
	<4AECA67A.7030708@freeswitch.org> <4AECBDCC.9000105@gmail.com>
Message-ID: <191c3a030910311455j2408a0a2r6caa49375ed10b78@mail.gmail.com>

Meftah,

He is 100% correct.  Please do not insult my volunteer developers. Without
help from him you would not have any FreeSWITCH right now so please drop
this subject we are not using pjsip.



On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb <tayeb.meftah at gmail.com>wrote:

>  hi,
> Pjsip support ICE, STUN and TURN!
> to STKN:
> if you don't pjsip, please stop talking or exit the discution
> we want to kype Freeswitch Clean and universal
> Stefan Knoblich a ?crit :
>
> Michael S Collins wrote:
>
>
>  I can guarantee that the FS devs are well aware of pj-sip. If it was/
> is a viable alternative then it would be considered. The fact that it
> isn't being used is a pretty good indication that it isn't suitable
> for FS at this time.
>
> -MV
>
> Sent from my iPhone
>
>
>
> We already mentioned some of the reasons why it did get dropped 3 years ago
> (first two points from memory, last two from old IRC logs):
>
>
> [License incompatible (GPL), but i think tony tried to negotiate on alternate license terms]
>
> Not possible to have multiple SIP profiles (due to global variables being used in the lib).
>
> A race-condition under high load, that couldn't be resolved back then (with the help of the pjsip developers).
>
> And the sofia module "just working" and surviving the scalability tests, so all efforts were focussed on
> mod_sofia and pjsip got dropped.
>
>
>
> stkn
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
> http://www.eset.com
>
>
>
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus
> signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From tayeb.meftah at gmail.com  Sat Oct 31 16:06:45 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sat, 31 Oct 2009 23:06:45 +0000
Subject: [Freeswitch-users] Mod_pjsip
In-Reply-To: <191c3a030910311455j2408a0a2r6caa49375ed10b78@mail.gmail.com>
References: <4AEC9C40.502@gmail.com>	<14A50F45-F0E7-4FEF-8C1C-66EA24CA64A7@freeswitch.org>	<4AECA67A.7030708@freeswitch.org>
	<4AECBDCC.9000105@gmail.com>
	<191c3a030910311455j2408a0a2r6caa49375ed10b78@mail.gmail.com>
Message-ID: <4AECC305.5000805@gmail.com>

hi Anthony
i agry
i say that because STKN hate all my suggestions.
about pjsip, i will contribute aditional module in the contrib.
thanks Anthony
Anthony Minessale a ?crit :
> Meftah,
>
> He is 100% correct.  Please do not insult my volunteer developers. 
> Without help from him you would not have any FreeSWITCH right now so 
> please drop this subject we are not using pjsip.
>
>
>
> On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb <tayeb.meftah at gmail.com 
> <mailto:tayeb.meftah at gmail.com>> wrote:
>
>     hi,
>     Pjsip support ICE, STUN and TURN!
>     to STKN:
>     if you don't pjsip, please stop talking or exit the discution
>     we want to kype Freeswitch Clean and universal
>     Stefan Knoblich a ?crit :
>>     Michael S Collins wrote:
>>       
>>>     I can guarantee that the FS devs are well aware of pj-sip. If it was/ 
>>>     is a viable alternative then it would be considered. The fact that it  
>>>     isn't being used is a pretty good indication that it isn't suitable  
>>>     for FS at this time.
>>>
>>>     -MV
>>>
>>>     Sent from my iPhone
>>>         
>>     We already mentioned some of the reasons why it did get dropped 3
>>     years ago (first two points from memory, last two from old IRC
>>     logs): [License incompatible (GPL), but i think tony tried to
>>     negotiate on alternate license terms] Not possible to have
>>     multiple SIP profiles (due to global variables being used in the
>>     lib). A race-condition under high load, that couldn't be resolved
>>     back then (with the help of the pjsip developers). And the sofia
>>     module "just working" and surviving the scalability tests, so all
>>     efforts were focussed on mod_sofia and pjsip got dropped. stkn
>>     _______________________________________________ FreeSWITCH-users
>>     mailing list FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>
>>
>>
>>     __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________
>>
>>     The message was checked by ESET NOD32 Antivirus.
>>
>>     http://www.eset.com
>>
>>
>>       
>
>
>
>
>     __________ Information from ESET NOD32 Antivirus, version of virus
>     signature database 4539 (20091024) __________
>
>     The message was checked by ESET NOD32 Antivirus.
>
>     http://www.eset.com
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>   




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The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From anthony.minessale at gmail.com  Sat Oct 31 15:17:19 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 31 Oct 2009 17:17:19 -0500
Subject: [Freeswitch-users] Mod_pjsip
In-Reply-To: <4AECC305.5000805@gmail.com>
References: <4AEC9C40.502@gmail.com>
	<14A50F45-F0E7-4FEF-8C1C-66EA24CA64A7@freeswitch.org>
	<4AECA67A.7030708@freeswitch.org> <4AECBDCC.9000105@gmail.com>
	<191c3a030910311455j2408a0a2r6caa49375ed10b78@mail.gmail.com>
	<4AECC305.5000805@gmail.com>
Message-ID: <191c3a030910311517v46a9a830xa59350f6f8e2f025@mail.gmail.com>

Meftah,
Feel free.

P.S.

STKN was the guy who made the first mod_pjsip for FS that we abandoned years
ago. So you should believe him.
Both him and I agreed it was not working out.  So if you don't believe me,
find out for yourself.




On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb <tayeb.meftah at gmail.com>wrote:

>  hi Anthony
> i agry
> i say that because STKN hate all my suggestions.
> about pjsip, i will contribute aditional module in the contrib.
> thanks Anthony
> Anthony Minessale a ?crit :
>
> Meftah,
>
> He is 100% correct.  Please do not insult my volunteer developers. Without
> help from him you would not have any FreeSWITCH right now so please drop
> this subject we are not using pjsip.
>
>
>
> On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb <tayeb.meftah at gmail.com>wrote:
>
> hi,
> Pjsip support ICE, STUN and TURN!
> to STKN:
> if you don't pjsip, please stop talking or exit the discution
> we want to kype Freeswitch Clean and universal
> Stefan Knoblich a ?crit :
>
>  Michael S Collins wrote:
>
>
>  I can guarantee that the FS devs are well aware of pj-sip. If it was/
> is a viable alternative then it would be considered. The fact that it
> isn't being used is a pretty good indication that it isn't suitable
> for FS at this time.
>
> -MV
>
> Sent from my iPhone
>
>
>
> We already mentioned some of the reasons why it did get dropped 3 years ago
> (first two points from memory, last two from old IRC logs):
>
>
> [License incompatible (GPL), but i think tony tried to negotiate on alternate license terms]
>
> Not possible to have multiple SIP profiles (due to global variables being used in the lib).
>
> A race-condition under high load, that couldn't be resolved back then (with the help of the pjsip developers).
>
> And the sofia module "just working" and surviving the scalability tests, so all efforts were focussed on
> mod_sofia and pjsip got dropped.
>
>
>
> stkn
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
> http://www.eset.com
>
>
>
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus
> signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
> http://www.eset.com
>
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus
> signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From tayeb.meftah at gmail.com  Sat Oct 31 16:27:27 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sat, 31 Oct 2009 23:27:27 +0000
Subject: [Freeswitch-users] Mod_pjsip
In-Reply-To: <191c3a030910311455j2408a0a2r6caa49375ed10b78@mail.gmail.com>
References: <4AEC9C40.502@gmail.com>	<14A50F45-F0E7-4FEF-8C1C-66EA24CA64A7@freeswitch.org>	<4AECA67A.7030708@freeswitch.org>
	<4AECBDCC.9000105@gmail.com>
	<191c3a030910311455j2408a0a2r6caa49375ed10b78@mail.gmail.com>
Message-ID: <4AECC7DF.9060209@gmail.com>

hello anthony
we are realy sory about this subject
"Ok, I apologize, I really love you and respect you, and I believe you"
i will drop it from now.
Anthony Minessale a ?crit :
> Meftah,
>
> He is 100% correct.  Please do not insult my volunteer developers. 
> Without help from him you would not have any FreeSWITCH right now so 
> please drop this subject we are not using pjsip.
>
>
>
> On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb <tayeb.meftah at gmail.com 
> <mailto:tayeb.meftah at gmail.com>> wrote:
>
>     hi,
>     Pjsip support ICE, STUN and TURN!
>     to STKN:
>     if you don't pjsip, please stop talking or exit the discution
>     we want to kype Freeswitch Clean and universal
>     Stefan Knoblich a ?crit :
>>     Michael S Collins wrote:
>>       
>>>     I can guarantee that the FS devs are well aware of pj-sip. If it was/ 
>>>     is a viable alternative then it would be considered. The fact that it  
>>>     isn't being used is a pretty good indication that it isn't suitable  
>>>     for FS at this time.
>>>
>>>     -MV
>>>
>>>     Sent from my iPhone
>>>         
>>     We already mentioned some of the reasons why it did get dropped 3
>>     years ago (first two points from memory, last two from old IRC
>>     logs): [License incompatible (GPL), but i think tony tried to
>>     negotiate on alternate license terms] Not possible to have
>>     multiple SIP profiles (due to global variables being used in the
>>     lib). A race-condition under high load, that couldn't be resolved
>>     back then (with the help of the pjsip developers). And the sofia
>>     module "just working" and surviving the scalability tests, so all
>>     efforts were focussed on mod_sofia and pjsip got dropped. stkn
>>     _______________________________________________ FreeSWITCH-users
>>     mailing list FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>
>>
>>
>>     __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________
>>
>>     The message was checked by ESET NOD32 Antivirus.
>>
>>     http://www.eset.com
>>
>>
>>       
>
>
>
>
>     __________ Information from ESET NOD32 Antivirus, version of virus
>     signature database 4539 (20091024) __________
>
>     The message was checked by ESET NOD32 Antivirus.
>
>     http://www.eset.com
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>   




__________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From tayeb.meftah at gmail.com  Sat Oct 31 16:32:03 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sat, 31 Oct 2009 23:32:03 +0000
Subject: [Freeswitch-users] Mod_pjsip
In-Reply-To: <191c3a030910311517v46a9a830xa59350f6f8e2f025@mail.gmail.com>
References: <4AEC9C40.502@gmail.com>	<14A50F45-F0E7-4FEF-8C1C-66EA24CA64A7@freeswitch.org>	<4AECA67A.7030708@freeswitch.org>
	<4AECBDCC.9000105@gmail.com>	<191c3a030910311455j2408a0a2r6caa49375ed10b78@mail.gmail.com>	<4AECC305.5000805@gmail.com>
	<191c3a030910311517v46a9a830xa59350f6f8e2f025@mail.gmail.com>
Message-ID: <4AECC8F3.7090208@gmail.com>

Anthony Minessale a ?crit :
> Meftah,
> Feel free.

> thanks

> P.S.
>
> STKN was the guy who made the first mod_pjsip for FS that we abandoned 
> years ago. So you should believe him.
> Both him and I agreed it was not working out.  So if you don't believe 
> me, find out for yourself.
> anthony, why i don't believe  you?
never say that.
i believe you and all Freeswitch Staf and thank you and to all 
Freeswitch Staf.
>
>
>
> On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb <tayeb.meftah at gmail.com 
> <mailto:tayeb.meftah at gmail.com>> wrote:
>
>     hi Anthony
>     i agry
>     i say that because STKN hate all my suggestions.
>     about pjsip, i will contribute aditional module in the contrib.
>     thanks Anthony
>     Anthony Minessale a ?crit :
>>     Meftah,
>>
>>     He is 100% correct.  Please do not insult my volunteer
>>     developers. Without help from him you would not have any
>>     FreeSWITCH right now so please drop this subject we are not using
>>     pjsip.
>>
>>
>>
>>     On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb
>>     <tayeb.meftah at gmail.com <mailto:tayeb.meftah at gmail.com>> wrote:
>>
>>         hi,
>>         Pjsip support ICE, STUN and TURN!
>>         to STKN:
>>         if you don't pjsip, please stop talking or exit the discution
>>         we want to kype Freeswitch Clean and universal
>>         Stefan Knoblich a ?crit :
>>>         Michael S Collins wrote:
>>>           
>>>>         I can guarantee that the FS devs are well aware of pj-sip. If it was/ 
>>>>         is a viable alternative then it would be considered. The fact that it  
>>>>         isn't being used is a pretty good indication that it isn't suitable  
>>>>         for FS at this time.
>>>>
>>>>         -MV
>>>>
>>>>         Sent from my iPhone
>>>>             
>>>         We already mentioned some of the reasons why it did get
>>>         dropped 3 years ago (first two points from memory, last two
>>>         from old IRC logs): [License incompatible (GPL), but i think
>>>         tony tried to negotiate on alternate license terms] Not
>>>         possible to have multiple SIP profiles (due to global
>>>         variables being used in the lib). A race-condition under
>>>         high load, that couldn't be resolved back then (with the
>>>         help of the pjsip developers). And the sofia module "just
>>>         working" and surviving the scalability tests, so all efforts
>>>         were focussed on mod_sofia and pjsip got dropped. stkn
>>>         _______________________________________________
>>>         FreeSWITCH-users mailing list
>>>         FreeSWITCH-users at lists.freeswitch.org
>>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>         http://www.freeswitch.org
>>>
>>>
>>>
>>>         __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________
>>>
>>>         The message was checked by ESET NOD32 Antivirus.
>>>
>>>         http://www.eset.com
>>>
>>>
>>>           
>>
>>
>>
>>
>>         __________ Information from ESET NOD32 Antivirus, version of
>>         virus signature database 4539 (20091024) __________
>>
>>         The message was checked by ESET NOD32 Antivirus.
>>
>>         http://www.eset.com
>>
>>         _______________________________________________
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>
>>     -- 
>>     Anthony Minessale II
>>
>>     FreeSWITCH http://www.freeswitch.org/
>>     ClueCon http://www.cluecon.com/
>>     Twitter: http://twitter.com/FreeSWITCH_wire
>>
>>     AIM: anthm
>>     MSN:anthony_minessale at hotmail.com
>>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>
>>     FreeSWITCH Developer Conference
>>     sip:888 at conference.freeswitch.org
>>     <mailto:sip%3A888 at conference.freeswitch.org>
>>     iax:guest at conference.freeswitch.org/888
>>     <http://iax:guest at conference.freeswitch.org/888>
>>     googletalk:conf+888 at conference.freeswitch.org
>>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>     pstn:213-799-1400
>>      
>>     _______________________________________________ FreeSWITCH-users
>>     mailing list FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org __________ Information from ESET NOD32
>>     Antivirus, version of virus signature database 4539 (20091024)
>>     __________ The message was checked by ESET NOD32 Antivirus.
>>     http://www.eset.com
>
>
>
>     __________ Information from ESET NOD32 Antivirus, version of virus
>     signature database 4539 (20091024) __________
>
>     The message was checked by ESET NOD32 Antivirus.
>
>     http://www.eset.com
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>   




__________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From dome at tel.co.th  Sat Oct 31 19:02:07 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Sun, 1 Nov 2009 09:02:07 +0700
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <b39273530910311117k3b4bcfa5s98a5b720b1775e20@mail.gmail.com>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
	<140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
	<AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
	<8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com>
	<EEDE7B6D-F722-4557-BBFE-AC0031C0BA86@freeswitch.org>
	<ebbc2d550910311018m7bc9b86bi120649c36dbf3413@mail.gmail.com>
	<8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com>
	<b39273530910311117k3b4bcfa5s98a5b720b1775e20@mail.gmail.com>
Message-ID: <8ccbff060910311902u7434bc6m76caba40178ecc1e@mail.gmail.com>

Yes. i user api_hangup_hook for do callback.
<action application="set" data="api_hangup_hook=originate
{origination_caller_id_number=${caller_id_number},callback_uuid=${uuid},ignore_early_media=true}${route_1}|${route_2}|${route_3}
call_ivr${caller_id_number} XML public "/>

may be need originate_timeout

Dome C.


2009/11/1 Rupa Schomaker <rupa at rupa.com>:
> fscore_pb has been updated.
>
> Next time put in the pastebin url. ?This one was 10911.
>
> Ok, a suspicious thread looks below. ?If I read it right, you are
> doing an originate from within the hangup handler. ?That doesn't look
> normal or sane to me.
>
> Thread 41 (process 30732):
> #0 ?0xb7a5e007 in select () from /lib/libc.so.6
> #1 ?0xb7dde39f in apr_sleep (t=1000) at time/unix/time.c:246
> #2 ?0xb7db12c0 in do_sleep (t=1000) at src/switch_time.c:109
> #3 ?0xb7db15ac in switch_cond_next () at src/switch_time.c:196
> #4 ?0xb7d862b8 in switch_ivr_originate (session=0x0, bleg=0xb48c2f00,
> cause=0xb48c2ed4, bridgeto=0xb1133340
> "{origination_caller_id_number=0857151465,callback_uuid=c32f54cc-c6b8-11de-af77-774e15238dbe,ignore_early_media=true}sofia/external/66857151465 at 115.178.62.191:7050|sofia/external/66857151465 at 115.178.62"...,
> timelimit_sec=60, table=0x0, cid_name_override=0xb7e8c976
> "FreeSWITCH", cid_num_override=0x90ec0800 "0857151465",
> caller_profile_override=0x0, ovars=0x0, flags=0) at
> src/switch_ivr_originate.c:2132
> #5 ?0xb6f0def1 in originate_function (cmd=0x90164ce2
> "{origination_caller_id_number=0857151465,callback_uuid=c32f54cc-c6b8-11de-af77-774e15238dbe,ignore_early_media=true}sofia/external/66857151465 at 115.178.62.191:7050|sofia/external/66857151465 at 115.178.62"...,
> session=0x0, stream=0xb48c3004) at mod_commands.c:2413
> #6 ?0xb7d66fb1 in switch_api_execute (cmd=0x90164cd8 "originate",
> arg=0x90164ce2 "{origination_caller_id_number=0857151465,callback_uuid=c32f54cc-c6b8-11de-af77-774e15238dbe,ignore_early_media=true}sofia/external/66857151465 at 115.178.62.191:7050|sofia/external/66857151465 at 115.178.62"...,
> session=0x0, stream=0xb48c3004) at src/switch_loadable_module.c:1567
> #7 ?0xb7d53c4a in switch_core_session_hangup_state
> (session=0x958ab678) at src/switch_core_state_machine.c:506
> #8 ?0xb7d385e6 in switch_channel_perform_hangup (channel=0x958afd70,
> file=0xb6af8788 "mod_dptools.c", func=0xb6af8ba8 "hangup_function",
> line=637, hangup_cause=SWITCH_CAUSE_NORMAL_CLEARING) at
> src/switch_channel.c:1898
> #9 ?0xb6aee906 in hangup_function (session=0x958ab678, data=0xb7e82ea5
> "") at mod_dptools.c:637
> #10 0xb7d4fe1f in switch_core_session_exec (session=0x958ab678,
> application_interface=0xb6e10ec0, arg=0xb7e82ea5 "") at
> src/switch_core_session.c:1552
> #11 0xb7d4f73e in switch_core_session_execute_application
> (session=0x958ab678, app=0x90163a80 "hangup", arg=0xb7e82ea5 "") at
> src/switch_core_session.c:1474
> #12 0xb7d51131 in switch_core_standard_on_execute (session=0x958ab678)
> at src/switch_core_state_machine.c:166
> #13 0xb7d5215a in switch_core_session_run (session=0x958ab678) at
> src/switch_core_state_machine.c:340
> #14 0xb7d4e199 in switch_core_session_thread (thread=0x9020bd98,
> obj=0x958ab678) at src/switch_core_session.c:1135
> #15 0xb7ddc2fc in dummy_worker (opaque=0x9020bd98) at
> threadproc/unix/thread.c:138
> #16 0xb7caaf3b in start_thread () from /lib/libpthread.so.0
> #17 0xb7a64bee in clone () from /lib/libc.so.6
>
>
> On Sat, Oct 31, 2009 at 12:34 PM, Dome Charoenyost <dome at tel.co.th> wrote:
>> I found bug in fscore_pb ? $pwd should be $mypwd
>>
>> Now i post already. please help me check.
>>
>> Dome C.
>>
>>
>> 2009/11/1 Jo?o Mesquita <jmesquita at freeswitch.org>:
>>> No, mod_nibblebill definetely needs to be enhanced but it is not the problem
>>> and it can be used with high load traffic.
>>>
>>> The one I am not sure about is odbc_query since it was not developed for
>>> that.
>>>
>>> Do what Rupa said, please.
>>>
>>> Regards,
>>>
>>> JM
>>>
>>> On Sat, Oct 31, 2009 at 3:11 PM, Brian West <brian at freeswitch.org> wrote:
>>>>
>>>> I think once you get the backtrace like rupa said we can see that maybe
>>>> odbc_query is really hanging or something similar.
>>>> /b
>>>> On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:
>>>>
>>>> 2009/10/31 Brian West <brian at freeswitch.org>:
>>>>
>>>> You should never do billing inline with the session thread is all I'm
>>>>
>>>> saying.
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From anthony.minessale at gmail.com  Sat Oct 31 19:23:23 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 31 Oct 2009 21:23:23 -0500
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <8ccbff060910311902u7434bc6m76caba40178ecc1e@mail.gmail.com>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
	<AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
	<8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com>
	<EEDE7B6D-F722-4557-BBFE-AC0031C0BA86@freeswitch.org>
	<ebbc2d550910311018m7bc9b86bi120649c36dbf3413@mail.gmail.com>
	<8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com>
	<b39273530910311117k3b4bcfa5s98a5b720b1775e20@mail.gmail.com>
	<8ccbff060910311902u7434bc6m76caba40178ecc1e@mail.gmail.com>
Message-ID: <191c3a030910311923t5f81019fyc4e7487cd61502b8@mail.gmail.com>

Use bgapi originate ...

On Oct 31, 2009 9:09 PM, "Dome Charoenyost" <dome at tel.co.th> wrote:

Yes. i user api_hangup_hook for do callback.
<action application="set" data="api_hangup_hook=originate
{origination_caller_id_number=${caller_id_number},callback_uuid=${uuid},ignore_early_media=true}${route_1}|${route_2}|${route_3}
call_ivr${caller_id_number} XML public "/>

may be need originate_timeout

Dome C.


2009/11/1 Rupa Schomaker <rupa at rupa.com>:

> fscore_pb has been updated. > > Next time put in the pastebin url.  This
one was 10911. > > Ok, a ...
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From chris at cloudtel.com  Sat Oct 31 19:24:47 2009
From: chris at cloudtel.com (Chris Burns)
Date: Sat, 31 Oct 2009 22:24:47 -0400
Subject: [Freeswitch-users] Mod_pjsip
In-Reply-To: <4AECC8F3.7090208@gmail.com>
References: <4AEC9C40.502@gmail.com>
	<191c3a030910311517v46a9a830xa59350f6f8e2f025@mail.gmail.com>
	<4AECC8F3.7090208@gmail.com>
Message-ID: <200910312224.47236.chris@cloudtel.com>

My favorite part of this 'civilized' discussion on IRC was when DelphiWord and 
diegoviola sat around tryin to take the piss outta stkn on this issue for 
seemingly no reason. Thanks for making the channel a cool place, guys ;)

On October 31, 2009 07:32:03 pm Meftah Tayeb wrote:
> Anthony Minessale a ?crit :
> > Meftah,
> > Feel free.
> >
> > thanks
> >
> > P.S.
> >
> > STKN was the guy who made the first mod_pjsip for FS that we abandoned
> > years ago. So you should believe him.
> > Both him and I agreed it was not working out.  So if you don't believe
> > me, find out for yourself.
> > anthony, why i don't believe  you?
>
> never say that.
> i believe you and all Freeswitch Staf and thank you and to all
> Freeswitch Staf.
>
> > On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb <tayeb.meftah at gmail.com
> > <mailto:tayeb.meftah at gmail.com>> wrote:
> >
> >     hi Anthony
> >     i agry
> >     i say that because STKN hate all my suggestions.
> >     about pjsip, i will contribute aditional module in the contrib.
> >     thanks Anthony
> >
> >     Anthony Minessale a ?crit :
> >>     Meftah,
> >>
> >>     He is 100% correct.  Please do not insult my volunteer
> >>     developers. Without help from him you would not have any
> >>     FreeSWITCH right now so please drop this subject we are not using
> >>     pjsip.
> >>
> >>
> >>
> >>     On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb
> >>     <tayeb.meftah at gmail.com <mailto:tayeb.meftah at gmail.com>> wrote:
> >>
> >>         hi,
> >>         Pjsip support ICE, STUN and TURN!
> >>         to STKN:
> >>         if you don't pjsip, please stop talking or exit the discution
> >>         we want to kype Freeswitch Clean and universal
> >>
> >>         Stefan Knoblich a ?crit :
> >>>         Michael S Collins wrote:
> >>>>         I can guarantee that the FS devs are well aware of pj-sip. If
> >>>> it was/ is a viable alternative then it would be considered. The fact
> >>>> that it isn't being used is a pretty good indication that it isn't
> >>>> suitable for FS at this time.
> >>>>
> >>>>         -MV
> >>>>
> >>>>         Sent from my iPhone
> >>>
> >>>         We already mentioned some of the reasons why it did get
> >>>         dropped 3 years ago (first two points from memory, last two
> >>>         from old IRC logs): [License incompatible (GPL), but i think
> >>>         tony tried to negotiate on alternate license terms] Not
> >>>         possible to have multiple SIP profiles (due to global
> >>>         variables being used in the lib). A race-condition under
> >>>         high load, that couldn't be resolved back then (with the
> >>>         help of the pjsip developers). And the sofia module "just
> >>>         working" and surviving the scalability tests, so all efforts
> >>>         were focussed on mod_sofia and pjsip got dropped. stkn
> >>>         _______________________________________________
> >>>         FreeSWITCH-users mailing list
> >>>         FreeSWITCH-users at lists.freeswitch.org
> >>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>        
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> >>>s http://www.freeswitch.org
> >>>
> >>>
> >>>
> >>>         __________ Information from ESET NOD32 Antivirus, version of
> >>> virus signature database 4539 (20091024) __________
> >>>
> >>>         The message was checked by ESET NOD32 Antivirus.
> >>>
> >>>         http://www.eset.com
> >>
> >>         __________ Information from ESET NOD32 Antivirus, version of
> >>         virus signature database 4539 (20091024) __________
> >>
> >>         The message was checked by ESET NOD32 Antivirus.
> >>
> >>         http://www.eset.com
> >>
> >>         _______________________________________________
> >>         FreeSWITCH-users mailing list
> >>         FreeSWITCH-users at lists.freeswitch.org
> >>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>        
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> >>     --
> >>     Anthony Minessale II
> >>
> >>     FreeSWITCH http://www.freeswitch.org/
> >>     ClueCon http://www.cluecon.com/
> >>     Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >>     AIM: anthm
> >>     MSN:anthony_minessale at hotmail.com
> >>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
> >>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> >>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >>
> >>     FreeSWITCH Developer Conference
> >>     sip:888 at conference.freeswitch.org
> >>     <mailto:sip%3A888 at conference.freeswitch.org>
> >>     iax:guest at conference.freeswitch.org/888
> >>     <http://iax:guest at conference.freeswitch.org/888>
> >>     googletalk:conf+888 at conference.freeswitch.org
> >>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>     pstn:213-799-1400
> >>
> >>     _______________________________________________ FreeSWITCH-users
> >>     mailing list FreeSWITCH-users at lists.freeswitch.org
> >>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>    
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org __________ Information from ESET NOD32
> >> Antivirus, version of virus signature database 4539 (20091024)
> >> __________ The message was checked by ESET NOD32 Antivirus.
> >> http://www.eset.com
> >
> >     __________ Information from ESET NOD32 Antivirus, version of virus
> >     signature database 4539 (20091024) __________
> >
> >     The message was checked by ESET NOD32 Antivirus.
> >
> >     http://www.eset.com
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >    
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com
> > <mailto:MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> > <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org
> > <mailto:sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > <http://iax:guest at conference.freeswitch.org/888>
> > googletalk:conf+888 at conference.freeswitch.org
> > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:213-799-1400
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > __________ Information from ESET NOD32 Antivirus, version of virus
> > signature database 4539 (20091024) __________
> >
> > The message was checked by ESET NOD32 Antivirus.
> >
> > http://www.eset.com
>
> __________ Information from ESET NOD32 Antivirus, version of virus
> signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com




From dome at tel.co.th  Sat Oct 31 19:45:25 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Sun, 1 Nov 2009 09:45:25 +0700
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <191c3a030910311923t5f81019fyc4e7487cd61502b8@mail.gmail.com>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
	<AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
	<8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com>
	<EEDE7B6D-F722-4557-BBFE-AC0031C0BA86@freeswitch.org>
	<ebbc2d550910311018m7bc9b86bi120649c36dbf3413@mail.gmail.com>
	<8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com>
	<b39273530910311117k3b4bcfa5s98a5b720b1775e20@mail.gmail.com>
	<8ccbff060910311902u7434bc6m76caba40178ecc1e@mail.gmail.com>
	<191c3a030910311923t5f81019fyc4e7487cd61502b8@mail.gmail.com>
Message-ID: <8ccbff060910311945j5be2fa85j95e7e1b422dcc3a9@mail.gmail.com>

How to use bgapi in my flow.
1. user call did
2. FS send ringing and check balance, LCR from DB (by mod_odbc_quey)
3. Hangup (by use or timeout)
4. FS callback to user and bridge to IVR

I'm not sure  bgapi can do after channel hangup.

Dome C.


2009/11/1 Anthony Minessale <anthony.minessale at gmail.com>:
> Use bgapi originate ...
>
> On Oct 31, 2009 9:09 PM, "Dome Charoenyost" <dome at tel.co.th> wrote:
>
> Yes. i user api_hangup_hook for do callback.
> <action application="set" data="api_hangup_hook=originate
> {origination_caller_id_number=${caller_id_number},callback_uuid=${uuid},ignore_early_media=true}${route_1}|${route_2}|${route_3}
> call_ivr${caller_id_number} XML public "/>
>
> may be need originate_timeout
>
> Dome C.
>
>
> 2009/11/1 Rupa Schomaker <rupa at rupa.com>:
>
>> fscore_pb has been updated. > > Next time put in the pastebin url. ?This
>> one was 10911. > > Ok, a ...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From mike at jerris.com  Sat Oct 31 19:57:32 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 31 Oct 2009 22:57:32 -0400
Subject: [Freeswitch-users] Mod_pjsip
In-Reply-To: <4AEC9C40.502@gmail.com>
References: <4AEC9C40.502@gmail.com>
Message-ID: <BC20832A-CC71-471E-9E2A-62742334F3F2@jerris.com>

As the other reasons have already been covered about the history of  
mod_pjsip, I'll just make a few comments here:

If someone is really interested in resurecting mod_pjsip, the code is  
around here somewhere (although I can't find it on my quick look).   
Just let us know.

The licensing is not really an issue, that was resolved long ago.

There is no major thread safety issue in sofia.  There are some minor  
issues which we easily address in the module and an interest in thread  
pooling, which would increase performance.

pjsip and sofia always performed quite similar to each other, they are  
both quite good sip libraries

We had extensive thread safety and race condition issues in pjsip that  
never got resolved while we were using it.  It is quite possible they  
have been fixed sense, but many months were spent trying to fix them  
to no success.

We already support ice in the freeswitch rtp stack.  A sip stack  
supporting ice and turn does little for us and would be the worst way  
to implement it in freeswitch.  Sofia already supports sdp parsing for  
all of this, if you are interested in ICE/TURN support in freeswitch,  
it needs to be done in the RTP stack mostly, regardless of which sip  
stack.

Sofia has been shown to run well already in embedded environments.

Mike

On Oct 31, 2009, at 4:21 PM, Meftah Tayeb wrote:

> 		Hi my FreeSWITCH friends
> Today we talked about FreeSWITCH libraries and we found pjsip, which  
> is
> a lot better than sofia, it can do everything sofia can do and more,
> it's thread-safe and performs well, I would love to see a mod_pjsip.
> The library is GPL but they also offer alternative licensing, don't  
> get
> me wrong, Sofia is great but I think FreeSWITCH will benefit from a
> thread-safe SIP library.
> don't say that i need some stuf for free, but we need to make fs a  
> very
> also we love freeswitch to run in a Embedded PC like the blackfin
> CPU/boards, Pjsip is very good for Embedded SIP Devicesawesome Open
> Source project
> With much love,



From gromovd at gmail.com  Sat Oct 31 22:02:18 2009
From: gromovd at gmail.com (Dmitry Gromov)
Date: Sun, 1 Nov 2009 01:02:18 -0400
Subject: [Freeswitch-users] SIP Proxy with direct media path
Message-ID: <e21d97a50910312202s5082b8cch5f809da20fddb0ab@mail.gmail.com>

Hi!

I was a long time user of MySipSwitch service (www.mysipswitch.com) but it
looks like they are planning to close their service. I can't use the
replacement they have and would actually prefer some locally installed
solution.

Question - is it possible to use FreeSwitch exactly like MySipSwitch - just
SIP Proxy with very flexible dial plan?
With MySipSwitch it is possible to:
- register service with multiple SIP providers as client
- register your ATA with MySipSwitch
- create smart dial plan (in Ruby) to route calls based on prefixes and/or
call source and some other logic
- accept and route calls from all registered providers

MySipSwitch does not have any media capabilities - RTP is always established
directly between call parties (no re-Invites either).

If this is possible - is there any beginner guide for this?

Also - I know that FreeSwitch could run in Windows as well as in Linux -
would it be possible to use the above functionality in both OSes?

Thanks,
Dmitry

-- 
DG
NJ
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From mike at jerris.com  Sat Oct 31 22:15:26 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 1 Nov 2009 01:15:26 -0400
Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.5pre3 is
	available for testing!
In-Reply-To: <8cc991dd0910281740w2c7a9d10q93dd5b5f0b4dbd6@mail.gmail.com>
References: <87f2f3b90910281000r4c2021b8u7232320614737e30@mail.gmail.com>
	<8cc991dd0910281740w2c7a9d10q93dd5b5f0b4dbd6@mail.gmail.com>
Message-ID: <91BCC471-AA4C-4E5A-973C-8380E54DAB51@jerris.com>

We still do plan on branching 1.0 into bugfix only.  This has not yet  
happened but may happen at some point after 1.0.5.  In the mean time,  
the vast majority of the work lately has been fixes with small feature  
improvements, most all of this would stay in a 1.0 branch even if we  
were already branched.  Trunk remains for the most part very stable  
with new features mostly coming in new modules.

Mike

On Oct 28, 2009, at 8:40 PM, Craig Askings wrote:

> Is the plan to run 1.0.5 as a stable branch with bug fix updates or  
> will it be the case of just follow the trunk like it is with any  
> problems you encounter in 1.0.4?
>
> 2009/10/29 Michael Collins <msc at freeswitch.org>
> Hello FreeSWITCHers,
>
> The latest FreeSWITCH version is now available for download on the  
> files site. The announcement story is in the main FreeSWITCH site.  
> Please download and test, and then test some more. We need your  
> feedback. The sooner we get your feedback, the more quickly we can  
> roll the official 1.0.5.
>
> -- 
> Craig Askings
>
> Network Engineer | Over the Wire Pty Ltd
> craig at overthewire.com.au | www.overthewire.com.au
> Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From jason at jasonjgw.net  Sat Oct 31 22:21:19 2009
From: jason at jasonjgw.net (Jason White)
Date: Sun, 1 Nov 2009 16:21:19 +1100
Subject: [Freeswitch-users] SIP Proxy with direct media path
In-Reply-To: <e21d97a50910312202s5082b8cch5f809da20fddb0ab@mail.gmail.com>
References: <e21d97a50910312202s5082b8cch5f809da20fddb0ab@mail.gmail.com>
Message-ID: <20091101052119.GA28137@jdc.jasonjgw.net>

> With MySipSwitch it is possible to:
> - register service with multiple SIP providers as client

This is easy once you understand the FreeSWITCH XML configuration.
> - register your ATA with MySipSwitch

That's really a special case of the above.
> - create smart dial plan (in Ruby) to route calls based on prefixes and/or
> call source and some other logic

FreeSWITCH can do better than that: there are lots of variables which can be
matched by Perl-compatible regular expressions, including the destination
number.
> - accept and route calls from all registered providers

That's handled in the same way as the previous item.
> 
> MySipSwitch does not have any media capabilities - RTP is always established
> directly between call parties (no re-Invites either).

FreeSWITCH gives you a choice: you can use proxy media or bypass media modes
if you don't want it to handle the media.
> 
> If this is possible - is there any beginner guide for this?

http://wiki.freeswitch.org/ is the best there is.

It is being improved by the community over time.

You can also take advantage of the IRC channel, the FreeSWITCH conference and
of course the mailing list.



From msc at freeswitch.org  Sat Oct 31 22:24:59 2009
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 31 Oct 2009 22:24:59 -0700
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179
In-Reply-To: <509453DC-57BF-43D0-B704-FF0F2BB58EC0@jerris.com>
References: <mailman.1516.1256123625.2495.freeswitch-users@lists.freeswitch.org>
	<a60fed5e0910212108v4755203y693b8bdec5501f5f@mail.gmail.com>
	<87f2f3b90910212141w72b8f5a2oce032adb974d71e2@mail.gmail.com>
	<b39273530910220551q14d22150ldf8d4bff0b6fd011@mail.gmail.com>
	<87f2f3b90910221054l8abc3far9a18a66f0959f950@mail.gmail.com>
	<509453DC-57BF-43D0-B704-FF0F2BB58EC0@jerris.com>
Message-ID: <87f2f3b90910312224s6968f278i86e35469c781d60c@mail.gmail.com>

How would you do an expression like: if $x < 24 in a condition tag? Just
curious. I would like to make sure that is properly documented.
-MC

On Sat, Oct 31, 2009 at 10:42 AM, Michael Jerris <mike at jerris.com> wrote:

> You actually can use these in conditions.  Just need to be careful that the
> var you are conditioning on is already set.
>
> Mike
>
> On Oct 22, 2009, at 1:54 PM, Michael Collins wrote:
>
>
>
> On Thu, Oct 22, 2009 at 5:51 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>
>> cond would be helpful here?  I updated the wiki on this one just now
>> with a bit more detail.  It is a api call. so, you'd use it like:
>>
>> ${cond(eval ? trueval : falseval)}
>>
>> so to get a value of ERR if the var my myvar is > 15 you could:
>>
>> ${cond(${myvar} > 15 ? ERR : OK)}
>>
>> If both sides of the comparison operator are numeric then it does
>> numeric comparison otherwise it does lexical string comparison.
>>
>
> Rupa,
>
> Yes, you can do the set/cond API trick but you can only do it in the action
> or anti-action tags, not in the condition tags. I'm sure you know that but I
> want all those reading this thread to make the connection.
> -MC
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Sat Oct 31 22:33:18 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 1 Nov 2009 01:33:18 -0400
Subject: [Freeswitch-users] Setting up Conference with Moderator
In-Reply-To: <D7D48F51-0917-4D9A-B8B4-DE8F552BB254@gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local>
	<3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local>
	<E97CD3D0-52FF-44D1-BB89-23816045CE21@gmail.com>
	<71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>,
	<114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local>
	<3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local>
	<F88E7B78-4CB8-45BD-98CE-81195862A891@gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local>
	<D7D48F51-0917-4D9A-B8B4-DE8F552BB254@gmail.com>
Message-ID: <89D54263-7234-4F9A-8E22-40139F103DD3@jerris.com>

Have you answered the call?

On Oct 30, 2009, at 11:34 AM, Rob Forman wrote:

> Hm, strange.  I haven't seen that before.  Can you pastebin your logs
> at debug level?
>
> On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote:
>
>> It's strange... a tcpdump tells me that there is no DTMF from my
>> provider when using IVR, but when I call into a TN that goes
>> directly into the Conference App, I see DTMF from the provider.
>>



From mike at jerris.com  Sat Oct 31 22:34:38 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 1 Nov 2009 01:34:38 -0400
Subject: [Freeswitch-users] mod_t38gateway
In-Reply-To: <65d96fc80910301452x2a831733h92b05861d6c94123@mail.gmail.com>
References: <65d96fc80910301452x2a831733h92b05861d6c94123@mail.gmail.com>
Message-ID: <ECDA870C-7549-450A-A338-3BFFF600E592@jerris.com>

This is a non working module, just a shell for development.

Mike

On Oct 30, 2009, at 5:52 PM, Tihomir Culjaga wrote:

> does anybody know how does it work and how to use it in a dialplan?
>



From mike at jerris.com  Sat Oct 31 22:38:10 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 1 Nov 2009 01:38:10 -0400
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179
In-Reply-To: <87f2f3b90910312224s6968f278i86e35469c781d60c@mail.gmail.com>
References: <mailman.1516.1256123625.2495.freeswitch-users@lists.freeswitch.org>
	<a60fed5e0910212108v4755203y693b8bdec5501f5f@mail.gmail.com>
	<87f2f3b90910212141w72b8f5a2oce032adb974d71e2@mail.gmail.com>
	<b39273530910220551q14d22150ldf8d4bff0b6fd011@mail.gmail.com>
	<87f2f3b90910221054l8abc3far9a18a66f0959f950@mail.gmail.com>
	<509453DC-57BF-43D0-B704-FF0F2BB58EC0@jerris.com>
	<87f2f3b90910312224s6968f278i86e35469c781d60c@mail.gmail.com>
Message-ID: <AE75D709-C68C-458D-9A18-C6AB0E14C5F0@jerris.com>

see rupa's explanation below.


On Nov 1, 2009, at 1:24 AM, Michael Collins wrote:

>> How would you do an expression like: if $x < 24 in a condition tag?  
>> Just curious. I would like to make sure that is properly documented.
>> -MC
>>
>>
>>>
>>>
>>> On Thu, Oct 22, 2009 at 5:51 AM, Rupa Schomaker <rupa at rupa.com>  
>>> wrote:
>>> ${cond(${myvar} > 15 ? ERR : OK)}
>>>
>>> If both sides of the comparison operator are numeric then it does
>>> numeric comparison otherwise it does lexical string comparison.
>>

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From helmut.kuper at ewetel.de  Thu Oct  1 01:52:59 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Thu, 01 Oct 2009 10:52:59 +0200
Subject: [Freeswitch-users] Problem with subscription expire
Message-ID: <4AC46DEB.3090506@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

it seems exired subsciptions are never cleared in FS.

A look into sofia_presence.c confirms explains this

/* negative in exptime means keep bumping up sub time to avoid a snafu
where every device has it's own rules about subscriptions
                           that somehow barely resemble the RFC not that
I blame them because the RFC MAY be amibiguous and SHOULD be deleted.
                           So to avoid the problem we keep resetting the
expiration date of the subscription so it never expires.

                           Eybeam completely ignores this option and
most other subscription-state: directives from rfc3265 and still expires.
                           Polycom is happy to keep upping the
subscription expiry back to the original time on each new notify.
                           The rest ... who knows...?

                        */

For some reasons subscriptions created by Snom phones are filling up the
sip_subscriptions table over time. This leads to some kind of DOS by FS
against the subscribing phone ... The subscribtions are differentiate by
call-id. This can be explained by RFC 3842 chapter 3.6 where expired
subscriptions must be renewed with a NEW call-id. Because there is no
hint about unsubscribing the old subscription I guess the clean up
process has to be done by FS.

Any way to get FS to do this job? Since there is no creation date or
expire value which represents the expire as a timestamp I have no way to
clean up the table manually via sql and cronjob - except cleaning the
whole table ...


A further (but background) question is, why do the subscriptions expire
in snom phones at all ...



regards
helmut
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pO+XUioXrBKJ/ozUOy1ZqeA=
=nZaf
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From nagalenoj at gmail.com  Thu Oct  1 06:50:00 2009
From: nagalenoj at gmail.com (Nagalenoj)
Date: Thu, 1 Oct 2009 06:50:00 -0700 (PDT)
Subject: [Freeswitch-users]  Listening to a connected call [barge in]
Message-ID: <25696889.post@talk.nabble.com>


In ES outbound, I need to do the following, 
* A calls 2000(FS ES outbound extension)
* In the script, It'll answer the call, play some files and get the reply
from A(as voice).
* Simultaneously(when doing the above), the script has to call B.
* When B attends the call, B has to listen to the live conversation between
2000 & A.

How should I do.?

I've tried this with async mode and by listening to the events. But I
couldn't do it. Help me to do this..

Regards,
Nagalenoj
-- 
View this message in context: http://www.nabble.com/Listening-to-a-connected-call--barge-in--tp25696889p25696889.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From vinuth.madinur at gmail.com  Thu Oct  1 07:03:19 2009
From: vinuth.madinur at gmail.com (Vinuth Madinur)
Date: Thu, 1 Oct 2009 19:33:19 +0530
Subject: [Freeswitch-users] Listening to a connected call [barge in]
In-Reply-To: <25696889.post@talk.nabble.com>
References: <25696889.post@talk.nabble.com>
Message-ID: <910309030910010703j151d71v3c9e5700a6355939@mail.gmail.com>

Hi,
Use the eavesdrop command.
Just supply it with the call UUID and the extension of B.
Wiki has more details.

Thanks,
Vinuth.


On Thu, Oct 1, 2009 at 7:20 PM, Nagalenoj <nagalenoj at gmail.com> wrote:

>
> In ES outbound, I need to do the following,
> * A calls 2000(FS ES outbound extension)
> * In the script, It'll answer the call, play some files and get the reply
> from A(as voice).
> * Simultaneously(when doing the above), the script has to call B.
> * When B attends the call, B has to listen to the live conversation between
> 2000 & A.
>
> How should I do.?
>
> I've tried this with async mode and by listening to the events. But I
> couldn't do it. Help me to do this..
>
> Regards,
> Nagalenoj
> --
> View this message in context:
> http://www.nabble.com/Listening-to-a-connected-call--barge-in--tp25696889p25696889.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sicfslist at gmail.com  Thu Oct  1 07:27:47 2009
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Thu, 1 Oct 2009 09:27:47 -0500
Subject: [Freeswitch-users] Dialplan Issue
Message-ID: <35b355e90910010727v2efcbff6p97a0a883857818e4@mail.gmail.com>

Hello:

I asked this on IRC yesterday and I think I confused everyone involved.  So
I apologize in advance here for reposting the question and if I wasted
anyone's time.

So here is the issue I'm having.  I'm trying to use FS as a redirect server
(specifically to serve up LNP queries via 302 redirects).  But I'm having an
issue where based on the string in the dialplan FS will respond with a 500
internal error message instead of a 300 redirect.

The call flow should be this:
   -- remote party sends an Invite to my FS instance
   -- FS should respond with a 302

The following works as expected (FS will send a 302 when it receives an
Invite):

<action application="redirect" data="sip:${destination_number}@
${network_addr};rn=${rn};npdi=yes"/>

However if I do this (which is the way the response should look) FS will
respond with a 500 internal server error:

<action application="redirect"
data="sip:${destination_number};rn=${rn};npdi=yes@${network_addr}"/>

So the issue is the placement of the user params .... if they are before the
@ FS will send a 500 internal server error ... if they are after the @ FS
will send a 302.  Unfortunately placing the user params after the @ doesn't
quite conform to the way other devices expect to receive the 302 for this
application.

Any help would be greatly appreciated.

Shelby

PS ... hats off to the author of mod_memcache ... that is extremely useful!
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From testa at voicetechnology.com.br  Thu Oct  1 07:35:24 2009
From: testa at voicetechnology.com.br (Fernando Testa)
Date: Thu, 1 Oct 2009 11:35:24 -0300
Subject: [Freeswitch-users] REGISTER fails with 407 after minutes of
	success register
In-Reply-To: <7B7B543E-ACC9-4D00-A246-6C8D2608B960@freeswitch.org>
References: <9cb0e15e0909291728y2d75cea0k96547ea727d3dff9@mail.gmail.com>
	<9F9D51D2-2EBE-441D-BABF-6CA1DE8A9372@freeswitch.org>
	<9cb0e15e0909300506g6306d86dg39ade3fafa3ae750@mail.gmail.com>
	<7B7B543E-ACC9-4D00-A246-6C8D2608B960@freeswitch.org>
Message-ID: <9cb0e15e0910010735q3364cca8m8333740b1fd1a31@mail.gmail.com>

In the link below you have the entire SIP trace from system startup until
start receiving this annoying 407 Proxy Auth Required, preventing FS to
register successfully on the Ericsson Pabx.You can notice multiple
registrations from named ericsson_1050 to ericsson_1064 that starts failing
after ~50 minutes after the boot. Issuing a 'sofia external profile restart'
solves the registration problems.
Brian, thanks for reply, but I really didn't get your point.
Thank you, I apreciate any help.

http://dl.getdropbox.com/u/410277/sip.log.gz

On Wed, Sep 30, 2009 at 10:33 AM, Brian West <brian at freeswitch.org> wrote:

> I don't see a challenge in your 407 so how can we answer properly
> against the far end if they don't challenge us?
>
> /b
>
> On Sep 30, 2009, at 7:06 AM, Fernando Testa wrote:
>
> > Brian,
> >
> > Thanks for the reply. The SIP trace is mixed with the log (+ sofia
> > loglevel all 9) on the pastebin I mention on the previous email
> > (http://pastebin.freeswitch.org/10517 ). That log is from FS 1.0.4.
> >
> >
> > On Tue, Sep 29, 2009 at 10:13 PM, Brian West <brian at freeswitch.org>
> > wrote:
> >> I need the sip trace.
> >>
> >> /b
> >>
> >> On Sep 29, 2009, at 7:28 PM, Fernando Testa wrote:
> >>
> >>> Hi all,
> >>>
> >>> I have a FS that registers on an Ericsson pabx as gateway under
> >>> sip_external.
> >>> This gateway start registering on the Ericsson ok, but after a
> >>> while,
> >>> around 50mins, it fails with the logs below.
> >>> If I hit *sofia profile external restart* on fs_cli then the gateway
> >>> returns to register with success (that means, we get 200 OK from
> >>> Ericsson).
> >>> This happens with FS 1.0.4 release tarball, and trunk r15011.
> >>> I found similar situations on these links, but not actually found a
> >>> solution.
> >>> Any help is very welcome.
> >>>
> >>> *OS*
> >>> CentOS 5.3 x86_64
> >>> 4Gb RAM
> >>>
> >>> log at http://pastebin.freeswitch.org/10517
> >>>
> >>> conf/sip_profiles/external/ericsson.xml:
> >>>  <gateway name="ericsson_1064">
> >>>    <param name="username" value="1064"/>
> >>>    <param name="realm" value="10.227.0.3"/>
> >>>    <param name="password" value="5080"/>
> >>>    <param name="register" value="true"/>
> >>>    <param name="register-proxy" value="10.227.0.3"/>
> >>>    <param name="expire-seconds" value="60"/>
> >>>    <param name="retry-seconds" value="10"/>
> >>>  </gateway>
> >>>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Fernando Gregianin Testa
Voice Technology Ltda
+55 11 35882166
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From brian at freeswitch.org  Thu Oct  1 07:42:23 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 1 Oct 2009 09:42:23 -0500
Subject: [Freeswitch-users] Dialplan Issue
In-Reply-To: <35b355e90910010727v2efcbff6p97a0a883857818e4@mail.gmail.com>
References: <35b355e90910010727v2efcbff6p97a0a883857818e4@mail.gmail.com>
Message-ID: <3DA34BA8-5E67-4A0C-A163-B72EF5F8638F@freeswitch.org>


On Oct 1, 2009, at 9:27 AM, Shelby Ramsey wrote:

> <action application="redirect" data="sip:${destination_number};rn=${rn};npdi=yes@${network_addr} 
> "/>

This will produce an INVALID sip uri... You can not feed this to sofia  
it'll get PISSED.

Its missing the host portion.

>
> So the issue is the placement of the user params .... if they are  
> before the @ FS will send a 500 internal server error ... if they  
> are after the @ FS will send a 302.  Unfortunately placing the user  
> params after the @ doesn't quite conform to the way other devices  
> expect to receive the 302 for this application.
>
> Any help would be greatly appreciated.
>
> Shelby
>
> PS ... hats off to the author of mod_memcache ... that is extremely  
> useful!

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From brian at freeswitch.org  Thu Oct  1 07:46:53 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 1 Oct 2009 09:46:53 -0500
Subject: [Freeswitch-users] REGISTER fails with 407 after minutes of
	success register
In-Reply-To: <9cb0e15e0910010735q3364cca8m8333740b1fd1a31@mail.gmail.com>
References: <9cb0e15e0909291728y2d75cea0k96547ea727d3dff9@mail.gmail.com>
	<9F9D51D2-2EBE-441D-BABF-6CA1DE8A9372@freeswitch.org>
	<9cb0e15e0909300506g6306d86dg39ade3fafa3ae750@mail.gmail.com>
	<7B7B543E-ACC9-4D00-A246-6C8D2608B960@freeswitch.org>
	<9cb0e15e0910010735q3364cca8m8333740b1fd1a31@mail.gmail.com>
Message-ID: <F5F435C3-9295-402D-9D65-A886232E7859@freeswitch.org>

Thanks for posting the logs... But I'm not going to spend the time to  
download it.. unzip it and look at it... I would rather just click a  
link with the logs in plain text and read them in my browser.

I'll do it now but next time lets not add steps to the process that  
are not needed.  This goes for Jira too don't upload zip files of text  
logs that just makes it harder for us to quickly help you.

This isn't going to help me much know why Sofia/FreeSWITCH isn't  
working.

sofia profile xxx siptrace on
press F8
sofia loglevel all 9

Then post that please.

/b

On Oct 1, 2009, at 9:35 AM, Fernando Testa wrote:

> In the link below you have the entire SIP trace from system startup  
> until start receiving this annoying 407 Proxy Auth Required,  
> preventing FS to register successfully on the Ericsson Pabx.
> You can notice multiple registrations from named ericsson_1050 to  
> ericsson_1064 that starts failing after ~50 minutes after the boot.  
> Issuing a 'sofia external profile restart' solves the registration  
> problems.
> Brian, thanks for reply, but I really didn't get your point.
> Thank you, I apreciate any help.
>
> http://dl.getdropbox.com/u/410277/sip.log.gz

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From anthony.minessale at gmail.com  Thu Oct  1 07:55:54 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 1 Oct 2009 09:55:54 -0500
Subject: [Freeswitch-users] Dialplan Issue
In-Reply-To: <35b355e90910010727v2efcbff6p97a0a883857818e4@mail.gmail.com>
References: <35b355e90910010727v2efcbff6p97a0a883857818e4@mail.gmail.com>
Message-ID: <191c3a030910010755l387faba8x1d8f9bbf51c8ffa1@mail.gmail.com>

you left too fast.

1) I told you if you put <> around the sip uri it will work.
2) I told you I added a patch in tree to add one for you if it's not
supplied so update to trunk.


On Thu, Oct 1, 2009 at 9:27 AM, Shelby Ramsey <sicfslist at gmail.com> wrote:

> Hello:
>
> I asked this on IRC yesterday and I think I confused everyone involved.  So
> I apologize in advance here for reposting the question and if I wasted
> anyone's time.
>
> So here is the issue I'm having.  I'm trying to use FS as a redirect server
> (specifically to serve up LNP queries via 302 redirects).  But I'm having an
> issue where based on the string in the dialplan FS will respond with a 500
> internal error message instead of a 300 redirect.
>
> The call flow should be this:
>    -- remote party sends an Invite to my FS instance
>    -- FS should respond with a 302
>
> The following works as expected (FS will send a 302 when it receives an
> Invite):
>
> <action application="redirect" data="sip:${destination_number}@
> ${network_addr};rn=${rn};npdi=yes"/>
>
> However if I do this (which is the way the response should look) FS will
> respond with a 500 internal server error:
>
> <action application="redirect"
> data="sip:${destination_number};rn=${rn};npdi=yes@${network_addr}"/>
>
> So the issue is the placement of the user params .... if they are before
> the @ FS will send a 500 internal server error ... if they are after the @
> FS will send a 302.  Unfortunately placing the user params after the @
> doesn't quite conform to the way other devices expect to receive the 302 for
> this application.
>
> Any help would be greatly appreciated.
>
> Shelby
>
> PS ... hats off to the author of mod_memcache ... that is extremely useful!
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From jerry.richards at teotech.com  Thu Oct  1 07:57:29 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Thu, 1 Oct 2009 07:57:29 -0700
Subject: [Freeswitch-users] FS Does Not Relay Presence PUBLISH To
	Subscribing Phones
Message-ID: <B4B4D2747EEC4852B41A7D8D2DA36006@greyhawk.tonecommander.com>


By the way, I see the following lines at the FS console, which might be a
clue as to why this is happening.  Could someone point me toward what might
cause this?  I set the "manage-presence" parameter to "true" in each XML
file where I saw it defined.

[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
   ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
   ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
   ...
[WARNING] sofia_presence.c:565  192.168.72.38 is an alias, skipping

 
Best Regards,
Jerry
 

-----Original Message-----
From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Wednesday, September 30, 2009 9:12 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones

I have two phones configured to subscribe to each other's presence status.
When I change the presence status in one phone, I see the SIP PUBLISH
message going to FS, but I don't see FS relaying that presence status to the
subscribing phone.  Does anyone know why?

Best Regards,
Jerry




From sicfslist at gmail.com  Thu Oct  1 08:59:53 2009
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Thu, 1 Oct 2009 10:59:53 -0500
Subject: [Freeswitch-users] Dialplan Issue
In-Reply-To: <3DA34BA8-5E67-4A0C-A163-B72EF5F8638F@freeswitch.org>
References: <35b355e90910010727v2efcbff6p97a0a883857818e4@mail.gmail.com>
	<3DA34BA8-5E67-4A0C-A163-B72EF5F8638F@freeswitch.org>
Message-ID: <35b355e90910010859t5c7cd401n7a33cfe7a13ec962@mail.gmail.com>

Brian,

Thanks for the info.  I guess I'll go read section 19.1 of RFC3261 again.  I
do think the above has a valid host portion (I don't think the port is
required).

I'm not so sure that putting params in the user portion of the uri is valid
(from the RFC it states sip:user:password at host:port;uri-parameters?headers).

The issue is that in the real world this is done all the time .... SIP is
fantastic :)

Shelby


On Thu, Oct 1, 2009 at 9:42 AM, Brian West <brian at freeswitch.org> wrote:

>
> On Oct 1, 2009, at 9:27 AM, Shelby Ramsey wrote:
>
> <action application="redirect" data="
> sip:${destination_number};rn=${rn};npdi=yes@${network_addr}"/>
>
>
> This will produce an INVALID sip uri... You can not feed this to sofia
> it'll get PISSED.
>
> Its missing the host portion.
>
>
> So the issue is the placement of the user params .... if they are before
> the @ FS will send a 500 internal server error ... if they are after the @
> FS will send a 302.  Unfortunately placing the user params after the @
> doesn't quite conform to the way other devices expect to receive the 302 for
> this application.
>
> Any help would be greatly appreciated.
>
> Shelby
>
> PS ... hats off to the author of mod_memcache ... that is extremely useful!
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sicfslist at gmail.com  Thu Oct  1 09:01:37 2009
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Thu, 1 Oct 2009 11:01:37 -0500
Subject: [Freeswitch-users] Dialplan Issue
In-Reply-To: <191c3a030910010755l387faba8x1d8f9bbf51c8ffa1@mail.gmail.com>
References: <35b355e90910010727v2efcbff6p97a0a883857818e4@mail.gmail.com>
	<191c3a030910010755l387faba8x1d8f9bbf51c8ffa1@mail.gmail.com>
Message-ID: <35b355e90910010901j63a36531m7550eba26564f01@mail.gmail.com>

Tony,

Once again ... you are the man!

I'll try this right now.

SDR
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From brian at freeswitch.org  Thu Oct  1 09:15:33 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 1 Oct 2009 11:15:33 -0500
Subject: [Freeswitch-users] Dialplan Issue
In-Reply-To: <35b355e90910010859t5c7cd401n7a33cfe7a13ec962@mail.gmail.com>
References: <35b355e90910010727v2efcbff6p97a0a883857818e4@mail.gmail.com>
	<3DA34BA8-5E67-4A0C-A163-B72EF5F8638F@freeswitch.org>
	<35b355e90910010859t5c7cd401n7a33cfe7a13ec962@mail.gmail.com>
Message-ID: <DB2DDCF0-3E7A-4EF9-B35A-E700676B3F0B@freeswitch.org>

I wouldn't go that far! :P  You might be able to get away with it on  
the patch tony wrote but not sure.

/b

On Oct 1, 2009, at 10:59 AM, Shelby Ramsey wrote:

> SIP is fantastic :)




From sicfslist at gmail.com  Thu Oct  1 09:18:01 2009
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Thu, 1 Oct 2009 11:18:01 -0500
Subject: [Freeswitch-users] Dialplan Issue
In-Reply-To: <35b355e90910010901j63a36531m7550eba26564f01@mail.gmail.com>
References: <35b355e90910010727v2efcbff6p97a0a883857818e4@mail.gmail.com>
	<191c3a030910010755l387faba8x1d8f9bbf51c8ffa1@mail.gmail.com>
	<35b355e90910010901j63a36531m7550eba26564f01@mail.gmail.com>
Message-ID: <35b355e90910010918w7a01c4d1l7bc8b7986d2a3434@mail.gmail.com>

Just to confirm ... works like a champ.

Thanks again!!!
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From anthony.minessale at gmail.com  Thu Oct  1 09:35:34 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 1 Oct 2009 11:35:34 -0500
Subject: [Freeswitch-users] FS Does Not Relay Presence PUBLISH To
	Subscribing Phones
In-Reply-To: <B4B4D2747EEC4852B41A7D8D2DA36006@greyhawk.tonecommander.com>
References: <B4B4D2747EEC4852B41A7D8D2DA36006@greyhawk.tonecommander.com>
Message-ID: <191c3a030910010935s69e65569v76f4d26503a94d96@mail.gmail.com>

which phone is it,
we tested it with eyebeam and it appears to work for us.


On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>
> By the way, I see the following lines at the FS console, which might be a
> clue as to why this is happening.  Could someone point me toward what might
> cause this?  I set the "manage-presence" parameter to "true" in each XML
> file where I saw it defined.
>
> [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal)
> [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
>   ...
> [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6)
> [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
>   ...
> [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external)
> [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
>   ...
> [WARNING] sofia_presence.c:565  192.168.72.38 is an alias, skipping
>
>
> Best Regards,
> Jerry
>
>
> -----Original Message-----
> From: Jerry Richards [mailto:jerry.richards at teotech.com]
> Sent: Wednesday, September 30, 2009 9:12 AM
> To: 'freeswitch-users at lists.freeswitch.org'
> Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones
>
> I have two phones configured to subscribe to each other's presence status.
> When I change the presence status in one phone, I see the SIP PUBLISH
> message going to FS, but I don't see FS relaying that presence status to
> the
> subscribing phone.  Does anyone know why?
>
> Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mike at van.lammeren.net  Thu Oct  1 09:45:23 2009
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Thu, 1 Oct 2009 12:45:23 -0400
Subject: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
In-Reply-To: <5d2828f0909290720y6bb4997uc4b9afba4b0fe74c@mail.gmail.com>
References: <5d2828f0909290720y6bb4997uc4b9afba4b0fe74c@mail.gmail.com>
Message-ID: <5d2828f0910010945q30ed76e2x50d6ab3b13bc8f88@mail.gmail.com>

Guess what? I have two FreeSWITCH servers working behind UltraMonkey, using
heartbeat and ldirectord for load-balancing, fail-over and high
availability! I'm probably not the first one to do it, but as near as Google
and I can tell, I'm the first one to write about it.
Here's how you can duplicate my setup:

1. Install Ubuntu Server 8 on four machines, either real or VM.
2. Compile and install FreeSWITCH v1.0.4 from source on two machines,
following these instructions:
http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start
<http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start>3. Configure both
FreeSWITCH boxes, and make sure they are both working.
4. Follow (most of) these instructions from Daniel Aliaman's blog. They were
written for Asterisk, but since a SIP connection is a SIP connection, most
of the document applies to FreeSWITCH:
http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf

<http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf>The one
problem I ran into was the IP address and port to which FreeSWITCH was
bound. The default is to use the primary address, which works great
out-of-the-box for everything else. When a client tried to register, all it
got back was an ICMP error -- Destination Unreachable, Port Unreachable.
That error is returned when no sockets are listening for UDP packets. To get
FreeSWITCH to listen for your Virtual IP, you need to set it in two places:

5. In /opt/freeswitch/conf/vars.xml, set "bind_server_ip".
6. In /opt/freeswitch/conf/sip_profiles/internal.xml, set "sip-ip".

That should do it! If you have any success, please report to this list.

Keep in mind that if you want to do something like conferencing between two
registered clients, then you have to deal with the fact that the clients may
or may not be on the same box.

Mike van Lammeren



On Tue, Sep 29, 2009 at 10:20 AM, Mike van Lammeren
<mike at van.lammeren.net>wrote:

>
> On Mon, Sep 28, 2009 at 9:05 PM, "Even Andr? Fiskvik" <grevenx at me.com
> > wrote:
>
>> From: "Even Andr? Fiskvik" <grevenx at me.com>
>> To: freeswitch-users at lists.freeswitch.org
>> Date: Mon, 28 Sep 2009 22:52:13 +0200
>> Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
>> I have been working with a similar setup myself, but for some reason I
>> ended up ditching theUltraMonkey setup because I just couldn't get it to
>> work right.
>>
>> It's been quite a while since my effort, so I don't remember what the
>> exact issue was.
>> I got registrations to work, but had some other sip-dialog issues.
>>
>> We have since then changed over to running OpenSIPs as a loadbalancer in
>> front of
>> multiple FreeSWITCH instances. This setup is still in testing, but
>> seemlingy works fine
>> (and if it doesn't, it's my own fault for writing a bad opensips config).
>>
>> After we have done some more testing I can create a wiki-page with config
>> details.
>>
>>
>> Best regards,
>> Even Andr?
>>
>>
> Thanks, Even, that would be great! I might have to give up on the
> ultramonkey solution, since I can't find anyone who has made it work. It's
> too bad, because it would fit well with the rest of our architecture.
>
> Mike van Lammeren
>
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From Russell.Mosemann at cune.org  Thu Oct  1 07:37:10 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Thu, 1 Oct 2009 14:37:10 -0000
Subject: [Freeswitch-users] Connecting FS to Hicom 300
Message-ID: <20091001143710.AE8063C4C4D@mail.cune.org>

We have connected FS to a Siemens Hicomm 300. As you might guess, it's
not working right. Here is what we are working with.

Dell 1750 (dual socket, dual core Xeon 2.8GHz)
Debian 5
FS (15029), OpenZAP (without libpri)
TE110P T1 card (Zaptel driver)
Handles 71xx extensions

Siemens Hicom 300
TMDN64P T1 card
Handles 74xx extensions

We are pretty much using the stock FS configuration, yet, because we're
trying to get this to work. I have configured OpenZAP and the associated
files like the examples on the wiki (see below) to work with a PRI T1.
There are 23 B channels and 1 D channel. The Zaptel side looks fine.
OpenZAP is able to open the channels when FS boots. So far, so good.

When a call is made from 74xx (Rolmphone 624) to 71xx (X-Lite 4.0 beta
from CounterPath on an office PC), X-Lite rings. The call can be
answered, and the conversation sounds fine. That means the routing,
registration and authorization are working on the network between X-Lite
and FS. It also means that FS is able to communicate with the Hicom over
the T1. Great.

When the caller presses the transfer button on the 74xx phone, the Hicom
sends a message over the D channel, and the call is disconnected
(watching with fs_cli). As best I can interpret the bytes in the message,
the Hicom sends a disconnect message when 74xx presses the transfer key.

In order to call 74xx, I created dialplan/default/02_hicom.xml. The
contents are

<include>
  <extension name="hicom">
    <condition field="destination_number" expression="^(74\d{2})$">
      <action application="bridge" data="openzap/1/a/$1"/>
    </condition>
  </extension>
</include>

If a call is made from 71xx to 74xx, the Hicom forwards the call to the
switchboard with "7100->7445 connection not possible" (or whatever
extensions) in the switchboard display.

1. Are these issues related to the way I have configured FS?

The Hicom is maintained by the local phone company. I do not have access
to view or configure the T1 card on the Hicom. According to the phone
guy, there isn't anything else that needs to be configured on the Hicom.
He believes that if 74xx can call 71xx, then 71xx should be able to call
74xx.

I suspect that something more needs to be done on the Hicom in order to
accept calls from FS and bridge/transfer them to a local extension on the
Hicom. It's as if the Hicom doesn't know how or is not permitted to route
incoming calls on the T1 to local extensions. I have no way to know,
though. I'm hoping someone else has connected FS to a Hicom 300 and can
provide configuration details. If I could tell the phone guy something
like, "You need to look at <this>," that would help him out.

2. Should I receive CID/ANI from the Hicom?

X-Lite displays "OpenZAP" as the call and "1" as Other when the call
comes in, which is the information for the endpoint. Is there something I
need to do in the FS configuration to capture CID/ANI information from
the Hicom and make it available (or is it not being provided by the Hicom)?

3. When dialing from the Rolmphone is there a way for FS to send the
called name back to the Hicom for it to appear in the display?

When dialing 74xx to 74xx, of course, it shows the called number and name
in the display. We also have a HiPath 4000 connected to the Hicom 300.
When dialing an extension on the HiPath from the Hicom, the HiPath ships
the called name back to the Hicom for display on the phone. It would be
nice to do that from FS.

Let me know if you need additional information. Thanks for any pointers
or insight as to how things work.

-- 
Russell Mosemann


openzap.conf
[span zt PRI_1]
name => OpenZAP
number => 1
trunk_type => t1
b-channel => 1-23
d-channel => 24

zt.conf
[defaults]
codec_ms => 20
wink_ms => 150
flash_ms => 750
echo_cancel_level => 64
rxgain => 0.0
txgain => 0.0

openzap.conf
<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    <param name="debug" value="0"/>
    <!--<param name="hold-music" value="$${moh_uri}"/>-->
    <!--<param name="enable-analog-option" value="call-swap"/>-->
    <!--<param name="enable-analog-option" value="3-way"/>-->
  </settings>
   <pri_spans>
     <span name="PRI_1">
       <param name="q921loglevel" value="alert"/>
       <param name="q931loglevel" value="alert"/>
       <param name="mode" value="user"/>
       <param name="dialect" value="national"/>
       <param name="dialplan" value="XML"/>
       <param name="context" value="public"/>
     </span>
   </pri_spans>
</configuration>

zaptel.conf
# Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" (MASTER)
span=1,1,0,esf,b8zs
# termtype: te
bchan=1-23
dchan=24

# Global data
loadzone        = us
defaultzone     = us


________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!




From dmitry.bely at gmail.com  Thu Oct  1 10:01:10 2009
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Thu, 1 Oct 2009 21:01:10 +0400
Subject: [Freeswitch-users] How to limit the number of incoming+outgoing
	calls via specific gateway?
Message-ID: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>

My SIP provider allows only one call (incoming or outgoing) via one
SIP account. For FreeSWITCH I have configured it as public DID
extension and outgoing gateway. Now I would like to transfer to
another gw (or generate "limit exceded") when one tries to place an
outgoing call while incoming call is in progress. How tho do that?
Limiting the number of outgoing calls is easy (mod_limit), but how to
take into account incoming one?

- Dmitry Bely



From jerry.richards at teotech.com  Thu Oct  1 10:28:40 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Thu, 1 Oct 2009 10:28:40 -0700
Subject: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
	ToSubscribing Phones
In-Reply-To: <191c3a030910010935s69e65569v76f4d26503a94d96@mail.gmail.com>
References: <B4B4D2747EEC4852B41A7D8D2DA36006@greyhawk.tonecommander.com>
	<191c3a030910010935s69e65569v76f4d26503a94d96@mail.gmail.com>
Message-ID: <C92528AAF40D4D41B0F6B69A4CE644F5@greyhawk.tonecommander.com>

I am using two Bria Professional Version 2.5.4 Build 54835 softphones.
 
Thanks,
Jerry


  _____  

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Thursday, October 01, 2009 9:36 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
ToSubscribing Phones


which phone is it,
we tested it with eyebeam and it appears to work for us.



On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards <jerry.richards at teotech.com>
wrote:



By the way, I see the following lines at the FS console, which might be a
clue as to why this is happening.  Could someone point me toward what might
cause this?  I set the "manage-presence" parameter to "true" in each XML
file where I saw it defined.

[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[WARNING] sofia_presence.c:565  192.168.72.38 is an alias, skipping


Best Regards,
Jerry



-----Original Message-----
From: Jerry Richards [mailto:jerry.richards at teotech.com]
Sent: Wednesday, September 30, 2009 9:12 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones

I have two phones configured to subscribe to each other's presence status.
When I change the presence status in one phone, I see the SIP PUBLISH
message going to FS, but I don't see FS relaying that presence status to the
subscribing phone.  Does anyone know why?

Best Regards,
Jerry


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:213-799-1400


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From raffaele.p.guidi at gmail.com  Thu Oct  1 11:02:25 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Thu, 1 Oct 2009 20:02:25 +0200
Subject: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
In-Reply-To: <5d2828f0910010945q30ed76e2x50d6ab3b13bc8f88@mail.gmail.com>
References: <5d2828f0909290720y6bb4997uc4b9afba4b0fe74c@mail.gmail.com>
	<5d2828f0910010945q30ed76e2x50d6ab3b13bc8f88@mail.gmail.com>
Message-ID: <b0a8792f0910011102x691ba8a9s6a06cfd42483c935@mail.gmail.com>

And, should someone succed replicating this setup, consider writing about it
on the wiki :)

On Thu, Oct 1, 2009 at 18:45, Mike van Lammeren <mike at van.lammeren.net>wrote:

> Guess what? I have two FreeSWITCH servers working behind UltraMonkey, using
> heartbeat and ldirectord for load-balancing, fail-over and high
> availability! I'm probably not the first one to do it, but as near as Google
> and I can tell, I'm the first one to write about it.
> Here's how you can duplicate my setup:
>
> 1. Install Ubuntu Server 8 on four machines, either real or VM.
> 2. Compile and install FreeSWITCH v1.0.4 from source on two machines,
> following these instructions:
> http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start
>  <http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start>3. Configure both
> FreeSWITCH boxes, and make sure they are both working.
> 4. Follow (most of) these instructions from Daniel Aliaman's blog. They
> were written for Asterisk, but since a SIP connection is a SIP connection,
> most of the document applies to FreeSWITCH:
> http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf
>
> <http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf>The one
> problem I ran into was the IP address and port to which FreeSWITCH was
> bound. The default is to use the primary address, which works great
> out-of-the-box for everything else. When a client tried to register, all it
> got back was an ICMP error -- Destination Unreachable, Port Unreachable.
> That error is returned when no sockets are listening for UDP packets. To get
> FreeSWITCH to listen for your Virtual IP, you need to set it in two places:
>
> 5. In /opt/freeswitch/conf/vars.xml, set "bind_server_ip".
> 6. In /opt/freeswitch/conf/sip_profiles/internal.xml, set "sip-ip".
>
> That should do it! If you have any success, please report to this list.
>
> Keep in mind that if you want to do something like conferencing between two
> registered clients, then you have to deal with the fact that the clients may
> or may not be on the same box.
>
> Mike van Lammeren
>
>
>
> On Tue, Sep 29, 2009 at 10:20 AM, Mike van Lammeren <mike at van.lammeren.net
> > wrote:
>
>>
>> On Mon, Sep 28, 2009 at 9:05 PM, "Even Andr? Fiskvik" <grevenx at me.com
>> > wrote:
>>
>>> From: "Even Andr? Fiskvik" <grevenx at me.com>
>>> To: freeswitch-users at lists.freeswitch.org
>>> Date: Mon, 28 Sep 2009 22:52:13 +0200
>>> Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
>>> I have been working with a similar setup myself, but for some reason I
>>> ended up ditching theUltraMonkey setup because I just couldn't get it to
>>> work right.
>>>
>>> It's been quite a while since my effort, so I don't remember what the
>>> exact issue was.
>>> I got registrations to work, but had some other sip-dialog issues.
>>>
>>> We have since then changed over to running OpenSIPs as a loadbalancer in
>>> front of
>>> multiple FreeSWITCH instances. This setup is still in testing, but
>>> seemlingy works fine
>>> (and if it doesn't, it's my own fault for writing a bad opensips config).
>>>
>>> After we have done some more testing I can create a wiki-page with config
>>> details.
>>>
>>>
>>> Best regards,
>>> Even Andr?
>>>
>>>
>> Thanks, Even, that would be great! I might have to give up on the
>> ultramonkey solution, since I can't find anyone who has made it work. It's
>> too bad, because it would fit well with the rest of our architecture.
>>
>> Mike van Lammeren
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From grevenx at me.com  Thu Oct  1 11:12:59 2009
From: grevenx at me.com (=?iso-8859-1?Q?Even_Andr=E9_Fiskvik?=)
Date: Thu, 01 Oct 2009 20:12:59 +0200
Subject: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
In-Reply-To: <5d2828f0910010945q30ed76e2x50d6ab3b13bc8f88@mail.gmail.com>
References: <5d2828f0909290720y6bb4997uc4b9afba4b0fe74c@mail.gmail.com>
	<5d2828f0910010945q30ed76e2x50d6ab3b13bc8f88@mail.gmail.com>
Message-ID: <32C47FA5-774E-4584-890B-0A066F5CFE14@me.com>

That's very cool Mike!

I'm going to try to configure four boxes with this as well (Btw, did  
you use physical hardware or virtualization?)
and see how it goes. I followed Daniel Aliaman's blog as well, but I  
can try it again with the tips
you provided on FreeSWITCH config to see if I can get it working  
properly this time.
We did the setup on CentOS, but I wouldn't think that would be any  
issue.

Perhaps you or we could write up a complete guide about this on the  
wiki since this is an scenario
commonly used? Also it would be great if we could outline possible  
issues (and even better solutions)
to this kind of setup with regards to stuff like conferencing,  
bridging between registered users and presence.


Best regards,
Even Andr?


On 1. okt. 2009, at 18.45, Mike van Lammeren wrote:

> Guess what? I have two FreeSWITCH servers working behind  
> UltraMonkey, using heartbeat and ldirectord for load-balancing, fail- 
> over and high availability! I'm probably not the first one to do it,  
> but as near as Google and I can tell, I'm the first one to write  
> about it.
>
> Here's how you can duplicate my setup:
>
> 1. Install Ubuntu Server 8 on four machines, either real or VM.
> 2. Compile and install FreeSWITCH v1.0.4 from source on two  
> machines, following these instructions: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start
> 3. Configure both FreeSWITCH boxes, and make sure they are both  
> working.
> 4. Follow (most of) these instructions from Daniel Aliaman's blog.  
> They were written for Asterisk, but since a SIP connection is a SIP  
> connection, most of the document applies to FreeSWITCH:
> http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf
>
> The one problem I ran into was the IP address and port to which  
> FreeSWITCH was bound. The default is to use the primary address,  
> which works great out-of-the-box for everything else. When a client  
> tried to register, all it got back was an ICMP error -- Destination  
> Unreachable, Port Unreachable. That error is returned when no  
> sockets are listening for UDP packets. To get FreeSWITCH to listen  
> for your Virtual IP, you need to set it in two places:
>
> 5. In /opt/freeswitch/conf/vars.xml, set "bind_server_ip".
> 6. In /opt/freeswitch/conf/sip_profiles/internal.xml, set "sip-ip".
>
> That should do it! If you have any success, please report to this  
> list.
>
> Keep in mind that if you want to do something like conferencing  
> between two registered clients, then you have to deal with the fact  
> that the clients may or may not be on the same box.
>
> Mike van Lammeren
>
>
>
> On Tue, Sep 29, 2009 at 10:20 AM, Mike van Lammeren <mike at van.lammeren.net 
> > wrote:
>
> On Mon, Sep 28, 2009 at 9:05 PM, "Even Andr? Fiskvik"  
> <grevenx at me.com> wrote:
> From: "Even Andr? Fiskvik" <grevenx at me.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Mon, 28 Sep 2009 22:52:13 +0200
> Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with  
> Ultramonkey
> I have been working with a similar setup myself, but for some reason  
> I ended up ditching the
> UltraMonkey setup because I just couldn't get it to work right.
>
> It's been quite a while since my effort, so I don't remember what  
> the exact issue was.
> I got registrations to work, but had some other sip-dialog issues.
>
> We have since then changed over to running OpenSIPs as a  
> loadbalancer in front of
> multiple FreeSWITCH instances. This setup is still in testing, but  
> seemlingy works fine
> (and if it doesn't, it's my own fault for writing a bad opensips  
> config).
>
> After we have done some more testing I can create a wiki-page with  
> config details.
>
>
> Best regards,
> Even Andr?
>
>
> Thanks, Even, that would be great! I might have to give up on the  
> ultramonkey solution, since I can't find anyone who has made it  
> work. It's too bad, because it would fit well with the rest of our  
> architecture.
>
> Mike van Lammeren
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From jerry.richards at teotech.com  Thu Oct  1 11:29:20 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Thu, 1 Oct 2009 11:29:20 -0700
Subject: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
	ToSubscribing Phones
References: <B4B4D2747EEC4852B41A7D8D2DA36006@greyhawk.tonecommander.com>
	<191c3a030910010935s69e65569v76f4d26503a94d96@mail.gmail.com> 
Message-ID: <9B5214D274044021A57C6857844C10BA@greyhawk.tonecommander.com>

If you have time to take a look, I could put a trace in the pastebin?
 
Jerry


  _____  

From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Thursday, October 01, 2009 10:29 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
ToSubscribing Phones


I am using two Bria Professional Version 2.5.4 Build 54835 softphones.
 
Thanks,
Jerry


  _____  

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Thursday, October 01, 2009 9:36 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
ToSubscribing Phones


which phone is it,
we tested it with eyebeam and it appears to work for us.



On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards <jerry.richards at teotech.com>
wrote:



By the way, I see the following lines at the FS console, which might be a
clue as to why this is happening.  Could someone point me toward what might
cause this?  I set the "manage-presence" parameter to "true" in each XML
file where I saw it defined.

[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[WARNING] sofia_presence.c:565  192.168.72.38 is an alias, skipping


Best Regards,
Jerry



-----Original Message-----
From: Jerry Richards [mailto:jerry.richards at teotech.com]
Sent: Wednesday, September 30, 2009 9:12 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones

I have two phones configured to subscribe to each other's presence status.
When I change the presence status in one phone, I see the SIP PUBLISH
message going to FS, but I don't see FS relaying that presence status to the
subscribing phone.  Does anyone know why?

Best Regards,
Jerry


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:213-799-1400


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From anthony.minessale at gmail.com  Thu Oct  1 12:14:14 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 1 Oct 2009 14:14:14 -0500
Subject: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
In-Reply-To: <32C47FA5-774E-4584-890B-0A066F5CFE14@me.com>
References: <5d2828f0909290720y6bb4997uc4b9afba4b0fe74c@mail.gmail.com>
	<5d2828f0910010945q30ed76e2x50d6ab3b13bc8f88@mail.gmail.com>
	<32C47FA5-774E-4584-890B-0A066F5CFE14@me.com>
Message-ID: <191c3a030910011214i3a9eaad1xf242ecf19dc80da3@mail.gmail.com>

can we do it without advertising to use ubuntu =D
We don't like encouraging our users to use bleeding edge OS for our own
sanity with debugging.
Not to say you are not allowed to I just don't want to encourage it =p



On Thu, Oct 1, 2009 at 1:12 PM, Even Andr? Fiskvik <grevenx at me.com> wrote:

> That's very cool Mike!
> I'm going to try to configure four boxes with this as well (Btw, did you
> use physical hardware or virtualization?)
> and see how it goes. I followed Daniel Aliaman's blog as well, but I can
> try it again with the tips
> you provided on FreeSWITCH config to see if I can get it working properly
> this time.
> We did the setup on CentOS, but I wouldn't think that would be any issue.
>
> Perhaps you or we could write up a complete guide about this on the wiki
> since this is an scenario
> commonly used? Also it would be great if we could outline possible issues
> (and even better solutions)
> to this kind of setup with regards to stuff like conferencing, bridging
> between registered users and presence.
>
>
> Best regards,
> Even Andr?
>
>
> On 1. okt. 2009, at 18.45, Mike van Lammeren wrote:
>
> Guess what? I have two FreeSWITCH servers working behind UltraMonkey, using
> heartbeat and ldirectord for load-balancing, fail-over and high
> availability! I'm probably not the first one to do it, but as near as Google
> and I can tell, I'm the first one to write about it.
> Here's how you can duplicate my setup:
>
> 1. Install Ubuntu Server 8 on four machines, either real or VM.
> 2. Compile and install FreeSWITCH v1.0.4 from source on two machines,
> following these instructions:
> http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start
>  <http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start>3. Configure both
> FreeSWITCH boxes, and make sure they are both working.
> 4. Follow (most of) these instructions from Daniel Aliaman's blog. They
> were written for Asterisk, but since a SIP connection is a SIP connection,
> most of the document applies to FreeSWITCH:
> http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf
>
> <http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf>The one
> problem I ran into was the IP address and port to which FreeSWITCH was
> bound. The default is to use the primary address, which works great
> out-of-the-box for everything else. When a client tried to register, all it
> got back was an ICMP error -- Destination Unreachable, Port Unreachable.
> That error is returned when no sockets are listening for UDP packets. To get
> FreeSWITCH to listen for your Virtual IP, you need to set it in two places:
>
> 5. In /opt/freeswitch/conf/vars.xml, set "bind_server_ip".
> 6. In /opt/freeswitch/conf/sip_profiles/internal.xml, set "sip-ip".
>
> That should do it! If you have any success, please report to this list.
>
> Keep in mind that if you want to do something like conferencing between two
> registered clients, then you have to deal with the fact that the clients may
> or may not be on the same box.
>
> Mike van Lammeren
>
>
>
> On Tue, Sep 29, 2009 at 10:20 AM, Mike van Lammeren <mike at van.lammeren.net
> > wrote:
>
>>
>> On Mon, Sep 28, 2009 at 9:05 PM, "Even Andr? Fiskvik" <grevenx at me.com
>> > wrote:
>>
>>> From: "Even Andr? Fiskvik" <grevenx at me.com>
>>> To: freeswitch-users at lists.freeswitch.org
>>> Date: Mon, 28 Sep 2009 22:52:13 +0200
>>> Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
>>> I have been working with a similar setup myself, but for some reason I
>>> ended up ditching theUltraMonkey setup because I just couldn't get it to
>>> work right.
>>>
>>> It's been quite a while since my effort, so I don't remember what the
>>> exact issue was.
>>> I got registrations to work, but had some other sip-dialog issues.
>>>
>>> We have since then changed over to running OpenSIPs as a loadbalancer in
>>> front of
>>> multiple FreeSWITCH instances. This setup is still in testing, but
>>> seemlingy works fine
>>> (and if it doesn't, it's my own fault for writing a bad opensips config).
>>>
>>> After we have done some more testing I can create a wiki-page with config
>>> details.
>>>
>>>
>>> Best regards,
>>> Even Andr?
>>>
>>>
>> Thanks, Even, that would be great! I might have to give up on the
>> ultramonkey solution, since I can't find anyone who has made it work. It's
>> too bad, because it would fit well with the rest of our architecture.
>>
>> Mike van Lammeren
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
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FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From anthony.minessale at gmail.com  Thu Oct  1 12:17:11 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 1 Oct 2009 14:17:11 -0500
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <20091001143710.AE8063C4C4D@mail.cune.org>
References: <20091001143710.AE8063C4C4D@mail.cune.org>
Message-ID: <191c3a030910011217o6aed6b2ag1ded3d26a62430dc@mail.gmail.com>

You might want to try the ozmod_pri instead of ozmod_isdn until the new
revision of ozmod_isdn is published into the source tree.


On Thu, Oct 1, 2009 at 9:37 AM, <Russell.Mosemann at cune.org> wrote:

> We have connected FS to a Siemens Hicomm 300. As you might guess, it's
> not working right. Here is what we are working with.
>
> Dell 1750 (dual socket, dual core Xeon 2.8GHz)
> Debian 5
> FS (15029), OpenZAP (without libpri)
> TE110P T1 card (Zaptel driver)
> Handles 71xx extensions
>
> Siemens Hicom 300
> TMDN64P T1 card
> Handles 74xx extensions
>
> We are pretty much using the stock FS configuration, yet, because we're
> trying to get this to work. I have configured OpenZAP and the associated
> files like the examples on the wiki (see below) to work with a PRI T1.
> There are 23 B channels and 1 D channel. The Zaptel side looks fine.
> OpenZAP is able to open the channels when FS boots. So far, so good.
>
> When a call is made from 74xx (Rolmphone 624) to 71xx (X-Lite 4.0 beta
> from CounterPath on an office PC), X-Lite rings. The call can be
> answered, and the conversation sounds fine. That means the routing,
> registration and authorization are working on the network between X-Lite
> and FS. It also means that FS is able to communicate with the Hicom over
> the T1. Great.
>
> When the caller presses the transfer button on the 74xx phone, the Hicom
> sends a message over the D channel, and the call is disconnected
> (watching with fs_cli). As best I can interpret the bytes in the message,
> the Hicom sends a disconnect message when 74xx presses the transfer key.
>
> In order to call 74xx, I created dialplan/default/02_hicom.xml. The
> contents are
>
> <include>
>  <extension name="hicom">
>    <condition field="destination_number" expression="^(74\d{2})$">
>      <action application="bridge" data="openzap/1/a/$1"/>
>    </condition>
>  </extension>
> </include>
>
> If a call is made from 71xx to 74xx, the Hicom forwards the call to the
> switchboard with "7100->7445 connection not possible" (or whatever
> extensions) in the switchboard display.
>
> 1. Are these issues related to the way I have configured FS?
>
> The Hicom is maintained by the local phone company. I do not have access
> to view or configure the T1 card on the Hicom. According to the phone
> guy, there isn't anything else that needs to be configured on the Hicom.
> He believes that if 74xx can call 71xx, then 71xx should be able to call
> 74xx.
>
> I suspect that something more needs to be done on the Hicom in order to
> accept calls from FS and bridge/transfer them to a local extension on the
> Hicom. It's as if the Hicom doesn't know how or is not permitted to route
> incoming calls on the T1 to local extensions. I have no way to know,
> though. I'm hoping someone else has connected FS to a Hicom 300 and can
> provide configuration details. If I could tell the phone guy something
> like, "You need to look at <this>," that would help him out.
>
> 2. Should I receive CID/ANI from the Hicom?
>
> X-Lite displays "OpenZAP" as the call and "1" as Other when the call
> comes in, which is the information for the endpoint. Is there something I
> need to do in the FS configuration to capture CID/ANI information from
> the Hicom and make it available (or is it not being provided by the Hicom)?
>
> 3. When dialing from the Rolmphone is there a way for FS to send the
> called name back to the Hicom for it to appear in the display?
>
> When dialing 74xx to 74xx, of course, it shows the called number and name
> in the display. We also have a HiPath 4000 connected to the Hicom 300.
> When dialing an extension on the HiPath from the Hicom, the HiPath ships
> the called name back to the Hicom for display on the phone. It would be
> nice to do that from FS.
>
> Let me know if you need additional information. Thanks for any pointers
> or insight as to how things work.
>
> --
> Russell Mosemann
>
>
> openzap.conf
> [span zt PRI_1]
> name => OpenZAP
> number => 1
> trunk_type => t1
> b-channel => 1-23
> d-channel => 24
>
> zt.conf
> [defaults]
> codec_ms => 20
> wink_ms => 150
> flash_ms => 750
> echo_cancel_level => 64
> rxgain => 0.0
> txgain => 0.0
>
> openzap.conf
> <configuration name="openzap.conf" description="OpenZAP Configuration">
>  <settings>
>    <param name="debug" value="0"/>
>    <!--<param name="hold-music" value="$${moh_uri}"/>-->
>    <!--<param name="enable-analog-option" value="call-swap"/>-->
>    <!--<param name="enable-analog-option" value="3-way"/>-->
>  </settings>
>   <pri_spans>
>     <span name="PRI_1">
>       <param name="q921loglevel" value="alert"/>
>       <param name="q931loglevel" value="alert"/>
>       <param name="mode" value="user"/>
>       <param name="dialect" value="national"/>
>       <param name="dialplan" value="XML"/>
>       <param name="context" value="public"/>
>     </span>
>   </pri_spans>
> </configuration>
>
> zaptel.conf
> # Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" (MASTER)
> span=1,1,0,esf,b8zs
> # termtype: te
> bchan=1-23
> dchan=24
>
> # Global data
> loadzone        = us
> defaultzone     = us
>
>
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From nicolas at medularis.com  Thu Oct  1 12:29:09 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Thu, 1 Oct 2009 15:29:09 -0400
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <1b46b4e80909291542k71440965g75bb3bfcf38e8005@mail.gmail.com>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<191c3a030909291122t770af037j5b1068197b31de87@mail.gmail.com>
	<1b46b4e80909291542k71440965g75bb3bfcf38e8005@mail.gmail.com>
Message-ID: <1b46b4e80910011229k31f7233du10b0fc661730d3f3@mail.gmail.com>

Any ideas about this?

The SIP provider is offering H323, but I'm not quite sure about that, is
mod_opal working right?

Thanks!

Nicolas

On Tue, Sep 29, 2009 at 6:42 PM, Nicolas Brenner <nicolas at medularis.com>wrote:

> Anthony, thanks. Below are my config files for the two gateways from the
> sip trace. Both files are located in conf/directory/default.
>
> ---------------------
>
> redvoiss.xml (the one that works)
>
> <include>
>   <user id="gateway_redvoiss">
>     <gateways>
>       <gateway name="redvoiss-pp">
>         <param name="username" value="xxxxxxx"/>
>         <param name="password" value="xxxxxxx"/>
>         <param name="from-domain" value="pxextmy.redvoiss.net"/>
>         <param name="realm" value="pxextmy.redvoiss.net"/>
>         <param name="proxy" value="pxextmy.redvoiss.net"/>
>         <param name="from-user" value="xxxxxxx"/>
>         <param name="caller-id-in-from" value="false"/>
>         <param name="expire-seconds" value="600"/>
>         <param name="register" value="true"/>
>         <param name="retry_seconds" value="5"/>
>         <param name="extension" value="2010"/>
>         <param name="context" value="public"/>
>         <param name="codec-prefs" value="G729"/>
>         <param name="rfc2833-pt" value="101"/>
>       </gateway>
>     </gateways>
>     <params>
>       <param name="password" value="4321"/>
>     </params>
>   </user>
> </include>
>
> ---------------------
>
> orange.xml (the one that doesn't work)
>
> <include>
>   <user id="gateway_orange">
>     <gateways>
>       <gateway name="orange">
>         <param name="username" value="xxxxxxx"/>
>         <param name="password" value="xxxxxxx"/>
>         <param name="from-domain" value="216.72.10.39"/>
>         <param name="realm" value="216.72.10.39"/>
>         <param name="proxy" value="216.72.10.39"/>
>         <param name="from-user" value="xxxxxxx"/>
>         <param name="caller-id-in-from" value="false"/>
>         <param name="expire-seconds" value="600"/>
>         <param name="register" value="true"/>
>         <param name="retry_seconds" value="5"/>
>         <param name="extension" value="2011"/>
>         <param name="context" value="public"/>
>         <param name="codec-prefs" value="G729"/>
>         <param name="rfc2833-pt" value="101"/>
>       </gateway>
>     </gateways>
>     <params>
>       <param name="password" value="4321"/>
>     </params>
>   </user>
> </include>
>
> ---------------------
>
> If I remove the register=true param for the non-working gateway, I don't
> get the registration error on the cli, but then all call attempts get
> rejected with a 401 Unauthorized, and I get a hangup cause of
> NORMAL_UNSPECIFIED.
>
>
> Best,
>
> Nicolas
>
>
>
> On Tue, Sep 29, 2009 at 2:22 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> 900 level errors are sofia internal errors so probably something is wrong
>> with your gateway config xml.
>> if you want to send it with any critical info replaced with XXX maybe we
>> can see the issue for you.
>>
>>
>>
>> On Tue, Sep 29, 2009 at 1:05 PM, Nicolas Brenner <nicolas at medularis.com>wrote:
>>
>>> Hello everyone,
>>>
>>> I am trying to add a gateway, but after configuring it just like the
>>> others gateways I have, it is failing to register with a message like this:
>>>
>>> 2009-09-29 12:54:40.853440 [ERR] sofia_reg.c:1402 orange Registration
>>> Failed with status Operation has no matching challenge  [904]. failure #1
>>> 2009-09-29 12:54:40.906798 [WARNING] sofia_reg.c:364 orange Failed
>>> Registration, setting retry to 10 seconds.
>>>
>>> I captured the sip traffic and noticed that when trying to register with
>>> one gateway (the one that works), I get a "Trying" reply immediately
>>> followed by a "401 Unauthorized" which contains a "WWW-Authenticate: digest"
>>> with a "qop=auth" parameter. Then Freeswitch replies with a second REGISTER
>>> including a large "Authorization: digest" section with cnonce and
>>> nc=00000001 parameters.
>>>
>>> The gateway which doesn't register, doesn't send the "qop=auth" parameter
>>> together with the "401 Unauthorized", and then Freeswitch sends a
>>> "Authorization: digest" section on the second REGISTER with no cnonce or nc
>>> parameters.
>>>
>>> I know very little abouth SIP, so I'm wondering what this "qop=auth"
>>> parameter means and how does it affect the registration process. Is there
>>> any way to do without the qop=auth parameter?
>>>
>>> Also, I tried registering with X-Lite directly to the gateway, and it
>>> worked, so it appears to be a problem in the Freeswitch/gateway combination.
>>> (Note: X-Lite sends an "Authorization: digest" section on the _first_
>>> REGISTER, apparently this makes a difference)
>>>
>>> Attached is a sip trace for the registration traffic when doing "sofia
>>> profile external restart reloadxml" on the cli, captured with "tshark -i
>>> eth0 -o "rtp.heuristic_rtp: TRUE" -w /tmp/capture.pcap -b filesize:51200 -b
>>> files:100 -R 'sip or rtp or icmp or dns or rtcp or t38'"
>>>
>>> Thanks!
>>>
>>> Nicolas
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From hads at nice.net.nz  Thu Oct  1 12:41:24 2009
From: hads at nice.net.nz (Hadley Rich)
Date: Fri, 02 Oct 2009 08:41:24 +1300
Subject: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
In-Reply-To: <191c3a030910011214i3a9eaad1xf242ecf19dc80da3@mail.gmail.com>
References: <5d2828f0909290720y6bb4997uc4b9afba4b0fe74c@mail.gmail.com>
	<5d2828f0910010945q30ed76e2x50d6ab3b13bc8f88@mail.gmail.com>
	<32C47FA5-774E-4584-890B-0A066F5CFE14@me.com>
	<191c3a030910011214i3a9eaad1xf242ecf19dc80da3@mail.gmail.com>
Message-ID: <1254426084.3779.29.camel@sodium>

On Thu, 2009-10-01 at 14:14 -0500, Anthony Minessale wrote:
> can we do it without advertising to use ubuntu =D
> We don't like encouraging our users to use bleeding edge OS for our
> own sanity with debugging. 

I understand your stance, though if we're talking about Ubuntu 8.04 LTS
(Long Term Support - 5 years) it's not really bleeding edge anymore. 18
months ago when it was released it may have been a little, but the LTS
releases still aren't as bleeding edge as the standard support in
between releases.

hads
-- 
http://nicegear.co.nz
New Zealand's Open Source Hardware Supplier




From peder at networkoblivion.com  Thu Oct  1 12:44:08 2009
From: peder at networkoblivion.com (Peder)
Date: Thu, 1 Oct 2009 14:44:08 -0500
Subject: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
In-Reply-To: <191c3a030910011214i3a9eaad1xf242ecf19dc80da3@mail.gmail.com>
References: <5d2828f0909290720y6bb4997uc4b9afba4b0fe74c@mail.gmail.com>	<5d2828f0910010945q30ed76e2x50d6ab3b13bc8f88@mail.gmail.com>	<32C47FA5-774E-4584-890B-0A066F5CFE14@me.com>
	<191c3a030910011214i3a9eaad1xf242ecf19dc80da3@mail.gmail.com>
Message-ID: <0ca401ca42cf$8a709d20$9f51d760$@com>

Looking thru the example, it looks like each box has a real address of 21,
22 or 23 and they all have a loopback of .17, right?  So even though they
connections are being load balanced, each box really thinks it is .17 and
each client that connects thinks it is connecting to .17, right?  If that?s
the case, how does a client on one box call a client on the other box?
Since every box thinks it is .17 how would you bridge to another user on
another box that also thinks it is .17?  Or am I totally missing how it
works?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Thursday, October 01, 2009 2:14 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey

 

can we do it without advertising to use ubuntu =D
We don't like encouraging our users to use bleeding edge OS for our own
sanity with debugging.
Not to say you are not allowed to I just don't want to encourage it =p




On Thu, Oct 1, 2009 at 1:12 PM, Even Andr? Fiskvik <grevenx at me.com> wrote:

That's very cool Mike!

 

I'm going to try to configure four boxes with this as well (Btw, did you use
physical hardware or virtualization?)

and see how it goes. I followed Daniel Aliaman's blog as well, but I can try
it again with the tips

you provided on FreeSWITCH config to see if I can get it working properly
this time.

We did the setup on CentOS, but I wouldn't think that would be any issue.

 

Perhaps you or we could write up a complete guide about this on the wiki
since this is an scenario

commonly used? Also it would be great if we could outline possible issues
(and even better solutions)

to this kind of setup with regards to stuff like conferencing, bridging
between registered users and presence.

 

 

Best regards,

Even Andr?

 

 

On 1. okt. 2009, at 18.45, Mike van Lammeren wrote:

 

Guess what? I have two FreeSWITCH servers working behind UltraMonkey, using
heartbeat and ldirectord for load-balancing, fail-over and high
availability! I'm probably not the first one to do it, but as near as Google
and I can tell, I'm the first one to write about it.

 

Here's how you can duplicate my setup:

 

1. Install Ubuntu Server 8 on four machines, either real or VM.

2. Compile and install FreeSWITCH v1.0.4 from source on two machines,
following these instructions:
http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start

3. Configure both FreeSWITCH boxes, and make sure they are both working.

4. Follow (most of) these instructions from Daniel Aliaman's blog. They were
written for Asterisk, but since a SIP connection is a SIP connection, most
of the document applies to FreeSWITCH:

http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf

 

The one problem I ran into was the IP address and port to which FreeSWITCH
was bound. The default is to use the primary address, which works great
out-of-the-box for everything else. When a client tried to register, all it
got back was an ICMP error -- Destination Unreachable, Port Unreachable.
That error is returned when no sockets are listening for UDP packets. To get
FreeSWITCH to listen for your Virtual IP, you need to set it in two places:

 

5. In /opt/freeswitch/conf/vars.xml, set "bind_server_ip".

6. In /opt/freeswitch/conf/sip_profiles/internal.xml, set "sip-ip".

 

That should do it! If you have any success, please report to this list.

 

Keep in mind that if you want to do something like conferencing between two
registered clients, then you have to deal with the fact that the clients may
or may not be on the same box.

 

Mike van Lammeren

 

 

 

On Tue, Sep 29, 2009 at 10:20 AM, Mike van Lammeren <mike at van.lammeren.net>
wrote:

 

On Mon, Sep 28, 2009 at 9:05 PM, "Even Andr? Fiskvik" <grevenx at me.com>
wrote:

From: "Even Andr? Fiskvik" <grevenx at me.com>
To: freeswitch-users at lists.freeswitch.org
Date: Mon, 28 Sep 2009 22:52:13 +0200
Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey

I have been working with a similar setup myself, but for some reason I ended
up ditching the

UltraMonkey setup because I just couldn't get it to work right.

 

It's been quite a while since my effort, so I don't remember what the exact
issue was.

I got registrations to work, but had some other sip-dialog issues.

 

We have since then changed over to running OpenSIPs as a loadbalancer in
front of 

multiple FreeSWITCH instances. This setup is still in testing, but seemlingy
works fine

(and if it doesn't, it's my own fault for writing a bad opensips config).

 

After we have done some more testing I can create a wiki-page with config
details.

 

 

Best regards,

Even Andr?

 

 

Thanks, Even, that would be great! I might have to give up on the
ultramonkey solution, since I can't find anyone who has made it work. It's
too bad, because it would fit well with the rest of our architecture.

 

Mike van Lammeren

 

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:213-799-1400

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From siniypin at gmail.com  Thu Oct  1 14:03:38 2009
From: siniypin at gmail.com (RobertT)
Date: Fri, 2 Oct 2009 01:03:38 +0400
Subject: [Freeswitch-users] conference participant from behind NAT
In-Reply-To: <A3EEB6BE-19E2-49F3-8E41-FC74307A18E2@jerris.com>
References: <2160023e0909290132p2a5d0b62jbcb7d37625686866@mail.gmail.com>
	<681a20520909290618i2fbe7199rb18ff32933bba952@mail.gmail.com>
	<2160023e0909290634h3102d81bj9e3dd3d4d56f43e0@mail.gmail.com>
	<A3EEB6BE-19E2-49F3-8E41-FC74307A18E2@jerris.com>
Message-ID: <2160023e0910011403r5e9d0bb0ib5699f0523aaa606@mail.gmail.com>

I am still experiencing problem with lost media in conference on a client
behind NAT.
This is what I've done - disabled VAD on a NATed client and asked my friend
to produce lots of animal sounds in order to keep channel busy. But at the
end of minute sounds of wild nature disapeared again. We reproduced that
without security with tcp SIP transport and got the same result.
Then I started to dig into SIP trace and this is what I found.
This client (behind NAT) recieve subsequent INVITE message from FS which
seem to destroy dialog and causes client app to close media stream after a
session being established normally. I performed the same call from box with
public ip and saw no subsequent INVITE's from FS. How come FS sends an
INVITE message to already connected client? Is it OK? Should client handle
this normally?

Below is client's SIP trace:

INVITE sip:1.conference.dw at 74.208.167.44:5081;transport=TLS SIP/2.0
...
User-Agent: DoxWox SIP user agent
..

SIP/2.0 100 Trying
..
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M
..

SIP/2.0 407 Proxy Authentication Required
..
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M
..

ACK sip:1.conference.dw at 74.208.167.44:5081;transport=TLS SIP/2.0
..

SIP/2.0 100 Trying
..
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M
..

SIP/2.0 183 Session Progress
..
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M
..

SIP/2.0 200 OK
..
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M
..

ACK sip:1.conference.dw at 74.208.167.44:5081;transport=tls SIP/2.0
..

*Finally, this message cause media stream closing*
INVITE sip:1001 at 87.184.52.45:64183;transport=tls SIP/2.0
Via: SIP/2.0/TLS 74.208.167.44:5081;branch=z9hG4bK8269NDyXQNjyH
Max-Forwards: 70
From: <sip:1.conference.dw at 74.208.167.44<sip%3A1.conference.dw at 74.208.167.44>>;tag=vQH234QtN2U8Q

To: <sip:1001 at 74.208.167.44
<sip%3A1001 at 74.208.167.44>>;tag=3a231ba86c894ceca81d5021b68d3b6c

Call-ID: 37edc38329f64fe98c36cc0a6ddcbd9d
CSeq: 121093810 INVITE
Contact: <sip:1.conference.dw at 74.208.167.44:5081;transport=tls>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY
Supported: timer, precondition, path, replaces
Session-Expires: 120;refresher=uac
Min-SE: 120
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 340
v=0
o=FreeSWITCH 1254396503 1254396504 IN IP4 74.208.167.44
s=FreeSWITCH
c=IN IP4 74.208.167.44
t=0 0
m=audio 27726 RTP/SAVP 103 101
a=rtpmap:103 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:J6x2UgRVUY8GfbwjCuyyttrtnXnwwuWA9Pt+o3VW
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From frank at carmickle.com  Thu Oct  1 14:26:02 2009
From: frank at carmickle.com (Frank Carmickle)
Date: Thu, 1 Oct 2009 17:26:02 -0400
Subject: [Freeswitch-users] register problem
Message-ID: <20091001212602.GC17256@base.carmickle.com>

Can someone point out what is wrong here.  Thanks.

Siptrace at http://carmickle.com/fs.txt

<include>
      <gateway name="voiptalk.org">
        <param name="username" value="81100068"/>
        <param name="password" value="my_pass"/>
        <param name="from-user" value="81100068"/>
        <param name="from-domain" value="voiptalk.org"/>
    <param name="proxy" value="voiptalk.org"/>
        <param name="expire-seconds" value="600"/>
        <param name="register" value="true"/>
        <param name="retry-seconds" value="30"/>
        <param name="extension" value="1000"/>
        <param name="contact-params" value="domain_name=$${domain}"/>
        <param name="context" value="public"/>
      </gateway>
</include>
 
--FC



From lists at venturevoip.com  Thu Oct  1 14:36:32 2009
From: lists at venturevoip.com (Matt Riddell)
Date: Fri, 02 Oct 2009 10:36:32 +1300
Subject: [Freeswitch-users] bgapi jobid to uuid
Message-ID: <4AC520E0.4020507@venturevoip.com>

Hi,

I decided to go with a linked list for current channels and maintain 
that through state changes.

So, basically it works like this:

1. Originate a call using bgapi (we get a jobid in response)
2. Receive an event with the jobid and a uuid
3. Lookup the linked list for the jobid, set the uuid
4. Receive hangup etc (with uuid) remove from linked list

The problem is that sometimes I receive the hangup with the uuid before 
I'm told what the match between the jobid and the uuid are.

Any ideas?

-- 
Cheers,

Matt Riddell
Director
_______________________________________________

http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)



From Russell.Mosemann at cune.org  Thu Oct  1 14:19:14 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Thu, 1 Oct 2009 21:19:14 -0000
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <191c3a030910011217o6aed6b2ag1ded3d26a62430dc@mail.gmail.com>
Message-ID: <20091001211914.6533641A1FA@mail.cune.org>

Anthony Minessale <anthony.minessale at gmail.com> said:

> You might want to try the ozmod_pri instead of ozmod_isdn until the new
> revision of ozmod_isdn is published into the source tree.

libpri took care of the problem with the transfer. Now, someone can call
into FS from the Hicomm and then transfer the call to another extension
on the Hicomm.

A call from FS to the Hicomm still transfers to the switchboard. I'm not
seeing any CID/ANI on the X-Lite. It shows up as "OpenZAP" and "1". Is
there something like ngrep for the D channel of a PRI? It would be nice
to see what data is being sent between FS and the Hicom.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!




From siniypin at gmail.com  Thu Oct  1 14:44:22 2009
From: siniypin at gmail.com (RobertT)
Date: Fri, 2 Oct 2009 01:44:22 +0400
Subject: [Freeswitch-users] conference participant from behind NAT
In-Reply-To: <2160023e0910011403r5e9d0bb0ib5699f0523aaa606@mail.gmail.com>
References: <2160023e0909290132p2a5d0b62jbcb7d37625686866@mail.gmail.com>
	<681a20520909290618i2fbe7199rb18ff32933bba952@mail.gmail.com>
	<2160023e0909290634h3102d81bj9e3dd3d4d56f43e0@mail.gmail.com>
	<A3EEB6BE-19E2-49F3-8E41-FC74307A18E2@jerris.com>
	<2160023e0910011403r5e9d0bb0ib5699f0523aaa606@mail.gmail.com>
Message-ID: <2160023e0910011444y24448djede1bc24fc6d9c1@mail.gmail.com>

And here is a short piece of log from the server side:
...
nua(): refersh session after 62 seconds (in [55..65])...
send INVITE ...
rcv OK...
send ACK...
rcv BYE...

I see now that sdp for natted client has additional lines in OK response
compared to client with public ip.
Session-Expires: 120;refresher=uas
Min-SE: 120

How come that they differs? And how do I resolve this situation? Should
client handle these refresher messages normally?

Best regards, Robert.
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From anthony.minessale at gmail.com  Thu Oct  1 14:50:04 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 1 Oct 2009 16:50:04 -0500
Subject: [Freeswitch-users] bgapi jobid to uuid
In-Reply-To: <4AC520E0.4020507@venturevoip.com>
References: <4AC520E0.4020507@venturevoip.com>
Message-ID: <191c3a030910011450r1e0c1645r395c2024a25dcde9@mail.gmail.com>

if you pick your own job-uuid you can set it in the originate too

send

job-uuid: 1234

in your bgapi event

and

{job_uuid=1234}sofia/internal/foo at bar.com

in your dial string.

*shrug* i am not sure exactly what the goal is so maybe it's not a useful
suggestion...



On Thu, Oct 1, 2009 at 4:36 PM, Matt Riddell <lists at venturevoip.com> wrote:

> Hi,
>
> I decided to go with a linked list for current channels and maintain
> that through state changes.
>
> So, basically it works like this:
>
> 1. Originate a call using bgapi (we get a jobid in response)
> 2. Receive an event with the jobid and a uuid
> 3. Lookup the linked list for the jobid, set the uuid
> 4. Receive hangup etc (with uuid) remove from linked list
>
> The problem is that sometimes I receive the hangup with the uuid before
> I'm told what the match between the jobid and the uuid are.
>
> Any ideas?
>
> --
> Cheers,
>
> Matt Riddell
> Director
> _______________________________________________
>
> http://www.venturevoip.com/news.php (Daily Asterisk News)
> http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
> http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Thu Oct  1 14:52:39 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 1 Oct 2009 16:52:39 -0500
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <20091001211914.6533641A1FA@mail.cune.org>
References: <191c3a030910011217o6aed6b2ag1ded3d26a62430dc@mail.gmail.com>
	<20091001211914.6533641A1FA@mail.cune.org>
Message-ID: <191c3a030910011452r2c163a83g7b5d0988d54facd7@mail.gmail.com>

there was a feature to generate a pcap from the debug logs but i forgot who
posted it.


On Thu, Oct 1, 2009 at 4:19 PM, <Russell.Mosemann at cune.org> wrote:

> Anthony Minessale <anthony.minessale at gmail.com> said:
>
> > You might want to try the ozmod_pri instead of ozmod_isdn until the new
> > revision of ozmod_isdn is published into the source tree.
>
> libpri took care of the problem with the transfer. Now, someone can call
> into FS from the Hicomm and then transfer the call to another extension
> on the Hicomm.
>
> A call from FS to the Hicomm still transfers to the switchboard. I'm not
> seeing any CID/ANI on the X-Lite. It shows up as "OpenZAP" and "1". Is
> there something like ngrep for the D channel of a PRI? It would be nice
> to see what data is being sent between FS and the Hicom.
>
> --
> Russell Mosemann
>
>
>
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From lists at venturevoip.com  Thu Oct  1 15:02:41 2009
From: lists at venturevoip.com (Matt Riddell)
Date: Fri, 02 Oct 2009 11:02:41 +1300
Subject: [Freeswitch-users] bgapi jobid to uuid
In-Reply-To: <191c3a030910011450r1e0c1645r395c2024a25dcde9@mail.gmail.com>
References: <4AC520E0.4020507@venturevoip.com>
	<191c3a030910011450r1e0c1645r395c2024a25dcde9@mail.gmail.com>
Message-ID: <4AC52701.4020504@venturevoip.com>

On 2/10/09 10:50 AM, Anthony Minessale wrote:
> if you pick your own job-uuid you can set it in the originate too
>
> send
>
> job-uuid: 1234
>
> in your bgapi event
>
> and
>
> {job_uuid=1234}sofia/internal/foo at bar.com <mailto:foo at bar.com>
>
> in your dial string.
>
> *shrug* i am not sure exactly what the goal is so maybe it's not a
> useful suggestion...

Does the job-uuid get used as the actual call uuid?  The problem is that 
the uuid for the call changes once it's answered (or hungup etc) and 
that the mapping doesn't come till after occasionally.

I'm getting the job uuid no problem, but it's the replacement of the 
job-uuid with a call-uuid that is the issue.

If this doesn't make too much sense I can post an example.

-- 
Cheers,

Matt Riddell
Director
_______________________________________________

http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)



From anthony.minessale at gmail.com  Thu Oct  1 15:36:14 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 1 Oct 2009 17:36:14 -0500
Subject: [Freeswitch-users] bgapi jobid to uuid
In-Reply-To: <4AC52701.4020504@venturevoip.com>
References: <4AC520E0.4020507@venturevoip.com>
	<191c3a030910011450r1e0c1645r395c2024a25dcde9@mail.gmail.com>
	<4AC52701.4020504@venturevoip.com>
Message-ID: <191c3a030910011536p403e4681u9375b53f594616a4@mail.gmail.com>

if you make your own uuids you could set them in the originate string with
{origination_uuuid=foo}

where foo is a real uuid.
if you have no other way to make them you can ask FS for one with the
create_uuid api call.


On Thu, Oct 1, 2009 at 5:02 PM, Matt Riddell <lists at venturevoip.com> wrote:

> On 2/10/09 10:50 AM, Anthony Minessale wrote:
> > if you pick your own job-uuid you can set it in the originate too
> >
> > send
> >
> > job-uuid: 1234
> >
> > in your bgapi event
> >
> > and
> >
> > {job_uuid=1234}sofia/internal/foo at bar.com <mailto:foo at bar.com>
> >
> > in your dial string.
> >
> > *shrug* i am not sure exactly what the goal is so maybe it's not a
> > useful suggestion...
>
> Does the job-uuid get used as the actual call uuid?  The problem is that
> the uuid for the call changes once it's answered (or hungup etc) and
> that the mapping doesn't come till after occasionally.
>
> I'm getting the job uuid no problem, but it's the replacement of the
> job-uuid with a call-uuid that is the issue.
>
> If this doesn't make too much sense I can post an example.
>
> --
> Cheers,
>
> Matt Riddell
> Director
> _______________________________________________
>
> http://www.venturevoip.com/news.php (Daily Asterisk News)
> http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
> http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Thu Oct  1 16:33:53 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 1 Oct 2009 16:33:53 -0700
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <20091001211914.6533641A1FA@mail.cune.org>
References: <191c3a030910011217o6aed6b2ag1ded3d26a62430dc@mail.gmail.com>
	<20091001211914.6533641A1FA@mail.cune.org>
Message-ID: <87f2f3b90910011633o1f561b76vf77e8e152d21e7a4@mail.gmail.com>

On Thu, Oct 1, 2009 at 2:19 PM, <Russell.Mosemann at cune.org> wrote:

> Anthony Minessale <anthony.minessale at gmail.com> said:
>
> > You might want to try the ozmod_pri instead of ozmod_isdn until the new
> > revision of ozmod_isdn is published into the source tree.
>
> libpri took care of the problem with the transfer. Now, someone can call
> into FS from the Hicomm and then transfer the call to another extension
> on the Hicomm.
>
> A call from FS to the Hicomm still transfers to the switchboard. I'm not
> seeing any CID/ANI on the X-Lite. It shows up as "OpenZAP" and "1". Is
> there something like ngrep for the D channel of a PRI? It would be nice
> to see what data is being sent between FS and the Hicom.
>
> I believe the "OpenZAP" and "1" are coming from your conf file:
openzap.conf
[span zt PRI_1]
name => OpenZAP
number => 1

As far as debugging with ozmod_libpri I believe the syntax is:
oz libpri debug 1 all

It will do a traditional libpri-style debug, just like "pri debug span 1" in
Asterisk.
-MC
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From msc at freeswitch.org  Thu Oct  1 16:41:02 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 1 Oct 2009 16:41:02 -0700
Subject: [Freeswitch-users] bgapi jobid to uuid
In-Reply-To: <4AC520E0.4020507@venturevoip.com>
References: <4AC520E0.4020507@venturevoip.com>
Message-ID: <87f2f3b90910011641l7f8f90c6td19ad8f70fbf914b@mail.gmail.com>

On Thu, Oct 1, 2009 at 2:36 PM, Matt Riddell <lists at venturevoip.com> wrote:

> Hi,
>
> I decided to go with a linked list for current channels and maintain
> that through state changes.
>
> So, basically it works like this:
>
> 1. Originate a call using bgapi (we get a jobid in response)
> 2. Receive an event with the jobid and a uuid
> 3. Lookup the linked list for the jobid, set the uuid
> 4. Receive hangup etc (with uuid) remove from linked list
>
> The problem is that sometimes I receive the hangup with the uuid before
> I'm told what the match between the jobid and the uuid are.
>
> Any ideas?
>

If I may ask, what's the application? Are you working on Vicidial-ish stuff
for FreeSWITCH? Also, you might want to call the FreeSWITCH conference
tomorrow. We have some Q&A time with the FS devs and this kind of thing
might work better in realtime rather than email threads.
-MC
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From lists at venturevoip.com  Thu Oct  1 16:46:01 2009
From: lists at venturevoip.com (Matt Riddell)
Date: Fri, 02 Oct 2009 12:46:01 +1300
Subject: [Freeswitch-users] bgapi jobid to uuid
In-Reply-To: <191c3a030910011536p403e4681u9375b53f594616a4@mail.gmail.com>
References: <4AC520E0.4020507@venturevoip.com>	<191c3a030910011450r1e0c1645r395c2024a25dcde9@mail.gmail.com>	<4AC52701.4020504@venturevoip.com>	<191c3a030910011536p403e4681u9375b53f594616a4@mail.gmail.com>
Message-ID: <4AC53F39.4030302@venturevoip.com>

On 2/10/09 11:36 AM, Anthony Minessale wrote:
> if you make your own uuids you could set them in the originate string with
> {origination_uuuid=foo}
>
> where foo is a real uuid.
> if you have no other way to make them you can ask FS for one with the
> create_uuid api call.

Awesome, thanks - will give it a whirl over the weekend.

-- 
Cheers,

Matt Riddell
Director
_______________________________________________

http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)



From jmesquita at freeswitch.org  Thu Oct  1 20:14:12 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 2 Oct 2009 00:14:12 -0300
Subject: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
	ToSubscribing Phones
In-Reply-To: <9B5214D274044021A57C6857844C10BA@greyhawk.tonecommander.com>
References: <B4B4D2747EEC4852B41A7D8D2DA36006@greyhawk.tonecommander.com>
	<191c3a030910010935s69e65569v76f4d26503a94d96@mail.gmail.com>
	<9B5214D274044021A57C6857844C10BA@greyhawk.tonecommander.com>
Message-ID: <ebbc2d550910012014ic655d84tbf000c084a98e51d@mail.gmail.com>

Piece of advice, don't ask, just do it. ;)

jmesquita

On Thu, Oct 1, 2009 at 3:29 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>  If you have time to take a look, I could put a trace in the pastebin?
>
> Jerry
>
>  ------------------------------
> *From:* Jerry Richards [mailto:jerry.richards at teotech.com]
> *Sent:* Thursday, October 01, 2009 10:29 AM
> *To:* 'freeswitch-users at lists.freeswitch.org'
> *Subject:* RE: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
> ToSubscribing Phones
>
>  I am using two Bria Professional Version 2.5.4 Build 54835 softphones.
>
> Thanks,
> Jerry
>
>  ------------------------------
> *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com]
> *Sent:* Thursday, October 01, 2009 9:36 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
> ToSubscribing Phones
>
> which phone is it,
> we tested it with eyebeam and it appears to work for us.
>
>
> On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards <jerry.richards at teotech.com
> > wrote:
>
>>
>> By the way, I see the following lines at the FS console, which might be a
>> clue as to why this is happening.  Could someone point me toward what
>> might
>> cause this?  I set the "manage-presence" parameter to "true" in each XML
>> file where I saw it defined.
>>
>> [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal)
>> [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
>>   ...
>> [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6)
>> [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
>>   ...
>> [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external)
>> [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
>>   ...
>> [WARNING] sofia_presence.c:565  192.168.72.38 is an alias, skipping
>>
>>
>> Best Regards,
>> Jerry
>>
>>
>> -----Original Message-----
>> From: Jerry Richards [mailto:jerry.richards at teotech.com]
>> Sent: Wednesday, September 30, 2009 9:12 AM
>> To: 'freeswitch-users at lists.freeswitch.org'
>> Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones
>>
>> I have two phones configured to subscribe to each other's presence status.
>> When I change the presence status in one phone, I see the SIP PUBLISH
>> message going to FS, but I don't see FS relaying that presence status to
>> the
>> subscribing phone.  Does anyone know why?
>>
>> Best Regards,
>> Jerry
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dule.maillist at gmail.com  Thu Oct  1 20:59:54 2009
From: dule.maillist at gmail.com (Dan Le)
Date: Thu, 1 Oct 2009 23:59:54 -0400
Subject: [Freeswitch-users] Minimum audio length for uuid_record
In-Reply-To: <20091001033138.GA18723@jdc.jasonjgw.net>
References: <914fc92a0909301944l116455b3tc25009fa9ea499d5@mail.gmail.com>
	<20091001033138.GA18723@jdc.jasonjgw.net>
Message-ID: <914fc92a0910012059v16bd018bpca397ec6c7b339d8@mail.gmail.com>

Thanks, I think I found the thread you were referring to
("[Freeswitch-users] session record does not for very short calls"), which
doesn't seem to be a solution for my situation. However, I did find that
using session:recordFile() didn't delete the file if it was really short.
And following that thread lead me to an interesting channel variable that
could be useful to us, record_ms, but having trouble getting it to reflect
the audio length. I can see the variable when printing data from the info
application, but it's always 0. My snippet of code is very simple:
if session:ready() then
   session:recordFile("C:/Temp/recording.wav", 30, 600, 6);

   local record_length = session:getVariable("record_ms");
   freeswitch.consoleLog("INFO", "Recorded a " .. record_length .. " ms
file.\n");
end

Another side question, the silence secs parameter (in this example, 6), is
that 6s silence hits during the entire recording session or 6s of
consecutive silence? From a few tests, it seems to be the former, but just
wanted to verify, something that would make a good addition to the wiki.

Dan


On Wed, Sep 30, 2009 at 11:31 PM, Jason White <jason at jasonjgw.net> wrote:

> Dan Le <dule.maillist at gmail.com> wrote:
> > We're running into a problem with the minimum file size when recording
> using
> > uuid_record. It seems if the audio is too short it deletes the audio
> file.
> > Is there a way to override that?
>
> Yes. It was discussed on the list recently. I suggest searching the list
> archives. Someone may have documented it on the wiki by now also.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From irmatov at gmail.com  Fri Oct  2 01:10:39 2009
From: irmatov at gmail.com (Timur Irmatov)
Date: Fri, 2 Oct 2009 13:10:39 +0500
Subject: [Freeswitch-users] internal & external ip addresses of freeswitch
Message-ID: <241d382f0910020110l5f3728e7k3e521e81353b5d6f@mail.gmail.com>

Hi.

We have a local network 192.168.1.0/24, where all the users are. Out
FreeSWITCH server is connected to this network, and has ip address
192.168.1.242. Through different network card it is connected to
external gateway, and has address 172.16.12.11 in this network.

I set up a test client with softphone. When incoming call is
deliviered to this client, call is set up normally, but client can't
hang it up. It sends BYE to external address - 172.16.12.11 - which is
not reachable from the client. It seems this address is coming from
Contact: field in INVITE that FreeSWITCH sends:

U 192.168.1.242:5060 -> 192.168.1.34:37169
INVITE sip:100 at 192.168.1.34:37169 SIP/2.0.
Via: SIP/2.0/UDP 172.16.12.11;rport;branch=z9hG4bKrvp6jm3myyaaF.
Max-Forwards: 70.
From: "FreeSWITCH" <sip:0000000000 at 172.16.12.11>;tag=v817pS9c6v6Fe.
To: <sip:100 at 192.168.1.34:37169>.
Call-ID: 797bd088-29cd-122d-9b93-0060979d54c5.
CSeq: 121117089 INVITE.
Contact: <sip:mod_sofia at 172.16.12.11:5060>.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14898.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 267.
Remote-Party-ID: "FreeSWITCH"
<sip:0000000000 at 172.16.12.11>;party=calling;screen=yes;privacy=off.

What should I tweak in freeswitch to change this behaviour?


-- 
Timur Irmatov, xmpp:irmatov at jabber.ru



From siniypin at gmail.com  Fri Oct  2 01:32:26 2009
From: siniypin at gmail.com (RobertT)
Date: Fri, 2 Oct 2009 12:32:26 +0400
Subject: [Freeswitch-users] conference participant from behind NAT
In-Reply-To: <2160023e0910011444y24448djede1bc24fc6d9c1@mail.gmail.com>
References: <2160023e0909290132p2a5d0b62jbcb7d37625686866@mail.gmail.com>
	<681a20520909290618i2fbe7199rb18ff32933bba952@mail.gmail.com>
	<2160023e0909290634h3102d81bj9e3dd3d4d56f43e0@mail.gmail.com>
	<A3EEB6BE-19E2-49F3-8E41-FC74307A18E2@jerris.com>
	<2160023e0910011403r5e9d0bb0ib5699f0523aaa606@mail.gmail.com>
	<2160023e0910011444y24448djede1bc24fc6d9c1@mail.gmail.com>
Message-ID: <2160023e0910020132i3be2d3a2o615cb18dba992878@mail.gmail.com>

Hi folks!

Suddenly I found this
http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-February/002015.htmltopic
and that explains a lot.
>From there I see that sofia sends refresher messages for NATed client in
order to check if it still alive.
It means I have problems in my client. Sorry for the mess.

Cheers, Robert.
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From mcampbellsmith at gmail.com  Fri Oct  2 01:58:58 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Fri, 2 Oct 2009 18:58:58 +1000
Subject: [Freeswitch-users] [ERR] mod_dingaling.c:980 Stun Failed!
In-Reply-To: <33c87fa30909290415y178b6506x120a6e5c39839ce0@mail.gmail.com>
References: <33c87fa30909290415y178b6506x120a6e5c39839ce0@mail.gmail.com>
Message-ID: <33c87fa30910020158q7589e18fj53bd4cc95ff3926f@mail.gmail.com>

Anyone have this issue?

On Tue, Sep 29, 2009 at 9:15 PM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> Hi!
>
> I have just started to use dingaling again, and noticed I constantly
> get a stun error.
>
> 2009-09-29 21:11:03.175002 [ERR] mod_dingaling.c:980 Stun Failed!
> stun.fwdnet.net:3478 [Remote Address Error!]
>
> I have tried with stun.freeswitch.org and stun.fwdnet.net stun servers
> and keep getting this error with dingaling. ?I have no problems with
> inbound sip calls, so I don't think ?its the actual stun server.
>
> Has anyone else seen this? ?I am using: FreeSWITCH Version 1.0.trunk (14952)
>
> Thanks!
>



From shaheryarkh at googlemail.com  Fri Oct  2 03:00:03 2009
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Fri, 2 Oct 2009 16:00:03 +0600
Subject: [Freeswitch-users] [ERR] mod_dingaling.c:980 Stun Failed!
In-Reply-To: <33c87fa30910020158q7589e18fj53bd4cc95ff3926f@mail.gmail.com>
References: <33c87fa30909290415y178b6506x120a6e5c39839ce0@mail.gmail.com>
	<33c87fa30910020158q7589e18fj53bd4cc95ff3926f@mail.gmail.com>
Message-ID: <b16156850910020300x6cd7dd9cm2348942b04c79030@mail.gmail.com>

Yes, i had same problem, then i changed stun server to one of our own
servers. You may try some of public stun servers listed on below link,

http://www.voip-info.org/wiki/view/STUN

Thank you.


On Fri, Oct 2, 2009 at 2:58 PM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> Anyone have this issue?
>
> On Tue, Sep 29, 2009 at 9:15 PM, Mark Campbell-Smith
> <mcampbellsmith at gmail.com> wrote:
> > Hi!
> >
> > I have just started to use dingaling again, and noticed I constantly
> > get a stun error.
> >
> > 2009-09-29 21:11:03.175002 [ERR] mod_dingaling.c:980 Stun Failed!
> > stun.fwdnet.net:3478 [Remote Address Error!]
> >
> > I have tried with stun.freeswitch.org and stun.fwdnet.net stun servers
> > and keep getting this error with dingaling.  I have no problems with
> > inbound sip calls, so I don't think  its the actual stun server.
> >
> > Has anyone else seen this?  I am using: FreeSWITCH Version 1.0.trunk
> (14952)
> >
> > Thanks!
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
________________________________________________________
|
                     |
| FATAL ERROR                                                           ---
O X |
|_______________________________________________________|
|                        You have moved the mouse.
 |
| Windows must be restarted for the changes to take effect.   |
|                                        <OK>
               |
####################################/


Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From aep.lists at it46.se  Fri Oct  2 04:11:23 2009
From: aep.lists at it46.se (Alberto Escudero)
Date: Fri, 2 Oct 2009 13:11:23 +0200
Subject: [Freeswitch-users] How to limit the number of incoming+outgoing
 calls via specific gateway?
In-Reply-To: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>
References: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>
Message-ID: <e403cddca1c90c83ce4fb6f85b026e63.squirrel@correo.nodo50.org>


You can use the api and check that the channel is occupied with "show
channels"?
You can write a small javascript that checks if the channel is occupied by
means of session.execute api.

/aep
-- 
Stopping junk mailers is good for the environment

> My SIP provider allows only one call (incoming or outgoing) via one
> SIP account. For FreeSWITCH I have configured it as public DID
> extension and outgoing gateway. Now I would like to transfer to
> another gw (or generate "limit exceded") when one tries to place an
> outgoing call while incoming call is in progress. How tho do that?
> Limiting the number of outgoing calls is easy (mod_limit), but how to
> take into account incoming one?
>
> - Dmitry Bely
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>





From Russell.Mosemann at cune.org  Fri Oct  2 04:48:35 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Fri, 2 Oct 2009 11:48:35 -0000
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <87f2f3b90910011633o1f561b76vf77e8e152d21e7a4@mail.gmail.com>
Message-ID: <20091002114835.4C0FA419B7A@mail.cune.org>

Michael Collins <msc at freeswitch.org> said:

> > I believe the "OpenZAP" and "1" are coming from your conf file:
> openzap.conf
> [span zt PRI_1]
> name => OpenZAP
> number => 1

That is correct. If that information is removed, then X-Lite displays

FreeSWITCH
[Other: 0000000000]

Are there any variables to set to get CID, or is OpenZap supposed to be
filling that in?

> As far as debugging with ozmod_libpri I believe the syntax is:
> oz libpri debug 1 all

That works. Now, I have to figure out what some of these abbreviations mean.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!




From tculjaga at gmail.com  Fri Oct  2 05:05:55 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 2 Oct 2009 14:05:55 +0200
Subject: [Freeswitch-users] configure FS: config.status: error: cannot find
	input file: Makefile.in
Message-ID: <65d96fc80910020505t48a6237ao19884c23fedf228d@mail.gmail.com>

hello,
i just got the last trunk and tried to compile it on one of my development
machines... Well configure fails on tiff-3.8.2 where it is unable to find
Makefile.in ... Can someone advice?



checking if g++ static flag -static works... yes
checking if g++ supports -c -o file.o... yes
checking if g++ supports -c -o file.o... (cached) yes
checking whether the g++ linker (/usr/bin/ld) supports shared libraries...
yes
checking dynamic linker characteristics... GNU/Linux ld.so
checking how to hardcode library paths into programs... immediate
checking for OpenGL Utility library... no
checking for GLUT library... no
configure: creating ./config.status
config.status: error: cannot find input file: Makefile.in



tculjaga at subZero:~/freeswitch-trunk/libs/tiff-3.8.2$ ls -l
total 2224
-rw-r--r--  1 tculjaga tculjaga  23741 2009-10-02 13:19 acinclude.m4
-rw-r--r--  1 tculjaga tculjaga 316978 2009-10-02 13:28 aclocal.m4
-rwxr-xr-x  1 tculjaga tculjaga    121 2009-10-02 13:19 autogen.sh
-rw-r--r--  1 tculjaga tculjaga 124047 2009-10-02 13:19 ChangeLog
drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:28 config
-rw-r--r--  1 tculjaga tculjaga 103926 2009-10-02 14:00 config.log
-rwxr-xr-x  1 tculjaga tculjaga  73065 2009-10-02 14:00 config.status
-rwxr-xr-x  1 tculjaga tculjaga 740145 2009-10-02 13:28 configure
-rw-r--r--  1 tculjaga tculjaga  20492 2009-10-02 13:19 configure.ac
-rwxr-xr-x  1 tculjaga tculjaga     56 2009-10-02 13:19 configure.gnu
-rwxr-xr-x  1 tculjaga tculjaga 737794 2009-10-02 13:57 configure.lineno
drwxr-xr-x 16 tculjaga tculjaga   4096 2009-10-02 13:19 contrib
-rw-r--r--  1 tculjaga tculjaga   1146 2009-10-02 13:19 COPYRIGHT
-rw-r--r--  1 tculjaga tculjaga   1570 2009-10-02 13:19 HOWTO-RELEASE
drwxr-xr-x  5 tculjaga tculjaga   4096 2009-10-02 13:19 html
drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:28 libtiff
drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 m4
-rw-r--r--  1 tculjaga tculjaga   1908 2009-10-02 13:19 Makefile.am
-rw-r--r--  1 tculjaga tculjaga   1724 2009-10-02 13:19 Makefile.vc
drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 man
-rw-r--r--  1 tculjaga tculjaga   6270 2009-10-02 13:19 nmake.opt
drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 port
-rw-r--r--  1 tculjaga tculjaga   2363 2009-10-02 13:19 README
-rw-r--r--  1 tculjaga tculjaga      9 2009-10-02 13:19 RELEASE-DATE
-rw-r--r--  1 tculjaga tculjaga   5893 2009-10-02 13:19 SConstruct
drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 test
-rw-r--r--  1 tculjaga tculjaga    433 2009-10-02 13:19 TODO
drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 tools
-rw-r--r--  1 tculjaga tculjaga      6 2009-10-02 13:19 VERSION
tculjaga at subZero:~/freeswitch-trunk/libs/tiff-3.8.2$
tculjaga at subZero:~/freeswitch-trunk/libs/tiff-3.8.2$
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From tculjaga at gmail.com  Fri Oct  2 05:32:31 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 2 Oct 2009 14:32:31 +0200
Subject: [Freeswitch-users] How to limit the number of incoming+outgoing
	calls via specific gateway?
In-Reply-To: <e403cddca1c90c83ce4fb6f85b026e63.squirrel@correo.nodo50.org>
References: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>
	<e403cddca1c90c83ce4fb6f85b026e63.squirrel@correo.nodo50.org>
Message-ID: <65d96fc80910020532i4ba33353g7ac141624a4bb89e@mail.gmail.com>

what if you are running some huge traffic e.g. 2000 calls with media?

a typical application for that is an IVR system handling several different
services. I'd like to "dedicate" some capacity for inbound on per service
basis.


e.g.

DID 10001 limit to 500 calls
DID 10002 limit to 400 calls
DID 10003 limit to 100 calls
DID 10005 limit to 1000 calls


This will be a total of 2000 calls.


don't you think js is simply too weak for that? It should cont
calls/channels, brake counts per service/DID and update the counters on
every call hit.




in the DP you would have something like this for every DID:


<include>
  <extension name="MY_DID_NUM">
    <condition field="destination_number" expression="^MY_DID_NUMBER$">
    <action application="set" data="SERVICE_LIMIT=500"/>

      <!--
             count number of active channels going towards MY_DID_NUMBER and
store it into COUNT_MY_DID_NUMBER
      -->

      <action application="transfer" data="do_MY_SERVICE XML public"/>
    </condition>
  </extension>
</include>



<include>
  <extension name="SERVICE1">
    <condition field="destination_number" expression="^do_MY_SERVICE$"/>
    <condition field="${COUNT_MY_DID_NUMBER}" expression="^SERVICE_LIMIT$">

       <!-- do your service here -->
      <action application="playback" data="I_Accept_Your_Call.wav"/>
      <action application="hangup" data="NORMAL_CLEARING"/>

       <!-- do your limitation here -->
      <anti-action application="respond" data="403 Forbidden"/> <= put your
response here!

    </condition>
  </extension>
</include>





but the question is ... how powerful a JavaScript can be? Will it be enough
to handle that load?



Tihomir.





On Fri, Oct 2, 2009 at 1:11 PM, Alberto Escudero <aep.lists at it46.se> wrote:

>
> You can use the api and check that the channel is occupied with "show
> channels"?
> You can write a small javascript that checks if the channel is occupied by
> means of session.execute api.
>
> /aep
> --
> Stopping junk mailers is good for the environment
>
> > My SIP provider allows only one call (incoming or outgoing) via one
> > SIP account. For FreeSWITCH I have configured it as public DID
> > extension and outgoing gateway. Now I would like to transfer to
> > another gw (or generate "limit exceded") when one tries to place an
> > outgoing call while incoming call is in progress. How tho do that?
> > Limiting the number of outgoing calls is easy (mod_limit), but how to
> > take into account incoming one?
> >
> > - Dmitry Bely
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tculjaga at gmail.com  Fri Oct  2 05:38:30 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 2 Oct 2009 14:38:30 +0200
Subject: [Freeswitch-users] Dialplan Issue
In-Reply-To: <35b355e90910010918w7a01c4d1l7bc8b7986d2a3434@mail.gmail.com>
References: <35b355e90910010727v2efcbff6p97a0a883857818e4@mail.gmail.com>
	<191c3a030910010755l387faba8x1d8f9bbf51c8ffa1@mail.gmail.com>
	<35b355e90910010901j63a36531m7550eba26564f01@mail.gmail.com>
	<35b355e90910010918w7a01c4d1l7bc8b7986d2a3434@mail.gmail.com>
Message-ID: <65d96fc80910020538s51c47425vf9142d4fe47e16bb@mail.gmail.com>

anyhow, this is how it works for me!




<include>
  <context name="public">

      <extension name="LNP">
                <condition field="destination_number"
expression="(^300000)(.*)">
                      <action application="lnp_getprefix" data="in $2, out
reroutingalias"/>
                        <action application="redirect" data="sip:${
reroutingalias}@10.4.13.11:5060"/>
                </condition>
        </extension>


        <extension name="LBS">
                <condition field="destination_number"
expression="(^300010)(.*)">
                        <action application="lbs_getpublicphone" data="in
${caller_id_number}, in $2, out reroutingalias"/>
                        <action application="redirect" data="sip:${
reroutingalias}@10.4.13.11:5060"/>
              </condition>
        </extension>

        <extension name="CPS">
                <condition field="destination_number"
expression="(^300020)(.*)">
                        <action application="cps_verifyphone" data="in
${caller_id_number}, in $2, out radiusacc"/>
                 </condition>
              <condition field="radiusacc" expression="1">
                            <action application="redirect" data="sip:${
caller_id_number}@10.4.13.11:5060"/>
                            <anti-action application="respond" data="403
Forbidden"/>
                </condition>
        </extension>



   <extension name="ServiceLookup">
      <condition field="destination_number" expression="(^300030)(.*)">
         <action application="lookup_service_destination" data="in
${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, in $
1, in ${network_addr}:5060, out red_contact, out authResult"/>
         <action application="log" data="INFO ########################
ServiceLookup ########################\n"/>
         <action application="log" data="INFO ########################
contact = '${red_contact}' ##############\n"/>
         <action application="log" data="INFO ########################
CallerNum = '${caller_id_number:6:16}' ##########\n"/>
         <action application="log" data="INFO ########################
RADIUS auth = '${authResult}' ##########\n"/>

         <action application="execute_extension" data="doRedirect XML
public"/>
        </condition>
   </extension>


   <extension name="doRedirect">
      <condition field="destination_number" expression="^doRedirect$"/>
      <condition field="${authResult}" expression="^0$|^60$">
         <action application="log" data="INFO ########################
RADIUS auth OK!!!' ##########\n"/>
         <action application="redirect" data="${red_contact}"/>
         <anti-action application="log" data="INFO ########################
RADIUS auth NOK!! ##########\n"/>
         <anti-action application="respond" data="403 Forbidden"/>
      </condition>

   </extension>


  </context>
</include>

On Thu, Oct 1, 2009 at 6:18 PM, Shelby Ramsey <sicfslist at gmail.com> wrote:

> Just to confirm ... works like a champ.
>
> Thanks again!!!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From orien at tx.rr.com  Thu Oct  1 18:11:40 2009
From: orien at tx.rr.com (Orien Love)
Date: Thu, 01 Oct 2009 20:11:40 -0500
Subject: [Freeswitch-users] New to freeswitch and have a few questions
Message-ID: <4AC5534C.6050202@tx.rr.com>

Hello Everybody,
    I am new to freeswitch, so forgive me if I ask stupid questions. I 
am planning a test setup consisting of:
    1 - Pfsense router with the freeswitch package installed.
    1 - Cisco WS-C3524-PWR-XL-EN 24 FE Switch, POE for the phones.
    1 - LINKSYS SPA3000 to connect to my existing land line and phones.
    2 - POLYCOM SOUNDPOINT IP601 SIP IP PHONEs

The first question I have, Are the IP601 phones supported? The wiki 
lists 320, 431, 501, 550, 650 but not the 601.

Second, is there a place that helps a person new to the IP phone world 
learn what is needed to set up a PBX using freeswitch at a small office?

Finally is my test setup a good one? is there something I am missing or 
that I need to get the learning process started, I have found in the 
past, with a little information and a test system, I can learn what I am 
doing by breaking and fixing the test bed.

Thanks for your time
Orien



From msc at freeswitch.org  Fri Oct  2 09:00:45 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 2 Oct 2009 09:00:45 -0700
Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Starting,
	Please Call In!
Message-ID: <87f2f3b90910020900k20f65141rcb2414402ce388a1@mail.gmail.com>

Hey folks, the weekly conference call is starting. Please see the agenda for
instructions on dialing:

http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_02

Looking forward to speaking with you all!

-Michael
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From msc at freeswitch.org  Fri Oct  2 09:08:03 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 2 Oct 2009 09:08:03 -0700
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <20091002114835.4C0FA419B7A@mail.cune.org>
References: <87f2f3b90910011633o1f561b76vf77e8e152d21e7a4@mail.gmail.com>
	<20091002114835.4C0FA419B7A@mail.cune.org>
Message-ID: <87f2f3b90910020908p1aa5f5d4o8b69894cd4a58bef@mail.gmail.com>

On Fri, Oct 2, 2009 at 4:48 AM, <Russell.Mosemann at cune.org> wrote:

> Michael Collins <msc at freeswitch.org> said:
>
> > > I believe the "OpenZAP" and "1" are coming from your conf file:
> > openzap.conf
> > [span zt PRI_1]
> > name => OpenZAP
> > number => 1
>
> That is correct. If that information is removed, then X-Lite displays
>
> FreeSWITCH
> [Other: 0000000000]
>

do something like:
name => XYZ Corp
number => 8005551212

>
> Are there any variables to set to get CID, or is OpenZap supposed to be
> filling that in?
>
> > As far as debugging with ozmod_libpri I believe the syntax is:
> > oz libpri debug 1 all
>
> That works. Now, I have to figure out what some of these abbreviations
> mean.
>
> Welcome to the wacky world of Q931. The wiki has info:
http://wiki.freeswitch.org/wiki/ISDN:_Integrated_Services_Digital_Network

-MC
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From mgende at gendesign.com  Fri Oct  2 09:10:30 2009
From: mgende at gendesign.com (Michael Gende)
Date: Fri, 2 Oct 2009 11:10:30 -0500
Subject: [Freeswitch-users] New to freeswitch and have a few questions
In-Reply-To: <4AC5534C.6050202@tx.rr.com>
References: <4AC5534C.6050202@tx.rr.com>
Message-ID: <a87d43fd0910020910l5db88c5fv980c8c8fe312f380@mail.gmail.com>

Hey Orien,

I'm not using exactly your set up, but am using pfsense/FreeBSD. Since
you're using that, I assume you're going "dual homed". I've got a starter
guide that might help you out. If nothing else, I'd be interested in a
candid assessment of its usefulness or lack thereof, especially to a guy
like you.

I've included it here. Its all just text at the moment so be advised. Also
be advised that there's a lot of great information on the freeswitch site
and in this group. The goal of my document was so that someone just starting
would have to hunt a little less.

Hope its good for something, let me know either way, especially if you find
errors.

Regards,

Mike G.

On Thu, Oct 1, 2009 at 8:11 PM, Orien Love <orien at tx.rr.com> wrote:

> Hello Everybody,
>    I am new to freeswitch, so forgive me if I ask stupid questions. I
> am planning a test setup consisting of:
>    1 - Pfsense router with the freeswitch package installed.
>    1 - Cisco WS-C3524-PWR-XL-EN 24 FE Switch, POE for the phones.
>    1 - LINKSYS SPA3000 to connect to my existing land line and phones.
>    2 - POLYCOM SOUNDPOINT IP601 SIP IP PHONEs
>
> The first question I have, Are the IP601 phones supported? The wiki
> lists 320, 431, 501, 550, 650 but not the 601.
>
> Second, is there a place that helps a person new to the IP phone world
> learn what is needed to set up a PBX using freeswitch at a small office?
>
> Finally is my test setup a good one? is there something I am missing or
> that I need to get the learning process started, I have found in the
> past, with a little information and a test system, I can learn what I am
> doing by breaking and fixing the test bed.
>
> Thanks for your time
> Orien
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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-------------- next part --------------
FREESWITCH FOR DUMMIES: DUAL-HOMED HOST EXAMPLE mgende at gendesign.com September, 2009. 

 - Stuff One Ought to Know Before Starting 
 - set FS as a dual-homed host (seperate LAN and WAN interfaces), 
 - use the pre-set 1000-1019 extension numbers to configure and register your local SIP phones, 
 - set up FS to register with your SIP/PSTN provider, 
 - tell FS how to use that provider gateway for out-bound calls, 
 - Group your LAN phone extensions in a useful way, 
 - tell FS to ring all LAN SIP phones when a call comes in. 
 - Add a main Voice Mail account
 - Ring a Group, if no answer, transfer to Main VM 
 - Adding Another DID

STUFF ONE OUGHT TO KNOW BEFORE STARTING

PURPOSE OF THIS DOCUMENT

When I was getting my own FS up and running, I did a lot of Wiki reading (and still do). There's a lot of information (including a really good "Getting Started" guide) on the FreeSwitch site, but I nonetheless found myself looking here and there. So, I tried to put into one place some things I had found about the basics. While I keep this pretty straightforward, there is the assumption that one is comfortable with computer systems, networking, VoIP in general, XML, these kinds of things. This is not developer documentation, just "getting it going and trying to understand what's what." You'll get some of my opinions, keep 'em or chuck 'em, up to you. But, one will also learn some important basics about how to do some hopfully useful things, where things are, and how to get FS to act like a phone system (in this case). From there, the sky's the limit. The idea here is to provide a foothold to step up higher.

Note that this document has much in common with the "Getting Started" section on the FS website. This is more tuned to dual-homed setups, thus its creation. If you find it helpful, great. Find errors, point them out to me, please, and I'll correct them. If you hate it, then delete your copy.  

Oh, and any external static IP addresses, DIDs, etc, in this document are all pure fiction, so please don't waste any time trying to hack them, if you were so inclined (which I'm sure you were not).


PRE-INSTALL CONSIDERATIONS

THE COMPUTER:

Before one starts anything, ask yourself a few questions about how you would like FS to be deployed in your situation. One way, and a simple one at that, is to add a computer (or use an existing one) to your LAN. Just a box with, say, a Gb of RAM and a dual-core CPU will do it (see the FS site for more precise details on system requirements). FS comes set up to work this way without much fooling around. However, using a dual-homed host has its advantages. The largest one being lessening the traffic introduced to your LAN because a WAN interface is available. I find this a good thing and worth the extra effort. You may not. 


THE OS:

FS can run on a variety of OS's (the top three being MacOS, Windows, and Linux I believe). If you can work with Linux, BSD and the like, you'll be fine. The FS site has a list of ways to install on different distros, take your pick. We initially went the CentOS, which is a good Red Hat Linux compatable distro. For a computer with only one Ethernet internface on your LAN, that'd be my personal choice. However, if you use a computer with seperate LAN and WAN interfaces, I'd opt for PfSense (pfsense.org). This is a very nice FreeBSD system that is already a powerful firewall (and other great stuff). If you're connecting direct to the Internet, you'll need that. Further, with PfSense the instalation of the OS and Firewall, via install disk image they provide, is pretty straightforward. Once FreeBSD/PfSense is installed, one adds FS as a package (see the directions at pfsense.org) with just a couple of mouse clicks. While FS on PfSense has some handy interfaces to provide aids in administration, I tend to run and configure it from the command line just the same. I do so because I want to know and understand the FS directory structure and XML files that make things work.  


SIDEBAR: Want to run FS on a dual homed host? 

That's what we do. One Ethernet port is set for the LAN while the other connects direct to the Internet. We opt for a static WAN IP address. I personally suggest using a fixed IP if you're going to go this route. It will make configuring FS, as well as well as potentially registering with your SIP providor, easier in my opinion. Also, one will have less new LAN traffic with the advent of you new VoIP system as one would with a single-homed host. This is what pursuaded me. However, please note the FS scripts are pretty intellegent and can sense an IP change on your WAN port, letting the system know about it. However, for clarity and ease, largely my own, I'll go on about a static IP on the WAN and a seperate IP address for the LAN port of your FS box. 

NOTE: Just call your cable provider or DSL provider for info on getting a static IP if you'd like one. We have both DSL and Cable at my office, it was pretty simple to set up. These Internet providors, they'll do anything for money. If you have a T1 or better, you already have a static IP. Also note: going this way (dual-homed, fixed IP) gets your hands a little dirty, and I do mean "a little", with FS. That's something I think you'll want to do to understand more about how things work. It sure helped me.


MOVING ON: A FEW IMPORTANT FS COMMANDS FOR STARTERS

At this point, I'm assuming that you'll have the computer of your choice up and running with an OS and FS installed and waiting for your commands (if you don't, pretend). When I was getting FS up and running, there were a few concepts and commands that helped me get on my feet. So, for your edification, here they are:

To Start FS: The executable is in /usr/local/freeswitch/bin/freeswitch. Invoking it long-hand (or going to the ~/bin directory and typing ./freeswitch) will start FS in the foreground, attach to your terminal, and provide output and a command line. 

To Check FS Status: If the screen is full of "junk", hit enter. You should see a prompt something like this: "freeswitch at FreeSwitch.local>".  From there, you can tell FS to do things and ask about status.  You'll use commands like "sofia status", "sofia status profile internal", and "sofia status profile external". One can accomplish quite a lot with just those few. More on these later in the document. Or, go try 'em out.  

To Stop FS: At the command line mentioned above, just enter the command "shutdown" (no quotes). Or, one could kill it if its running in the background (as it does by default in PfSense on FreeBSD). I've done so with a "kill -15" followed by the FS PID (ps uax | grep free) and things shut down fine. There's probably a better way to do that, but this works.  

To Reload XML Files into FS: You'll likely be making changes to the XML files that configure FS, especially initially. Once you edit an XML file, FS has to be told about this so your changes will take effect. That can be done with the "reloadxml" command. Alternatively, one can issue a "stop" command and then start FS from the command line. Take your choice. 

For More Info: one can type "help" at the FS command prompt. There's a lot there; don't let it intimidate you. 


THE FS FILE SYSTEM: WHERE STUFF IS FOR STARTERS:

Here's the basics on where things are. Master this and you've got a good, starting grasp on how to "get around" when configuring and maintaining your FS:
NOTE: Everythings starts with /usr/local/freeswitch, so I'm just going to assume that with the tilde (~) here to save me from typing it again and again. 

~/bin is where all the binaries, like the FS executable itself, resides. 
~/conf is a starting point for your exloration. Some important files here, like vars.xml, and more. 
~/conf/sip_profiles is very important. Here are all the SIP User Agent instructions. UAs listen for registrations of SIP Phones, etc. 
~/conf/dialplan is where one sets up instructions telling FS about events and what to do when they take place.
~/conf/directory is where information on physical SIP phone extensions (and their groupings) is stored. 

These directories have subdirectories (which themselves have subdirectories) that you'll want to get familiar with. I'll go into more detail later on. Also, I've left out where language stuff goes, and more. But, one has to start somewhere. Now, let's get to work. 



SETTING THE SIP PROFILES TO USE DIFFERENT ETHERNET PORTS

INTERNAL LAN

Back to "dual-homed host setup with fixed IP on the WAN port". To make this work, one has to tell FS about where to be listening for SIP registrations, and other SIP traffic. That's done in the /usr/local/freeswitch/conf/sip_profiles directory. There, one finds what are conveniently called "sip profiles". One profile is for internal traffic (in my case called "internal.xml", for the phones on your LAN, etc) the other is for outside traffic (in my case called "external.xml", outside phone registrations, etc).

In the internal.xml file, we'll make some changes to accommodate the LAN network. Its a big file, but just a small tweak needs to be made. I'd edit the file and search for rtp-ip, sip-ip, ext-rtp-ip, and ext-sip-ip, respectively. Initially, you'll likely see "value=$${local_ip_v4}" in these four lines. Change the "value=" as below, using your own LAN IP address, for those four lines. Note that in the actual file there's a lot more stuff.

 <param name="rtp-ip" value="192.168.0.199"/>

 <param name="sip-ip" value="192.168.0.199"/>

Since I'm writting about a dual-homed example, we don't need a STUN server (i.e, we don't really need external rtp or sip) you can either leave the following two enteries alone or set them as I have below:

 <param name="ext-rtp-ip" value="192.168.0.199"/>

 <param name="ext-sip-ip" value="192.168.0.199"/>

Another reminder to use the LAN IP of your computer running FS, not mine. Also remember this is a dual-homed host in this particular example. You don't need to mess with this - the the internal.xml or external.xml files - if you have a single Ethernet port connected to a LAN with a gateway to the Internet. The settings "from the factory" will do just fine "out of the box" with a single homed host (though you will likely need STUN, not covered here). But, what fun is that?


EXTERNAL WAN

So, now we've got to do the same sort of thing to the /usr/local/freeswitch/conf/sip_profiles/external.xml file too right? Actually, while you could, one does not need to. Strangely enough, and to one's great happiness, this profile will go out and get the address of the WAN side and make the proper adjustments. Again, it will also, if a dynamic IP is being used, sense the address and any changes to it. I prefer the static IP and that's what I'm writting about here. The long and short of it is you don't have to change this file at this time. 


GETTING YOUR LAN PHONES REGISTERED

Also, we make a similar change elsewhere. In the /usr/local/freeswitch/conf/directory area sits a file that is - I hope - called default.xml. This file also has something to do with SIP on your LAN. In this case, what end-points (phones, soft phones) FS should expect to see, and other info (such as grouping these end-points into, well, groups). So, edit that file but use your LAN IP instead of mine, just like in the profiles above:

<!--the domain or ip (the right hand side of the @ in the addr-->
  <domain name="192.168.0.199">

While we're in the /usr/local/freeswitch/conf/directory area, let's consider how the SIP phones on your LAN are going to register with FS. I suggest using the pre-set 1000-1017 extension numbers - already set up for you - to configure and register your local SIP phones. If you need more, you'll know where/how to enter them in. So, while still in the default.xml file here - remember there are other default.xml files in other directories doing different things - note everything between the <groups> and </groups> tags. There are singular <group> </group> tags nested inside, setting up logical groupings of extension numbers into Sales, Billing, and Support. You can change these groupings to suite yourself, our go with what's there. Either way, you can see where this configuration is set now.

Also in this file, near the top, you'll see a pre-processor directive to go a directory below and include any .XML files found there. Leave this file, and cd down to the ./default directory. Just to be clear, we in directory /usr/local/freeswitch/conf/directory/default. Here's 1015.xml, one of the several you'll find there:

<include>
  <user id="1015">
    <params>
      <param name="password" value="$${default_password}"/>
      <param name="vm-password" value="1015"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="1015"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="Extension 1015"/>
      <!-- The value Extension 1015 can be personalized to your name, for instance, if you'd like -->
      <variable name="effective_caller_id_number" value="1015"/>
      <!-- Again, one could type in your main DID phone number instead of just 1015 if desired -->
      <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
      <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>

That's the whole XML file. Now, these can get more complicated, but this is right there and ready to go. While the variables pretty much tell you what they do, the point here is that if you need to modify something about a LAN SIP phone's "personal info", you come to /usr/local/freeswitch/conf/directory/default and look at the *.xml files here. And/or go "up" one level and look in the default.xml file there. NOTE: On my system, there's another default.xml file here. Keep in mind that while the names are the same, where they are has a lot to do with their function, so associate the place with the name. It gets easier with time.  

CHANGING THE PASSWORD FOR SIP REGISTRATION ON YOUR LAN

Last thing: You'll want to change the default password that your LAN SIP phones (or external phones for that matter) will use when authenticating. Leaving the one set up "out of the box" is a security risk since anyone could know it. So, make up your own. Edit the /usr/local/freeswitch/conf/vars.xml file. Near the top you'll see:

 <X-PRE-PROCESS cmd="set" data="default_password=1234"/>
  <!-- Did you change it yet? -->

Change the default_password parameter to some other four numbers that appeal to you. Now, only SIP phones that know the secret can register. Having done all this, set up your LAN SIP phones. Assign each a number between 1000 and 1017 for starters because these are already configured. Point your SIP phones the the IP address we've hard coded in the profiles (192.168.0.199 in my example). Be sure to use the new password you just set up.  


APPLYING YOUR CHANGES AND CHECKING YOUR WORK

Once you've made all these changes, either reloadxml the FS or simply stop and start it (don't know how? See "Stuff I Ought to Know Before Starting" section). To check, after the reload or restart, issue this command at the FS command line:

freeswitch at FreeSwitch.local> sofia status profile internal

You'll get something like the following if all is well:

API CALL [sofia(status profile internal)] output:
=================================================================================================
Name             	internal
Domain Name      	N/A
DBName           	sofia_reg_internal
Pres Hosts       	
Dialplan         	XML
Context          	public
Challenge Realm  	auto_to
RTP-IP           	192.168.0.199
Ext-RTP-IP       	192.168.0.199
SIP-IP           	192.168.0.199
Ext-SIP-IP       	192.168.0.199
URL              	sip:mod_sofia at 192.168.0.199:5060
BIND-URL         	sip:mod_sofia at 192.168.0.199:5060
HOLD-MUSIC       	local_stream://moh
OUTBOUND-PROXY   	N/A
CODECS           	G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
TEL-EVENT        	101
DTMF-MODE        	rfc2833
CNG              	13
SESSION-TO       	0
MAX-DIALOG       	0
NOMEDIA          	false
LATE-NEG         	false
PROXY-MEDIA      	false
AGGRESSIVENAT    	false
STUN-ENABLED     	true
STUN-AUTO-DISABLE	false
CALLS-IN         	2
FAILED-CALLS-IN  	0
CALLS-OUT        	3
FAILED-CALLS-OUT 	0

Registrations:
=================================================================================================
Call-ID:    	e6c864e9c4a3d at 192.168.0.80
User:       	1013 at 192.168.0.199
Contact:    	"user" <sip:1013 at 192.168.0.80:5062;transport=udp;fs_nat=yes;fs_path=sip%013%40192.168.0.80%062%3Btransport%3Dudp>
Agent:      	Grandstream GXP2000 1.1.6.46
Status:     	Registered(UDP-NAT)(unknown) EXP(2009-09-23 16:37:10)
Host:       	FreeSwitch.local
IP:         	192.168.0.80
Port:       	5062
Auth-User:  	1013
Auth-Realm: 	192.168.0.199

Call-ID:    	88ee646b068da at 192.168.0.41
User:       	1012 at 192.168.0.199
Contact:    	"user" <sip:1012 at 192.168.0.41:5062;transport=udp;user=phone;fs_nat=yes;fs_path=sip%012%192.168.0.41%3A5062%3Btransport%udp%3Buser%3Dphone>
Agent:      	Grandstream GXP2000 1.1.5.15
Status:     	Registered(UDP-NAT)(unknown) EXP(2009-09-23 16:37:12)
Host:       	FreeSwitch.local
IP:         	192.168.0.41
Port:       	5062
Auth-User:  	1012
Auth-Realm: 	192.168.0.199

Call-ID:    	45ee6e1b083da at 192.168.0.57
User:       	1017 at 192.168.0.199
Contact:    	"user" <sip:1017 at 192.168.0.57:5062;transport=udp;fs_nat=yes;fs_path=sip%17%40192.168.0.57%62%3Btransport%3Dudp>
Agent:      	Grandstream GXP2000 1.1.5.15
Status:     	Registered(UDP-NAT)(unknown) EXP(2009-09-23 16:37:15)
Host:       	FreeSwitch.local
IP:         	192.168.0.57
Port:       	5062
Auth-User:  	1017
Auth-Realm: 	192.168.0.199

=================================================================================================

Note that everwhere there's an IP address (RTP_IP, SIP_IP, EXT_RTP_IP, EXT_SIP_IP), its the LAN Ethernet address on our FS box or the address of a LAN SIP phone (under Registrations). That's what we want. The internal.xml profile is listening ONLY on the LAN. 

Do the same thing for the external profile:

freeswitch at FreeSwitch.local> sofia status profile external
API CALL [sofia(status profile external)] output:
=================================================================================================
Name             	external
Domain Name      	N/A
DBName           	sofia_reg_external
Pres Hosts       	
Dialplan         	XML
Context          	public
Challenge Realm  	auto_to
RTP-IP           	yy.yy.yy.yy
Ext-RTP-IP       	yy.yy.yy.yy
SIP-IP           	yy.yy.yy.yy
Ext-SIP-IP       	yy.yy.yy.yy
URL              	sip:mod_sofia at yy.yy.yy.yy:5080
BIND-URL         	sip:mod_sofia at yy.yy.yy.yy:5080
HOLD-MUSIC       	local_stream://moh
OUTBOUND-PROXY   	N/A
CODECS           	PCMU,PCMA,GSM
TEL-EVENT        	101
DTMF-MODE        	rfc2833
CNG              	13
SESSION-TO       	0
MAX-DIALOG       	0
NOMEDIA          	false
LATE-NEG         	false
PROXY-MEDIA      	false
AGGRESSIVENAT    	false
STUN-ENABLED     	true
STUN-AUTO-DISABLE	false
CALLS-IN         	1
FAILED-CALLS-IN  	0
CALLS-OUT        	1
FAILED-CALLS-OUT 	0

Registrations:
=================================================================================================
=================================================================================================

Here, we have only the external Ethernet static-ip address showing up. Again, that's as it should be. This profile is listening only to the WAN and not interfearing with the LAN. Note that in my case, I don't have any registrations on this profile. That's OK, but I could set things up for phones to register from the WAN if that was desirable (which in our case, it is not as yet). As we'll see, these two will work together well for out-going and in-coming calls. Read on.



REGISTERING WITH A SIP PROVIDER

So that we can make and receive calls, we'll need to register with a SIP/PSTN provider. Getting one is outside the scope of this document, but check out the Internet, many are available. You'll have to have FS register with your "gateway provider" so things can work. Here's how we did it: In /usr/local/freeswitch/conf/sip_profiles/external, I created a file called urbancom.xml (happens to be the name of my provider). As long as this file is in that directory, FS will use it to register with my SIP/PSTN provider, making calling "in and out" possible. Here's what's inside:

<include>
  <gateway name="xxx.xxx.xxx.xxx">
     <!-- Remember, you'll need this name for your "out-bound calls" extension -->
     <param name="username" value="8158381212"/>
     <!-- Natch', your own username, ours is like a NPANXXTN -->
     <param name="password" value="somekindapassw0rd"/>
     <!-- Natch', your own password -->
     <param name="expire-seconds" value="60"/>
     <!-- Time out value, 60 works for us -->
     <param name="register" value="true"/>
     <!-- You want to register, so "true" -->
     <param name="register-transport" value="udp"/>
     <!-- You'll probably use udp, but ask your provider -->
     <param name="retry-seconds" value="30"/>
     <!-- Retry parameter, 30 seconds works for us -->
     <param name="caller-id-in-from" value="false"/>
     <!-- Use the callerid of an inbound call in the from field on outbound calls via this gateway. We don't, up to you. -->
     <param name="contact-params" value="tport=5060"/>
     <!-- Port 5060 is customary, but again, ask the provider -->
     <param name="ping" value="25"/>
     <!-- We like to send a ping just to keep some traffic traversing and keep us registered -->
  </gateway>
</include>

That's the whole file. The "name" has to be something DNS can translate or an actual IP address. Clearly, I used the latter. Username and password come from the provider, as do the transport value (could be tcp, ask the provider) and the tport value (5060 is typical, but ask the...you get the message). I like the ping value, kinda making sure the link is kept alive in the absence of other traffic.

Now, we want to set something to happen "call-wise". That is, when our LAN phones register with FS and dial a phone number, something good will transpire. Here's one way to get that done: We'll add an "extension" - not to be confused with a physical telephone extension - to what's called a "context". Contexts are just containers for extensions. These context extensions are really directions to FS, telling it how to react when certain things take place. Our "thing" in this case is a registered sip phone dialing a 10 digit number, trying to call someone on the outside.

To make this work, we add the following "extension" to the "context" in the /usr/local/freeswitch/conf/dialplan/default.xml file. Don't confuse this default.xml with the one in /usr/local/freeswitch/conf/directory, different files doing different things. Enter the following in as the first extension in the file, note the <extension> tags that delimit the text you're inputting. What's the point? Well, this paragraph lets FS know what to do when an outgoing call is dialed. Note the <action application> tag. This directs FS to use a specific gateway, the one we just set up, to "get the call out".

   <! -- Mike stole from elsewhere: Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here -->
   <extension name="UrbanOut">
     <! -- Call this what you want, above. Can be anything that makes sense to you. -->
     <condition field="destination_number" expression="^(1{0,1}\d{10})$">
     <!-- The above translates to: "a 10 digit number, with or without a preceeding 0 or 1" -->
       <action application="set" data="effective_caller_id_number=18158381212"/>
       <!-- Change the above effective_caller_id_number to your own DID. -->
       <!-- If your provider does not provide ringback (180 or 183) you may simulate ringback by uncommenting the following line. -->
       <!-- action application="ringback" /-->
       <action application="bridge" data="sofia/gateway/xxx.xxx.xxx.xxx/$1"/>
       <!-- Change the above to your own gateway name, between "/gateway/" and "/$1" -->
     </condition>
   </extension>

NOTE: One can know if the gateway is registered or not, plus other very useful information, by using the following command at the FS command line:

freeswitch at FreeSwitch.local> sofia status

Don't forget to hit "enter". That will get you something like the following:

API CALL [sofia(status)] output:

    Name                Type                       Data                        State
=================================================================================================
   internal            profile          sip:mod_sofia at 192.168.0.199:5060      RUNNING (0)
   external            profile            sip:mod_sofia at yy.yy.yy.yy:5080      RUNNING (0)
   xxx.xxx.xxx.xxx     gateway           sip:8158381212 at xxx.xxx.xxx.xxx       REGED
   default               alias                                  internal      ALIASED
   192.168.0.199         alias                                  internal      ALIASED
=================================================================================================
2 profiles 2 aliases

Note that my gateway (check the "Type" column) is in state REGED, meaning it is registered with my provider and all is well with the world. If it isn't, check with your provider. If they're nice, they'll look at your registration attempt and provide clues as to what's not right. Hey, its a start.

While were at it, also note that the internal profile is up and running. That's what we messed with in the first place to direct calls to the gateway. Lo, the beautiful symmetry of it all. Further, the internal profile is listening on the LAN address (192.168.0.199 happens to be ours) while the external listens to the outside world on a different Ethernet static IP (it isn't our real one so don't try to hack it) as well as a different UDP port (a logical port for UDP, 5080 for the outside, 5060 for the inside here). All make sense? The alias types are just that, different names for the internal and external profiles. Used for some FS shorthand we won't talk about at the moment.


LETS MAKE SOME CALLS (OUT-GOING)

At this point, your FS has SIP phones registered locally, a SIP/PSTN gateway that it is registered to, and SIP profiles listening for work to do. Your SIP phone - there are many types so I assume you know how to configure and use your device - should itself show as registered to the FS. So, go ahead. Make that call out. I called my cellphone the first time. If all is well, and if your output from above looks like it should then all IS well, your outside phone is ringing. Rejoice.   


IN-COMING CALLS, RINGING A GROUP

OK, that was fun. But, one would want people to be able to call FS from the outside too. It would also be nice if FS would ring my SIP phone so I could take the call. As a matter of fact, at least in my office, we like all the phones to ring and whoever is available takes the call. One way to set up incoming calls in this way (ring all extensions in a group) is the following:

First, one has to have a group to ring. Forget, for a second, that there are groups already setup in the /usr/local/freeswitch/conf/directory. There is a feature of FS that allows one, from the SIP phone itself, to join a group that FS can then use. Go to each SIP phone in turn and dial extension 8101 (or *8101 on my PfSense version). 81 means "I'm putting this phone into a group". The last two digits tell what group, 01 in this case. You'll hear a tone letting you know it worked. Hang up and go join the next phone to group 01 until you've done 'em all. 

NOTE: Want to delete a phone from the group (it will still be able to call out)? Dial extension 8001 (or *8001 in my case on PfSense) and that particular phone will be deleted from the group. Keep in mind that once you are in the group, that group will persist in FS. You can re-register, restart FS or even reboot the computer FS is running on. You have to take action to get an extension out of the group (which is a good thing). 

Also note that if you'd like to see how 8001, 8101, or any other "logic" works in FS on your LAN, edit the /usr/local/freeswitch/conf/dialplan/default.xml file. Search for 80, for instance. This file contains "contexts" which themselves containg "extensions". Here, and extention is logic for taking some action when FS sees a particular event take place (like someone dialing 8201). We are at the heart of FS in this file, don't forget about it or what its for. 

Now, we have our group, 01 in this case (could be anything between 00 and 99). Dial extension 8201 (or, again, *8201 depending) and all phones joined to group 01 will ring. Fun stuff. Now, let's get FS to use that group to ring the same group. To do so, go to the /usr/local/freeswitch/conf/dialplan/public directory. I created a file that would be read first by using the file name 00_inbound_did.xml. The name is significant, especially the leading 00s. In this case, its the only file here but one could have many. Anyway, enter the following in that file, correcting for your own private info: 

<include>
  <extension name="Urban1212">
  <!-- Call it what you'd like. -->
    <condition field="destination_number" expression="^(8158381212)$">
    <!-- When FS sees the incoming number above - please use your own DID - it will engage the actions below -->
      <action application="set" data="domain_name=192.168.0.199"/>
      <!-- again, use the domain for your LAN from earlier in this document -->
      <action application="transfer" data="*8201 XML default"/>
      <!-- Here's the group ring. I need the *, you may not. But, ringing 8201, just like from your phone, will ring the whole group -->
      <!-- The "*8201 XML default" string means: find extension *8201, its in the default.xml file in the dialplan directory. -->
    </condition>
  </extension>
</include>

OK, having done the above, reloadxml or stop and start your FS. Then, dial in from a cellphone or other outside line. All phone in your group should ring. The DID of your calling phone should show. Now, we've got something useful going.  

Now, this is pretty powerful stuff. You could create another file in /usr/local/freeswitch/conf/dialplan/public and call it 01_inbound_did.xml. There one could set up another group (or single phone) to be rung if one has another DID. Much is possible. But, we'll keep to getting things functional and operational. 



MOVING ON TO MORE ADVANCED TOPICS


VOICE MAIL (OR "HOW TO ACCESS SOME THINGS THAT FS IS PREPARED TO DO")

If one has a registered phone (at this point in our document, only possible on the internal profile) then you have a voicemail extension at your disposal too. To check if an extension is in fact registered with your FS, go to your console screen and type:

freeswitch at FreeSwitch.local> sofia status profile internal

As in the scection on APPLYING YOUR CHANGES AND CHECKING YOUR WORK, one sees output about the internal profile itself, followed by all registered extensions for the profile. 

Now, we're going to use the FS dialplan file to figure out how to access voicemail. You'll likely want to do things like create a message for people to hear when they get VM. Also, you will want to hear what messages are left there, delteing some, leaving others. How? We'll answer that question; in the process learning  more about FS dialplans, contexts, extensions, and how to understand and use them. 

NOTE: You may recall a line in the REGISTERING WITH A SIP PROVIDOR section stating in passing that a dial plan containted "contexts". Contexts themselves are containters for instruction sets called "extensions". Don't confuse the term  "extension" here with a SIP phone or other device (which has an extension number) and these instruction blocks within contexts.


A CLOSER LOOK AT YOUR DEFAULT DIAL PLAN

Let's edit the /usr/local/freeswitch/conf/dialplan/default.xml file. Near the top of the file, one notes a <context=default> tag. All the way at the bottom of the file, one sees the closing </context> tag. In between these tags, one can see many groupings, deliniated with the <extension="somename"> and </extension> tags. Each of these extensions is actually a reaction to an event. The event being serviced is seen in the <condition> tag set, inside each extension. Usually, that condition is a number that someone has dialed. That could be an outside caller coming into your FS via your gateway provider, or a registered phone on your LAN doing the dialing. When FS "sees" a number, coming or going, it looks at the dial plans, matching that number to each extension's <condition>. FS then performs the <actions> within the matching extension.

NOTE: If the pattern matching syntax within the  "expresion=" of the <condition> tag makes no sense, please have a look at "regular expressions" ("regex" for short) and how they work. There's a great regex primer on the freeswitch.org site that's well worth finding and reading it you need it. However, that's outside the scope of this document. Nonetheless, you'll need some familiarity with regex's to understand conditions.

So, now that you're in the file and have the gist of what's going on here, search for the string "voicemail". In mine, courtesy of PfSense (pfsense.org), I find five extensions dealing with voicemail. We'll talk about each a little bit. Once finished, you'll know a lot more about voice mail on my FS. Yours will likely be similar. Here's my five, right from ~/conf/dialplan/default.xml on my system:

    <!-- voicemail operator extension -->
    <extension name="operator">
      <condition field="destination_number" expression="^\*operator$|^0$">
        <action application="set" data="transfer_ringback=$${hold_music}"/>
        <action application="transfer" data="1000 XML features"/>
      </condition>
    </extension>

The "operator" extension above is invoked by the string *operator or a dialed 0. On PfSense, that author likes to use a "*" with special extensions. Others do not. You can set it how you like it. If you have an extension like my "operator", go to a registered phone and hit 0. Note that you'll need a "real" extension 1000 on your LAN to be able to use this. We'll provide a way to put this to work later.   

    <!-- voicemail main2 extension -->
    <extension name="vmain2">
      <condition field="destination_number" expression="^vmain2$|^\*97$|^\*4000$">
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="voicemail" data="check default ${domain_name}"/>
      </condition>
    </extension>

In extension "vmain2", one can type *97, *4000 or a literal vmain2 to access the <action> here. Looking at the last <action> tag, one notes voicemail being invoked to check a user's VM contents. You'll need your user ID and password to gain access. Try this one out, if you have it, and follow the prompts. 

    <!-- voicemail main extension -->
    <extension name="vmain">
      <condition field="destination_number" expression="^vmain$|^\*98$">
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="voicemail" data="check default ${domain_name} ${sip_from_user}"/>
      </condition>
    </extension>

Much like vmai2, you can type *98 (or set it how you'd like it) for this VM entry. Note the <action> here works thus: if you call from your own phone extension, you'll only need the password to gain entry. The invocation of voicemail here uses the number you are dialing from as the user. 

    <extension name="send_to_voicemail_5digits">
      <condition field="destination_number" expression="^\*99(\d{5})$">
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
        <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
      </condition>
    </extension>

Above, a handy extension to transfer a caller to a five-digit extension, if you have them. Mine are all four digit, 1000-1019, the defaults. You could change this to handle any extension lenth you like. 

    <extension name="send_to_voicemail_4digits">
      <condition field="destination_number" expression="^\*99(\d{4})$">
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
        <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
      </condition>
    </extension>

Finally, an extension so that anyone taking a call can transfer that party to any four digit extension they'd like (much like the 5 digit above this). Just hit "tranfer" on your phone, and then enter *991013 to go to 1013's voicemail without ringing his/her phone. Nice. Remember, if your FS isn't configured as above, you can do so with the extension as you see them here, or by modifiying them to suit yourself. 

Now that you can get around in this particular default.xml, have a look at other .xml files in the /usr/local/freeswitch/conf/dialplan directory. Note that we've been here before in the  IN-COMING CALLS, RINGING A GROUP section of this document. There, we were directing in-coming calls to the proper default.xml extension via the ~/conf/dialplan/public subdirectory. Hopefully, this is all beginning to make sense as a whole. 

So, back to Voicemail. Now, you can access VM from your registered phone, set up your own voice prompt, and administer VM in general. How did we figure that out? By reading the "instructions" in the default dial plan. There's more to know, and a lot more that can be done. The idea is to build on what you've learned up to this point. Now, let's tie that in with calls coming into the system to make our FS even more useful.  



INCOMING CALLS RING A GROUP AND GOES TO VM AFTER NO ANSWER

This time, I'm just going to "lay out" some files that are accessed, in order, when an incoming call hits FS. Actually, FS isn't looking at the literal file, it has read them into the running application (assuming you've told it to). If not, FS can't "see" them so be sure you've reloadxml'd or shutdown/started FS. So, have a look and you'll see the flow FS follows to find out what to do. Let's say someone on the "outside" dials us in Illinois at 8158381212.


First, the call "comes in" and FS wants to know where to send it. Since this is a public incoming call, it looks for a match in ~/conf/dialplan/public/ and finds the match - in my case - in 01_incoming_did.xml:

     <include>
       <extension name="Urban1212">
       <!-- Call it what you'd like. -->
         <condition field="destination_number" expression="^(8158381212)$">
         <!-- When FS sees the incoming number above  please use your own DID  it will engage the actions below -->
           <action application="set" data="domain_name=192.168.0.199"/>
           <!-- again, use the domain for your LAN from earlier in this document -->
           <action application="transfer" data="8201 XML default"/>
           <!-- Go to this extension, 8201, to find more instructions. -->
         </condition>
       </extension>
     </include>

Notice above that I changed the transfer extension from *8201 (IN-COMING CALLS, RINGING A GROUP) to a new extension, 8201 (no preceeding *), seen below. I did that so I could add the <actions> that I needed and also demonstrate another way to ring groups, as you will see:  


Next, FS looks in ~/conf/dialplan/default.xml, just as it was told to, above, for the 8201 extension. Here it is:

    <!-- Mike Added: Incoming Urban Calls -->
    <extension name="UrbanIn">
      <condition field="destination_number" expression="^(8201)$">
         <action application="set" data="call_timeout=20"/>
         <action application="set" data="domain_name=192.168.0.199"/>
         <action application="set" data="hangup_after_bridge=true"/>
         <action application="set" data="continue_on_fail=true"/>
         <action application="bridge" data="group/support@${domain_name}"/>
         <action application="answer"/>
         <action application="sleep" data="1000"/>
         <action application="voicemail" data="default 192.168.0.199 1013"/>
      </condition>
    </extension>

In this extension, I have the included actions do a few things. I set a timeout value to 20. That means if no one answers in that amount of time (from when the bridge is invoked later on), FS will skip down to the next action, invoking voicemail (via dialplan default.xml) for extension 1013. 

Also notice that instead of bridging to *8201, which I could have still done, I opted to use one of the groups already set up for me in ~/conf/directory/default.xml (There's a lot of default.xml files. But, being in different directories, they do completely different things. Actually, one could call them anything you'd like. For the moment, I'm sticking to the names as installed). 


Here's a snippet of ~/conf/directory/default.xml. I simply located the <group name> set equal to "support", and added the users I wanted to that group. Note that the same users can be in different groups if you'd like: 

    <group name="support">
      <users>
        <user id="1010" type="pointer"/>
        <user id="1012" type="pointer"/>
        <user id="1013" type="pointer"/>
        <user id="1017" type="pointer"/>
      </users>
    </group>

Assuming you've set up extension 1013 (the physical phone's voicemail session) with a message about everybody being busy and please leave us a call back number, you're good to go (remember to reloadxml or shutdown/start if you've changed XML files). My phone's (admittedly an older Grandstream GXP-2000) message light blinked red as soon as I called in and left one (I'm 1013). Using the appropriate numbers from A CLOSER LOOK AT YOUR DIAL PLAN, you can now get your VM and deal with it as you'd like.  



ADDING ANOTHER DID

Let's say that you would like to have more than one direct inward dial number for your home or business. We saw in the section on REGISTERING WITH A SIP PROVIDER who to have one DID registered. To have more than one, naturally, you will have to have another <gateway> set up as that one is. However, if you want to use the same providor, we'll have to change the syntax of the XML file in /usr/local/freeswitch/conf/sip_profiles/external. Have a look: 

my ~/conf/sip_profiles/external/urban.xml file for registering and using one DID:

<include>
  <gateway name="xxx.xxx.xxx.xxx">
     <!-- Remember, you'll need this name for your "out-bound calls" extension -->
     <param name="username" value="8158381212"/>
     <!-- Natch', your own username, ours is like a NPANXXTN -->
     <param name="password" value="somekindapassw0rd"/>
     <!-- Natch', your own password -->
     <param name="expire-seconds" value="60"/>
     <!-- Time out value, 60 works for us -->
     <param name="register" value="true"/>
     <!-- You want to register, so "true" -->
     <param name="register-transport" value="udp"/>
     <!-- You'll probably use udp, but ask your provider -->
     <param name="retry-seconds" value="30"/>
     <!-- Retry parameter, 30 seconds works for us -->
     <param name="caller-id-in-from" value="false"/>
     <!-- Use the callerid of an inbound call in the from field on outbound calls via this gateway. We don't, up to you. -->
     <param name="contact-params" value="tport=5060"/>
     <!-- Port 5060 is customary, but again, ask the provider -->
     <param name="ping" value="25"/>
     <!-- We like to send a ping just to keep some traffic traversing and keep us registered -->
  </gateway>
</include>


Here's the same file, registering with the same providor, but now we have two registrations, one for each DID. Check the comments in the XML code below:


<include>
  <gateway name="urban1212">
	<!-- Any name that makes sense -->
        <param name="username" value="8158381212"/>
        <param name="password" value="somekindapassw0rd"/>
        <param name="realm" value="xxx.xxx.xxx.xxx"/>
	<!-- The gateway name from the "One DID" example. -->
        <param name="expire-seconds" value="60"/>>
        <!--/// do not register if value="false"///-->
        <param name="register" value="true"/>
        <param name="register-transport" value="udp"/>
        <param name="retry-seconds" value="30"/>
        <!--Use the callerid of an inbound call in the from field on outbound calls via this gateway -->
        <param name="caller-id-in-from" value="false"/>
        <param name="contact-params" value="tport=5060"/>
        <!--send an options ping every x seconds, failure will unregister and/or mark it down-->
        <param name="ping" value="25"/>
  </gateway>
  <gateway name="urban1234">
	<!-- Any name that makes sense, but not the one from above. The gateway names must be unique. --> 
        <param name="username" value="8158381234"/>
        <param name="password" value="anotherpassword1"/>
        <param name="realm" value="xxx.xxx.xxx.xxx"/>
	<!-- The gateway name from the "One DID" example. This is the same as above, but its OK here. -->
        <param name="expire-seconds" value="60"/>>
        <!--/// do not register if value="false"///-->
        <param name="register" value="true"/>
        <param name="register-transport" value="udp"/>
        <param name="retry-seconds" value="30"/>
        <!--Use the callerid of an inbound call in the from field on outbound calls via this gateway -->
        <param name="caller-id-in-from" value="false"/>
        <param name="contact-params" value="tport=5060"/>
        <!--send an options ping every x seconds, failure will unregister and/or mark it down-->
        <param name="ping" value="25"/>
  </gateway>
</include>


Notice that I had to use a different gateway name in this example for each DID registration, even though the same providor is registering both DIDs. This was done using the <param name="realm" value=xxx.xxx.xxx.xxx"/> tag. Of course, use your own gatway IP address. If you just double the entries of the single DID example, only one will actually register. You could use the syntax of the two DID example for one DID. As you can see, there's more than one way to do things. 

IMPORTANT: If you only make the changes above - assuming you have a working system with the "one DID" setup - you'll find that you can not make outgoing calls anymore (but incoming calls still work). Why? Remember in REGISTERING WITH A SIP PROVIDOR, we told FS what to do with out-going calls. That is, to use your newly registered gatwway. In the "two DID" example, we've changed the name of that gateway. Had to, can't use the same one twice. So, pick one of your DID gateway names above and update the extension in ~/conf/dialplan/default.xml for out-going calls (I use the urban1212 gateway, see below).

    <!-- Mike Added: Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here -->
    <extension name="UrbanOut">
     <condition field="destination_number" expression="^(1{0,1}\d{10})$">
       <action application="set" data="effective_caller_id_number=18158384096"/>
       <!-- If your provider does not provide ringback (180 or 183) you may simulate ringback by uncommenting the following line. -->
       <!-- action application="ringback" /-->
       <action application="bridge" data="sofia/gateway/urban1212/$1"/>
     </condition>
    </extension>

From jerry.richards at teotech.com  Fri Oct  2 09:40:28 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 2 Oct 2009 09:40:28 -0700
Subject: [Freeswitch-users] Call Forward All/Busy/No-Answer
Message-ID: <ABB03FEA28D4411393F2389EF02C9EA0@greyhawk.tonecommander.com>

How would I configure FS to Call Forward All or Call Forward when Busy or
Call Forward when No-Answer?  Can this be done at the server, rather than at
the phone?

Best Regards,
Jerry




From mike at van.lammeren.net  Fri Oct  2 09:51:11 2009
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Fri, 2 Oct 2009 12:51:11 -0400
Subject: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
In-Reply-To: <1254426084.3779.29.camel@sodium>
References: <5d2828f0909290720y6bb4997uc4b9afba4b0fe74c@mail.gmail.com>
	<5d2828f0910010945q30ed76e2x50d6ab3b13bc8f88@mail.gmail.com>
	<32C47FA5-774E-4584-890B-0A066F5CFE14@me.com>
	<191c3a030910011214i3a9eaad1xf242ecf19dc80da3@mail.gmail.com>
	<1254426084.3779.29.camel@sodium>
Message-ID: <5d2828f0910020951n784c569ak5af4bf4f25a8ac00@mail.gmail.com>

I only mentioned the OS I used as a reference for people. If they want to do
the same thing on another OS, then they might not have apt-get, etc.
Mike van Lammeren

On Thu, Oct 1, 2009 at 3:41 PM, Hadley Rich <hads at nice.net.nz> wrote:

> On Thu, 2009-10-01 at 14:14 -0500, Anthony Minessale wrote:
> > can we do it without advertising to use ubuntu =D
> > We don't like encouraging our users to use bleeding edge OS for our
> > own sanity with debugging.
>
> I understand your stance, though if we're talking about Ubuntu 8.04 LTS
> (Long Term Support - 5 years) it's not really bleeding edge anymore. 18
> months ago when it was released it may have been a little, but the LTS
> releases still aren't as bleeding edge as the standard support in
> between releases.
>
> hads
> --
> http://nicegear.co.nz
> New Zealand's Open Source Hardware Supplier
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at van.lammeren.net  Fri Oct  2 09:56:22 2009
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Fri, 2 Oct 2009 12:56:22 -0400
Subject: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
In-Reply-To: <0ca401ca42cf$8a709d20$9f51d760$@com>
References: <5d2828f0909290720y6bb4997uc4b9afba4b0fe74c@mail.gmail.com>
	<5d2828f0910010945q30ed76e2x50d6ab3b13bc8f88@mail.gmail.com>
	<32C47FA5-774E-4584-890B-0A066F5CFE14@me.com>
	<191c3a030910011214i3a9eaad1xf242ecf19dc80da3@mail.gmail.com>
	<0ca401ca42cf$8a709d20$9f51d760$@com>
Message-ID: <5d2828f0910020956w5c3fd2dfi481bb2952b0beb85@mail.gmail.com>

The load balancer listens to the virtual IP address, and port-forwards to
one of the FreeSWITCH boxes. Each FreeSWITCH box listens for the same
virtual IP address for SIP registrations and connections, which is what
FreeSWITCH needs to bind to. All other traffic actually travels over their
real IP address, which is what the FreeSWITCH servers would use to talk to
each other.

On Thu, Oct 1, 2009 at 3:44 PM, Peder <peder at networkoblivion.com> wrote:

>  Looking thru the example, it looks like each box has a real address of
> 21, 22 or 23 and they all have a loopback of .17, right?  So even though
> they connections are being load balanced, each box really thinks it is .17
> and each client that connects thinks it is connecting to .17, right?  If
> that?s the case, how does a client on one box call a client on the other
> box?  Since every box thinks it is .17 how would you bridge to another user
> on another box that also thinks it is .17?  Or am I totally missing how it
> works?
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Thursday, October 01, 2009 2:14 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
>
>
>
> can we do it without advertising to use ubuntu =D
> We don't like encouraging our users to use bleeding edge OS for our own
> sanity with debugging.
> Not to say you are not allowed to I just don't want to encourage it =p
>
>
>  On Thu, Oct 1, 2009 at 1:12 PM, Even Andr? Fiskvik <grevenx at me.com>
> wrote:
>
> That's very cool Mike!
>
>
>
> I'm going to try to configure four boxes with this as well (Btw, did you
> use physical hardware or virtualization?)
>
> and see how it goes. I followed Daniel Aliaman's blog as well, but I can
> try it again with the tips
>
> you provided on FreeSWITCH config to see if I can get it working properly
> this time.
>
> We did the setup on CentOS, but I wouldn't think that would be any issue.
>
>
>
> Perhaps you or we could write up a complete guide about this on the wiki
> since this is an scenario
>
> commonly used? Also it would be great if we could outline possible issues
> (and even better solutions)
>
> to this kind of setup with regards to stuff like conferencing, bridging
> between registered users and presence.
>
>
>
>
>
> Best regards,
>
> Even Andr?
>
>
>
>
>
> On 1. okt. 2009, at 18.45, Mike van Lammeren wrote:
>
>
>
>  Guess what? I have two FreeSWITCH servers working behind UltraMonkey,
> using heartbeat and ldirectord for load-balancing, fail-over and high
> availability! I'm probably not the first one to do it, but as near as Google
> and I can tell, I'm the first one to write about it.
>
>
>
> Here's how you can duplicate my setup:
>
>
>
> 1. Install Ubuntu Server 8 on four machines, either real or VM.
>
> 2. Compile and install FreeSWITCH v1.0.4 from source on two machines,
> following these instructions:
> http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start
>
> 3. Configure both FreeSWITCH boxes, and make sure they are both working.
>
> 4. Follow (most of) these instructions from Daniel Aliaman's blog. They
> were written for Asterisk, but since a SIP connection is a SIP connection,
> most of the document applies to FreeSWITCH:
>
> http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf
>
>
>
> The one problem I ran into was the IP address and port to which FreeSWITCH
> was bound. The default is to use the primary address, which works great
> out-of-the-box for everything else. When a client tried to register, all it
> got back was an ICMP error -- Destination Unreachable, Port Unreachable.
> That error is returned when no sockets are listening for UDP packets. To get
> FreeSWITCH to listen for your Virtual IP, you need to set it in two places:
>
>
>
> 5. In /opt/freeswitch/conf/vars.xml, set "bind_server_ip".
>
> 6. In /opt/freeswitch/conf/sip_profiles/internal.xml, set "sip-ip".
>
>
>
> That should do it! If you have any success, please report to this list.
>
>
>
> Keep in mind that if you want to do something like conferencing between two
> registered clients, then you have to deal with the fact that the clients may
> or may not be on the same box.
>
>
>
> Mike van Lammeren
>
>
>
>
>
>
>
> On Tue, Sep 29, 2009 at 10:20 AM, Mike van Lammeren <mike at van.lammeren.net>
> wrote:
>
>
>
> On Mon, Sep 28, 2009 at 9:05 PM, "Even Andr? Fiskvik" <grevenx at me.com
> > wrote:
>
> From: "Even Andr? Fiskvik" <grevenx at me.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Mon, 28 Sep 2009 22:52:13 +0200
> Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
>
> I have been working with a similar setup myself, but for some reason I
> ended up ditching the
>
> UltraMonkey setup because I just couldn't get it to work right.
>
>
>
> It's been quite a while since my effort, so I don't remember what the exact
> issue was.
>
> I got registrations to work, but had some other sip-dialog issues.
>
>
>
> We have since then changed over to running OpenSIPs as a loadbalancer in
> front of
>
> multiple FreeSWITCH instances. This setup is still in testing, but
> seemlingy works fine
>
> (and if it doesn't, it's my own fault for writing a bad opensips config).
>
>
>
> After we have done some more testing I can create a wiki-page with config
> details.
>
>
>
>
>
> Best regards,
>
> Even Andr?
>
>
>
>
>
> Thanks, Even, that would be great! I might have to give up on the
> ultramonkey solution, since I can't find anyone who has made it work. It's
> too bad, because it would fit well with the rest of our architecture.
>
>
>
> Mike van Lammeren
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at van.lammeren.net  Fri Oct  2 09:58:16 2009
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Fri, 2 Oct 2009 12:58:16 -0400
Subject: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
In-Reply-To: <32C47FA5-774E-4584-890B-0A066F5CFE14@me.com>
References: <5d2828f0909290720y6bb4997uc4b9afba4b0fe74c@mail.gmail.com>
	<5d2828f0910010945q30ed76e2x50d6ab3b13bc8f88@mail.gmail.com>
	<32C47FA5-774E-4584-890B-0A066F5CFE14@me.com>
Message-ID: <5d2828f0910020958o1176ab71rc361d9b2e96d6e58@mail.gmail.com>

I am running the servers on the free version of VMware's ESX platform, but
only for development purposes. We will be setting up real machines sometime
in Spring 2010.

On Thu, Oct 1, 2009 at 2:12 PM, Even Andr? Fiskvik <grevenx at me.com> wrote:

> That's very cool Mike!
> I'm going to try to configure four boxes with this as well (Btw, did you
> use physical hardware or virtualization?)
> and see how it goes. I followed Daniel Aliaman's blog as well, but I can
> try it again with the tips
> you provided on FreeSWITCH config to see if I can get it working properly
> this time.
> We did the setup on CentOS, but I wouldn't think that would be any issue.
>
> Perhaps you or we could write up a complete guide about this on the wiki
> since this is an scenario
> commonly used? Also it would be great if we could outline possible issues
> (and even better solutions)
> to this kind of setup with regards to stuff like conferencing, bridging
> between registered users and presence.
>
>
> Best regards,
> Even Andr?
>
>
> On 1. okt. 2009, at 18.45, Mike van Lammeren wrote:
>
> Guess what? I have two FreeSWITCH servers working behind UltraMonkey, using
> heartbeat and ldirectord for load-balancing, fail-over and high
> availability! I'm probably not the first one to do it, but as near as Google
> and I can tell, I'm the first one to write about it.
> Here's how you can duplicate my setup:
>
> 1. Install Ubuntu Server 8 on four machines, either real or VM.
> 2. Compile and install FreeSWITCH v1.0.4 from source on two machines,
> following these instructions:
> http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start
>  <http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start>3. Configure both
> FreeSWITCH boxes, and make sure they are both working.
> 4. Follow (most of) these instructions from Daniel Aliaman's blog. They
> were written for Asterisk, but since a SIP connection is a SIP connection,
> most of the document applies to FreeSWITCH:
> http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf
>
> <http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf>The one
> problem I ran into was the IP address and port to which FreeSWITCH was
> bound. The default is to use the primary address, which works great
> out-of-the-box for everything else. When a client tried to register, all it
> got back was an ICMP error -- Destination Unreachable, Port Unreachable.
> That error is returned when no sockets are listening for UDP packets. To get
> FreeSWITCH to listen for your Virtual IP, you need to set it in two places:
>
> 5. In /opt/freeswitch/conf/vars.xml, set "bind_server_ip".
> 6. In /opt/freeswitch/conf/sip_profiles/internal.xml, set "sip-ip".
>
> That should do it! If you have any success, please report to this list.
>
> Keep in mind that if you want to do something like conferencing between two
> registered clients, then you have to deal with the fact that the clients may
> or may not be on the same box.
>
> Mike van Lammeren
>
>
>
> On Tue, Sep 29, 2009 at 10:20 AM, Mike van Lammeren <mike at van.lammeren.net
> > wrote:
>
>>
>> On Mon, Sep 28, 2009 at 9:05 PM, "Even Andr? Fiskvik" <grevenx at me.com
>> > wrote:
>>
>>> From: "Even Andr? Fiskvik" <grevenx at me.com>
>>> To: freeswitch-users at lists.freeswitch.org
>>> Date: Mon, 28 Sep 2009 22:52:13 +0200
>>> Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
>>> I have been working with a similar setup myself, but for some reason I
>>> ended up ditching theUltraMonkey setup because I just couldn't get it to
>>> work right.
>>>
>>> It's been quite a while since my effort, so I don't remember what the
>>> exact issue was.
>>> I got registrations to work, but had some other sip-dialog issues.
>>>
>>> We have since then changed over to running OpenSIPs as a loadbalancer in
>>> front of
>>> multiple FreeSWITCH instances. This setup is still in testing, but
>>> seemlingy works fine
>>> (and if it doesn't, it's my own fault for writing a bad opensips config).
>>>
>>> After we have done some more testing I can create a wiki-page with config
>>> details.
>>>
>>>
>>> Best regards,
>>> Even Andr?
>>>
>>>
>> Thanks, Even, that would be great! I might have to give up on the
>> ultramonkey solution, since I can't find anyone who has made it work. It's
>> too bad, because it would fit well with the rest of our architecture.
>>
>> Mike van Lammeren
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From davis.erwin at gmail.com  Fri Oct  2 08:50:58 2009
From: davis.erwin at gmail.com (Erwin Davis)
Date: Fri, 2 Oct 2009 11:50:58 -0400
Subject: [Freeswitch-users] looking for qualified and cheap TISP
Message-ID: <a38a91d50910020850v5e19b8e0r751823b9d1e4ba5d@mail.gmail.com>

Hi, I installed internal freeSWITCH in my LAN and want to see if I can dial
out / dial in. Could anyone suggest one Telephone Service provider which is
capable of connecting with FreeSWITCH and CHEAP/even FREE if possible? At
this moment, I want to prove it is working with the real outside world.
Thanks,

e
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From jerry.richards at teotech.com  Fri Oct  2 10:12:45 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 2 Oct 2009 10:12:45 -0700
Subject: [Freeswitch-users] FS Does Not Relay Presence
	PUBLISHToSubscribing Phones
In-Reply-To: <ebbc2d550910012014ic655d84tbf000c084a98e51d@mail.gmail.com>
References: <B4B4D2747EEC4852B41A7D8D2DA36006@greyhawk.tonecommander.com><191c3a030910010935s69e65569v76f4d26503a94d96@mail.gmail.com><9B5214D274044021A57C6857844C10BA@greyhawk.tonecommander.com>
	<ebbc2d550910012014ic655d84tbf000c084a98e51d@mail.gmail.com>
Message-ID: <E407DD09040D408F8A130C2BFAFFCD17@greyhawk.tonecommander.com>

Okay, I put a log up on the pastebin that shows the PUBLISH event coming
from a CounterPath Bria Professional phone.  For some reason, FS is getting
an error and not relaying the presence status to the subscriber.
 
Best Regards,
Jerry


  _____  

From: Jo?o Mesquita [mailto:jmesquita at freeswitch.org] 
Sent: Thursday, October 01, 2009 8:14 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Does Not Relay Presence
PUBLISHToSubscribing Phones


Piece of advice, don't ask, just do it. ;)

jmesquita


On Thu, Oct 1, 2009 at 3:29 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:


If you have time to take a look, I could put a trace in the pastebin?
 
Jerry


  _____  


From: Jerry Richards [mailto:jerry.richards at teotech.com] 

Sent: Thursday, October 01, 2009 10:29 AM 

To: 'freeswitch-users at lists.freeswitch.org'

Subject: RE: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
ToSubscribing Phones


I am using two Bria Professional Version 2.5.4 Build 54835 softphones.
 
Thanks,
Jerry


  _____  


From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 

Sent: Thursday, October 01, 2009 9:36 AM 

To: freeswitch-users at lists.freeswitch.org

Subject: Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
ToSubscribing Phones


which phone is it,
we tested it with eyebeam and it appears to work for us.



On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards <jerry.richards at teotech.com>
wrote:



By the way, I see the following lines at the FS console, which might be a
clue as to why this is happening.  Could someone point me toward what might
cause this?  I set the "manage-presence" parameter to "true" in each XML
file where I saw it defined.

[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[WARNING] sofia_presence.c:565  192.168.72.38 is an alias, skipping


Best Regards,
Jerry



-----Original Message-----
From: Jerry Richards [mailto:jerry.richards at teotech.com]
Sent: Wednesday, September 30, 2009 9:12 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones

I have two phones configured to subscribe to each other's presence status.
When I change the presence status in one phone, I see the SIP PUBLISH
message going to FS, but I don't see FS relaying that presence status to the
subscribing phone.  Does anyone know why?

Best Regards,
Jerry


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:213-799-1400



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




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From Russell.Mosemann at cune.org  Fri Oct  2 10:24:18 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Fri, 2 Oct 2009 17:24:18 -0000
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <87f2f3b90910020908p1aa5f5d4o8b69894cd4a58bef@mail.gmail.com>
Message-ID: <20091002172418.9E8AF2E19CE@mail.cune.org>

Michael Collins <msc at freeswitch.org> said:

> do something like:
> name => XYZ Corp
> number => 8005551212

I was expecting that information to be filled with the caller name and
number. It doesn't really help if someone calls from outside the
business, and it looks like my business is calling me. Doesn't OpenZAP
extract caller information from a PRI T1?

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!




From mgende at gendesign.com  Fri Oct  2 10:27:55 2009
From: mgende at gendesign.com (Michael Gende)
Date: Fri, 2 Oct 2009 12:27:55 -0500
Subject: [Freeswitch-users] looking for qualified and cheap TISP
In-Reply-To: <a38a91d50910020850v5e19b8e0r751823b9d1e4ba5d@mail.gmail.com>
References: <a38a91d50910020850v5e19b8e0r751823b9d1e4ba5d@mail.gmail.com>
Message-ID: <a87d43fd0910021027q4d4766c9n2930a92d1b7fce3f@mail.gmail.com>

Hey Erwin,

Can't give any personal recommendations, but on the FS site, there's several
examples. Some have "free" or "cheap" in the name. Might be a good place to
start, plus the means to connect is demonstrated to-boot.

http://wiki.freeswitch.org/wiki/SIP_Provider_Examples

Regards,

Mike G.

On Fri, Oct 2, 2009 at 10:50 AM, Erwin Davis <davis.erwin at gmail.com> wrote:

> Hi, I installed internal freeSWITCH in my LAN and want to see if I can dial
> out / dial in. Could anyone suggest one Telephone Service provider which is
> capable of connecting with FreeSWITCH and CHEAP/even FREE if possible? At
> this moment, I want to prove it is working with the real outside world.
> Thanks,
>
>
>
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From mike at van.lammeren.net  Fri Oct  2 10:30:45 2009
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Fri, 2 Oct 2009 13:30:45 -0400
Subject: [Freeswitch-users] looking for qualified and cheap TISP
In-Reply-To: <a38a91d50910020850v5e19b8e0r751823b9d1e4ba5d@mail.gmail.com>
References: <a38a91d50910020850v5e19b8e0r751823b9d1e4ba5d@mail.gmail.com>
Message-ID: <2F7DF2F3-4A48-426C-91F6-9A2F79E17B06@van.lammeren.net>

Hello!

For dialing in, there are a number of sites that provide free DIDs,  
such as http://freephonelines.ca/ .

For dialing out, you can get 1.5 cents per minute calling to N.  
America from http://les.net/ .

Mike

On 2009-10-02, at 11:50 AM, Erwin Davis <davis.erwin at gmail.com> wrote:

> Hi, I installed internal freeSWITCH in my LAN and want to see if I  
> can dial out / dial in. Could anyone suggest one Telephone Service  
> provider which is capable of connecting with FreeSWITCH and CHEAP/ 
> even FREE if possible? At this moment, I want to prove it is working  
> with the real outside world. Thanks,
>
> e
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From anthony.minessale at gmail.com  Fri Oct  2 10:52:17 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 2 Oct 2009 12:52:17 -0500
Subject: [Freeswitch-users] FS Does Not Relay Presence
	PUBLISHToSubscribing Phones
In-Reply-To: <E407DD09040D408F8A130C2BFAFFCD17@greyhawk.tonecommander.com>
References: <B4B4D2747EEC4852B41A7D8D2DA36006@greyhawk.tonecommander.com>
	<191c3a030910010935s69e65569v76f4d26503a94d96@mail.gmail.com>
	<9B5214D274044021A57C6857844C10BA@greyhawk.tonecommander.com>
	<ebbc2d550910012014ic655d84tbf000c084a98e51d@mail.gmail.com>
	<E407DD09040D408F8A130C2BFAFFCD17@greyhawk.tonecommander.com>
Message-ID: <191c3a030910021052y2e074b4r648024507206e25@mail.gmail.com>

connect to sqlite directly with sqlite3 app and try that sql stmt and see
why it doesn't match anything.

sqlite3 /usr/local/freeswitch/db/sofia_reg_internal.db

select
sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Away','away','
192.168.72.38',sip_presence.status,sip_presence.rpid from sip_subscriptions
left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user
and sip_subscriptions.sub_to_host=sip_presence.sip_host and
sip_subscriptions.profile_name=sip_presence.profile_name) where
(event='presence'
or event='presence') and sub_to_user='1001' and (sub_to_host='192.168.72.38'
or presence_hosts like '%192.168.72.38%') and (sip_subscriptions.profile_name
= 'external' or sip_subscriptions.presence_hosts !=
sip_subscriptions.sub_to_host)


On Fri, Oct 2, 2009 at 12:12 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>  Okay, I put a log up on the pastebin that shows the PUBLISH event coming
> from a CounterPath Bria Professional phone.  For some reason, FS is getting
> an error and not relaying the presence status to the subscriber.
>
> Best Regards,
> Jerry
>
>  ------------------------------
> *From:* Jo?o Mesquita [mailto:jmesquita at freeswitch.org]
> *Sent:* Thursday, October 01, 2009 8:14 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] FS Does Not Relay Presence
> PUBLISHToSubscribing Phones
>
> Piece of advice, don't ask, just do it. ;)
>
> jmesquita
>
> On Thu, Oct 1, 2009 at 3:29 PM, Jerry Richards <jerry.richards at teotech.com
> > wrote:
>
>>  If you have time to take a look, I could put a trace in the pastebin?
>>
>> Jerry
>>
>>  ------------------------------
>>  *From:* Jerry Richards [mailto:jerry.richards at teotech.com]
>> *Sent:* Thursday, October 01, 2009 10:29 AM
>> *To:* 'freeswitch-users at lists.freeswitch.org'
>> *Subject:* RE: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
>> ToSubscribing Phones
>>
>>  I am using two Bria Professional Version 2.5.4 Build 54835 softphones.
>>
>> Thanks,
>> Jerry
>>
>>  ------------------------------
>>  *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com]
>> *Sent:* Thursday, October 01, 2009 9:36 AM
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
>> ToSubscribing Phones
>>
>>   which phone is it,
>> we tested it with eyebeam and it appears to work for us.
>>
>>
>> On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards <
>> jerry.richards at teotech.com> wrote:
>>
>>>
>>> By the way, I see the following lines at the FS console, which might be a
>>> clue as to why this is happening.  Could someone point me toward what
>>> might
>>> cause this?  I set the "manage-presence" parameter to "true" in each XML
>>> file where I saw it defined.
>>>
>>> [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal)
>>> [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
>>>   ...
>>> [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6)
>>> [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
>>>   ...
>>> [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external)
>>> [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
>>>   ...
>>> [WARNING] sofia_presence.c:565  192.168.72.38 is an alias, skipping
>>>
>>>
>>> Best Regards,
>>> Jerry
>>>
>>>
>>> -----Original Message-----
>>> From: Jerry Richards [mailto:jerry.richards at teotech.com]
>>> Sent: Wednesday, September 30, 2009 9:12 AM
>>> To: 'freeswitch-users at lists.freeswitch.org'
>>> Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones
>>>
>>> I have two phones configured to subscribe to each other's presence
>>> status.
>>> When I change the presence status in one phone, I see the SIP PUBLISH
>>> message going to FS, but I don't see FS relaying that presence status to
>>> the
>>> subscribing phone.  Does anyone know why?
>>>
>>> Best Regards,
>>> Jerry
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From csa at nowthor.com  Fri Oct  2 11:12:01 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Fri, 02 Oct 2009 14:12:01 -0400
Subject: [Freeswitch-users] looking for qualified and cheap TISP
In-Reply-To: <a38a91d50910020850v5e19b8e0r751823b9d1e4ba5d@mail.gmail.com>
References: <a38a91d50910020850v5e19b8e0r751823b9d1e4ba5d@mail.gmail.com>
Message-ID: <4AC64271.2050707@nowthor.com>

Hi!

Callcentric 
<http://wiki.freeswitch.org/wiki/Provider_Configuration:_Callcentric> 
offers a package called IP Freedom 
<http://www.callcentric.com/rate_plans01.php>. It costs nothing and will 
allow you to test FS.

Carlos

Erwin Davis wrote:
> Hi, I installed internal freeSWITCH in my LAN and want to see if I can 
> dial out / dial in. Could anyone suggest one Telephone Service 
> provider which is capable of connecting with FreeSWITCH and CHEAP/even 
> FREE if possible? At this moment, I want to prove it is working with 
> the real outside world. Thanks,
>
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From msc at freeswitch.org  Fri Oct  2 11:16:58 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 2 Oct 2009 11:16:58 -0700
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <20091002172418.9E8AF2E19CE@mail.cune.org>
References: <87f2f3b90910020908p1aa5f5d4o8b69894cd4a58bef@mail.gmail.com>
	<20091002172418.9E8AF2E19CE@mail.cune.org>
Message-ID: <87f2f3b90910021116l209a44bcg124016e58735ca37@mail.gmail.com>

On Fri, Oct 2, 2009 at 10:24 AM, <Russell.Mosemann at cune.org> wrote:

> Michael Collins <msc at freeswitch.org> said:
>
> > do something like:
> > name => XYZ Corp
> > number => 8005551212
>
> I was expecting that information to be filled with the caller name and
> number. It doesn't really help if someone calls from outside the
> business, and it looks like my business is calling me. Doesn't OpenZAP
> extract caller information from a PRI T1?
>

Can you pastebin a dialplan snippet (or put it here) so I can see what
you're doing?
-MC
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From jerry.richards at teotech.com  Fri Oct  2 11:28:07 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 2 Oct 2009 11:28:07 -0700
Subject: [Freeswitch-users] FS Does Not Relay
	PresencePUBLISHToSubscribing Phones
In-Reply-To: <191c3a030910021052y2e074b4r648024507206e25@mail.gmail.com>
References: <B4B4D2747EEC4852B41A7D8D2DA36006@greyhawk.tonecommander.com><191c3a030910010935s69e65569v76f4d26503a94d96@mail.gmail.com><9B5214D274044021A57C6857844C10BA@greyhawk.tonecommander.com><ebbc2d550910012014ic655d84tbf000c084a98e51d@mail.gmail.com><E407DD09040D408F8A130C2BFAFFCD17@greyhawk.tonecommander.com>
	<191c3a030910021052y2e074b4r648024507206e25@mail.gmail.com>
Message-ID: <57502048DB624686B96B8AE76F697AA7@greyhawk.tonecommander.com>

I put the sqlite3 select query in the paste bin, and prior to that, I
entered the .dump command.  The select command came back with a "...>"
prompt which I don't understand.  I don't know enough about sqlite3 to know
what that means?
 
Best Regards,
Jerry


  _____  

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Friday, October 02, 2009 10:52 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Does Not Relay
PresencePUBLISHToSubscribing Phones


connect to sqlite directly with sqlite3 app and try that sql stmt and see
why it doesn't match anything.

sqlite3 /usr/local/freeswitch/db/sofia_reg_internal.db

select
sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_hos
t,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscripti
ons.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscripti
ons.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subsc
riptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,
'Away','away','192.168.72.38',sip_presence.status,sip_presence.rpid from
sip_subscriptions left join sip_presence on
(sip_subscriptions.sub_to_user=sip_presence.sip_user and
sip_subscriptions.sub_to_host=sip_presence.sip_host and
sip_subscriptions.profile_name=sip_presence.profile_name) where
(event='presence' or event='presence') and sub_to_user='1001' and
(sub_to_host='192.168.72.38' or presence_hosts like '%192.168.72.38%') and
(sip_subscriptions.profile_name = 'external' or
sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host)



On Fri, Oct 2, 2009 at 12:12 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:


Okay, I put a log up on the pastebin that shows the PUBLISH event coming
from a CounterPath Bria Professional phone.  For some reason, FS is getting
an error and not relaying the presence status to the subscriber.
 
Best Regards,
Jerry


  _____  

From: Jo?o Mesquita [mailto:jmesquita at freeswitch.org] 
Sent: Thursday, October 01, 2009 8:14 PM 

To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Does Not Relay Presence
PUBLISHToSubscribing Phones


Piece of advice, don't ask, just do it. ;)

jmesquita


On Thu, Oct 1, 2009 at 3:29 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:


If you have time to take a look, I could put a trace in the pastebin?
 
Jerry


  _____  


From: Jerry Richards [mailto:jerry.richards at teotech.com] 

Sent: Thursday, October 01, 2009 10:29 AM 

To: 'freeswitch-users at lists.freeswitch.org'

Subject: RE: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
ToSubscribing Phones


I am using two Bria Professional Version 2.5.4 Build 54835 softphones.
 
Thanks,
Jerry


  _____  


From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 

Sent: Thursday, October 01, 2009 9:36 AM 

To: freeswitch-users at lists.freeswitch.org

Subject: Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
ToSubscribing Phones


which phone is it,
we tested it with eyebeam and it appears to work for us.



On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards <jerry.richards at teotech.com>
wrote:



By the way, I see the following lines at the FS console, which might be a
clue as to why this is happening.  Could someone point me toward what might
cause this?  I set the "manage-presence" parameter to "true" in each XML
file where I saw it defined.

[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[WARNING] sofia_presence.c:565  192.168.72.38 is an alias, skipping


Best Regards,
Jerry



-----Original Message-----
From: Jerry Richards [mailto:jerry.richards at teotech.com]
Sent: Wednesday, September 30, 2009 9:12 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones

I have two phones configured to subscribe to each other's presence status.
When I change the presence status in one phone, I see the SIP PUBLISH
message going to FS, but I don't see FS relaying that presence status to the
subscribing phone.  Does anyone know why?

Best Regards,
Jerry


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:213-799-1400



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org






-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:213-799-1400


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From rupa at rupa.com  Fri Oct  2 11:42:29 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 2 Oct 2009 13:42:29 -0500
Subject: [Freeswitch-users] FS Does Not Relay
	PresencePUBLISHToSubscribing Phones
In-Reply-To: <57502048DB624686B96B8AE76F697AA7@greyhawk.tonecommander.com>
References: <B4B4D2747EEC4852B41A7D8D2DA36006@greyhawk.tonecommander.com>
	<191c3a030910010935s69e65569v76f4d26503a94d96@mail.gmail.com>
	<9B5214D274044021A57C6857844C10BA@greyhawk.tonecommander.com>
	<ebbc2d550910012014ic655d84tbf000c084a98e51d@mail.gmail.com>
	<E407DD09040D408F8A130C2BFAFFCD17@greyhawk.tonecommander.com>
	<191c3a030910021052y2e074b4r648024507206e25@mail.gmail.com>
	<57502048DB624686B96B8AE76F697AA7@greyhawk.tonecommander.com>
Message-ID: <b39273530910021142p546680felc6c6ec19d86c922f@mail.gmail.com>

You are missing the trailing ;

On Fri, Oct 2, 2009 at 1:28 PM, Jerry Richards
<jerry.richards at teotech.com> wrote:
> I put the sqlite3 select query in the paste bin, and prior to that, I
> entered the .dump command.? The select command came back with a "...>"
> prompt which I don't understand.? I don't know enough about sqlite3 to know
> what that means?
>
> Best Regards,
> Jerry


-- 
-Rupa



From Russell.Mosemann at cune.org  Fri Oct  2 11:53:59 2009
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Fri, 2 Oct 2009 13:53:59 -0500
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <87f2f3b90910021116l209a44bcg124016e58735ca37@mail.gmail.com>
References: <87f2f3b90910020908p1aa5f5d4o8b69894cd4a58bef@mail.gmail.com><20091002172418.9E8AF2E19CE@mail.cune.org>
	<87f2f3b90910021116l209a44bcg124016e58735ca37@mail.gmail.com>
Message-ID: <E2C130F6CC9B4F59B069CEF8D0B0AC42@cune.pri>

> Can you pastebin a dialplan snippet (or put it here) so I can see what
> you're doing?
> -MC

It is the stock FS configuration with a small change. We're still testing things, getting them to work. This is from public.xml. It detects calls to internal 71xx extensions and transfers them. The transfer works. Do some additional variables need to be set here?

<extension name="public_extensions">
  <condition field="destination_number" expression="^(10[01][0-9]|71\d{2})$">
    <action application="transfer" data="$1 XML default"/>
  </condition>
</extension>

--
Russell Mosemann




From ujjval at simplesignal.com  Fri Oct  2 12:10:40 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Fri, 2 Oct 2009 12:10:40 -0700
Subject: [Freeswitch-users] Asterisk vs Freeswitch
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701C61EE3@EXMBXCLUS01.citservers.local>

Is there benchmark test results on how many simultaneous calls Freeswtich can do (with RTP anchored through it) vs the Asterisk.

For any hardware/CPU/Mem  that anyone may have performed this performance testing.

Any numbers on average how much Freeswitch scores over the Asterisk in terms of capacity will help.


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From msc at freeswitch.org  Fri Oct  2 12:18:30 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 2 Oct 2009 12:18:30 -0700
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <E2C130F6CC9B4F59B069CEF8D0B0AC42@cune.pri>
References: <87f2f3b90910020908p1aa5f5d4o8b69894cd4a58bef@mail.gmail.com>
	<20091002172418.9E8AF2E19CE@mail.cune.org>
	<87f2f3b90910021116l209a44bcg124016e58735ca37@mail.gmail.com>
	<E2C130F6CC9B4F59B069CEF8D0B0AC42@cune.pri>
Message-ID: <87f2f3b90910021218i434f67cbjc33073f8550c8d1a@mail.gmail.com>

On Fri, Oct 2, 2009 at 11:53 AM, Russell Mosemann <Russell.Mosemann at cune.org
> wrote:

> > Can you pastebin a dialplan snippet (or put it here) so I can see what
> > you're doing?
> > -MC
>
> It is the stock FS configuration with a small change. We're still testing
> things, getting them to work. This is from public.xml. It detects calls to
> internal 71xx extensions and transfers them. The transfer works. Do some
> additional variables need to be set here?
>
> <extension name="public_extensions">
>  <condition field="destination_number"
> expression="^(10[01][0-9]|71\d{2})$">
>    <action application="transfer" data="$1 XML default"/>
>  </condition>
> </extension>
>
> cool. can you pastebin a debug log on an incoming call?
-MC
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From kadantsev.d at gmail.com  Fri Oct  2 13:00:42 2009
From: kadantsev.d at gmail.com (Dmitry Kadantsev)
Date: Fri, 2 Oct 2009 22:00:42 +0200
Subject: [Freeswitch-users] Asterisk vs Freeswitch
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71701C61EE3@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701C61EE3@EXMBXCLUS01.citservers.local>
Message-ID: <681a20520910021300g131cd87j7c1fa6c6c9ec7f3e@mail.gmail.com>

Hi,

for example here: http://blogs.zdnet.com/Greenfield/?p=214

We *replaced* a cluster of *10 Asterisk* servers with a *single
FreeSwitch*server, said Chris Parker, director of systems for a large
publicly traded
CLEC. Parker says hes getting several hundred concurrent calls on a single,
dual-core box thats also doing all of the media processing, a
computationally intensive task.




--
Best regards,
Dmitry Kadantsev

http://www.kadantsev.com - Home page (MS Silverlight required)
http://www.doxwox.com - Best web meeting and online collaboration tool


On Fri, Oct 2, 2009 at 9:10 PM, Ujjval Karihaloo <ujjval at simplesignal.com>wrote:

>  Is there benchmark test results on how many simultaneous calls Freeswtich
> can do (with RTP anchored through it) vs the Asterisk.
>
>
>
> For any hardware/CPU/Mem  that anyone may have performed this performance
> testing.
>
>
>
> Any numbers on average how much Freeswitch scores over the Asterisk in
> terms of capacity will help.
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Russell.Mosemann at cune.org  Fri Oct  2 14:53:54 2009
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Fri, 2 Oct 2009 16:53:54 -0500
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <87f2f3b90910021218i434f67cbjc33073f8550c8d1a@mail.gmail.com>
References: <87f2f3b90910020908p1aa5f5d4o8b69894cd4a58bef@mail.gmail.com><20091002172418.9E8AF2E19CE@mail.cune.org><87f2f3b90910021116l209a44bcg124016e58735ca37@mail.gmail.com><E2C130F6CC9B4F59B069CEF8D0B0AC42@cune.pri>
	<87f2f3b90910021218i434f67cbjc33073f8550c8d1a@mail.gmail.com>
Message-ID: <8E079E8CB37E4FA49363D5315F6E878E@cune.pri>

> cool. can you pastebin a debug log on an incoming call?
> -MC

Here you go.

http://pastebin.freeswitch.org/10570

One thing I notice is that in the second line, the caller number is missing.

2009-10-02 16:42:39.539736 [NOTICE] ozmod_libpri.c:772 -- Ring on channel 1:1 (from  to 7100)

If libpri doesn't know the number, then it's probably not being sent by the Hicomm.

--
Russell Mosemann




From msc at freeswitch.org  Fri Oct  2 17:00:08 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 2 Oct 2009 17:00:08 -0700
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <8E079E8CB37E4FA49363D5315F6E878E@cune.pri>
References: <87f2f3b90910020908p1aa5f5d4o8b69894cd4a58bef@mail.gmail.com>
	<20091002172418.9E8AF2E19CE@mail.cune.org>
	<87f2f3b90910021116l209a44bcg124016e58735ca37@mail.gmail.com>
	<E2C130F6CC9B4F59B069CEF8D0B0AC42@cune.pri>
	<87f2f3b90910021218i434f67cbjc33073f8550c8d1a@mail.gmail.com>
	<8E079E8CB37E4FA49363D5315F6E878E@cune.pri>
Message-ID: <87f2f3b90910021700h3431bd78q3b4f6ced255ecee5@mail.gmail.com>

On Fri, Oct 2, 2009 at 2:53 PM, Russell Mosemann
<Russell.Mosemann at cune.org>wrote:

> > cool. can you pastebin a debug log on an incoming call?
> > -MC
>
> Here you go.
>
> http://pastebin.freeswitch.org/10570
>
> One thing I notice is that in the second line, the caller number is
> missing.
>
> 2009-10-02 16:42:39.539736 [NOTICE] ozmod_libpri.c:772 -- Ring on channel
> 1:1 (from  to 7100)
>
> If libpri doesn't know the number, then it's probably not being sent by the
> Hicomm.
>
> Exactly. Turn on q931 debugging and try again:

oz libpri debug 1 all

PB the results again and we'll check it out.
-MC
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From Russell.Mosemann at cune.org  Fri Oct  2 17:54:34 2009
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Fri, 2 Oct 2009 19:54:34 -0500
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <87f2f3b90910021700h3431bd78q3b4f6ced255ecee5@mail.gmail.com>
References: <87f2f3b90910020908p1aa5f5d4o8b69894cd4a58bef@mail.gmail.com><20091002172418.9E8AF2E19CE@mail.cune.org><87f2f3b90910021116l209a44bcg124016e58735ca37@mail.gmail.com><E2C130F6CC9B4F59B069CEF8D0B0AC42@cune.pri><87f2f3b90910021218i434f67cbjc33073f8550c8d1a@mail.gmail.com><8E079E8CB37E4FA49363D5315F6E878E@cune.pri>
	<87f2f3b90910021700h3431bd78q3b4f6ced255ecee5@mail.gmail.com>
Message-ID: <191DBBEF043B4EFE8F680B82C1038FC7@cune.pri>

> Exactly. Turn on q931 debugging and try again:
> 
> oz libpri debug 1 all
> PB the results again and we'll check it out.
> -MC

Here's the next one. I'm not sure what to look for, but nothing pops out right away.

http://pastebin.freeswitch.org/10571

--
Russell Mosemann




From thangappan143 at gmail.com  Fri Oct  2 21:24:42 2009
From: thangappan143 at gmail.com (Thangappan.M)
Date: Sat, 3 Oct 2009 09:54:42 +0530
Subject: [Freeswitch-users] Need Help in Getting DTMF
Message-ID: <7aa29e790910022124h320de58dp16f1a19aca08bb6@mail.gmail.com>

Dear all,

          I am in the process of implementing IVR server in Perl using event
outbound socket.

Let take the following scenario.

         There are three menus in the IVR. First menu will authenticate you,
second menu get the option value from you,. third menu will give the you the
result.

           You already know  all the numbers that you could give. So when
the call answered you are giving the value in ONE SHOT.

         Is it possible to get all the DTMF values in one shot in
freeswitch?

   It should have facility to recollect DTMF values and clear the DTMF
values.

-- 
Regards,
Thangappan.M
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From vinuth.madinur at gmail.com  Fri Oct  2 22:21:17 2009
From: vinuth.madinur at gmail.com (Vinuth Madinur)
Date: Sat, 3 Oct 2009 10:51:17 +0530
Subject: [Freeswitch-users] Need Help in Getting DTMF
In-Reply-To: <7aa29e790910022124h320de58dp16f1a19aca08bb6@mail.gmail.com>
References: <7aa29e790910022124h320de58dp16f1a19aca08bb6@mail.gmail.com>
Message-ID: <910309030910022221h31a1fcd0h9f41e1aa17f0c7a3@mail.gmail.com>

You can use play_and_get_digits command or the read command.
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_read
<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_read>

Thanks,
Vinuth.


On Sat, Oct 3, 2009 at 9:54 AM, Thangappan.M <thangappan143 at gmail.com>wrote:

> Dear all,
>
>           I am in the process of implementing IVR server in Perl using
> event outbound socket.
>
> Let take the following scenario.
>
>          There are three menus in the IVR. First menu will authenticate
> you, second menu get the option value from you,. third menu will give the
> you the result.
>
>            You already know  all the numbers that you could give. So when
> the call answered you are giving the value in ONE SHOT.
>
>          Is it possible to get all the DTMF values in one shot in
> freeswitch?
>
>    It should have facility to recollect DTMF values and clear the DTMF
> values.
>
> --
> Regards,
> Thangappan.M
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From keith.wood2000 at gmail.com  Fri Oct  2 23:44:50 2009
From: keith.wood2000 at gmail.com (Keith Wood)
Date: Sat, 3 Oct 2009 14:44:50 +0800
Subject: [Freeswitch-users] wav files compression
Message-ID: <b7737fe70910022344y72cdebc4ieb0b580ce7b7905f@mail.gmail.com>

I am working on an implementation for managing thousands of IVR within an
organization.  Right now, I am storing all audio files in wav format, but it
quickly become unmanagable because the size of these wav files ( 8 bits mono
) quickly consuming a lot of the disk space.

Is there anyway I can store those audio files and still have high quality
audio for IVR?  I know mp3 is smaller but freeswitch does not support it.

any ideas?

keith
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From dujinfang at gmail.com  Sat Oct  3 00:12:57 2009
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 3 Oct 2009 15:12:57 +0800
Subject: [Freeswitch-users] wav files compression
In-Reply-To: <b7737fe70910022344y72cdebc4ieb0b580ce7b7905f@mail.gmail.com>
References: <b7737fe70910022344y72cdebc4ieb0b580ce7b7905f@mail.gmail.com>
Message-ID: <23f91030910030012k572a544tcefa32c33dc7efef@mail.gmail.com>

FS support recording to mp3 directly through mod_shout but you might not
want to use that for performance reason.

You can use lame to convert .wav to .mp3 regularly( by crontab if you on
linux) or immediately after record(by using iwatch, or listening to event
socket to see when the record is done ).

2009/10/3 Keith Wood <keith.wood2000 at gmail.com>

>
> I am working on an implementation for managing thousands of IVR within an
> organization.  Right now, I am storing all audio files in wav format, but it
> quickly become unmanagable because the size of these wav files ( 8 bits mono
> ) quickly consuming a lot of the disk space.
>
> Is there anyway I can store those audio files and still have high quality
> audio for IVR?  I know mp3 is smaller but freeswitch does not support it.
>
> any ideas?
>
> keith
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From thangappan143 at gmail.com  Sat Oct  3 03:42:55 2009
From: thangappan143 at gmail.com (Thangappan.M)
Date: Sat, 3 Oct 2009 16:12:55 +0530
Subject: [Freeswitch-users] Need Help in Getting DTMF
In-Reply-To: <7aa29e790910022124h320de58dp16f1a19aca08bb6@mail.gmail.com>
References: <7aa29e790910022124h320de58dp16f1a19aca08bb6@mail.gmail.com>
Message-ID: <7aa29e790910030342n222d7d9q8170f0a8fa537840@mail.gmail.com>

Can you please give some example?

Because I have tried it using playAndGetDigits() application only.

my need is " User can give input at any time " It should be captured.

Is there any internal mechanism avail in freeswitch?
Or

Shall we do it?


On Sat, Oct 3, 2009 at 9:54 AM, Thangappan.M <thangappan143 at gmail.com>wrote:

> Dear all,
>
>           I am in the process of implementing IVR server in Perl using
> event outbound socket.
>
> Let take the following scenario.
>
>          There are three menus in the IVR. First menu will authenticate
> you, second menu get the option value from you,. third menu will give the
> you the result.
>
>            You already know  all the numbers that you could give. So when
> the call answered you are giving the value in ONE SHOT.
>
>          Is it possible to get all the DTMF values in one shot in
> freeswitch?
>
>    It should have facility to recollect DTMF values and clear the DTMF
> values.
>
> --
> Regards,
> Thangappan.M
>



-- 
Regards,
Thangappan.M
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From brian at freeswitch.org  Sat Oct  3 07:52:54 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 3 Oct 2009 09:52:54 -0500
Subject: [Freeswitch-users] wav files compression
In-Reply-To: <b7737fe70910022344y72cdebc4ieb0b580ce7b7905f@mail.gmail.com>
References: <b7737fe70910022344y72cdebc4ieb0b580ce7b7905f@mail.gmail.com>
Message-ID: <709C5BD7-C916-4757-9AEE-C862D7F60053@freeswitch.org>

MP3 is NOT recommend and if WAV files are too large you can mosey on  
down to the local Best Buy and snag 1.5TB of disk for like $119  
dollars.  Disk is cheap.

/b

On Oct 3, 2009, at 1:44 AM, Keith Wood wrote:

>
> I am working on an implementation for managing thousands of IVR  
> within an organization.  Right now, I am storing all audio files in  
> wav format, but it quickly become unmanagable because the size of  
> these wav files ( 8 bits mono ) quickly consuming a lot of the disk  
> space.
>
> Is there anyway I can store those audio files and still have high  
> quality audio for IVR?  I know mp3 is smaller but freeswitch does  
> not support it.
>
> any ideas?
>
> keith
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From frank at carmickle.com  Sat Oct  3 09:16:36 2009
From: frank at carmickle.com (Frank Carmickle)
Date: Sat, 3 Oct 2009 12:16:36 -0400
Subject: [Freeswitch-users] voiptalk.org register 904
Message-ID: <20091003161635.GG17256@base.carmickle.com>

I hope this is more helpful.

http://carmickle.com/fs-2009-10-02.txt

<include>
      <gateway name="voiptalk.org">
        <param name="username" value="81100068"/>
        <param name="password" value="my_pass"/>
        <param name="from-user" value="81100068"/>
        <param name="from-domain" value="voiptalk.org"/>
    <param name="proxy" value="voiptalk.org"/>
        <param name="expire-seconds" value="600"/>
        <param name="register" value="true"/>
        <param name="retry-seconds" value="30"/>
        <param name="extension" value="1000"/>
        <param name="contact-params" value="domain_name=$${domain}"/>
        <param name="context" value="public"/>
      </gateway>
</include>
 
Let me know what you think.  Thanks.
--FC



From diego.viola at gmail.com  Sat Oct  3 10:07:39 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 3 Oct 2009 17:07:39 +0000
Subject: [Freeswitch-users] wav files compression
In-Reply-To: <709C5BD7-C916-4757-9AEE-C862D7F60053@freeswitch.org>
References: <b7737fe70910022344y72cdebc4ieb0b580ce7b7905f@mail.gmail.com>
	<709C5BD7-C916-4757-9AEE-C862D7F60053@freeswitch.org>
Message-ID: <86a32abc0910031007ted4b6bfk1a567574af91ecae@mail.gmail.com>

Why is not recommended?

On Sat, Oct 3, 2009 at 2:52 PM, Brian West <brian at freeswitch.org> wrote:

> MP3 is NOT recommend and if WAV files are too large you can mosey on
> down to the local Best Buy and snag 1.5TB of disk for like $119
> dollars.  Disk is cheap.
>
> /b
>
> On Oct 3, 2009, at 1:44 AM, Keith Wood wrote:
>
> >
> > I am working on an implementation for managing thousands of IVR
> > within an organization.  Right now, I am storing all audio files in
> > wav format, but it quickly become unmanagable because the size of
> > these wav files ( 8 bits mono ) quickly consuming a lot of the disk
> > space.
> >
> > Is there anyway I can store those audio files and still have high
> > quality audio for IVR?  I know mp3 is smaller but freeswitch does
> > not support it.
> >
> > any ideas?
> >
> > keith
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sat Oct  3 10:27:54 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 3 Oct 2009 12:27:54 -0500
Subject: [Freeswitch-users] wav files compression
In-Reply-To: <86a32abc0910031007ted4b6bfk1a567574af91ecae@mail.gmail.com>
References: <b7737fe70910022344y72cdebc4ieb0b580ce7b7905f@mail.gmail.com>
	<709C5BD7-C916-4757-9AEE-C862D7F60053@freeswitch.org>
	<86a32abc0910031007ted4b6bfk1a567574af91ecae@mail.gmail.com>
Message-ID: <84976809-0961-410D-8733-0C0449365B1A@freeswitch.org>

Lets see... mp3 decoding is heavy compared to wav files.. its 2009 and  
disk is cheap and fast why worry about it?  Not sure you wanna scale  
mp3 playback to the same level you can wav files.

/b

On Oct 3, 2009, at 12:07 PM, Diego Viola wrote:

> Why is not recommended?
>
> On Sat, Oct 3, 2009 at 2:52 PM, Brian West <brian at freeswitch.org>  
> wrote:
> MP3 is NOT recommend and if WAV files are too large you can mosey on
> down to the local Best Buy and snag 1.5TB of disk for like $119
> dollars.  Disk is cheap.
>
> /b

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From steveu at coppice.org  Sat Oct  3 10:47:22 2009
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 04 Oct 2009 01:47:22 +0800
Subject: [Freeswitch-users] wav files compression
In-Reply-To: <86a32abc0910031007ted4b6bfk1a567574af91ecae@mail.gmail.com>
References: <b7737fe70910022344y72cdebc4ieb0b580ce7b7905f@mail.gmail.com>	<709C5BD7-C916-4757-9AEE-C862D7F60053@freeswitch.org>
	<86a32abc0910031007ted4b6bfk1a567574af91ecae@mail.gmail.com>
Message-ID: <4AC78E2A.2090501@coppice.org>

On 10/04/2009 01:07 AM, Diego Viola wrote:
> Why is not recommended?

Square peg. Round hole.
> On Sat, Oct 3, 2009 at 2:52 PM, Brian West <brian at freeswitch.org 
> <mailto:brian at freeswitch.org>> wrote:
>
>     MP3 is NOT recommend and if WAV files are too large you can mosey on
>     down to the local Best Buy and snag 1.5TB of disk for like $119
>     dollars.  Disk is cheap.
>
>     /b
>
>     On Oct 3, 2009, at 1:44 AM, Keith Wood wrote:
>
>     >
>     > I am working on an implementation for managing thousands of IVR
>     > within an organization.  Right now, I am storing all audio files in
>     > wav format, but it quickly become unmanagable because the size of
>     > these wav files ( 8 bits mono ) quickly consuming a lot of the disk
>     > space.
>     >
>     > Is there anyway I can store those audio files and still have high
>     > quality audio for IVR?  I know mp3 is smaller but freeswitch does
>     > not support it.
>     >
>     > any ideas?
>     >
>     > keith
>
Steve




From diego.viola at gmail.com  Sat Oct  3 11:17:26 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 3 Oct 2009 18:17:26 +0000
Subject: [Freeswitch-users] wav files compression
In-Reply-To: <4AC78E2A.2090501@coppice.org>
References: <b7737fe70910022344y72cdebc4ieb0b580ce7b7905f@mail.gmail.com>
	<709C5BD7-C916-4757-9AEE-C862D7F60053@freeswitch.org>
	<86a32abc0910031007ted4b6bfk1a567574af91ecae@mail.gmail.com>
	<4AC78E2A.2090501@coppice.org>
Message-ID: <86a32abc0910031117k17109746s905d65640d548e43@mail.gmail.com>

I see, does Ogg/Vorbis have the same problem?

Diego

On Sat, Oct 3, 2009 at 5:47 PM, Steve Underwood <steveu at coppice.org> wrote:

> On 10/04/2009 01:07 AM, Diego Viola wrote:
> > Why is not recommended?
>
> Square peg. Round hole.
> > On Sat, Oct 3, 2009 at 2:52 PM, Brian West <brian at freeswitch.org
> > <mailto:brian at freeswitch.org>> wrote:
> >
> >     MP3 is NOT recommend and if WAV files are too large you can mosey on
> >     down to the local Best Buy and snag 1.5TB of disk for like $119
> >     dollars.  Disk is cheap.
> >
> >     /b
> >
> >     On Oct 3, 2009, at 1:44 AM, Keith Wood wrote:
> >
> >     >
> >     > I am working on an implementation for managing thousands of IVR
> >     > within an organization.  Right now, I am storing all audio files in
> >     > wav format, but it quickly become unmanagable because the size of
> >     > these wav files ( 8 bits mono ) quickly consuming a lot of the disk
> >     > space.
> >     >
> >     > Is there anyway I can store those audio files and still have high
> >     > quality audio for IVR?  I know mp3 is smaller but freeswitch does
> >     > not support it.
> >     >
> >     > any ideas?
> >     >
> >     > keith
> >
> Steve
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sat Oct  3 11:39:03 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 3 Oct 2009 13:39:03 -0500
Subject: [Freeswitch-users] wav files compression
In-Reply-To: <86a32abc0910031117k17109746s905d65640d548e43@mail.gmail.com>
References: <b7737fe70910022344y72cdebc4ieb0b580ce7b7905f@mail.gmail.com>
	<709C5BD7-C916-4757-9AEE-C862D7F60053@freeswitch.org>
	<86a32abc0910031007ted4b6bfk1a567574af91ecae@mail.gmail.com>
	<4AC78E2A.2090501@coppice.org>
	<86a32abc0910031117k17109746s905d65640d548e43@mail.gmail.com>
Message-ID: <635CFF1F-BFDB-4C00-B04C-D1B59F1648AE@freeswitch.org>

Yes... Why add layers of bullshit on top of audio that is going to  
traverse the public phone network?  PCM raw or ulaw/alaw are the most  
optimal formats.

/b

On Oct 3, 2009, at 1:17 PM, Diego Viola wrote:

> I see, does Ogg/Vorbis have the same problem?
>
> Diego




From steveu at coppice.org  Sat Oct  3 11:47:07 2009
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 04 Oct 2009 02:47:07 +0800
Subject: [Freeswitch-users] wav files compression
In-Reply-To: <86a32abc0910031117k17109746s905d65640d548e43@mail.gmail.com>
References: <b7737fe70910022344y72cdebc4ieb0b580ce7b7905f@mail.gmail.com>	<709C5BD7-C916-4757-9AEE-C862D7F60053@freeswitch.org>	<86a32abc0910031007ted4b6bfk1a567574af91ecae@mail.gmail.com>	<4AC78E2A.2090501@coppice.org>
	<86a32abc0910031117k17109746s905d65640d548e43@mail.gmail.com>
Message-ID: <4AC79C2B.6020505@coppice.org>

On 10/04/2009 02:17 AM, Diego Viola wrote:
> I see, does Ogg/Vorbis have the same problem?
Yep. Anything designed for general purpose audio is going to be a poor 
choice when you want to achieve compact storage of narrowband voice.

> Diego
>
> On Sat, Oct 3, 2009 at 5:47 PM, Steve Underwood <steveu at coppice.org 
> <mailto:steveu at coppice.org>> wrote:
>
>     On 10/04/2009 01:07 AM, Diego Viola wrote:
>     > Why is not recommended?
>
>     Square peg. Round hole.
>     > On Sat, Oct 3, 2009 at 2:52 PM, Brian West <brian at freeswitch.org
>     <mailto:brian at freeswitch.org>
>     > <mailto:brian at freeswitch.org <mailto:brian at freeswitch.org>>> wrote:
>     >
>     >     MP3 is NOT recommend and if WAV files are too large you can
>     mosey on
>     >     down to the local Best Buy and snag 1.5TB of disk for like $119
>     >     dollars.  Disk is cheap.
>     >
>     >     /b
>     >
>     >     On Oct 3, 2009, at 1:44 AM, Keith Wood wrote:
>     >
>     > >
>     > > I am working on an implementation for managing thousands of IVR
>     > > within an organization.  Right now, I am storing all audio
>     files in
>     > > wav format, but it quickly become unmanagable because the size of
>     > > these wav files ( 8 bits mono ) quickly consuming a lot of the
>     disk
>     > > space.
>     > >
>     > > Is there anyway I can store those audio files and still have high
>     > > quality audio for IVR?  I know mp3 is smaller but freeswitch does
>     > > not support it.
>     > >
>     > > any ideas?
>
Steve




From diego.viola at gmail.com  Sat Oct  3 14:26:20 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 3 Oct 2009 21:26:20 +0000
Subject: [Freeswitch-users] CDR page reworked
Message-ID: <86a32abc0910031426q79417376n541edf2a22740275@mail.gmail.com>

Hi FreeSWITCH community.

I just wanted to say that I have reworked this page a bit as it was a bit
poor, feel free to add anything else on it.

http://wiki.freeswitch.org/wiki/CDR

Regards,

Diego
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From tculjaga at gmail.com  Sat Oct  3 16:38:22 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Sun, 4 Oct 2009 01:38:22 +0200
Subject: [Freeswitch-users] wav files compression
In-Reply-To: <86a32abc0910031007ted4b6bfk1a567574af91ecae@mail.gmail.com>
References: <b7737fe70910022344y72cdebc4ieb0b580ce7b7905f@mail.gmail.com>
	<709C5BD7-C916-4757-9AEE-C862D7F60053@freeswitch.org>
	<86a32abc0910031007ted4b6bfk1a567574af91ecae@mail.gmail.com>
Message-ID: <65d96fc80910031638q52c3a595w8cce2209c6cf61f5@mail.gmail.com>

also, you can store files in PCMA/PCMU format and avoid transcoding at
all... and as said disk space is cheap.. go get some...

On Sat, Oct 3, 2009 at 7:07 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Why is not recommended?
>
>
> On Sat, Oct 3, 2009 at 2:52 PM, Brian West <brian at freeswitch.org> wrote:
>
>> MP3 is NOT recommend and if WAV files are too large you can mosey on
>> down to the local Best Buy and snag 1.5TB of disk for like $119
>> dollars.  Disk is cheap.
>>
>> /b
>>
>> On Oct 3, 2009, at 1:44 AM, Keith Wood wrote:
>>
>> >
>> > I am working on an implementation for managing thousands of IVR
>> > within an organization.  Right now, I am storing all audio files in
>> > wav format, but it quickly become unmanagable because the size of
>> > these wav files ( 8 bits mono ) quickly consuming a lot of the disk
>> > space.
>> >
>> > Is there anyway I can store those audio files and still have high
>> > quality audio for IVR?  I know mp3 is smaller but freeswitch does
>> > not support it.
>> >
>> > any ideas?
>> >
>> > keith
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> > users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From nandy1925 at gmail.com  Sun Oct  4 00:28:38 2009
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Sun, 4 Oct 2009 15:28:38 +0800
Subject: [Freeswitch-users] wav files compression
In-Reply-To: <65d96fc80910031638q52c3a595w8cce2209c6cf61f5@mail.gmail.com>
References: <b7737fe70910022344y72cdebc4ieb0b580ce7b7905f@mail.gmail.com>
	<709C5BD7-C916-4757-9AEE-C862D7F60053@freeswitch.org>
	<86a32abc0910031007ted4b6bfk1a567574af91ecae@mail.gmail.com>
	<65d96fc80910031638q52c3a595w8cce2209c6cf61f5@mail.gmail.com>
Message-ID: <7d0bfd8c0910040028y79ad956cxc289eec410beb8c1@mail.gmail.com>

agree that WAV/PCMA/PCMU formats are best for performance. you can use
mp3/ogg ONLY to archive recorded files.
/nandy

On Sun, Oct 4, 2009 at 7:38 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> also, you can store files in PCMA/PCMU format and avoid transcoding at
> all... and as said disk space is cheap.. go get some...
>
>
> On Sat, Oct 3, 2009 at 7:07 PM, Diego Viola <diego.viola at gmail.com> wrote:
>
>> Why is not recommended?
>>
>>
>> On Sat, Oct 3, 2009 at 2:52 PM, Brian West <brian at freeswitch.org> wrote:
>>
>>> MP3 is NOT recommend and if WAV files are too large you can mosey on
>>> down to the local Best Buy and snag 1.5TB of disk for like $119
>>> dollars.  Disk is cheap.
>>>
>>> /b
>>>
>>> On Oct 3, 2009, at 1:44 AM, Keith Wood wrote:
>>>
>>> >
>>> > I am working on an implementation for managing thousands of IVR
>>> > within an organization.  Right now, I am storing all audio files in
>>> > wav format, but it quickly become unmanagable because the size of
>>> > these wav files ( 8 bits mono ) quickly consuming a lot of the disk
>>> > space.
>>> >
>>> > Is there anyway I can store those audio files and still have high
>>> > quality audio for IVR?  I know mp3 is smaller but freeswitch does
>>> > not support it.
>>> >
>>> > any ideas?
>>> >
>>> > keith
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> > users
>>> > http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From raffaele.p.guidi at gmail.com  Sun Oct  4 03:13:05 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Sun, 4 Oct 2009 12:13:05 +0200
Subject: [Freeswitch-users]  Freeswitch as a softphone - presence?
Message-ID: <b0a8792f0910040313s3db46076sa527dd04110bc9f2@mail.gmail.com>

Hi, I was wondering how FreeSWITCH could notify presence through sofia
gateways - the basic idea is to use it as a softphone, of course.
Thanks,
   Raffaele
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From mcampbellsmith at gmail.com  Sun Oct  4 06:03:05 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sun, 4 Oct 2009 23:03:05 +1000
Subject: [Freeswitch-users] Detecting a fax
Message-ID: <33c87fa30910040603p7929df26l88f51fc7d256408f@mail.gmail.com>

Hi

I was hoping someone could help me to setup the fax detection / tone
detection application.

I want to be able to transfer an incoming fax to a specific extension.
 In my default.xml file, I have the following (extracted):

    <extension name="1000">
      <condition field="destination_number" expression="^(10[01][0-9])$">
        <action application="fax_detect"/>
        <action application="tone_detect" data="fax 1100 r +5000
transfer fax XML features" />

I can't get the fax to be detected and transferred.  Is there any way
this can be done?

Thanks!



From mike at jerris.com  Sun Oct  4 14:23:02 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 4 Oct 2009 17:23:02 -0400
Subject: [Freeswitch-users] Outbound INVITE rejected with 480 Temp
	Unavail, Reason "MANDATORY_IE_MISSING"
In-Reply-To: <6789F39C3E7544B7BDBCD0CA706B060E@greyhawk.tonecommander.com>
References: <6789F39C3E7544B7BDBCD0CA706B060E@greyhawk.tonecommander.com>
Message-ID: <0E5C8DD9-FED6-4367-AC53-62FCCE3FFFC1@jerris.com>

there is a profile param to enable 3pcc.  It should be documented in  
the default configs.

Mike

On Sep 29, 2009, at 5:22 PM, Jerry Richards wrote:

> Hello All,
>
> I have an internal extension that needs to send an INVITE without  
> SDP body
> (Content Length 0).  Freeswitch is replying with 480 Temporarily  
> Unavailable
> with reason "MANDATORY_IE_MISSING".  Would anyone know what I need  
> to do to
> enable this?




From mike at jerris.com  Sun Oct  4 14:34:01 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 4 Oct 2009 17:34:01 -0400
Subject: [Freeswitch-users] Problem with subscription expire
In-Reply-To: <4AC46DEB.3090506@ewetel.de>
References: <4AC46DEB.3090506@ewetel.de>
Message-ID: <B9D110F1-7602-4107-9151-B622F7F04A46@jerris.com>

This sounds like a bug in the snom to me, we keep changing the expire  
on to the future so it should never expire in the first place.  You  
will have to look at a longer running sip trace to see what exactly is  
going on.

Mike

On Oct 1, 2009, at 4:52 AM, Helmut Kuper wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> it seems exired subsciptions are never cleared in FS.
>
> A look into sofia_presence.c confirms explains this
>
> /* negative in exptime means keep bumping up sub time to avoid a snafu
> where every device has it's own rules about subscriptions
>                           that somehow barely resemble the RFC not  
> that
> I blame them because the RFC MAY be amibiguous and SHOULD be deleted.
>                           So to avoid the problem we keep resetting  
> the
> expiration date of the subscription so it never expires.
>
>                           Eybeam completely ignores this option and
> most other subscription-state: directives from rfc3265 and still  
> expires.
>                           Polycom is happy to keep upping the
> subscription expiry back to the original time on each new notify.
>                           The rest ... who knows...?
>
>                        */
>
> For some reasons subscriptions created by Snom phones are filling up  
> the
> sip_subscriptions table over time. This leads to some kind of DOS by  
> FS
> against the subscribing phone ... The subscribtions are  
> differentiate by
> call-id. This can be explained by RFC 3842 chapter 3.6 where expired
> subscriptions must be renewed with a NEW call-id. Because there is no
> hint about unsubscribing the old subscription I guess the clean up
> process has to be done by FS.
>
> Any way to get FS to do this job? Since there is no creation date or
> expire value which represents the expire as a timestamp I have no  
> way to
> clean up the table manually via sql and cronjob - except cleaning the
> whole table ...
>
>
> A further (but background) question is, why do the subscriptions  
> expire
> in snom phones at all ...




From mike at jerris.com  Sun Oct  4 14:40:30 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 4 Oct 2009 17:40:30 -0400
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <1b46b4e80910011229k31f7233du10b0fc661730d3f3@mail.gmail.com>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<191c3a030909291122t770af037j5b1068197b31de87@mail.gmail.com>
	<1b46b4e80909291542k71440965g75bb3bfcf38e8005@mail.gmail.com>
	<1b46b4e80910011229k31f7233du10b0fc661730d3f3@mail.gmail.com>
Message-ID: <F86F950D-9106-49C9-81CA-38B3EB2072F4@jerris.com>

can you send a link of a text sip trace please.

On Oct 1, 2009, at 3:29 PM, Nicolas Brenner wrote:

> Any ideas about this?
>
> The SIP provider is offering H323, but I'm not quite sure about  
> that, is mod_opal working right?
>
> Thanks!
>
> Nicolas
>
> On Tue, Sep 29, 2009 at 6:42 PM, Nicolas Brenner <nicolas at medularis.com 
> > wrote:
> Anthony, thanks. Below are my config files for the two gateways from  
> the sip trace. Both files are located in conf/directory/default.
>
> ---------------------
>
> redvoiss.xml (the one that works)
>
> <include>
>   <user id="gateway_redvoiss">
>     <gateways>
>       <gateway name="redvoiss-pp">
>         <param name="username" value="xxxxxxx"/>
>         <param name="password" value="xxxxxxx"/>
>         <param name="from-domain" value="pxextmy.redvoiss.net"/>
>         <param name="realm" value="pxextmy.redvoiss.net"/>
>         <param name="proxy" value="pxextmy.redvoiss.net"/>
>         <param name="from-user" value="xxxxxxx"/>
>         <param name="caller-id-in-from" value="false"/>
>         <param name="expire-seconds" value="600"/>
>         <param name="register" value="true"/>
>         <param name="retry_seconds" value="5"/>
>         <param name="extension" value="2010"/>
>         <param name="context" value="public"/>
>         <param name="codec-prefs" value="G729"/>
>         <param name="rfc2833-pt" value="101"/>
>       </gateway>
>     </gateways>
>     <params>
>       <param name="password" value="4321"/>
>     </params>
>   </user>
> </include>
>
> ---------------------
>
> orange.xml (the one that doesn't work)
>
> <include>
>   <user id="gateway_orange">
>     <gateways>
>       <gateway name="orange">
>         <param name="username" value="xxxxxxx"/>
>         <param name="password" value="xxxxxxx"/>
>         <param name="from-domain" value="216.72.10.39"/>
>         <param name="realm" value="216.72.10.39"/>
>         <param name="proxy" value="216.72.10.39"/>
>         <param name="from-user" value="xxxxxxx"/>
>         <param name="caller-id-in-from" value="false"/>
>         <param name="expire-seconds" value="600"/>
>         <param name="register" value="true"/>
>         <param name="retry_seconds" value="5"/>
>         <param name="extension" value="2011"/>
>         <param name="context" value="public"/>
>         <param name="codec-prefs" value="G729"/>
>         <param name="rfc2833-pt" value="101"/>
>       </gateway>
>     </gateways>
>     <params>
>       <param name="password" value="4321"/>
>     </params>
>   </user>
> </include>
>
> ---------------------
>
> If I remove the register=true param for the non-working gateway, I  
> don't get the registration error on the cli, but then all call  
> attempts get rejected with a 401 Unauthorized, and I get a hangup  
> cause of NORMAL_UNSPECIFIED.
>
>
> Best,
>
> Nicolas
>
>
>
> On Tue, Sep 29, 2009 at 2:22 PM, Anthony Minessale <anthony.minessale at gmail.com 
> > wrote:
> 900 level errors are sofia internal errors so probably something is  
> wrong with your gateway config xml.
> if you want to send it with any critical info replaced with XXX  
> maybe we can see the issue for you.
>
>
>
> On Tue, Sep 29, 2009 at 1:05 PM, Nicolas Brenner <nicolas at medularis.com 
> > wrote:
> Hello everyone,
>
> I am trying to add a gateway, but after configuring it just like the  
> others gateways I have, it is failing to register with a message  
> like this:
>
> 2009-09-29 12:54:40.853440 [ERR] sofia_reg.c:1402 orange  
> Registration Failed with status Operation has no matching challenge   
> [904]. failure #1
> 2009-09-29 12:54:40.906798 [WARNING] sofia_reg.c:364 orange Failed  
> Registration, setting retry to 10 seconds.
>
> I captured the sip traffic and noticed that when trying to register  
> with one gateway (the one that works), I get a "Trying" reply  
> immediately followed by a "401 Unauthorized" which contains a "WWW- 
> Authenticate: digest" with a "qop=auth" parameter. Then Freeswitch  
> replies with a second REGISTER including a large "Authorization:  
> digest" section with cnonce and nc=00000001 parameters.
>
> The gateway which doesn't register, doesn't send the "qop=auth"  
> parameter together with the "401 Unauthorized", and then Freeswitch  
> sends a "Authorization: digest" section on the second REGISTER with  
> no cnonce or nc parameters.
>
> I know very little abouth SIP, so I'm wondering what this "qop=auth"  
> parameter means and how does it affect the registration process. Is  
> there any way to do without the qop=auth parameter?
>
> Also, I tried registering with X-Lite directly to the gateway, and  
> it worked, so it appears to be a problem in the Freeswitch/gateway  
> combination. (Note: X-Lite sends an "Authorization: digest" section  
> on the _first_ REGISTER, apparently this makes a difference)
>
> Attached is a sip trace for the registration traffic when doing  
> "sofia profile external restart reloadxml" on the cli, captured with  
> "tshark -i eth0 -o "rtp.heuristic_rtp: TRUE" -w /tmp/capture.pcap -b  
> filesize:51200 -b files:100 -R 'sip or rtp or icmp or dns or rtcp or  
> t38'"
>
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From mike at jerris.com  Sun Oct  4 14:41:20 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 4 Oct 2009 17:41:20 -0400
Subject: [Freeswitch-users] register problem
In-Reply-To: <20091001212602.GC17256@base.carmickle.com>
References: <20091001212602.GC17256@base.carmickle.com>
Message-ID: <37090A74-55D8-4551-94A9-F540B60AD726@jerris.com>

Your problem is that the url below returns a Not found.


On Oct 1, 2009, at 5:26 PM, Frank Carmickle wrote:

> Can someone point out what is wrong here.  Thanks.
>
> Siptrace at http://carmickle.com/fs.txt
>
> <include>
>      <gateway name="voiptalk.org">
>        <param name="username" value="81100068"/>
>        <param name="password" value="my_pass"/>
>        <param name="from-user" value="81100068"/>
>        <param name="from-domain" value="voiptalk.org"/>
>    <param name="proxy" value="voiptalk.org"/>
>        <param name="expire-seconds" value="600"/>
>        <param name="register" value="true"/>
>        <param name="retry-seconds" value="30"/>
>        <param name="extension" value="1000"/>
>        <param name="contact-params" value="domain_name=$${domain}"/>
>        <param name="context" value="public"/>
>      </gateway>
> </include>




From mike at jerris.com  Sun Oct  4 14:45:53 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 4 Oct 2009 17:45:53 -0400
Subject: [Freeswitch-users] internal & external ip addresses of
	freeswitch
In-Reply-To: <241d382f0910020110l5f3728e7k3e521e81353b5d6f@mail.gmail.com>
References: <241d382f0910020110l5f3728e7k3e521e81353b5d6f@mail.gmail.com>
Message-ID: <C7EA8BED-2517-4E1C-94F6-148FD9C790B4@jerris.com>

You will need to setup 2 sip profiles for this setup, one for each  
interface.

Mike

On Oct 2, 2009, at 4:10 AM, Timur Irmatov wrote:

> Hi.
>
> We have a local network 192.168.1.0/24, where all the users are. Out
> FreeSWITCH server is connected to this network, and has ip address
> 192.168.1.242. Through different network card it is connected to
> external gateway, and has address 172.16.12.11 in this network.
>
> I set up a test client with softphone. When incoming call is
> deliviered to this client, call is set up normally, but client can't
> hang it up. It sends BYE to external address - 172.16.12.11 - which is
> not reachable from the client. It seems this address is coming from
> Contact: field in INVITE that FreeSWITCH sends:
>
> U 192.168.1.242:5060 -> 192.168.1.34:37169
> INVITE sip:100 at 192.168.1.34:37169 SIP/2.0.
> Via: SIP/2.0/UDP 172.16.12.11;rport;branch=z9hG4bKrvp6jm3myyaaF.
> Max-Forwards: 70.
> From: "FreeSWITCH" <sip:0000000000 at 172.16.12.11>;tag=v817pS9c6v6Fe.
> To: <sip:100 at 192.168.1.34:37169>.
> Call-ID: 797bd088-29cd-122d-9b93-0060979d54c5.
> CSeq: 121117089 INVITE.
> Contact: <sip:mod_sofia at 172.16.12.11:5060>.
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14898.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 267.
> Remote-Party-ID: "FreeSWITCH"
> <sip:0000000000 at 172.16.12.11>;party=calling;screen=yes;privacy=off.
>
> What should I tweak in freeswitch to change this behaviour?




From frank at carmickle.com  Sun Oct  4 15:00:56 2009
From: frank at carmickle.com (Frank Carmickle)
Date: Sun, 4 Oct 2009 18:00:56 -0400
Subject: [Freeswitch-users] register problem
In-Reply-To: <37090A74-55D8-4551-94A9-F540B60AD726@jerris.com>
References: <20091001212602.GC17256@base.carmickle.com>
	<37090A74-55D8-4551-94A9-F540B60AD726@jerris.com>
Message-ID: <20091004220055.GK17256@base.carmickle.com>

On Sun, Oct 04, Michael Jerris wrote:
> Your problem is that the url below returns a Not found.

I sent another message yesterday with a different link with more output.

http://carmickle.com/fs-2009-10-02.txt

Thanks
--FRank



From nicolas at medularis.com  Sun Oct  4 15:19:37 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Sun, 4 Oct 2009 18:19:37 -0400
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <F86F950D-9106-49C9-81CA-38B3EB2072F4@jerris.com>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<191c3a030909291122t770af037j5b1068197b31de87@mail.gmail.com>
	<1b46b4e80909291542k71440965g75bb3bfcf38e8005@mail.gmail.com>
	<1b46b4e80910011229k31f7233du10b0fc661730d3f3@mail.gmail.com>
	<F86F950D-9106-49C9-81CA-38B3EB2072F4@jerris.com>
Message-ID: <1b46b4e80910041519h3581d3d2x7b84041e8c18588@mail.gmail.com>

Here it is:

- http://pastebin.freeswitch.org/10582

(it is the pcap file I sent on the first email of this thread, converted to
text with 'tshark -V -r')

On Sun, Oct 4, 2009 at 5:40 PM, Michael Jerris <mike at jerris.com> wrote:

> can you send a link of a text sip trace please.
>
> On Oct 1, 2009, at 3:29 PM, Nicolas Brenner wrote:
>
> Any ideas about this?
>
> The SIP provider is offering H323, but I'm not quite sure about that, is
> mod_opal working right?
>
> Thanks!
>
> Nicolas
>
> On Tue, Sep 29, 2009 at 6:42 PM, Nicolas Brenner <nicolas at medularis.com>wrote:
>
>> Anthony, thanks. Below are my config files for the two gateways from the
>> sip trace. Both files are located in conf/directory/default.
>>
>> ---------------------
>>
>> redvoiss.xml (the one that works)
>>
>> <include>
>>   <user id="gateway_redvoiss">
>>     <gateways>
>>       <gateway name="redvoiss-pp">
>>         <param name="username" value="xxxxxxx"/>
>>         <param name="password" value="xxxxxxx"/>
>>         <param name="from-domain" value="pxextmy.redvoiss.net"/>
>>         <param name="realm" value="pxextmy.redvoiss.net"/>
>>         <param name="proxy" value="pxextmy.redvoiss.net"/>
>>         <param name="from-user" value="xxxxxxx"/>
>>         <param name="caller-id-in-from" value="false"/>
>>         <param name="expire-seconds" value="600"/>
>>         <param name="register" value="true"/>
>>         <param name="retry_seconds" value="5"/>
>>         <param name="extension" value="2010"/>
>>         <param name="context" value="public"/>
>>         <param name="codec-prefs" value="G729"/>
>>         <param name="rfc2833-pt" value="101"/>
>>       </gateway>
>>     </gateways>
>>     <params>
>>       <param name="password" value="4321"/>
>>     </params>
>>   </user>
>> </include>
>>
>> ---------------------
>>
>> orange.xml (the one that doesn't work)
>>
>> <include>
>>   <user id="gateway_orange">
>>     <gateways>
>>       <gateway name="orange">
>>         <param name="username" value="xxxxxxx"/>
>>         <param name="password" value="xxxxxxx"/>
>>         <param name="from-domain" value="216.72.10.39"/>
>>         <param name="realm" value="216.72.10.39"/>
>>         <param name="proxy" value="216.72.10.39"/>
>>         <param name="from-user" value="xxxxxxx"/>
>>         <param name="caller-id-in-from" value="false"/>
>>         <param name="expire-seconds" value="600"/>
>>         <param name="register" value="true"/>
>>         <param name="retry_seconds" value="5"/>
>>         <param name="extension" value="2011"/>
>>         <param name="context" value="public"/>
>>         <param name="codec-prefs" value="G729"/>
>>         <param name="rfc2833-pt" value="101"/>
>>       </gateway>
>>     </gateways>
>>     <params>
>>       <param name="password" value="4321"/>
>>     </params>
>>   </user>
>> </include>
>>
>> ---------------------
>>
>> If I remove the register=true param for the non-working gateway, I don't
>> get the registration error on the cli, but then all call attempts get
>> rejected with a 401 Unauthorized, and I get a hangup cause of
>> NORMAL_UNSPECIFIED.
>>
>>
>> Best,
>>
>> Nicolas
>>
>>
>>
>> On Tue, Sep 29, 2009 at 2:22 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> 900 level errors are sofia internal errors so probably something is wrong
>>> with your gateway config xml.
>>> if you want to send it with any critical info replaced with XXX maybe we
>>> can see the issue for you.
>>>
>>>
>>>
>>> On Tue, Sep 29, 2009 at 1:05 PM, Nicolas Brenner <nicolas at medularis.com>wrote:
>>>
>>>> Hello everyone,
>>>>
>>>> I am trying to add a gateway, but after configuring it just like the
>>>> others gateways I have, it is failing to register with a message like this:
>>>>
>>>> 2009-09-29 12:54:40.853440 [ERR] sofia_reg.c:1402 orange Registration
>>>> Failed with status Operation has no matching challenge  [904]. failure #1
>>>> 2009-09-29 12:54:40.906798 [WARNING] sofia_reg.c:364 orange Failed
>>>> Registration, setting retry to 10 seconds.
>>>>
>>>> I captured the sip traffic and noticed that when trying to register with
>>>> one gateway (the one that works), I get a "Trying" reply immediately
>>>> followed by a "401 Unauthorized" which contains a "WWW-Authenticate: digest"
>>>> with a "qop=auth" parameter. Then Freeswitch replies with a second REGISTER
>>>> including a large "Authorization: digest" section with cnonce and
>>>> nc=00000001 parameters.
>>>>
>>>> The gateway which doesn't register, doesn't send the "qop=auth"
>>>> parameter together with the "401 Unauthorized", and then Freeswitch sends a
>>>> "Authorization: digest" section on the second REGISTER with no cnonce or nc
>>>> parameters.
>>>>
>>>> I know very little abouth SIP, so I'm wondering what this "qop=auth"
>>>> parameter means and how does it affect the registration process. Is there
>>>> any way to do without the qop=auth parameter?
>>>>
>>>> Also, I tried registering with X-Lite directly to the gateway, and it
>>>> worked, so it appears to be a problem in the Freeswitch/gateway combination.
>>>> (Note: X-Lite sends an "Authorization: digest" section on the _first_
>>>> REGISTER, apparently this makes a difference)
>>>>
>>>> Attached is a sip trace for the registration traffic when doing "sofia
>>>> profile external restart reloadxml" on the cli, captured with "tshark -i
>>>> eth0 -o "rtp.heuristic_rtp: TRUE" -w /tmp/capture.pcap -b filesize:51200 -b
>>>> files:100 -R 'sip or rtp or icmp or dns or rtcp or t38'"
>>>>
>>>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Sun Oct  4 15:26:21 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 4 Oct 2009 18:26:21 -0400
Subject: [Freeswitch-users] register problem
In-Reply-To: <20091004220055.GK17256@base.carmickle.com>
References: <20091001212602.GC17256@base.carmickle.com>
	<37090A74-55D8-4551-94A9-F540B60AD726@jerris.com>
	<20091004220055.GK17256@base.carmickle.com>
Message-ID: <1EF6E903-1014-4042-8653-AA0B0C6C6401@jerris.com>

Is there any info of what I am looking at here, I just went through  
1000's  of lines that look like repeated good registers and a working  
call.. What exactly is not working?

Mike

On Oct 4, 2009, at 6:00 PM, Frank Carmickle wrote:

> On Sun, Oct 04, Michael Jerris wrote:
>> Your problem is that the url below returns a Not found.
>
> I sent another message yesterday with a different link with more  
> output.
>
> http://carmickle.com/fs-2009-10-02.txt
>
> Thanks
> --FRank
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From mike at jerris.com  Sun Oct  4 15:30:15 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 4 Oct 2009 18:30:15 -0400
Subject: [Freeswitch-users] How to limit the number of incoming+outgoing
	calls via specific gateway?
In-Reply-To: <65d96fc80910020532i4ba33353g7ac141624a4bb89e@mail.gmail.com>
References: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>
	<e403cddca1c90c83ce4fb6f85b026e63.squirrel@correo.nodo50.org>
	<65d96fc80910020532i4ba33353g7ac141624a4bb89e@mail.gmail.com>
Message-ID: <C1AD67AA-F969-4BA1-8415-5E381BDF18FA@jerris.com>

http://wiki.freeswitch.org/wiki/Mod_limit


On Oct 2, 2009, at 8:32 AM, Tihomir Culjaga wrote:

> what if you are running some huge traffic e.g. 2000 calls with media?
>
> a typical application for that is an IVR system handling several  
> different services. I'd like to "dedicate" some capacity for inbound  
> on per service basis.
>
>
> e.g.
>
> DID 10001 limit to 500 calls
> DID 10002 limit to 400 calls
> DID 10003 limit to 100 calls
> DID 10005 limit to 1000 calls
>
>
> This will be a total of 2000 calls.
>
>
> don't you think js is simply too weak for that? It should cont calls/ 
> channels, brake counts per service/DID and update the counters on  
> every call hit.
>
>
>
>
> in the DP you would have something like this for every DID:
>
>
> <include>
>   <extension name="MY_DID_NUM">
>     <condition field="destination_number" expression="^MY_DID_NUMBER 
> $">
>     <action application="set" data="SERVICE_LIMIT=500"/>
>
>       <!--
>              count number of active channels going towards  
> MY_DID_NUMBER and store it into COUNT_MY_DID_NUMBER
>       -->
>
>       <action application="transfer" data="do_MY_SERVICE XML public"/>
>     </condition>
>   </extension>
> </include>
>
>
>
> <include>
>   <extension name="SERVICE1">
>     <condition field="destination_number" expression="^do_MY_SERVICE 
> $"/>
>     <condition field="${COUNT_MY_DID_NUMBER}"  
> expression="^SERVICE_LIMIT$">
>
>        <!-- do your service here -->
>       <action application="playback" data="I_Accept_Your_Call.wav"/>
>       <action application="hangup" data="NORMAL_CLEARING"/>
>
>        <!-- do your limitation here -->
>       <anti-action application="respond" data="403 Forbidden"/> <=  
> put your response here!
>
>     </condition>
>   </extension>
> </include>
>
>
>
>
>
> but the question is ... how powerful a JavaScript can be? Will it be  
> enough to handle that load?
>
>
>
> Tihomir.
>
>
>
>
>
> On Fri, Oct 2, 2009 at 1:11 PM, Alberto Escudero <aep.lists at it46.se>  
> wrote:
>
> You can use the api and check that the channel is occupied with "show
> channels"?
> You can write a small javascript that checks if the channel is  
> occupied by
> means of session.execute api.
>
> /aep
> --
> Stopping junk mailers is good for the environment
>
> > My SIP provider allows only one call (incoming or outgoing) via one
> > SIP account. For FreeSWITCH I have configured it as public DID
> > extension and outgoing gateway. Now I would like to transfer to
> > another gw (or generate "limit exceded") when one tries to place an
> > outgoing call while incoming call is in progress. How tho do that?
> > Limiting the number of outgoing calls is easy (mod_limit), but how  
> to
> > take into account incoming one?
> >
> > - Dmitry Bely
> >
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From mike at jerris.com  Sun Oct  4 15:31:28 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 4 Oct 2009 18:31:28 -0400
Subject: [Freeswitch-users] configure FS: config.status: error: cannot
	find input file: Makefile.in
In-Reply-To: <65d96fc80910020505t48a6237ao19884c23fedf228d@mail.gmail.com>
References: <65d96fc80910020505t48a6237ao19884c23fedf228d@mail.gmail.com>
Message-ID: <4690D578-FBC2-4230-BDF0-C43CDF6669A0@jerris.com>

I updated the tiff lib to build better inline, try make  tiff-reconf

Mike

On Oct 2, 2009, at 8:05 AM, Tihomir Culjaga wrote:

> hello,
> i just got the last trunk and tried to compile it on one of my  
> development machines... Well configure fails on tiff-3.8.2 where it  
> is unable to find Makefile.in ... Can someone advice?
>
>
>
> checking if g++ static flag -static works... yes
> checking if g++ supports -c -o file.o... yes
> checking if g++ supports -c -o file.o... (cached) yes
> checking whether the g++ linker (/usr/bin/ld) supports shared  
> libraries... yes
> checking dynamic linker characteristics... GNU/Linux ld.so
> checking how to hardcode library paths into programs... immediate
> checking for OpenGL Utility library... no
> checking for GLUT library... no
> configure: creating ./config.status
> config.status: error: cannot find input file: Makefile.in
>
>
>
> tculjaga at subZero:~/freeswitch-trunk/libs/tiff-3.8.2$ ls -l
> total 2224
> -rw-r--r--  1 tculjaga tculjaga  23741 2009-10-02 13:19 acinclude.m4
> -rw-r--r--  1 tculjaga tculjaga 316978 2009-10-02 13:28 aclocal.m4
> -rwxr-xr-x  1 tculjaga tculjaga    121 2009-10-02 13:19 autogen.sh
> -rw-r--r--  1 tculjaga tculjaga 124047 2009-10-02 13:19 ChangeLog
> drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:28 config
> -rw-r--r--  1 tculjaga tculjaga 103926 2009-10-02 14:00 config.log
> -rwxr-xr-x  1 tculjaga tculjaga  73065 2009-10-02 14:00 config.status
> -rwxr-xr-x  1 tculjaga tculjaga 740145 2009-10-02 13:28 configure
> -rw-r--r--  1 tculjaga tculjaga  20492 2009-10-02 13:19 configure.ac
> -rwxr-xr-x  1 tculjaga tculjaga     56 2009-10-02 13:19 configure.gnu
> -rwxr-xr-x  1 tculjaga tculjaga 737794 2009-10-02 13:57  
> configure.lineno
> drwxr-xr-x 16 tculjaga tculjaga   4096 2009-10-02 13:19 contrib
> -rw-r--r--  1 tculjaga tculjaga   1146 2009-10-02 13:19 COPYRIGHT
> -rw-r--r--  1 tculjaga tculjaga   1570 2009-10-02 13:19 HOWTO-RELEASE
> drwxr-xr-x  5 tculjaga tculjaga   4096 2009-10-02 13:19 html
> drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:28 libtiff
> drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 m4
> -rw-r--r--  1 tculjaga tculjaga   1908 2009-10-02 13:19 Makefile.am
> -rw-r--r--  1 tculjaga tculjaga   1724 2009-10-02 13:19 Makefile.vc
> drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 man
> -rw-r--r--  1 tculjaga tculjaga   6270 2009-10-02 13:19 nmake.opt
> drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 port
> -rw-r--r--  1 tculjaga tculjaga   2363 2009-10-02 13:19 README
> -rw-r--r--  1 tculjaga tculjaga      9 2009-10-02 13:19 RELEASE-DATE
> -rw-r--r--  1 tculjaga tculjaga   5893 2009-10-02 13:19 SConstruct
> drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 test
> -rw-r--r--  1 tculjaga tculjaga    433 2009-10-02 13:19 TODO
> drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 tools
> -rw-r--r--  1 tculjaga tculjaga      6 2009-10-02 13:19 VERSION
> tculjaga at subZero:~/freeswitch-trunk/libs/tiff-3.8.2$
> tculjaga at subZero:~/freeswitch-trunk/libs/tiff-3.8.2$
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From mike at jerris.com  Sun Oct  4 15:35:58 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 4 Oct 2009 18:35:58 -0400
Subject: [Freeswitch-users] New to freeswitch and have a few questions
In-Reply-To: <a87d43fd0910020910l5db88c5fv980c8c8fe312f380@mail.gmail.com>
References: <4AC5534C.6050202@tx.rr.com>
	<a87d43fd0910020910l5db88c5fv980c8c8fe312f380@mail.gmail.com>
Message-ID: <8F97B3DC-C0B5-403C-9117-A6B4E64A0492@jerris.com>

Getting documentation on like this on the wiki would be awesome.

Mike

On Oct 2, 2009, at 12:10 PM, Michael Gende wrote:

> Hey Orien,
>
> I'm not using exactly your set up, but am using pfsense/FreeBSD.  
> Since you're using that, I assume you're going "dual homed". I've  
> got a starter guide that might help you out. If nothing else, I'd be  
> interested in a candid assessment of its usefulness or lack thereof,  
> especially to a guy like you.
>
> I've included it here. Its all just text at the moment so be  
> advised. Also be advised that there's a lot of great information on  
> the freeswitch site and in this group. The goal of my document was  
> so that someone just starting would have to hunt a little less.
>
> Hope its good for something, let me know either way, especially if  
> you find errors.
>
> Regards,
>
> Mike G.
>
> On Thu, Oct 1, 2009 at 8:11 PM, Orien Love <orien at tx.rr.com> wrote:
> Hello Everybody,
>    I am new to freeswitch, so forgive me if I ask stupid questions. I
> am planning a test setup consisting of:
>    1 - Pfsense router with the freeswitch package installed.
>    1 - Cisco WS-C3524-PWR-XL-EN 24 FE Switch, POE for the phones.
>    1 - LINKSYS SPA3000 to connect to my existing land line and phones.
>    2 - POLYCOM SOUNDPOINT IP601 SIP IP PHONEs
>
> The first question I have, Are the IP601 phones supported? The wiki
> lists 320, 431, 501, 550, 650 but not the 601.
>
> Second, is there a place that helps a person new to the IP phone world
> learn what is needed to set up a PBX using freeswitch at a small  
> office?
>
> Finally is my test setup a good one? is there something I am missing  
> or
> that I need to get the learning process started, I have found in the
> past, with a little information and a test system, I can learn what  
> I am
> doing by breaking and fixing the test bed.
>
> Thanks for your time
> Orien
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
> <freeswitch.txt>_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From frank at carmickle.com  Sun Oct  4 15:40:44 2009
From: frank at carmickle.com (Frank Carmickle)
Date: Sun, 4 Oct 2009 18:40:44 -0400
Subject: [Freeswitch-users] register problem
In-Reply-To: <1EF6E903-1014-4042-8653-AA0B0C6C6401@jerris.com>
References: <20091001212602.GC17256@base.carmickle.com>
	<37090A74-55D8-4551-94A9-F540B60AD726@jerris.com>
	<20091004220055.GK17256@base.carmickle.com>
	<1EF6E903-1014-4042-8653-AA0B0C6C6401@jerris.com>
Message-ID: <20091004224044.GL17256@base.carmickle.com>

On Sun, Oct 04, Michael Jerris wrote:
> Is there any info of what I am looking at here, I just went through  
> 1000's  of lines that look like repeated good registers and a working  
> call.. What exactly is not working?

The register fails and then never registers again.  See line 31268.  The failure is

2009-10-03 01:56:01.942233 [ERR] sofia_reg.c:1419 voiptalk.org Registration Failed with status Operation has no matching challenge  [904]. failure #1

Thanks for looking at this.

--FC


> 
> Mike
> 
> On Oct 4, 2009, at 6:00 PM, Frank Carmickle wrote:
> 
> > On Sun, Oct 04, Michael Jerris wrote:
> >> Your problem is that the url below returns a Not found.
> >
> > I sent another message yesterday with a different link with more  
> > output.
> >
> > http://carmickle.com/fs-2009-10-02.txt
> >
> > Thanks
> > --FRank
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> > users
> > http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mike at jerris.com  Sun Oct  4 15:48:17 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 4 Oct 2009 18:48:17 -0400
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <1b46b4e80910041519h3581d3d2x7b84041e8c18588@mail.gmail.com>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<191c3a030909291122t770af037j5b1068197b31de87@mail.gmail.com>
	<1b46b4e80909291542k71440965g75bb3bfcf38e8005@mail.gmail.com>
	<1b46b4e80910011229k31f7233du10b0fc661730d3f3@mail.gmail.com>
	<F86F950D-9106-49C9-81CA-38B3EB2072F4@jerris.com>
	<1b46b4e80910041519h3581d3d2x7b84041e8c18588@mail.gmail.com>
Message-ID: <C887D143-F7A1-44B9-B8A2-D34A4B4E582E@jerris.com>

I've never been able to read these, why exactly do I need a text  
protocol to be decoded for me?  Ends up being too much noise so I just  
don't bother.

Mike

On Oct 4, 2009, at 6:19 PM, Nicolas Brenner wrote:

> Here it is:
>
> - http://pastebin.freeswitch.org/10582
>
> (it is the pcap file I sent on the first email of this thread,  
> converted to text with 'tshark -V -r')
>
> On Sun, Oct 4, 2009 at 5:40 PM, Michael Jerris <mike at jerris.com>  
> wrote:
> can you send a link of a text sip trace please.
>
> On Oct 1, 2009, at 3:29 PM, Nicolas Brenner wrote:
>
>> Any ideas about this?
>>
>> The SIP provider is offering H323, but I'm not quite sure about  
>> that, is mod_opal working right?
>>
>> Thanks!
>>
>> Nicolas
>>
>> On Tue, Sep 29, 2009 at 6:42 PM, Nicolas Brenner <nicolas at medularis.com 
>> > wrote:
>> Anthony, thanks. Below are my config files for the two gateways  
>> from the sip trace. Both files are located in conf/directory/default.
>>
>> ---------------------
>>
>> redvoiss.xml (the one that works)
>>
>> <include>
>>   <user id="gateway_redvoiss">
>>     <gateways>
>>       <gateway name="redvoiss-pp">
>>         <param name="username" value="xxxxxxx"/>
>>         <param name="password" value="xxxxxxx"/>
>>         <param name="from-domain" value="pxextmy.redvoiss.net"/>
>>         <param name="realm" value="pxextmy.redvoiss.net"/>
>>         <param name="proxy" value="pxextmy.redvoiss.net"/>
>>         <param name="from-user" value="xxxxxxx"/>
>>         <param name="caller-id-in-from" value="false"/>
>>         <param name="expire-seconds" value="600"/>
>>         <param name="register" value="true"/>
>>         <param name="retry_seconds" value="5"/>
>>         <param name="extension" value="2010"/>
>>         <param name="context" value="public"/>
>>         <param name="codec-prefs" value="G729"/>
>>         <param name="rfc2833-pt" value="101"/>
>>       </gateway>
>>     </gateways>
>>     <params>
>>       <param name="password" value="4321"/>
>>     </params>
>>   </user>
>> </include>
>>
>> ---------------------
>>
>> orange.xml (the one that doesn't work)
>>
>> <include>
>>   <user id="gateway_orange">
>>     <gateways>
>>       <gateway name="orange">
>>         <param name="username" value="xxxxxxx"/>
>>         <param name="password" value="xxxxxxx"/>
>>         <param name="from-domain" value="216.72.10.39"/>
>>         <param name="realm" value="216.72.10.39"/>
>>         <param name="proxy" value="216.72.10.39"/>
>>         <param name="from-user" value="xxxxxxx"/>
>>         <param name="caller-id-in-from" value="false"/>
>>         <param name="expire-seconds" value="600"/>
>>         <param name="register" value="true"/>
>>         <param name="retry_seconds" value="5"/>
>>         <param name="extension" value="2011"/>
>>         <param name="context" value="public"/>
>>         <param name="codec-prefs" value="G729"/>
>>         <param name="rfc2833-pt" value="101"/>
>>       </gateway>
>>     </gateways>
>>     <params>
>>       <param name="password" value="4321"/>
>>     </params>
>>   </user>
>> </include>
>>
>> ---------------------
>>
>> If I remove the register=true param for the non-working gateway, I  
>> don't get the registration error on the cli, but then all call  
>> attempts get rejected with a 401 Unauthorized, and I get a hangup  
>> cause of NORMAL_UNSPECIFIED.
>>
>>
>> Best,
>>
>> Nicolas
>>
>>
>>
>> On Tue, Sep 29, 2009 at 2:22 PM, Anthony Minessale <anthony.minessale at gmail.com 
>> > wrote:
>> 900 level errors are sofia internal errors so probably something is  
>> wrong with your gateway config xml.
>> if you want to send it with any critical info replaced with XXX  
>> maybe we can see the issue for you.
>>
>>
>>
>> On Tue, Sep 29, 2009 at 1:05 PM, Nicolas Brenner <nicolas at medularis.com 
>> > wrote:
>> Hello everyone,
>>
>> I am trying to add a gateway, but after configuring it just like  
>> the others gateways I have, it is failing to register with a  
>> message like this:
>>
>> 2009-09-29 12:54:40.853440 [ERR] sofia_reg.c:1402 orange  
>> Registration Failed with status Operation has no matching  
>> challenge  [904]. failure #1
>> 2009-09-29 12:54:40.906798 [WARNING] sofia_reg.c:364 orange Failed  
>> Registration, setting retry to 10 seconds.
>>
>> I captured the sip traffic and noticed that when trying to register  
>> with one gateway (the one that works), I get a "Trying" reply  
>> immediately followed by a "401 Unauthorized" which contains a "WWW- 
>> Authenticate: digest" with a "qop=auth" parameter. Then Freeswitch  
>> replies with a second REGISTER including a large "Authorization:  
>> digest" section with cnonce and nc=00000001 parameters.
>>
>> The gateway which doesn't register, doesn't send the "qop=auth"  
>> parameter together with the "401 Unauthorized", and then Freeswitch  
>> sends a "Authorization: digest" section on the second REGISTER with  
>> no cnonce or nc parameters.
>>
>> I know very little abouth SIP, so I'm wondering what this  
>> "qop=auth" parameter means and how does it affect the registration  
>> process. Is there any way to do without the qop=auth parameter?
>>
>> Also, I tried registering with X-Lite directly to the gateway, and  
>> it worked, so it appears to be a problem in the Freeswitch/gateway  
>> combination. (Note: X-Lite sends an "Authorization: digest" section  
>> on the _first_ REGISTER, apparently this makes a difference)
>>
>> Attached is a sip trace for the registration traffic when doing  
>> "sofia profile external restart reloadxml" on the cli, captured  
>> with "tshark -i eth0 -o "rtp.heuristic_rtp: TRUE" -w /tmp/ 
>> capture.pcap -b filesize:51200 -b files:100 -R 'sip or rtp or icmp  
>> or dns or rtcp or t38'"
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From mike at jerris.com  Sun Oct  4 15:56:54 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 4 Oct 2009 18:56:54 -0400
Subject: [Freeswitch-users] Detecting a fax
In-Reply-To: <33c87fa30910040603p7929df26l88f51fc7d256408f@mail.gmail.com>
References: <33c87fa30910040603p7929df26l88f51fc7d256408f@mail.gmail.com>
Message-ID: <91832961-FBA6-465F-AFD8-6A1BE604F6BC@jerris.com>

check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect

Note, you can't just have tone_detect as your last iten in the  
dialplan as the call will just get hung up.

Mike

On Oct 4, 2009, at 9:03 AM, Mark Campbell-Smith wrote:

> Hi
>
> I was hoping someone could help me to setup the fax detection / tone
> detection application.
>
> I want to be able to transfer an incoming fax to a specific extension.
> In my default.xml file, I have the following (extracted):
>
>    <extension name="1000">
>      <condition field="destination_number" expression="^(10[01][0-9]) 
> $">
>        <action application="fax_detect"/>
>        <action application="tone_detect" data="fax 1100 r +5000
> transfer fax XML features" />
>
> I can't get the fax to be detected and transferred.  Is there any way
> this can be done?
>
> Thanks!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From nicolas at medularis.com  Sun Oct  4 16:09:34 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Sun, 4 Oct 2009 19:09:34 -0400
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <C887D143-F7A1-44B9-B8A2-D34A4B4E582E@jerris.com>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<191c3a030909291122t770af037j5b1068197b31de87@mail.gmail.com>
	<1b46b4e80909291542k71440965g75bb3bfcf38e8005@mail.gmail.com>
	<1b46b4e80910011229k31f7233du10b0fc661730d3f3@mail.gmail.com>
	<F86F950D-9106-49C9-81CA-38B3EB2072F4@jerris.com>
	<1b46b4e80910041519h3581d3d2x7b84041e8c18588@mail.gmail.com>
	<C887D143-F7A1-44B9-B8A2-D34A4B4E582E@jerris.com>
Message-ID: <1b46b4e80910041609x2f41ceb9u142cb484951a21ed@mail.gmail.com>

Mike, how exactly should I format the file? I got the pcap file, how do I
convert it to text so that you can easily read it?


On Sun, Oct 4, 2009 at 6:48 PM, Michael Jerris <mike at jerris.com> wrote:

> I've never been able to read these, why exactly do I need a text protocol
> to be decoded for me?  Ends up being too much noise so I just don't bother.
> Mike
>
> On Oct 4, 2009, at 6:19 PM, Nicolas Brenner wrote:
>
> Here it is:
>
> - http://pastebin.freeswitch.org/10582
>
> (it is the pcap file I sent on the first email of this thread, converted to
> text with 'tshark -V -r')
>
> On Sun, Oct 4, 2009 at 5:40 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> can you send a link of a text sip trace please.
>>
>> On Oct 1, 2009, at 3:29 PM, Nicolas Brenner wrote:
>>
>> Any ideas about this?
>>
>> The SIP provider is offering H323, but I'm not quite sure about that, is
>> mod_opal working right?
>>
>> Thanks!
>>
>> Nicolas
>>
>> On Tue, Sep 29, 2009 at 6:42 PM, Nicolas Brenner <nicolas at medularis.com>wrote:
>>
>>> Anthony, thanks. Below are my config files for the two gateways from the
>>> sip trace. Both files are located in conf/directory/default.
>>>
>>> ---------------------
>>>
>>> redvoiss.xml (the one that works)
>>>
>>> <include>
>>>   <user id="gateway_redvoiss">
>>>     <gateways>
>>>       <gateway name="redvoiss-pp">
>>>         <param name="username" value="xxxxxxx"/>
>>>         <param name="password" value="xxxxxxx"/>
>>>         <param name="from-domain" value="pxextmy.redvoiss.net"/>
>>>         <param name="realm" value="pxextmy.redvoiss.net"/>
>>>         <param name="proxy" value="pxextmy.redvoiss.net"/>
>>>         <param name="from-user" value="xxxxxxx"/>
>>>         <param name="caller-id-in-from" value="false"/>
>>>         <param name="expire-seconds" value="600"/>
>>>         <param name="register" value="true"/>
>>>         <param name="retry_seconds" value="5"/>
>>>         <param name="extension" value="2010"/>
>>>         <param name="context" value="public"/>
>>>         <param name="codec-prefs" value="G729"/>
>>>         <param name="rfc2833-pt" value="101"/>
>>>       </gateway>
>>>     </gateways>
>>>     <params>
>>>       <param name="password" value="4321"/>
>>>     </params>
>>>   </user>
>>> </include>
>>>
>>> ---------------------
>>>
>>> orange.xml (the one that doesn't work)
>>>
>>> <include>
>>>   <user id="gateway_orange">
>>>     <gateways>
>>>       <gateway name="orange">
>>>         <param name="username" value="xxxxxxx"/>
>>>         <param name="password" value="xxxxxxx"/>
>>>         <param name="from-domain" value="216.72.10.39"/>
>>>         <param name="realm" value="216.72.10.39"/>
>>>         <param name="proxy" value="216.72.10.39"/>
>>>         <param name="from-user" value="xxxxxxx"/>
>>>         <param name="caller-id-in-from" value="false"/>
>>>         <param name="expire-seconds" value="600"/>
>>>         <param name="register" value="true"/>
>>>         <param name="retry_seconds" value="5"/>
>>>         <param name="extension" value="2011"/>
>>>         <param name="context" value="public"/>
>>>         <param name="codec-prefs" value="G729"/>
>>>         <param name="rfc2833-pt" value="101"/>
>>>       </gateway>
>>>     </gateways>
>>>     <params>
>>>       <param name="password" value="4321"/>
>>>     </params>
>>>   </user>
>>> </include>
>>>
>>> ---------------------
>>>
>>> If I remove the register=true param for the non-working gateway, I don't
>>> get the registration error on the cli, but then all call attempts get
>>> rejected with a 401 Unauthorized, and I get a hangup cause of
>>> NORMAL_UNSPECIFIED.
>>>
>>>
>>> Best,
>>>
>>> Nicolas
>>>
>>>
>>>
>>> On Tue, Sep 29, 2009 at 2:22 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> 900 level errors are sofia internal errors so probably something is
>>>> wrong with your gateway config xml.
>>>> if you want to send it with any critical info replaced with XXX maybe we
>>>> can see the issue for you.
>>>>
>>>>
>>>>
>>>> On Tue, Sep 29, 2009 at 1:05 PM, Nicolas Brenner <nicolas at medularis.com
>>>> > wrote:
>>>>
>>>>> Hello everyone,
>>>>>
>>>>> I am trying to add a gateway, but after configuring it just like the
>>>>> others gateways I have, it is failing to register with a message like this:
>>>>>
>>>>> 2009-09-29 12:54:40.853440 [ERR] sofia_reg.c:1402 orange Registration
>>>>> Failed with status Operation has no matching challenge  [904]. failure #1
>>>>> 2009-09-29 12:54:40.906798 [WARNING] sofia_reg.c:364 orange Failed
>>>>> Registration, setting retry to 10 seconds.
>>>>>
>>>>> I captured the sip traffic and noticed that when trying to register
>>>>> with one gateway (the one that works), I get a "Trying" reply immediately
>>>>> followed by a "401 Unauthorized" which contains a "WWW-Authenticate: digest"
>>>>> with a "qop=auth" parameter. Then Freeswitch replies with a second REGISTER
>>>>> including a large "Authorization: digest" section with cnonce and
>>>>> nc=00000001 parameters.
>>>>>
>>>>> The gateway which doesn't register, doesn't send the "qop=auth"
>>>>> parameter together with the "401 Unauthorized", and then Freeswitch sends a
>>>>> "Authorization: digest" section on the second REGISTER with no cnonce or nc
>>>>> parameters.
>>>>>
>>>>> I know very little abouth SIP, so I'm wondering what this "qop=auth"
>>>>> parameter means and how does it affect the registration process. Is there
>>>>> any way to do without the qop=auth parameter?
>>>>>
>>>>> Also, I tried registering with X-Lite directly to the gateway, and it
>>>>> worked, so it appears to be a problem in the Freeswitch/gateway combination.
>>>>> (Note: X-Lite sends an "Authorization: digest" section on the _first_
>>>>> REGISTER, apparently this makes a difference)
>>>>>
>>>>> Attached is a sip trace for the registration traffic when doing "sofia
>>>>> profile external restart reloadxml" on the cli, captured with "tshark -i
>>>>> eth0 -o "rtp.heuristic_rtp: TRUE" -w /tmp/capture.pcap -b filesize:51200 -b
>>>>> files:100 -R 'sip or rtp or icmp or dns or rtcp or t38'"
>>>>>
>>>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Sun Oct  4 16:14:47 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 4 Oct 2009 19:14:47 -0400
Subject: [Freeswitch-users] register problem
In-Reply-To: <20091004224044.GL17256@base.carmickle.com>
References: <20091001212602.GC17256@base.carmickle.com>
	<37090A74-55D8-4551-94A9-F540B60AD726@jerris.com>
	<20091004220055.GK17256@base.carmickle.com>
	<1EF6E903-1014-4042-8653-AA0B0C6C6401@jerris.com>
	<20091004224044.GL17256@base.carmickle.com>
Message-ID: <CD304EEF-8F23-463E-ABBB-8AAE5847FA2D@jerris.com>

It appears to be in the case of the far end sending a 100 on register,  
and no 200 or any other terminal response after that.  I sent the  
realevant part of this trace off to the developer of the sip library  
for advice.  Please file a bug on jira.freeswitch.org on this, with a  
log of a coupe good registers before the failure and a few after the  
initial failure.

Mike

On Oct 4, 2009, at 6:40 PM, Frank Carmickle wrote:

> On Sun, Oct 04, Michael Jerris wrote:
>> Is there any info of what I am looking at here, I just went through
>> 1000's  of lines that look like repeated good registers and a working
>> call.. What exactly is not working?
>
> The register fails and then never registers again.  See line 31268.   
> The failure is
>
> 2009-10-03 01:56:01.942233 [ERR] sofia_reg.c:1419 voiptalk.org  
> Registration Failed with status Operation has no matching challenge   
> [904]. failure #1
>
> Thanks for looking at this.
>
> --FC
>
>
>>
>> Mike
>>
>> On Oct 4, 2009, at 6:00 PM, Frank Carmickle wrote:
>>
>>> On Sun, Oct 04, Michael Jerris wrote:
>>>> Your problem is that the url below returns a Not found.
>>>
>>> I sent another message yesterday with a different link with more
>>> output.
>>>
>>> http://carmickle.com/fs-2009-10-02.txt
>>>
>>> Thanks
>>> --FRank
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From kjv at ken-ton.com  Sun Oct  4 16:16:07 2009
From: kjv at ken-ton.com (Karl Vesterling)
Date: Sun, 4 Oct 2009 19:16:07 -0400
Subject: [Freeswitch-users] Youtube - FreeSWITCH Promo Video
Message-ID: <9A87E77C-AD85-4F1E-8ED0-F9F06AA451EE@ken-ton.com>

Folks;

Here's something that I did playing around w/ learning Apple Motion.
It's my first Apple Motion production, so don't be too hard on the  
ratings...

http://www.youtube.com/watch?v=9Katqjx5RJ4

Best Regards,
Karl J. Vesterling
kjv at ken-ton.com
202-461-3231 x0

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From diego.viola at gmail.com  Sun Oct  4 16:51:39 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 4 Oct 2009 23:51:39 +0000
Subject: [Freeswitch-users] Youtube - FreeSWITCH Promo Video
In-Reply-To: <9A87E77C-AD85-4F1E-8ED0-F9F06AA451EE@ken-ton.com>
References: <9A87E77C-AD85-4F1E-8ED0-F9F06AA451EE@ken-ton.com>
Message-ID: <86a32abc0910041651rb6a3ff9k1f6ed7d0b1a9b440@mail.gmail.com>

Very nice :)

On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling <kjv at ken-ton.com> wrote:

> Folks;
>
> Here's something that I did playing around w/ learning Apple Motion.
> It's my first Apple Motion production, so don't be too hard on the
> ratings...
>
> http://www.youtube.com/watch?v=9Katqjx5RJ4
>
> Best Regards,
> Karl J. Vesterling
> kjv at ken-ton.com
> 202-461-3231 x0
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From srinivas.ksvreddy at gmail.com  Sun Oct  4 23:24:45 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Mon, 5 Oct 2009 11:54:45 +0530
Subject: [Freeswitch-users] Fail over
Message-ID: <f8af5740910042324t76810797hb770f86ddeeba028@mail.gmail.com>

Hi,

can any know how to implement fail over with freeswitch, please help me

Regards
-- 
Srinivasula Reddy K
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From gmaruzz at celliax.org  Sun Oct  4 23:59:34 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 5 Oct 2009 08:59:34 +0200
Subject: [Freeswitch-users] Fail over
In-Reply-To: <f8af5740910042324t76810797hb770f86ddeeba028@mail.gmail.com>
References: <f8af5740910042324t76810797hb770f86ddeeba028@mail.gmail.com>
Message-ID: <7b197bef0910042359p7dc9c89agfeaac71771ea0a4@mail.gmail.com>

On Mon, Oct 5, 2009 at 8:24 AM, srinivasula reddy
<srinivas.ksvreddy at gmail.com> wrote:

> can any know how to implement fail over with freeswitch, please help me
>

This issue has been debated many many times in the mailing lists.
(hint: no live call failover, HA with OpenSERet similia as load-balancers).

Please have a look at the archives:

http://lists.freeswitch.org/pipermail/freeswitch-dev/

http://lists.freeswitch.org/pipermail/freeswitch-users/

-gm


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From gmaruzz at celliax.org  Mon Oct  5 00:04:50 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 5 Oct 2009 09:04:50 +0200
Subject: [Freeswitch-users] Youtube - FreeSWITCH Promo Video
In-Reply-To: <86a32abc0910041651rb6a3ff9k1f6ed7d0b1a9b440@mail.gmail.com>
References: <9A87E77C-AD85-4F1E-8ED0-F9F06AA451EE@ken-ton.com> 
	<86a32abc0910041651rb6a3ff9k1f6ed7d0b1a9b440@mail.gmail.com>
Message-ID: <7b197bef0910050004v71c623c8w68a013bef9fea596@mail.gmail.com>

On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola <diego.viola at gmail.com> wrote:
> Very nice :)
>
> On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling <kjv at ken-ton.com> wrote:
>>
>> Folks;
>> Here's something that I did playing around w/ learning Apple Motion.

Me too: very nice!

-gmaruzz

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From lakindia89 at gmail.com  Mon Oct  5 00:07:25 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Mon, 5 Oct 2009 12:37:25 +0530
Subject: [Freeswitch-users] oz dump Saying error
Message-ID: <7d79b3930910050007h5af4def2u622d34592a56b6ba@mail.gmail.com>

HI all,
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From diego.viola at gmail.com  Mon Oct  5 00:13:11 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Mon, 5 Oct 2009 07:13:11 +0000
Subject: [Freeswitch-users] oz dump Saying error
In-Reply-To: <7d79b3930910050007h5af4def2u622d34592a56b6ba@mail.gmail.com>
References: <7d79b3930910050007h5af4def2u622d34592a56b6ba@mail.gmail.com>
Message-ID: <86a32abc0910050013w23c6294fma3411facf5fa7c05@mail.gmail.com>

Hello?

On Mon, Oct 5, 2009 at 7:07 AM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> HI all,
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lakindia89 at gmail.com  Mon Oct  5 00:20:37 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Mon, 5 Oct 2009 12:50:37 +0530
Subject: [Freeswitch-users] oz debug says error
Message-ID: <7d79b3930910050020i1b96541dk26c8d72c220ded8d@mail.gmail.com>

Hi all,
I've compiled the freeswitch with libpri support. But when I execute

oz libpri debug 1 all, I got the following error.

API CALL [oz(libpri debug 1 all )] output:
src/ozmod/ozmod_libpri/ozmod_libpri.c: -ERR invalid span.


Here is my openzap configurations.

openzap.conf

[span zt PRI_1]
trunk_type => e1
b-channel => 1:1-15
d-channel=> 1:16
b-channel => 1:17-31


openzap.conf.xml

<configuration name="openzap.conf" description="OpenZAP Configuration">
<settings>
<param name="debug" value="9"/>
</settings>
<pri_spans>
<span id="1" name="PRI_1">
<param name="q921loglevel" value="alert"/>
<param name="q931loglevel" value="alert"/>
<param name="mode" value="user"/>
<param name="dialect" value="q931"/>
<param name="dialplan" value="XML"/>
<param name="context" value="default"/>
</span>
</pri_spans>
</configuration>

I feel something I've missed in configurations. Please tell me how to get
rid of that error.
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From lakindia89 at gmail.com  Mon Oct  5 00:22:46 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Mon, 5 Oct 2009 12:52:46 +0530
Subject: [Freeswitch-users] oz dump Saying error
In-Reply-To: <86a32abc0910050013w23c6294fma3411facf5fa7c05@mail.gmail.com>
References: <7d79b3930910050007h5af4def2u622d34592a56b6ba@mail.gmail.com>
	<86a32abc0910050013w23c6294fma3411facf5fa7c05@mail.gmail.com>
Message-ID: <7d79b3930910050022j1a9626cau93bb229436d6a92e@mail.gmail.com>

Sorry my mail client has some problem. I've send another mail with my
question. Kindly ignore this one.

On Mon, Oct 5, 2009 at 12:43 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Hello?
>
> On Mon, Oct 5, 2009 at 7:07 AM, lakshmanan ganapathy <lakindia89 at gmail.com
> > wrote:
>
>> HI all,
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gmaruzz at celliax.org  Mon Oct  5 00:27:38 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 5 Oct 2009 09:27:38 +0200
Subject: [Freeswitch-users] Fail over
In-Reply-To: <7b197bef0910042359p7dc9c89agfeaac71771ea0a4@mail.gmail.com>
References: <f8af5740910042324t76810797hb770f86ddeeba028@mail.gmail.com> 
	<7b197bef0910042359p7dc9c89agfeaac71771ea0a4@mail.gmail.com>
Message-ID: <7b197bef0910050027v3baadd2era04e469ee34ca997@mail.gmail.com>

On Mon, Oct 5, 2009 at 8:59 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
> On Mon, Oct 5, 2009 at 8:24 AM, srinivasula reddy
> <srinivas.ksvreddy at gmail.com> wrote:
>
>> can any know how to implement fail over with freeswitch, please help me
>>
>
> This issue has been debated many many times in the mailing lists.
> (hint: no live call failover, HA with OpenSERet similia as load-balancers).
>
> Please have a look at the archives:
>
> http://lists.freeswitch.org/pipermail/freeswitch-dev/
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/
>

On Mon, Oct 5, 2009 at 9:15 AM, srinivasula reddy
<srinivas.ksvreddy at gmail.com> wrote:
> Hi Giovanni Maruzzelli
>
> Thanks for your reply,
> i am new to is there any way to do live call failover.

Srinivas, are you joking ? Please take the time to read the answer,
when you ask a question.

In my previous mail, I have replied to you:

This issue has been debated many many times in the mailing lists.

(hint: no live call failover, HA with OpenSERet similia as load-balancers).

Please have a look at the archives:

http://lists.freeswitch.org/pipermail/freeswitch-dev/

http://lists.freeswitch.org/pipermail/freeswitch-users/

-gm
-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From janvb at live.com  Mon Oct  5 02:15:24 2009
From: janvb at live.com (Jan Berger)
Date: Mon, 5 Oct 2009 11:15:24 +0200
Subject: [Freeswitch-users] Fail over
In-Reply-To: <7b197bef0910042359p7dc9c89agfeaac71771ea0a4@mail.gmail.com>
References: <f8af5740910042324t76810797hb770f86ddeeba028@mail.gmail.com>
Message-ID: <BLU127-W222B2636075E6063A62964D2CF0@phx.gbl>


hi,

 

FreeSWITCH "as is" have no live fail-over support, but this will change.

 

A live fail-over and redundancy mechanism is part of what SIGTRAN provides of added values. I am working on this, but it will take time before this is on a functional stage and available.

 

Also - SIGTRAN only provide failover and redundancy on L2/L3 signalling or higher.

 

This will however not provide failover on E1/T1 hardware level. The later can be achieved by different techniques, but the easiest is if the xternal switch is configured to re-connect a lost call on a different E1/T1. Most proper switches can provide this service. 

 

Jan


 
> From: gmaruzz at celliax.org
> Date: Mon, 5 Oct 2009 08:59:34 +0200
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Fail over
> 
> On Mon, Oct 5, 2009 at 8:24 AM, srinivasula reddy
> <srinivas.ksvreddy at gmail.com> wrote:
> 
> > can any know how to implement fail over with freeswitch, please help me
> >
> 
> This issue has been debated many many times in the mailing lists.
> (hint: no live call failover, HA with OpenSERet similia as load-balancers).
> 
> Please have a look at the archives:
> 
> http://lists.freeswitch.org/pipermail/freeswitch-dev/
> 
> http://lists.freeswitch.org/pipermail/freeswitch-users/
> 
> -gm
> 
> 
> -- 
> Sincerely,
> 
> Giovanni Maruzzelli
> Cell : +39-347-2665618
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
_________________________________________________________________
Windows Live: Make it easier for your friends to see what you?re up to on Facebook.
http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_2:092009
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From mcampbellsmith at gmail.com  Mon Oct  5 03:28:37 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Mon, 5 Oct 2009 21:28:37 +1100
Subject: [Freeswitch-users] Detecting a fax
In-Reply-To: <91832961-FBA6-465F-AFD8-6A1BE604F6BC@jerris.com>
References: <33c87fa30910040603p7929df26l88f51fc7d256408f@mail.gmail.com>
	<91832961-FBA6-465F-AFD8-6A1BE604F6BC@jerris.com>
Message-ID: <33c87fa30910050328l3d119939pd638b44b219a0140@mail.gmail.com>

Thanks for the response Mike,

I read that page and this one (among others)
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_fax_detect, but
I'm still lost.  This is an extract of my dialplan

    <extension name="Local">
      <condition field="destination_number" expression="^(10[01][0-9])$">
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
        <action application="set" data="ringback=${au-ring}"/>
        <action application="fax_detect"/>
        <action application="tone_detect" data="fax 1100 r +5000
transfer fax XML features" />
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="bridge" data="user/${dialed_extension}@${domain}"/>

I would assume that on detecting a fax, the dialplan 'fax' is called
in context features.  This never happens.

When is the fax tone detected?   Is it while the call is ringing or
can it be detected after the call is answered?  My goal is to be able
to have the same extension for a voice and fax call.  i assume that
the fax 'tones' are standardised and the ones on the wiki are correct?
 Also, I guess this doesn't work with media bypass (which I don't
use).

Thanks!


On Mon, Oct 5, 2009 at 9:56 AM, Michael Jerris <mike at jerris.com> wrote:
> check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
>
> Note, you can't just have tone_detect as your last iten in the
> dialplan as the call will just get hung up.
>
> Mike
>
> On Oct 4, 2009, at 9:03 AM, Mark Campbell-Smith wrote:
>
>> Hi
>>
>> I was hoping someone could help me to setup the fax detection / tone
>> detection application.
>>
>> I want to be able to transfer an incoming fax to a specific extension.
>> In my default.xml file, I have the following (extracted):
>>
>> ? ?<extension name="1000">
>> ? ? ?<condition field="destination_number" expression="^(10[01][0-9])
>> $">
>> ? ? ? ?<action application="fax_detect"/>
>> ? ? ? ?<action application="tone_detect" data="fax 1100 r +5000
>> transfer fax XML features" />
>>
>> I can't get the fax to be detected and transferred. ?Is there any way
>> this can be done?
>>
>> Thanks!
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From Russell.Mosemann at cune.org  Mon Oct  5 03:30:26 2009
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Mon, 5 Oct 2009 05:30:26 -0500
Subject: [Freeswitch-users] oz debug says error
In-Reply-To: <7d79b3930910050020i1b96541dk26c8d72c220ded8d@mail.gmail.com>
References: <7d79b3930910050020i1b96541dk26c8d72c220ded8d@mail.gmail.com>
Message-ID: <9F7D1E871F0C4184AA8ED441CF415C8A@cune.pri>

> On Behalf Of lakshmanan ganapathy
...
> I've compiled the freeswitch with libpri support. But when I execute
> oz libpri debug 1 all, I got the following error.
> 
> API CALL [oz(libpri debug 1 all )] output:
> src/ozmod/ozmod_libpri/ozmod_libpri.c: -ERR invalid span.

If you would start freeswitch from the command line or look at freeswitch/log/freeswitch.log, you will see during startup that libpri does not find a span (note the ozmod lines). That's because the configuration below is not for libpri.

> openzap.conf.xml
...
> <pri_spans>

You need to use a libpri span configuration.

http://wiki.freeswitch.org/wiki/Openzap.conf.xml_Examples#Using_with_PRI_.28libpri_compatibility_stack.29

--
Russell Moseman




From tculjaga at gmail.com  Mon Oct  5 04:13:04 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 5 Oct 2009 13:13:04 +0200
Subject: [Freeswitch-users] configure FS: config.status: error: cannot
	find input file: Makefile.in
In-Reply-To: <4690D578-FBC2-4230-BDF0-C43CDF6669A0@jerris.com>
References: <65d96fc80910020505t48a6237ao19884c23fedf228d@mail.gmail.com>
	<4690D578-FBC2-4230-BDF0-C43CDF6669A0@jerris.com>
Message-ID: <65d96fc80910050413je176581i5fe06b9877ada592@mail.gmail.com>

it works,

thx!


T.

On Mon, Oct 5, 2009 at 12:31 AM, Michael Jerris <mike at jerris.com> wrote:

> I updated the tiff lib to build better inline, try make  tiff-reconf
> Mike
>
> On Oct 2, 2009, at 8:05 AM, Tihomir Culjaga wrote:
>
> hello,
> i just got the last trunk and tried to compile it on one of my development
> machines... Well configure fails on tiff-3.8.2 where it is unable to find
> Makefile.in ... Can someone advice?
>
>
>
> checking if g++ static flag -static works... yes
> checking if g++ supports -c -o file.o... yes
> checking if g++ supports -c -o file.o... (cached) yes
> checking whether the g++ linker (/usr/bin/ld) supports shared libraries...
> yes
> checking dynamic linker characteristics... GNU/Linux ld.so
> checking how to hardcode library paths into programs... immediate
> checking for OpenGL Utility library... no
> checking for GLUT library... no
> configure: creating ./config.status
> config.status: error: cannot find input file: Makefile.in
>
>
>
> tculjaga at subZero:~/freeswitch-trunk/libs/tiff-3.8.2$ ls -l
> total 2224
> -rw-r--r--  1 tculjaga tculjaga  23741 2009-10-02 13:19 acinclude.m4
> -rw-r--r--  1 tculjaga tculjaga 316978 2009-10-02 13:28 aclocal.m4
> -rwxr-xr-x  1 tculjaga tculjaga    121 2009-10-02 13:19 autogen.sh
> -rw-r--r--  1 tculjaga tculjaga 124047 2009-10-02 13:19 ChangeLog
> drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:28 config
> -rw-r--r--  1 tculjaga tculjaga 103926 2009-10-02 14:00 config.log
> -rwxr-xr-x  1 tculjaga tculjaga  73065 2009-10-02 14:00 config.status
> -rwxr-xr-x  1 tculjaga tculjaga 740145 2009-10-02 13:28 configure
> -rw-r--r--  1 tculjaga tculjaga  20492 2009-10-02 13:19 configure.ac
> -rwxr-xr-x  1 tculjaga tculjaga     56 2009-10-02 13:19 configure.gnu
> -rwxr-xr-x  1 tculjaga tculjaga 737794 2009-10-02 13:57 configure.lineno
> drwxr-xr-x 16 tculjaga tculjaga   4096 2009-10-02 13:19 contrib
> -rw-r--r--  1 tculjaga tculjaga   1146 2009-10-02 13:19 COPYRIGHT
> -rw-r--r--  1 tculjaga tculjaga   1570 2009-10-02 13:19 HOWTO-RELEASE
> drwxr-xr-x  5 tculjaga tculjaga   4096 2009-10-02 13:19 html
> drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:28 libtiff
> drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 m4
> -rw-r--r--  1 tculjaga tculjaga   1908 2009-10-02 13:19 Makefile.am
> -rw-r--r--  1 tculjaga tculjaga   1724 2009-10-02 13:19 Makefile.vc
> drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 man
> -rw-r--r--  1 tculjaga tculjaga   6270 2009-10-02 13:19 nmake.opt
> drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 port
> -rw-r--r--  1 tculjaga tculjaga   2363 2009-10-02 13:19 README
> -rw-r--r--  1 tculjaga tculjaga      9 2009-10-02 13:19 RELEASE-DATE
> -rw-r--r--  1 tculjaga tculjaga   5893 2009-10-02 13:19 SConstruct
> drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 test
> -rw-r--r--  1 tculjaga tculjaga    433 2009-10-02 13:19 TODO
> drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 tools
> -rw-r--r--  1 tculjaga tculjaga      6 2009-10-02 13:19 VERSION
> tculjaga at subZero:~/freeswitch-trunk/libs/tiff-3.8.2$
> tculjaga at subZero:~/freeswitch-trunk/libs/tiff-3.8.2$
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lakindia89 at gmail.com  Mon Oct  5 05:17:30 2009
From: lakindia89 at gmail.com (lakshmanan)
Date: Mon, 5 Oct 2009 05:17:30 -0700 (PDT)
Subject: [Freeswitch-users] oz debug says error
In-Reply-To: <9F7D1E871F0C4184AA8ED441CF415C8A@cune.pri>
References: <7d79b3930910050020i1b96541dk26c8d72c220ded8d@mail.gmail.com>
	<9F7D1E871F0C4184AA8ED441CF415C8A@cune.pri>
Message-ID: <25749736.post@talk.nabble.com>


Thanks for pointing that.
I also tried that.
But in that case, I'm not able to make a call through openzap.

When I say 
  originate openzap/1/A/number number

It reported the following error
2009-10-05 17:45:47.733495 [ERR] ozmod_libpri.c:88 Can't destroy call 0!
API CALL [originate(openzap/1/1/9952248266 9952248266)] output:
-ERR INVALID_IE_CONTENTS

I also gone and looked up the hangup_cause page for the reason. But I was
unable to understand that.
Can u please tell why it is reporting this error?



Russell.Mosemann wrote:
> 
>> On Behalf Of lakshmanan ganapathy
> ...
>> I've compiled the freeswitch with libpri support. But when I execute
>> oz libpri debug 1 all, I got the following error.
>> 
>> API CALL [oz(libpri debug 1 all )] output:
>> src/ozmod/ozmod_libpri/ozmod_libpri.c: -ERR invalid span.
> 
> If you would start freeswitch from the command line or look at
> freeswitch/log/freeswitch.log, you will see during startup that libpri
> does not find a span (note the ozmod lines). That's because the
> configuration below is not for libpri.
> 
>> openzap.conf.xml
> ...
>> <pri_spans>
> 
> You need to use a libpri span configuration.
> 
> http://wiki.freeswitch.org/wiki/Openzap.conf.xml_Examples#Using_with_PRI_.28libpri_compatibility_stack.29
> 
> --
> Russell Moseman
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/oz-debug-says-error-tp25746215p25749736.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From maciej.aniserowicz at gmail.com  Mon Oct  5 00:10:59 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Mon, 5 Oct 2009 09:10:59 +0200
Subject: [Freeswitch-users] Bad sound quality while eavesdropping
Message-ID: <41A44DD027064988A914974405788C2E@procent>

Hello,
When I use eavesdropping in FreeSWITCH, the sound quality is really bad. Is there any way to improve it? Is this a known problem?
Br/
Maciej Aniserowicz
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From maciej.aniserowicz at gmail.com  Mon Oct  5 00:13:43 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Mon, 5 Oct 2009 09:13:43 +0200
Subject: [Freeswitch-users] gateway FS informs it's client FS about users
	hanged up with a long delay
Message-ID: <1A16501E57CA4727A593226A8C810308@procent>

Hi,
When I use two FreeSWITCH instances ('internal' and 'external'), all users register to the 'external' instance which acts as a gateway by 'internal' instance (which in turn is controlled by my applicaiton with commands sent by socket).
When user hangs up, the 'hanged up' event is propagated to the 'internal' instance after a long time (~3 minutes) instead of being propagated immediately.
What can cause this issue?

Br/
Maciej Aniserowicz
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From maciej.aniserowicz at gmail.com  Mon Oct  5 00:16:10 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Mon, 5 Oct 2009 09:16:10 +0200
Subject: [Freeswitch-users] Recording creates a 388-byte long file and
	deletes it
Message-ID: <4ED3AB65AFE34242AACDE97127FE1248@procent>

Hi,
When I record a call in FS, it only creates a 388-byte-long wav file. The conversation is no written there, and FS deletes the file when the session finishes.
What can cause this strange behavior?

Br/
Maciej Aniserowicz
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From mike at jerris.com  Mon Oct  5 05:32:47 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 5 Oct 2009 08:32:47 -0400
Subject: [Freeswitch-users] Detecting a fax
In-Reply-To: <33c87fa30910050328l3d119939pd638b44b219a0140@mail.gmail.com>
References: <33c87fa30910040603p7929df26l88f51fc7d256408f@mail.gmail.com>
	<91832961-FBA6-465F-AFD8-6A1BE604F6BC@jerris.com>
	<33c87fa30910050328l3d119939pd638b44b219a0140@mail.gmail.com>
Message-ID: <402585BB-EBEA-42B8-9A2A-58FBA19B066F@jerris.com>

Fax tones are not played by the remote machine until after answer, the  
tone_detect application starts a media bug that listens for the tone,  
can you confirm the tone is happening at all.  Maybe the issue here is  
the timeout, try making that longer, or doing the tone_detect in  
execute_on_answer

Mike

On Oct 5, 2009, at 6:28 AM, Mark Campbell-Smith wrote:

> Thanks for the response Mike,
>
> I read that page and this one (among others)
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_fax_detect, but
> I'm still lost.  This is an extract of my dialplan
>
>    <extension name="Local">
>      <condition field="destination_number" expression="^(10[01][0-9]) 
> $">
>        <action application="set" data="dialed_extension=$1"/>
>        <action application="export" data="dialed_extension=$1"/>
>        <action application="set" data="ringback=${au-ring}"/>
>        <action application="fax_detect"/>
>        <action application="tone_detect" data="fax 1100 r +5000
> transfer fax XML features" />
>        <action application="set" data="hangup_after_bridge=true"/>
>        <action application="set" data="continue_on_fail=true"/>
>        <action application="bridge" data="user/${dialed_extension}@$ 
> {domain}"/>
>
> I would assume that on detecting a fax, the dialplan 'fax' is called
> in context features.  This never happens.
>
> When is the fax tone detected?   Is it while the call is ringing or
> can it be detected after the call is answered?  My goal is to be able
> to have the same extension for a voice and fax call.  i assume that
> the fax 'tones' are standardised and the ones on the wiki are correct?
> Also, I guess this doesn't work with media bypass (which I don't
> use).
>
> Thanks!
>
>
> On Mon, Oct 5, 2009 at 9:56 AM, Michael Jerris <mike at jerris.com>  
> wrote:
>> check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
>>
>> Note, you can't just have tone_detect as your last iten in the
>> dialplan as the call will just get hung up.
>>
>> Mike
>>
>> On Oct 4, 2009, at 9:03 AM, Mark Campbell-Smith wrote:
>>
>>> Hi
>>>
>>> I was hoping someone could help me to setup the fax detection / tone
>>> detection application.
>>>
>>> I want to be able to transfer an incoming fax to a specific  
>>> extension.
>>> In my default.xml file, I have the following (extracted):
>>>
>>>    <extension name="1000">
>>>      <condition field="destination_number" expression="^(10[01] 
>>> [0-9])
>>> $">
>>>        <action application="fax_detect"/>
>>>        <action application="tone_detect" data="fax 1100 r +5000
>>> transfer fax XML features" />
>>>
>>> I can't get the fax to be detected and transferred.  Is there any  
>>> way
>>> this can be done?
>>>
>>> Thanks!




From xengelpublicx at gmail.com  Mon Oct  5 05:43:26 2009
From: xengelpublicx at gmail.com (Vladimir Elizarov)
Date: Mon, 05 Oct 2009 16:43:26 +0400
Subject: [Freeswitch-users] stun not working in fs 1.0.4?
Message-ID: <4AC9E9EE.8090805@gmail.com>

Hello.

I'm trying to configure stun in fs 1.0.4

vars.xml
<X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>
<X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:example.com"/>
<X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:example.com"/>

external.xml
    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <param name="ext-sip-ip" value="$${external_sip_ip}"/>
    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>

In this configuration, the address in ext_rtp_ip transmitted literally
(it is: "stun:stun.exmaple.com")

If you do not specify ext_rtp_ip stun then allegedly began to work. But
as in bug SFSIP-163
(http://jira.freeswitch.org/browse/SFSIP-163?page=com.atlassian.jira.plugin.system.issuetabpanels%3Acomment-tabpanel
) Via: IP is replaced, and SDP - no.

How do I fix this? Upgrade fs to trunk?

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From Russell.Mosemann at cune.org  Mon Oct  5 05:48:58 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Mon, 5 Oct 2009 12:48:58 -0000
Subject: [Freeswitch-users] oz debug says error
In-Reply-To: <25749736.post@talk.nabble.com>
Message-ID: <20091005124858.81857415806@mail.cune.org>

lakshmanan <lakindia89 at gmail.com> said:
> Thanks for pointing that.
> I also tried that.
> But in that case, I'm not able to make a call through openzap.

What is in openzap.conf.xml? If you start fs_cli and enter "oz list",
what does it show? Copy the ozmod lines from freeswitch.log to
pastebin.freeswitch.org and post the link here so that we can see what
openzap does when freeswitch starts.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!




From nagalenoj at gmail.com  Mon Oct  5 06:46:34 2009
From: nagalenoj at gmail.com (Nagalenoj)
Date: Mon, 5 Oct 2009 06:46:34 -0700 (PDT)
Subject: [Freeswitch-users] Re corded file as voicemail.
Message-ID: <25751158.post@talk.nabble.com>


Is it possible to treat a recorded voice as voice mail?

Assume that, I've recorded a conversation and I want this recorded file to
be treated like voicemail. So, I could check it like voicemail!!

-- 
View this message in context: http://www.nabble.com/Recorded-file-as-voicemail.-tp25751158p25751158.html
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From nagalenoj at gmail.com  Mon Oct  5 06:50:39 2009
From: nagalenoj at gmail.com (Nagalenoj)
Date: Mon, 5 Oct 2009 06:50:39 -0700 (PDT)
Subject: [Freeswitch-users]  UUID of the newly originated call?
Message-ID: <25751228.post@talk.nabble.com>


Dear friends,
     I am trying with ESL outbound socket. I'm trying to make a call when I
receive ANSWER event. Now, I would want to do something like,
     * Receive the events only for this uuid - I have done by registering
all events and filtering only for this uuid($uuid).
     * If it is CHANNEL_ANSWER, originate a new call.

Now, How can I get the uuid of the new call and receive events for this new
call.? I want to receive the events for both uuids.



-- 
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From anthony.minessale at gmail.com  Mon Oct  5 07:31:18 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 5 Oct 2009 09:31:18 -0500
Subject: [Freeswitch-users] Bad sound quality while eavesdropping
In-Reply-To: <41A44DD027064988A914974405788C2E@procent>
References: <41A44DD027064988A914974405788C2E@procent>
Message-ID: <191c3a030910050731m2d74979ep4598e5a1945d58ae@mail.gmail.com>

That's is a somewhat vague position.

You did not mention which version of FreeSWITCH you are running, the phones
being used in your example, your configuration, the codecs in use etc.

BTW,
I think you should only ask one question at a time on this list.  The list
is run by volunteers and it's sort of rude to expect 3 or 4 threads to be
tended to concerning the same one individual.


2009/10/5 Maciej Aniserowicz <maciej.aniserowicz at gmail.com>

>  Hello,
> When I use eavesdropping in FreeSWITCH, the sound quality is really bad. Is
> there any way to improve it? Is this a known problem?
> Br/
> Maciej Aniserowicz
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From tculjaga at gmail.com  Mon Oct  5 07:32:32 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 5 Oct 2009 16:32:32 +0200
Subject: [Freeswitch-users] Detecting a fax
In-Reply-To: <402585BB-EBEA-42B8-9A2A-58FBA19B066F@jerris.com>
References: <33c87fa30910040603p7929df26l88f51fc7d256408f@mail.gmail.com>
	<91832961-FBA6-465F-AFD8-6A1BE604F6BC@jerris.com>
	<33c87fa30910050328l3d119939pd638b44b219a0140@mail.gmail.com>
	<402585BB-EBEA-42B8-9A2A-58FBA19B066F@jerris.com>
Message-ID: <65d96fc80910050732g2f414dffs45375cfb16b08c39@mail.gmail.com>

hi Mark,

This is an inbound call leg and media channel (so far)  is open in reverse
direction only (application ringback). I'm afraid you have to answer the
call to be able to "hear" the fax tone.

T.



On Mon, Oct 5, 2009 at 2:32 PM, Michael Jerris <mike at jerris.com> wrote:

> Fax tones are not played by the remote machine until after answer, the
> tone_detect application starts a media bug that listens for the tone,
> can you confirm the tone is happening at all.  Maybe the issue here is
> the timeout, try making that longer, or doing the tone_detect in
> execute_on_answer
>
> Mike
>
> On Oct 5, 2009, at 6:28 AM, Mark Campbell-Smith wrote:
>
> > Thanks for the response Mike,
> >
> > I read that page and this one (among others)
> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_fax_detect, but
> > I'm still lost.  This is an extract of my dialplan
> >
> >    <extension name="Local">
> >      <condition field="destination_number" expression="^(10[01][0-9])
> > $">
> >        <action application="set" data="dialed_extension=$1"/>
> >        <action application="export" data="dialed_extension=$1"/>
> >        <action application="set" data="ringback=${au-ring}"/>
> >        <action application="fax_detect"/>
> >        <action application="tone_detect" data="fax 1100 r +5000
> > transfer fax XML features" />
> >        <action application="set" data="hangup_after_bridge=true"/>
> >        <action application="set" data="continue_on_fail=true"/>
> >        <action application="bridge" data="user/${dialed_extension}@$
> > {domain}"/>
> >
> > I would assume that on detecting a fax, the dialplan 'fax' is called
> > in context features.  This never happens.
> >
> > When is the fax tone detected?   Is it while the call is ringing or
> > can it be detected after the call is answered?  My goal is to be able
> > to have the same extension for a voice and fax call.  i assume that
> > the fax 'tones' are standardised and the ones on the wiki are correct?
> > Also, I guess this doesn't work with media bypass (which I don't
> > use).
> >
> > Thanks!
> >
> >
> > On Mon, Oct 5, 2009 at 9:56 AM, Michael Jerris <mike at jerris.com>
> > wrote:
> >> check out
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
> >>
> >> Note, you can't just have tone_detect as your last iten in the
> >> dialplan as the call will just get hung up.
> >>
> >> Mike
> >>
> >> On Oct 4, 2009, at 9:03 AM, Mark Campbell-Smith wrote:
> >>
> >>> Hi
> >>>
> >>> I was hoping someone could help me to setup the fax detection / tone
> >>> detection application.
> >>>
> >>> I want to be able to transfer an incoming fax to a specific
> >>> extension.
> >>> In my default.xml file, I have the following (extracted):
> >>>
> >>>    <extension name="1000">
> >>>      <condition field="destination_number" expression="^(10[01]
> >>> [0-9])
> >>> $">
> >>>        <action application="fax_detect"/>
> >>>        <action application="tone_detect" data="fax 1100 r +5000
> >>> transfer fax XML features" />
> >>>
> >>> I can't get the fax to be detected and transferred.  Is there any
> >>> way
> >>> this can be done?
> >>>
> >>> Thanks!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From rupa at rupa.com  Mon Oct  5 07:45:30 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 5 Oct 2009 08:45:30 -0600
Subject: [Freeswitch-users] stun not working in fs 1.0.4?
In-Reply-To: <4AC9E9EE.8090805@gmail.com>
References: <4AC9E9EE.8090805@gmail.com>
Message-ID: <b39273530910050745r659d8cd6vf52ec1748bff25a2@mail.gmail.com>

Yes, the stun thing was fixed after 1.4 I believe.

On Mon, Oct 5, 2009 at 6:43 AM, Vladimir Elizarov
<xengelpublicx at gmail.com> wrote:
>
> How do I fix this? Upgrade fs to trunk?



-- 
-Rupa



From woodydickson at gmail.com  Mon Oct  5 07:49:39 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Mon, 5 Oct 2009 22:49:39 +0800
Subject: [Freeswitch-users] overriding conference preference
Message-ID: <a44688a0910050749s1ae1f9a4vba96477d159d377f@mail.gmail.com>

Hi,

Is is possible to override any of the setting specified in the conference
profile?

What I want to do is to have a default profile, and be able to modify
certain fields if necessary in the dialplan.


Alternatively, I would prefer to have a dynamic profile setting for the
conference to obtain those parameters from odbc.

Is it possible?

woody
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From anthony.minessale at gmail.com  Mon Oct  5 08:02:44 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 5 Oct 2009 10:02:44 -0500
Subject: [Freeswitch-users] Youtube - FreeSWITCH Promo Video
In-Reply-To: <7b197bef0910050004v71c623c8w68a013bef9fea596@mail.gmail.com>
References: <9A87E77C-AD85-4F1E-8ED0-F9F06AA451EE@ken-ton.com>
	<86a32abc0910041651rb6a3ff9k1f6ed7d0b1a9b440@mail.gmail.com>
	<7b197bef0910050004v71c623c8w68a013bef9fea596@mail.gmail.com>
Message-ID: <191c3a030910050802v59b2a9du6d2af7ade03bc509@mail.gmail.com>

neat,

Here's some suggestions for your next ones. =p

Have them standing around the hologram trying to destroy the "Death
Star(tm)" that happens to look a lot like a giant 3d unix '*' character.
Then have one rebel say, "wait!, why are we wasting our time... watch
this... and dial a number on his cellphone as the whole thing explodes in
the background.

Have Darth Forkium face Luke ThreadSpawner in a dual.  "I see you have
fashioned your own TDM card...." vroom......
"Join me and together we can make linked lists and monolithic processes",
"NEVER!..." vroom vroom Master Coda has taught you well....."You are no
match for me...JOIN THE ORANGE SIDE OF THE FORCE"





On Mon, Oct 5, 2009 at 2:04 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola <diego.viola at gmail.com> wrote:
> > Very nice :)
> >
> > On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling <kjv at ken-ton.com>
> wrote:
> >>
> >> Folks;
> >> Here's something that I did playing around w/ learning Apple Motion.
>
> Me too: very nice!
>
> -gmaruzz
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mike at jerris.com  Mon Oct  5 08:03:04 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 5 Oct 2009 11:03:04 -0400
Subject: [Freeswitch-users] Re corded file as voicemail.
In-Reply-To: <25751158.post@talk.nabble.com>
References: <25751158.post@talk.nabble.com>
Message-ID: <BC536E8C-6813-4A69-9B80-EA11E41A79FC@jerris.com>

http://wiki.freeswitch.org/wiki/Mod_voicemail#voicemail_inject

On Oct 5, 2009, at 9:46 AM, Nagalenoj wrote:

>
> Is it possible to treat a recorded voice as voice mail?
>
> Assume that, I've recorded a conversation and I want this recorded  
> file to
> be treated like voicemail. So, I could check it like voicemail!!




From brian at freeswitch.org  Mon Oct  5 08:04:23 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Oct 2009 10:04:23 -0500
Subject: [Freeswitch-users] stun not working in fs 1.0.4?
In-Reply-To: <4AC9E9EE.8090805@gmail.com>
References: <4AC9E9EE.8090805@gmail.com>
Message-ID: <7C489261-F5F3-4B57-B807-21EC4A3947A2@freeswitch.org>

Yes!

/b

On Oct 5, 2009, at 7:43 AM, Vladimir Elizarov wrote:

> How do I fix this? Upgrade fs to trunk?




From mike at jerris.com  Mon Oct  5 08:05:15 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 5 Oct 2009 11:05:15 -0400
Subject: [Freeswitch-users] overriding conference preference
In-Reply-To: <a44688a0910050749s1ae1f9a4vba96477d159d377f@mail.gmail.com>
References: <a44688a0910050749s1ae1f9a4vba96477d159d377f@mail.gmail.com>
Message-ID: <E773DB8B-AC74-49DE-9B38-7AFC441882F5@jerris.com>


On Oct 5, 2009, at 10:49 AM, Woody Dickson wrote:

> Is is possible to override any of the setting specified in the  
> conference profile?

Just the flags you can pass per user such as pin and mute

>
> What I want to do is to have a default profile, and be able to  
> modify certain fields if necessary in the dialplan.
>
>
> Alternatively, I would prefer to have a dynamic profile setting for  
> the conference to obtain those parameters from odbc.

you can do this with mod_xml_curl

Mike




From kevin at johnnyvoip.com  Mon Oct  5 08:13:20 2009
From: kevin at johnnyvoip.com (Kevin Green)
Date: Mon, 5 Oct 2009 11:13:20 -0400
Subject: [Freeswitch-users] Gateways, Limits, & Routes
Message-ID: <ab57a3e80910050813i1e517723t1107fd70d05214af@mail.gmail.com>

It seems many people are looking for ways to control gateways, resiliency of
termination, and limit on connections easily in FS. Here are some of the
thoughts I had, and I would like to hear what others think of this.

In tradition phone hardware you would define lines, put them into a group,
and then assign a route to go through that group. For resiliency you could
group multiple routes together into a route list, if the first route failed,
was all busy, or unavailable it would start to try the second route. If you
go out a secondary route you can also play a warning tone to indicate this
might be going out a more expensive connection. For example you may have an
IP link between two boxes, then fail back to TDM if the IP links go down,
the TDM would be more expensive so you would want to warn business users so
they don't spend hours on the phone.

Inbound and outbound calls could both go on the lines so when a call comes
in or goes out a route it takes up one slot.

My thought is, why don't we create something similar to this that will allow
us to handle a lot of these cases without complex dialplans. We could create
routes that are assigned to gateways and limit the number of incoming,
outgoing, and total connections that can be on that route. We could also
specify what cases we would consider as failure to move onto the next route
if we are in a route list. Route lists would then be similar to using
multiple limits, and failover, but in this case it would simply be a list.

The following is an example of how I think this could be prgrammed.

<routes>
   <route id="route1" gateway="provider1" max-inbound="5" max-outbound="5"
max-total="5"
countinue_on_fail="NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION"/>
   <route id="route2" gateway="provider2" max-inbound="0" max-outbound="5"
max-total="5" countinue_on_fail="NORMAL_TEMPORARY_FAILURE"/>
   <route id="route3" gateway="provider3" max-inbound="5" max-outbound="5"
max-total="10" countinue_on_fail="NO_ROUTE_DESTINATION"/>
</routes>
<route_lists>
   <route_list id="list1">
          <route id="route1"/>
          <route id="route2" warning_tone="expensive.wav"/>
          <route id="route3" warning_tone="really_expensive.wav"/>
   </route_list>
</route_litsts>

Then in your dialplan you would simply put:

<extension name="outbound_rout">
   <condition field="destination_number" expression="^(\d{10})$">
      <action application="bridge" data="route/route2/$1"/>
   </condition>
</extension>

OR

<extension name="outbound_route">
   <condition field="destination_number" expression="^(\d{10})$">
      <action application="bridge" data="route_list/list1/$1"/>
   </condition>
</extension>

   The module would need to track not only outbound calls, but also inbound
calls that come in through the specified gateways. This would help track
cases where you have say only two channels that can be used for both
incoming and outgoing calls.

   I'm not 100% sure if this is all feasable or if it would be that much of
an improvement compaired to what is already there so I put it out to all of
you for feedback.

Regards,
   Kevin Green

JohnnyVoIP
350 Legget Drive
Kanata, ON, Canada
K2K 2W7

Phone:   613 271 5993 Ext 1203
Fax:       613 271 9810
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From fredyg at negosat.com  Mon Oct  5 08:29:19 2009
From: fredyg at negosat.com (Fredy Gonzales)
Date: Mon, 5 Oct 2009 10:29:19 -0500
Subject: [Freeswitch-users] Youtube - FreeSWITCH Promo Video
References: <9A87E77C-AD85-4F1E-8ED0-F9F06AA451EE@ken-ton.com>
	<86a32abc0910041651rb6a3ff9k1f6ed7d0b1a9b440@mail.gmail.com>
	<7b197bef0910050004v71c623c8w68a013bef9fea596@mail.gmail.com>
Message-ID: <11A305EDADB9418888807581F6759586@gcg.com.pe>

Espectacular!!.

FG
----- Original Message ----- 
From: "Giovanni Maruzzelli" <gmaruzz at celliax.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Monday, October 05, 2009 2:04 AM
Subject: Re: [Freeswitch-users] Youtube - FreeSWITCH Promo Video


> On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola <diego.viola at gmail.com> wrote:
>> Very nice :)
>>
>> On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling <kjv at ken-ton.com> wrote:
>>>
>>> Folks;
>>> Here's something that I did playing around w/ learning Apple Motion.
>
> Me too: very nice!
>
> -gmaruzz
>
> -- 
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 




From msc at freeswitch.org  Mon Oct  5 08:43:57 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 5 Oct 2009 08:43:57 -0700
Subject: [Freeswitch-users] New to freeswitch and have a few questions
In-Reply-To: <8F97B3DC-C0B5-403C-9117-A6B4E64A0492@jerris.com>
References: <4AC5534C.6050202@tx.rr.com>
	<a87d43fd0910020910l5db88c5fv980c8c8fe312f380@mail.gmail.com>
	<8F97B3DC-C0B5-403C-9117-A6B4E64A0492@jerris.com>
Message-ID: <87f2f3b90910050843m2ec3cbf9rb232ad62fa313917@mail.gmail.com>

I've got a copy of Mike G's document that I'm reviewing. Like he said, it's
not 100% complete, however at first glance it looks like it would be a
perfect fit for a config example page. I'll get with Mike G shortly and
we'll have it up on the wiki in the next few days for everyone to review.

-MC

On Sun, Oct 4, 2009 at 3:35 PM, Michael Jerris <mike at jerris.com> wrote:

> Getting documentation on like this on the wiki would be awesome.
> Mike
>
> On Oct 2, 2009, at 12:10 PM, Michael Gende wrote:
>
> Hey Orien,
>
> I'm not using exactly your set up, but am using pfsense/FreeBSD. Since
> you're using that, I assume you're going "dual homed". I've got a starter
> guide that might help you out. If nothing else, I'd be interested in a
> candid assessment of its usefulness or lack thereof, especially to a guy
> like you.
>
> I've included it here. Its all just text at the moment so be advised. Also
> be advised that there's a lot of great information on the freeswitch site
> and in this group. The goal of my document was so that someone just starting
> would have to hunt a little less.
>
> Hope its good for something, let me know either way, especially if you find
> errors.
>
> Regards,
>
> Mike G.
>
> On Thu, Oct 1, 2009 at 8:11 PM, Orien Love <orien at tx.rr.com> wrote:
>
>> Hello Everybody,
>>    I am new to freeswitch, so forgive me if I ask stupid questions. I
>> am planning a test setup consisting of:
>>    1 - Pfsense router with the freeswitch package installed.
>>    1 - Cisco WS-C3524-PWR-XL-EN 24 FE Switch, POE for the phones.
>>    1 - LINKSYS SPA3000 to connect to my existing land line and phones.
>>    2 - POLYCOM SOUNDPOINT IP601 SIP IP PHONEs
>>
>> The first question I have, Are the IP601 phones supported? The wiki
>> lists 320, 431, 501, 550, 650 but not the 601.
>>
>> Second, is there a place that helps a person new to the IP phone world
>> learn what is needed to set up a PBX using freeswitch at a small office?
>>
>> Finally is my test setup a good one? is there something I am missing or
>> that I need to get the learning process started, I have found in the
>> past, with a little information and a test system, I can learn what I am
>> doing by breaking and fixing the test bed.
>>
>> Thanks for your time
>> Orien
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> <freeswitch.txt>_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Oct  5 09:05:35 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 5 Oct 2009 09:05:35 -0700
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <191DBBEF043B4EFE8F680B82C1038FC7@cune.pri>
References: <87f2f3b90910020908p1aa5f5d4o8b69894cd4a58bef@mail.gmail.com>
	<20091002172418.9E8AF2E19CE@mail.cune.org>
	<87f2f3b90910021116l209a44bcg124016e58735ca37@mail.gmail.com>
	<E2C130F6CC9B4F59B069CEF8D0B0AC42@cune.pri>
	<87f2f3b90910021218i434f67cbjc33073f8550c8d1a@mail.gmail.com>
	<8E079E8CB37E4FA49363D5315F6E878E@cune.pri>
	<87f2f3b90910021700h3431bd78q3b4f6ced255ecee5@mail.gmail.com>
	<191DBBEF043B4EFE8F680B82C1038FC7@cune.pri>
Message-ID: <87f2f3b90910050905g68ad8ec2xb7f834ba90f40d0b@mail.gmail.com>

On Fri, Oct 2, 2009 at 5:54 PM, Russell Mosemann
<Russell.Mosemann at cune.org>wrote:

> > Exactly. Turn on q931 debugging and try again:
> >
> > oz libpri debug 1 all
> > PB the results again and we'll check it out.
> > -MC
>
> Here's the next one. I'm not sure what to look for, but nothing pops out
> right away.
>
> http://pastebin.freeswitch.org/10571
>
>
Confirmed: the Hicomm isn't sending anything at all in the SETUP message
except the usual stuff: dialed number, channel number, etc. Does the Hicomm
have any config parameters, like Caller ID presentation?

-MC
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From msc at freeswitch.org  Mon Oct  5 09:20:57 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 5 Oct 2009 09:20:57 -0700
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <1b46b4e80910041609x2f41ceb9u142cb484951a21ed@mail.gmail.com>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<191c3a030909291122t770af037j5b1068197b31de87@mail.gmail.com>
	<1b46b4e80909291542k71440965g75bb3bfcf38e8005@mail.gmail.com>
	<1b46b4e80910011229k31f7233du10b0fc661730d3f3@mail.gmail.com>
	<F86F950D-9106-49C9-81CA-38B3EB2072F4@jerris.com>
	<1b46b4e80910041519h3581d3d2x7b84041e8c18588@mail.gmail.com>
	<C887D143-F7A1-44B9-B8A2-D34A4B4E582E@jerris.com>
	<1b46b4e80910041609x2f41ceb9u142cb484951a21ed@mail.gmail.com>
Message-ID: <87f2f3b90910050920w4d17033fl4c386a0018107bdf@mail.gmail.com>

On Sun, Oct 4, 2009 at 4:09 PM, Nicolas Brenner <nicolas at medularis.com>wrote:

> Mike, how exactly should I format the file? I got the pcap file, how do I
> convert it to text so that you can easily read it?
>
>
you can open it with wireshark, follow the TCP or UDP stream, then just copy
& paste the text as needed...

-MC
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From jerry.richards at teotech.com  Mon Oct  5 09:25:08 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 5 Oct 2009 09:25:08 -0700
Subject: [Freeswitch-users] FS Does Not
	RelayPresencePUBLISHToSubscribing Phones
In-Reply-To: <b39273530910021142p546680felc6c6ec19d86c922f@mail.gmail.com>
References: <B4B4D2747EEC4852B41A7D8D2DA36006@greyhawk.tonecommander.com><191c3a030910010935s69e65569v76f4d26503a94d96@mail.gmail.com><9B5214D274044021A57C6857844C10BA@greyhawk.tonecommander.com><ebbc2d550910012014ic655d84tbf000c084a98e51d@mail.gmail.com><E407DD09040D408F8A130C2BFAFFCD17@greyhawk.tonecommander.com><191c3a030910021052y2e074b4r648024507206e25@mail.gmail.com><57502048DB624686B96B8AE76F697AA7@greyhawk.tonecommander.com>
	<b39273530910021142p546680felc6c6ec19d86c922f@mail.gmail.com>
Message-ID: <6F7998CFEDEC4CDC83915045AF03000C@greyhawk.tonecommander.com>

Okay, I added the ";" at the end of the sqlite3 "select" command and it just
returned to the "sqlite>" prompt.  No error was returned.  Do you see
anything in my database (in the pastebin) that is incorrect?  By the way,
the "select" command I put in the pastebin refers to the "external" config,
but the "internal" config does the  same thing.

Best Regards,
Jerry


-----Original Message-----
From: Rupa Schomaker [mailto:rupa at rupa.com] 
Sent: Friday, October 02, 2009 11:42 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Does Not
RelayPresencePUBLISHToSubscribing Phones

You are missing the trailing ;

On Fri, Oct 2, 2009 at 1:28 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:
> I put the sqlite3 select query in the paste bin, and prior to that, I 
> entered the .dump command.? The select command came back with a "...>"
> prompt which I don't understand.? I don't know enough about sqlite3 to 
> know what that means?
>
> Best Regards,
> Jerry


--
-Rupa






From fraunhofer.lists.freeswitch-001 at traced.net  Mon Oct  5 10:39:39 2009
From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer)
Date: Mon, 5 Oct 2009 19:39:39 +0200
Subject: [Freeswitch-users] UUID of the newly originated call?
In-Reply-To: <25751228.post@talk.nabble.com>
References: <25751228.post@talk.nabble.com>
Message-ID: <a37510170910051039i1fdeb6dcm7cc31f47e88f0c62@mail.gmail.com>

Hi,

2009/10/5 Nagalenoj <nagalenoj at gmail.com>:
> ? ? * Receive the events only for this uuid - I have done by registering
> all events and filtering only for this uuid($uuid).
> ? ? * If it is CHANNEL_ANSWER, originate a new call.

it's a "filter in", not "filter out" :)

> Now, How can I get the uuid of the new call and receive events for this new
> call.? I want to receive the events for both uuids.

You can specify the UUID of an originated call by doing the following:

    * Use create_uuid to generate a UUID to use.
    * This will allow you to kill an originated call before it is
answered by using uuid_kill.
    * The UUID of the answered call leg will not be the same UUID as
the origination_uuid specified (Each call leg always gets its own
UUID)

    originate [origination_uuid=...]user/100 at domain.name.com

shamelessly ripped from
http://wiki.freeswitch.org/wiki/Mod_commands#originate

at least it worked for me.

  Beni.



From Russell.Mosemann at cune.org  Mon Oct  5 10:50:38 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Mon, 5 Oct 2009 17:50:38 -0000
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <87f2f3b90910050905g68ad8ec2xb7f834ba90f40d0b@mail.gmail.com>
Message-ID: <20091005175038.CD7D73F642F@mail.cune.org>

Michael Collins <msc at freeswitch.org> said:

> Confirmed: the Hicomm isn't sending anything at all in the SETUP message
> except the usual stuff: dialed number, channel number, etc. Does the Hicomm
> have any config parameters, like Caller ID presentation?

I believe it does, but I don't have access to the Hicom. I have to go
through the phone guy. It's kind of a delicate situation. At least I have
something to suggest for him to investigate. I appreciate the confirmation.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!




From xengelpublicx at gmail.com  Mon Oct  5 10:50:37 2009
From: xengelpublicx at gmail.com (Vladimir Elizarov)
Date: Mon, 05 Oct 2009 21:50:37 +0400
Subject: [Freeswitch-users] stun not working in fs 1.0.4?
In-Reply-To: <7C489261-F5F3-4B57-B807-21EC4A3947A2@freeswitch.org>
References: <4AC9E9EE.8090805@gmail.com>
	<7C489261-F5F3-4B57-B807-21EC4A3947A2@freeswitch.org>
Message-ID: <4ACA31ED.1000002@gmail.com>

Brian West ?????:
> Yes!
>   
Ok. Brian, why fs no two branches of the stable and trunk?
> /b
>
> On Oct 5, 2009, at 7:43 AM, Vladimir Elizarov wrote:
>
>   
>> How do I fix this? Upgrade fs to trunk?
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


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From jerry.richards at teotech.com  Mon Oct  5 10:58:10 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 5 Oct 2009 10:58:10 -0700
Subject: [Freeswitch-users] SLAs and BLAs
Message-ID: <5B5F50E0B1D34BC0BCBCC05E1FE05C8A@greyhawk.tonecommander.com>


I can see how BLFs and Presence are managed, however I haven't found much
documentation on SLAs and BLAs.  What is the RFC(s) that Freeswitch used to
implement SLAs and BLAs?  Do they differ from BLFs?

Best Regards,
Jerry




From brian at freeswitch.org  Mon Oct  5 10:59:33 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Oct 2009 12:59:33 -0500
Subject: [Freeswitch-users] stun not working in fs 1.0.4?
In-Reply-To: <4ACA31ED.1000002@gmail.com>
References: <4AC9E9EE.8090805@gmail.com>
	<7C489261-F5F3-4B57-B807-21EC4A3947A2@freeswitch.org>
	<4ACA31ED.1000002@gmail.com>
Message-ID: <4AA6F4E2-2138-439C-86E8-7554A5D110D4@freeswitch.org>

Because TRUNK is stable... its only fixes going in usually and if  
things do break they don't stay broken for long.

Ask anyone our trunk is more table then most commercial products.

/b

On Oct 5, 2009, at 12:50 PM, Vladimir Elizarov wrote:

> Ok. Brian, why fs no two branches of the stable and trunk?




From brian at freeswitch.org  Mon Oct  5 11:01:38 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Oct 2009 13:01:38 -0500
Subject: [Freeswitch-users] SLAs and BLAs
In-Reply-To: <5B5F50E0B1D34BC0BCBCC05E1FE05C8A@greyhawk.tonecommander.com>
References: <5B5F50E0B1D34BC0BCBCC05E1FE05C8A@greyhawk.tonecommander.com>
Message-ID: <3F7AC380-CACB-432F-8B94-80FBA18628C0@freeswitch.org>

First off what phones are you going to be using?

/b

On Oct 5, 2009, at 12:58 PM, Jerry Richards wrote:

>
> I can see how BLFs and Presence are managed, however I haven't found  
> much
> documentation on SLAs and BLAs.  What is the RFC(s) that Freeswitch  
> used to
> implement SLAs and BLAs?  Do they differ from BLFs?
>
> Best Regards,
> Jerry




From msc at freeswitch.org  Mon Oct  5 11:25:13 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 5 Oct 2009 11:25:13 -0700
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <20091005175038.CD7D73F642F@mail.cune.org>
References: <87f2f3b90910050905g68ad8ec2xb7f834ba90f40d0b@mail.gmail.com>
	<20091005175038.CD7D73F642F@mail.cune.org>
Message-ID: <87f2f3b90910051125p5932e254yc622b1d9454e0223@mail.gmail.com>

On Mon, Oct 5, 2009 at 10:50 AM, <Russell.Mosemann at cune.org> wrote:

> Michael Collins <msc at freeswitch.org> said:
>
> > Confirmed: the Hicomm isn't sending anything at all in the SETUP message
> > except the usual stuff: dialed number, channel number, etc. Does the
> Hicomm
> > have any config parameters, like Caller ID presentation?
>
> I believe it does, but I don't have access to the Hicom. I have to go
> through the phone guy. It's kind of a delicate situation. At least I have
> something to suggest for him to investigate. I appreciate the confirmation.
>
>
If you need proverbial ammo let me know. If he speaks Q931 then your
pastebin is the ultimate proof for him that the hicom is not sending any
caller ID info. In any case, I'm here if you need assistance.
-MC
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From diego.viola at gmail.com  Mon Oct  5 11:34:31 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Mon, 5 Oct 2009 18:34:31 +0000
Subject: [Freeswitch-users] Youtube - FreeSWITCH Promo Video
In-Reply-To: <191c3a030910050802v59b2a9du6d2af7ade03bc509@mail.gmail.com>
References: <9A87E77C-AD85-4F1E-8ED0-F9F06AA451EE@ken-ton.com>
	<86a32abc0910041651rb6a3ff9k1f6ed7d0b1a9b440@mail.gmail.com>
	<7b197bef0910050004v71c623c8w68a013bef9fea596@mail.gmail.com>
	<191c3a030910050802v59b2a9du6d2af7ade03bc509@mail.gmail.com>
Message-ID: <86a32abc0910051134mbb2c39fr57e29caeb27121a5@mail.gmail.com>

Nice script Anthony, that would be amazing to have on video ;)

On Mon, Oct 5, 2009 at 3:02 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> neat,
>
> Here's some suggestions for your next ones. =p
>
> Have them standing around the hologram trying to destroy the "Death
> Star(tm)" that happens to look a lot like a giant 3d unix '*' character.
> Then have one rebel say, "wait!, why are we wasting our time... watch
> this... and dial a number on his cellphone as the whole thing explodes in
> the background.
>
> Have Darth Forkium face Luke ThreadSpawner in a dual.  "I see you have
> fashioned your own TDM card...." vroom......
> "Join me and together we can make linked lists and monolithic processes",
> "NEVER!..." vroom vroom Master Coda has taught you well....."You are no
> match for me...JOIN THE ORANGE SIDE OF THE FORCE"
>
>
>
>
>
>
> On Mon, Oct 5, 2009 at 2:04 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:
>
>> On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola <diego.viola at gmail.com>
>> wrote:
>> > Very nice :)
>> >
>> > On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling <kjv at ken-ton.com>
>> wrote:
>> >>
>> >> Folks;
>> >> Here's something that I did playing around w/ learning Apple Motion.
>>
>> Me too: very nice!
>>
>> -gmaruzz
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Oct  5 12:03:27 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 5 Oct 2009 12:03:27 -0700
Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Followup,
	Agenda For This Week
Message-ID: <87f2f3b90910051203v7bdbca5amfc82d193f3a4f1a3@mail.gmail.com>

Ladies and Gentlemen,

Thank you for calling in to the weekly FreeSWITCH conference call. Last
week's agenda was rather light, so if you have things that you would like to
have discussed please be sure to add them here:

http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_09

Here are a few updates for everyone to keep in mind:
* Starting with the upcoming meeting (Oct 9) the conference will support
48kHz CELT codec.
* When you call in, please mute your phone if you are just listening or if
you will not be speaking a lot. Background noise from one user isn't usually
too bad, but when 10 or 15 people are not muted it can get a little
distracting. :)
* We are always looking for people to help out. If you are looking for ways
to help out then please by all means call the conference. If you cannot call
the conference for some reason but still want to help out, please email me
off list. We have numerous documentation, janitorial, code review, etc.
sub-projects that FreeSWITCH users can help with.
* Be sure to update to latest SVN trunk and test test test! We are prepping
for 1.0.5 and more people testing means a more stable release delivered more
quickly.

One topic that came up was the testing of Mike van Lammerman's Ultramonkey
setup. (http://bit.ly/8zZUS) Several people agreed to try it out. We would
love to see others try it out and report back their experiences.

Lastly, we'd like to thank everyone who has been helping out with testing,
documentation, and answering questions on the email list and IRC channels. A
special thanks to Diego Viola for doing lots of wiki cleanup lately.
Everyone's efforts are appreciated.

-Michael
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From Russell.Mosemann at cune.org  Mon Oct  5 12:15:52 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Mon, 5 Oct 2009 19:15:52 -0000
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <87f2f3b90910051125p5932e254yc622b1d9454e0223@mail.gmail.com>
Message-ID: <20091005191552.0D0093BA4C5@mail.cune.org>

Michael Collins <msc at freeswitch.org> said:

> If you need proverbial ammo let me know. If he speaks Q931 then your
> pastebin is the ultimate proof for him that the hicom is not sending any
> caller ID info. In any case, I'm here if you need assistance.

Heh, I have the exact opposite problem. I don't think he configured a PRI
T1 before, and the debug output would be meaningless to him. He usually
handles everyday "my phone doesn't work" kind of issues. I try to nudge
in the right direction without being pushy. :-)

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!




From brian at freeswitch.org  Mon Oct  5 12:30:10 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Oct 2009 14:30:10 -0500
Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Followup,
	Agenda For This Week
In-Reply-To: <87f2f3b90910051203v7bdbca5amfc82d193f3a4f1a3@mail.gmail.com>
References: <87f2f3b90910051203v7bdbca5amfc82d193f3a4f1a3@mail.gmail.com>
Message-ID: <B5ECAE34-B846-4408-8AC1-6D94ADCC672C@freeswitch.org>

It always supported 48kHz CELT but the conference itself was running  
at 32kHz so everyone 48k had to be down sampled.  Now you all get to  
be up sampled.  w00t!

/b

On Oct 5, 2009, at 2:03 PM, Michael Collins wrote:

> * Starting with the upcoming meeting (Oct 9) the conference will  
> support 48kHz CELT codec.




From dmitry.bely at gmail.com  Mon Oct  5 12:57:47 2009
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Mon, 5 Oct 2009 23:57:47 +0400
Subject: [Freeswitch-users] How to limit the number of incoming+outgoing
	calls via specific gateway?
In-Reply-To: <C1AD67AA-F969-4BA1-8415-5E381BDF18FA@jerris.com>
References: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>
	<e403cddca1c90c83ce4fb6f85b026e63.squirrel@correo.nodo50.org>
	<65d96fc80910020532i4ba33353g7ac141624a4bb89e@mail.gmail.com>
	<C1AD67AA-F969-4BA1-8415-5E381BDF18FA@jerris.com>
Message-ID: <90823c940910051257u403108afg3478c4c13ab26283@mail.gmail.com>

On Mon, Oct 5, 2009 at 2:30 AM, Michael Jerris <mike at jerris.com> wrote:
> http://wiki.freeswitch.org/wiki/Mod_limit

Of course I've read that. But it only affects the number of outgoing
calls (at least for gateways - chapter "Using mod_limit with an
outbound gateway"). But I would like to limit the number of all calls
(incoming+outgoing) via specific gateway. Any idea?

- Dmitry Bely



From sprice at gmail.com  Mon Oct  5 13:19:13 2009
From: sprice at gmail.com (SP)
Date: Mon, 5 Oct 2009 15:19:13 -0500
Subject: [Freeswitch-users] How to limit the number of incoming+outgoing
	calls via specific gateway?
In-Reply-To: <90823c940910051257u403108afg3478c4c13ab26283@mail.gmail.com>
References: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>
	<e403cddca1c90c83ce4fb6f85b026e63.squirrel@correo.nodo50.org>
	<65d96fc80910020532i4ba33353g7ac141624a4bb89e@mail.gmail.com>
	<C1AD67AA-F969-4BA1-8415-5E381BDF18FA@jerris.com>
	<90823c940910051257u403108afg3478c4c13ab26283@mail.gmail.com>
Message-ID: <7e2ac3270910051319xfdfc9d3wa583c2da7e932339@mail.gmail.com>

Direction doesn't matter, it uses realm's and a few other vars.  Use
the same vars for both directions.

On Mon, Oct 5, 2009 at 14:57, Dmitry Bely <dmitry.bely at gmail.com> wrote:
> On Mon, Oct 5, 2009 at 2:30 AM, Michael Jerris <mike at jerris.com> wrote:
>> http://wiki.freeswitch.org/wiki/Mod_limit
>
> Of course I've read that. But it only affects the number of outgoing
> calls (at least for gateways - chapter "Using mod_limit with an
> outbound gateway"). But I would like to limit the number of all calls
> (incoming+outgoing) via specific gateway. Any idea?
>
> - Dmitry Bely
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Shannon



From monemran at gmail.com  Mon Oct  5 13:32:50 2009
From: monemran at gmail.com (M.Emran)
Date: Tue, 6 Oct 2009 02:32:50 +0600
Subject: [Freeswitch-users] How to limit the number of incoming+outgoing
	calls via specific gateway?
In-Reply-To: <7e2ac3270910051319xfdfc9d3wa583c2da7e932339@mail.gmail.com>
References: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>
	<e403cddca1c90c83ce4fb6f85b026e63.squirrel@correo.nodo50.org>
	<65d96fc80910020532i4ba33353g7ac141624a4bb89e@mail.gmail.com>
	<C1AD67AA-F969-4BA1-8415-5E381BDF18FA@jerris.com>
	<90823c940910051257u403108afg3478c4c13ab26283@mail.gmail.com>
	<7e2ac3270910051319xfdfc9d3wa583c2da7e932339@mail.gmail.com>
Message-ID: <b95af1c0910051332p3113156aj89a1a7b2fc3ae22a@mail.gmail.com>

Incoming Call-Limit http://www.howtonix.com/?p=89

Outbound Call-Limit http://www.howtonix.com/?p=86



On Tue, Oct 6, 2009 at 2:19 AM, SP <sprice at gmail.com> wrote:

> Direction doesn't matter, it uses realm's and a few other vars.  Use
> the same vars for both directions.
>
> On Mon, Oct 5, 2009 at 14:57, Dmitry Bely <dmitry.bely at gmail.com> wrote:
> > On Mon, Oct 5, 2009 at 2:30 AM, Michael Jerris <mike at jerris.com> wrote:
> >> http://wiki.freeswitch.org/wiki/Mod_limit
> >
> > Of course I've read that. But it only affects the number of outgoing
> > calls (at least for gateways - chapter "Using mod_limit with an
> > outbound gateway"). But I would like to limit the number of all calls
> > (incoming+outgoing) via specific gateway. Any idea?
> >
> > - Dmitry Bely
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> Shannon
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Regards
----------
M Emran
Managing Director
E-SOFT BILLING PTE. LTD.
Web: www.e-softbilling.com
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From dmitry.bely at gmail.com  Mon Oct  5 13:39:56 2009
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Tue, 6 Oct 2009 00:39:56 +0400
Subject: [Freeswitch-users] How to limit the number of incoming+outgoing
	calls via specific gateway?
In-Reply-To: <7e2ac3270910051319xfdfc9d3wa583c2da7e932339@mail.gmail.com>
References: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>
	<e403cddca1c90c83ce4fb6f85b026e63.squirrel@correo.nodo50.org>
	<65d96fc80910020532i4ba33353g7ac141624a4bb89e@mail.gmail.com>
	<C1AD67AA-F969-4BA1-8415-5E381BDF18FA@jerris.com>
	<90823c940910051257u403108afg3478c4c13ab26283@mail.gmail.com>
	<7e2ac3270910051319xfdfc9d3wa583c2da7e932339@mail.gmail.com>
Message-ID: <90823c940910051339h3614be97o48a6319f68da7977@mail.gmail.com>

On Tue, Oct 6, 2009 at 12:19 AM, SP <sprice at gmail.com> wrote:
> Direction doesn't matter, it uses realm's and a few other vars. ?Use
> the same vars for both directions.

Unfortunately it does.

      <action application="limit" data="$${domain} my_provider 1"/>

generates limit_exceeded for the second outbound call, but if an
incoming call is active FreeSWITCH still tries to use this gateway.

> On Mon, Oct 5, 2009 at 14:57, Dmitry Bely <dmitry.bely at gmail.com> wrote:
>> On Mon, Oct 5, 2009 at 2:30 AM, Michael Jerris <mike at jerris.com> wrote:
>>> http://wiki.freeswitch.org/wiki/Mod_limit
>>
>> Of course I've read that. But it only affects the number of outgoing
>> calls (at least for gateways - chapter "Using mod_limit with an
>> outbound gateway"). But I would like to limit the number of all calls
>> (incoming+outgoing) via specific gateway. Any idea?
>>
>> - Dmitry Bely
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Shannon
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

 Dmitry Bely



From dmitry.bely at gmail.com  Mon Oct  5 13:41:49 2009
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Tue, 6 Oct 2009 00:41:49 +0400
Subject: [Freeswitch-users] How to limit the number of incoming+outgoing
	calls via specific gateway?
In-Reply-To: <b95af1c0910051332p3113156aj89a1a7b2fc3ae22a@mail.gmail.com>
References: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>
	<e403cddca1c90c83ce4fb6f85b026e63.squirrel@correo.nodo50.org>
	<65d96fc80910020532i4ba33353g7ac141624a4bb89e@mail.gmail.com>
	<C1AD67AA-F969-4BA1-8415-5E381BDF18FA@jerris.com>
	<90823c940910051257u403108afg3478c4c13ab26283@mail.gmail.com>
	<7e2ac3270910051319xfdfc9d3wa583c2da7e932339@mail.gmail.com>
	<b95af1c0910051332p3113156aj89a1a7b2fc3ae22a@mail.gmail.com>
Message-ID: <90823c940910051341w2060e9c3m59f2b59f20ac328d@mail.gmail.com>

On Tue, Oct 6, 2009 at 12:32 AM, M.Emran <monemran at gmail.com> wrote:
> Incoming Call-Limit http://www.howtonix.com/?p=89
>
> Outbound Call-Limit http://www.howtonix.com/?p=86

But what if I need to limit the total number of calls (in my case == 1)?

> On Tue, Oct 6, 2009 at 2:19 AM, SP <sprice at gmail.com> wrote:
>>
>> Direction doesn't matter, it uses realm's and a few other vars. ?Use
>> the same vars for both directions.
>>
>> On Mon, Oct 5, 2009 at 14:57, Dmitry Bely <dmitry.bely at gmail.com> wrote:
>> > On Mon, Oct 5, 2009 at 2:30 AM, Michael Jerris <mike at jerris.com> wrote:
>> >> http://wiki.freeswitch.org/wiki/Mod_limit
>> >
>> > Of course I've read that. But it only affects the number of outgoing
>> > calls (at least for gateways - chapter "Using mod_limit with an
>> > outbound gateway"). But I would like to limit the number of all calls
>> > (incoming+outgoing) via specific gateway. Any idea?

- Dmitry Bely



From sprice at gmail.com  Mon Oct  5 13:52:51 2009
From: sprice at gmail.com (SP)
Date: Mon, 5 Oct 2009 15:52:51 -0500
Subject: [Freeswitch-users] How to limit the number of incoming+outgoing
	calls via specific gateway?
In-Reply-To: <90823c940910051339h3614be97o48a6319f68da7977@mail.gmail.com>
References: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>
	<e403cddca1c90c83ce4fb6f85b026e63.squirrel@correo.nodo50.org>
	<65d96fc80910020532i4ba33353g7ac141624a4bb89e@mail.gmail.com>
	<C1AD67AA-F969-4BA1-8415-5E381BDF18FA@jerris.com>
	<90823c940910051257u403108afg3478c4c13ab26283@mail.gmail.com>
	<7e2ac3270910051319xfdfc9d3wa583c2da7e932339@mail.gmail.com>
	<90823c940910051339h3614be97o48a6319f68da7977@mail.gmail.com>
Message-ID: <7e2ac3270910051352p7d052a5aife6f07827173767@mail.gmail.com>

did you use the application limit on the inbound call?  You'll need to
in order to account for it.

On Mon, Oct 5, 2009 at 15:39, Dmitry Bely <dmitry.bely at gmail.com> wrote:
> On Tue, Oct 6, 2009 at 12:19 AM, SP <sprice at gmail.com> wrote:
>> Direction doesn't matter, it uses realm's and a few other vars. ?Use
>> the same vars for both directions.
>
> Unfortunately it does.
>
> ? ? ?<action application="limit" data="$${domain} my_provider 1"/>
>
> generates limit_exceeded for the second outbound call, but if an
> incoming call is active FreeSWITCH still tries to use this gateway.
>
>> On Mon, Oct 5, 2009 at 14:57, Dmitry Bely <dmitry.bely at gmail.com> wrote:
>>> On Mon, Oct 5, 2009 at 2:30 AM, Michael Jerris <mike at jerris.com> wrote:
>>>> http://wiki.freeswitch.org/wiki/Mod_limit
>>>
>>> Of course I've read that. But it only affects the number of outgoing
>>> calls (at least for gateways - chapter "Using mod_limit with an
>>> outbound gateway"). But I would like to limit the number of all calls
>>> (incoming+outgoing) via specific gateway. Any idea?
>>>
>>> - Dmitry Bely
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Shannon
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> ?Dmitry Bely
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Shannon



From msc at freeswitch.org  Mon Oct  5 14:01:14 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 5 Oct 2009 14:01:14 -0700
Subject: [Freeswitch-users] Connecting FS to Hicom 300
In-Reply-To: <20091005191552.0D0093BA4C5@mail.cune.org>
References: <87f2f3b90910051125p5932e254yc622b1d9454e0223@mail.gmail.com>
	<20091005191552.0D0093BA4C5@mail.cune.org>
Message-ID: <87f2f3b90910051401y31a1faf0v91b6a1b2443f519c@mail.gmail.com>

On Mon, Oct 5, 2009 at 12:15 PM, <Russell.Mosemann at cune.org> wrote:

> Michael Collins <msc at freeswitch.org> said:
>
> > If you need proverbial ammo let me know. If he speaks Q931 then your
> > pastebin is the ultimate proof for him that the hicom is not sending any
> > caller ID info. In any case, I'm here if you need assistance.
>
> Heh, I have the exact opposite problem. I don't think he configured a PRI
> T1 before, and the debug output would be meaningless to him. He usually
> handles everyday "my phone doesn't work" kind of issues. I try to nudge
> in the right direction without being pushy. :-)
>
> Haha, good luck w/ that. :P
-MC
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From ChristianDamianidis at globalive.com  Mon Oct  5 11:25:03 2009
From: ChristianDamianidis at globalive.com (Christian Damianidis)
Date: Mon, 5 Oct 2009 14:25:03 -0400
Subject: [Freeswitch-users] mod_xml_curl http POST is inconsistent/bug
Message-ID: <66EA3166EB339A4489B06286C0876A8A0C0C1432@mailserv.Globalive.local>

Trying to achieve dynamic binding of user directory. In short: It's not
doing the authorization properly. I can use curl in the command line and
it works perfectly, specifying BASIC auth.. however with the freeswitch
module it returns HTTP 401. 

 

So I've taken a close look at the network packets being sent and there
are some issues.

 

This is between the <bindings></bindings> tags in my xml_curl.conf.xml
(1.2.3.4 represents my webserver's IP)

 

        <binding name="users">

                <param name="gateway-url"
value="http://1.2.3.4:2000/users.aspx" bindings="directory"/>

                <param name="gateway-credentials"
value="username:password"/>

        </binding>

 

When I run "curl -basic -u username http://1.2.3.4:2000/users.aspx" it
asks me for a password and returns the correct thing. I use tshark to
monitor, and it sends a GET request, with the correct authorization
credentials in the header. I receive an HTTP 200 OK packet and the xml
follows. 

 

When I startup freeswitch, I guess the xml curl module gets to run, and
it makes the request. However this time it's a POST, and oddly DOES NOT
include the Authorization: Basic <encodedstring> line in the packet. I
get back two HTTP 401 Unauthorized responses, and then freeswitch sends
out another POST, this time includes the authorization line, and I get
back an OK with the xml.

My user directory is updated and we're all good.

 

The inconsistent POST request sent by the module causes freeswitch to
hang for 1-2 minutes during start-up.

 

Has anyone else had this issue? Is this a bug or intended functionality
(ping the server before making a real request?). I'd love to sort this
out, otherwise getting an updated directory isn't real-time, thus
defeating the purpose.

Thanks,

 

Christian

 

 

 

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From vdc1048 at tx.rr.com  Mon Oct  5 13:49:58 2009
From: vdc1048 at tx.rr.com (David Clark)
Date: Mon, 05 Oct 2009 15:49:58 -0500
Subject: [Freeswitch-users] Basic compile question.
In-Reply-To: <7e2ac3270910051319xfdfc9d3wa583c2da7e932339@mail.gmail.com
 >
References: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>
	<e403cddca1c90c83ce4fb6f85b026e63.squirrel@correo.nodo50.org>
	<65d96fc80910020532i4ba33353g7ac141624a4bb89e@mail.gmail.com>
	<C1AD67AA-F969-4BA1-8415-5E381BDF18FA@jerris.com>
	<90823c940910051257u403108afg3478c4c13ab26283@mail.gmail.com>
	<7e2ac3270910051319xfdfc9d3wa583c2da7e932339@mail.gmail.com>
Message-ID: <6.2.3.4.2.20091005154158.02b632a8@pop-server.tx.rr.com>

Ok using windows xp x64 here.  I download from the trunk as expected.
I fire up VS 2005 and I open the VS 2005 solution.  Yes it does say 
unsupported.

But I get two missing header files:
freeswitch\src\mod\endpoints\mod_opal\mod_opal.h(33) : fatal error 
C1083: Cannot open include file: 'ptlib.h': No such file or directory
.\mod_voipcodecs.c(36) : fatal error C1083: Cannot open include file: 
'spandsp.h': No such file or directory

Even if the project file is wrong or out of date I should be able to 
find the include files some place in the fileset.
I can't find either file in the freeswitch directory or below it.

Any idea what is up?

Thanks,
David Clark




From vdc1048 at tx.rr.com  Mon Oct  5 14:14:08 2009
From: vdc1048 at tx.rr.com (David Clark)
Date: Mon, 05 Oct 2009 16:14:08 -0500
Subject: [Freeswitch-users] UPDATED: Basic compile question.
Message-ID: <6.2.3.4.2.20091005161228.02b6bcc8@pop-server.tx.rr.com>

Ok I found spandsp.h.  It is a case of the project file being out of 
date.  No surprise.  ptlib.h is still not found.

------------------------------------------------------------------------------------------------------------------------------------------------------------
Ok using windows xp x64 here.  I download from the trunk as expected.
I fire up VS 2005 and I open the VS 2005 solution.  Yes it does say 
unsupported.

But I get two missing header files:
freeswitch\src\mod\endpoints\mod_opal\mod_opal.h(33) : fatal error 
C1083: Cannot open include file: 'ptlib.h': No such file or directory
.\mod_voipcodecs.c(36) : fatal error C1083: Cannot open include file: 
'spandsp.h': No such file or directory


Even if the project file is wrong or out of date I should be able to 
find the include files some place in the fileset.
I can't find either file in the freeswitch directory or below it.

Any idea what is up?

Thanks,
David Clark




From brian at freeswitch.org  Mon Oct  5 14:41:39 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Oct 2009 16:41:39 -0500
Subject: [Freeswitch-users] Basic compile question.
In-Reply-To: <6.2.3.4.2.20091005154158.02b632a8@pop-server.tx.rr.com>
References: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>
	<e403cddca1c90c83ce4fb6f85b026e63.squirrel@correo.nodo50.org>
	<65d96fc80910020532i4ba33353g7ac141624a4bb89e@mail.gmail.com>
	<C1AD67AA-F969-4BA1-8415-5E381BDF18FA@jerris.com>
	<90823c940910051257u403108afg3478c4c13ab26283@mail.gmail.com>
	<7e2ac3270910051319xfdfc9d3wa583c2da7e932339@mail.gmail.com>
	<6.2.3.4.2.20091005154158.02b632a8@pop-server.tx.rr.com>
Message-ID: <4C985F33-8B02-4F8F-95B5-36BA6B40690F@freeswitch.org>

Have you updated today?

/b

On Oct 5, 2009, at 3:49 PM, David Clark wrote:

> Any idea what is up?




From brian at freeswitch.org  Mon Oct  5 14:42:36 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Oct 2009 16:42:36 -0500
Subject: [Freeswitch-users] mod_xml_curl http POST is inconsistent/bug
In-Reply-To: <66EA3166EB339A4489B06286C0876A8A0C0C1432@mailserv.Globalive.local>
References: <66EA3166EB339A4489B06286C0876A8A0C0C1432@mailserv.Globalive.local>
Message-ID: <276FEE28-70CF-422D-9F6F-7CF747FDDB22@freeswitch.org>

Are you using something other than apache?

/b

On Oct 5, 2009, at 1:25 PM, Christian Damianidis wrote:

>
> The inconsistent POST request sent by the module causes freeswitch  
> to hang for 1-2 minutes during start-up.

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From mike at jerris.com  Mon Oct  5 14:58:51 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 5 Oct 2009 17:58:51 -0400
Subject: [Freeswitch-users] UPDATED: Basic compile question.
In-Reply-To: <6.2.3.4.2.20091005161228.02b6bcc8@pop-server.tx.rr.com>
References: <6.2.3.4.2.20091005161228.02b6bcc8@pop-server.tx.rr.com>
Message-ID: <8EA2BE7E-128C-4D56-A761-C2912ED097E0@jerris.com>

voip codecs is fixed, ptlib I can't recall if we ever did full build  
integration or if you needed to manually download the libraries, can  
someone who has done mod_opal build on windows comment?

Mike


On Oct 5, 2009, at 5:14 PM, David Clark wrote:

> Ok I found spandsp.h.  It is a case of the project file being out of
> date.  No surprise.  ptlib.h is still not found.
>
> ------------------------------------------------------------------------------------------------------------------------------------------------------------
> Ok using windows xp x64 here.  I download from the trunk as expected.
> I fire up VS 2005 and I open the VS 2005 solution.  Yes it does say
> unsupported.
>
> But I get two missing header files:
> freeswitch\src\mod\endpoints\mod_opal\mod_opal.h(33) : fatal error
> C1083: Cannot open include file: 'ptlib.h': No such file or directory
> .\mod_voipcodecs.c(36) : fatal error C1083: Cannot open include file:
> 'spandsp.h': No such file or directory
>
>
> Even if the project file is wrong or out of date I should be able to
> find the include files some place in the fileset.
> I can't find either file in the freeswitch directory or below it.




From jaybinks at gmail.com  Mon Oct  5 15:16:50 2009
From: jaybinks at gmail.com (Jay Binks)
Date: Tue, 6 Oct 2009 08:16:50 +1000
Subject: [Freeswitch-users] Youtube - FreeSWITCH Promo Video
In-Reply-To: <191c3a030910050802v59b2a9du6d2af7ade03bc509@mail.gmail.com>
References: <9A87E77C-AD85-4F1E-8ED0-F9F06AA451EE@ken-ton.com>
	<86a32abc0910041651rb6a3ff9k1f6ed7d0b1a9b440@mail.gmail.com>
	<7b197bef0910050004v71c623c8w68a013bef9fea596@mail.gmail.com>
	<191c3a030910050802v59b2a9du6d2af7ade03bc509@mail.gmail.com>
Message-ID: <73572B80-F192-48CE-A6BA-F829AF8F788B@gmail.com>

Haha classic !!!

Can't wait for the next installment in the series !!

J


On 06/10/2009, at 1:02, Anthony Minessale  
<anthony.minessale at gmail.com> wrote:

> neat,
>
> Here's some suggestions for your next ones. =p
>
> Have them standing around the hologram trying to destroy the "Death  
> Star(tm)" that happens to look a lot like a giant 3d unix '*'  
> character.  Then have one rebel say, "wait!, why are we wasting our  
> time... watch this... and dial a number on his cellphone as the  
> whole thing explodes in the background.
>
> Have Darth Forkium face Luke ThreadSpawner in a dual.  "I see you  
> have fashioned your own TDM card...." vroom......
> "Join me and together we can make linked lists and monolithic  
> processes", "NEVER!..." vroom vroom Master Coda has taught you  
> well....."You are no match for me...JOIN THE ORANGE SIDE OF THE FORCE"
>
>
>
>
>
> On Mon, Oct 5, 2009 at 2:04 AM, Giovanni Maruzzelli <gmaruzz at celliax.org 
> > wrote:
> On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola <diego.viola at gmail.com>  
> wrote:
> > Very nice :)
> >
> > On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling <kjv at ken-ton.com>  
> wrote:
> >>
> >> Folks;
> >> Here's something that I did playing around w/ learning Apple  
> Motion.
>
> Me too: very nice!
>
> -gmaruzz
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From jerry.richards at teotech.com  Mon Oct  5 15:24:07 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 5 Oct 2009 15:24:07 -0700
Subject: [Freeswitch-users] SLAs and BLAs
In-Reply-To: <3F7AC380-CACB-432F-8B94-80FBA18628C0@freeswitch.org>
References: <5B5F50E0B1D34BC0BCBCC05E1FE05C8A@greyhawk.tonecommander.com>
	<3F7AC380-CACB-432F-8B94-80FBA18628C0@freeswitch.org>
Message-ID: <910A6955DAC94B23A0CB430A4A863E33@greyhawk.tonecommander.com>

We are building our own in-house developed Teo phones.  I also have
CounterPath's Bria Professional phone.  For test purposes, I have one snom
phone and a couple Polycomm phones.

Jerry
 

-----Original Message-----
From: Brian West [mailto:brian at freeswitch.org] 
Sent: Monday, October 05, 2009 11:02 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] SLAs and BLAs

First off what phones are you going to be using?

/b

On Oct 5, 2009, at 12:58 PM, Jerry Richards wrote:

>
> I can see how BLFs and Presence are managed, however I haven't found 
> much documentation on SLAs and BLAs.  What is the RFC(s) that 
> Freeswitch used to implement SLAs and BLAs?  Do they differ from BLFs?
>
> Best Regards,
> Jerry







From msc at freeswitch.org  Mon Oct  5 15:27:41 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 5 Oct 2009 15:27:41 -0700
Subject: [Freeswitch-users] Wiki Info: Multi Homed Tutorial
Message-ID: <87f2f3b90910051527x638f55b8hda75d5302267c251@mail.gmail.com>

FYI,

For those who've been following the thread about Michael Gende's tutorial I
just wanted to let you know that I his document on the wiki. It can be found
here:

http://wiki.freeswitch.org/wiki/Multi_home_tutorial

Please feel free to get in there and try it out, make editorial changes,
etc. I'm sure you won't hurt Mike's feelings by adding your thoughts. :)

If you have any questions please reply to this thread and we'll take it from
there.

Thanks,
MC
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From gmaruzz at celliax.org  Mon Oct  5 15:35:07 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 6 Oct 2009 00:35:07 +0200
Subject: [Freeswitch-users] Youtube - FreeSWITCH Promo Video
In-Reply-To: <73572B80-F192-48CE-A6BA-F829AF8F788B@gmail.com>
References: <9A87E77C-AD85-4F1E-8ED0-F9F06AA451EE@ken-ton.com> 
	<86a32abc0910041651rb6a3ff9k1f6ed7d0b1a9b440@mail.gmail.com> 
	<7b197bef0910050004v71c623c8w68a013bef9fea596@mail.gmail.com> 
	<191c3a030910050802v59b2a9du6d2af7ade03bc509@mail.gmail.com> 
	<73572B80-F192-48CE-A6BA-F829AF8F788B@gmail.com>
Message-ID: <7b197bef0910051535w4f1a888fj6bccdfbb240c7bcd@mail.gmail.com>

"The Revenge of the Sip"

On Tue, Oct 6, 2009 at 12:16 AM, Jay Binks <jaybinks at gmail.com> wrote:
> Haha classic !!!
> Can't wait for the next installment in the series !!
>
> J
>
>
> On 06/10/2009, at 1:02, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
>
> neat,
>
> Here's some suggestions for your next ones. =p
>
> Have them standing around the hologram trying to destroy the "Death
> Star(tm)" that happens to look a lot like a giant 3d unix '*' character.
> Then have one rebel say, "wait!, why are we wasting our time... watch
> this... and dial a number on his cellphone as the whole thing explodes in
> the background.
>
> Have Darth Forkium face Luke ThreadSpawner in a dual.? "I see you have
> fashioned your own TDM card...." vroom......
> "Join me and together we can make linked lists and monolithic processes",
> "NEVER!..." vroom vroom Master Coda has taught you well....."You are no
> match for me...JOIN THE ORANGE SIDE OF THE FORCE"
>
>
>
>
>
> On Mon, Oct 5, 2009 at 2:04 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>
> wrote:
>>
>> On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola <diego.viola at gmail.com> wrote:
>> > Very nice :)
>> >
>> > On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling <kjv at ken-ton.com>
>> > wrote:
>> >>
>> >> Folks;
>> >> Here's something that I did playing around w/ learning Apple Motion.
>>
>> Me too: very nice!
>>
>> -gmaruzz
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From kjv at ken-ton.com  Mon Oct  5 16:19:52 2009
From: kjv at ken-ton.com (Karl Vesterling)
Date: Mon, 5 Oct 2009 19:19:52 -0400
Subject: [Freeswitch-users] Youtube - FreeSWITCH Promo Video
In-Reply-To: <86a32abc0910051134mbb2c39fr57e29caeb27121a5@mail.gmail.com>
References: <9A87E77C-AD85-4F1E-8ED0-F9F06AA451EE@ken-ton.com>
	<86a32abc0910041651rb6a3ff9k1f6ed7d0b1a9b440@mail.gmail.com>
	<7b197bef0910050004v71c623c8w68a013bef9fea596@mail.gmail.com>
	<191c3a030910050802v59b2a9du6d2af7ade03bc509@mail.gmail.com>
	<86a32abc0910051134mbb2c39fr57e29caeb27121a5@mail.gmail.com>
Message-ID: <E9AC4ED5-CD35-4770-A8E1-8D0DC01C1D38@ken-ton.com>


Sadly the budget of time and props can't afford such extravagance...
But OMG I'm still laughing...

I was e-mailing earlier off the list, and came up with some nice  
"names" that could be put in the "credits"...

Like:

Anthony Minnessale -as- Obi-Code-Kenobi  (But I do like "Master Coda"  
from below.)

Mike Jerris -as- Luke Skypewalker

Richard Stallman -as- cpp30

Stuff like that...

I corrected to "speech" vs "speach" (my bad, sorry...)

Now, as far as the below, I imagine we could pull that off using the  
REAL footage, and dubbing in the audio if someone can do a decent  
Darth Spencer, errr Vader voice...  But something tells me that Mr.  
Lucas might get a bit peeved at such a thing...

Best Regards,
Karl J. Vesterling
kjv at ken-ton.com
202-461-3231 x0

On Oct 5, 2009, at 2:34 PM, Diego Viola wrote:

> Nice script Anthony, that would be amazing to have on video ;)
>
> On Mon, Oct 5, 2009 at 3:02 PM, Anthony Minessale <anthony.minessale at gmail.com 
> > wrote:
> neat,
>
> Here's some suggestions for your next ones. =p
>
> Have them standing around the hologram trying to destroy the "Death  
> Star(tm)" that happens to look a lot like a giant 3d unix '*'  
> character.  Then have one rebel say, "wait!, why are we wasting our  
> time... watch this... and dial a number on his cellphone as the  
> whole thing explodes in the background.
>
> Have Darth Forkium face Luke ThreadSpawner in a dual.  "I see you  
> have fashioned your own TDM card...." vroom......
> "Join me and together we can make linked lists and monolithic  
> processes", "NEVER!..." vroom vroom Master Coda has taught you  
> well....."You are no match for me...JOIN THE ORANGE SIDE OF THE FORCE"
>
>
>
>
>
>
> On Mon, Oct 5, 2009 at 2:04 AM, Giovanni Maruzzelli <gmaruzz at celliax.org 
> > wrote:
> On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola <diego.viola at gmail.com>  
> wrote:
> > Very nice :)
> >
> > On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling <kjv at ken-ton.com>  
> wrote:
> >>
> >> Folks;
> >> Here's something that I did playing around w/ learning Apple  
> Motion.
>
> Me too: very nice!
>
> -gmaruzz
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From kjv at ken-ton.com  Mon Oct  5 16:45:17 2009
From: kjv at ken-ton.com (Karl Vesterling)
Date: Mon, 5 Oct 2009 19:45:17 -0400
Subject: [Freeswitch-users] Youtube - FreeSWITCH Promo Video
In-Reply-To: <7b197bef0910051535w4f1a888fj6bccdfbb240c7bcd@mail.gmail.com>
References: <9A87E77C-AD85-4F1E-8ED0-F9F06AA451EE@ken-ton.com>
	<86a32abc0910041651rb6a3ff9k1f6ed7d0b1a9b440@mail.gmail.com>
	<7b197bef0910050004v71c623c8w68a013bef9fea596@mail.gmail.com>
	<191c3a030910050802v59b2a9du6d2af7ade03bc509@mail.gmail.com>
	<73572B80-F192-48CE-A6BA-F829AF8F788B@gmail.com>
	<7b197bef0910051535w4f1a888fj6bccdfbb240c7bcd@mail.gmail.com>
Message-ID: <7053A8C1-0EB3-447E-BFF7-C975E8087ADD@ken-ton.com>

NOW THAT might be worth doing!


Best Regards,
Karl J. Vesterling
kjv at ken-ton.com
202-461-3231 x0

On Oct 5, 2009, at 6:35 PM, Giovanni Maruzzelli wrote:

> "The Revenge of the Sip"
>
> On Tue, Oct 6, 2009 at 12:16 AM, Jay Binks <jaybinks at gmail.com> wrote:
>> Haha classic !!!
>> Can't wait for the next installment in the series !!
>>
>> J
>>
>>
>> On 06/10/2009, at 1:02, Anthony Minessale <anthony.minessale at gmail.com 
>> >
>> wrote:
>>
>> neat,
>>
>> Here's some suggestions for your next ones. =p
>>
>> Have them standing around the hologram trying to destroy the "Death
>> Star(tm)" that happens to look a lot like a giant 3d unix '*'  
>> character.
>> Then have one rebel say, "wait!, why are we wasting our time... watch
>> this... and dial a number on his cellphone as the whole thing  
>> explodes in
>> the background.
>>
>> Have Darth Forkium face Luke ThreadSpawner in a dual.  "I see you  
>> have
>> fashioned your own TDM card...." vroom......
>> "Join me and together we can make linked lists and monolithic  
>> processes",
>> "NEVER!..." vroom vroom Master Coda has taught you well....."You  
>> are no
>> match for me...JOIN THE ORANGE SIDE OF THE FORCE"
>>
>>
>>
>>
>>
>> On Mon, Oct 5, 2009 at 2:04 AM, Giovanni Maruzzelli <gmaruzz at celliax.org 
>> >
>> wrote:
>>>
>>> On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola  
>>> <diego.viola at gmail.com> wrote:
>>>> Very nice :)
>>>>
>>>> On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling <kjv at ken-ton.com>
>>>> wrote:
>>>>>
>>>>> Folks;
>>>>> Here's something that I did playing around w/ learning Apple  
>>>>> Motion.
>>>
>>> Me too: very nice!
>>>
>>> -gmaruzz
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>>> users
>>> http://www.freeswitch.org
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>>
>
>
>
> -- 
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>




From lakindia89 at gmail.com  Mon Oct  5 20:49:33 2009
From: lakindia89 at gmail.com (lakshmanan)
Date: Mon, 5 Oct 2009 20:49:33 -0700 (PDT)
Subject: [Freeswitch-users] oz debug says error
In-Reply-To: <20091005124858.81857415806@mail.cune.org>
References: <7d79b3930910050020i1b96541dk26c8d72c220ded8d@mail.gmail.com>
	<9F7D1E871F0C4184AA8ED441CF415C8A@cune.pri>
	<25749736.post@talk.nabble.com>
	<20091005124858.81857415806@mail.cune.org>
Message-ID: <25762469.post@talk.nabble.com>


Openzap.conf.xml
<configuration name="openzap.conf" description="OpenZAP Configuration">
        <settings>
                
        </settings>
        <libpri_spans>
                
                        
                        
                        
                        
                
        </libpri_spans>
</configuration>

Output of oz list in fs_cli

span: 1 (PRI_1)
type: isdn
chan_count: 47
dialplan: XML
context: default
dial_regex:
fail_dial_regex:
hold_music:
analog_options none

freeswitch.log

http://pastebin.freeswitch.org/10604


Russell.Mosemann wrote:
> 
> lakshmanan <lakindia89 at gmail.com> said:
>> Thanks for pointing that.
>> I also tried that.
>> But in that case, I'm not able to make a call through openzap.
> 
> What is in openzap.conf.xml? If you start fs_cli and enter "oz list",
> what does it show? Copy the ozmod lines from freeswitch.log to
> pastebin.freeswitch.org and post the link here so that we can see what
> openzap does when freeswitch starts.
> 
> -- 
> Russell Mosemann
> 
> 
> 
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/oz-debug-says-error-tp25746215p25762469.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From lakindia89 at gmail.com  Mon Oct  5 20:51:08 2009
From: lakindia89 at gmail.com (lakshmanan)
Date: Mon, 5 Oct 2009 20:51:08 -0700 (PDT)
Subject: [Freeswitch-users] oz debug says error
Message-ID: <25762469.post@talk.nabble.com>


Openzap.conf.xml

<configuration name="openzap.conf" description="OpenZAP Configuration">
        <settings>
                
        </settings>
        <libpri_spans>
                
                        
                        
                        
                        
                
        </libpri_spans>
</configuration>

Output of oz list in fs_cli

span: 1 (PRI_1)
type: isdn
chan_count: 47
dialplan: XML
context: default
dial_regex:
fail_dial_regex:
hold_music:
analog_options none

freeswitch.log

http://pastebin.freeswitch.org/10604


Russell.Mosemann wrote:
> 
> lakshmanan <lakindia89 at gmail.com> said:
>> Thanks for pointing that.
>> I also tried that.
>> But in that case, I'm not able to make a call through openzap.
> 
> What is in openzap.conf.xml? If you start fs_cli and enter "oz list",
> what does it show? Copy the ozmod lines from freeswitch.log to
> pastebin.freeswitch.org and post the link here so that we can see what
> openzap does when freeswitch starts.
> 
> -- 
> Russell Mosemann
> 
> 
> 
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/oz-debug-says-error-tp25746215p25762469.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From lakindia89 at gmail.com  Mon Oct  5 21:06:36 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Tue, 6 Oct 2009 09:36:36 +0530
Subject: [Freeswitch-users] oz debug says error
In-Reply-To: <20091005124858.81857415806@mail.cune.org>
References: <25749736.post@talk.nabble.com>
	<20091005124858.81857415806@mail.cune.org>
Message-ID: <7d79b3930910052106n59cf9cbema68e4d6ccc274034@mail.gmail.com>

Openzap.conf.xml
<configuration name="openzap.conf" description="OpenZAP Configuration">
        <settings>
                <param name="debug" value="7"/>
        </settings>
        <libpri_spans>
                <span name="PRI_1">
                        <param name="node" value="cpe"/>
                        <param name="switch" value="euroisdn"/>
                        <param name="dialplan" value="XML"/>
                        <param name="context" value="default"/>
                </span>
        </libpri_spans>
</configuration>

Output of oz list in fs_cli

span: 1 (PRI_1)
type: isdn
chan_count: 47
dialplan: XML
context: default
dial_regex:
fail_dial_regex:
hold_music:
analog_options none


freeswitch.log

http://pastebin.freeswitch.org/10604





On Mon, Oct 5, 2009 at 6:18 PM, <Russell.Mosemann at cune.org> wrote:

> lakshmanan <lakindia89 at gmail.com> said:
> > Thanks for pointing that.
> > I also tried that.
> > But in that case, I'm not able to make a call through openzap.
>
> What is in openzap.conf.xml? If you start fs_cli and enter "oz list",
> what does it show? Copy the ozmod lines from freeswitch.log to
> pastebin.freeswitch.org and post the link here so that we can see what
> openzap does when freeswitch starts.
>
> --
> Russell Mosemann
>
>
>
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mcampbellsmith at gmail.com  Mon Oct  5 21:15:21 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 6 Oct 2009 15:15:21 +1100
Subject: [Freeswitch-users] Detecting a fax
In-Reply-To: <65d96fc80910050732g2f414dffs45375cfb16b08c39@mail.gmail.com>
References: <33c87fa30910040603p7929df26l88f51fc7d256408f@mail.gmail.com>
	<91832961-FBA6-465F-AFD8-6A1BE604F6BC@jerris.com>
	<33c87fa30910050328l3d119939pd638b44b219a0140@mail.gmail.com>
	<402585BB-EBEA-42B8-9A2A-58FBA19B066F@jerris.com>
	<65d96fc80910050732g2f414dffs45375cfb16b08c39@mail.gmail.com>
Message-ID: <33c87fa30910052115m268b6e48q33d90946585334f9@mail.gmail.com>

Thanks for your help Mike and Tihomir.

A little more playing around and I found that having <action
application="fax_detect"/> as well as <action
application="tone_detect" data="fax 1100 r +5000 transfer fax XML
features" /> do not work together.

Simply by removing fax_detect, the fax is detected beautifully.

My problem now is trying to email the fax.  I followed the
instructions on the wiki at http://wiki.freeswitch.org/wiki/Mod_fax,
but the dialplan <action application="system" .../> is not executed
after the rxfax command.  I know the script works because if I put the
system command in another part of the dialplan and hard code the
filename to attach, then the email is sent.

    <extension name="receivefax">
      <condition field="destination_number" expression="^receivefax$">
        <action application="answer" />
        <action application="playback" data="silence_stream://2000"/>
        <action application="rxfax"
data="//usr//local//freeswitch//storage//${caller_id_number}-${uuid}.rxfax.tiff"/>
        <action application="system"
data="/usr/local/freeswitch/scripts/emailfax.sh
/usr/local/freeswitch/storage/${caller_id_number}-${uuid}.rxfax.tiff"/>
        <action application="hangup"/>
      </condition>
    </extension>

ideas?
Thanks!

On Tue, Oct 6, 2009 at 1:32 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
> hi Mark,
>
> This is an inbound call leg and media channel (so far)? is open in reverse
> direction only (application ringback). I'm afraid you have to answer the
> call to be able to "hear" the fax tone.
>
> T.
>
>
>
> On Mon, Oct 5, 2009 at 2:32 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>> Fax tones are not played by the remote machine until after answer, the
>> tone_detect application starts a media bug that listens for the tone,
>> can you confirm the tone is happening at all. ?Maybe the issue here is
>> the timeout, try making that longer, or doing the tone_detect in
>> execute_on_answer
>>
>> Mike
>>
>> On Oct 5, 2009, at 6:28 AM, Mark Campbell-Smith wrote:
>>
>> > Thanks for the response Mike,
>> >
>> > I read that page and this one (among others)
>> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_fax_detect, but
>> > I'm still lost. ?This is an extract of my dialplan
>> >
>> > ? ?<extension name="Local">
>> > ? ? ?<condition field="destination_number" expression="^(10[01][0-9])
>> > $">
>> > ? ? ? ?<action application="set" data="dialed_extension=$1"/>
>> > ? ? ? ?<action application="export" data="dialed_extension=$1"/>
>> > ? ? ? ?<action application="set" data="ringback=${au-ring}"/>
>> > ? ? ? ?<action application="fax_detect"/>
>> > ? ? ? ?<action application="tone_detect" data="fax 1100 r +5000
>> > transfer fax XML features" />
>> > ? ? ? ?<action application="set" data="hangup_after_bridge=true"/>
>> > ? ? ? ?<action application="set" data="continue_on_fail=true"/>
>> > ? ? ? ?<action application="bridge" data="user/${dialed_extension}@$
>> > {domain}"/>
>> >
>> > I would assume that on detecting a fax, the dialplan 'fax' is called
>> > in context features. ?This never happens.
>> >
>> > When is the fax tone detected? ? Is it while the call is ringing or
>> > can it be detected after the call is answered? ?My goal is to be able
>> > to have the same extension for a voice and fax call. ?i assume that
>> > the fax 'tones' are standardised and the ones on the wiki are correct?
>> > Also, I guess this doesn't work with media bypass (which I don't
>> > use).
>> >
>> > Thanks!
>> >
>> >
>> > On Mon, Oct 5, 2009 at 9:56 AM, Michael Jerris <mike at jerris.com>
>> > wrote:
>> >> check out
>> >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
>> >>
>> >> Note, you can't just have tone_detect as your last iten in the
>> >> dialplan as the call will just get hung up.
>> >>
>> >> Mike
>> >>
>> >> On Oct 4, 2009, at 9:03 AM, Mark Campbell-Smith wrote:
>> >>
>> >>> Hi
>> >>>
>> >>> I was hoping someone could help me to setup the fax detection / tone
>> >>> detection application.
>> >>>
>> >>> I want to be able to transfer an incoming fax to a specific
>> >>> extension.
>> >>> In my default.xml file, I have the following (extracted):
>> >>>
>> >>> ? ?<extension name="1000">
>> >>> ? ? ?<condition field="destination_number" expression="^(10[01]
>> >>> [0-9])
>> >>> $">
>> >>> ? ? ? ?<action application="fax_detect"/>
>> >>> ? ? ? ?<action application="tone_detect" data="fax 1100 r +5000
>> >>> transfer fax XML features" />
>> >>>
>> >>> I can't get the fax to be detected and transferred. ?Is there any
>> >>> way
>> >>> this can be done?
>> >>>
>> >>> Thanks!
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From nicolas at medularis.com  Mon Oct  5 21:19:18 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Tue, 6 Oct 2009 00:19:18 -0400
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <87f2f3b90910050920w4d17033fl4c386a0018107bdf@mail.gmail.com>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<191c3a030909291122t770af037j5b1068197b31de87@mail.gmail.com>
	<1b46b4e80909291542k71440965g75bb3bfcf38e8005@mail.gmail.com>
	<1b46b4e80910011229k31f7233du10b0fc661730d3f3@mail.gmail.com>
	<F86F950D-9106-49C9-81CA-38B3EB2072F4@jerris.com>
	<1b46b4e80910041519h3581d3d2x7b84041e8c18588@mail.gmail.com>
	<C887D143-F7A1-44B9-B8A2-D34A4B4E582E@jerris.com>
	<1b46b4e80910041609x2f41ceb9u142cb484951a21ed@mail.gmail.com>
	<87f2f3b90910050920w4d17033fl4c386a0018107bdf@mail.gmail.com>
Message-ID: <1b46b4e80910052119me12a0d9y4653edd831f9555e@mail.gmail.com>

There was no sane way of doing that, so I ended up logging the trace from
the cli.

Here's the bad registration:

- http://pastebin.freeswitch.org/10605

Here's the good one:

- http://pastebin.freeswitch.org/10606

I am not sure if the second one is complete because for some reason the
first few packages don't appear on the console when doing 'sofia profile
external restart reloadxml' and 'sofia profile external siptrace on' or
viceversa.

Anyway, thanks for your time, and I hope those traces help in figuring out
what's going on.


Nicolas


PS: Is there anyway to get the same format from a pcap dump as with the
siptrace feature on the cli?


On Mon, Oct 5, 2009 at 12:20 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Sun, Oct 4, 2009 at 4:09 PM, Nicolas Brenner <nicolas at medularis.com>wrote:
>
>> Mike, how exactly should I format the file? I got the pcap file, how do I
>> convert it to text so that you can easily read it?
>>
>>
> you can open it with wireshark, follow the TCP or UDP stream, then just
> copy & paste the text as needed...
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mgende at gendesign.com  Mon Oct  5 21:20:55 2009
From: mgende at gendesign.com (Michael Gende)
Date: Mon, 5 Oct 2009 23:20:55 -0500
Subject: [Freeswitch-users] Wiki Info: Multi Homed Tutorial
In-Reply-To: <87f2f3b90910051527x638f55b8hda75d5302267c251@mail.gmail.com>
References: <87f2f3b90910051527x638f55b8hda75d5302267c251@mail.gmail.com>
Message-ID: <a87d43fd0910052120l76da1536ra92766174dc938d3@mail.gmail.com>

Michael,

Thanks for "wiki-fying" my text-only attempt at some user doc. I should have
done that for you. I actually have an updated version with many corrections
and the end tabs filled in. Can you point me to info on how I can amend and
append what you have kindly put up?

Mike G.

On Mon, Oct 5, 2009 at 5:27 PM, Michael Collins <msc at freeswitch.org> wrote:

> FYI,
>
> For those who've been following the thread about Michael Gende's tutorial I
> just wanted to let you know that I his document on the wiki. It can be found
> here:
>
> http://wiki.freeswitch.org/wiki/Multi_home_tutorial
>
> Please feel free to get in there and try it out, make editorial changes,
> etc. I'm sure you won't hurt Mike's feelings by adding your thoughts. :)
>
> If you have any questions please reply to this thread and we'll take it
> from there.
>
> Thanks,
> MC
>
>
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From vdc1048 at tx.rr.com  Mon Oct  5 21:30:02 2009
From: vdc1048 at tx.rr.com (David Clark)
Date: Mon, 05 Oct 2009 23:30:02 -0500
Subject: [Freeswitch-users] Basic compile question.
In-Reply-To: <4C985F33-8B02-4F8F-95B5-36BA6B40690F@freeswitch.org>
References: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>
	<e403cddca1c90c83ce4fb6f85b026e63.squirrel@correo.nodo50.org>
	<65d96fc80910020532i4ba33353g7ac141624a4bb89e@mail.gmail.com>
	<C1AD67AA-F969-4BA1-8415-5E381BDF18FA@jerris.com>
	<90823c940910051257u403108afg3478c4c13ab26283@mail.gmail.com>
	<7e2ac3270910051319xfdfc9d3wa583c2da7e932339@mail.gmail.com>
	<6.2.3.4.2.20091005154158.02b632a8@pop-server.tx.rr.com>
	<4C985F33-8B02-4F8F-95B5-36BA6B40690F@freeswitch.org>
Message-ID: <6.2.3.4.2.20091005232815.03231998@pop-server.tx.rr.com>

No I found the one header.  I added it to the include list for the 
project.  It included something else, added that.  etc.  Basically I 
think I am going to need the VC 2008
compiler  and to use the other project file.

At 04:41 PM 10/5/2009, Brian West wrote:
>Have you updated today?
>
>/b
>
>On Oct 5, 2009, at 3:49 PM, David Clark wrote:
>
> > Any idea what is up?
>
>
>_______________________________________________
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org




From nicolas at medularis.com  Mon Oct  5 21:42:32 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Tue, 6 Oct 2009 00:42:32 -0400
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <1b46b4e80910052119me12a0d9y4653edd831f9555e@mail.gmail.com>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<191c3a030909291122t770af037j5b1068197b31de87@mail.gmail.com>
	<1b46b4e80909291542k71440965g75bb3bfcf38e8005@mail.gmail.com>
	<1b46b4e80910011229k31f7233du10b0fc661730d3f3@mail.gmail.com>
	<F86F950D-9106-49C9-81CA-38B3EB2072F4@jerris.com>
	<1b46b4e80910041519h3581d3d2x7b84041e8c18588@mail.gmail.com>
	<C887D143-F7A1-44B9-B8A2-D34A4B4E582E@jerris.com>
	<1b46b4e80910041609x2f41ceb9u142cb484951a21ed@mail.gmail.com>
	<87f2f3b90910050920w4d17033fl4c386a0018107bdf@mail.gmail.com>
	<1b46b4e80910052119me12a0d9y4653edd831f9555e@mail.gmail.com>
Message-ID: <1b46b4e80910052142jaa09f08l72e5e4b2b24d10fc@mail.gmail.com>

Ignore my previous email, the traces were incomplete, got much better (and
complete) traces with ngrep (found a suggestion from Brian in the list
archive, thanks!)

The gateway that registers:

- http://pastebin.freeswitch.org/10607

The one that doesn't:

- http://pastebin.freeswitch.org/10608


Thanks again for your time and help!


Nicolas


On Tue, Oct 6, 2009 at 12:19 AM, Nicolas Brenner <nicolas at medularis.com>wrote:

> There was no sane way of doing that, so I ended up logging the trace from
> the cli.
>
> Here's the bad registration:
>
> - http://pastebin.freeswitch.org/10605
>
> Here's the good one:
>
> - http://pastebin.freeswitch.org/10606
>
> I am not sure if the second one is complete because for some reason the
> first few packages don't appear on the console when doing 'sofia profile
> external restart reloadxml' and 'sofia profile external siptrace on' or
> viceversa.
>
> Anyway, thanks for your time, and I hope those traces help in figuring out
> what's going on.
>
>
> Nicolas
>
>
> PS: Is there anyway to get the same format from a pcap dump as with the
> siptrace feature on the cli?
>
>
> On Mon, Oct 5, 2009 at 12:20 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Sun, Oct 4, 2009 at 4:09 PM, Nicolas Brenner <nicolas at medularis.com>wrote:
>>
>>> Mike, how exactly should I format the file? I got the pcap file, how do I
>>> convert it to text so that you can easily read it?
>>>
>>>
>> you can open it with wireshark, follow the TCP or UDP stream, then just
>> copy & paste the text as needed...
>>
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From mcampbellsmith at gmail.com  Mon Oct  5 21:43:41 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 6 Oct 2009 15:43:41 +1100
Subject: [Freeswitch-users] Detecting a fax
In-Reply-To: <33c87fa30910040603p7929df26l88f51fc7d256408f@mail.gmail.com>
References: <33c87fa30910040603p7929df26l88f51fc7d256408f@mail.gmail.com>
Message-ID: <33c87fa30910052143w9c9913chff5eacc2d7a2303e@mail.gmail.com>

Further playing around and everything is working fine (even the
emailing).  I'm not sure what I changed though to document it.

cheers
/M



On Mon, Oct 5, 2009 at 12:03 AM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> Hi
>
> I was hoping someone could help me to setup the fax detection / tone
> detection application.
>
> I want to be able to transfer an incoming fax to a specific extension.
> ?In my default.xml file, I have the following (extracted):
>
> ? ?<extension name="1000">
> ? ? ?<condition field="destination_number" expression="^(10[01][0-9])$">
> ? ? ? ?<action application="fax_detect"/>
> ? ? ? ?<action application="tone_detect" data="fax 1100 r +5000
> transfer fax XML features" />
>
> I can't get the fax to be detected and transferred. ?Is there any way
> this can be done?
>
> Thanks!
>



From sprice at gmail.com  Mon Oct  5 21:45:39 2009
From: sprice at gmail.com (SP)
Date: Mon, 5 Oct 2009 23:45:39 -0500
Subject: [Freeswitch-users] Detecting a fax
In-Reply-To: <33c87fa30910052115m268b6e48q33d90946585334f9@mail.gmail.com>
References: <33c87fa30910040603p7929df26l88f51fc7d256408f@mail.gmail.com>
	<91832961-FBA6-465F-AFD8-6A1BE604F6BC@jerris.com>
	<33c87fa30910050328l3d119939pd638b44b219a0140@mail.gmail.com>
	<402585BB-EBEA-42B8-9A2A-58FBA19B066F@jerris.com>
	<65d96fc80910050732g2f414dffs45375cfb16b08c39@mail.gmail.com>
	<33c87fa30910052115m268b6e48q33d90946585334f9@mail.gmail.com>
Message-ID: <7e2ac3270910052145s8c7ac0cqa9c1bd6c4549089b@mail.gmail.com>

try using the hanup hook

<action application="set" data="api_hangup_hook=system ${syscmd}"/>

On Mon, Oct 5, 2009 at 23:15, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> Thanks for your help Mike and Tihomir.
>
> A little more playing around and I found that having <action
> application="fax_detect"/> as well as <action
> application="tone_detect" data="fax 1100 r +5000 transfer fax XML
> features" /> do not work together.
>
> Simply by removing fax_detect, the fax is detected beautifully.
>
> My problem now is trying to email the fax. ?I followed the
> instructions on the wiki at http://wiki.freeswitch.org/wiki/Mod_fax,
> but the dialplan <action application="system" .../> is not executed
> after the rxfax command. ?I know the script works because if I put the
> system command in another part of the dialplan and hard code the
> filename to attach, then the email is sent.
>
> ? ?<extension name="receivefax">
> ? ? ?<condition field="destination_number" expression="^receivefax$">
> ? ? ? ?<action application="answer" />
> ? ? ? ?<action application="playback" data="silence_stream://2000"/>
> ? ? ? ?<action application="rxfax"
> data="//usr//local//freeswitch//storage//${caller_id_number}-${uuid}.rxfax.tiff"/>
> ? ? ? ?<action application="system"
> data="/usr/local/freeswitch/scripts/emailfax.sh
> /usr/local/freeswitch/storage/${caller_id_number}-${uuid}.rxfax.tiff"/>
> ? ? ? ?<action application="hangup"/>
> ? ? ?</condition>
> ? ?</extension>
>
> ideas?
> Thanks!
>
> On Tue, Oct 6, 2009 at 1:32 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>> hi Mark,
>>
>> This is an inbound call leg and media channel (so far)? is open in reverse
>> direction only (application ringback). I'm afraid you have to answer the
>> call to be able to "hear" the fax tone.
>>
>> T.
>>
>>
>>
>> On Mon, Oct 5, 2009 at 2:32 PM, Michael Jerris <mike at jerris.com> wrote:
>>>
>>> Fax tones are not played by the remote machine until after answer, the
>>> tone_detect application starts a media bug that listens for the tone,
>>> can you confirm the tone is happening at all. ?Maybe the issue here is
>>> the timeout, try making that longer, or doing the tone_detect in
>>> execute_on_answer
>>>
>>> Mike
>>>
>>> On Oct 5, 2009, at 6:28 AM, Mark Campbell-Smith wrote:
>>>
>>> > Thanks for the response Mike,
>>> >
>>> > I read that page and this one (among others)
>>> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_fax_detect, but
>>> > I'm still lost. ?This is an extract of my dialplan
>>> >
>>> > ? ?<extension name="Local">
>>> > ? ? ?<condition field="destination_number" expression="^(10[01][0-9])
>>> > $">
>>> > ? ? ? ?<action application="set" data="dialed_extension=$1"/>
>>> > ? ? ? ?<action application="export" data="dialed_extension=$1"/>
>>> > ? ? ? ?<action application="set" data="ringback=${au-ring}"/>
>>> > ? ? ? ?<action application="fax_detect"/>
>>> > ? ? ? ?<action application="tone_detect" data="fax 1100 r +5000
>>> > transfer fax XML features" />
>>> > ? ? ? ?<action application="set" data="hangup_after_bridge=true"/>
>>> > ? ? ? ?<action application="set" data="continue_on_fail=true"/>
>>> > ? ? ? ?<action application="bridge" data="user/${dialed_extension}@$
>>> > {domain}"/>
>>> >
>>> > I would assume that on detecting a fax, the dialplan 'fax' is called
>>> > in context features. ?This never happens.
>>> >
>>> > When is the fax tone detected? ? Is it while the call is ringing or
>>> > can it be detected after the call is answered? ?My goal is to be able
>>> > to have the same extension for a voice and fax call. ?i assume that
>>> > the fax 'tones' are standardised and the ones on the wiki are correct?
>>> > Also, I guess this doesn't work with media bypass (which I don't
>>> > use).
>>> >
>>> > Thanks!
>>> >
>>> >
>>> > On Mon, Oct 5, 2009 at 9:56 AM, Michael Jerris <mike at jerris.com>
>>> > wrote:
>>> >> check out
>>> >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
>>> >>
>>> >> Note, you can't just have tone_detect as your last iten in the
>>> >> dialplan as the call will just get hung up.
>>> >>
>>> >> Mike
>>> >>
>>> >> On Oct 4, 2009, at 9:03 AM, Mark Campbell-Smith wrote:
>>> >>
>>> >>> Hi
>>> >>>
>>> >>> I was hoping someone could help me to setup the fax detection / tone
>>> >>> detection application.
>>> >>>
>>> >>> I want to be able to transfer an incoming fax to a specific
>>> >>> extension.
>>> >>> In my default.xml file, I have the following (extracted):
>>> >>>
>>> >>> ? ?<extension name="1000">
>>> >>> ? ? ?<condition field="destination_number" expression="^(10[01]
>>> >>> [0-9])
>>> >>> $">
>>> >>> ? ? ? ?<action application="fax_detect"/>
>>> >>> ? ? ? ?<action application="tone_detect" data="fax 1100 r +5000
>>> >>> transfer fax XML features" />
>>> >>>
>>> >>> I can't get the fax to be detected and transferred. ?Is there any
>>> >>> way
>>> >>> this can be done?
>>> >>>
>>> >>> Thanks!
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Shannon



From xengelpublicx at gmail.com  Tue Oct  6 00:21:03 2009
From: xengelpublicx at gmail.com (Vladimir Elizarov)
Date: Tue, 06 Oct 2009 11:21:03 +0400
Subject: [Freeswitch-users] stun not working in fs 1.0.4?
In-Reply-To: <4AA6F4E2-2138-439C-86E8-7554A5D110D4@freeswitch.org>
References: <4AC9E9EE.8090805@gmail.com>	<7C489261-F5F3-4B57-B807-21EC4A3947A2@freeswitch.org>	<4ACA31ED.1000002@gmail.com>
	<4AA6F4E2-2138-439C-86E8-7554A5D110D4@freeswitch.org>
Message-ID: <4ACAEFDF.4030604@gmail.com>

Brian West ?????:
> Because TRUNK is stable... its only fixes going in usually and if  
> things do break they don't stay broken for long.
>
> Ask anyone our trunk is more table then most commercial products.
>   
This separation of the branches a very bad influence on the packaging.
That is gathered deb-package trunk 15094. Man found in the trunk bug.
Must again rebuild the package from the new trunk...
> /b
>
> On Oct 5, 2009, at 12:50 PM, Vladimir Elizarov wrote:
>
>   
>> Ok. Brian, why fs no two branches of the stable and trunk?
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


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From moizchinoy at gmail.com  Tue Oct  6 00:58:17 2009
From: moizchinoy at gmail.com (Moiz Chinoy)
Date: Tue, 6 Oct 2009 11:58:17 +0400
Subject: [Freeswitch-users] Mobile Phone As GSM Gateway....
Message-ID: <29b888f80910060058m4298a829s414a322f8548ff81@mail.gmail.com>

Hi,

Is it possible to connect a mobile phone (GSM phone) to Freeswitch and
use this as a GSM gateway?

-- 
Regards,
Moiz Chinoy.



From yehavi.bourvine at gmail.com  Tue Oct  6 01:14:31 2009
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 6 Oct 2009 10:14:31 +0200
Subject: [Freeswitch-users] Bridge application with shared lines
Message-ID: <d361ca7b0910060114g3f23a102nb3508cf6e1e196fc@mail.gmail.com>

Hello,

  We have Polycom and SNOM phones running with FreeSwitch. The Polycoms have
shared lines defined and the SNOMs have both shared lines and BLFs (defined
as extensions in the phone config). I've tried supporting both, but have
some incompatibility:


   - When calling the Bridge application with data parameter of *sofia*/*
   profile-name/number at domain* the BLF works ok, but not the shared lines
   (i.e only one of the phones  rings).
   - When calling the Bridge application with data parameter of *
   ${sofia_contact(*/*profile-name/number at domain*)} shared lines work ok but
   BLF doesn't fire up.


How do I support both? Is there a way to know whether the destination is a
shared one and then chose one of the above formats?

                                  Thanks! __Yehavi:
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From itamar at ispbrasil.com.br  Tue Oct  6 01:17:44 2009
From: itamar at ispbrasil.com.br (Itamar Reis Peixoto)
Date: Tue, 6 Oct 2009 05:17:44 -0300
Subject: [Freeswitch-users] Mobile Phone As GSM Gateway....
In-Reply-To: <29b888f80910060058m4298a829s414a322f8548ff81@mail.gmail.com>
References: <29b888f80910060058m4298a829s414a322f8548ff81@mail.gmail.com>
Message-ID: <d3ceea380910060117mc02fb80lb059e92c8d0164f7@mail.gmail.com>

a gsm phone not.

take a look

http://portech.com.tw/

I think a portech product can do what you need.




On Tue, Oct 6, 2009 at 4:58 AM, Moiz Chinoy <moizchinoy at gmail.com> wrote:
> Hi,
>
> Is it possible to connect a mobile phone (GSM phone) to Freeswitch and
> use this as a GSM gateway?
>
> --
> Regards,
> Moiz Chinoy.




-- 
------------

Itamar Reis Peixoto

e-mail/msn: itamar at ispbrasil.com.br
sip: itamar at ispbrasil.com.br
skype: itamarjp
icq: 81053601
+55 11 4063 5033
+55 34 3221 8599



From woodydickson at gmail.com  Tue Oct  6 02:00:40 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Tue, 6 Oct 2009 17:00:40 +0800
Subject: [Freeswitch-users] problem with compiling freeswith
Message-ID: <a44688a0910060200q6b6bc15cjf5100b17e63b5bd4@mail.gmail.com>

Hi,

Is this just me who is having this problem?  I can't compile the latest
freeswitch source code and here is the error:

checking for gcc option to accept ANSI C... none needed
checking for style of include used by make... GNU
checking dependency style of gcc... gcc3
checking whether gcc and cc understand -c and -o together... yes
./configure: line 3377: syntax error near unexpected token `echo'
./configure: line 3377: `echo "$as_me:$LINENO: checking for a BSD-compatible
install" >&5'
configure: error: /bin/sh './configure.gnu' failed for libs/tiff-3.8.2
[root at localhost freeswitch-snapshot]#

Does anyone know why?

woody
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From rs at runsolutions.com  Tue Oct  6 02:02:46 2009
From: rs at runsolutions.com (Raimund Sacherer)
Date: Tue, 6 Oct 2009 11:02:46 +0200
Subject: [Freeswitch-users] Youtube - FreeSWITCH Promo Video
In-Reply-To: <191c3a030910050802v59b2a9du6d2af7ade03bc509@mail.gmail.com>
References: <9A87E77C-AD85-4F1E-8ED0-F9F06AA451EE@ken-ton.com>
	<86a32abc0910041651rb6a3ff9k1f6ed7d0b1a9b440@mail.gmail.com>
	<7b197bef0910050004v71c623c8w68a013bef9fea596@mail.gmail.com>
	<191c3a030910050802v59b2a9du6d2af7ade03bc509@mail.gmail.com>
Message-ID: <91D6B861-B8B1-4559-8494-A68D90A0FB4A@runsolutions.com>

Yeah, on top of it would'nt it be nice if:

when they call the giant 3d unix '*' character with the cell phone

* switch inside the con of the giant 3d unix '*'
* People nervous and shouting about "incoming" (like in the fight szenes
    when they call incoming if missiles are fired)
* One is calling "Oh no, one of our THREADs is blocking!!"
    (THREAD = Thermal Heat REAction Device, eg, self deploying cooling  
pipe)
    (or maybe Throttle Heat REAction Device), view of a pipe on the  
point of blocking
    the heat-xchange (e.g. simulate somehow that nothing goes further  
anymore).
* A high ranked officer is shouting
    "Core Dump, Core Dump, Leave the ships through all available  
Channels!"
* switch to outer scene
* you see an anatomically hinting, but technically correct crack  
forming at the rear
    bottom side of the giant 3d unix '*'
* out of this crack comes "the dumped core"
* switch to open scene view
* you see the people leaving the giant 3d unix '*' at all available  
channels
    (e.g. light pulses which look like little ships driven by rockets  
in every direction out of the
    gicant 3d unix '*')
* When the CoreDump is finished, the giant 3d unix '*' implodes in  
itself and leaves
    nothing but the fouly stench of the dumped core (which has to  
resemble somehow
    the same stench you feel after a callcenter full of agents again  
lost all connections
    because of a fat dumb giant 3d unix '*' could not cope with it's  
pipes and dumped its core)



....

:-)

-- 
Raimund Sacherer
-
RunSolutions
     Open Source It Consulting
-
Email: rs at runsolutions.com
tel: 625 40 32 08

Parc Bit - Centro Empresarial Son Espanyol
Edificio Estel - Local 3D
07121 -  Palma de Mallorca
Baleares

On Oct 5, 2009, at 5:02 PM, Anthony Minessale wrote:

> neat,
>
> Here's some suggestions for your next ones. =p
>
> Have them standing around the hologram trying to destroy the "Death  
> Star(tm)" that happens to look a lot like a giant 3d unix '*'  
> character.  Then have one rebel say, "wait!, why are we wasting our  
> time... watch this... and dial a number on his cellphone as the  
> whole thing explodes in the background.
>
> Have Darth Forkium face Luke ThreadSpawner in a dual.  "I see you  
> have fashioned your own TDM card...." vroom......
> "Join me and together we can make linked lists and monolithic  
> processes", "NEVER!..." vroom vroom Master Coda has taught you  
> well....."You are no match for me...JOIN THE ORANGE SIDE OF THE FORCE"
>
>
>
>
>
> On Mon, Oct 5, 2009 at 2:04 AM, Giovanni Maruzzelli <gmaruzz at celliax.org 
> > wrote:
> On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola <diego.viola at gmail.com>  
> wrote:
> > Very nice :)
> >
> > On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling <kjv at ken-ton.com>  
> wrote:
> >>
> >> Folks;
> >> Here's something that I did playing around w/ learning Apple  
> Motion.
>
> Me too: very nice!
>
> -gmaruzz
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From jason at jasonjgw.net  Tue Oct  6 02:18:58 2009
From: jason at jasonjgw.net (Jason White)
Date: Tue, 6 Oct 2009 20:18:58 +1100
Subject: [Freeswitch-users] problem with compiling freeswith
In-Reply-To: <a44688a0910060200q6b6bc15cjf5100b17e63b5bd4@mail.gmail.com>
References: <a44688a0910060200q6b6bc15cjf5100b17e63b5bd4@mail.gmail.com>
Message-ID: <20091006091858.GA18854@jdc.jasonjgw.net>

Woody Dickson <woodydickson at gmail.com> wrote:
> Is this just me who is having this problem?  I can't compile the latest
> freeswitch source code and here is the error:

Try starting with a fresh checkout from the repository.

If the problem persists, please report the operating system and version
thereof, so that someone with access to a similar environment can try to
reproduce the issue.




From m.krivushin at imarto.net  Tue Oct  6 02:25:38 2009
From: m.krivushin at imarto.net (Mikhail Krivushin)
Date: Tue, 6 Oct 2009 16:25:38 +0700
Subject: [Freeswitch-users] fs_path not work
Message-ID: <5be734a50910060225mcef587dw69b12132303dadab@mail.gmail.com>

Hello!

I try to use ;fs_path in originate command, but this seems to not work:
bgapi originate
{origination_caller_id_name=qwe,origination_caller_id_number=qwe,sip_auth_username=qwe,sip_auth_password=qwe,origination_uuid=1ebf2ef8-b259-11de-b7f9-000c29cf246f,fsc_call=1ebf0432-b259-11de-b7f9-000c29cf246f,fsc_leg=legB}sofia/gateway/qwe_gw/
test05 at service.deep.com;fs_path=sip:vm3.deep.com &park

Where I am wrong?

-- 
? ?????????, ???????? ??????
?. ????? ???. +7 913 865 78 66
icq: 218 744 127
xmpp: KrivushinME at jabber.ru
skype: mkrivushin
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From m.krivushin at imarto.net  Tue Oct  6 02:41:43 2009
From: m.krivushin at imarto.net (Mikhail Krivushin)
Date: Tue, 6 Oct 2009 16:41:43 +0700
Subject: [Freeswitch-users] fs_path not work
In-Reply-To: <5be734a50910060225mcef587dw69b12132303dadab@mail.gmail.com>
References: <5be734a50910060225mcef587dw69b12132303dadab@mail.gmail.com>
Message-ID: <5be734a50910060241r707d7018m95337f464577a4fb@mail.gmail.com>

I also try to use "proxy" param in gateway, but this doesnt work too. INVITE
dont going to proxy pointed by me.

-- 
? ?????????, ???????? ??????
?. ????? ???. +7 913 865 78 66
icq: 218 744 127
xmpp: KrivushinME at jabber.ru
skype: mkrivushin
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From ahmedmunir007 at gmail.com  Tue Oct  6 03:11:03 2009
From: ahmedmunir007 at gmail.com (Ahmed Munir)
Date: Tue, 6 Oct 2009 15:11:03 +0500
Subject: [Freeswitch-users] Cannot connect to ODBC driver/database
	freeswitchdb
Message-ID: <a60fed5e0910060311w4ceb749cu520c877bb47c47b8@mail.gmail.com>

Hi,
I've installed FS on Ubuntu 9.04 and I want to run mod_nibbles on it. I
follow the steps to configure my ODBC connection with MySQL as explained in
wiki (mod_nibbles and mod_spidermonkey). But  FS, unable to connect it. The
error I got is listed below when I restart FS,

2009-10-06 15:47:21.164590 [ERR] switch_odbc.c:188 STATE: IM002 CODE 0
ERROR: [unixODBC][Driver Manager]Data source name not found, and no default
driver specified
2009-10-06 15:47:21.164617 [CRIT] mod_nibblebill.c:221 Cannot connect to
ODBC driver/database freeswitchdb (user: root / pass password)
2009-10-06 15:47:21.164650 [CONSOLE] switch_loadable_module.c:889
Successfully Loaded [mod_nibblebill]
2009-10-06 15:47:21.164664 [NOTICE] switch_loadable_module.c:248 Adding
Application 'nibblebill'
2009-10-06 15:47:21.164710 [NOTICE] switch_loadable_module.c:270 Adding API
Function 'nibblebill'

But when I use isql it accepts my odbc connection i.e. isql MySQL-freeswitch

I'm listing my settings of odbc.ini and odbcinst.ini as listed below;

odbc.ini
--------------
[MySQL-freeswitch]
Driver       = MySQL
#Driver       = /usr/lib/odbc/libodbcmyS.so
Description  = Connector/ODBC Driver DSN With FreeSwitch
SERVER       = localhost
PORT         = 3306
USER         = root
Password     = password
Database     = freeswitchdb

odbcinst.ini
-------------------
[MySQL]
Description    = ODBC for MySQL
Driver         = /usr/lib/odbc/libmyodbc.so
Setup          = /usr/lib/odbc/libodbcmyS.so
FileUsage      = 1


odbc.ini and odbcinst.ini are located at /etc/. Even I set my odbc
connection setting as I provide with this link;
http://dev.mysql.com/doc/refman/5.0/en/connector-odbc-configuration-dsn-unix.html

But unfortunately my problem is unresolved then.


Kindly advise me, how can I resolve this problem?

-- 
Regards,

Ahmed Munir
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From asannucci at gmail.com  Tue Oct  6 03:27:46 2009
From: asannucci at gmail.com (bakko)
Date: Tue, 6 Oct 2009 05:27:46 -0500
Subject: [Freeswitch-users] Cannot connect to ODBC
	driver/databasefreeswitchdb
In-Reply-To: <a60fed5e0910060311w4ceb749cu520c877bb47c47b8@mail.gmail.com>
References: <a60fed5e0910060311w4ceb749cu520c877bb47c47b8@mail.gmail.com>
Message-ID: <EA1280B80AF74924955FA379860DE26E@voztovoice>

Hi,

are you configured correctly the nibblebill.conf.xml file?

BR
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From lakindia89 at gmail.com  Tue Oct  6 03:30:03 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Tue, 6 Oct 2009 16:00:03 +0530
Subject: [Freeswitch-users] Outgoing via openzap is not working
Message-ID: <7d79b3930910060330s77a2bbbv88d8014b73f91663@mail.gmail.com>

Hi I'm using freeswitch1.0.4. This post is moreover similar to my previous
post.
When I make an outgoing call, it is saying INVALID_IE_CONTENTS.
Here are the details.
openzap.conf.xml

<configuration name="openzap.conf" description="OpenZAP Configuration">
<settings>
<param name="debug" value="7"/>
</settings>
<libpri_spans>
<span name="PRI_1">
<param name="node" value="cpe"/>
<param name="switch" value="euroisdn"/>
<param name="dialplan" value="XML"/>
<param name="context" value="default"/>
</span>
</libpri_spans>
</configuration>

openzap.conf
[span zt PRI_1]
trunk_type => e1
b-channel => 1:1-15
d-channel=> 1:16
b-channel => 1:17-31
oz libpri debug 1 all

API CALL [oz(libpri debug 1 all)] output:
src/ozmod/ozmod_libpri/ozmod_libpri.c: +OK debug set.

oz list

API CALL [oz(list)] output:
+OK
span: 1 (PRI_1)
type: isdn
chan_count: 47
dialplan: XML
context: default
dial_regex:
fail_dial_regex:
hold_music:
analog_options none

Freeswitch startup log:
http://pastebin.freeswitch.org/10609
After saying originate openzap/1/1/9952248266 openzap/1/1/9952248266
http://pastebin.freeswitch.org/10610

Please help me to solve this.
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From dujinfang at gmail.com  Tue Oct  6 05:04:58 2009
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 6 Oct 2009 20:04:58 +0800
Subject: [Freeswitch-users] Mobile Phone As GSM Gateway....
In-Reply-To: <29b888f80910060058m4298a829s414a322f8548ff81@mail.gmail.com>
References: <29b888f80910060058m4298a829s414a322f8548ff81@mail.gmail.com>
Message-ID: <23f91030910060504y3d060990l44ec913fb130954c@mail.gmail.com>

maybe you can check this: http://www.gsmopen.org/

2009/10/6 Moiz Chinoy <moizchinoy at gmail.com>

> Hi,
>
> Is it possible to connect a mobile phone (GSM phone) to Freeswitch and
> use this as a GSM gateway?
>
> --
> Regards,
> Moiz Chinoy.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From srinivas.ksvreddy at gmail.com  Tue Oct  6 05:29:25 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 6 Oct 2009 17:59:25 +0530
Subject: [Freeswitch-users] Dynamic updation of groups in default.xml
Message-ID: <f8af5740910060529v5d415d6bla4e54121d8ad2ff@mail.gmail.com>

Hi,
Can any one tell me how to add users dynamically to groups in default.xml,
with out restart the freeswitch.


Thanks
Srinivasula Reddy K
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From srinivas.ksvreddy at gmail.com  Tue Oct  6 05:31:52 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 6 Oct 2009 18:01:52 +0530
Subject: [Freeswitch-users] add users dynamically to groups in default.xml
Message-ID: <f8af5740910060531j58158057h881d9862602d360c@mail.gmail.com>

Hi,
Can any one tell me how to add users dynamically to groups in default.xml,
with out restart the freeswitch.

Thanks
Srinivasula Reddy K
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From dujinfang at gmail.com  Tue Oct  6 05:38:50 2009
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 6 Oct 2009 20:38:50 +0800
Subject: [Freeswitch-users] Dynamic updation of groups in default.xml
In-Reply-To: <f8af5740910060529v5d415d6bla4e54121d8ad2ff@mail.gmail.com>
References: <f8af5740910060529v5d415d6bla4e54121d8ad2ff@mail.gmail.com>
Message-ID: <23f91030910060538x6b1b2bddwa9f287e4e5c0251d@mail.gmail.com>

change the xml and execute "reloadxml" in FS console or fs_cli

or you can check mod_xml_curl

2009/10/6 srinivasula reddy <srinivas.ksvreddy at gmail.com>

> Hi,
> Can any one tell me how to add users dynamically to groups in default.xml,
> with out restart the freeswitch.
>
>
> Thanks
> Srinivasula Reddy K
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tomabroad at gmail.com  Tue Oct  6 05:44:51 2009
From: tomabroad at gmail.com (tom)
Date: Tue, 6 Oct 2009 08:44:51 -0400
Subject: [Freeswitch-users] Wiki Info: Multi Homed Tutorial
In-Reply-To: <a87d43fd0910052120l76da1536ra92766174dc938d3@mail.gmail.com>
References: <87f2f3b90910051527x638f55b8hda75d5302267c251@mail.gmail.com>
	<a87d43fd0910052120l76da1536ra92766174dc938d3@mail.gmail.com>
Message-ID: <6f7c60c40910060544h5d1f7251p7194bcaa944a627c@mail.gmail.com>

thx guys, that helped me alot!


On Tue, Oct 6, 2009 at 12:20 AM, Michael Gende <mgende at gendesign.com> wrote:

> Michael,
>
> Thanks for "wiki-fying" my text-only attempt at some user doc. I should
> have done that for you. I actually have an updated version with many
> corrections and the end tabs filled in. Can you point me to info on how I
> can amend and append what you have kindly put up?
>
> Mike G.
>
>
> On Mon, Oct 5, 2009 at 5:27 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> FYI,
>>
>> For those who've been following the thread about Michael Gende's tutorial
>> I just wanted to let you know that I his document on the wiki. It can be
>> found here:
>>
>> http://wiki.freeswitch.org/wiki/Multi_home_tutorial
>>
>> Please feel free to get in there and try it out, make editorial changes,
>> etc. I'm sure you won't hurt Mike's feelings by adding your thoughts. :)
>>
>> If you have any questions please reply to this thread and we'll take it
>> from there.
>>
>> Thanks,
>> MC
>>
>>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kjv at ken-ton.com  Tue Oct  6 05:51:05 2009
From: kjv at ken-ton.com (Karl Vesterling)
Date: Tue, 6 Oct 2009 08:51:05 -0400
Subject: [Freeswitch-users] Youtube - FreeSWITCH Promo Video
In-Reply-To: <91D6B861-B8B1-4559-8494-A68D90A0FB4A@runsolutions.com>
References: <9A87E77C-AD85-4F1E-8ED0-F9F06AA451EE@ken-ton.com>
	<86a32abc0910041651rb6a3ff9k1f6ed7d0b1a9b440@mail.gmail.com>
	<7b197bef0910050004v71c623c8w68a013bef9fea596@mail.gmail.com>
	<191c3a030910050802v59b2a9du6d2af7ade03bc509@mail.gmail.com>
	<91D6B861-B8B1-4559-8494-A68D90A0FB4A@runsolutions.com>
Message-ID: <E83DE12B-8728-4778-B2E6-B44E1A0A347F@ken-ton.com>


I'm flattered that you consider my abilities so capable, but time nor  
budget are available to afford such extravagance as outlined below.

Besides, bashing something doesn't really gain you any respect (but I  
do think that _*_ does ever sooo much deserve bashing).

What I have done is added to the end of the movie a Cisco 7900 series  
(top left), a Snom 360 (top right), and a Grandstream video phone  
(bottom right.)

I need more though...

I was thinking the point of the video was more along the lines to get  
people interested in FreeSWITCH, not bash such an easy target as  
(insert * here).

I think that can be done by showing various equipment from multiple  
manufacturers at the end of the movie, with another title declaring,  
"INTEROPERABILITY"...

Fade some in, fade out, fade some more in, perhaps including a sangoma  
card, over and over again...

So far I've got artwork for Cisco 7900, Snom, Grandstream...  I could  
put in a SPA-942, etc...
These are the type of "suggestions" I'm looking for... Although I  
really do like the drama with, "Oh no!  One of our threads is  
blocking!" (still chuckling...)  It did give me a few ideas too, so  
keep em' coming!

Best Regards,
Karl J. Vesterling
kjv at ken-ton.com
202-461-3231 x0

On Oct 6, 2009, at 5:02 AM, Raimund Sacherer wrote:

> Yeah, on top of it would'nt it be nice if:
>
> when they call the giant 3d unix '*' character with the cell phone
>
> * switch inside the con of the giant 3d unix '*'
> * People nervous and shouting about "incoming" (like in the fight  
> szenes
>    when they call incoming if missiles are fired)
> * One is calling "Oh no, one of our THREADs is blocking!!"
>    (THREAD = Thermal Heat REAction Device, eg, self deploying cooling
> pipe)
>    (or maybe Throttle Heat REAction Device), view of a pipe on the
> point of blocking
>    the heat-xchange (e.g. simulate somehow that nothing goes further
> anymore).
> * A high ranked officer is shouting
>    "Core Dump, Core Dump, Leave the ships through all available
> Channels!"
> * switch to outer scene
> * you see an anatomically hinting, but technically correct crack
> forming at the rear
>    bottom side of the giant 3d unix '*'
> * out of this crack comes "the dumped core"
> * switch to open scene view
> * you see the people leaving the giant 3d unix '*' at all available
> channels
>    (e.g. light pulses which look like little ships driven by rockets
> in every direction out of the
>    gicant 3d unix '*')
> * When the CoreDump is finished, the giant 3d unix '*' implodes in
> itself and leaves
>    nothing but the fouly stench of the dumped core (which has to
> resemble somehow
>    the same stench you feel after a callcenter full of agents again
> lost all connections
>    because of a fat dumb giant 3d unix '*' could not cope with it's
> pipes and dumped its core)
>
>
>
> ....
>
> :-)
>
> -- 
> Raimund Sacherer
> -
> RunSolutions
>     Open Source It Consulting
> -
> Email: rs at runsolutions.com
> tel: 625 40 32 08
>
> Parc Bit - Centro Empresarial Son Espanyol
> Edificio Estel - Local 3D
> 07121 -  Palma de Mallorca
> Baleares
>
> On Oct 5, 2009, at 5:02 PM, Anthony Minessale wrote:
>
>> neat,
>>
>> Here's some suggestions for your next ones. =p
>>
>> Have them standing around the hologram trying to destroy the "Death
>> Star(tm)" that happens to look a lot like a giant 3d unix '*'
>> character.  Then have one rebel say, "wait!, why are we wasting our
>> time... watch this... and dial a number on his cellphone as the
>> whole thing explodes in the background.
>>
>> Have Darth Forkium face Luke ThreadSpawner in a dual.  "I see you
>> have fashioned your own TDM card...." vroom......
>> "Join me and together we can make linked lists and monolithic
>> processes", "NEVER!..." vroom vroom Master Coda has taught you
>> well....."You are no match for me...JOIN THE ORANGE SIDE OF THE  
>> FORCE"
>>
>>
>>
>>
>>
>> On Mon, Oct 5, 2009 at 2:04 AM, Giovanni Maruzzelli <gmaruzz at celliax.org
>>> wrote:
>> On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola <diego.viola at gmail.com>
>> wrote:
>>> Very nice :)
>>>
>>> On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling <kjv at ken-ton.com>
>> wrote:
>>>>
>>>> Folks;
>>>> Here's something that I did playing around w/ learning Apple
>> Motion.
>>
>> Me too: very nice!
>>
>> -gmaruzz
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>>
>>
>> -- 
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:213-799-1400
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>




From Russell.Mosemann at cune.org  Tue Oct  6 05:59:50 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Tue, 6 Oct 2009 12:59:50 -0000
Subject: [Freeswitch-users] Outgoing via openzap is not working
In-Reply-To: <7d79b3930910060330s77a2bbbv88d8014b73f91663@mail.gmail.com>
Message-ID: <20091006125950.EB7073DECDF@mail.cune.org>

lakshmanan ganapathy <lakindia89 at gmail.com> said:

> When I make an outgoing call, it is saying INVALID_IE_CONTENTS.
> Here are the details.

There are a couple of things that might be OK, but they seem odd. When
ozmod starts, the first 15 channels can't be configured, because they are
busy. The output from "oz list" shows 47 channels, as if 15 channels are
added to the E1's 31 channels (and one control channel). When libpri is
checking the numbers, it shows the caller's number as '0000000000'.

The output from libpri shows that the call is being released by the other
side of the E1 because of a protocol error involving the redirecting
number (invalid contents). My guess is that you aren't providing a valid
caller number, but I could be completely wrong. Someone with more
experience will need to interpret the information. I don't see the
originate command in the debug output. Did you supply both the calling
and called number?

http://wiki.freeswitch.org/wiki/Mod_commands#originate

-- 
Russell Mosemann

________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!




From anthony.minessale at gmail.com  Tue Oct  6 06:49:32 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 6 Oct 2009 08:49:32 -0500
Subject: [Freeswitch-users] fs_path not work
In-Reply-To: <5be734a50910060241r707d7018m95337f464577a4fb@mail.gmail.com>
References: <5be734a50910060225mcef587dw69b12132303dadab@mail.gmail.com>
	<5be734a50910060241r707d7018m95337f464577a4fb@mail.gmail.com>
Message-ID: <191c3a030910060649o16f15cd9q7443cbf406f13041@mail.gmail.com>

gateway calls do not contain any uri data
sofia/gateway/mygw/1000
is all you can do

if you want all that other stuff you need to formulate a direct url
connection


On Tue, Oct 6, 2009 at 4:41 AM, Mikhail Krivushin <m.krivushin at imarto.net>wrote:

> I also try to use "proxy" param in gateway, but this doesnt work too.
> INVITE dont going to proxy pointed by me.
>
>
> --
> ? ?????????, ???????? ??????
> ?. ????? ???. +7 913 865 78 66
> icq: 218 744 127
> xmpp: KrivushinME at jabber.ru
> skype: mkrivushin
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From brian at freeswitch.org  Tue Oct  6 07:31:06 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 6 Oct 2009 09:31:06 -0500
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <1b46b4e80910052142jaa09f08l72e5e4b2b24d10fc@mail.gmail.com>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<191c3a030909291122t770af037j5b1068197b31de87@mail.gmail.com>
	<1b46b4e80909291542k71440965g75bb3bfcf38e8005@mail.gmail.com>
	<1b46b4e80910011229k31f7233du10b0fc661730d3f3@mail.gmail.com>
	<F86F950D-9106-49C9-81CA-38B3EB2072F4@jerris.com>
	<1b46b4e80910041519h3581d3d2x7b84041e8c18588@mail.gmail.com>
	<C887D143-F7A1-44B9-B8A2-D34A4B4E582E@jerris.com>
	<1b46b4e80910041609x2f41ceb9u142cb484951a21ed@mail.gmail.com>
	<87f2f3b90910050920w4d17033fl4c386a0018107bdf@mail.gmail.com>
	<1b46b4e80910052119me12a0d9y4653edd831f9555e@mail.gmail.com>
	<1b46b4e80910052142jaa09f08l72e5e4b2b24d10fc@mail.gmail.com>
Message-ID: <083DBCB9-48B2-4E61-81B5-B7D00A42EAB5@freeswitch.org>

This looks like you have an ALG messing with packets... notice it says  
rport 5080 but we are sending to 5060.

/b

On Oct 5, 2009, at 11:42 PM, Nicolas Brenner wrote:

> Ignore my previous email, the traces were incomplete, got much  
> better (and complete) traces with ngrep (found a suggestion from  
> Brian in the list archive, thanks!)
>
> The gateway that registers:
>
> - http://pastebin.freeswitch.org/10607
>
> The one that doesn't:
>
> - http://pastebin.freeswitch.org/10608
>
>
> Thanks again for your time and help!
>
>
> Nicolas
>
>
> On Tue, Oct 6, 2009 at 12:19 AM, Nicolas Brenner <nicolas at medularis.com 
> > wrote:
> There was no sane way of doing that, so I ended up logging the trace  
> from the cli.

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From srinivas.ksvreddy at gmail.com  Tue Oct  6 07:38:57 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 6 Oct 2009 20:08:57 +0530
Subject: [Freeswitch-users] Dynamic updation of groups in default.xml
In-Reply-To: <23f91030910060538x6b1b2bddwa9f287e4e5c0251d@mail.gmail.com>
References: <f8af5740910060529v5d415d6bla4e54121d8ad2ff@mail.gmail.com>
	<23f91030910060538x6b1b2bddwa9f287e4e5c0251d@mail.gmail.com>
Message-ID: <f8af5740910060738l111b6ab7y2c5b05c505951c4e@mail.gmail.com>

Thank u very much seven du.

Regards
Srinvas


On Tue, Oct 6, 2009 at 6:08 PM, Seven Du <dujinfang at gmail.com> wrote:

> change the xml and execute "reloadxml" in FS console or fs_cli
>
> or you can check mod_xml_curl
>
> 2009/10/6 srinivasula reddy <srinivas.ksvreddy at gmail.com>
>
>> Hi,
>> Can any one tell me how to add users dynamically to groups in
>> default.xml,  with out restart the freeswitch.
>>
>>
>> Thanks
>> Srinivasula Reddy K
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Srinivasula Reddy K
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From ChristianDamianidis at globalive.com  Tue Oct  6 07:39:27 2009
From: ChristianDamianidis at globalive.com (Christian Damianidis)
Date: Tue, 6 Oct 2009 10:39:27 -0400
Subject: [Freeswitch-users] mod_xml_curl http POST is inconsistent/bug
In-Reply-To: <276FEE28-70CF-422D-9F6F-7CF747FDDB22@freeswitch.org>
References: <66EA3166EB339A4489B06286C0876A8A0C0C1432@mailserv.Globalive.local>
	<276FEE28-70CF-422D-9F6F-7CF747FDDB22@freeswitch.org>
Message-ID: <66EA3166EB339A4489B06286C0876A8A0C0C1BA9@mailserv.Globalive.local>

This web request goes to a server running IIS on Windows Server 2003.

 

From: Brian West [mailto:brian at freeswitch.org] 
Sent: Monday, October 05, 2009 5:43 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_xml_curl http POST is
inconsistent/bug

 

Are you using something other than apache?

 

/b

 

On Oct 5, 2009, at 1:25 PM, Christian Damianidis wrote:





 

The inconsistent POST request sent by the module causes freeswitch to
hang for 1-2 minutes during start-up.

 

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From anthony.minessale at gmail.com  Tue Oct  6 08:01:46 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 6 Oct 2009 10:01:46 -0500
Subject: [Freeswitch-users] mod_xml_curl http POST is inconsistent/bug
In-Reply-To: <66EA3166EB339A4489B06286C0876A8A0C0C1BA9@mailserv.Globalive.local>
References: <66EA3166EB339A4489B06286C0876A8A0C0C1432@mailserv.Globalive.local>
	<276FEE28-70CF-422D-9F6F-7CF747FDDB22@freeswitch.org>
	<66EA3166EB339A4489B06286C0876A8A0C0C1BA9@mailserv.Globalive.local>
Message-ID: <191c3a030910060801x3ed48e98x6d69b53c6beaca69@mail.gmail.com>

My guess is that we configure the curl to support the full range of http
auth methods.
Some of them like Digest require a challenge and realm etc so it's probably
asking without auth header because it cannot create one until it gets that
data.  In the case of Basic you can send the login and pass right away but
it does not know in advance that it will be basic.

Here is a snippet from the libcurl api docs:
-------------------------------------------------------------------------------------------------------------------------------------------------------------

Both these options allow you to set multiple types (by ORing them together),
to make libcurl pick the most secure one out of the types the server/proxy
claims to support. This method does however add a round-trip since libcurl
must first ask the server what it supports:

 curl_easy_setopt(easyhandle, CURLOPT_HTTPAUTH,
 CURLAUTH_DIGEST|CURLAUTH_BASIC);
-------------------------------------------------------------------------------------------------------------------------------------------------------------

So my guess is that if we set it to only support basic, then it would work
how you expect so if you want to test it for me I can make it into a
parameter.

edit: /usr/src/freeswitch.trunk/src/mod/xml_int/mod_xml_curl/mod_xml_curl.c
line 220
change

curl_easy_setopt(curl_handle, CURLOPT_HTTPAUTH, CURLAUTH_ANY);

to

curl_easy_setopt(curl_handle, CURLOPT_HTTPAUTH, CURLAUTH_BASIC);


If this works i'll think about exposing the auth methods so you can choose
them in the config.


On Tue, Oct 6, 2009 at 9:39 AM, Christian Damianidis <
ChristianDamianidis at globalive.com> wrote:

>  This web request goes to a server running IIS on Windows Server 2003.
>
>
>
> *From:* Brian West [mailto:brian at freeswitch.org]
> *Sent:* Monday, October 05, 2009 5:43 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] mod_xml_curl http POST is
> inconsistent/bug
>
>
>
> Are you using something other than apache?
>
>
>
> /b
>
>
>
> On Oct 5, 2009, at 1:25 PM, Christian Damianidis wrote:
>
>
>
>
>
> The inconsistent POST request sent by the module causes freeswitch to hang
> for 1-2 minutes during start-up.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From woodydickson at gmail.com  Tue Oct  6 08:05:23 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Tue, 6 Oct 2009 23:05:23 +0800
Subject: [Freeswitch-users] overriding conference preference
In-Reply-To: <E773DB8B-AC74-49DE-9B38-7AFC441882F5@jerris.com>
References: <a44688a0910050749s1ae1f9a4vba96477d159d377f@mail.gmail.com>
	<E773DB8B-AC74-49DE-9B38-7AFC441882F5@jerris.com>
Message-ID: <a44688a0910060805o49d549b8vbd4d545b229c75d@mail.gmail.com>

Hi,

Is there anyway of using curl without having to setup a standalone http
service?  Is it possible to generate curl xml using scripts?

woody

On Mon, Oct 5, 2009 at 11:05 PM, Michael Jerris <mike at jerris.com> wrote:

>
> On Oct 5, 2009, at 10:49 AM, Woody Dickson wrote:
>
> > Is is possible to override any of the setting specified in the
> > conference profile?
>
> Just the flags you can pass per user such as pin and mute
>
> >
> > What I want to do is to have a default profile, and be able to
> > modify certain fields if necessary in the dialplan.
> >
> >
> > Alternatively, I would prefer to have a dynamic profile setting for
> > the conference to obtain those parameters from odbc.
>
> you can do this with mod_xml_curl
>
> Mike
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From todd.baumgartner at gmail.com  Tue Oct  6 07:31:06 2009
From: todd.baumgartner at gmail.com (Todd Baumgartner)
Date: Tue, 6 Oct 2009 10:31:06 -0400
Subject: [Freeswitch-users] Cannot connect to ODBC driver/database
	freeswitchdb
In-Reply-To: <a60fed5e0910060311w4ceb749cu520c877bb47c47b8@mail.gmail.com>
References: <a60fed5e0910060311w4ceb749cu520c877bb47c47b8@mail.gmail.com>
Message-ID: <dc8dcf060910060731j2e3961a1ra65f580101ec411d@mail.gmail.com>

Ahmed,

I believe you need to specify the database name as it is configured in the
odbc.ini

I am assuming you have something like this in your nibblebill.conf.xml

    <param name="db_dsn" value="freeswitchdb"/>

Try changing it to this as it is named in the odbc.ini:

    <param name="db_dsn" value="MySQL-freeswitch"/>


Thanks,

Todd



On Tue, Oct 6, 2009 at 6:11 AM, Ahmed Munir <ahmedmunir007 at gmail.com> wrote:

> Hi,
> I've installed FS on Ubuntu 9.04 and I want to run mod_nibbles on it. I
> follow the steps to configure my ODBC connection with MySQL as explained in
> wiki (mod_nibbles and mod_spidermonkey). But  FS, unable to connect it. The
> error I got is listed below when I restart FS,
>
> 2009-10-06 15:47:21.164590 [ERR] switch_odbc.c:188 STATE: IM002 CODE 0
> ERROR: [unixODBC][Driver Manager]Data source name not found, and no default
> driver specified
> 2009-10-06 15:47:21.164617 [CRIT] mod_nibblebill.c:221 Cannot connect to
> ODBC driver/database freeswitchdb (user: root / pass password)
> 2009-10-06 15:47:21.164650 [CONSOLE] switch_loadable_module.c:889
> Successfully Loaded [mod_nibblebill]
> 2009-10-06 15:47:21.164664 [NOTICE] switch_loadable_module.c:248 Adding
> Application 'nibblebill'
> 2009-10-06 15:47:21.164710 [NOTICE] switch_loadable_module.c:270 Adding API
> Function 'nibblebill'
>
> But when I use isql it accepts my odbc connection i.e. isql
> MySQL-freeswitch
>
> I'm listing my settings of odbc.ini and odbcinst.ini as listed below;
>
> odbc.ini
> --------------
> [MySQL-freeswitch]
> Driver       = MySQL
> #Driver       = /usr/lib/odbc/libodbcmyS.so
> Description  = Connector/ODBC Driver DSN With FreeSwitch
> SERVER       = localhost
> PORT         = 3306
> USER         = root
> Password     = password
> Database     = freeswitchdb
>
> odbcinst.ini
> -------------------
> [MySQL]
> Description    = ODBC for MySQL
> Driver         = /usr/lib/odbc/libmyodbc.so
> Setup          = /usr/lib/odbc/libodbcmyS.so
> FileUsage      = 1
>
>
> odbc.ini and odbcinst.ini are located at /etc/. Even I set my odbc
> connection setting as I provide with this link;
> http://dev.mysql.com/doc/refman/5.0/en/connector-odbc-configuration-dsn-unix.html
>
> But unfortunately my problem is unresolved then.
>
>
> Kindly advise me, how can I resolve this problem?
>
> --
> Regards,
>
> Ahmed Munir
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Tue Oct  6 08:12:58 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 6 Oct 2009 11:12:58 -0400
Subject: [Freeswitch-users] Basic compile question.
In-Reply-To: <6.2.3.4.2.20091005232815.03231998@pop-server.tx.rr.com>
References: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>
	<e403cddca1c90c83ce4fb6f85b026e63.squirrel@correo.nodo50.org>
	<65d96fc80910020532i4ba33353g7ac141624a4bb89e@mail.gmail.com>
	<C1AD67AA-F969-4BA1-8415-5E381BDF18FA@jerris.com>
	<90823c940910051257u403108afg3478c4c13ab26283@mail.gmail.com>
	<7e2ac3270910051319xfdfc9d3wa583c2da7e932339@mail.gmail.com>
	<6.2.3.4.2.20091005154158.02b632a8@pop-server.tx.rr.com>
	<4C985F33-8B02-4F8F-95B5-36BA6B40690F@freeswitch.org>
	<6.2.3.4.2.20091005232815.03231998@pop-server.tx.rr.com>
Message-ID: <B5B67EEE-6103-4BEF-A064-A2D6110C77E4@jerris.com>

As I said in the duplicate thread, the voip codecs issue has been  
resolved in trunk, I had a change 1/2 done waiting for testing and it  
is now complete.

Mike

On Oct 6, 2009, at 12:30 AM, David Clark wrote:

> No I found the one header.  I added it to the include list for the
> project.  It included something else, added that.  etc.  Basically I
> think I am going to need the VC 2008
> compiler  and to use the other project file.
>
> At 04:41 PM 10/5/2009, Brian West wrote:
>> Have you updated today?
>>
>> /b
>>
>> On Oct 5, 2009, at 3:49 PM, David Clark wrote:
>>
>>> Any idea what is up?
>>




From mike at jerris.com  Tue Oct  6 08:15:10 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 6 Oct 2009 11:15:10 -0400
Subject: [Freeswitch-users] stun not working in fs 1.0.4?
In-Reply-To: <4ACAEFDF.4030604@gmail.com>
References: <4AC9E9EE.8090805@gmail.com>	<7C489261-F5F3-4B57-B807-21EC4A3947A2@freeswitch.org>	<4ACA31ED.1000002@gmail.com>
	<4AA6F4E2-2138-439C-86E8-7554A5D110D4@freeswitch.org>
	<4ACAEFDF.4030604@gmail.com>
Message-ID: <177D2C13-C1CC-4C00-BD6B-18A8F2F69366@jerris.com>

I am not sure what you mean, do you think that fixes from today should  
somehow go somewhere else before we do a release?


On Oct 6, 2009, at 3:21 AM, Vladimir Elizarov wrote:

> Brian West ?????:
>> Because TRUNK is stable... its only fixes going in usually and if
>> things do break they don't stay broken for long.
>>
>> Ask anyone our trunk is more table then most commercial products.
>>
> This separation of the branches a very bad influence on the packaging.
> That is gathered deb-package trunk 15094. Man found in the trunk bug.
> Must again rebuild the package from the new trunk...
>> /b
>>
>> On Oct 5, 2009, at 12:50 PM, Vladimir Elizarov wrote:
>>
>>
>>> Ok. Brian, why fs no two branches of the stable and trunk?
>>>




From mike at jerris.com  Tue Oct  6 08:19:27 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 6 Oct 2009 11:19:27 -0400
Subject: [Freeswitch-users] overriding conference preference
In-Reply-To: <a44688a0910060805o49d549b8vbd4d545b229c75d@mail.gmail.com>
References: <a44688a0910050749s1ae1f9a4vba96477d159d377f@mail.gmail.com>
	<E773DB8B-AC74-49DE-9B38-7AFC441882F5@jerris.com>
	<a44688a0910060805o49d549b8vbd4d545b229c75d@mail.gmail.com>
Message-ID: <CF2AF67F-68EF-4AD5-9E83-7C53B054379A@jerris.com>

There are xml hooks in several of the embedded including mod_perl and  
mod_lua.  Not sure how well those scale as I have not seen anyone use  
them heavily.

Mike

On Oct 6, 2009, at 11:05 AM, Woody Dickson wrote:

> Hi,
>
> Is there anyway of using curl without having to setup a standalone  
> http service?  Is it possible to generate curl xml using scripts?
>
> woody
>
> On Mon, Oct 5, 2009 at 11:05 PM, Michael Jerris <mike at jerris.com>  
> wrote:
>
> On Oct 5, 2009, at 10:49 AM, Woody Dickson wrote:
>
> > Is is possible to override any of the setting specified in the
> > conference profile?
>
> Just the flags you can pass per user such as pin and mute
>
> >
> > What I want to do is to have a default profile, and be able to
> > modify certain fields if necessary in the dialplan.
> >
> >
> > Alternatively, I would prefer to have a dynamic profile setting for
> > the conference to obtain those parameters from odbc.
>
> you can do this with mod_xml_curl

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From brian at freeswitch.org  Tue Oct  6 07:31:06 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 6 Oct 2009 09:31:06 -0500
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <1b46b4e80910052142jaa09f08l72e5e4b2b24d10fc@mail.gmail.com>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<191c3a030909291122t770af037j5b1068197b31de87@mail.gmail.com>
	<1b46b4e80909291542k71440965g75bb3bfcf38e8005@mail.gmail.com>
	<1b46b4e80910011229k31f7233du10b0fc661730d3f3@mail.gmail.com>
	<F86F950D-9106-49C9-81CA-38B3EB2072F4@jerris.com>
	<1b46b4e80910041519h3581d3d2x7b84041e8c18588@mail.gmail.com>
	<C887D143-F7A1-44B9-B8A2-D34A4B4E582E@jerris.com>
	<1b46b4e80910041609x2f41ceb9u142cb484951a21ed@mail.gmail.com>
	<87f2f3b90910050920w4d17033fl4c386a0018107bdf@mail.gmail.com>
	<1b46b4e80910052119me12a0d9y4653edd831f9555e@mail.gmail.com>
	<1b46b4e80910052142jaa09f08l72e5e4b2b24d10fc@mail.gmail.com>
Message-ID: <083DBCB9-48B2-4E61-81B5-B7D00A42EAB5@freeswitch.org>

This looks like you have an ALG messing with packets... notice it says  
rport 5080 but we are sending to 5060.

/b

On Oct 5, 2009, at 11:42 PM, Nicolas Brenner wrote:

> Ignore my previous email, the traces were incomplete, got much  
> better (and complete) traces with ngrep (found a suggestion from  
> Brian in the list archive, thanks!)
>
> The gateway that registers:
>
> - http://pastebin.freeswitch.org/10607
>
> The one that doesn't:
>
> - http://pastebin.freeswitch.org/10608
>
>
> Thanks again for your time and help!
>
>
> Nicolas
>
>
> On Tue, Oct 6, 2009 at 12:19 AM, Nicolas Brenner <nicolas at medularis.com 
> > wrote:
> There was no sane way of doing that, so I ended up logging the trace  
> from the cli.

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From larclap at yahoo.com  Tue Oct  6 08:41:29 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Tue, 6 Oct 2009 08:41:29 -0700
Subject: [Freeswitch-users] Upgrading causes no answer
Message-ID: <00ea01ca469b$7929faf0$6b7df0d0$@com>

http://pastebin.freeswitch.org/10612

 

I having been running v14996 OK for a while. I have upgraded a couple of
times after, but every time, an inbound call is hung up on. The only thing
that has changed is the upgrade. This morning I upgraded to v15098 and the
problem persists.

 

I believe it has to do with a lua script I use for inbound calls. Reading
from the log, just after the script is launched, the following two lines
appear:

 

switch_cpp.cpp:1116  session not ready

switch_cpp.cpp:925 destroy/unlink session from object

 

Has something changed recently with lua processing? Is there something in
the lua script which is causing the problem?

 

I would appreciate any help.

 

Thanks, Lars

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From msc at freeswitch.org  Tue Oct  6 09:06:02 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 6 Oct 2009 09:06:02 -0700
Subject: [Freeswitch-users] oz debug says error
In-Reply-To: <7d79b3930910052106n59cf9cbema68e4d6ccc274034@mail.gmail.com>
References: <25749736.post@talk.nabble.com>
	<20091005124858.81857415806@mail.cune.org>
	<7d79b3930910052106n59cf9cbema68e4d6ccc274034@mail.gmail.com>
Message-ID: <87f2f3b90910060906y4b087d7coa91de38d78a3456b@mail.gmail.com>

Pastebin your openzap.conf file. Also, is this Sangoma or zaptel-based
hardware? If it's Sangoma, pastebin your wanpipe1.conf file. If zaptel,
please paste your zaptel.conf file.

-MC

On Mon, Oct 5, 2009 at 9:06 PM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Openzap.conf.xml
> <configuration name="openzap.conf" description="OpenZAP Configuration">
>         <settings>
>                 <param name="debug" value="7"/>
>         </settings>
>         <libpri_spans>
>                 <span name="PRI_1">
>                         <param name="node" value="cpe"/>
>                         <param name="switch" value="euroisdn"/>
>                         <param name="dialplan" value="XML"/>
>                         <param name="context" value="default"/>
>                 </span>
>         </libpri_spans>
> </configuration>
>
> Output of oz list in fs_cli
>
> span: 1 (PRI_1)
> type: isdn
> chan_count: 47
> dialplan: XML
> context: default
> dial_regex:
> fail_dial_regex:
> hold_music:
> analog_options none
>
>
> freeswitch.log
>
> http://pastebin.freeswitch.org/10604
>
>
>
>
>
> On Mon, Oct 5, 2009 at 6:18 PM, <Russell.Mosemann at cune.org> wrote:
>
>> lakshmanan <lakindia89 at gmail.com> said:
>> > Thanks for pointing that.
>> > I also tried that.
>> > But in that case, I'm not able to make a call through openzap.
>>
>> What is in openzap.conf.xml? If you start fs_cli and enter "oz list",
>> what does it show? Copy the ozmod lines from freeswitch.log to
>> pastebin.freeswitch.org and post the link here so that we can see what
>> openzap does when freeswitch starts.
>>
>> --
>> Russell Mosemann
>>
>>
>>
>> ________________________________________________________
>> Concordia University, Nebraska
>> See http://www.cune.edu/ for the latest news and events!
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Oct  6 09:15:42 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 6 Oct 2009 09:15:42 -0700
Subject: [Freeswitch-users] Wiki Info: Multi Homed Tutorial
In-Reply-To: <a87d43fd0910052120l76da1536ra92766174dc938d3@mail.gmail.com>
References: <87f2f3b90910051527x638f55b8hda75d5302267c251@mail.gmail.com>
	<a87d43fd0910052120l76da1536ra92766174dc938d3@mail.gmail.com>
Message-ID: <87f2f3b90910060915g167bafa7jd8b0ed5e1b3b2b20@mail.gmail.com>

On Mon, Oct 5, 2009 at 9:20 PM, Michael Gende <mgende at gendesign.com> wrote:

> Michael,
>
> Thanks for "wiki-fying" my text-only attempt at some user doc. I should
> have done that for you. I actually have an updated version with many
> corrections and the end tabs filled in. Can you point me to info on how I
> can amend and append what you have kindly put up?
>
> Mike G.
>
> First, go to wiki.freeswitch.org and create a wiki account. Then, go to
the Multi_home_tutorial page and click edit (top of the page). You'll see
that there is wiki markup to learn. If you have questions let me know. Just
edit the wiki text and click Show Preview to see what the real thing looks
like. Then click Save Page to save your changes.

Welcome to the world of MediaWiki and thanks for your help!
-MC
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From msc at freeswitch.org  Tue Oct  6 09:18:32 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 6 Oct 2009 09:18:32 -0700
Subject: [Freeswitch-users] add users dynamically to groups in
	default.xml
In-Reply-To: <f8af5740910060531j58158057h881d9862602d360c@mail.gmail.com>
References: <f8af5740910060531j58158057h881d9862602d360c@mail.gmail.com>
Message-ID: <87f2f3b90910060918q7d5d4d4fu454031120d098174@mail.gmail.com>

On Tue, Oct 6, 2009 at 5:31 AM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

> Hi,
> Can any one tell me how to add users dynamically to groups in default.xml,
> with out restart the freeswitch.
>
> Thanks
> Srinivasula Reddy K
>
>
Changes to the dialplan xml files get updated with a simple 'reloadxml'
command at the CLI. If you want truly dynamic configuration then you'll want
to investigate mod_xml_curl.
-MC
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From msc at freeswitch.org  Tue Oct  6 09:40:33 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 6 Oct 2009 09:40:33 -0700
Subject: [Freeswitch-users] Outgoing via openzap is not working
In-Reply-To: <7d79b3930910060330s77a2bbbv88d8014b73f91663@mail.gmail.com>
References: <7d79b3930910060330s77a2bbbv88d8014b73f91663@mail.gmail.com>
Message-ID: <87f2f3b90910060940k543bfd35rc7d6b2505b6d853@mail.gmail.com>

On Tue, Oct 6, 2009 at 3:30 AM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Hi I'm using freeswitch1.0.4. This post is moreover similar to my previous
> post.
> When I make an outgoing call, it is saying INVALID_IE_CONTENTS.
> Here are the details.
> openzap.conf.xml
>
> <configuration name="openzap.conf" description="OpenZAP Configuration">
> <settings>
> <param name="debug" value="7"/>
> </settings>
> <libpri_spans>
> <span name="PRI_1">
> <param name="node" value="cpe"/>
> <param name="switch" value="euroisdn"/>
> <param name="dialplan" value="XML"/>
> <param name="context" value="default"/>
> </span>
> </libpri_spans>
> </configuration>
>
> openzap.conf
> [span zt PRI_1]
> trunk_type => e1
> b-channel => 1:1-15
> d-channel=> 1:16
> b-channel => 1:17-31
> oz libpri debug 1 all
>
> API CALL [oz(libpri debug 1 all)] output:
> src/ozmod/ozmod_libpri/ozmod_libpri.c: +OK debug set.
>
> oz list
>
> API CALL [oz(list)] output:
> +OK
> span: 1 (PRI_1)
> type: isdn
> chan_count: 47
> dialplan: XML
> context: default
> dial_regex:
> fail_dial_regex:
> hold_music:
> analog_options none
>
> Freeswitch startup log:
> http://pastebin.freeswitch.org/10609
> After saying originate openzap/1/1/9952248266 openzap/1/1/9952248266
>

Can you confirm your originate line? What you type above is incorrect
syntax. Correct syntax:

openzap/1/a/99522448266 1234

Where the 'a' means select first available b chan and the 1234 is just an
extension number. You can put any number that works for your dialplan. The
originate command originates a call leg and then connects it to a second
call leg in the dialplan. Make sure that you are using this command properly
before continuing your debugging.
-MC



> http://pastebin.freeswitch.org/10610
>
> Please help me to solve this.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Oct  6 09:45:13 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 6 Oct 2009 09:45:13 -0700
Subject: [Freeswitch-users] overriding conference preference
In-Reply-To: <a44688a0910060805o49d549b8vbd4d545b229c75d@mail.gmail.com>
References: <a44688a0910050749s1ae1f9a4vba96477d159d377f@mail.gmail.com>
	<E773DB8B-AC74-49DE-9B38-7AFC441882F5@jerris.com>
	<a44688a0910060805o49d549b8vbd4d545b229c75d@mail.gmail.com>
Message-ID: <87f2f3b90910060945ge3f2a44ocf836b83b038736c@mail.gmail.com>

On Tue, Oct 6, 2009 at 8:05 AM, Woody Dickson <woodydickson at gmail.com>wrote:

> Hi,
>
> Is there anyway of using curl without having to setup a standalone http
> service?  Is it possible to generate curl xml using scripts?
>
> woody
>
>
Check out this page on the wiki:
http://wiki.freeswitch.org/wiki/Mod_xml_curl#Two_types_of_data_sources

Depending on your needs you might find an option that helps, such as using a
static file (which you can update in a 3rd party process).
-MC


> On Mon, Oct 5, 2009 at 11:05 PM, Michael Jerris <mike at jerris.com> wrote:
>
>>
>> On Oct 5, 2009, at 10:49 AM, Woody Dickson wrote:
>>
>> > Is is possible to override any of the setting specified in the
>> > conference profile?
>>
>> Just the flags you can pass per user such as pin and mute
>>
>> >
>> > What I want to do is to have a default profile, and be able to
>> > modify certain fields if necessary in the dialplan.
>> >
>> >
>> > Alternatively, I would prefer to have a dynamic profile setting for
>> > the conference to obtain those parameters from odbc.
>>
>> you can do this with mod_xml_curl
>>
>> Mike
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ChristianDamianidis at globalive.com  Tue Oct  6 09:55:05 2009
From: ChristianDamianidis at globalive.com (Christian Damianidis)
Date: Tue, 6 Oct 2009 12:55:05 -0400
Subject: [Freeswitch-users] mod_xml_curl http POST is inconsistent/bug
In-Reply-To: <191c3a030910060801x3ed48e98x6d69b53c6beaca69@mail.gmail.com>
References: <66EA3166EB339A4489B06286C0876A8A0C0C1432@mailserv.Globalive.local><276FEE28-70CF-422D-9F6F-7CF747FDDB22@freeswitch.org><66EA3166EB339A4489B06286C0876A8A0C0C1BA9@mailserv.Globalive.local>
	<191c3a030910060801x3ed48e98x6d69b53c6beaca69@mail.gmail.com>
Message-ID: <66EA3166EB339A4489B06286C0876A8A0C0C1F04@mailserv.Globalive.local>

I've tested this and making the change from ANY to BASIC worked. Thanks
for the help. 

It no longer sends the initial post without auth.

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Tuesday, October 06, 2009 11:02 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_xml_curl http POST is
inconsistent/bug

 

My guess is that we configure the curl to support the full range of http
auth methods.
Some of them like Digest require a challenge and realm etc so it's
probably asking without auth header because it cannot create one until
it gets that data.  In the case of Basic you can send the login and pass
right away but it does not know in advance that it will be basic.

Here is a snippet from the libcurl api docs:
------------------------------------------------------------------------
------------------------------------------------------------------------
-------------

Both these options allow you to set multiple types (by ORing them
together), to make libcurl pick the most secure one out of the types the
server/proxy claims to support. This method does however add a
round-trip since libcurl must first ask the server what it supports: 

 curl_easy_setopt(easyhandle, CURLOPT_HTTPAUTH,
CURLAUTH_DIGEST|CURLAUTH_BASIC); 

------------------------------------------------------------------------
------------------------------------------------------------------------
-------------

So my guess is that if we set it to only support basic, then it would
work how you expect so if you want to test it for me I can make it into
a parameter.

edit:
/usr/src/freeswitch.trunk/src/mod/xml_int/mod_xml_curl/mod_xml_curl.c
line 220
change

curl_easy_setopt(curl_handle, CURLOPT_HTTPAUTH, CURLAUTH_ANY);

to

curl_easy_setopt(curl_handle, CURLOPT_HTTPAUTH, CURLAUTH_BASIC);


If this works i'll think about exposing the auth methods so you can
choose them in the config.



On Tue, Oct 6, 2009 at 9:39 AM, Christian Damianidis
<ChristianDamianidis at globalive.com> wrote:

This web request goes to a server running IIS on Windows Server 2003.

 

From: Brian West [mailto:brian at freeswitch.org] 
Sent: Monday, October 05, 2009 5:43 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_xml_curl http POST is
inconsistent/bug

 

Are you using something other than apache?

 

/b

 

On Oct 5, 2009, at 1:25 PM, Christian Damianidis wrote:

 

 

The inconsistent POST request sent by the module causes freeswitch to
hang for 1-2 minutes during start-up.

 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:213-799-1400

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From tculjaga at gmail.com  Tue Oct  6 10:37:08 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 6 Oct 2009 19:37:08 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
Message-ID: <65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>

hello guys,


i was playing with mod_opal to see if i can make it working ... well it
seems SIP-H323 interworking is not tuned at all.

I have a call from a registered sip user (1001) to PSTN via mod_opal


<include>
  <extension name="EMERGENCY">
    <condition field="destination_number" expression="^0(112|9[23456])$">
      <action application="set"
data="effective_caller_id_number=1001282122"/>
      <action application="set" data="NCX_IP=10.4.4.254"/>
      <action application="set" data="call_timeout=30"/>
      <action application="set" data="hangup_after_bridge=true"/>

      <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
    </condition>
  </extension>

  <extension name="SPECIAL_SERVICES">
    <condition field="destination_number" expression="^0(9[01789]\d{3,4})$">
      <action application="set"
data="effective_caller_id_number=1001282122"/>
      <action application="set" data="NCX_IP=10.4.4.254"/>
      <action application="set" data="call_timeout=30"/>
      <action application="set" data="hangup_after_bridge=true"/>

      <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
    </condition>
  </extension>

  <extension name="ENYTHING_ELSE">
    <condition field="destination_number"
expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
      <action application="set"
data="effective_caller_id_number=1001282122"/>
      <action application="set" data="NCX_IP=10.4.4.254"/>
      <action application="set" data="call_timeout=30"/>
      <action application="set" data="hangup_after_bridge=true"/>

      <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
    </condition>
  </extension>
</include>



One of the many issues i sow is that FS connects the call on SIP leg before
it actually receives H.225 connect from H323 leg... as it is configured to
send 200 OK on the 1st H.225 message containing a FastStart element/OLC.


Attached is the tcpdump i took on FS machine... just use this filter: "h225
or h245 or q931 or sip"
Also, you can check the attac CDR,,,, this is an unanswered call i placed to
PSTN and FS billed it 23 seconds.



Can anyone tell where i can do correct SIP - H323 message mappings to avoid
this?



T.
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From dome at tel.co.th  Tue Oct  6 10:51:27 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Wed, 7 Oct 2009 00:51:27 +0700
Subject: [Freeswitch-users] Memory Question ?
Message-ID: <8ccbff060910061051p12b39c67kddf201113dda7c48@mail.gmail.com>

Dear All,
             why freeswitch use more memory after send and receuve
call. i attach htop capture screen.
you can compare to asterisk it use 0.7% for long time. but FS use 7.7%
(start from 1.2%)  after running about 4 hr.

Best Regards.

Dome C.
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From quentusrex at gmail.com  Tue Oct  6 10:57:40 2009
From: quentusrex at gmail.com (William King)
Date: Tue, 06 Oct 2009 10:57:40 -0700
Subject: [Freeswitch-users] Memory Question ?
In-Reply-To: <8ccbff060910061051p12b39c67kddf201113dda7c48@mail.gmail.com>
References: <8ccbff060910061051p12b39c67kddf201113dda7c48@mail.gmail.com>
Message-ID: <4ACB8514.2050201@gmail.com>

What operating system?

-William King

Dome Charoenyost wrote:
> Dear All,
>              why freeswitch use more memory after send and receuve
> call. i attach htop capture screen.
> you can compare to asterisk it use 0.7% for long time. but FS use 7.7%
> (start from 1.2%)  after running about 4 hr.
>
> Best Regards.
>
> Dome C.
>   
>
> ------------------------------------------------------------------------
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From brian at freeswitch.org  Tue Oct  6 11:07:05 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 6 Oct 2009 13:07:05 -0500
Subject: [Freeswitch-users] Memory Question ?
In-Reply-To: <8ccbff060910061051p12b39c67kddf201113dda7c48@mail.gmail.com>
References: <8ccbff060910061051p12b39c67kddf201113dda7c48@mail.gmail.com>
Message-ID: <DB335ACC-EEC9-41A0-83E4-8829E12FE6A8@freeswitch.org>

We use memory pools and its not uncommon to use what you displayed.

/b

On Oct 6, 2009, at 12:51 PM, Dome Charoenyost wrote:

> Dear All,
>             why freeswitch use more memory after send and receuve
> call. i attach htop capture screen.
> you can compare to asterisk it use 0.7% for long time. but FS use 7.7%
> (start from 1.2%)  after running about 4 hr.
>
> Best Regards.
>
> Dome C.
> <Screenshot-1.png>_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From msc at freeswitch.org  Tue Oct  6 11:12:02 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 6 Oct 2009 11:12:02 -0700
Subject: [Freeswitch-users] Memory Question ?
In-Reply-To: <8ccbff060910061051p12b39c67kddf201113dda7c48@mail.gmail.com>
References: <8ccbff060910061051p12b39c67kddf201113dda7c48@mail.gmail.com>
Message-ID: <87f2f3b90910061112h221d37b2p23d82342b9ac9a09@mail.gmail.com>

On Tue, Oct 6, 2009 at 10:51 AM, Dome Charoenyost <dome at tel.co.th> wrote:

> Dear All,
>             why freeswitch use more memory after send and receuve
> call. i attach htop capture screen.
> you can compare to asterisk it use 0.7% for long time. but FS use 7.7%
> (start from 1.2%)  after running about 4 hr.
>
> Best Regards.
>
> Dome C.
>
> Do you have lots of modules loaded? The default config has lots of stuff
loaded. If memory usage is a big issue then check your modules.conf.xml file
and don't load unnecessary modules. Of course, you could use Asterisk but
then you save memory at the expense of fewer features and more crashes...

-MC
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From dome at tel.co.th  Tue Oct  6 11:43:10 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Wed, 7 Oct 2009 01:43:10 +0700
Subject: [Freeswitch-users] Memory Question ?
In-Reply-To: <4ACB8514.2050201@gmail.com>
References: <8ccbff060910061051p12b39c67kddf201113dda7c48@mail.gmail.com>
	<4ACB8514.2050201@gmail.com>
Message-ID: <8ccbff060910061143gc0d0edcl46a2b5274a74c876@mail.gmail.com>

Debian Squeeze  i386 32bit
And Debian Lenny are same

Dome C.

2009/10/7 William King <quentusrex at gmail.com>:
> What operating system?
>
> -William King
>
> Dome Charoenyost wrote:
>> Dear All,
>> ? ? ? ? ? ? ?why freeswitch use more memory after send and receuve
>> call. i attach htop capture screen.
>> you can compare to asterisk it use 0.7% for long time. but FS use 7.7%
>> (start from 1.2%) ?after running about 4 hr.
>>
>> Best Regards.
>>
>> Dome C.
>>
>>
>> ------------------------------------------------------------------------
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From dome at tel.co.th  Tue Oct  6 11:50:25 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Wed, 7 Oct 2009 01:50:25 +0700
Subject: [Freeswitch-users] Memory Question ?
In-Reply-To: <87f2f3b90910061112h221d37b2p23d82342b9ac9a09@mail.gmail.com>
References: <8ccbff060910061051p12b39c67kddf201113dda7c48@mail.gmail.com>
	<87f2f3b90910061112h221d37b2p23d82342b9ac9a09@mail.gmail.com>
Message-ID: <8ccbff060910061150h2ca1c495u339e2ee046b8c21a@mail.gmail.com>

2009/10/7 Michael Collins <msc at freeswitch.org>:
>
>
> On Tue, Oct 6, 2009 at 10:51 AM, Dome Charoenyost <dome at tel.co.th> wrote:
>>
>> Dear All,
>> ? ? ? ? ? ? why freeswitch use more memory after send and receuve
>> call. i attach htop capture screen.
>> you can compare to asterisk it use 0.7% for long time. but FS use 7.7%
>> (start from 1.2%) ?after running about 4 hr.
>>
>> Best Regards.
>>
>> Dome C.
>>
> Do you have lots of modules loaded? The default config has lots of stuff
I use LCR , nibblebill , odbcquery.
> loaded. If memory usage is a big issue then check your modules.conf.xml file
My question is why FS need more memory when start about 4 hr. is is
normal or got problem ?
So what's happen when running about 1 weeks ?
some time my FS crash and i report bug in jira. and bug maintainer ask
me is memory run out ?
i think yes.

> and don't load unnecessary modules. Of course, you could use Asterisk but
> then you save memory at the expense of fewer features and more crashes...
No. i move all from asterisk to FS right now :)

>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From quentusrex at gmail.com  Tue Oct  6 11:50:49 2009
From: quentusrex at gmail.com (William King)
Date: Tue, 06 Oct 2009 11:50:49 -0700
Subject: [Freeswitch-users] Memory Question ?
In-Reply-To: <8ccbff060910061143gc0d0edcl46a2b5274a74c876@mail.gmail.com>
References: <8ccbff060910061051p12b39c67kddf201113dda7c48@mail.gmail.com>	<4ACB8514.2050201@gmail.com>
	<8ccbff060910061143gc0d0edcl46a2b5274a74c876@mail.gmail.com>
Message-ID: <4ACB9189.5070504@gmail.com>

Which modules do you have loaded?

-William King

Dome Charoenyost wrote:
> Debian Squeeze  i386 32bit
> And Debian Lenny are same
>
> Dome C.
>
> 2009/10/7 William King <quentusrex at gmail.com>:
>   
>> What operating system?
>>
>> -William King
>>
>> Dome Charoenyost wrote:
>>     
>>> Dear All,
>>>              why freeswitch use more memory after send and receuve
>>> call. i attach htop capture screen.
>>> you can compare to asterisk it use 0.7% for long time. but FS use 7.7%
>>> (start from 1.2%)  after running about 4 hr.
>>>
>>> Best Regards.
>>>
>>> Dome C.
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From dome at tel.co.th  Tue Oct  6 11:57:49 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Wed, 7 Oct 2009 01:57:49 +0700
Subject: [Freeswitch-users] Memory Question ?
In-Reply-To: <4ACB9189.5070504@gmail.com>
References: <8ccbff060910061051p12b39c67kddf201113dda7c48@mail.gmail.com>
	<4ACB8514.2050201@gmail.com>
	<8ccbff060910061143gc0d0edcl46a2b5274a74c876@mail.gmail.com>
	<4ACB9189.5070504@gmail.com>
Message-ID: <8ccbff060910061157g2220a8fay1585593303bf593e@mail.gmail.com>

2009/10/7 William King <quentusrex at gmail.com>:
> Which modules do you have loaded?
default config and nibllebill , lcr , odbcquery

>
> -William King
>
> Dome Charoenyost wrote:
>> Debian Squeeze ?i386 32bit
>> And Debian Lenny are same
>>
>> Dome C.
>>
>> 2009/10/7 William King <quentusrex at gmail.com>:
>>
>>> What operating system?
>>>
>>> -William King
>>>
>>> Dome Charoenyost wrote:
>>>
>>>> Dear All,
>>>> ? ? ? ? ? ? ?why freeswitch use more memory after send and receuve
>>>> call. i attach htop capture screen.
>>>> you can compare to asterisk it use 0.7% for long time. but FS use 7.7%
>>>> (start from 1.2%) ?after running about 4 hr.
>>>>
>>>> Best Regards.
>>>>
>>>> Dome C.
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From xengelpublicx at gmail.com  Tue Oct  6 12:06:01 2009
From: xengelpublicx at gmail.com (Vladimir Elizarov)
Date: Tue, 06 Oct 2009 23:06:01 +0400
Subject: [Freeswitch-users] linksys spa-3102 fax t38 + freeswitch media proxy
Message-ID: <4ACB9519.9000805@gmail.com>

Hello.

I want to set up faxing via the gateway linksys spa-3102 (with support
for t38) via SIP.

SIP-client -> linksys spa3102 -> fs -> provider

    <gateway name="callwithus">
        ....
        <param name="inbound-late-negotiation" value="true"/>
        <param name="stun-enabled" value="true"/>
    </gateway>

<extension name="Fax">
    <condition field="destination_number" expression="^(30)(\d{11})$">
            <action application="set" data="proxy_media=true"/>
            <action application="set" data="bypass_media=false"/>
            <action application="bridge"
data="sofia/gateway/callwithus/*4087$2 at sip.callwithus.com"/>
        </condition>
</extension>

I make a call, press start on the fax. Fax not sent. Get this in the log
(UA freeswitch not trunk-12805):

   ------------------------------------------------------------------------
   INVITE sip:*408774957558438 at sip.callwithus.com SIP/2.0
   Via: SIP/2.0/UDP 77.239.230.202:5080;rport;branch=z9hG4bK8yjrHeUprX2Hr
   Max-Forwards: 99
   From: "Mihailova Ludmila"
<sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
   To: <sip:*408774957558438 at sip.callwithus.com>
   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
   CSeq: 121302430 INVITE
   Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 250
   Remote-Party-ID: "Mihailova Ludmila"
<sip:214 at sip.callwithus.com>;party=calling;screen=yes;privacy=off
   
   v=0
   o=- 216896551 216896551 IN IP4 192.168.50.51
   s=-
   c=IN IP4 77.239.230.202
   t=0 0
   m=audio 32404 RTP/AVP 0 100 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:100 NSE/8000
   a=fmtp:100 192-193
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=ptime:30

   ------------------------------------------------------------------------
send 1333 bytes to udp/[38.99.70.232]:5060 at 15:05:34.485226:
   ------------------------------------------------------------------------
   INVITE sip:*408774957558438 at sip.callwithus.com SIP/2.0
   Via: SIP/2.0/UDP 77.239.230.202:5080;rport;branch=z9hG4bK97BHK9BtN6r4K
   Max-Forwards: 99
   From: "Mihailova Ludmila"
<sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
   To: <sip:*408774957558438 at sip.callwithus.com>
   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
   CSeq: 121302431 INVITE
   Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
   Expires: 3600
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 250
   Remote-Party-ID: "Mihailova Ludmila"
<sip:214 at sip.callwithus.com>;party=calling;screen=yes;privacy=off
   
   v=0
   o=- 216896551 216896551 IN IP4 192.168.50.51
   s=-
   c=IN IP4 77.239.230.202
   t=0 0
   m=audio 32404 RTP/AVP 0 100 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:100 NSE/8000
   a=fmtp:100 192-193
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=ptime:30
   ------------------------------------------------------------------------
   SIP/2.0 100 Giving a try
   Via: SIP/2.0/UDP
192.168.50.10:5080;rport=5080;branch=z9hG4bK97BHK9BtN6r4K
   From: "Mihailova Ludmila"
<sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
   To: <sip:*408774957558438 at sip.callwithus.com>
   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
   CSeq: 121302431 INVITE
   Server: CWU SIP GW
   Content-Length: 0
   
   ------------------------------------------------------------------------
recv 836 bytes from udp/[38.99.70.232]:5060 at 15:05:49.641312:
   ------------------------------------------------------------------------
   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP
192.168.50.10:5080;rport=5080;branch=z9hG4bK97BHK9BtN6r4K
   Record-Route:
<sip:38.99.70.232;lr=on;ftag=HXjrDKpHHX8ZN;did=e8e.089fc2b6>
   From: "Mihailova Ludmila"
<sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
   To: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
   CSeq: 121302431 INVITE
   User-Agent: CWU SIP-GW
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
   Supported: replaces
   Contact: <sip:*408774957558438 at 204.74.213.1:5062>
   Content-Type: application/sdp
   Content-Length: 209
   
   v=0
   o=root 31797 31797 IN IP4 4.55.17.66
   s=session
   c=IN IP4 4.55.17.66
   t=0 0
   m=audio 14104 RTP/AVP 0 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   a=sendrecv
   ------------------------------------------------------------------------
v=0
o=root 31797 31797 IN IP4 4.55.17.66
s=session
c=IN IP4 4.55.17.66
t=0 0
m=audio 14104 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

2009-10-06 19:03:31.112988 [NOTICE] sofia.c:3449 Pre-Answer
sofia/external/*408774957558438 at sip.callwithus.com!
2009-10-06 19:03:31.112988 [DEBUG] switch_channel.c:1822 Send signal
sofia/internal/214 at pbx0.tssec.lan [BREAK]
2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2095 Set Codec
sofia/external/*408774957558438 at sip.callwithus.com PROXY/8000 20 ms 160
samples
2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2372 PROXY AUDIO RTP
[sofia/external/*408774957558438 at sip.callwithus.com]
192.168.50.10:32404->4.55.17.66:14104 codec: 0 ms: 20
2009-10-06 19:03:31.112988 [DEBUG] switch_rtp.c:1163 Not using a timer
2009-10-06 19:03:31.112988 [DEBUG] switch_ivr_originate.c:2154
sofia/internal/214 at pbx0.tssec.lan receive message [PROGRESS]
2009-10-06 19:03:31.112988 [INFO] switch_ivr_originate.c:2154 Sending
early media
2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:1322
sofia/internal/214 at pbx0.tssec.lan Patched SDP
---
v=0
o=root 31797 31797 IN IP4 4.55.17.66
s=session
c=IN IP4 4.55.17.66
t=0 0
m=audio 14104 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

+++
v=0
o=root 31797 31797 IN IP4 4.55.17.66
s=session
c=IN IP4 192.168.50.10
t=0 0
m=audio 32698 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2372 PROXY AUDIO RTP
[sofia/internal/214 at pbx0.tssec.lan]
192.168.50.10:32698->192.168.50.51:16474 codec: 0 ms: 30
2009-10-06 19:03:31.112988 [DEBUG] switch_rtp.c:1163 Not using a timer
2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2508 Set comfort noise
payload to 13
2009-10-06 19:03:31.112988 [DEBUG] sofia.c:3462
sofia/internal/214 at pbx0.tssec.lan receive message [PROGRESS]
2009-10-06 19:03:31.116989 [NOTICE] mod_sofia.c:1521 Pre-Answer
sofia/internal/214 at pbx0.tssec.lan!
2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal
sofia/internal/214 at pbx0.tssec.lan [BREAK]
2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_originate.c:2196 Originate
Resulted in Success: [sofia/external/*408774957558438 at sip.callwithus.com]
2009-10-06 19:03:31.116989 [DEBUG] switch_channel.c:182
sofia/external/*408774957558438 at sip.callwithus.com receive message
[AUDIO_SYNC]
2009-10-06 19:03:31.116989 [DEBUG] switch_channel.c:182
sofia/internal/214 at pbx0.tssec.lan receive message [AUDIO_SYNC]
2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_bridge.c:911
sofia/external/*408774957558438 at sip.callwithus.com receive message [BRIDGE]
2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal
sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_bridge.c:918
sofia/internal/214 at pbx0.tssec.lan receive message [BRIDGE]
2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal
sofia/internal/214 at pbx0.tssec.lan [BREAK]
2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_bridge.c:962
(sofia/external/*408774957558438 at sip.callwithus.com) State Change
CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2009-10-06 19:03:31.116989 [DEBUG] sofia.c:3359 Channel
sofia/internal/214 at pbx0.tssec.lan entering state [early][183]
2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal
sofia/internal/214 at pbx0.tssec.lan [BREAK]
2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
2009-10-06 19:03:31.116989 [DEBUG] switch_core_state_machine.c:306
(sofia/external/*408774957558438 at sip.callwithus.com) Running State
Change CS_EXCHANGE_MEDIA
2009-10-06 19:03:31.116989 [DEBUG] switch_core_state_machine.c:343
(sofia/external/*408774957558438 at sip.callwithus.com) State EXCHANGE_MEDIA
2009-10-06 19:03:31.116989 [DEBUG] mod_sofia.c:430 SOFIA LOOPBACK
recv 822 bytes from udp/[38.99.70.232]:5060 at 15:05:58.503701:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP
192.168.50.10:5080;rport=5080;branch=z9hG4bK97BHK9BtN6r4K
   Record-Route:
<sip:38.99.70.232;lr=on;ftag=HXjrDKpHHX8ZN;did=e8e.089fc2b6>
   From: "Mihailova Ludmila"
<sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
   To: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
   CSeq: 121302431 INVITE
   User-Agent: CWU SIP-GW
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
   Supported: replaces
   Contact: <sip:*408774957558438 at 204.74.213.1:5062>
   Content-Type: application/sdp
   Content-Length: 209
   
   v=0
   o=root 31797 31798 IN IP4 4.55.17.66
   s=session
   c=IN IP4 4.55.17.66
   t=0 0
   m=audio 14104 RTP/AVP 0 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   a=sendrecv
   ------------------------------------------------------------------------
send 762 bytes to udp/[38.99.70.232]:5060 at 15:05:58.507702:
   ------------------------------------------------------------------------
   ACK sip:*408774957558438 at 204.74.213.1:5062 SIP/2.0
   Via: SIP/2.0/UDP 192.168.50.10:5080;rport;branch=z9hG4bKaH59m4vXjFFQF
   Route: <sip:38.99.70.232;lr=on;ftag=HXjrDKpHHX8ZN;did=e8e.089fc2b6>
   Max-Forwards: 70
   From: "Mihailova Ludmila"
<sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
   To: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
   CSeq: 121302431 ACK
   Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
   Content-Length: 0
   
   ------------------------------------------------------------------------
2009-10-06 19:03:39.975377 [DEBUG] sofia.c:3359 Channel
sofia/external/*408774957558438 at sip.callwithus.com entering state
[ready][200]
2009-10-06 19:03:39.975377 [DEBUG] switch_channel.c:1935 Send signal
sofia/internal/214 at pbx0.tssec.lan [BREAK]
2009-10-06 19:03:39.975377 [NOTICE] sofia.c:3814 Channel
[sofia/external/*408774957558438 at sip.callwithus.com] has been answered
2009-10-06 19:03:39.975377 [DEBUG] switch_channel.c:182
sofia/external/*408774957558438 at sip.callwithus.com receive message
[AUDIO_SYNC]
2009-10-06 19:03:39.975377 [DEBUG] sofia.c:3819
sofia/internal/214 at pbx0.tssec.lan receive message [ANSWER]
2009-10-06 19:03:39.975377 [DEBUG] sofia_glue.c:1322
sofia/internal/214 at pbx0.tssec.lan Patched SDP
---
v=0
o=root 31797 31797 IN IP4 4.55.17.66
s=session
c=IN IP4 4.55.17.66
t=0 0
m=audio 14104 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

+++
v=0
o=root 31797 31797 IN IP4 4.55.17.66
s=session
c=IN IP4 192.168.50.10
t=0 0
m=audio 32698 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

2009-10-06 19:03:39.975377 [DEBUG] switch_core_session.c:630 Send signal
sofia/internal/214 at pbx0.tssec.lan [BREAK]
2009-10-06 19:03:39.975377 [NOTICE] sofia.c:3819 Channel
[sofia/internal/214 at pbx0.tssec.lan] has been answered
2009-10-06 19:03:39.975377 [DEBUG] switch_channel.c:182
sofia/internal/214 at pbx0.tssec.lan receive message [AUDIO_SYNC]
2009-10-06 19:03:39.975377 [DEBUG] sofia.c:3359 Channel
sofia/internal/214 at pbx0.tssec.lan entering state [completed][200]
2009-10-06 19:03:39.991381 [DEBUG] sofia.c:3359 Channel
sofia/internal/214 at pbx0.tssec.lan entering state [ready][200]
recv 1054 bytes from udp/[38.99.70.232]:5060 at 15:06:05.577607:
   ------------------------------------------------------------------------
   INVITE sip:gw+callwithus at 192.168.50.10:5080;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.0
   Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK41d48157;rport=5062
   From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
   To: "Mihailova Ludmila"
<sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
   Contact: <sip:*408774957558438 at 204.74.213.1:5062>
   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
   CSeq: 102 INVITE
   User-Agent: CWU SIP-GW
   Max-Forwards: 69
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
   Supported: replaces
   X-asterisk-info: SIP re-invite (T38 switchover)
   Content-Type: application/sdp
   Content-Length: 350
   
   v=0
   o=root 31797 31799 IN IP4 32.50.63.192
   s=session
   c=IN IP4 32.50.63.192
   t=0 0
   m=image 14104 udptl t38
   a=T38FaxVersion:0
   a=T38MaxBitRate:9600
   a=T38FaxFillBitRemoval:0
   a=T38FaxTranscodingMMR:0
   a=T38FaxTranscodingJBIG:0
   a=T38FaxRateManagement:transferredTCF
   a=T38FaxMaxBuffer:176
   a=T38FaxMaxDatagram:176
   a=T38FaxUdpEC:t38UDPRedundancy
   ------------------------------------------------------------------------
send 443 bytes to udp/[38.99.70.232]:5060 at 15:06:05.577607:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.0
   Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK41d48157;rport=5062
   From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
   To: "Mihailova Ludmila"
<sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
   CSeq: 102 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
   Content-Length: 0
   
   ------------------------------------------------------------------------
2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3359 Channel
sofia/external/*408774957558438 at sip.callwithus.com entering state
[received][100]
2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3366 Remote SDP:
v=0
o=root 31797 31799 IN IP4 32.50.63.192
s=session
c=IN IP4 32.50.63.192
t=0 0
m=image 14104 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:176
a=T38FaxMaxDatagram:176
a=T38FaxUdpEC:t38UDPRedundancy

2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3683 Passing SDP to other leg.
v=0
o=root 31797 31799 IN IP4 32.50.63.192
s=session
c=IN IP4 32.50.63.192
t=0 0
m=image 14104 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:176
a=T38FaxMaxDatagram:176
a=T38FaxUdpEC:t38UDPRedundancy

2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3696
sofia/internal/214 at pbx0.tssec.lan receive message [MEDIA_REDIRECT]
2009-10-06 19:03:47.049284 [DEBUG] mod_sofia.c:1195
sofia/internal/214 at pbx0.tssec.lan Sending media re-direct:
v=0
o=root 31797 31799 IN IP4 32.50.63.192
s=session
c=IN IP4 32.50.63.192
t=0 0
m=image 14104 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:176
a=T38FaxMaxDatagram:176
a=T38FaxUdpEC:t38UDPRedundancy

2009-10-06 19:03:47.049284 [DEBUG] sofia_glue.c:1101 Remote address:port
[192.168.50.51:16474] has not changed.
2009-10-06 19:03:47.049284 [DEBUG] sofia_glue.c:1322
sofia/internal/214 at pbx0.tssec.lan Patched SDP
---
v=0
o=root 31797 31799 IN IP4 32.50.63.192
s=session
c=IN IP4 32.50.63.192
t=0 0
m=image 14104 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:176
a=T38FaxMaxDatagram:176
a=T38FaxUdpEC:t38UDPRedundancy

+++
v=0
o=root 31797 31799 IN IP4 32.50.63.192
s=session
c=IN IP4 192.168.50.10
t=0 0
m=image 32698 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:176
a=T38FaxMaxDatagram:176
a=T38FaxUdpEC:t38UDPRedundancy

2009-10-06 19:03:47.049284 [DEBUG] switch_core_session.c:630 Send signal
sofia/internal/214 at pbx0.tssec.lan [BREAK]
2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3359 Channel
sofia/internal/214 at pbx0.tssec.lan entering state [calling][0]
2009-10-06 19:03:47.077291 [DEBUG] sofia_glue.c:1101 Remote address:port
[192.168.50.51:16474] has not changed.
2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3158 Passing 200 OK to other leg
2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3170
sofia/external/*408774957558438 at sip.callwithus.com receive message [RESPOND]
2009-10-06 19:03:47.077291 [DEBUG] mod_sofia.c:1427 Responding with 200 [OK]
2009-10-06 19:03:47.077291 [DEBUG] sofia_glue.c:1322
sofia/external/*408774957558438 at sip.callwithus.com Patched SDP
---
v=0
o=- 216899808 216899808 IN IP4 192.168.50.51
s=-
c=IN IP4 192.168.50.51
t=0 0
m=image 16474 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:200
a=T38FaxUdpEC:t38UDPRedundancy

+++
v=0
o=- 216899808 216899808 IN IP4 192.168.50.51
s=-
c=IN IP4 77.239.230.202
t=0 0
m=image 32404 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:200
a=T38FaxUdpEC:t38UDPRedundancy

2009-10-06 19:03:47.077291 [DEBUG] sofia_glue.c:1110 AUDIO RTP CHANGING
DEST TO: [32.50.63.192:14104]
2009-10-06 19:03:47.077291 [DEBUG] switch_core_session.c:630 Send signal
sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3359 Channel
sofia/internal/214 at pbx0.tssec.lan entering state [ready][200]
2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3366 Remote SDP:
v=0
o=- 216899808 216899808 IN IP4 192.168.50.51
s=-
c=IN IP4 192.168.50.51
t=0 0
m=image 16474 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:200
a=T38FaxUdpEC:t38UDPRedundancy

send 1003 bytes to udp/[38.99.70.232]:5060 at 15:06:05.605615:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.0
   Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK41d48157;rport=5062
   From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
   To: "Mihailova Ludmila"
<sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
   CSeq: 102 INVITE
   Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 274
   
   v=0
   o=- 216896551 216896552 IN IP4 192.168.50.51
   s=-
   c=IN IP4 77.239.230.202
   t=0 0
   m=image 32404 udptl t38
   a=T38FaxVersion:0
   a=T38MaxBitRate:14400
   a=T38FaxRateManagement:transferredTCF
   a=T38FaxMaxBuffer:200
   a=T38FaxMaxDatagram:200
   a=T38FaxUdpEC:t38UDPRedundancy
   ------------------------------------------------------------------------
2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3359 Channel
sofia/external/*408774957558438 at sip.callwithus.com entering state
[completed][200]
2009-10-06 19:03:47.285347 [INFO] switch_rtp.c:1905 Auto Changing port
from 32.50.63.192:14104 to 4.55.17.66:14104
recv 527 bytes from udp/[38.99.70.232]:5060 at 15:06:05.817672:
   ------------------------------------------------------------------------
   ACK sip:gw+callwithus at 192.168.50.10:5080;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.2
   Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK5ffd72cd;rport=5062
   From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
   To: "Mihailova Ludmila"
<sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
   Contact: <sip:*408774957558438 at 204.74.213.1:5062>
   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
   CSeq: 102 ACK
   User-Agent: CWU SIP-GW
   Max-Forwards: 69
   Content-Length: 0
   
   ------------------------------------------------------------------------
2009-10-06 19:03:47.289349 [DEBUG] sofia.c:3359 Channel
sofia/external/*408774957558438 at sip.callwithus.com entering state
[ready][200]
recv 476 bytes from udp/[38.99.70.232]:5060 at 15:06:47.588933:
   ------------------------------------------------------------------------
   BYE sip:gw+callwithus at 192.168.50.10:5080;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKd406.f8553e14.0
   Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK0f2cff01;rport=5062
   From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
   To: "Mihailova Ludmila"
<sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
   CSeq: 103 BYE
   User-Agent: CWU SIP-GW
   Max-Forwards: 69
   Content-Length: 0
   
   ------------------------------------------------------------------------
2009-10-06 19:04:29.060609 [NOTICE] sofia.c:328 Hangup
sofia/external/*408774957558438 at sip.callwithus.com [CS_EXCHANGE_MEDIA]
[NORMAL_CLEARING]
2009-10-06 19:04:29.060609 [DEBUG] switch_channel.c:1726 Send signal
sofia/external/*408774957558438 at sip.callwithus.com [KILL]
2009-10-06 19:04:29.060609 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
2009-10-06 19:04:29.060609 [DEBUG] switch_core_state_machine.c:437
thread mismatch skipping state handler.
send 574 bytes to udp/[38.99.70.232]:5060 at 15:06:47.588933:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKd406.f8553e14.0
   Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK0f2cff01;rport=5062
   From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
   To: "Mihailova Ludmila"
<sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
   CSeq: 103 BYE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Content-Length: 0
   
   ------------------------------------------------------------------------


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From quentusrex at gmail.com  Tue Oct  6 12:14:09 2009
From: quentusrex at gmail.com (William King)
Date: Tue, 06 Oct 2009 12:14:09 -0700
Subject: [Freeswitch-users] Memory Question ?
In-Reply-To: <8ccbff060910061157g2220a8fay1585593303bf593e@mail.gmail.com>
References: <8ccbff060910061051p12b39c67kddf201113dda7c48@mail.gmail.com>	<4ACB8514.2050201@gmail.com>	<8ccbff060910061143gc0d0edcl46a2b5274a74c876@mail.gmail.com>	<4ACB9189.5070504@gmail.com>
	<8ccbff060910061157g2220a8fay1585593303bf593e@mail.gmail.com>
Message-ID: <4ACB9701.2010707@gmail.com>

It will use enough ram to load all of your modules, and applications.
That is the initial ram usage on startup.

So how much ram is FS using on startup? less than 250-300 MB of ram on
initial load with all the modules and applications isn't unreasonable. I
think by default it loads tons of stuff you probably wouldn't use in
production.

Next up is how much ram is used per call. This should hardly be anything.

And last up, if you have freeswitch running for days and you see the
memory usage grow from ~150MB to ~800MB, even if it takes weeks. Then
there might be an issue.

If you want to start finding where the RAM usage is located, then unload
some modules and see if it does it still. Also, do you have any custom
scripts in the dialplan? they might not be ending properly. Thus they
stick around when you didn't intend them to.

-William King

Dome Charoenyost wrote:
> 2009/10/7 William King <quentusrex at gmail.com>:
>   
>> Which modules do you have loaded?
>>     
> default config and nibllebill , lcr , odbcquery
>
>   
>> -William King
>>
>> Dome Charoenyost wrote:
>>     
>>> Debian Squeeze  i386 32bit
>>> And Debian Lenny are same
>>>
>>> Dome C.
>>>
>>> 2009/10/7 William King <quentusrex at gmail.com>:
>>>
>>>       
>>>> What operating system?
>>>>
>>>> -William King
>>>>
>>>> Dome Charoenyost wrote:
>>>>
>>>>         
>>>>> Dear All,
>>>>>              why freeswitch use more memory after send and receuve
>>>>> call. i attach htop capture screen.
>>>>> you can compare to asterisk it use 0.7% for long time. but FS use 7.7%
>>>>> (start from 1.2%)  after running about 4 hr.
>>>>>
>>>>> Best Regards.
>>>>>
>>>>> Dome C.
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>           
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>         
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From anthony.minessale at gmail.com  Tue Oct  6 12:25:03 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 6 Oct 2009 14:25:03 -0500
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
Message-ID: <191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>

pcap is not as useful as FS console log on debug with:
sofia profile internal siptrace on

you should be reporting issues to jira under mod_opal not to the mailing
list.
http://jira.freeswitch.org

FYI
There is little financial support from the community for h323 which prevents
the mod_opal from getting much attention.
We actually have to contract the author of opal to help with these issues
including the original writing of the module that he did with very little
funding and nobody ever wants to pay him to improve it.

That does not mean your issue will not be addressed but there is no promise
how fast it will be.



On Tue, Oct 6, 2009 at 12:37 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> hello guys,
>
>
> i was playing with mod_opal to see if i can make it working ... well it
> seems SIP-H323 interworking is not tuned at all.
>
> I have a call from a registered sip user (1001) to PSTN via mod_opal
>
>
> <include>
>   <extension name="EMERGENCY">
>     <condition field="destination_number" expression="^0(112|9[23456])$">
>       <action application="set"
> data="effective_caller_id_number=1001282122"/>
>       <action application="set" data="NCX_IP=10.4.4.254"/>
>       <action application="set" data="call_timeout=30"/>
>       <action application="set" data="hangup_after_bridge=true"/>
>
>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>     </condition>
>   </extension>
>
>   <extension name="SPECIAL_SERVICES">
>     <condition field="destination_number"
> expression="^0(9[01789]\d{3,4})$">
>       <action application="set"
> data="effective_caller_id_number=1001282122"/>
>       <action application="set" data="NCX_IP=10.4.4.254"/>
>       <action application="set" data="call_timeout=30"/>
>       <action application="set" data="hangup_after_bridge=true"/>
>
>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>     </condition>
>   </extension>
>
>   <extension name="ENYTHING_ELSE">
>     <condition field="destination_number"
> expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
>       <action application="set"
> data="effective_caller_id_number=1001282122"/>
>       <action application="set" data="NCX_IP=10.4.4.254"/>
>       <action application="set" data="call_timeout=30"/>
>       <action application="set" data="hangup_after_bridge=true"/>
>
>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>     </condition>
>   </extension>
> </include>
>
>
>
> One of the many issues i sow is that FS connects the call on SIP leg before
> it actually receives H.225 connect from H323 leg... as it is configured to
> send 200 OK on the 1st H.225 message containing a FastStart element/OLC.
>
>
> Attached is the tcpdump i took on FS machine... just use this filter: "h225
> or h245 or q931 or sip"
> Also, you can check the attac CDR,,,, this is an unanswered call i placed
> to PSTN and FS billed it 23 seconds.
>
>
>
> Can anyone tell where i can do correct SIP - H323 message mappings to avoid
> this?
>
>
>
> T.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From tculjaga at gmail.com  Tue Oct  6 13:47:19 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 6 Oct 2009 22:47:19 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
Message-ID: <65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>

hi Anthony,

it is somewhere here:

         switch_status_t
FSConnection::receive_message(switch_core_session_message_t *msg)


anyhow, i will open an issue jira of course.


I understand your financial point of view, but anyhow while the entire world
is wants sip and trying to move to sip, the reality is quite different. The
majority of voice traffic exchanged via IP is still H323. This means a
working SIP - H323 interworking is really needed... pity nobody wants/has
time to work in this direction to produce a decent mod_h323.



T.





On Tue, Oct 6, 2009 at 9:25 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> pcap is not as useful as FS console log on debug with:
> sofia profile internal siptrace on
>
> you should be reporting issues to jira under mod_opal not to the mailing
> list.
> http://jira.freeswitch.org
>
> FYI
> There is little financial support from the community for h323 which
> prevents the mod_opal from getting much attention.
> We actually have to contract the author of opal to help with these issues
> including the original writing of the module that he did with very little
> funding and nobody ever wants to pay him to improve it.
>
> That does not mean your issue will not be addressed but there is no promise
> how fast it will be.
>
>
>
> On Tue, Oct 6, 2009 at 12:37 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> hello guys,
>>
>>
>> i was playing with mod_opal to see if i can make it working ... well it
>> seems SIP-H323 interworking is not tuned at all.
>>
>> I have a call from a registered sip user (1001) to PSTN via mod_opal
>>
>>
>> <include>
>>   <extension name="EMERGENCY">
>>     <condition field="destination_number" expression="^0(112|9[23456])$">
>>       <action application="set"
>> data="effective_caller_id_number=1001282122"/>
>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>       <action application="set" data="call_timeout=30"/>
>>       <action application="set" data="hangup_after_bridge=true"/>
>>
>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>     </condition>
>>   </extension>
>>
>>   <extension name="SPECIAL_SERVICES">
>>     <condition field="destination_number"
>> expression="^0(9[01789]\d{3,4})$">
>>       <action application="set"
>> data="effective_caller_id_number=1001282122"/>
>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>       <action application="set" data="call_timeout=30"/>
>>       <action application="set" data="hangup_after_bridge=true"/>
>>
>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>     </condition>
>>   </extension>
>>
>>   <extension name="ENYTHING_ELSE">
>>     <condition field="destination_number"
>> expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
>>       <action application="set"
>> data="effective_caller_id_number=1001282122"/>
>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>       <action application="set" data="call_timeout=30"/>
>>       <action application="set" data="hangup_after_bridge=true"/>
>>
>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>     </condition>
>>   </extension>
>> </include>
>>
>>
>>
>> One of the many issues i sow is that FS connects the call on SIP leg
>> before it actually receives H.225 connect from H323 leg... as it is
>> configured to send 200 OK on the 1st H.225 message containing a FastStart
>> element/OLC.
>>
>>
>> Attached is the tcpdump i took on FS machine... just use this filter:
>> "h225 or h245 or q931 or sip"
>> Also, you can check the attac CDR,,,, this is an unanswered call i placed
>> to PSTN and FS billed it 23 seconds.
>>
>>
>>
>> Can anyone tell where i can do correct SIP - H323 message mappings to
>> avoid this?
>>
>>
>>
>> T.
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From nicolas at medularis.com  Tue Oct  6 14:22:31 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Tue, 6 Oct 2009 17:22:31 -0400
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <083DBCB9-48B2-4E61-81B5-B7D00A42EAB5@freeswitch.org>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<1b46b4e80910011229k31f7233du10b0fc661730d3f3@mail.gmail.com>
	<F86F950D-9106-49C9-81CA-38B3EB2072F4@jerris.com>
	<1b46b4e80910041519h3581d3d2x7b84041e8c18588@mail.gmail.com>
	<C887D143-F7A1-44B9-B8A2-D34A4B4E582E@jerris.com>
	<1b46b4e80910041609x2f41ceb9u142cb484951a21ed@mail.gmail.com>
	<87f2f3b90910050920w4d17033fl4c386a0018107bdf@mail.gmail.com>
	<1b46b4e80910052119me12a0d9y4653edd831f9555e@mail.gmail.com>
	<1b46b4e80910052142jaa09f08l72e5e4b2b24d10fc@mail.gmail.com>
	<083DBCB9-48B2-4E61-81B5-B7D00A42EAB5@freeswitch.org>
Message-ID: <1b46b4e80910061422q7be2a21fg5d2caaef7e307abb@mail.gmail.com>

That happens with both gateways though, one works and the other doesn't.
Would the rport have anything to do with the registration failing?

The big difference to me is that the working gateway replies a 401
Unauthorized containing:

 WWW-Authenticate: Digest realm="pxextmy.redvoiss.net",
nonce="4acac8fe248a9075a13773274684392a65a40240", qop="auth".

Whereas the non-working gateway's 401 has:

WWW-Authenticate: Digest realm="216.72.10.39",
nonce="4acac08249c439decb2bea539282faf755c80b0c".


What does the qop parameter stand for? Apparently because of that parameter,
FS sends a new REGISTER including this:

Authorization: Digest username="xxxxxxxxx", realm="pxextmy.redvoiss.net",
nonce="4acac8fe248a9075a13773274684392a65a40240",
cnonce="h1DCSizTEi2eMQAdCe9KJA", algorithm=MD5, uri="sip:
pxextmy.redvoiss.net", response="05adb2a7f9d7772e57dc846257484f5d",
qop=auth, nc=00000001.

Instead, on the non-working gateway case, FS sends a REGISTER with this:

Authorization: Digest username="yyyyyyyyy", realm="216.72.10.39",
nonce="4acac08249c439decb2bea539282faf755c80b0c", algorithm=MD5, uri="sip:
216.72.10.39", response="8311db7666779df89d5223e16a611826".


Notice the absence of the qop and nc parameters. I'm guessing the lack of
those parameters causes the gateway (SIP server) to use another nonce and
hence reject the mismatching REGISTER.

BTW, registration from an X-Lite softphone works.


Thanks!

Nicolas


On Tue, Oct 6, 2009 at 10:31 AM, Brian West <brian at freeswitch.org> wrote:

> This looks like you have an ALG messing with packets... notice it says
> rport 5080 but we are sending to 5060.
> /b
>
> On Oct 5, 2009, at 11:42 PM, Nicolas Brenner wrote:
>
> Ignore my previous email, the traces were incomplete, got much better (and
> complete) traces with ngrep (found a suggestion from Brian in the list
> archive, thanks!)
>
> The gateway that registers:
>
> - http://pastebin.freeswitch.org/10607
>
> The one that doesn't:
>
> - http://pastebin.freeswitch.org/10608
>
>
> Thanks again for your time and help!
>
>
> Nicolas
>
>
> On Tue, Oct 6, 2009 at 12:19 AM, Nicolas Brenner <nicolas at medularis.com>wrote:
>
>> There was no sane way of doing that, so I ended up logging the trace from
>> the cli.
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Tue Oct  6 14:45:10 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 6 Oct 2009 16:45:10 -0500
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <1b46b4e80910061422q7be2a21fg5d2caaef7e307abb@mail.gmail.com>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<1b46b4e80910011229k31f7233du10b0fc661730d3f3@mail.gmail.com>
	<F86F950D-9106-49C9-81CA-38B3EB2072F4@jerris.com>
	<1b46b4e80910041519h3581d3d2x7b84041e8c18588@mail.gmail.com>
	<C887D143-F7A1-44B9-B8A2-D34A4B4E582E@jerris.com>
	<1b46b4e80910041609x2f41ceb9u142cb484951a21ed@mail.gmail.com>
	<87f2f3b90910050920w4d17033fl4c386a0018107bdf@mail.gmail.com>
	<1b46b4e80910052119me12a0d9y4653edd831f9555e@mail.gmail.com>
	<1b46b4e80910052142jaa09f08l72e5e4b2b24d10fc@mail.gmail.com>
	<083DBCB9-48B2-4E61-81B5-B7D00A42EAB5@freeswitch.org>
	<1b46b4e80910061422q7be2a21fg5d2caaef7e307abb@mail.gmail.com>
Message-ID: <9C8FB847-FBFB-4382-8C60-865C9FB5CAE7@freeswitch.org>

First off you have to fully understand how SIP authentication works  
the two authorization line are different because one is for a  
challenge and one is a response to a challenge.

http://en.wikipedia.org/wiki/Digest_access_authentication


On Oct 6, 2009, at 4:22 PM, Nicolas Brenner wrote:

> That happens with both gateways though, one works and the other  
> doesn't. Would the rport have anything to do with the registration  
> failing?
>
> The big difference to me is that the working gateway replies a 401  
> Unauthorized containing:
>
>  WWW-Authenticate: Digest realm="pxextmy.redvoiss.net",  
> nonce="4acac8fe248a9075a13773274684392a65a40240", qop="auth".
>
> Whereas the non-working gateway's 401 has:
>
> WWW-Authenticate: Digest realm="216.72.10.39",  
> nonce="4acac08249c439decb2bea539282faf755c80b0c".

What is this gateway? You might actually put the realm param INTO the  
gateway config for this gateway.


> What does the qop parameter stand for? Apparently because of that  
> parameter, FS sends a new REGISTER including this:

Quality of Protection, qop is assumed auth if excluded.

>
> Authorization: Digest username="xxxxxxxxx", realm="pxextmy.redvoiss.net 
> ", nonce="4acac8fe248a9075a13773274684392a65a40240",  
> cnonce="h1DCSizTEi2eMQAdCe9KJA", algorithm=MD5, uri="sip:pxextmy.redvoiss.net 
> ", response="05adb2a7f9d7772e57dc846257484f5d", qop=auth, nc=00000001.

This is a response to a challenge.

> Instead, on the non-working gateway case, FS sends a REGISTER with  
> this:
>
> Authorization: Digest username="yyyyyyyyy", realm="216.72.10.39",  
> nonce="4acac08249c439decb2bea539282faf755c80b0c", algorithm=MD5,  
> uri="sip:216.72.10.39", response="8311db7666779df89d5223e16a611826".

This is a challenge.

> Notice the absence of the qop and nc parameters. I'm guessing the  
> lack of those parameters causes the gateway (SIP server) to use  
> another nonce and hence reject the mismatching REGISTER.

Again challenge vs response.

>
> BTW, registration from an X-Lite softphone works.
>
>
> Thanks!
>
> Nicolas

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From diego.viola at gmail.com  Tue Oct  6 14:45:25 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 6 Oct 2009 21:45:25 +0000
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
Message-ID: <86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>

Instead of complaining and demanding things for free, people should start to
put their money where their mouth is.

Diego

On Tue, Oct 6, 2009 at 8:47 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> hi Anthony,
>
> it is somewhere here:
>
>          switch_status_t
> FSConnection::receive_message(switch_core_session_message_t *msg)
>
>
> anyhow, i will open an issue jira of course.
>
>
> I understand your financial point of view, but anyhow while the entire
> world is wants sip and trying to move to sip, the reality is quite
> different. The majority of voice traffic exchanged via IP is still H323.
> This means a working SIP - H323 interworking is really needed... pity nobody
> wants/has time to work in this direction to produce a decent mod_h323.
>
>
>
> T.
>
>
>
>
>
>
> On Tue, Oct 6, 2009 at 9:25 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> pcap is not as useful as FS console log on debug with:
>> sofia profile internal siptrace on
>>
>> you should be reporting issues to jira under mod_opal not to the mailing
>> list.
>> http://jira.freeswitch.org
>>
>> FYI
>> There is little financial support from the community for h323 which
>> prevents the mod_opal from getting much attention.
>> We actually have to contract the author of opal to help with these issues
>> including the original writing of the module that he did with very little
>> funding and nobody ever wants to pay him to improve it.
>>
>> That does not mean your issue will not be addressed but there is no
>> promise how fast it will be.
>>
>>
>>
>> On Tue, Oct 6, 2009 at 12:37 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>
>>> hello guys,
>>>
>>>
>>> i was playing with mod_opal to see if i can make it working ... well it
>>> seems SIP-H323 interworking is not tuned at all.
>>>
>>> I have a call from a registered sip user (1001) to PSTN via mod_opal
>>>
>>>
>>> <include>
>>>   <extension name="EMERGENCY">
>>>     <condition field="destination_number" expression="^0(112|9[23456])$">
>>>       <action application="set"
>>> data="effective_caller_id_number=1001282122"/>
>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>       <action application="set" data="call_timeout=30"/>
>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>
>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>     </condition>
>>>   </extension>
>>>
>>>   <extension name="SPECIAL_SERVICES">
>>>     <condition field="destination_number"
>>> expression="^0(9[01789]\d{3,4})$">
>>>       <action application="set"
>>> data="effective_caller_id_number=1001282122"/>
>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>       <action application="set" data="call_timeout=30"/>
>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>
>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>     </condition>
>>>   </extension>
>>>
>>>   <extension name="ENYTHING_ELSE">
>>>     <condition field="destination_number"
>>> expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
>>>       <action application="set"
>>> data="effective_caller_id_number=1001282122"/>
>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>       <action application="set" data="call_timeout=30"/>
>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>
>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>     </condition>
>>>   </extension>
>>> </include>
>>>
>>>
>>>
>>> One of the many issues i sow is that FS connects the call on SIP leg
>>> before it actually receives H.225 connect from H323 leg... as it is
>>> configured to send 200 OK on the 1st H.225 message containing a FastStart
>>> element/OLC.
>>>
>>>
>>> Attached is the tcpdump i took on FS machine... just use this filter:
>>> "h225 or h245 or q931 or sip"
>>> Also, you can check the attac CDR,,,, this is an unanswered call i placed
>>> to PSTN and FS billed it 23 seconds.
>>>
>>>
>>>
>>> Can anyone tell where i can do correct SIP - H323 message mappings to
>>> avoid this?
>>>
>>>
>>>
>>> T.
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Tue Oct  6 14:46:59 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 6 Oct 2009 16:46:59 -0500
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <1b46b4e80910061422q7be2a21fg5d2caaef7e307abb@mail.gmail.com>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<1b46b4e80910011229k31f7233du10b0fc661730d3f3@mail.gmail.com>
	<F86F950D-9106-49C9-81CA-38B3EB2072F4@jerris.com>
	<1b46b4e80910041519h3581d3d2x7b84041e8c18588@mail.gmail.com>
	<C887D143-F7A1-44B9-B8A2-D34A4B4E582E@jerris.com>
	<1b46b4e80910041609x2f41ceb9u142cb484951a21ed@mail.gmail.com>
	<87f2f3b90910050920w4d17033fl4c386a0018107bdf@mail.gmail.com>
	<1b46b4e80910052119me12a0d9y4653edd831f9555e@mail.gmail.com>
	<1b46b4e80910052142jaa09f08l72e5e4b2b24d10fc@mail.gmail.com>
	<083DBCB9-48B2-4E61-81B5-B7D00A42EAB5@freeswitch.org>
	<1b46b4e80910061422q7be2a21fg5d2caaef7e307abb@mail.gmail.com>
Message-ID: <9106C8E1-498F-4AE8-8D34-7F6AAD4DC8AE@freeswitch.org>

btw My mistake it doesn't assume auth it just calculates the response  
hash differently on this case where qop isn't present.

/b

On Oct 6, 2009, at 4:22 PM, Nicolas Brenner wrote:

>
> What does the qop parameter stand for? Apparently because of that  
> parameter, FS sends a new REGISTER including this:




From jason at jasonjgw.net  Tue Oct  6 15:58:24 2009
From: jason at jasonjgw.net (Jason White)
Date: Wed, 7 Oct 2009 09:58:24 +1100
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
Message-ID: <20091006225824.GA7451@jdc.jasonjgw.net>

Tihomir Culjaga <tculjaga at gmail.com> wrote:
 
> 
> I understand your financial point of view, but anyhow while the entire world
> is wants sip and trying to move to sip, the reality is quite different. The
> majority of voice traffic exchanged via IP is still H323. 

Is there any evidence in support of the above assertion (e.g., survey results
of VoIP traffic)? I've heard of H323 but I don't know anyone who uses it, or
any phones that implement it.

The lack of interest in this forum and the absence of financial support to
improve the H323 support in FreeSWITCH suggest that the level of demand for
this is quite low, relative to SIP.

Of course, improvements are always welcome, so if you're interested in funding
better H323 support, or helping with the module I'm sure the FreeSWITCH
community would welcome your efforts.




From tculjaga at gmail.com  Tue Oct  6 16:41:13 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 7 Oct 2009 01:41:13 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
Message-ID: <65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>

Diego,

what i'm pointing here is the situation where you have a great product that
lacks in one of most common protocol. It is true H323 is going to disappear
(eventually), it is true that the community prefers SIP/IAX instead ... but
the reality still remains. H323 is going to be used for quite a long time to
exchange a lot of traffic while FS will be left aside. Today, when you setup
an IP peering interconnection 80% of carriers will prefer H323.

Of course, developing something costs "time" (and we all know what time
stands for...) and as i said, i understand the financial point of view and i
really understand if nobody is going to work on that, but let's face it FS
doesn't have any usable module to reliably handle H323 protocol.


said that, i don't intend to offend anyone... just facing the reality.


regarding the h323 module, we don't have any issue fixing the existing or
developing a new one... but before we go developing something it is always
better check if the thing you want already exists in an usable state or
not... that's what i did today.


So, I'm interested in a reliable module handling H323v4... anyone else?


T.





On Tue, Oct 6, 2009 at 11:45 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Instead of complaining and demanding things for free, people should start
> to put their money where their mouth is.
>
> Diego
>
>
> On Tue, Oct 6, 2009 at 8:47 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> hi Anthony,
>>
>> it is somewhere here:
>>
>>          switch_status_t
>> FSConnection::receive_message(switch_core_session_message_t *msg)
>>
>>
>> anyhow, i will open an issue jira of course.
>>
>>
>> I understand your financial point of view, but anyhow while the entire
>> world is wants sip and trying to move to sip, the reality is quite
>> different. The majority of voice traffic exchanged via IP is still H323.
>> This means a working SIP - H323 interworking is really needed... pity nobody
>> wants/has time to work in this direction to produce a decent mod_h323.
>>
>>
>>
>> T.
>>
>>
>>
>>
>>
>>
>> On Tue, Oct 6, 2009 at 9:25 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> pcap is not as useful as FS console log on debug with:
>>> sofia profile internal siptrace on
>>>
>>> you should be reporting issues to jira under mod_opal not to the mailing
>>> list.
>>> http://jira.freeswitch.org
>>>
>>> FYI
>>> There is little financial support from the community for h323 which
>>> prevents the mod_opal from getting much attention.
>>> We actually have to contract the author of opal to help with these issues
>>> including the original writing of the module that he did with very little
>>> funding and nobody ever wants to pay him to improve it.
>>>
>>> That does not mean your issue will not be addressed but there is no
>>> promise how fast it will be.
>>>
>>>
>>>
>>> On Tue, Oct 6, 2009 at 12:37 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>>
>>>> hello guys,
>>>>
>>>>
>>>> i was playing with mod_opal to see if i can make it working ... well it
>>>> seems SIP-H323 interworking is not tuned at all.
>>>>
>>>> I have a call from a registered sip user (1001) to PSTN via mod_opal
>>>>
>>>>
>>>> <include>
>>>>   <extension name="EMERGENCY">
>>>>     <condition field="destination_number"
>>>> expression="^0(112|9[23456])$">
>>>>       <action application="set"
>>>> data="effective_caller_id_number=1001282122"/>
>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>>       <action application="set" data="call_timeout=30"/>
>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>
>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>>     </condition>
>>>>   </extension>
>>>>
>>>>   <extension name="SPECIAL_SERVICES">
>>>>     <condition field="destination_number"
>>>> expression="^0(9[01789]\d{3,4})$">
>>>>       <action application="set"
>>>> data="effective_caller_id_number=1001282122"/>
>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>>       <action application="set" data="call_timeout=30"/>
>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>
>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>>     </condition>
>>>>   </extension>
>>>>
>>>>   <extension name="ENYTHING_ELSE">
>>>>     <condition field="destination_number"
>>>> expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
>>>>       <action application="set"
>>>> data="effective_caller_id_number=1001282122"/>
>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>>       <action application="set" data="call_timeout=30"/>
>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>
>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>>     </condition>
>>>>   </extension>
>>>> </include>
>>>>
>>>>
>>>>
>>>> One of the many issues i sow is that FS connects the call on SIP leg
>>>> before it actually receives H.225 connect from H323 leg... as it is
>>>> configured to send 200 OK on the 1st H.225 message containing a FastStart
>>>> element/OLC.
>>>>
>>>>
>>>> Attached is the tcpdump i took on FS machine... just use this filter:
>>>> "h225 or h245 or q931 or sip"
>>>> Also, you can check the attac CDR,,,, this is an unanswered call i
>>>> placed to PSTN and FS billed it 23 seconds.
>>>>
>>>>
>>>>
>>>> Can anyone tell where i can do correct SIP - H323 message mappings to
>>>> avoid this?
>>>>
>>>>
>>>>
>>>> T.
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:213-799-1400
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Tue Oct  6 17:09:25 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 7 Oct 2009 02:09:25 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <20091006225824.GA7451@jdc.jasonjgw.net>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<20091006225824.GA7451@jdc.jasonjgw.net>
Message-ID: <65d96fc80910061709t2bc9aa5ex8769ef7aa3e27671@mail.gmail.com>

thanks for your e-mail,

H323 is mainly used for trunking purpose, inter-carrier traffic exchange...
it is not used to control IP phones :P

well, believe me, I've heard enough of H323 that i'm sick of it :P
What i can tell you comes from my own experience on daily activities i'm
doing for living... Of course, there might be part of the world where H323
dispersed completely but over here in Europe things tend to stick on
tradition :P

Yep, you are right... the forum wants SIP and that's understandable...

anyhow you might check this:
http://www.dailypayload.com/content/3111


T.


On Wed, Oct 7, 2009 at 12:58 AM, Jason White <jason at jasonjgw.net> wrote:

> Tihomir Culjaga <tculjaga at gmail.com> wrote:
>
> >
> > I understand your financial point of view, but anyhow while the entire
> world
> > is wants sip and trying to move to sip, the reality is quite different.
> The
> > majority of voice traffic exchanged via IP is still H323.
>
> Is there any evidence in support of the above assertion (e.g., survey
> results
> of VoIP traffic)? I've heard of H323 but I don't know anyone who uses it,
> or
> any phones that implement it.
>
> The lack of interest in this forum and the absence of financial support to
> improve the H323 support in FreeSWITCH suggest that the level of demand for
> this is quite low, relative to SIP.
>
> Of course, improvements are always welcome, so if you're interested in
> funding
> better H323 support, or helping with the module I'm sure the FreeSWITCH
> community would welcome your efforts.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From diego.viola at gmail.com  Tue Oct  6 17:22:13 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Wed, 7 Oct 2009 00:22:13 +0000
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
Message-ID: <86a32abc0910061722t67ff23ddr62e4f62365539bd4@mail.gmail.com>

Yeah I understand your point of view, but saying "I want a H.323 module" or
"I want a Ferrari" wont magically make it happen.

We need to work on it ourselves or pay to the people that knows how to do it
to do it for us.

There is no other way I think.

Diego



On Tue, Oct 6, 2009 at 11:41 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> Diego,
>
> what i'm pointing here is the situation where you have a great product that
> lacks in one of most common protocol. It is true H323 is going to disappear
> (eventually), it is true that the community prefers SIP/IAX instead ... but
> the reality still remains. H323 is going to be used for quite a long time to
> exchange a lot of traffic while FS will be left aside. Today, when you setup
> an IP peering interconnection 80% of carriers will prefer H323.
>
> Of course, developing something costs "time" (and we all know what time
> stands for...) and as i said, i understand the financial point of view and i
> really understand if nobody is going to work on that, but let's face it FS
> doesn't have any usable module to reliably handle H323 protocol.
>
>
> said that, i don't intend to offend anyone... just facing the reality.
>
>
> regarding the h323 module, we don't have any issue fixing the existing or
> developing a new one... but before we go developing something it is always
> better check if the thing you want already exists in an usable state or
> not... that's what i did today.
>
>
> So, I'm interested in a reliable module handling H323v4... anyone else?
>
>
> T.
>
>
>
>
>
>
> On Tue, Oct 6, 2009 at 11:45 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> Instead of complaining and demanding things for free, people should start
>> to put their money where their mouth is.
>>
>> Diego
>>
>>
>> On Tue, Oct 6, 2009 at 8:47 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>
>>> hi Anthony,
>>>
>>> it is somewhere here:
>>>
>>>          switch_status_t
>>> FSConnection::receive_message(switch_core_session_message_t *msg)
>>>
>>>
>>> anyhow, i will open an issue jira of course.
>>>
>>>
>>> I understand your financial point of view, but anyhow while the entire
>>> world is wants sip and trying to move to sip, the reality is quite
>>> different. The majority of voice traffic exchanged via IP is still H323.
>>> This means a working SIP - H323 interworking is really needed... pity nobody
>>> wants/has time to work in this direction to produce a decent mod_h323.
>>>
>>>
>>>
>>> T.
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Tue, Oct 6, 2009 at 9:25 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> pcap is not as useful as FS console log on debug with:
>>>> sofia profile internal siptrace on
>>>>
>>>> you should be reporting issues to jira under mod_opal not to the mailing
>>>> list.
>>>> http://jira.freeswitch.org
>>>>
>>>> FYI
>>>> There is little financial support from the community for h323 which
>>>> prevents the mod_opal from getting much attention.
>>>> We actually have to contract the author of opal to help with these
>>>> issues including the original writing of the module that he did with very
>>>> little funding and nobody ever wants to pay him to improve it.
>>>>
>>>> That does not mean your issue will not be addressed but there is no
>>>> promise how fast it will be.
>>>>
>>>>
>>>>
>>>> On Tue, Oct 6, 2009 at 12:37 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>>>
>>>>> hello guys,
>>>>>
>>>>>
>>>>> i was playing with mod_opal to see if i can make it working ... well it
>>>>> seems SIP-H323 interworking is not tuned at all.
>>>>>
>>>>> I have a call from a registered sip user (1001) to PSTN via mod_opal
>>>>>
>>>>>
>>>>> <include>
>>>>>   <extension name="EMERGENCY">
>>>>>     <condition field="destination_number"
>>>>> expression="^0(112|9[23456])$">
>>>>>       <action application="set"
>>>>> data="effective_caller_id_number=1001282122"/>
>>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>>>       <action application="set" data="call_timeout=30"/>
>>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>>
>>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>>>     </condition>
>>>>>   </extension>
>>>>>
>>>>>   <extension name="SPECIAL_SERVICES">
>>>>>     <condition field="destination_number"
>>>>> expression="^0(9[01789]\d{3,4})$">
>>>>>       <action application="set"
>>>>> data="effective_caller_id_number=1001282122"/>
>>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>>>       <action application="set" data="call_timeout=30"/>
>>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>>
>>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>>>     </condition>
>>>>>   </extension>
>>>>>
>>>>>   <extension name="ENYTHING_ELSE">
>>>>>     <condition field="destination_number"
>>>>> expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
>>>>>       <action application="set"
>>>>> data="effective_caller_id_number=1001282122"/>
>>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>>>       <action application="set" data="call_timeout=30"/>
>>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>>
>>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>>>     </condition>
>>>>>   </extension>
>>>>> </include>
>>>>>
>>>>>
>>>>>
>>>>> One of the many issues i sow is that FS connects the call on SIP leg
>>>>> before it actually receives H.225 connect from H323 leg... as it is
>>>>> configured to send 200 OK on the 1st H.225 message containing a FastStart
>>>>> element/OLC.
>>>>>
>>>>>
>>>>> Attached is the tcpdump i took on FS machine... just use this filter:
>>>>> "h225 or h245 or q931 or sip"
>>>>> Also, you can check the attac CDR,,,, this is an unanswered call i
>>>>> placed to PSTN and FS billed it 23 seconds.
>>>>>
>>>>>
>>>>>
>>>>> Can anyone tell where i can do correct SIP - H323 message mappings to
>>>>> avoid this?
>>>>>
>>>>>
>>>>>
>>>>> T.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:213-799-1400
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Tue Oct  6 17:58:20 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 6 Oct 2009 19:58:20 -0500
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <86a32abc0910061722t67ff23ddr62e4f62365539bd4@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<86a32abc0910061722t67ff23ddr62e4f62365539bd4@mail.gmail.com>
Message-ID: <191c3a030910061758r2ddea5fand388f0af2c4dfc33@mail.gmail.com>

I didn't mean to start anything.  I'm just saying we work very long hours
and barely get anybody asking about h.323.
I wanted to support it and that's why we took up a collection to get funding
for mod_opal but when only 1 donor showed any interest we were forced to
proceed in our spare time which is very limited.

The developers of opal are not part of our project and they need financial
compensation to be motivated to work on it.  Its not even related to me its
only fair that an outside developer who makes his living as a consultant
would want money to integrate his work into our project.

Like I said, I will do my best to point your issue to the opal devs but I
cannot force them to work on it.




On Tue, Oct 6, 2009 at 7:22 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Yeah I understand your point of view, but saying "I want a H.323 module" or
> "I want a Ferrari" wont magically make it happen.
>
> We need to work on it ourselves or pay to the people that knows how to do
> it to do it for us.
>
> There is no other way I think.
>
> Diego
>
>
>
>
> On Tue, Oct 6, 2009 at 11:41 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> Diego,
>>
>> what i'm pointing here is the situation where you have a great product
>> that lacks in one of most common protocol. It is true H323 is going to
>> disappear (eventually), it is true that the community prefers SIP/IAX
>> instead ... but the reality still remains. H323 is going to be used for
>> quite a long time to exchange a lot of traffic while FS will be left aside.
>> Today, when you setup an IP peering interconnection 80% of carriers will
>> prefer H323.
>>
>> Of course, developing something costs "time" (and we all know what time
>> stands for...) and as i said, i understand the financial point of view and i
>> really understand if nobody is going to work on that, but let's face it FS
>> doesn't have any usable module to reliably handle H323 protocol.
>>
>>
>> said that, i don't intend to offend anyone... just facing the reality.
>>
>>
>> regarding the h323 module, we don't have any issue fixing the existing or
>> developing a new one... but before we go developing something it is always
>> better check if the thing you want already exists in an usable state or
>> not... that's what i did today.
>>
>>
>> So, I'm interested in a reliable module handling H323v4... anyone else?
>>
>>
>> T.
>>
>>
>>
>>
>>
>>
>> On Tue, Oct 6, 2009 at 11:45 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>
>>> Instead of complaining and demanding things for free, people should start
>>> to put their money where their mouth is.
>>>
>>> Diego
>>>
>>>
>>> On Tue, Oct 6, 2009 at 8:47 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>>
>>>> hi Anthony,
>>>>
>>>> it is somewhere here:
>>>>
>>>>          switch_status_t
>>>> FSConnection::receive_message(switch_core_session_message_t *msg)
>>>>
>>>>
>>>> anyhow, i will open an issue jira of course.
>>>>
>>>>
>>>> I understand your financial point of view, but anyhow while the entire
>>>> world is wants sip and trying to move to sip, the reality is quite
>>>> different. The majority of voice traffic exchanged via IP is still H323.
>>>> This means a working SIP - H323 interworking is really needed... pity nobody
>>>> wants/has time to work in this direction to produce a decent mod_h323.
>>>>
>>>>
>>>>
>>>> T.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Tue, Oct 6, 2009 at 9:25 PM, Anthony Minessale <
>>>> anthony.minessale at gmail.com> wrote:
>>>>
>>>>> pcap is not as useful as FS console log on debug with:
>>>>> sofia profile internal siptrace on
>>>>>
>>>>> you should be reporting issues to jira under mod_opal not to the
>>>>> mailing list.
>>>>> http://jira.freeswitch.org
>>>>>
>>>>> FYI
>>>>> There is little financial support from the community for h323 which
>>>>> prevents the mod_opal from getting much attention.
>>>>> We actually have to contract the author of opal to help with these
>>>>> issues including the original writing of the module that he did with very
>>>>> little funding and nobody ever wants to pay him to improve it.
>>>>>
>>>>> That does not mean your issue will not be addressed but there is no
>>>>> promise how fast it will be.
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Oct 6, 2009 at 12:37 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>>>>
>>>>>> hello guys,
>>>>>>
>>>>>>
>>>>>> i was playing with mod_opal to see if i can make it working ... well
>>>>>> it seems SIP-H323 interworking is not tuned at all.
>>>>>>
>>>>>> I have a call from a registered sip user (1001) to PSTN via mod_opal
>>>>>>
>>>>>>
>>>>>> <include>
>>>>>>   <extension name="EMERGENCY">
>>>>>>     <condition field="destination_number"
>>>>>> expression="^0(112|9[23456])$">
>>>>>>       <action application="set"
>>>>>> data="effective_caller_id_number=1001282122"/>
>>>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>>>>       <action application="set" data="call_timeout=30"/>
>>>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>>>
>>>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>>>>     </condition>
>>>>>>   </extension>
>>>>>>
>>>>>>   <extension name="SPECIAL_SERVICES">
>>>>>>     <condition field="destination_number"
>>>>>> expression="^0(9[01789]\d{3,4})$">
>>>>>>       <action application="set"
>>>>>> data="effective_caller_id_number=1001282122"/>
>>>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>>>>       <action application="set" data="call_timeout=30"/>
>>>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>>>
>>>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>>>>     </condition>
>>>>>>   </extension>
>>>>>>
>>>>>>   <extension name="ENYTHING_ELSE">
>>>>>>     <condition field="destination_number"
>>>>>> expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
>>>>>>       <action application="set"
>>>>>> data="effective_caller_id_number=1001282122"/>
>>>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>>>>       <action application="set" data="call_timeout=30"/>
>>>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>>>
>>>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>>>>     </condition>
>>>>>>   </extension>
>>>>>> </include>
>>>>>>
>>>>>>
>>>>>>
>>>>>> One of the many issues i sow is that FS connects the call on SIP leg
>>>>>> before it actually receives H.225 connect from H323 leg... as it is
>>>>>> configured to send 200 OK on the 1st H.225 message containing a FastStart
>>>>>> element/OLC.
>>>>>>
>>>>>>
>>>>>> Attached is the tcpdump i took on FS machine... just use this filter:
>>>>>> "h225 or h245 or q931 or sip"
>>>>>> Also, you can check the attac CDR,,,, this is an unanswered call i
>>>>>> placed to PSTN and FS billed it 23 seconds.
>>>>>>
>>>>>>
>>>>>>
>>>>>> Can anyone tell where i can do correct SIP - H323 message mappings to
>>>>>> avoid this?
>>>>>>
>>>>>>
>>>>>>
>>>>>> T.
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>> iax:guest at conference.freeswitch.org/888
>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>> pstn:213-799-1400
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mgende at gendesign.com  Tue Oct  6 18:52:40 2009
From: mgende at gendesign.com (Michael Gende)
Date: Tue, 6 Oct 2009 20:52:40 -0500
Subject: [Freeswitch-users] Wiki Info: Multi Homed Tutorial
In-Reply-To: <87f2f3b90910060915g167bafa7jd8b0ed5e1b3b2b20@mail.gmail.com>
References: <87f2f3b90910051527x638f55b8hda75d5302267c251@mail.gmail.com>
	<a87d43fd0910052120l76da1536ra92766174dc938d3@mail.gmail.com>
	<87f2f3b90910060915g167bafa7jd8b0ed5e1b3b2b20@mail.gmail.com>
Message-ID: <a87d43fd0910061852y3542f23p2bab39365b4b350b@mail.gmail.com>

Thanks Michael,

I found this right after posting you and have already begun
adding/correcting the guide. Thanks again for getting it up there in the
first place.

Mike G.

On Tue, Oct 6, 2009 at 11:15 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Oct 5, 2009 at 9:20 PM, Michael Gende <mgende at gendesign.com>wrote:
>
>> Michael,
>>
>> Thanks for "wiki-fying" my text-only attempt at some user doc. I should
>> have done that for you. I actually have an updated version with many
>> corrections and the end tabs filled in. Can you point me to info on how I
>> can amend and append what you have kindly put up?
>>
>> Mike G.
>>
>> First, go to wiki.freeswitch.org and create a wiki account. Then, go to
> the Multi_home_tutorial page and click edit (top of the page). You'll see
> that there is wiki markup to learn. If you have questions let me know. Just
> edit the wiki text and click Show Preview to see what the real thing looks
> like. Then click Save Page to save your changes.
>
> Welcome to the world of MediaWiki and thanks for your help!
> -MC
>
>
>
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From mgende at gendesign.com  Tue Oct  6 18:53:14 2009
From: mgende at gendesign.com (Michael Gende)
Date: Tue, 6 Oct 2009 20:53:14 -0500
Subject: [Freeswitch-users] Wiki Info: Multi Homed Tutorial
In-Reply-To: <6f7c60c40910060544h5d1f7251p7194bcaa944a627c@mail.gmail.com>
References: <87f2f3b90910051527x638f55b8hda75d5302267c251@mail.gmail.com>
	<a87d43fd0910052120l76da1536ra92766174dc938d3@mail.gmail.com>
	<6f7c60c40910060544h5d1f7251p7194bcaa944a627c@mail.gmail.com>
Message-ID: <a87d43fd0910061853k1d78e687jdc5d4e7249a1c69f@mail.gmail.com>

Sounds good.

On Tue, Oct 6, 2009 at 7:44 AM, tom <tomabroad at gmail.com> wrote:

> thx guys, that helped me alot!
>
>
> On Tue, Oct 6, 2009 at 12:20 AM, Michael Gende <mgende at gendesign.com>wrote:
>
>> Michael,
>>
>> Thanks for "wiki-fying" my text-only attempt at some user doc. I should
>> have done that for you. I actually have an updated version with many
>> corrections and the end tabs filled in. Can you point me to info on how I
>> can amend and append what you have kindly put up?
>>
>> Mike G.
>>
>>
>> On Mon, Oct 5, 2009 at 5:27 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> FYI,
>>>
>>> For those who've been following the thread about Michael Gende's tutorial
>>> I just wanted to let you know that I his document on the wiki. It can be
>>> found here:
>>>
>>> http://wiki.freeswitch.org/wiki/Multi_home_tutorial
>>>
>>> Please feel free to get in there and try it out, make editorial changes,
>>> etc. I'm sure you won't hurt Mike's feelings by adding your thoughts. :)
>>>
>>> If you have any questions please reply to this thread and we'll take it
>>> from there.
>>>
>>> Thanks,
>>> MC
>>>
>>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Tue Oct  6 20:04:23 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 6 Oct 2009 23:04:23 -0400
Subject: [Freeswitch-users] mod_xml_curl http POST is inconsistent/bug
In-Reply-To: <66EA3166EB339A4489B06286C0876A8A0C0C1F04@mailserv.Globalive.local>
References: <66EA3166EB339A4489B06286C0876A8A0C0C1432@mailserv.Globalive.local><276FEE28-70CF-422D-9F6F-7CF747FDDB22@freeswitch.org><66EA3166EB339A4489B06286C0876A8A0C0C1BA9@mailserv.Globalive.local>
	<191c3a030910060801x3ed48e98x6d69b53c6beaca69@mail.gmail.com>
	<66EA3166EB339A4489B06286C0876A8A0C0C1F04@mailserv.Globalive.local>
Message-ID: <1CD47AE5-ED16-47AE-9730-BA22551BDF31@jerris.com>

Could you open a bug on jira.freeswitch.org as a feature request to  
make this a configurable param.  (patches that do it even better)

Mike

On Oct 6, 2009, at 12:55 PM, Christian Damianidis wrote:

> I?ve tested this and making the change from ANY to BASIC worked.  
> Thanks for the help.
> It no longer sends the initial post without auth.
> From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
> Sent: Tuesday, October 06, 2009 11:02 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] mod_xml_curl http POST is  
> inconsistent/bug
>
> My guess is that we configure the curl to support the full range of  
> http auth methods.
> Some of them like Digest require a challenge and realm etc so it's  
> probably asking without auth header because it cannot create one  
> until it gets that data.  In the case of Basic you can send the  
> login and pass right away but it does not know in advance that it  
> will be basic.
>
> Here is a snippet from the libcurl api docs:
> -------------------------------------------------------------------------------------------------------------------------------------------------------------
> Both these options allow you to set multiple types (by ORing them  
> together), to make libcurl pick the most secure one out of the types  
> the server/proxy claims to support. This method does however add a  
> round-trip since libcurl must first ask the server what it supports:
>
>  curl_easy_setopt(easyhandle, CURLOPT_HTTPAUTH,  CURLAUTH_DIGEST| 
> CURLAUTH_BASIC);
>
> -------------------------------------------------------------------------------------------------------------------------------------------------------------
>
> So my guess is that if we set it to only support basic, then it  
> would work how you expect so if you want to test it for me I can  
> make it into a parameter.
>
> edit: /usr/src/freeswitch.trunk/src/mod/xml_int/mod_xml_curl/ 
> mod_xml_curl.c line 220
> change
>
> curl_easy_setopt(curl_handle, CURLOPT_HTTPAUTH, CURLAUTH_ANY);
>
> to
>
> curl_easy_setopt(curl_handle, CURLOPT_HTTPAUTH, CURLAUTH_BASIC);
>
>
> If this works i'll think about exposing the auth methods so you can  
> choose them in the config.
>
>
>
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From shaheryarkh at googlemail.com  Tue Oct  6 21:24:31 2009
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Wed, 7 Oct 2009 10:24:31 +0600
Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Followup,
	Agenda For This Week
In-Reply-To: <B5ECAE34-B846-4408-8AC1-6D94ADCC672C@freeswitch.org>
References: <87f2f3b90910051203v7bdbca5amfc82d193f3a4f1a3@mail.gmail.com>
	<B5ECAE34-B846-4408-8AC1-6D94ADCC672C@freeswitch.org>
Message-ID: <b16156850910062124h4bae532enc3624e18d1b20b2f@mail.gmail.com>

I got a lot of problem last week for making conference call. I was at home
(conference call starts at 2200hours PKST, my time) and unable to make SIP
call since the government has blocked it. So my only choice was Skype, but
unfortunately DTMF wasn't working, i get connected on Skypiax5 for about 7-8
times, but DTMF problem never let me get in to conference. Is there any
solution for this?

I really need to discuss a lot of things about FS documentation in
conference, like what FS community can expect from it and what not? how i
have planned it? how much progress has done? what are the problems me and my
team are facing (which has slow us down considerably)? etc. etc.

I have a solution for it but that needs testing. The plan is to use one of
my FS servers to connect my jingle calls from GTalk to conference server
over SIP. How can i test this setup with conference server, any ideas?

If any one else also interested in getting connected to weekly conference
call through this setup then i can also extend this setup as needed.

Thank you.


On Tue, Oct 6, 2009 at 1:30 AM, Brian West <brian at freeswitch.org> wrote:

> It always supported 48kHz CELT but the conference itself was running
> at 32kHz so everyone 48k had to be down sampled.  Now you all get to
> be up sampled.  w00t!
>
> /b
>
> On Oct 5, 2009, at 2:03 PM, Michael Collins wrote:
>
> > * Starting with the upcoming meeting (Oct 9) the conference will
> > support 48kHz CELT codec.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
________________________________________________________
|
                     |
| FATAL ERROR                                                           ---
O X |
|_______________________________________________________|
|                        You have moved the mouse.
 |
| Windows must be restarted for the changes to take effect.   |
|                                        <OK>
               |
####################################/


Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From srinivas.ksvreddy at gmail.com  Tue Oct  6 23:25:34 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Wed, 7 Oct 2009 11:55:34 +0530
Subject: [Freeswitch-users] mod_xml_curl
Message-ID: <f8af5740910062325u47da2813sc2ffc0cf9d688e44@mail.gmail.com>

Hi,

i want use mod_xml_curl, the xml files also there in my local system, i dont
want to take from any other system,
can any please tell me how to configure mox_xml.conf.xml,  how can i use
<bindings> to local folders

Thanks
Srinivasula Reddy K
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From srinivas.ksvreddy at gmail.com  Tue Oct  6 23:43:12 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Wed, 7 Oct 2009 12:13:12 +0530
Subject: [Freeswitch-users] mod_xml_curl
Message-ID: <f8af5740910062343q463fc2dt7e08cf610d28e351@mail.gmail.com>

Hi,

i want use mod_xml_curl, the xml files also there in my local system, i dont
want to take from any other system,
can any please tell me how to configure mox_xml.conf.xml,  how can i use
<bindings> to local folders

Thanks
Srinivasula Reddy K
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From brian at freeswitch.org  Tue Oct  6 23:52:16 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 7 Oct 2009 01:52:16 -0500
Subject: [Freeswitch-users] mod_xml_curl
In-Reply-To: <f8af5740910062343q463fc2dt7e08cf610d28e351@mail.gmail.com>
References: <f8af5740910062343q463fc2dt7e08cf610d28e351@mail.gmail.com>
Message-ID: <6F6B493F-6233-4249-B1AE-1628AB6333B6@freeswitch.org>

http://wiki.freeswitch.org/wiki/Mod_xml_curl  Is a great start.

/b

On Oct 7, 2009, at 1:43 AM, srinivasula reddy wrote:

> Hi,
>
> i want use mod_xml_curl, the xml files also there in my local  
> system, i dont want to take from any other system,
> can any please tell me how to configure mox_xml.conf.xml,  how can i  
> use <bindings> to local folders
>
> Thanks
> Srinivasula Reddy K _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From srinivas.ksvreddy at gmail.com  Wed Oct  7 00:28:37 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Wed, 7 Oct 2009 12:58:37 +0530
Subject: [Freeswitch-users] mod_xml_curl
In-Reply-To: <6F6B493F-6233-4249-B1AE-1628AB6333B6@freeswitch.org>
References: <f8af5740910062343q463fc2dt7e08cf610d28e351@mail.gmail.com>
	<6F6B493F-6233-4249-B1AE-1628AB6333B6@freeswitch.org>
Message-ID: <f8af5740910070028k3342b55fkc04e859008c212c5@mail.gmail.com>

Hi brain west,

Thank you very much for your reply,  i have gone through the link what u
have sent me, they have given how to access remote system through webserver,
but they have not given how to access from local system,

if u know please hellp me

Thanks

On Wed, Oct 7, 2009 at 12:22 PM, Brian West <brian at freeswitch.org> wrote:

> http://wiki.freeswitch.org/wiki/Mod_xml_curl  Is a great start.
> /b
>
> On Oct 7, 2009, at 1:43 AM, srinivasula reddy wrote:
>
> Hi,
>
> i want use mod_xml_curl, the xml files also there in my local system, i
> dont want to take from any other system,
> can any please tell me how to configure mox_xml.conf.xml,  how can i use
> <bindings> to local folders
>
> Thanks
> Srinivasula Reddy K _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Srinivasula Reddy K
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From jason at jasonjgw.net  Wed Oct  7 00:49:50 2009
From: jason at jasonjgw.net (Jason White)
Date: Wed, 7 Oct 2009 18:49:50 +1100
Subject: [Freeswitch-users] mod_xml_curl
In-Reply-To: <f8af5740910070028k3342b55fkc04e859008c212c5@mail.gmail.com>
References: <f8af5740910062343q463fc2dt7e08cf610d28e351@mail.gmail.com>
	<6F6B493F-6233-4249-B1AE-1628AB6333B6@freeswitch.org>
	<f8af5740910070028k3342b55fkc04e859008c212c5@mail.gmail.com>
Message-ID: <20091007074950.GA30931@jdc.jasonjgw.net>

srinivasula reddy <srinivas.ksvreddy at gmail.com> wrote:
> 
> Thank you very much for your reply,  i have gone through the link what u
> have sent me, they have given how to access remote system through webserver,
> but they have not given how to access from local system,

On that page, there is an example of how to use mod_curl to access a local
system. Do you know what localhost means?

You can't use a file:// URL, you have to use an HTTP URL and set up a (small)
HTTP server on localhost.




From moizchinoy at gmail.com  Wed Oct  7 00:53:02 2009
From: moizchinoy at gmail.com (Moiz Chinoy)
Date: Wed, 7 Oct 2009 11:53:02 +0400
Subject: [Freeswitch-users] Mobile Phone As GSM Gateway....
In-Reply-To: <23f91030910060504y3d060990l44ec913fb130954c@mail.gmail.com>
References: <29b888f80910060058m4298a829s414a322f8548ff81@mail.gmail.com>
	<23f91030910060504y3d060990l44ec913fb130954c@mail.gmail.com>
Message-ID: <29b888f80910070053j75b344c3n99c2d53045c08e00@mail.gmail.com>

Thanks for your replies....

gsmopen,org seems interesting but it does not have any documentation.
Can anyone point me where I can find information regarding this
project.


On Tue, Oct 6, 2009 at 4:04 PM, Seven Du <dujinfang at gmail.com> wrote:
> maybe you can check this: http://www.gsmopen.org/
>
> 2009/10/6 Moiz Chinoy <moizchinoy at gmail.com>
>>
>> Hi,
>>
>> Is it possible to connect a mobile phone (GSM phone) to Freeswitch and
>> use this as a GSM gateway?
>>
>> --
>> Regards,
>> Moiz Chinoy.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Regards,
Moiz Chinoy.



From gmaruzz at celliax.org  Wed Oct  7 01:16:18 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 7 Oct 2009 10:16:18 +0200
Subject: [Freeswitch-users] Mobile Phone As GSM Gateway....
In-Reply-To: <29b888f80910070053j75b344c3n99c2d53045c08e00@mail.gmail.com>
References: <29b888f80910060058m4298a829s414a322f8548ff81@mail.gmail.com> 
	<23f91030910060504y3d060990l44ec913fb130954c@mail.gmail.com> 
	<29b888f80910070053j75b344c3n99c2d53045c08e00@mail.gmail.com>
Message-ID: <7b197bef0910070116l7703f0fbk57244b0b763c933a@mail.gmail.com>

On Wed, Oct 7, 2009 at 9:53 AM, Moiz Chinoy <moizchinoy at gmail.com> wrote:
> Thanks for your replies....
>
> gsmopen,org seems interesting but it does not have any documentation.
> Can anyone point me where I can find information regarding this
> project.
>

:)
it is prealpha now, will available as alpha in couple week or so...

Docs will change a lot before being alpha, but... in the spirit of
openness... this is what is in the works :
http://wiki.freeswitch.org/wiki/GSMopen

>
> On Tue, Oct 6, 2009 at 4:04 PM, Seven Du <dujinfang at gmail.com> wrote:
>> maybe you can check this: http://www.gsmopen.org/
>>
>> 2009/10/6 Moiz Chinoy <moizchinoy at gmail.com>
>>>
>>> Hi,
>>>
>>> Is it possible to connect a mobile phone (GSM phone) to Freeswitch and
>>> use this as a GSM gateway?
>>>
>>> --
>>> Regards,
>>> Moiz Chinoy.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Regards,
> Moiz Chinoy.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From lyncker at lyth.de  Wed Oct  7 01:30:54 2009
From: lyncker at lyth.de (Filip Lyncker)
Date: Wed, 07 Oct 2009 10:30:54 +0200
Subject: [Freeswitch-users] how to Conference  with picup?
Message-ID: <4ACC51BE.6000201@lyth.de>

Hi List,

maybe someone can give me some hints to get faster into this stuff. Here 
the use case:

I am Ext A and talking to Ext B. Now Ext C is calling to another number 
: D.


How can I

1. Pic up the calling Ext C ?
2. Include it to my current call to Ext A ?


I would need some infos wich elements in my dialplan i have to use and 
how to configure them ... maybe you guys know some documentation else 
about this issue ...

thanks a lot


regards,

Filip

-- 
_________________________________
Filip Lyncker, Dipl.-Inform. (FH)


Lyncker & Theis GmbH
Wilhelmstr. 16
65185 Wiesbaden
Germany

Fon +49 611/9006951
Fax +49 611/9406125


Handelsregister: HRB 23156 Amtsgericht Wiesbaden
Steuernummer: 4023897051
USt-IdNr.: DE255806399

Gesch?ftsf?hrer:
Filip Lyncker,
Armin Theis 





From tculjaga at gmail.com  Wed Oct  7 01:37:50 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 7 Oct 2009 10:37:50 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <191c3a030910061758r2ddea5fand388f0af2c4dfc33@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<86a32abc0910061722t67ff23ddr62e4f62365539bd4@mail.gmail.com>
	<191c3a030910061758r2ddea5fand388f0af2c4dfc33@mail.gmail.com>
Message-ID: <65d96fc80910070137k52be89e3g846e55db72a8de5d@mail.gmail.com>

Anthony,

of course, nobody wants to start anything... we are all here to help making
FS a better product.

so, regarding the founding for mod_opal ... what is the amount you need?


Tihomir.



On Wed, Oct 7, 2009 at 2:58 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> I didn't mean to start anything.  I'm just saying we work very long hours
> and barely get anybody asking about h.323.
> I wanted to support it and that's why we took up a collection to get
> funding for mod_opal but when only 1 donor showed any interest we were
> forced to proceed in our spare time which is very limited.
>
> The developers of opal are not part of our project and they need financial
> compensation to be motivated to work on it.  Its not even related to me its
> only fair that an outside developer who makes his living as a consultant
> would want money to integrate his work into our project.
>
> Like I said, I will do my best to point your issue to the opal devs but I
> cannot force them to work on it.
>
>
>
>
>
> On Tue, Oct 6, 2009 at 7:22 PM, Diego Viola <diego.viola at gmail.com> wrote:
>
>> Yeah I understand your point of view, but saying "I want a H.323 module"
>> or "I want a Ferrari" wont magically make it happen.
>>
>> We need to work on it ourselves or pay to the people that knows how to do
>> it to do it for us.
>>
>> There is no other way I think.
>>
>> Diego
>>
>>
>>
>>
>> On Tue, Oct 6, 2009 at 11:41 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>
>>> Diego,
>>>
>>> what i'm pointing here is the situation where you have a great product
>>> that lacks in one of most common protocol. It is true H323 is going to
>>> disappear (eventually), it is true that the community prefers SIP/IAX
>>> instead ... but the reality still remains. H323 is going to be used for
>>> quite a long time to exchange a lot of traffic while FS will be left aside.
>>> Today, when you setup an IP peering interconnection 80% of carriers will
>>> prefer H323.
>>>
>>> Of course, developing something costs "time" (and we all know what time
>>> stands for...) and as i said, i understand the financial point of view and i
>>> really understand if nobody is going to work on that, but let's face it FS
>>> doesn't have any usable module to reliably handle H323 protocol.
>>>
>>>
>>> said that, i don't intend to offend anyone... just facing the reality.
>>>
>>>
>>> regarding the h323 module, we don't have any issue fixing the existing or
>>> developing a new one... but before we go developing something it is always
>>> better check if the thing you want already exists in an usable state or
>>> not... that's what i did today.
>>>
>>>
>>> So, I'm interested in a reliable module handling H323v4... anyone else?
>>>
>>>
>>> T.
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Tue, Oct 6, 2009 at 11:45 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>
>>>> Instead of complaining and demanding things for free, people should
>>>> start to put their money where their mouth is.
>>>>
>>>> Diego
>>>>
>>>>
>>>> On Tue, Oct 6, 2009 at 8:47 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>>>
>>>>> hi Anthony,
>>>>>
>>>>> it is somewhere here:
>>>>>
>>>>>          switch_status_t
>>>>> FSConnection::receive_message(switch_core_session_message_t *msg)
>>>>>
>>>>>
>>>>> anyhow, i will open an issue jira of course.
>>>>>
>>>>>
>>>>> I understand your financial point of view, but anyhow while the entire
>>>>> world is wants sip and trying to move to sip, the reality is quite
>>>>> different. The majority of voice traffic exchanged via IP is still H323.
>>>>> This means a working SIP - H323 interworking is really needed... pity nobody
>>>>> wants/has time to work in this direction to produce a decent mod_h323.
>>>>>
>>>>>
>>>>>
>>>>> T.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Oct 6, 2009 at 9:25 PM, Anthony Minessale <
>>>>> anthony.minessale at gmail.com> wrote:
>>>>>
>>>>>> pcap is not as useful as FS console log on debug with:
>>>>>> sofia profile internal siptrace on
>>>>>>
>>>>>> you should be reporting issues to jira under mod_opal not to the
>>>>>> mailing list.
>>>>>> http://jira.freeswitch.org
>>>>>>
>>>>>> FYI
>>>>>> There is little financial support from the community for h323 which
>>>>>> prevents the mod_opal from getting much attention.
>>>>>> We actually have to contract the author of opal to help with these
>>>>>> issues including the original writing of the module that he did with very
>>>>>> little funding and nobody ever wants to pay him to improve it.
>>>>>>
>>>>>> That does not mean your issue will not be addressed but there is no
>>>>>> promise how fast it will be.
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Tue, Oct 6, 2009 at 12:37 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>>>>>
>>>>>>> hello guys,
>>>>>>>
>>>>>>>
>>>>>>> i was playing with mod_opal to see if i can make it working ... well
>>>>>>> it seems SIP-H323 interworking is not tuned at all.
>>>>>>>
>>>>>>> I have a call from a registered sip user (1001) to PSTN via mod_opal
>>>>>>>
>>>>>>>
>>>>>>> <include>
>>>>>>>   <extension name="EMERGENCY">
>>>>>>>     <condition field="destination_number"
>>>>>>> expression="^0(112|9[23456])$">
>>>>>>>       <action application="set"
>>>>>>> data="effective_caller_id_number=1001282122"/>
>>>>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>>>>>       <action application="set" data="call_timeout=30"/>
>>>>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>>>>
>>>>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>>>>>     </condition>
>>>>>>>   </extension>
>>>>>>>
>>>>>>>   <extension name="SPECIAL_SERVICES">
>>>>>>>     <condition field="destination_number"
>>>>>>> expression="^0(9[01789]\d{3,4})$">
>>>>>>>       <action application="set"
>>>>>>> data="effective_caller_id_number=1001282122"/>
>>>>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>>>>>       <action application="set" data="call_timeout=30"/>
>>>>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>>>>
>>>>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>>>>>     </condition>
>>>>>>>   </extension>
>>>>>>>
>>>>>>>   <extension name="ENYTHING_ELSE">
>>>>>>>     <condition field="destination_number"
>>>>>>> expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
>>>>>>>       <action application="set"
>>>>>>> data="effective_caller_id_number=1001282122"/>
>>>>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>>>>>       <action application="set" data="call_timeout=30"/>
>>>>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>>>>
>>>>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>>>>>     </condition>
>>>>>>>   </extension>
>>>>>>> </include>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> One of the many issues i sow is that FS connects the call on SIP leg
>>>>>>> before it actually receives H.225 connect from H323 leg... as it is
>>>>>>> configured to send 200 OK on the 1st H.225 message containing a FastStart
>>>>>>> element/OLC.
>>>>>>>
>>>>>>>
>>>>>>> Attached is the tcpdump i took on FS machine... just use this filter:
>>>>>>> "h225 or h245 or q931 or sip"
>>>>>>> Also, you can check the attac CDR,,,, this is an unanswered call i
>>>>>>> placed to PSTN and FS billed it 23 seconds.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Can anyone tell where i can do correct SIP - H323 message mappings to
>>>>>>> avoid this?
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> T.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Anthony Minessale II
>>>>>>
>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>> ClueCon http://www.cluecon.com/
>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>
>>>>>> AIM: anthm
>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>
>>>>>> FreeSWITCH Developer Conference
>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>> pstn:213-799-1400
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lakindia89 at gmail.com  Wed Oct  7 03:00:24 2009
From: lakindia89 at gmail.com (lakshmanan)
Date: Wed, 7 Oct 2009 03:00:24 -0700 (PDT)
Subject: [Freeswitch-users] oz debug says error
In-Reply-To: <87f2f3b90910060906y4b087d7coa91de38d78a3456b@mail.gmail.com>
References: <7d79b3930910050020i1b96541dk26c8d72c220ded8d@mail.gmail.com>
	<9F7D1E871F0C4184AA8ED441CF415C8A@cune.pri>
	<25749736.post@talk.nabble.com>
	<20091005124858.81857415806@mail.cune.org>
	<7d79b3930910052106n59cf9cbema68e4d6ccc274034@mail.gmail.com>
	<87f2f3b90910060906y4b087d7coa91de38d78a3456b@mail.gmail.com>
Message-ID: <25783750.post@talk.nabble.com>


Here are the details

openzap.conf
[span zt PRI_1]
trunk_type => e1
b-channel => 1:1-15
d-channel=> 1:16
b-channel => 1:17-31

Zaptel.conf

span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16

loadzone        = us
defaultzone     = us



mercutioviz wrote:
> 
> Pastebin your openzap.conf file. Also, is this Sangoma or zaptel-based
> hardware? If it's Sangoma, pastebin your wanpipe1.conf file. If zaptel,
> please paste your zaptel.conf file.
> 
> -MC
> 
> On Mon, Oct 5, 2009 at 9:06 PM, lakshmanan ganapathy
> <lakindia89 at gmail.com>wrote:
> 
>> Openzap.conf.xml
>> <configuration name="openzap.conf" description="OpenZAP Configuration">
>>         <settings>
>>                 
>>         </settings>
>>         <libpri_spans>
>>                 
>>                         
>>                         
>>                         
>>                         
>>                 
>>         </libpri_spans>
>> </configuration>
>>
>> Output of oz list in fs_cli
>>
>> span: 1 (PRI_1)
>> type: isdn
>> chan_count: 47
>> dialplan: XML
>> context: default
>> dial_regex:
>> fail_dial_regex:
>> hold_music:
>> analog_options none
>>
>>
>> freeswitch.log
>>
>> http://pastebin.freeswitch.org/10604
>>
>>
>>
>>
>>
>> On Mon, Oct 5, 2009 at 6:18 PM, <Russell.Mosemann at cune.org> wrote:
>>
>>> lakshmanan <lakindia89 at gmail.com> said:
>>> > Thanks for pointing that.
>>> > I also tried that.
>>> > But in that case, I'm not able to make a call through openzap.
>>>
>>> What is in openzap.conf.xml? If you start fs_cli and enter "oz list",
>>> what does it show? Copy the ozmod lines from freeswitch.log to
>>> pastebin.freeswitch.org and post the link here so that we can see what
>>> openzap does when freeswitch starts.
>>>
>>> --
>>> Russell Mosemann
>>>
>>>
>>>
>>> ________________________________________________________
>>> Concordia University, Nebraska
>>> See http://www.cune.edu/ for the latest news and events!
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/oz-debug-says-error-tp25746215p25783750.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From lakindia89 at gmail.com  Wed Oct  7 03:34:50 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Wed, 7 Oct 2009 16:04:50 +0530
Subject: [Freeswitch-users] openzap Failure opening channel error
Message-ID: <7d79b3930910070334g53cef0c5m63037e264f1a9233@mail.gmail.com>

Hi,
Again I was struck in a problem, Here is the scenario.

On incomming call, I just call an event outboud socket. But what happens is,
for the first 15 call, it is working fine. But from the 16th call to 30th
call, it says the below error.

2009-10-07 15:07:48.201846 [WARNING] ozmod_libpri.c:761 --Failure opening
channel 1:17 (ignored)
2009-10-07 15:07:55.381861 [WARNING] ozmod_libpri.c:761 --Failure opening
channel 1:18 (ignored)
2009-10-07 15:07:58.569774 [WARNING] ozmod_libpri.c:761 --Failure opening
channel 1:19 (ignored)
2009-10-07 15:08:01.37824 [WARNING] ozmod_libpri.c:761 --Failure opening
channel 1:20 (ignored)
2009-10-07 15:08:03.129846 [WARNING] ozmod_libpri.c:761 --Failure opening
channel 1:21 (ignored)
2009-10-07 15:08:04.825851 [WARNING] ozmod_libpri.c:761 --Failure opening
channel 1:22 (ignored)
2009-10-07 15:08:06.289977 [WARNING] ozmod_libpri.c:761 --Failure opening
channel 1:23 (ignored)
2009-10-07 15:08:07.761961 [WARNING] ozmod_libpri.c:761 --Failure opening
channel 1:24 (ignored)
2009-10-07 15:08:09.737944 [WARNING] ozmod_libpri.c:761 --Failure opening
channel 1:25 (ignored)
2009-10-07 15:08:11.462018 [WARNING] ozmod_libpri.c:761 --Failure opening
channel 1:26 (ignored)
2009-10-07 15:08:13.566024 [WARNING] ozmod_libpri.c:761 --Failure opening
channel 1:27 (ignored)
2009-10-07 15:08:15.430163 [WARNING] ozmod_libpri.c:761 --Failure opening
channel 1:28 (ignored)
2009-10-07 15:08:17.446103 [WARNING] ozmod_libpri.c:761 --Failure opening
channel 1:29 (ignored)
2009-10-07 15:08:19.430118 [WARNING] ozmod_libpri.c:761 --Failure opening
channel 1:30 (ignored)
2009-10-07 15:08:21.358121 [WARNING] ozmod_libpri.c:761 --Failure opening
channel 1:31 (ignored)


But for the next call, when it is opening channel 1:1, it is executing my
dial plans.
I don't know why it failed when it is choosing 1:17-1:31. Any one has any
idea.

Below are my configuration details:

openzap.conf
[span zt PRI_1]
trunk_type => e1
b-channel => 1:1-15
d-channel=> 1:16
b-channel => 1:17-31

openzap.conf.xml
<configuration name="openzap.conf" description="OpenZAP Configuration">
        <settings>
                <param name="debug" value="7"/>
        </settings>
        <libpri_spans>
                <span name="PRI_1">
                        <param name="node" value="cpe"/>
                        <param name="switch" value="euroisdn"/>
                        <param name="dialplan" value="XML"/>
                        <param name="context" value="default"/>
                </span>
        </libpri_spans>
</configuration>

zaptel.conf

span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16

loadzone        = us
defaultzone     = us

oz list

span: 1 (PRI_1)
type: isdn
chan_count: 47
dialplan: XML
context: default
dial_regex:
fail_dial_regex:
hold_music:
analog_options none
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From woodydickson at gmail.com  Wed Oct  7 03:40:24 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Wed, 7 Oct 2009 18:40:24 +0800
Subject: [Freeswitch-users] unable to configure Digium TDM400P
Message-ID: <a44688a0910070340p4209e7b5yec9218b8c4fc692c@mail.gmail.com>

Hi,

I am trying to setup a Digium TDM400P following the instruction on the
wiki.
It is a 1 fxo and 1 fxs card, so I tried
loadzone=in
defaultzone=in
fxsks=2
fxoks=1

and

loadzone=in
defaultzone=in
fxsks=1
fxoks=2

None works.  Does anyone know how it should be configured?

Here is what I get by following the wiki.

[root at localhost zaptel]# ztcfg -vv

Zaptel Version: SVN-branch-1.4-r4629M
Echo Canceller: MG2
Configuration
======================


Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Slaves: 02)

2 channels to configure.

Changing signalling on channel 1 from FXO Kewlstart to FXS Kewlstart
Changing signalling on channel 2 from FXS Kewlstart to FXO Kewlstart


[root at localhost zaptel]# ztcfg -vv

Zaptel Version: SVN-branch-1.4-r4629M
Echo Canceller: MG2
Configuration
======================


Channel map:

Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)

2 channels to configure.

Changing signalling on channel 1 from FXS Kewlstart to FXO Kewlstart
Changing signalling on channel 2 from FXO Kewlstart to FXS Kewlstart


[root at localhost zaptel]# lspci
00:14.0 Ethernet controller: Digium, Inc. TDM400P (rev 11)
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From Russell.Mosemann at cune.org  Wed Oct  7 04:59:39 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Wed, 7 Oct 2009 11:59:39 -0000
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <7d79b3930910070334g53cef0c5m63037e264f1a9233@mail.gmail.com>
Message-ID: <20091007115939.4856239DEA0@mail.cune.org>

lakshmanan ganapathy <lakindia89 at gmail.com> said:

> On incomming call, I just call an event outboud socket. But what
happens is,
> for the first 15 call, it is working fine. But from the 16th call to 30th
> call, it says the below error.

What is displayed with

  ztcfg -vv

What is displayed with

  cat /proc/zaptel/1

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!




From lakindia89 at gmail.com  Wed Oct  7 05:12:23 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Wed, 7 Oct 2009 17:42:23 +0530
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <20091007115939.4856239DEA0@mail.cune.org>
References: <7d79b3930910070334g53cef0c5m63037e264f1a9233@mail.gmail.com>
	<20091007115939.4856239DEA0@mail.cune.org>
Message-ID: <7d79b3930910070512n5f2f795fob19eece48e2a872@mail.gmail.com>

[debian :~]# ztcfg -vvv
Zaptel Version: 1.4.11
Echo Canceller: MG2
Configuration
======================

SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)

Channel map:

Channel 01: Clear channel (Default) (Slaves: 01)
Channel 02: Clear channel (Default) (Slaves: 02)
Channel 03: Clear channel (Default) (Slaves: 03)
Channel 04: Clear channel (Default) (Slaves: 04)
Channel 05: Clear channel (Default) (Slaves: 05)
Channel 06: Clear channel (Default) (Slaves: 06)
Channel 07: Clear channel (Default) (Slaves: 07)
Channel 08: Clear channel (Default) (Slaves: 08)
Channel 09: Clear channel (Default) (Slaves: 09)
Channel 10: Clear channel (Default) (Slaves: 10)
Channel 11: Clear channel (Default) (Slaves: 11)
Channel 12: Clear channel (Default) (Slaves: 12)
Channel 13: Clear channel (Default) (Slaves: 13)
Channel 14: Clear channel (Default) (Slaves: 14)
Channel 15: Clear channel (Default) (Slaves: 15)
Channel 16: D-channel (Default) (Slaves: 16)
Channel 17: Clear channel (Default) (Slaves: 17)
Channel 18: Clear channel (Default) (Slaves: 18)
Channel 19: Clear channel (Default) (Slaves: 19)
Channel 20: Clear channel (Default) (Slaves: 20)
Channel 21: Clear channel (Default) (Slaves: 21)
Channel 22: Clear channel (Default) (Slaves: 22)
Channel 23: Clear channel (Default) (Slaves: 23)
Channel 24: Clear channel (Default) (Slaves: 24)
Channel 25: Clear channel (Default) (Slaves: 25)
Channel 26: Clear channel (Default) (Slaves: 26)
Channel 27: Clear channel (Default) (Slaves: 27)
Channel 28: Clear channel (Default) (Slaves: 28)
Channel 29: Clear channel (Default) (Slaves: 29)
Channel 30: Clear channel (Default) (Slaves: 30)
Channel 31: Clear channel (Default) (Slaves: 31)

31 channels to configure.

[debian :~]# cat /proc/zaptel/1
+ cat /proc/zaptel/1
Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS ClockSource
        Timing slips: 1

        1 TE2/0/1/1 Clear
        2 TE2/0/1/2 Clear
        3 TE2/0/1/3 Clear
        4 TE2/0/1/4 Clear
        5 TE2/0/1/5 Clear
        6 TE2/0/1/6 Clear
        7 TE2/0/1/7 Clear
        8 TE2/0/1/8 Clear
        9 TE2/0/1/9 Clear
        10 TE2/0/1/10 Clear
        11 TE2/0/1/11 Clear
        12 TE2/0/1/12 Clear
        13 TE2/0/1/13 Clear
        14 TE2/0/1/14 Clear
        15 TE2/0/1/15 Clear
        16 TE2/0/1/16 HDLCFCS
        17 TE2/0/1/17 Clear
        18 TE2/0/1/18 Clear
        19 TE2/0/1/19 Clear
        20 TE2/0/1/20 Clear
        21 TE2/0/1/21 Clear
        22 TE2/0/1/22 Clear
        23 TE2/0/1/23 Clear
        24 TE2/0/1/24 Clear
        25 TE2/0/1/25 Clear
        26 TE2/0/1/26 Clear
        27 TE2/0/1/27 Clear
        28 TE2/0/1/28 Clear
        29 TE2/0/1/29 Clear
        30 TE2/0/1/30 Clear
        31 TE2/0/1/31 Clear


On Wed, Oct 7, 2009 at 5:29 PM, <Russell.Mosemann at cune.org> wrote:

> lakshmanan ganapathy <lakindia89 at gmail.com> said:
>
> > On incomming call, I just call an event outboud socket. But what
> happens is,
> > for the first 15 call, it is working fine. But from the 16th call to 30th
> > call, it says the below error.
>
> What is displayed with
>
>  ztcfg -vv
>
> What is displayed with
>
>  cat /proc/zaptel/1
>
> --
> Russell Mosemann
>
>
>
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ahmedmunir007 at gmail.com  Wed Oct  7 05:34:54 2009
From: ahmedmunir007 at gmail.com (Ahmed Munir)
Date: Wed, 7 Oct 2009 17:34:54 +0500
Subject: [Freeswitch-users] Configuring Nibble Bill
Message-ID: <a60fed5e0910070534v6e00bad9r518c07c002e0ee6e@mail.gmail.com>

Hi,

Thanks for replying. The advise u gave resolved my problem. I want to ask
three questions related to nibble bills, as I 'm listing down below;

1- Can we select/use dynamic tables for billing using nibble bill?
2- Can we define more than two tables and attributes in nibblebill.conf.xml?
3- As Nibble bill is use to deduct amount of user account, Can we deduct
minutes instead of cash? Because my case is, if a user buy a package and I
only want to deducts his/her minutes. How we can resolve it by nibble bill?
/ What other way we can resolve it?

Kindly advise soon.



>
> ---------- Forwarded message ----------
> From: Todd Baumgartner <todd.baumgartner at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Tue, 6 Oct 2009 10:31:06 -0400
> Subject: Re: [Freeswitch-users] Cannot connect to ODBC driver/database
> freeswitchdb
> Ahmed,
>
> I believe you need to specify the database name as it is configured in the
> odbc.ini
>
> I am assuming you have something like this in your nibblebill.conf.xml
>
>     <param name="db_dsn" value="freeswitchdb"/>
>
> Try changing it to this as it is named in the odbc.ini:
>
>     <param name="db_dsn" value="MySQL-freeswitch"/>
>
>
> Thanks,
>
> Todd
>
>
>
> On Tue, Oct 6, 2009 at 6:11 AM, Ahmed Munir <ahmedmunir007 at gmail.com>wrote:
>
>> Hi,
>> I've installed FS on Ubuntu 9.04 and I want to run mod_nibbles on it. I
>> follow the steps to configure my ODBC connection with MySQL as explained in
>> wiki (mod_nibbles and mod_spidermonkey). But  FS, unable to connect it. The
>> error I got is listed below when I restart FS,
>>
>> 2009-10-06 15:47:21.164590 [ERR] switch_odbc.c:188 STATE: IM002 CODE 0
>> ERROR: [unixODBC][Driver Manager]Data source name not found, and no default
>> driver specified
>> 2009-10-06 15:47:21.164617 [CRIT] mod_nibblebill.c:221 Cannot connect to
>> ODBC driver/database freeswitchdb (user: root / pass password)
>> 2009-10-06 15:47:21.164650 [CONSOLE] switch_loadable_module.c:889
>> Successfully Loaded [mod_nibblebill]
>> 2009-10-06 15:47:21.164664 [NOTICE] switch_loadable_module.c:248 Adding
>> Application 'nibblebill'
>> 2009-10-06 15:47:21.164710 [NOTICE] switch_loadable_module.c:270 Adding
>> API Function 'nibblebill'
>>
>> But when I use isql it accepts my odbc connection i.e. isql
>> MySQL-freeswitch
>>
>> I'm listing my settings of odbc.ini and odbcinst.ini as listed below;
>>
>> odbc.ini
>> --------------
>> [MySQL-freeswitch]
>> Driver       = MySQL
>> #Driver       = /usr/lib/odbc/libodbcmyS.so
>> Description  = Connector/ODBC Driver DSN With FreeSwitch
>> SERVER       = localhost
>> PORT         = 3306
>> USER         = root
>> Password     = password
>> Database     = freeswitchdb
>>
>> odbcinst.ini
>> -------------------
>> [MySQL]
>> Description    = ODBC for MySQL
>> Driver         = /usr/lib/odbc/libmyodbc.so
>> Setup          = /usr/lib/odbc/libodbcmyS.so
>> FileUsage      = 1
>>
>>
>> odbc.ini and odbcinst.ini are located at /etc/. Even I set my odbc
>> connection setting as I provide with this link;
>> http://dev.mysql.com/doc/refman/5.0/en/connector-odbc-configuration-dsn-unix.html
>>
>> But unfortunately my problem is unresolved then.
>>
>>
>> Kindly advise me, how can I resolve this problem?
>>
>> --
>> Regards,
>>
>> Ahmed Munir
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,

Ahmed Munir
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From maciej.aniserowicz at gmail.com  Wed Oct  7 00:39:52 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Wed, 7 Oct 2009 00:39:52 -0700 (PDT)
Subject: [Freeswitch-users] Bad sound quality while eavesdropping
In-Reply-To: <191c3a030910050731m2d74979ep4598e5a1945d58ae@mail.gmail.com>
References: <41A44DD027064988A914974405788C2E@procent>
	<191c3a030910050731m2d74979ep4598e5a1945d58ae@mail.gmail.com>
Message-ID: <1254901192035-3780245.post@n2.nabble.com>


Sorry about posting several questions at once, I wasn't aware it's "rude". 
Let's concentrate on this issue then. 

I use FS rev 14994. Phones on extensions: 
1) x-lite 
2) cisco sip phone 
3) audio played by fs to the extension being eavesdropped 

I did not change any codec configuration, I just use the standard one that
comes with both FS and the phones. 
Some time ago someone on FS irc channel told me that this is just how FS
eavesdropping works... from your response I understand that this is not
entirely true? 

Maciej Aniserowicz



Anthony Minessale wrote:
> 
> That's is a somewhat vague position.
> 
> You did not mention which version of FreeSWITCH you are running, the
> phones
> being used in your example, your configuration, the codecs in use etc.
> 
> BTW,
> I think you should only ask one question at a time on this list.  The list
> is run by volunteers and it's sort of rude to expect 3 or 4 threads to be
> tended to concerning the same one individual.
> 
> 
> 2009/10/5 Maciej Aniserowicz <maciej.aniserowicz at gmail.com>
> 
>>  Hello,
>> When I use eavesdropping in FreeSWITCH, the sound quality is really bad.
>> Is
>> there any way to improve it? Is this a known problem?
>> Br/
>> Maciej Aniserowicz
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3780245.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From claudiu at globtel.ro  Wed Oct  7 05:40:57 2009
From: claudiu at globtel.ro (Claudiu Filip)
Date: Wed, 7 Oct 2009 15:40:57 +0300
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910070137k52be89e3g846e55db72a8de5d@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<86a32abc0910061722t67ff23ddr62e4f62365539bd4@mail.gmail.com>
	<191c3a030910061758r2ddea5fand388f0af2c4dfc33@mail.gmail.com>
	<65d96fc80910070137k52be89e3g846e55db72a8de5d@mail.gmail.com>
Message-ID: <14480284.20091007154057@globtel.ro>




Hi Tihomir,


   I've  done  some  tests  to  see  how  suitable  is freeswitch as a
   SIP/H323  translator  and  you  are  right about the fact that H323
   'alert+open  logical  channel'  will generate a SIP '200 OK'. I was
   able  to fix that with a couple of changes in mod_opal.cpp, however
   some  things  were changed on mod_sofia in the latest svn. (on this
   particular  issue,  open_logical_channel  is  processed  BEFORE the
   alerting,   so   the   call   is   in   SetupPhase  when  the  proc
   OnOpenMediaStream is triggered)
   
   
   The  most  important  problem  I'm having right now is that G729 is
   still  not  working (poor quality due to high buffering). Even with
   the  latest  Opal,  which  includes  the last week patch for jitter
   buffer.  If  you  dont  need G729, I could send you a patch for the
   latest svns (freeswitch, opal, ptlib), ofc no founding needed.
   
   There are a couple of bugs in opal itself and h323ing freeswitch
   with opal will bring them in. On the other hand, mod_opal is
   already there, it just needs a few adjustments.

   
Best wishes,

Claudiu Filip




Wednesday, October 7, 2009, 11:37:50 AM, you wrote:
Tihomir> Anthony, 

Tihomir> of course, nobody wants to start anything... we are all here
Tihomir> to help making FS a better product.

Tihomir> so, regarding the founding for mod_opal ... what is the amount you need?


Tihomir> Tihomir.


Tihomir>  
Tihomir>  On Wed, Oct 7, 2009 at 2:58 AM, Anthony Minessale
Tihomir> <anthony.minessale at gmail.com> wrote:
Tihomir>  I didn't mean to start anything.? I'm just saying we work
Tihomir> very long hours and barely get anybody asking about h.323.
Tihomir> I wanted to support it and that's why we took up a
Tihomir> collection to get funding for mod_opal but when only 1 donor
Tihomir> showed any interest we were forced to proceed in our spare time which is very limited.
Tihomir>   
Tihomir> The developers of opal are not part of our project and they
Tihomir> need financial compensation to be motivated to work on it.?
Tihomir> Its not even related to me its only fair that an outside
Tihomir> developer who makes his living as a consultant would want
Tihomir> money to integrate his work into our project.
Tihomir>   
Tihomir> Like I said, I will do my best to point your issue to the
Tihomir> opal devs but I cannot force them to work on it.





Tihomir> On Tue, Oct 6, 2009 at 7:22 PM, Diego Viola
Tihomir> <diego.viola at gmail.com> wrote:
Tihomir>   
Tihomir> Yeah I understand your point of view, but saying "I want a
Tihomir> H.323 module" or "I want a Ferrari" wont magically make it happen.
Tihomir>   
Tihomir> We need to work on it ourselves or pay to the people that
Tihomir> knows how to do it to do it for us.
Tihomir>  
Tihomir> There is no other way I think.

Tihomir> Diego




Tihomir> On Tue, Oct 6, 2009 at 11:41 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
Tihomir>   
Tihomir> Diego,

Tihomir> what i'm pointing here is the situation where you have a
Tihomir> great product that lacks in one of most common protocol. It
Tihomir> is true H323 is going to disappear (eventually), it is true
Tihomir> that the community prefers SIP/IAX instead ... but the
Tihomir> reality still remains. H323 is going to be used for quite a
Tihomir> long time to exchange a lot of traffic while FS will be left
Tihomir> aside. Today, when you setup an IP peering interconnection
Tihomir> 80% of carriers will prefer H323.
Tihomir>   
Tihomir> Of course, developing something costs "time" (and we all
Tihomir> know what time stands for...) and as i said, i understand the
Tihomir> financial point of view and i really understand if nobody is
Tihomir> going to work on that, but let's face it FS doesn't have any
Tihomir> usable module to reliably handle H323 protocol. 
Tihomir>   

Tihomir> said that, i don't intend to offend anyone... just facing the reality.


Tihomir> regarding the h323 module, we don't have any issue fixing
Tihomir> the existing or developing a new one... but before we go
Tihomir> developing something it is always better check if the thing
Tihomir> you want already exists in an usable state or not... that's what i did today.
Tihomir>   

Tihomir> So, I'm interested in a reliable module handling H323v4... anyone else?


Tihomir> T.






Tihomir> On Tue, Oct 6, 2009 at 11:45 PM, Diego Viola
Tihomir> <diego.viola at gmail.com> wrote:
Tihomir>   
Tihomir> Instead of complaining and demanding things for free, people
Tihomir> should start to put their money where their mouth is.
Tihomir>   
Tihomir> Diego


Tihomir> On Tue, Oct 6, 2009 at 8:47 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
Tihomir>   
Tihomir> hi Anthony,

Tihomir> it is somewhere here:

Tihomir> ? ? ? ?? switch_status_t
Tihomir> FSConnection::receive_message(switch_core_session_message_t *msg)
Tihomir>   

Tihomir> anyhow, i will open an issue jira of course.


Tihomir> I understand your financial point of view, but anyhow while
Tihomir> the entire world is wants sip and trying to move to sip, the
Tihomir> reality is quite different. The majority of voice traffic
Tihomir> exchanged via IP is still H323. This means a working SIP -
Tihomir> H323 interworking is really needed... pity nobody wants/has
Tihomir> time to work in this direction to produce a decent mod_h323. 
Tihomir>    


Tihomir> T.



Tihomir> ?


Tihomir> On Tue, Oct 6, 2009 at 9:25 PM, Anthony Minessale
Tihomir> <anthony.minessale at gmail.com> wrote:
Tihomir>   
Tihomir> pcap is not as useful as FS console log on debug with:
Tihomir> sofia profile internal siptrace on
Tihomir>   
Tihomir> you should be reporting issues to jira under mod_opal not to the mailing list.
Tihomir> http://jira.freeswitch.org
Tihomir>  
Tihomir> FYI
Tihomir> There is little financial support from the community for
Tihomir> h323 which prevents the mod_opal from getting much attention.
Tihomir> We actually have to contract the author of opal to help with
Tihomir> these issues including the original writing of the module
Tihomir> that he did with very little funding and nobody ever wants to pay him to improve it.
Tihomir>   
Tihomir> That does not mean your issue will not be addressed but
Tihomir> there is no promise how fast it will be.



Tihomir> On Tue, Oct 6, 2009 at 12:37 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
Tihomir>   
Tihomir> hello guys,


Tihomir>  i was playing with mod_opal to see if i can make it working
Tihomir> ... well it seems SIP-H323 interworking is not tuned at all.
Tihomir>  
Tihomir> I have a call from a registered sip user (1001) to PSTN via mod_opal
Tihomir>   

Tihomir> <include>
Tihomir> ? <extension name="EMERGENCY">
Tihomir> ??? <condition field="destination_number"
Tihomir> expression="^0(112|9[23456])$">
Tihomir> ????? <action application="set"
Tihomir> data="effective_caller_id_number=1001282122"/>
Tihomir>  ????? <action application="set" data="NCX_IP=10.4.4.254"/>
Tihomir> ????? <action application="set" data="call_timeout=30"/>
Tihomir> ????? <action application="set" data="hangup_after_bridge=true"/>
Tihomir>   
Tihomir> ????? <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
Tihomir> ??? </condition>
Tihomir> ? </extension>

Tihomir> ? <extension name="SPECIAL_SERVICES">
Tihomir> ??? <condition field="destination_number"
Tihomir> expression="^0(9[01789]\d{3,4})$">
Tihomir>  ????? <action application="set"
Tihomir> data="effective_caller_id_number=1001282122"/>
Tihomir> ????? <action application="set" data="NCX_IP=10.4.4.254"/>
Tihomir> ????? <action application="set" data="call_timeout=30"/>
Tihomir>  ????? <action application="set" data="hangup_after_bridge=true"/>

Tihomir> ????? <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
Tihomir> ??? </condition>
Tihomir>  ? </extension>

Tihomir> ? <extension name="ENYTHING_ELSE">
Tihomir> ??? <condition field="destination_number"
Tihomir> expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
Tihomir> ????? <action application="set"
Tihomir> data="effective_caller_id_number=1001282122"/>
Tihomir>  ????? <action application="set" data="NCX_IP=10.4.4.254"/>
Tihomir> ????? <action application="set" data="call_timeout=30"/>
Tihomir> ????? <action application="set" data="hangup_after_bridge=true"/>
Tihomir>   
Tihomir> ????? <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
Tihomir> ??? </condition>
Tihomir> ? </extension>
Tihomir> </include>



Tihomir> One of the many issues i sow is that FS connects the call on
Tihomir> SIP leg before it actually receives H.225 connect from H323
Tihomir> leg... as it is configured to send 200 OK on the 1st H.225
Tihomir> message containing a FastStart element/OLC.
Tihomir>   

Tihomir> Attached is the tcpdump i took on FS machine... just use
Tihomir> this filter: "h225 or h245 or q931 or sip"
Tihomir> Also, you can check the attac CDR,,,, this is an unanswered
Tihomir> call i placed to PSTN and FS billed it 23 seconds.
Tihomir>   


Tihomir> Can anyone tell where i can do correct SIP - H323 message mappings to avoid this?



Tihomir> T.


Tihomir>  

Tihomir>  
Tihomir> _______________________________________________
Tihomir>  FreeSWITCH-users mailing list
Tihomir>  FreeSWITCH-users at lists.freeswitch.org
Tihomir>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Tihomir> 
Tihomir> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Tihomir>  http://www.freeswitch.org
Tihomir>  




Tihomir> -- 
Tihomir> Anthony Minessale II

Tihomir> FreeSWITCH http://www.freeswitch.org/
Tihomir> ClueCon http://www.cluecon.com/
Tihomir>  Twitter: http://twitter.com/FreeSWITCH_wire

Tihomir> AIM: anthm
Tihomir> MSN:anthony_minessale at hotmail.com
Tihomir>  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
Tihomir>  IRC: irc.freenode.net #freeswitch

Tihomir> FreeSWITCH Developer Conference
Tihomir> sip:888 at conference.freeswitch.org
Tihomir>   iax:guest at conference.freeswitch.org/888
Tihomir>  googletalk:conf+888 at conference.freeswitch.org
Tihomir> pstn:213-799-1400
Tihomir>  
Tihomir> _______________________________________________
Tihomir>  FreeSWITCH-users mailing list
Tihomir>  FreeSWITCH-users at lists.freeswitch.org
Tihomir>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Tihomir> 
Tihomir> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Tihomir>  http://www.freeswitch.org
Tihomir>  


Tihomir>  

Tihomir> _______________________________________________
Tihomir>  FreeSWITCH-users mailing list
Tihomir>  FreeSWITCH-users at lists.freeswitch.org
Tihomir>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Tihomir> 
Tihomir> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Tihomir>  http://www.freeswitch.org
Tihomir>  


Tihomir>  

Tihomir> _______________________________________________
Tihomir>  FreeSWITCH-users mailing list
Tihomir>  FreeSWITCH-users at lists.freeswitch.org
Tihomir>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Tihomir> 
Tihomir> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Tihomir>  http://www.freeswitch.org
Tihomir>  


Tihomir>  

Tihomir> _______________________________________________
Tihomir>  FreeSWITCH-users mailing list
Tihomir>  FreeSWITCH-users at lists.freeswitch.org
Tihomir>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Tihomir> 
Tihomir> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Tihomir>  http://www.freeswitch.org
Tihomir>  


Tihomir>  

Tihomir> _______________________________________________
Tihomir>  FreeSWITCH-users mailing list
Tihomir>  FreeSWITCH-users at lists.freeswitch.org
Tihomir>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Tihomir> 
Tihomir> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Tihomir>  http://www.freeswitch.org
Tihomir>  








From bottleman at icf.org.ru  Wed Oct  7 05:50:47 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Wed, 7 Oct 2009 16:50:47 +0400 (MSD)
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>

On 2009-10-07 01:41 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

We are developing module to handle h323 proto now, we try to use mod_opal and try improve it, but no luck, 
there is many issues in libopal, and finaly we now move to h323plus library.

TC>Diego,
TC>
TC>what i'm pointing here is the situation where you have a great product that
TC>lacks in one of most common protocol. It is true H323 is going to disappear
TC>(eventually), it is true that the community prefers SIP/IAX instead ... but
TC>the reality still remains. H323 is going to be used for quite a long time to
TC>exchange a lot of traffic while FS will be left aside. Today, when you setup
TC>an IP peering interconnection 80% of carriers will prefer H323.
TC>
TC>Of course, developing something costs "time" (and we all know what time
TC>stands for...) and as i said, i understand the financial point of view and i
TC>really understand if nobody is going to work on that, but let's face it FS
TC>doesn't have any usable module to reliably handle H323 protocol.
TC>
TC>
TC>said that, i don't intend to offend anyone... just facing the reality.
TC>
TC>
TC>regarding the h323 module, we don't have any issue fixing the existing or
TC>developing a new one... but before we go developing something it is always
TC>better check if the thing you want already exists in an usable state or
TC>not... that's what i did today.
TC>
TC>
TC>So, I'm interested in a reliable module handling H323v4... anyone else?
TC>
TC>
TC>T.
TC>
TC>
TC>
TC>
TC>
TC>On Tue, Oct 6, 2009 at 11:45 PM, Diego Viola <diego.viola at gmail.com> wrote:
TC>
TC>> Instead of complaining and demanding things for free, people should start
TC>> to put their money where their mouth is.
TC>>
TC>> Diego
TC>>
TC>>
TC>> On Tue, Oct 6, 2009 at 8:47 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
TC>>
TC>>> hi Anthony,
TC>>>
TC>>> it is somewhere here:
TC>>>
TC>>>          switch_status_t
TC>>> FSConnection::receive_message(switch_core_session_message_t *msg)
TC>>>
TC>>>
TC>>> anyhow, i will open an issue jira of course.
TC>>>
TC>>>
TC>>> I understand your financial point of view, but anyhow while the entire
TC>>> world is wants sip and trying to move to sip, the reality is quite
TC>>> different. The majority of voice traffic exchanged via IP is still H323.
TC>>> This means a working SIP - H323 interworking is really needed... pity nobody
TC>>> wants/has time to work in this direction to produce a decent mod_h323.
TC>>>
TC>>>
TC>>>
TC>>> T.
TC>>>
TC>>>
TC>>>
TC>>>
TC>>>
TC>>>
TC>>> On Tue, Oct 6, 2009 at 9:25 PM, Anthony Minessale <
TC>>> anthony.minessale at gmail.com> wrote:
TC>>>
TC>>>> pcap is not as useful as FS console log on debug with:
TC>>>> sofia profile internal siptrace on
TC>>>>
TC>>>> you should be reporting issues to jira under mod_opal not to the mailing
TC>>>> list.
TC>>>> http://jira.freeswitch.org
TC>>>>
TC>>>> FYI
TC>>>> There is little financial support from the community for h323 which
TC>>>> prevents the mod_opal from getting much attention.
TC>>>> We actually have to contract the author of opal to help with these issues
TC>>>> including the original writing of the module that he did with very little
TC>>>> funding and nobody ever wants to pay him to improve it.
TC>>>>
TC>>>> That does not mean your issue will not be addressed but there is no
TC>>>> promise how fast it will be.
TC>>>>
TC>>>>
TC>>>>
TC>>>> On Tue, Oct 6, 2009 at 12:37 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
TC>>>>
TC>>>>> hello guys,
TC>>>>>
TC>>>>>
TC>>>>> i was playing with mod_opal to see if i can make it working ... well it
TC>>>>> seems SIP-H323 interworking is not tuned at all.
TC>>>>>
TC>>>>> I have a call from a registered sip user (1001) to PSTN via mod_opal
TC>>>>>
TC>>>>>
TC>>>>> <include>
TC>>>>>   <extension name="EMERGENCY">
TC>>>>>     <condition field="destination_number"
TC>>>>> expression="^0(112|9[23456])$">
TC>>>>>       <action application="set"
TC>>>>> data="effective_caller_id_number=1001282122"/>
TC>>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
TC>>>>>       <action application="set" data="call_timeout=30"/>
TC>>>>>       <action application="set" data="hangup_after_bridge=true"/>
TC>>>>>
TC>>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
TC>>>>>     </condition>
TC>>>>>   </extension>
TC>>>>>
TC>>>>>   <extension name="SPECIAL_SERVICES">
TC>>>>>     <condition field="destination_number"
TC>>>>> expression="^0(9[01789]\d{3,4})$">
TC>>>>>       <action application="set"
TC>>>>> data="effective_caller_id_number=1001282122"/>
TC>>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
TC>>>>>       <action application="set" data="call_timeout=30"/>
TC>>>>>       <action application="set" data="hangup_after_bridge=true"/>
TC>>>>>
TC>>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
TC>>>>>     </condition>
TC>>>>>   </extension>
TC>>>>>
TC>>>>>   <extension name="ENYTHING_ELSE">
TC>>>>>     <condition field="destination_number"
TC>>>>> expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
TC>>>>>       <action application="set"
TC>>>>> data="effective_caller_id_number=1001282122"/>
TC>>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
TC>>>>>       <action application="set" data="call_timeout=30"/>
TC>>>>>       <action application="set" data="hangup_after_bridge=true"/>
TC>>>>>
TC>>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
TC>>>>>     </condition>
TC>>>>>   </extension>
TC>>>>> </include>
TC>>>>>
TC>>>>>
TC>>>>>
TC>>>>> One of the many issues i sow is that FS connects the call on SIP leg
TC>>>>> before it actually receives H.225 connect from H323 leg... as it is
TC>>>>> configured to send 200 OK on the 1st H.225 message containing a FastStart
TC>>>>> element/OLC.
TC>>>>>
TC>>>>>
TC>>>>> Attached is the tcpdump i took on FS machine... just use this filter:
TC>>>>> "h225 or h245 or q931 or sip"
TC>>>>> Also, you can check the attac CDR,,,, this is an unanswered call i
TC>>>>> placed to PSTN and FS billed it 23 seconds.
TC>>>>>
TC>>>>>
TC>>>>>
TC>>>>> Can anyone tell where i can do correct SIP - H323 message mappings to
TC>>>>> avoid this?
TC>>>>>
TC>>>>>
TC>>>>>
TC>>>>> T.
TC>>>>>
TC>>>>>
TC>>>>>
TC>>>>>
TC>>>>> _______________________________________________
TC>>>>> FreeSWITCH-users mailing list
TC>>>>> FreeSWITCH-users at lists.freeswitch.org
TC>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>>>>> UNSUBSCRIBE:
TC>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>>>>> http://www.freeswitch.org
TC>>>>>
TC>>>>>
TC>>>>
TC>>>>
TC>>>> --
TC>>>> Anthony Minessale II
TC>>>>
TC>>>> FreeSWITCH http://www.freeswitch.org/
TC>>>> ClueCon http://www.cluecon.com/
TC>>>> Twitter: http://twitter.com/FreeSWITCH_wire
TC>>>>
TC>>>> AIM: anthm
TC>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
TC>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
TC>>>> IRC: irc.freenode.net #freeswitch
TC>>>>
TC>>>> FreeSWITCH Developer Conference
TC>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
TC>>>> iax:guest at conference.freeswitch.org/888
TC>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
TC>>>> pstn:213-799-1400
TC>>>>
TC>>>> _______________________________________________
TC>>>> FreeSWITCH-users mailing list
TC>>>> FreeSWITCH-users at lists.freeswitch.org
TC>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>>>> http://www.freeswitch.org
TC>>>>
TC>>>>
TC>>>
TC>>> _______________________________________________
TC>>> FreeSWITCH-users mailing list
TC>>> FreeSWITCH-users at lists.freeswitch.org
TC>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>>> http://www.freeswitch.org
TC>>>
TC>>>
TC>>
TC>> _______________________________________________
TC>> FreeSWITCH-users mailing list
TC>> FreeSWITCH-users at lists.freeswitch.org
TC>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> http://www.freeswitch.org
TC>>
TC>>
TC>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From mike at jerris.com  Wed Oct  7 06:06:26 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 7 Oct 2009 09:06:26 -0400
Subject: [Freeswitch-users] Bad sound quality while eavesdropping
In-Reply-To: <1254901192035-3780245.post@n2.nabble.com>
References: <41A44DD027064988A914974405788C2E@procent>
	<191c3a030910050731m2d74979ep4598e5a1945d58ae@mail.gmail.com>
	<1254901192035-3780245.post@n2.nabble.com>
Message-ID: <8437F5BC-7AFF-4A74-B8CD-C5B8219021F6@jerris.com>

What codecs are all the call legs using, also, please try current svn  
trunk.

Mike

On Oct 7, 2009, at 3:39 AM, Maciej Aniserowicz wrote:

>
> Sorry about posting several questions at once, I wasn't aware it's  
> "rude".
> Let's concentrate on this issue then.
>
> I use FS rev 14994. Phones on extensions:
> 1) x-lite
> 2) cisco sip phone
> 3) audio played by fs to the extension being eavesdropped
>
> I did not change any codec configuration, I just use the standard  
> one that
> comes with both FS and the phones.
> Some time ago someone on FS irc channel told me that this is just  
> how FS
> eavesdropping works... from your response I understand that this is  
> not
> entirely true?
>
> Maciej Aniserowicz
>
>
>
> Anthony Minessale wrote:
>>
>> That's is a somewhat vague position.
>>
>> You did not mention which version of FreeSWITCH you are running, the
>> phones
>> being used in your example, your configuration, the codecs in use  
>> etc.
>>
>> BTW,
>> I think you should only ask one question at a time on this list.   
>> The list
>> is run by volunteers and it's sort of rude to expect 3 or 4 threads  
>> to be
>> tended to concerning the same one individual.
>>
>>
>> 2009/10/5 Maciej Aniserowicz <maciej.aniserowicz at gmail.com>
>>
>>> Hello,
>>> When I use eavesdropping in FreeSWITCH, the sound quality is  
>>> really bad.
>>> Is
>>> there any way to improve it? Is this a known problem?
>>> Br/
>>> Maciej Aniserowicz
>>>




From andy at fabulous4.co.uk  Wed Oct  7 06:11:24 2009
From: andy at fabulous4.co.uk (Andy)
Date: Wed, 7 Oct 2009 14:11:24 +0100
Subject: [Freeswitch-users] NAT problems - sorry
Message-ID: <5BE289D2AE22492BA71A6919C8A440B3@D810>

Hi folks,
 
I know this problem comes up all the time so sorry to bring it up again but
I can't seem to find the answer in previous posts.
 
I have my freeswitch installation behind a DLink firewall so the freeswitch
server is natted. I have added what I believe are all the necessary rules to
the firewall to handle SIP correctly and the firewall isn't showing any
blocked traffic when calls come in. Calls are forwarded to my public ip
address from my VOIP provider Voiptalk. Here's the problem:
 
Incoming calls are successfully forwarded to the freeswitch server and
answered.
I can hear the IVR message played down the phone line so outgoing audio is
ok.
When I try to send DTMF key presses from my phone none of these register on
the switch.
When I hangup the call the call hangup is also not picked up by the switch.
 
I have followed the auto-nat instrcutions and have verified that the switch
correctly grabs and sets the external ip address. I have also tried setting
the external ip address manually but without success.
 
Can anyone offer any guidance as to what might be causing this problem or
what I could try to resolve it? I can post log/config snippets if that
helps.
 
Many thanks
Andy
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From claudiu at globtel.ro  Wed Oct  7 06:14:40 2009
From: claudiu at globtel.ro (Claudiu Filip)
Date: Wed, 7 Oct 2009 16:14:40 +0300
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <191c3a030910061758r2ddea5fand388f0af2c4dfc33@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<86a32abc0910061722t67ff23ddr62e4f62365539bd4@mail.gmail.com>
	<191c3a030910061758r2ddea5fand388f0af2c4dfc33@mail.gmail.com>
Message-ID: <716784809.20091007161440@globtel.ro>


Hi,

Wednesday, October 7, 2009, 3:58:20 AM, Anthony M. wrote:
> barely get anybody asking about h.323.


H323 may not be popular for small ITSPs or small/medium PBXes, but
it's widely used by the big players.. and freeswitch doesnt share the
same goals with asterisk.



Best wishes,


Claudiu Filip






From chris.chen2004 at gmail.com  Wed Oct  7 06:15:31 2009
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Wed, 7 Oct 2009 09:15:31 -0400
Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Followup,
	Agenda For This Week
In-Reply-To: <b16156850910062124h4bae532enc3624e18d1b20b2f@mail.gmail.com>
References: <87f2f3b90910051203v7bdbca5amfc82d193f3a4f1a3@mail.gmail.com>
	<B5ECAE34-B846-4408-8AC1-6D94ADCC672C@freeswitch.org>
	<b16156850910062124h4bae532enc3624e18d1b20b2f@mail.gmail.com>
Message-ID: <507898380910070615k9fb3c9ha2d040e31c0035ad@mail.gmail.com>

Hi Muhammad, the simple and reliable solution for you where SIP is being
blocked is add conf+888 at conference.freeswitch.org<conf%2B888 at conference.freeswitch.org>to
your Goolgetalk buddy list, and you can call from there to join the
conference, simple and straightforward.

Chris

On Wed, Oct 7, 2009 at 12:24 AM, Muhammad Shahzad <
shaheryarkh at googlemail.com> wrote:

> I got a lot of problem last week for making conference call. I was at home
> (conference call starts at 2200hours PKST, my time) and unable to make SIP
> call since the government has blocked it. So my only choice was Skype, but
> unfortunately DTMF wasn't working, i get connected on Skypiax5 for about 7-8
> times, but DTMF problem never let me get in to conference. Is there any
> solution for this?
>
> I really need to discuss a lot of things about FS documentation in
> conference, like what FS community can expect from it and what not? how i
> have planned it? how much progress has done? what are the problems me and my
> team are facing (which has slow us down considerably)? etc. etc.
>
> I have a solution for it but that needs testing. The plan is to use one of
> my FS servers to connect my jingle calls from GTalk to conference server
> over SIP. How can i test this setup with conference server, any ideas?
>
> If any one else also interested in getting connected to weekly conference
> call through this setup then i can also extend this setup as needed.
>
> Thank you.
>
>
>
> On Tue, Oct 6, 2009 at 1:30 AM, Brian West <brian at freeswitch.org> wrote:
>
>> It always supported 48kHz CELT but the conference itself was running
>> at 32kHz so everyone 48k had to be down sampled.  Now you all get to
>> be up sampled.  w00t!
>>
>> /b
>>
>> On Oct 5, 2009, at 2:03 PM, Michael Collins wrote:
>>
>> > * Starting with the upcoming meeting (Oct 9) the conference will
>> > support 48kHz CELT codec.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> ________________________________________________________
> |
>                      |
> | FATAL ERROR                                                           ---
> O X |
> |_______________________________________________________|
> |                        You have moved the mouse.
>    |
> | Windows must be restarted for the changes to take effect.   |
> |                                        <OK>
>                |
> ####################################/
>
>
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Wed Oct  7 06:21:51 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 7 Oct 2009 09:21:51 -0400
Subject: [Freeswitch-users] Recording creates a 388-byte long file and
	deletes it
In-Reply-To: <4ED3AB65AFE34242AACDE97127FE1248@procent>
References: <4ED3AB65AFE34242AACDE97127FE1248@procent>
Message-ID: <74D80E7B-119B-4CC1-82DC-9A370F940BA5@jerris.com>

switch_ivr_async.c:480

On Oct 5, 2009, at 3:16 AM, Maciej Aniserowicz wrote:

> Hi,
> When I record a call in FS, it only creates a 388-byte-long wav  
> file. The conversation is no written there, and FS deletes the file  
> when the session finishes.
> What can cause this strange behavior?

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From mike at jerris.com  Wed Oct  7 06:23:33 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 7 Oct 2009 09:23:33 -0400
Subject: [Freeswitch-users] gateway FS informs it's client FS about
	users hanged up with a long delay
In-Reply-To: <1A16501E57CA4727A593226A8C810308@procent>
References: <1A16501E57CA4727A593226A8C810308@procent>
Message-ID: <98F72CCB-88B9-47ED-AED3-B6BA6DE648C0@jerris.com>

Incorrect NAT configuration so one of the boxes is not actually  
getting a BYE.


On Oct 5, 2009, at 3:13 AM, Maciej Aniserowicz wrote:

> Hi,
> When I use two FreeSWITCH instances ('internal' and 'external'), all  
> users register to the 'external' instance which acts as a gateway by  
> 'internal' instance (which in turn is controlled by my applicaiton  
> with commands sent by socket).
> When user hangs up, the 'hanged up' event is propagated to the  
> 'internal' instance after a long time (~3 minutes) instead of being  
> propagated immediately.
> What can cause this issue?

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From mike at jerris.com  Wed Oct  7 06:26:31 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 7 Oct 2009 09:26:31 -0400
Subject: [Freeswitch-users] Bridge application with shared lines
In-Reply-To: <d361ca7b0910060114g3f23a102nb3508cf6e1e196fc@mail.gmail.com>
References: <d361ca7b0910060114g3f23a102nb3508cf6e1e196fc@mail.gmail.com>
Message-ID: <7F6D4A24-DCC7-4367-9EEE-73968A42DE98@jerris.com>


On Oct 6, 2009, at 4:14 AM, Yehavi Bourvine wrote:

> Hello,
>
>   We have Polycom and SNOM phones running with FreeSwitch. The  
> Polycoms have shared lines defined and the SNOMs have both shared  
> lines and BLFs (defined as extensions in the phone config). I've  
> tried supporting both, but have some incompatibility:
>
> When calling the Bridge application with data parameter of sofia/ 
> profile-name/number at domain the BLF works ok, but not the shared  
> lines (i.e only one of the phones  rings).
> When calling the Bridge application with data parameter of $ 
> {sofia_contact(/profile-name/number at domain)} shared lines work ok  
> but BLF doesn't fire up.
>
> How do I support both? Is there a way to know whether the  
> destination is a shared one and then chose one of the above formats?

You should probably always be using the second method or using a %  
instead of the @ in the first method to get the registered contact.   
can you provide more information about why the BLF "doesn't fire up" .

Mike

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From shaheryarkh at googlemail.com  Wed Oct  7 06:37:52 2009
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Wed, 7 Oct 2009 18:37:52 +0500
Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Followup,
	Agenda For This Week
In-Reply-To: <507898380910070615k9fb3c9ha2d040e31c0035ad@mail.gmail.com>
References: <87f2f3b90910051203v7bdbca5amfc82d193f3a4f1a3@mail.gmail.com>
	<B5ECAE34-B846-4408-8AC1-6D94ADCC672C@freeswitch.org>
	<b16156850910062124h4bae532enc3624e18d1b20b2f@mail.gmail.com>
	<507898380910070615k9fb3c9ha2d040e31c0035ad@mail.gmail.com>
Message-ID: <b16156850910070637w7c2b1f27s33ee87949f719d45@mail.gmail.com>

Great. Just added it. Is there any user limit on this?

Thank you.

On Wed, Oct 7, 2009 at 6:15 PM, Chris Chen <chris.chen2004 at gmail.com> wrote:

> Hi Muhammad, the simple and reliable solution for you where SIP is being
> blocked is add conf+888 at conference.freeswitch.org<conf%2B888 at conference.freeswitch.org>to your Goolgetalk buddy list, and you can call from there to join the
> conference, simple and straightforward.
>
> Chris
>
>
> On Wed, Oct 7, 2009 at 12:24 AM, Muhammad Shahzad <
> shaheryarkh at googlemail.com> wrote:
>
>> I got a lot of problem last week for making conference call. I was at home
>> (conference call starts at 2200hours PKST, my time) and unable to make SIP
>> call since the government has blocked it. So my only choice was Skype, but
>> unfortunately DTMF wasn't working, i get connected on Skypiax5 for about 7-8
>> times, but DTMF problem never let me get in to conference. Is there any
>> solution for this?
>>
>> I really need to discuss a lot of things about FS documentation in
>> conference, like what FS community can expect from it and what not? how i
>> have planned it? how much progress has done? what are the problems me and my
>> team are facing (which has slow us down considerably)? etc. etc.
>>
>> I have a solution for it but that needs testing. The plan is to use one of
>> my FS servers to connect my jingle calls from GTalk to conference server
>> over SIP. How can i test this setup with conference server, any ideas?
>>
>> If any one else also interested in getting connected to weekly conference
>> call through this setup then i can also extend this setup as needed.
>>
>> Thank you.
>>
>>
>>
>> On Tue, Oct 6, 2009 at 1:30 AM, Brian West <brian at freeswitch.org> wrote:
>>
>>> It always supported 48kHz CELT but the conference itself was running
>>> at 32kHz so everyone 48k had to be down sampled.  Now you all get to
>>> be up sampled.  w00t!
>>>
>>> /b
>>>
>>> On Oct 5, 2009, at 2:03 PM, Michael Collins wrote:
>>>
>>> > * Starting with the upcoming meeting (Oct 9) the conference will
>>> > support 48kHz CELT codec.
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> ________________________________________________________
>> |
>>                        |
>> | FATAL ERROR
>> --- O X |
>> |_______________________________________________________|
>> |                        You have moved the mouse.
>>    |
>> | Windows must be restarted for the changes to take effect.   |
>> |                                        <OK>
>>                  |
>> ####################################/
>>
>>
>> Muhammad Shahzad
>> -----------------------------------
>> CISCO Rich Media Communication Specialist (CRMCS)
>> CISCO Certified Network Associate (CCNA)
>> Cell: +92 334 422 40 88
>> MSN: shari_786pk at hotmail.com
>> Email: shaheryarkh at googlemail.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
________________________________________________________
|
                     |
| FATAL ERROR                                                           ---
O X |
|_______________________________________________________|
|                        You have moved the mouse.
 |
| Windows must be restarted for the changes to take effect.   |
|                                        <OK>
               |
####################################/


Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From nicolas at medularis.com  Wed Oct  7 06:48:59 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Wed, 7 Oct 2009 09:48:59 -0400
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <9106C8E1-498F-4AE8-8D34-7F6AAD4DC8AE@freeswitch.org>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<1b46b4e80910041519h3581d3d2x7b84041e8c18588@mail.gmail.com>
	<C887D143-F7A1-44B9-B8A2-D34A4B4E582E@jerris.com>
	<1b46b4e80910041609x2f41ceb9u142cb484951a21ed@mail.gmail.com>
	<87f2f3b90910050920w4d17033fl4c386a0018107bdf@mail.gmail.com>
	<1b46b4e80910052119me12a0d9y4653edd831f9555e@mail.gmail.com>
	<1b46b4e80910052142jaa09f08l72e5e4b2b24d10fc@mail.gmail.com>
	<083DBCB9-48B2-4E61-81B5-B7D00A42EAB5@freeswitch.org>
	<1b46b4e80910061422q7be2a21fg5d2caaef7e307abb@mail.gmail.com>
	<9106C8E1-498F-4AE8-8D34-7F6AAD4DC8AE@freeswitch.org>
Message-ID: <1b46b4e80910070648n622c3f9haacc22c2328b44ff@mail.gmail.com>

Is there some way to make FS register with the gateway that is rejecting the
authentication? is it FS or the SIP server at fault? Why would X-Lite work
and FS not?

Thanks again for your time and help.


On Tue, Oct 6, 2009 at 5:46 PM, Brian West <brian at freeswitch.org> wrote:

> btw My mistake it doesn't assume auth it just calculates the response
> hash differently on this case where qop isn't present.
>
> /b
>
> On Oct 6, 2009, at 4:22 PM, Nicolas Brenner wrote:
>
> >
> > What does the qop parameter stand for? Apparently because of that
> > parameter, FS sends a new REGISTER including this:
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Wed Oct  7 07:09:32 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 7 Oct 2009 09:09:32 -0500
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <716784809.20091007161440@globtel.ro>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<86a32abc0910061722t67ff23ddr62e4f62365539bd4@mail.gmail.com>
	<191c3a030910061758r2ddea5fand388f0af2c4dfc33@mail.gmail.com>
	<716784809.20091007161440@globtel.ro>
Message-ID: <191c3a030910070709r2f63cf1dm9f41fff66ab76d3@mail.gmail.com>

I am not commenting on how popular it is overall.
I am commenting on the specific demand presented to us.
I don't know what else to say to explain that I am completely neutral when
it comes to this topic.

One more time:

1) We are an open source project who volunteer most of our time as well as
paid commercial support.
2) We made mod_opal anyway despite only a small handful of requests.
3) The devs from OPAL are consultants for hire and like us, they will look
at what they can for free but you can hire them to heighten their attention
span.

This is not a discussion on the merits of h323, I am more than happy to
support it.  The OPAL guys are some of the best open source telephony guys
out there and they have been doing this for over a decade.  We Just can't
promise a whole lot of man hours towards it when we already work day and
night managing jira with only a few managers.  We also cannot make any
promises on the free time the OPAL devs have to devote to it either.




On Wed, Oct 7, 2009 at 8:14 AM, Claudiu Filip <claudiu at globtel.ro> wrote:

>
> Hi,
>
> Wednesday, October 7, 2009, 3:58:20 AM, Anthony M. wrote:
> > barely get anybody asking about h.323.
>
>
> H323 may not be popular for small ITSPs or small/medium PBXes, but
> it's widely used by the big players.. and freeswitch doesnt share the
> same goals with asterisk.
>
>
>
> Best wishes,
>
>
> Claudiu Filip
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Wed Oct  7 07:10:51 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 7 Oct 2009 09:10:51 -0500
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910070137k52be89e3g846e55db72a8de5d@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<86a32abc0910061722t67ff23ddr62e4f62365539bd4@mail.gmail.com>
	<191c3a030910061758r2ddea5fand388f0af2c4dfc33@mail.gmail.com>
	<65d96fc80910070137k52be89e3g846e55db72a8de5d@mail.gmail.com>
Message-ID: <191c3a030910070710s6722c2b5j4a7c6fd2b1bcc648@mail.gmail.com>

I think the way to determine the funding is to get all the most important
issues up on jira, try to deal with them and see if we need to put bounties
on any of them to get them done faster.


On Wed, Oct 7, 2009 at 3:37 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> Anthony,
>
> of course, nobody wants to start anything... we are all here to help making
> FS a better product.
>
> so, regarding the founding for mod_opal ... what is the amount you need?
>
>
> Tihomir.
>
>
>
>
> On Wed, Oct 7, 2009 at 2:58 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> I didn't mean to start anything.  I'm just saying we work very long hours
>> and barely get anybody asking about h.323.
>> I wanted to support it and that's why we took up a collection to get
>> funding for mod_opal but when only 1 donor showed any interest we were
>> forced to proceed in our spare time which is very limited.
>>
>> The developers of opal are not part of our project and they need financial
>> compensation to be motivated to work on it.  Its not even related to me its
>> only fair that an outside developer who makes his living as a consultant
>> would want money to integrate his work into our project.
>>
>> Like I said, I will do my best to point your issue to the opal devs but I
>> cannot force them to work on it.
>>
>>
>>
>>
>>
>> On Tue, Oct 6, 2009 at 7:22 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>
>>> Yeah I understand your point of view, but saying "I want a H.323 module"
>>> or "I want a Ferrari" wont magically make it happen.
>>>
>>> We need to work on it ourselves or pay to the people that knows how to do
>>> it to do it for us.
>>>
>>> There is no other way I think.
>>>
>>> Diego
>>>
>>>
>>>
>>>
>>> On Tue, Oct 6, 2009 at 11:41 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>>
>>>> Diego,
>>>>
>>>> what i'm pointing here is the situation where you have a great product
>>>> that lacks in one of most common protocol. It is true H323 is going to
>>>> disappear (eventually), it is true that the community prefers SIP/IAX
>>>> instead ... but the reality still remains. H323 is going to be used for
>>>> quite a long time to exchange a lot of traffic while FS will be left aside.
>>>> Today, when you setup an IP peering interconnection 80% of carriers will
>>>> prefer H323.
>>>>
>>>> Of course, developing something costs "time" (and we all know what time
>>>> stands for...) and as i said, i understand the financial point of view and i
>>>> really understand if nobody is going to work on that, but let's face it FS
>>>> doesn't have any usable module to reliably handle H323 protocol.
>>>>
>>>>
>>>> said that, i don't intend to offend anyone... just facing the reality.
>>>>
>>>>
>>>> regarding the h323 module, we don't have any issue fixing the existing
>>>> or developing a new one... but before we go developing something it is
>>>> always better check if the thing you want already exists in an usable state
>>>> or not... that's what i did today.
>>>>
>>>>
>>>> So, I'm interested in a reliable module handling H323v4... anyone else?
>>>>
>>>>
>>>> T.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Tue, Oct 6, 2009 at 11:45 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>>
>>>>> Instead of complaining and demanding things for free, people should
>>>>> start to put their money where their mouth is.
>>>>>
>>>>> Diego
>>>>>
>>>>>
>>>>> On Tue, Oct 6, 2009 at 8:47 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>>>>
>>>>>> hi Anthony,
>>>>>>
>>>>>> it is somewhere here:
>>>>>>
>>>>>>          switch_status_t
>>>>>> FSConnection::receive_message(switch_core_session_message_t *msg)
>>>>>>
>>>>>>
>>>>>> anyhow, i will open an issue jira of course.
>>>>>>
>>>>>>
>>>>>> I understand your financial point of view, but anyhow while the entire
>>>>>> world is wants sip and trying to move to sip, the reality is quite
>>>>>> different. The majority of voice traffic exchanged via IP is still H323.
>>>>>> This means a working SIP - H323 interworking is really needed... pity nobody
>>>>>> wants/has time to work in this direction to produce a decent mod_h323.
>>>>>>
>>>>>>
>>>>>>
>>>>>> T.
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Tue, Oct 6, 2009 at 9:25 PM, Anthony Minessale <
>>>>>> anthony.minessale at gmail.com> wrote:
>>>>>>
>>>>>>> pcap is not as useful as FS console log on debug with:
>>>>>>> sofia profile internal siptrace on
>>>>>>>
>>>>>>> you should be reporting issues to jira under mod_opal not to the
>>>>>>> mailing list.
>>>>>>> http://jira.freeswitch.org
>>>>>>>
>>>>>>> FYI
>>>>>>> There is little financial support from the community for h323 which
>>>>>>> prevents the mod_opal from getting much attention.
>>>>>>> We actually have to contract the author of opal to help with these
>>>>>>> issues including the original writing of the module that he did with very
>>>>>>> little funding and nobody ever wants to pay him to improve it.
>>>>>>>
>>>>>>> That does not mean your issue will not be addressed but there is no
>>>>>>> promise how fast it will be.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Tue, Oct 6, 2009 at 12:37 PM, Tihomir Culjaga <tculjaga at gmail.com
>>>>>>> > wrote:
>>>>>>>
>>>>>>>> hello guys,
>>>>>>>>
>>>>>>>>
>>>>>>>> i was playing with mod_opal to see if i can make it working ... well
>>>>>>>> it seems SIP-H323 interworking is not tuned at all.
>>>>>>>>
>>>>>>>> I have a call from a registered sip user (1001) to PSTN via mod_opal
>>>>>>>>
>>>>>>>>
>>>>>>>> <include>
>>>>>>>>   <extension name="EMERGENCY">
>>>>>>>>     <condition field="destination_number"
>>>>>>>> expression="^0(112|9[23456])$">
>>>>>>>>       <action application="set"
>>>>>>>> data="effective_caller_id_number=1001282122"/>
>>>>>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>>>>>>       <action application="set" data="call_timeout=30"/>
>>>>>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>>>>>
>>>>>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>>>>>>     </condition>
>>>>>>>>   </extension>
>>>>>>>>
>>>>>>>>   <extension name="SPECIAL_SERVICES">
>>>>>>>>     <condition field="destination_number"
>>>>>>>> expression="^0(9[01789]\d{3,4})$">
>>>>>>>>       <action application="set"
>>>>>>>> data="effective_caller_id_number=1001282122"/>
>>>>>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>>>>>>       <action application="set" data="call_timeout=30"/>
>>>>>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>>>>>
>>>>>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>>>>>>     </condition>
>>>>>>>>   </extension>
>>>>>>>>
>>>>>>>>   <extension name="ENYTHING_ELSE">
>>>>>>>>     <condition field="destination_number"
>>>>>>>> expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
>>>>>>>>       <action application="set"
>>>>>>>> data="effective_caller_id_number=1001282122"/>
>>>>>>>>       <action application="set" data="NCX_IP=10.4.4.254"/>
>>>>>>>>       <action application="set" data="call_timeout=30"/>
>>>>>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>>>>>
>>>>>>>>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
>>>>>>>>     </condition>
>>>>>>>>   </extension>
>>>>>>>> </include>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> One of the many issues i sow is that FS connects the call on SIP leg
>>>>>>>> before it actually receives H.225 connect from H323 leg... as it is
>>>>>>>> configured to send 200 OK on the 1st H.225 message containing a FastStart
>>>>>>>> element/OLC.
>>>>>>>>
>>>>>>>>
>>>>>>>> Attached is the tcpdump i took on FS machine... just use this
>>>>>>>> filter: "h225 or h245 or q931 or sip"
>>>>>>>> Also, you can check the attac CDR,,,, this is an unanswered call i
>>>>>>>> placed to PSTN and FS billed it 23 seconds.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Can anyone tell where i can do correct SIP - H323 message mappings
>>>>>>>> to avoid this?
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> T.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Anthony Minessale II
>>>>>>>
>>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>>> ClueCon http://www.cluecon.com/
>>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>>
>>>>>>> AIM: anthm
>>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>>
>>>>>>> FreeSWITCH Developer Conference
>>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>>> pstn:213-799-1400
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From brian at freeswitch.org  Wed Oct  7 07:20:20 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 7 Oct 2009 09:20:20 -0500
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <1b46b4e80910070648n622c3f9haacc22c2328b44ff@mail.gmail.com>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<1b46b4e80910041519h3581d3d2x7b84041e8c18588@mail.gmail.com>
	<C887D143-F7A1-44B9-B8A2-D34A4B4E582E@jerris.com>
	<1b46b4e80910041609x2f41ceb9u142cb484951a21ed@mail.gmail.com>
	<87f2f3b90910050920w4d17033fl4c386a0018107bdf@mail.gmail.com>
	<1b46b4e80910052119me12a0d9y4653edd831f9555e@mail.gmail.com>
	<1b46b4e80910052142jaa09f08l72e5e4b2b24d10fc@mail.gmail.com>
	<083DBCB9-48B2-4E61-81B5-B7D00A42EAB5@freeswitch.org>
	<1b46b4e80910061422q7be2a21fg5d2caaef7e307abb@mail.gmail.com>
	<9106C8E1-498F-4AE8-8D34-7F6AAD4DC8AE@freeswitch.org>
	<1b46b4e80910070648n622c3f9haacc22c2328b44ff@mail.gmail.com>
Message-ID: <8750A70D-6891-4C07-B4DF-C7F4ABA58D67@freeswitch.org>

I would suspect its a PEBKAC.  I mean if you could register to a  
gateway that rejected auth... what purpose would auth serve in the  
first place?

/b

On Oct 7, 2009, at 8:48 AM, Nicolas Brenner wrote:

> Is there some way to make FS register with the gateway that is  
> rejecting the authentication? is it FS or the SIP server at fault?  
> Why would X-Lite work and FS not?
>
> Thanks again for your time and help.




From brian at freeswitch.org  Wed Oct  7 07:20:42 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 7 Oct 2009 09:20:42 -0500
Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Followup,
	Agenda For This Week
In-Reply-To: <b16156850910070637w7c2b1f27s33ee87949f719d45@mail.gmail.com>
References: <87f2f3b90910051203v7bdbca5amfc82d193f3a4f1a3@mail.gmail.com>
	<B5ECAE34-B846-4408-8AC1-6D94ADCC672C@freeswitch.org>
	<b16156850910062124h4bae532enc3624e18d1b20b2f@mail.gmail.com>
	<507898380910070615k9fb3c9ha2d040e31c0035ad@mail.gmail.com>
	<b16156850910070637w7c2b1f27s33ee87949f719d45@mail.gmail.com>
Message-ID: <9FF9A0D6-0D40-4C62-9244-6607172BAEA7@freeswitch.org>

No!

/b

On Oct 7, 2009, at 8:37 AM, Muhammad Shahzad wrote:

> Great. Just added it. Is there any user limit on this?
>
> Thank you.




From brian at freeswitch.org  Wed Oct  7 07:21:43 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 7 Oct 2009 09:21:43 -0500
Subject: [Freeswitch-users] NAT problems - sorry
In-Reply-To: <5BE289D2AE22492BA71A6919C8A440B3@D810>
References: <5BE289D2AE22492BA71A6919C8A440B3@D810>
Message-ID: <5A70A0D4-70FA-43B2-BAF5-F8D6ECC64838@freeswitch.org>

Bet you its inband dtmf and you need to start the dtmf detector.

/b

On Oct 7, 2009, at 8:11 AM, Andy wrote:

> I can hear the IVR message played down the phone line so outgoing  
> audio is ok.

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From brian at freeswitch.org  Wed Oct  7 07:24:00 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 7 Oct 2009 09:24:00 -0500
Subject: [Freeswitch-users] Bridge application with shared lines
In-Reply-To: <7F6D4A24-DCC7-4367-9EEE-73968A42DE98@jerris.com>
References: <d361ca7b0910060114g3f23a102nb3508cf6e1e196fc@mail.gmail.com>
	<7F6D4A24-DCC7-4367-9EEE-73968A42DE98@jerris.com>
Message-ID: <09EC098B-3883-4FF3-B7F2-08360E39D374@freeswitch.org>

You'll need to set presence_id so it can work properly. SEE the  
default config that does exactly that with sofia_contact in the dial- 
string on the domain.

/b

On Oct 7, 2009, at 8:26 AM, Michael Jerris wrote:

>> When calling the Bridge application with data parameter of sofia/ 
>> profile-name/number at domain the BLF works ok, but not the shared  
>> lines (i.e only one of the phones  rings).
>> When calling the Bridge application with data parameter of $ 
>> {sofia_contact(/profile-name/number at domain)} shared lines work ok  
>> but BLF doesn't fire up.

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From nicolas at medularis.com  Wed Oct  7 07:43:44 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Wed, 7 Oct 2009 10:43:44 -0400
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <8750A70D-6891-4C07-B4DF-C7F4ABA58D67@freeswitch.org>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<1b46b4e80910041609x2f41ceb9u142cb484951a21ed@mail.gmail.com>
	<87f2f3b90910050920w4d17033fl4c386a0018107bdf@mail.gmail.com>
	<1b46b4e80910052119me12a0d9y4653edd831f9555e@mail.gmail.com>
	<1b46b4e80910052142jaa09f08l72e5e4b2b24d10fc@mail.gmail.com>
	<083DBCB9-48B2-4E61-81B5-B7D00A42EAB5@freeswitch.org>
	<1b46b4e80910061422q7be2a21fg5d2caaef7e307abb@mail.gmail.com>
	<9106C8E1-498F-4AE8-8D34-7F6AAD4DC8AE@freeswitch.org>
	<1b46b4e80910070648n622c3f9haacc22c2328b44ff@mail.gmail.com>
	<8750A70D-6891-4C07-B4DF-C7F4ABA58D67@freeswitch.org>
Message-ID: <1b46b4e80910070743h38d39e5ax362ff7944754067e@mail.gmail.com>

You are missing the point, it is only rejecting auth for FS, Asterisk and
X-Lite work fine with the same config for that gateway.


On Wed, Oct 7, 2009 at 10:20 AM, Brian West <brian at freeswitch.org> wrote:

> I would suspect its a PEBKAC.  I mean if you could register to a
> gateway that rejected auth... what purpose would auth serve in the
> first place?
>
> /b
>
> On Oct 7, 2009, at 8:48 AM, Nicolas Brenner wrote:
>
> > Is there some way to make FS register with the gateway that is
> > rejecting the authentication? is it FS or the SIP server at fault?
> > Why would X-Lite work and FS not?
> >
> > Thanks again for your time and help.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From srinivas.ksvreddy at gmail.com  Wed Oct  7 07:48:14 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Wed, 7 Oct 2009 20:18:14 +0530
Subject: [Freeswitch-users] mod_sofia.c registered calls how to know
Message-ID: <f8af5740910070748g1efee71ft5841b6fa4d2412c8@mail.gmail.com>

Hi

can any please tell me where registered calls are stored, so when incoming
call came to mod_sofia.c how it will check it is registered or not?\\

-- 
Srinivasula Reddy K
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From vhatz at kinetix.gr  Wed Oct  7 06:37:58 2009
From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI))
Date: Wed, 07 Oct 2009 16:37:58 +0300
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <716784809.20091007161440@globtel.ro>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>	<86a32abc0910061722t67ff23ddr62e4f62365539bd4@mail.gmail.com>	<191c3a030910061758r2ddea5fand388f0af2c4dfc33@mail.gmail.com>
	<716784809.20091007161440@globtel.ro>
Message-ID: <4ACC99B6.4070706@kinetix.gr>


Claudiu Filip wrote:
> Hi,
> 
> Wednesday, October 7, 2009, 3:58:20 AM, Anthony M. wrote:
>> barely get anybody asking about h.323.
> 
> 
> H323 may not be popular for small ITSPs or small/medium PBXes, but
> it's widely used by the big players.. and freeswitch doesnt share the
> same goals with asterisk.
> 
> 
> 
> Best wishes,
> 
> 
> Claudiu Filip
> 

I have to agree with Filip, H323 is not popular (or sometimes even 
known) to the VoIP end user and small PBX community. In fact many VoIP 
users that I've talked with even believe that VoIP started with SIP, 
which is not true.

H323 although diminishing in usage is still used among voice carriers 
because it better resembles telephony environments and is not prone to 
errors and problems of SIP (example: if there is a network error in 
mid-call H323 uses TCP for signaling and therefore almost immediately 
detects the error to hang-up the call, while SIP _typically_ uses UDP 
and cannot detect such issues via signaling but only if a media gateway 
monitors RTP activity etc.).

As it is already pointed out, many carriers switch to SIP steadily, but 
still H323 is important for carrier-to carrier interconnection.

FreeSWITCH is positioned/advertised as a high-capacity softswitch (not 
just a SME PBX) which implies interconnection with large carriers (who 
still prefer H323).

There are many SIP platforms out there, open source and closed source 
that can do SIP (most of them pretty good, some of them excellent).

There are many closed source platforms that can do SIP and H323 pretty 
good (and some of them excellent).

However, there is no open source platform that can do BOTH SIP and H323 
REALLY good. That could be a competitive advantage for an open source 
platform...

NOTE: All the above is just academic discussion. A reply to the above in 
the spirit "put your money where your mouth is" is not really 
constructive and will be countered by a series of valid arguments 
against it.

Best regards,
Vlasis Hatzistavrou.



From msc at freeswitch.org  Wed Oct  7 08:49:44 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 7 Oct 2009 08:49:44 -0700
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <7d79b3930910070334g53cef0c5m63037e264f1a9233@mail.gmail.com>
References: <7d79b3930910070334g53cef0c5m63037e264f1a9233@mail.gmail.com>
Message-ID: <87f2f3b90910070849t79b7cb90x4150bf1dd8501ca4@mail.gmail.com>

On Wed, Oct 7, 2009 at 3:34 AM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Hi,
> Again I was struck in a problem, Here is the scenario.
>
> On incomming call, I just call an event outboud socket. But what happens
> is, for the first 15 call, it is working fine. But from the 16th call to
> 30th call, it says the below error.
>
> 2009-10-07 15:07:48.201846 [WARNING] ozmod_libpri.c:761 --Failure opening
> channel 1:17 (ignored)
> 2009-10-07 15:07:55.381861 [WARNING] ozmod_libpri.c:761 --Failure opening
> channel 1:18 (ignored)
> 2009-10-07 15:07:58.569774 [WARNING] ozmod_libpri.c:761 --Failure opening
> channel 1:19 (ignored)
> 2009-10-07 15:08:01.37824 [WARNING] ozmod_libpri.c:761 --Failure opening
> channel 1:20 (ignored)
> 2009-10-07 15:08:03.129846 [WARNING] ozmod_libpri.c:761 --Failure opening
> channel 1:21 (ignored)
> 2009-10-07 15:08:04.825851 [WARNING] ozmod_libpri.c:761 --Failure opening
> channel 1:22 (ignored)
> 2009-10-07 15:08:06.289977 [WARNING] ozmod_libpri.c:761 --Failure opening
> channel 1:23 (ignored)
> 2009-10-07 15:08:07.761961 [WARNING] ozmod_libpri.c:761 --Failure opening
> channel 1:24 (ignored)
> 2009-10-07 15:08:09.737944 [WARNING] ozmod_libpri.c:761 --Failure opening
> channel 1:25 (ignored)
> 2009-10-07 15:08:11.462018 [WARNING] ozmod_libpri.c:761 --Failure opening
> channel 1:26 (ignored)
> 2009-10-07 15:08:13.566024 [WARNING] ozmod_libpri.c:761 --Failure opening
> channel 1:27 (ignored)
> 2009-10-07 15:08:15.430163 [WARNING] ozmod_libpri.c:761 --Failure opening
> channel 1:28 (ignored)
> 2009-10-07 15:08:17.446103 [WARNING] ozmod_libpri.c:761 --Failure opening
> channel 1:29 (ignored)
> 2009-10-07 15:08:19.430118 [WARNING] ozmod_libpri.c:761 --Failure opening
> channel 1:30 (ignored)
> 2009-10-07 15:08:21.358121 [WARNING] ozmod_libpri.c:761 --Failure opening
> channel 1:31 (ignored)
>
>
> But for the next call, when it is opening channel 1:1, it is executing my
> dial plans.
> I don't know why it failed when it is choosing 1:17-1:31. Any one has any
> idea.
>
> Below are my configuration details:
>
> openzap.conf
> [span zt PRI_1]
> trunk_type => e1
> b-channel => 1:1-15
> d-channel=> 1:16
> b-channel => 1:17-31
>

Just a hunch, but could you try this in openzap.conf:
b-channel => 1:1-15,17-31
d-channel => 1:16

I want to see what happens. I don't have an E1 setup to test with right now
otherwise I'd do it myself. Please report back. Thanks,
MC


>
> openzap.conf.xml
> <configuration name="openzap.conf" description="OpenZAP Configuration">
>         <settings>
>                 <param name="debug" value="7"/>
>         </settings>
>         <libpri_spans>
>                 <span name="PRI_1">
>                         <param name="node" value="cpe"/>
>                         <param name="switch" value="euroisdn"/>
>                         <param name="dialplan" value="XML"/>
>                         <param name="context" value="default"/>
>                 </span>
>         </libpri_spans>
> </configuration>
>
> zaptel.conf
>
> span=1,1,0,ccs,hdb3
> bchan=1-15,17-31
> dchan=16
>
> loadzone        = us
> defaultzone     = us
>
> oz list
>
> span: 1 (PRI_1)
> type: isdn
> chan_count: 47
> dialplan: XML
> context: default
> dial_regex:
> fail_dial_regex:
> hold_music:
> analog_options none
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Wed Oct  7 08:50:12 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 7 Oct 2009 17:50:12 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <14480284.20091007154057@globtel.ro>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<86a32abc0910061722t67ff23ddr62e4f62365539bd4@mail.gmail.com>
	<191c3a030910061758r2ddea5fand388f0af2c4dfc33@mail.gmail.com>
	<65d96fc80910070137k52be89e3g846e55db72a8de5d@mail.gmail.com>
	<14480284.20091007154057@globtel.ro>
Message-ID: <65d96fc80910070850h4c3735b7s7a7577b25b71e1f6@mail.gmail.com>

On Wed, Oct 7, 2009 at 2:40 PM, Claudiu Filip <claudiu at globtel.ro> wrote:

>
>
>
> Hi Tihomir,
>
>
>   I've  done  some  tests  to  see  how  suitable  is freeswitch as a
>   SIP/H323  translator  and  you  are  right about the fact that H323
>   'alert+open  logical  channel'  will generate a SIP '200 OK'. I was
>   able  to fix that with a couple of changes in mod_opal.cpp, however
>   some  things  were changed on mod_sofia in the latest svn. (on this
>   particular  issue,  open_logical_channel  is  processed  BEFORE the
>   alerting,   so   the   call   is   in   SetupPhase  when  the  proc
>   OnOpenMediaStream is triggered)
>
>
>


yep, thats correct ... i was just wondering why it hangs in SetUpPhase


2009-10-07 16:50:11.690451 [DEBUG] manager.cxx:718 OnOpenMediaStream
Call[n03f409711]-EP<h323>[localhost/3263],OpalRTPMediaStream-Source-G.711-ALaw-64k
2009-10-07 16:50:11.690451 [INFO] mod_opal.cpp:1283 opal/
h323:05492122 at 10.4.4.254 <h323%3A05492122 at 10.4.4.254> initialise
opal/h323:05492122 at 10.4.4.254read audio codec G.711-ALaw-64k for connection
FSMediaStream-Sink-G.711-ALaw-64k
2009-10-07 16:50:11.690451 [DEBUG] mod_opal.cpp:1313 Set read audio codec to
<< G.711-ALaw-64k for connection FSMediaStream-Sink-G.711-ALaw-64k
2009-10-07 16:50:11.691525 [DEBUG] manager.cxx:718 OnOpenMediaStream
Call[n03f409711]-EP<local>[1],FSMediaStream-Sink-G.711-ALaw-64k
*2009-10-07 16:50:11.691525 [CONSOLE] mod_opal.cpp:852 SetUpPhase =>
GetPhase() = '1'*
2009-10-07 16:50:11.691525 [DEBUG] connection.cxx:561 Opened sink stream
n03f409711_1 with format G.711-ALaw-64k
2009-10-07 16:50:11.691525 [DEBUG] patch.cxx:341 Created Sink:
format=G.711-ALaw-64k
2009-10-07 16:50:11.691525 [DEBUG] mediastrm.cxx:666 RTP data size cannot be
changed to 160, fixed at 2048
2009-10-07 16:50:11.691525 [DEBUG] patch.cxx:179 Added direct media stream
sink FSMediaStream-Sink-G.711-ALaw-64k



this is the original code, and it never triggers eraly_media as never
reaches AlertingPhase.

if (GetMediaStream(stream.GetSessionID(), stream.IsSink()) != NULL) {
        // Have open media in both directions.
        if (GetPhase() == AlertingPhase) {
            switch_channel_mark_pre_answered(m_fsChannel);
            switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CONSOLE, "LOG
==>\t Alerting => GetPhase() = '%d'\n",GetPhase());


        } else if (GetPhase() < ReleasingPhase) {
            switch_channel_mark_answered(m_fsChannel);
            switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CONSOLE, "LOG
==>\t GetPhase() = '%d'\n",GetPhase());
        }
    }




I tried this, it works for early media but i still need to open a full media
path and say the call actually connected ....


   if (GetMediaStream(stream.GetSessionID(), stream.IsSink()) != NULL) {

        // Have open media in both directions.
        if (GetPhase() < ConnectedPhase) {
            switch_channel_mark_pre_answered(m_fsChannel);
            switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CONSOLE, "EARLY
MEDIA => GetPhase() = '%d'\n",GetPhase());

        } else if (GetPhase() < ReleasingPhase) {
            switch_channel_mark_answered(m_fsChannel);
            switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CONSOLE, "FULL
MEDIA => GetPhase() = '%d'\n",GetPhase());

        }
    }






this is when i'm dong early_media:


2009-10-07 17:45:26.788082 [DEBUG] manager.cxx:718 OnOpenMediaStream
Call[c8dce50981]-EP<h323>[localhost/26906],OpalRTPMediaStream-Source-G.711-ALaw-64k
2009-10-07 17:45:26.789158 [INFO] mod_opal.cpp:1279 opal/
h323:05492122 at 10.4.4.254 <h323%3A05492122 at 10.4.4.254> initialise
opal/h323:05492122 at 10.4.4.254read audio codec G.711-ALaw-64k for connection
FSMediaStream-Sink-G.711-ALaw-64k
2009-10-07 17:45:26.789158 [DEBUG] mod_opal.cpp:1309 Set read audio codec to
<< G.711-ALaw-64k for connection FSMediaStream-Sink-G.711-ALaw-64k
2009-10-07 17:45:26.789158 [DEBUG] manager.cxx:718 OnOpenMediaStream
Call[c8dce50981]-EP<local>[1],FSMediaStream-Sink-G.711-ALaw-64k
2009-10-07 17:45:26.789158 [NOTICE] mod_opal.cpp:887 Pre-Answer opal/
h323:05492122 at 10.4.4.254 <h323%3A05492122 at 10.4.4.254>!
2009-10-07 17:45:26.789158 [DEBUG] switch_channel.c:1822 Send signal
sofia/internal/1001 at 10.4.62.7 [BREAK]
*2009-10-07 17:45:26.789158 [CONSOLE] mod_opal.cpp:888 EARLY MEDIA =>
GetPhase() = '1'*
2009-10-07 17:45:26.789158 [DEBUG] connection.cxx:561 Opened sink stream
c8dce50981_1 with format G.711-ALaw-64k
2009-10-07 17:45:26.789158 [DEBUG] patch.cxx:341 Created Sink:
format=G.711-ALaw-64k
2009-10-07 17:45:26.790236 [DEBUG] switch_ivr_originate.c:2154
sofia/internal/1001 at 10.4.62.7 receive message [PROGRESS]
2009-10-07 17:45:26.790236 [INFO] switch_ivr_originate.c:2154 Sending early
media
2009-10-07 17:45:26.790236 [DEBUG] sofia_glue.c:2329 AUDIO RTP
[sofia/internal/1001 at 10.4.62.7] 10.4.62.7 port 19594 -> 10.4.62.89 port 5004
codec: 8 ms: 20
2009-10-07 17:45:26.790236 [DEBUG] switch_rtp.c:1155 Starting timer [soft]
160 bytes per 20ms
2009-10-07 17:45:26.790236 [DEBUG] mediastrm.cxx:666 RTP data size cannot be
changed to 160, fixed at 2048
2009-10-07 17:45:26.790236 [DEBUG] patch.cxx:179 Added direct media stream
sink FSMediaStream-Sink-G.711-ALaw-64k
2009-10-07 17:45:26.790236 [DEBUG] connection.cxx:728 Sink stream of
connection Call[c8dce50981]-EP<local>[1] uses patch Patch
OpalRTPMediaStream-Source-G.711-ALaw-64k ->
FSMediaStream-Sink-G.711-ALaw-64k
2009-10-07 17:45:26.791311 [DEBUG] connection.cxx:728 Source stream of
connection Call[c8dce50981]-EP<h323>[localhost/26906] uses patch Patch
OpalRTPMediaStream-Source-G.711-ALaw-64k ->
FSMediaStream-Sink-G.711-ALaw-64k
2009-10-07 17:45:26.791311 [DEBUG] rtpconn.cxx:249 Adding RFC2833 receive
handler
2009-10-07 17:45:26.791311 [DEBUG] rtpconn.cxx:254 Adding Cisco NSE receive
handler
2009-10-07 17:45:26.791311 [DEBUG] h323ep.cxx:1118 Started receiving logical
channel: G.711-ALaw-64k <2>
2009-10-07 17:45:26.791311 [DEBUG] rtpconn.cxx:496 Releasing session 1
2009-10-07 17:45:26.792387 [DEBUG] rtpconn.cxx:496 Releasing session 1
2009-10-07 17:45:26.792387 [DEBUG] h323.cxx:2450 Fast starting 2 channels
2009-10-07 17:45:26.792387 [DEBUG] h323ep.cxx:1024 Received alerting PDU.
2009-10-07 17:45:26.792387 [DEBUG] manager.cxx:636 OnAlerting
Call[c8dce50981]-EP<h323>[localhost/26906]
2009-10-07 17:45:26.792387 [DEBUG] call.cxx:184 OnAlerting
Call[c8dce50981]-EP<h323>[localhost/26906]
2009-10-07 17:45:26.792387 [DEBUG] localep.cxx:214 SetAlerting(10.4.4.254)
2009-10-07 17:45:26.792387 [INFO] mod_sofia.c:1518 Ring SDP:
v=0
o=FreeSWITCH 1254910732 1254910733 IN IP4 10.4.62.7
s=FreeSWITCH
c=IN IP4 10.4.62.7
t=0 0
m=audio 19594 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2009-10-07 17:45:26.792387 [DEBUG] connection.cxx:1115 SetPhase from
SetUpPhase to AlertingPhase for Call[c8dce50981]-EP<local>[1]
2009-10-07 17:45:26.792387 [NOTICE] mod_sofia.c:1521 Pre-Answer
sofia/internal/1001 at 10.4.62.7!
2009-10-07 17:45:26.793458 [DEBUG] switch_core_session.c:630 Send signal
sofia/internal/1001 at 10.4.62.7 [BREAK]
2009-10-07 17:45:26.793458 [DEBUG] switch_ivr_originate.c:2196 Originate
Resulted in Success: [opal/h323:05492122 at 10.4.4.254<h323%3A05492122 at 10.4.4.254>
]
2009-10-07 17:45:26.793458 [DEBUG] switch_channel.c:182 opal/
h323:05492122 at 10.4.4.254 <h323%3A05492122 at 10.4.4.254> receive message
[AUDIO_SYNC]
2009-10-07 17:45:26.793458 [DEBUG] mod_opal.cpp:1156 Received message 20 on
connection Call[c8dce50981]-EP<local>[1]
2009-10-07 17:45:26.793458 [DEBUG] switch_channel.c:182 sofia/internal/
1001 at 10.4.62.7 receive message [AUDIO_SYNC]
2009-10-07 17:45:26.793458 [DEBUG] switch_ivr_bridge.c:911 opal/
h323:05492122 at 10.4.4.254 <h323%3A05492122 at 10.4.4.254> receive message
[BRIDGE]
2009-10-07 17:45:26.793458 [DEBUG] sofia.c:3359 Channel sofia/internal/
1001 at 10.4.62.7 entering state [early][183]
2009-10-07 17:45:26.793458 [DEBUG] mod_opal.cpp:1156 Received message 4 on
connection Call[c8dce50981]-EP<local>[1]
2009-10-07 17:45:26.793458 [DEBUG] switch_core_session.c:630 Send signal
opal/h323:05492122 at 10.4.4.254 <h323%3A05492122 at 10.4.4.254> [BREAK]
2009-10-07 17:45:26.794816 [DEBUG] mod_opal.cpp:999 Kill 3 on connection
Call[c8dce50981]-EP<local>[1]
2009-10-07 17:45:26.794816 [DEBUG] switch_ivr_bridge.c:918 sofia/internal/
1001 at 10.4.62.7 receive message [BRIDGE]
2009-10-07 17:45:26.794816 [DEBUG] switch_core_session.c:630 Send signal
sofia/internal/1001 at 10.4.62.7 [BREAK]
2009-10-07 17:45:26.794816 [DEBUG] switch_ivr_bridge.c:962 (opal/
h323:05492122 at 10.4.4.254 <h323%3A05492122 at 10.4.4.254>) State Change
CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA

< ----------------- snip ------------------>


 1 entries {
            [0]=dialedDigits "1001282122"
          }
        }
        h245Tunneling = true
      }
    }
  }
2009-10-07 17:45:31.892521 [DEBUG] h323.cxx:582 Handling PDU: Connect
callRef=26906
2009-10-07 17:45:31.892521 [DEBUG] connection.cxx:1115 SetPhase from
AlertingPhase to ConnectedPhase for
Call[c8dce50981]-EP<h323>[localhost/26906]
2009-10-07 17:45:31.892521 [DEBUG] connection.cxx:513 OnConnected for
Call[c8dce50981]-EP<h323>[localhost/26906]
2009-10-07 17:45:31.892521 [DEBUG] manager.cxx:652 OnConnected
Call[c8dce50981]-EP<h323>[localhost/26906]
2009-10-07 17:45:31.892521 [DEBUG] call.cxx:214 OnConnected
Call[c8dce50981]-EP<h323>[localhost/26906]
2009-10-07 17:45:31.892521 [DEBUG] connection.cxx:480 SetConnected for
Call[c8dce50981]-EP<local>[1]
2009-10-07 17:45:31.892521 [DEBUG] connection.cxx:1115 SetPhase from
AlertingPhase to ConnectedPhase for Call[c8dce50981]-EP<local>[1]
2009-10-07 17:45:31.892521 [DEBUG] connection.cxx:1115 SetPhase from
ConnectedPhase to EstablishedPhase for Call[c8dce50981]-EP<local>[1]
2009-10-07 17:45:31.892521 [DEBUG] connection.cxx:520 OnEstablished
Call[c8dce50981]-EP<local>[1]
2009-10-07 17:45:31.892521 [DEBUG] connection.cxx:617 Media stream threads
started.
2009-10-07 17:45:31.892521 [DEBUG] manager.cxx:660 OnEstablished
Call[c8dce50981]-EP<local>[1]
2009-10-07 17:45:31.892521 [DEBUG] call.cxx:245 OnEstablished
Call[c8dce50981]-EP<local>[1]
2009-10-07 17:45:31.892521 [DEBUG] connection.cxx:617 Media stream threads
started.
2009-10-07 17:45:31.892521 [DEBUG] connection.cxx:1115 SetPhase from
ConnectedPhase to EstablishedPhase for
Call[c8dce50981]-EP<h323>[localhost/26906]
2009-10-07 17:45:31.892521 [DEBUG] connection.cxx:520 OnEstablished
Call[c8dce50981]-EP<h323>[localhost/26906]
2009-10-07 17:45:31.892521 [DEBUG] connection.cxx:617 Media stream threads
started.
2009-10-07 17:45:31.893601 [DEBUG] manager.cxx:660 OnEstablished
Call[c8dce50981]-EP<h323>[localhost/26906]
2009-10-07 17:45:31.893601 [DEBUG] call.cxx:245 OnEstablished
Call[c8dce50981]-EP<h323>[localhost/26906]
2009-10-07 17:45:31.893601 [DEBUG] connection.cxx:617 Media stream threads
started.
2009-10-07 17:45:31.893601 [DEBUG] h323.cxx:945 Set protocol version to 5
2009-10-07 17:45:31.893601 [DEBUG] h323.cxx:1261 Set remote party name:
"10.4.4.254"
2009-10-07 17:45:31.893601 [DEBUG] h323.cxx:1269 Set remote application
name: "Avaya Multivantage R014x.00.1.731.2 181/19540 Vox Lucida Pty. Ltd."
2009-10-07 17:45:31.893601 [DEBUG] h323ep.cxx:1087 Received connect PDU.
2009-10-07 17:45:31.893601 [DEBUG] h323.cxx:2367 Fast start response with no
channels to open
2009-10-07 17:45:31.893601 [DEBUG] h323.cxx:3543
InternalEstablishedConnectionCheck: connectionState=HasExecutedSignalConnect
fastStartState=FastStartAcknowledged H.245 is ready



as ConnectMessage was never sent from opal to FS :P




what was your workaround?






>   The  most  important  problem  I'm having right now is that G729 is
>   still  not  working (poor quality due to high buffering). Even with
>   the  latest  Opal,  which  includes  the last week patch for jitter
>   buffer.  If  you  dont  need G729, I could send you a patch for the
>   latest svns (freeswitch, opal, ptlib), ofc no founding needed.
>
>   There are a couple of bugs in opal itself and h323ing freeswitch
>   with opal will bring them in. On the other hand, mod_opal is
>   already there, it just needs a few adjustments.
>
>
> Best wishes,
>
> Claudiu Filip
>
>
>
>
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From msc at freeswitch.org  Wed Oct  7 09:56:41 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 7 Oct 2009 09:56:41 -0700
Subject: [Freeswitch-users] unable to configure Digium TDM400P
In-Reply-To: <a44688a0910070340p4209e7b5yec9218b8c4fc692c@mail.gmail.com>
References: <a44688a0910070340p4209e7b5yec9218b8c4fc692c@mail.gmail.com>
Message-ID: <87f2f3b90910070956l520b2c51p7f33463128bf68b2@mail.gmail.com>

The ztcfg seems okay since it set the signaling to kewlstart. What part is
not working? Be sure to pastebin your openzap.conf and openzap.conf.xml
files as well as a full debug of what happens when loading mod_openzap.
-MC

On Wed, Oct 7, 2009 at 3:40 AM, Woody Dickson <woodydickson at gmail.com>wrote:

> Hi,
>
> I am trying to setup a Digium TDM400P following the instruction on the
> wiki.
> It is a 1 fxo and 1 fxs card, so I tried
> loadzone=in
> defaultzone=in
> fxsks=2
> fxoks=1
>
> and
>
> loadzone=in
> defaultzone=in
> fxsks=1
> fxoks=2
>
> None works.  Does anyone know how it should be configured?
>
> Here is what I get by following the wiki.
>
> [root at localhost zaptel]# ztcfg -vv
>
> Zaptel Version: SVN-branch-1.4-r4629M
> Echo Canceller: MG2
> Configuration
> ======================
>
>
> Channel map:
>
> Channel 01: FXS Kewlstart (Default) (Slaves: 01)
> Channel 02: FXO Kewlstart (Default) (Slaves: 02)
>
> 2 channels to configure.
>
> Changing signalling on channel 1 from FXO Kewlstart to FXS Kewlstart
> Changing signalling on channel 2 from FXS Kewlstart to FXO Kewlstart
>
>
> [root at localhost zaptel]# ztcfg -vv
>
> Zaptel Version: SVN-branch-1.4-r4629M
> Echo Canceller: MG2
> Configuration
> ======================
>
>
> Channel map:
>
> Channel 01: FXO Kewlstart (Default) (Slaves: 01)
> Channel 02: FXS Kewlstart (Default) (Slaves: 02)
>
> 2 channels to configure.
>
> Changing signalling on channel 1 from FXS Kewlstart to FXO Kewlstart
> Changing signalling on channel 2 from FXO Kewlstart to FXS Kewlstart
>
>
> [root at localhost zaptel]# lspci
> 00:14.0 Ethernet controller: Digium, Inc. TDM400P (rev 11)
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Prometheus001 at gmx.net  Wed Oct  7 10:11:27 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Wed, 07 Oct 2009 19:11:27 +0200
Subject: [Freeswitch-users] xml_curl configuration for failover cluster
Message-ID: <4ACCCBBF.4010308@gmx.net>

Hello,

I read in the wiki that binding blocks are processed in sequential order
in a failover matter.

So I  created the following bindings for XML-Curl:
However grepping the network traffic I can see that Freewitch always
fetches both servers fo one binding. So there is no real failover.
How can I avoid that?

Best regards
Peter


<configuration name="xml_curl.conf" description="CURL XML Gateway">
  <bindings>

    <!-- Configuration -->
    <!-- FIRST Application server -->
    <binding name="configuration">
      <param name="gateway-url"
value="http://localhost/xml_curls/configuration" bindings="configuration"/>
    </binding>
    <!-- SECOND Application server -->
    <binding name="configuration_backup">
      <param name="gateway-url"
value="http://10.0.0.104/xml_curls/configuration" bindings="configuration"/>
    </binding>

    <!-- Directory -->
    <!-- FIRST Application server -->
    <binding name="directory">
      <param name="gateway-url"
value="http://localhost/xml_curls/directory" bindings="directory"/>
    </binding>
    <!-- SECOND Application server -->
    <binding name="directory_backup">
      <param name="gateway-url"
value="http://10.0.0.104/xml_curls/directory" bindings="directory"/>
    </binding>

    <!-- Dialplan -->
    <!-- FIRST Application server -->
    <binding name="dialplan">
      <param name="gateway-url"
value="http://localhost/xml_curls/dialplan" bindings="dialplan"/>
    </binding>
    <!-- SECOND Application server -->
    <binding name="dialpla_backup">
      <param name="gateway-url"
value="http://10.0.0.104/xml_curls/dialplan" bindings="dialplan"/>
    </binding>

  </bindings>
</configuration>



From andy at fabulous4.co.uk  Wed Oct  7 10:44:00 2009
From: andy at fabulous4.co.uk (Andy)
Date: Wed, 7 Oct 2009 18:44:00 +0100
Subject: [Freeswitch-users] NAT problems - sorry
In-Reply-To: <5A70A0D4-70FA-43B2-BAF5-F8D6ECC64838@freeswitch.org>
References: <5BE289D2AE22492BA71A6919C8A440B3@D810>
	<5A70A0D4-70FA-43B2-BAF5-F8D6ECC64838@freeswitch.org>
Message-ID: <60E56B6B355C45BCA0D31F4425B9173A@D810>

Thanks Brian, sorry should have pre-empted that one as I've issue before.
start_dtmf is in the dialplan and occurs at the start of every call.
 
On closer inspection however it appears that only part of the nat setup is
taking place.
 
sofia status gives:
 
Name          Type                               Data     State
============================================================================
=====================
                 internal       profile       sip:mod_sofia at 10.10.0.2:5080
RUNNING (0)
                 external       profile       sip:mod_sofia at 10.10.0.2:5060
RUNNING (0)
                  default         alias                           internal
ALIASED
                10.10.0.2         alias                           internal
ALIASED
============================================================================
=====================
2 profiles 2 aliases

 
sofia status profile external gives (the external ip addresses are correct):

API CALL [sofia(status profile external)] output:
============================================================================
=====================
Name                    external
Domain Name             N/A
DBName                  sofia_reg_external
Pres Hosts
Dialplan                XML,enum
Context                 default
Challenge Realm         auto_to
RTP-IP                  10.10.0.2
Ext-RTP-IP              82.5.159.138
SIP-IP                  10.10.0.2
Ext-SIP-IP              82.5.159.138
URL                     sip:mod_sofia at 10.10.0.2:5060
BIND-URL                sip:mod_sofia at 10.10.0.2:5060
HOLD-MUSIC              N/A
OUTBOUND-PROXY          N/A
CODECS                  PCMU,PCMA,GSM
TEL-EVENT               101
DTMF-MODE               rfc2833
CNG                     13
SESSION-TO              0
MAX-DIALOG              0
NOMEDIA                 false
LATE-NEG                false
PROXY-MEDIA             false
AGGRESSIVENAT           false
STUN-ENABLED            true
STUN-AUTO-DISABLE       false
CALLS-IN                0
FAILED-CALLS-IN         0
CALLS-OUT               0
FAILED-CALLS-OUT        0
 
BUT......
 
nat_map status, gives:

API CALL [nat_map(status)] output:
false

And there is no mention of nat detection in the startup log.
 
Is this because I'm using port 5060 externally?
 
Cheers
Andy 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: 07 October 2009 15:22
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] NAT problems - sorry


Bet you its inband dtmf and you need to start the dtmf detector. 


/b


On Oct 7, 2009, at 8:11 AM, Andy wrote:


I can hear the IVR message played down the phone line so outgoing audio is
ok.


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From brian at freeswitch.org  Wed Oct  7 10:49:24 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 7 Oct 2009 12:49:24 -0500
Subject: [Freeswitch-users] NAT problems - sorry
In-Reply-To: <60E56B6B355C45BCA0D31F4425B9173A@D810>
References: <5BE289D2AE22492BA71A6919C8A440B3@D810>
	<5A70A0D4-70FA-43B2-BAF5-F8D6ECC64838@freeswitch.org>
	<60E56B6B355C45BCA0D31F4425B9173A@D810>
Message-ID: <6B2B7FFD-75BC-44CE-A11C-2B26447DA889@freeswitch.org>

No its because you're not behind a upnp/nat-pmp router so you'll have  
to manually forward everything... All the info you showed displaying  
the profile status is correct.

/b

On Oct 7, 2009, at 12:44 PM, Andy wrote:

> Is this because I'm using port 5060 externally?
>
> Cheers
> Andy

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From dome at tel.co.th  Wed Oct  7 10:52:35 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Thu, 8 Oct 2009 00:52:35 +0700
Subject: [Freeswitch-users] mod_sofia.c registered calls how to know
In-Reply-To: <f8af5740910070748g1efee71ft5841b6fa4d2412c8@mail.gmail.com>
References: <f8af5740910070748g1efee71ft5841b6fa4d2412c8@mail.gmail.com>
Message-ID: <8ccbff060910071052x5c27c7c0ub15c69c5c39ff83e@mail.gmail.com>

2009/10/7 srinivasula reddy <srinivas.ksvreddy at gmail.com>:
>
> Hi
>
> can any please tell me where registered calls are stored, so when incoming
> call came to mod_sofia.c how it will check it is registered or not?\\
I use 2 profile
   internal profile require register before make a call.
   external i use odbcquery  check ipaddress before process call. (in dialplan)

Dome C.
>
> --
> Srinivasula Reddy K
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From andy at fabulous4.co.uk  Wed Oct  7 10:58:19 2009
From: andy at fabulous4.co.uk (Andy)
Date: Wed, 7 Oct 2009 18:58:19 +0100
Subject: [Freeswitch-users] NAT problems - sorry
In-Reply-To: <6B2B7FFD-75BC-44CE-A11C-2B26447DA889@freeswitch.org>
References: <5BE289D2AE22492BA71A6919C8A440B3@D810><5A70A0D4-70FA-43B2-BAF5-F8D6ECC64838@freeswitch.org><60E56B6B355C45BCA0D31F4425B9173A@D810>
	<6B2B7FFD-75BC-44CE-A11C-2B26447DA889@freeswitch.org>
Message-ID: <703EEB9AC9AD43898C6C46F82904E6A9@D810>

Many thanks Brian, the firewall docs assure me it is uPnp but is probably
lying or a poor implementation. Could you point me to the right section of
the Wiki to tell me how to do this manually as I've been scouting for some
time and can;t seem to find the right thing. sorry if I'm being blind.

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: 07 October 2009 18:49
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] NAT problems - sorry


No its because you're not behind a upnp/nat-pmp router so you'll have to
manually forward everything... All the info you showed displaying the
profile status is correct. 

/b

On Oct 7, 2009, at 12:44 PM, Andy wrote:



Is this because I'm using port 5060 externally?
 
Cheers
Andy 


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From brian at freeswitch.org  Wed Oct  7 11:06:37 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 7 Oct 2009 13:06:37 -0500
Subject: [Freeswitch-users] NAT problems - sorry
In-Reply-To: <703EEB9AC9AD43898C6C46F82904E6A9@D810>
References: <5BE289D2AE22492BA71A6919C8A440B3@D810><5A70A0D4-70FA-43B2-BAF5-F8D6ECC64838@freeswitch.org><60E56B6B355C45BCA0D31F4425B9173A@D810>
	<6B2B7FFD-75BC-44CE-A11C-2B26447DA889@freeswitch.org>
	<703EEB9AC9AD43898C6C46F82904E6A9@D810>
Message-ID: <0A2A7A27-611E-40FA-8268-D2C451AA8B77@freeswitch.org>

s/auto-nat/$realip/ then forward the rtp ports and sip ports.

/b
PS chances are you have to ENABLE upnp.
On Oct 7, 2009, at 12:58 PM, Andy wrote:

> Many thanks Brian, the firewall docs assure me it is uPnp but is  
> probably lying or a poor implementation. Could you point me to the  
> right section of the Wiki to tell me how to do this manually as I've  
> been scouting for some time and can;t seem to find the right thing.  
> sorry if I'm being blind.

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From andy at fabulous4.co.uk  Wed Oct  7 11:48:33 2009
From: andy at fabulous4.co.uk (Andy)
Date: Wed, 7 Oct 2009 19:48:33 +0100
Subject: [Freeswitch-users] NAT problems - sorry
In-Reply-To: <0A2A7A27-611E-40FA-8268-D2C451AA8B77@freeswitch.org>
References: <5BE289D2AE22492BA71A6919C8A440B3@D810><5A70A0D4-70FA-43B2-BAF5-F8D6ECC64838@freeswitch.org><60E56B6B355C45BCA0D31F4425B9173A@D810><6B2B7FFD-75BC-44CE-A11C-2B26447DA889@freeswitch.org><703EEB9AC9AD43898C6C46F82904E6A9@D810>
	<0A2A7A27-611E-40FA-8268-D2C451AA8B77@freeswitch.org>
Message-ID: <1B629AB98E50412496D2673E1FE160D9@D810>

Thanks Brian,
 
I've now set the external ips manually to be my external ip and have forward
all ports through my firewall to the FS server. It's actually set up as a
DMZ to everything is being forwarded without restriction but sadly DTMF and
HANGUP messages are still not getting through. Have I misunderstood what is
required. Is there some additional forwarding within FS required. I'm really
sorry to keep coming back but I've been wrestling with this for a long time
now and not getting anywhere.
 
Many thanks
Andy 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: 07 October 2009 19:07
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] NAT problems - sorry


s/auto-nat/$realip/ then forward the rtp ports and sip ports. 

/b
PS chances are you have to ENABLE upnp.

On Oct 7, 2009, at 12:58 PM, Andy wrote:


Many thanks Brian, the firewall docs assure me it is uPnp but is probably
lying or a poor implementation. Could you point me to the right section of
the Wiki to tell me how to do this manually as I've been scouting for some
time and can;t seem to find the right thing. sorry if I'm being blind.



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From rupa at rupa.com  Wed Oct  7 12:30:16 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 7 Oct 2009 13:30:16 -0600
Subject: [Freeswitch-users] NAT problems - sorry
In-Reply-To: <1B629AB98E50412496D2673E1FE160D9@D810>
References: <5BE289D2AE22492BA71A6919C8A440B3@D810>
	<5A70A0D4-70FA-43B2-BAF5-F8D6ECC64838@freeswitch.org>
	<60E56B6B355C45BCA0D31F4425B9173A@D810>
	<6B2B7FFD-75BC-44CE-A11C-2B26447DA889@freeswitch.org>
	<703EEB9AC9AD43898C6C46F82904E6A9@D810>
	<0A2A7A27-611E-40FA-8268-D2C451AA8B77@freeswitch.org>
	<1B629AB98E50412496D2673E1FE160D9@D810>
Message-ID: <b39273530910071230x1e5c3ddawa72239545c1d86c6@mail.gmail.com>

Double check your firewall and:

1) ensure you've actually enabled UPNP
and
2) Ensure that any mention of a SIP ALG (application level gateway) is
turned off.  SIP ALGs tend to really screw things up.

On Wed, Oct 7, 2009 at 12:48 PM, Andy <andy at fabulous4.co.uk> wrote:
> I've now set the external ips manually to be my external ip and have forward
> all ports through my firewall to the FS server. It's actually set up as a
> DMZ to everything is?being forwarded without restriction but sadly DTMF and
> HANGUP messages are still not getting through. Have I misunderstood what is
> required. Is there some additional forwarding within FS required. I'm really
> sorry to keep?coming back but I've been wrestling with this?for a long time
> now and not?getting anywhere.
>



-- 
-Rupa



From kristian.kielhofner at gmail.com  Wed Oct  7 12:35:56 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Wed, 7 Oct 2009 15:35:56 -0400
Subject: [Freeswitch-users] linksys spa-3102 fax t38 + freeswitch media
	proxy
In-Reply-To: <4ACB9519.9000805@gmail.com>
References: <4ACB9519.9000805@gmail.com>
Message-ID: <2d9149cd0910071235k27c1429v2bc555c0e7397fc4@mail.gmail.com>

Since no one else has responded I'll chime in with some general advice.

It's troubling to see that your "provider" is using Asterisk to face
you (the customer).

I've never had any luck getting T.38 to work (at all, in any mode)
using Asterisk.  I've heard of other people making it work but
consider the chain here:

Their provider (they appear to be using DIDx, hmmm) -> Callwithus
(using Asterisk) -> You (FreeSWITCH) -> Your device

I don't see anything specific in your case but there are many, many
places for T.38 to go wrong in this scenario.  I'd recommend
attempting T.38 against a local gateway or known Tier 1 provider
supporting T.38 with known equipment.

Otherwise, do you have any updates for us?

On Tue, Oct 6, 2009 at 3:06 PM, Vladimir Elizarov
<xengelpublicx at gmail.com> wrote:
> Hello.
>
> I want to set up faxing via the gateway linksys spa-3102 (with support
> for t38) via SIP.
>
> SIP-client -> linksys spa3102 -> fs -> provider
>
> ? ?<gateway name="callwithus">
> ? ? ? ?....
> ? ? ? ?<param name="inbound-late-negotiation" value="true"/>
> ? ? ? ?<param name="stun-enabled" value="true"/>
> ? ?</gateway>
>
> <extension name="Fax">
> ? ?<condition field="destination_number" expression="^(30)(\d{11})$">
> ? ? ? ? ? ?<action application="set" data="proxy_media=true"/>
> ? ? ? ? ? ?<action application="set" data="bypass_media=false"/>
> ? ? ? ? ? ?<action application="bridge"
> data="sofia/gateway/callwithus/*4087$2 at sip.callwithus.com"/>
> ? ? ? ?</condition>
> </extension>
>
> I make a call, press start on the fax. Fax not sent. Get this in the log
> (UA freeswitch not trunk-12805):
>
> ? ------------------------------------------------------------------------
> ? INVITE sip:*408774957558438 at sip.callwithus.com SIP/2.0
> ? Via: SIP/2.0/UDP 77.239.230.202:5080;rport;branch=z9hG4bK8yjrHeUprX2Hr
> ? Max-Forwards: 99
> ? From: "Mihailova Ludmila"
> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
> ? To: <sip:*408774957558438 at sip.callwithus.com>
> ? Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
> ? CSeq: 121302430 INVITE
> ? Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
> ? Supported: timer, precondition, path, replaces
> ? Allow-Events: talk, refer
> ? Content-Type: application/sdp
> ? Content-Disposition: session
> ? Content-Length: 250
> ? Remote-Party-ID: "Mihailova Ludmila"
> <sip:214 at sip.callwithus.com>;party=calling;screen=yes;privacy=off
>
> ? v=0
> ? o=- 216896551 216896551 IN IP4 192.168.50.51
> ? s=-
> ? c=IN IP4 77.239.230.202
> ? t=0 0
> ? m=audio 32404 RTP/AVP 0 100 101
> ? a=rtpmap:0 PCMU/8000
> ? a=rtpmap:100 NSE/8000
> ? a=fmtp:100 192-193
> ? a=rtpmap:101 telephone-event/8000
> ? a=fmtp:101 0-15
> ? a=ptime:30
>
> ? ------------------------------------------------------------------------
> send 1333 bytes to udp/[38.99.70.232]:5060 at 15:05:34.485226:
> ? ------------------------------------------------------------------------
> ? INVITE sip:*408774957558438 at sip.callwithus.com SIP/2.0
> ? Via: SIP/2.0/UDP 77.239.230.202:5080;rport;branch=z9hG4bK97BHK9BtN6r4K
> ? Max-Forwards: 99
> ? From: "Mihailova Ludmila"
> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
> ? To: <sip:*408774957558438 at sip.callwithus.com>
> ? Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
> ? CSeq: 121302431 INVITE
> ? Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
> ? Expires: 3600
> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
> ? Supported: timer, precondition, path, replaces
> ? Allow-Events: talk, refer
> ? Content-Type: application/sdp
> ? Content-Disposition: session
> ? Content-Length: 250
> ? Remote-Party-ID: "Mihailova Ludmila"
> <sip:214 at sip.callwithus.com>;party=calling;screen=yes;privacy=off
>
> ? v=0
> ? o=- 216896551 216896551 IN IP4 192.168.50.51
> ? s=-
> ? c=IN IP4 77.239.230.202
> ? t=0 0
> ? m=audio 32404 RTP/AVP 0 100 101
> ? a=rtpmap:0 PCMU/8000
> ? a=rtpmap:100 NSE/8000
> ? a=fmtp:100 192-193
> ? a=rtpmap:101 telephone-event/8000
> ? a=fmtp:101 0-15
> ? a=ptime:30
> ? ------------------------------------------------------------------------
> ? SIP/2.0 100 Giving a try
> ? Via: SIP/2.0/UDP
> 192.168.50.10:5080;rport=5080;branch=z9hG4bK97BHK9BtN6r4K
> ? From: "Mihailova Ludmila"
> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
> ? To: <sip:*408774957558438 at sip.callwithus.com>
> ? Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
> ? CSeq: 121302431 INVITE
> ? Server: CWU SIP GW
> ? Content-Length: 0
>
> ? ------------------------------------------------------------------------
> recv 836 bytes from udp/[38.99.70.232]:5060 at 15:05:49.641312:
> ? ------------------------------------------------------------------------
> ? SIP/2.0 183 Session Progress
> ? Via: SIP/2.0/UDP
> 192.168.50.10:5080;rport=5080;branch=z9hG4bK97BHK9BtN6r4K
> ? Record-Route:
> <sip:38.99.70.232;lr=on;ftag=HXjrDKpHHX8ZN;did=e8e.089fc2b6>
> ? From: "Mihailova Ludmila"
> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
> ? To: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
> ? Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
> ? CSeq: 121302431 INVITE
> ? User-Agent: CWU SIP-GW
> ? Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> ? Supported: replaces
> ? Contact: <sip:*408774957558438 at 204.74.213.1:5062>
> ? Content-Type: application/sdp
> ? Content-Length: 209
>
> ? v=0
> ? o=root 31797 31797 IN IP4 4.55.17.66
> ? s=session
> ? c=IN IP4 4.55.17.66
> ? t=0 0
> ? m=audio 14104 RTP/AVP 0 101
> ? a=rtpmap:0 PCMU/8000
> ? a=rtpmap:101 telephone-event/8000
> ? a=fmtp:101 0-16
> ? a=ptime:20
> ? a=sendrecv
> ? ------------------------------------------------------------------------
> v=0
> o=root 31797 31797 IN IP4 4.55.17.66
> s=session
> c=IN IP4 4.55.17.66
> t=0 0
> m=audio 14104 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
> 2009-10-06 19:03:31.112988 [NOTICE] sofia.c:3449 Pre-Answer
> sofia/external/*408774957558438 at sip.callwithus.com!
> 2009-10-06 19:03:31.112988 [DEBUG] switch_channel.c:1822 Send signal
> sofia/internal/214 at pbx0.tssec.lan [BREAK]
> 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2095 Set Codec
> sofia/external/*408774957558438 at sip.callwithus.com PROXY/8000 20 ms 160
> samples
> 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2372 PROXY AUDIO RTP
> [sofia/external/*408774957558438 at sip.callwithus.com]
> 192.168.50.10:32404->4.55.17.66:14104 codec: 0 ms: 20
> 2009-10-06 19:03:31.112988 [DEBUG] switch_rtp.c:1163 Not using a timer
> 2009-10-06 19:03:31.112988 [DEBUG] switch_ivr_originate.c:2154
> sofia/internal/214 at pbx0.tssec.lan receive message [PROGRESS]
> 2009-10-06 19:03:31.112988 [INFO] switch_ivr_originate.c:2154 Sending
> early media
> 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:1322
> sofia/internal/214 at pbx0.tssec.lan Patched SDP
> ---
> v=0
> o=root 31797 31797 IN IP4 4.55.17.66
> s=session
> c=IN IP4 4.55.17.66
> t=0 0
> m=audio 14104 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
> +++
> v=0
> o=root 31797 31797 IN IP4 4.55.17.66
> s=session
> c=IN IP4 192.168.50.10
> t=0 0
> m=audio 32698 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
> 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2372 PROXY AUDIO RTP
> [sofia/internal/214 at pbx0.tssec.lan]
> 192.168.50.10:32698->192.168.50.51:16474 codec: 0 ms: 30
> 2009-10-06 19:03:31.112988 [DEBUG] switch_rtp.c:1163 Not using a timer
> 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2508 Set comfort noise
> payload to 13
> 2009-10-06 19:03:31.112988 [DEBUG] sofia.c:3462
> sofia/internal/214 at pbx0.tssec.lan receive message [PROGRESS]
> 2009-10-06 19:03:31.116989 [NOTICE] mod_sofia.c:1521 Pre-Answer
> sofia/internal/214 at pbx0.tssec.lan!
> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal
> sofia/internal/214 at pbx0.tssec.lan [BREAK]
> 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_originate.c:2196 Originate
> Resulted in Success: [sofia/external/*408774957558438 at sip.callwithus.com]
> 2009-10-06 19:03:31.116989 [DEBUG] switch_channel.c:182
> sofia/external/*408774957558438 at sip.callwithus.com receive message
> [AUDIO_SYNC]
> 2009-10-06 19:03:31.116989 [DEBUG] switch_channel.c:182
> sofia/internal/214 at pbx0.tssec.lan receive message [AUDIO_SYNC]
> 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_bridge.c:911
> sofia/external/*408774957558438 at sip.callwithus.com receive message [BRIDGE]
> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal
> sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
> 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_bridge.c:918
> sofia/internal/214 at pbx0.tssec.lan receive message [BRIDGE]
> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal
> sofia/internal/214 at pbx0.tssec.lan [BREAK]
> 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_bridge.c:962
> (sofia/external/*408774957558438 at sip.callwithus.com) State Change
> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
> 2009-10-06 19:03:31.116989 [DEBUG] sofia.c:3359 Channel
> sofia/internal/214 at pbx0.tssec.lan entering state [early][183]
> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal
> sofia/internal/214 at pbx0.tssec.lan [BREAK]
> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/*408774957558438 at sip.callwithus.com) Running State
> Change CS_EXCHANGE_MEDIA
> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_state_machine.c:343
> (sofia/external/*408774957558438 at sip.callwithus.com) State EXCHANGE_MEDIA
> 2009-10-06 19:03:31.116989 [DEBUG] mod_sofia.c:430 SOFIA LOOPBACK
> recv 822 bytes from udp/[38.99.70.232]:5060 at 15:05:58.503701:
> ? ------------------------------------------------------------------------
> ? SIP/2.0 200 OK
> ? Via: SIP/2.0/UDP
> 192.168.50.10:5080;rport=5080;branch=z9hG4bK97BHK9BtN6r4K
> ? Record-Route:
> <sip:38.99.70.232;lr=on;ftag=HXjrDKpHHX8ZN;did=e8e.089fc2b6>
> ? From: "Mihailova Ludmila"
> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
> ? To: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
> ? Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
> ? CSeq: 121302431 INVITE
> ? User-Agent: CWU SIP-GW
> ? Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> ? Supported: replaces
> ? Contact: <sip:*408774957558438 at 204.74.213.1:5062>
> ? Content-Type: application/sdp
> ? Content-Length: 209
>
> ? v=0
> ? o=root 31797 31798 IN IP4 4.55.17.66
> ? s=session
> ? c=IN IP4 4.55.17.66
> ? t=0 0
> ? m=audio 14104 RTP/AVP 0 101
> ? a=rtpmap:0 PCMU/8000
> ? a=rtpmap:101 telephone-event/8000
> ? a=fmtp:101 0-16
> ? a=ptime:20
> ? a=sendrecv
> ? ------------------------------------------------------------------------
> send 762 bytes to udp/[38.99.70.232]:5060 at 15:05:58.507702:
> ? ------------------------------------------------------------------------
> ? ACK sip:*408774957558438 at 204.74.213.1:5062 SIP/2.0
> ? Via: SIP/2.0/UDP 192.168.50.10:5080;rport;branch=z9hG4bKaH59m4vXjFFQF
> ? Route: <sip:38.99.70.232;lr=on;ftag=HXjrDKpHHX8ZN;did=e8e.089fc2b6>
> ? Max-Forwards: 70
> ? From: "Mihailova Ludmila"
> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
> ? To: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
> ? Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
> ? CSeq: 121302431 ACK
> ? Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
> ? Content-Length: 0
>
> ? ------------------------------------------------------------------------
> 2009-10-06 19:03:39.975377 [DEBUG] sofia.c:3359 Channel
> sofia/external/*408774957558438 at sip.callwithus.com entering state
> [ready][200]
> 2009-10-06 19:03:39.975377 [DEBUG] switch_channel.c:1935 Send signal
> sofia/internal/214 at pbx0.tssec.lan [BREAK]
> 2009-10-06 19:03:39.975377 [NOTICE] sofia.c:3814 Channel
> [sofia/external/*408774957558438 at sip.callwithus.com] has been answered
> 2009-10-06 19:03:39.975377 [DEBUG] switch_channel.c:182
> sofia/external/*408774957558438 at sip.callwithus.com receive message
> [AUDIO_SYNC]
> 2009-10-06 19:03:39.975377 [DEBUG] sofia.c:3819
> sofia/internal/214 at pbx0.tssec.lan receive message [ANSWER]
> 2009-10-06 19:03:39.975377 [DEBUG] sofia_glue.c:1322
> sofia/internal/214 at pbx0.tssec.lan Patched SDP
> ---
> v=0
> o=root 31797 31797 IN IP4 4.55.17.66
> s=session
> c=IN IP4 4.55.17.66
> t=0 0
> m=audio 14104 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
> +++
> v=0
> o=root 31797 31797 IN IP4 4.55.17.66
> s=session
> c=IN IP4 192.168.50.10
> t=0 0
> m=audio 32698 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
> 2009-10-06 19:03:39.975377 [DEBUG] switch_core_session.c:630 Send signal
> sofia/internal/214 at pbx0.tssec.lan [BREAK]
> 2009-10-06 19:03:39.975377 [NOTICE] sofia.c:3819 Channel
> [sofia/internal/214 at pbx0.tssec.lan] has been answered
> 2009-10-06 19:03:39.975377 [DEBUG] switch_channel.c:182
> sofia/internal/214 at pbx0.tssec.lan receive message [AUDIO_SYNC]
> 2009-10-06 19:03:39.975377 [DEBUG] sofia.c:3359 Channel
> sofia/internal/214 at pbx0.tssec.lan entering state [completed][200]
> 2009-10-06 19:03:39.991381 [DEBUG] sofia.c:3359 Channel
> sofia/internal/214 at pbx0.tssec.lan entering state [ready][200]
> recv 1054 bytes from udp/[38.99.70.232]:5060 at 15:06:05.577607:
> ? ------------------------------------------------------------------------
> ? INVITE sip:gw+callwithus at 192.168.50.10:5080;transport=udp SIP/2.0
> ? Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.0
> ? Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK41d48157;rport=5062
> ? From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
> ? To: "Mihailova Ludmila"
> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
> ? Contact: <sip:*408774957558438 at 204.74.213.1:5062>
> ? Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
> ? CSeq: 102 INVITE
> ? User-Agent: CWU SIP-GW
> ? Max-Forwards: 69
> ? Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> ? Supported: replaces
> ? X-asterisk-info: SIP re-invite (T38 switchover)
> ? Content-Type: application/sdp
> ? Content-Length: 350
>
> ? v=0
> ? o=root 31797 31799 IN IP4 32.50.63.192
> ? s=session
> ? c=IN IP4 32.50.63.192
> ? t=0 0
> ? m=image 14104 udptl t38
> ? a=T38FaxVersion:0
> ? a=T38MaxBitRate:9600
> ? a=T38FaxFillBitRemoval:0
> ? a=T38FaxTranscodingMMR:0
> ? a=T38FaxTranscodingJBIG:0
> ? a=T38FaxRateManagement:transferredTCF
> ? a=T38FaxMaxBuffer:176
> ? a=T38FaxMaxDatagram:176
> ? a=T38FaxUdpEC:t38UDPRedundancy
> ? ------------------------------------------------------------------------
> send 443 bytes to udp/[38.99.70.232]:5060 at 15:06:05.577607:
> ? ------------------------------------------------------------------------
> ? SIP/2.0 100 Trying
> ? Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.0
> ? Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK41d48157;rport=5062
> ? From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
> ? To: "Mihailova Ludmila"
> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
> ? Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
> ? CSeq: 102 INVITE
> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
> ? Content-Length: 0
>
> ? ------------------------------------------------------------------------
> 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3359 Channel
> sofia/external/*408774957558438 at sip.callwithus.com entering state
> [received][100]
> 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3366 Remote SDP:
> v=0
> o=root 31797 31799 IN IP4 32.50.63.192
> s=session
> c=IN IP4 32.50.63.192
> t=0 0
> m=image 14104 udptl t38
> a=T38FaxVersion:0
> a=T38MaxBitRate:9600
> a=T38FaxFillBitRemoval:0
> a=T38FaxTranscodingMMR:0
> a=T38FaxTranscodingJBIG:0
> a=T38FaxRateManagement:transferredTCF
> a=T38FaxMaxBuffer:176
> a=T38FaxMaxDatagram:176
> a=T38FaxUdpEC:t38UDPRedundancy
>
> 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3683 Passing SDP to other leg.
> v=0
> o=root 31797 31799 IN IP4 32.50.63.192
> s=session
> c=IN IP4 32.50.63.192
> t=0 0
> m=image 14104 udptl t38
> a=T38FaxVersion:0
> a=T38MaxBitRate:9600
> a=T38FaxFillBitRemoval:0
> a=T38FaxTranscodingMMR:0
> a=T38FaxTranscodingJBIG:0
> a=T38FaxRateManagement:transferredTCF
> a=T38FaxMaxBuffer:176
> a=T38FaxMaxDatagram:176
> a=T38FaxUdpEC:t38UDPRedundancy
>
> 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3696
> sofia/internal/214 at pbx0.tssec.lan receive message [MEDIA_REDIRECT]
> 2009-10-06 19:03:47.049284 [DEBUG] mod_sofia.c:1195
> sofia/internal/214 at pbx0.tssec.lan Sending media re-direct:
> v=0
> o=root 31797 31799 IN IP4 32.50.63.192
> s=session
> c=IN IP4 32.50.63.192
> t=0 0
> m=image 14104 udptl t38
> a=T38FaxVersion:0
> a=T38MaxBitRate:9600
> a=T38FaxFillBitRemoval:0
> a=T38FaxTranscodingMMR:0
> a=T38FaxTranscodingJBIG:0
> a=T38FaxRateManagement:transferredTCF
> a=T38FaxMaxBuffer:176
> a=T38FaxMaxDatagram:176
> a=T38FaxUdpEC:t38UDPRedundancy
>
> 2009-10-06 19:03:47.049284 [DEBUG] sofia_glue.c:1101 Remote address:port
> [192.168.50.51:16474] has not changed.
> 2009-10-06 19:03:47.049284 [DEBUG] sofia_glue.c:1322
> sofia/internal/214 at pbx0.tssec.lan Patched SDP
> ---
> v=0
> o=root 31797 31799 IN IP4 32.50.63.192
> s=session
> c=IN IP4 32.50.63.192
> t=0 0
> m=image 14104 udptl t38
> a=T38FaxVersion:0
> a=T38MaxBitRate:9600
> a=T38FaxFillBitRemoval:0
> a=T38FaxTranscodingMMR:0
> a=T38FaxTranscodingJBIG:0
> a=T38FaxRateManagement:transferredTCF
> a=T38FaxMaxBuffer:176
> a=T38FaxMaxDatagram:176
> a=T38FaxUdpEC:t38UDPRedundancy
>
> +++
> v=0
> o=root 31797 31799 IN IP4 32.50.63.192
> s=session
> c=IN IP4 192.168.50.10
> t=0 0
> m=image 32698 udptl t38
> a=T38FaxVersion:0
> a=T38MaxBitRate:9600
> a=T38FaxFillBitRemoval:0
> a=T38FaxTranscodingMMR:0
> a=T38FaxTranscodingJBIG:0
> a=T38FaxRateManagement:transferredTCF
> a=T38FaxMaxBuffer:176
> a=T38FaxMaxDatagram:176
> a=T38FaxUdpEC:t38UDPRedundancy
>
> 2009-10-06 19:03:47.049284 [DEBUG] switch_core_session.c:630 Send signal
> sofia/internal/214 at pbx0.tssec.lan [BREAK]
> 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3359 Channel
> sofia/internal/214 at pbx0.tssec.lan entering state [calling][0]
> 2009-10-06 19:03:47.077291 [DEBUG] sofia_glue.c:1101 Remote address:port
> [192.168.50.51:16474] has not changed.
> 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3158 Passing 200 OK to other leg
> 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3170
> sofia/external/*408774957558438 at sip.callwithus.com receive message [RESPOND]
> 2009-10-06 19:03:47.077291 [DEBUG] mod_sofia.c:1427 Responding with 200 [OK]
> 2009-10-06 19:03:47.077291 [DEBUG] sofia_glue.c:1322
> sofia/external/*408774957558438 at sip.callwithus.com Patched SDP
> ---
> v=0
> o=- 216899808 216899808 IN IP4 192.168.50.51
> s=-
> c=IN IP4 192.168.50.51
> t=0 0
> m=image 16474 udptl t38
> a=T38FaxVersion:0
> a=T38MaxBitRate:14400
> a=T38FaxRateManagement:transferredTCF
> a=T38FaxMaxBuffer:200
> a=T38FaxMaxDatagram:200
> a=T38FaxUdpEC:t38UDPRedundancy
>
> +++
> v=0
> o=- 216899808 216899808 IN IP4 192.168.50.51
> s=-
> c=IN IP4 77.239.230.202
> t=0 0
> m=image 32404 udptl t38
> a=T38FaxVersion:0
> a=T38MaxBitRate:14400
> a=T38FaxRateManagement:transferredTCF
> a=T38FaxMaxBuffer:200
> a=T38FaxMaxDatagram:200
> a=T38FaxUdpEC:t38UDPRedundancy
>
> 2009-10-06 19:03:47.077291 [DEBUG] sofia_glue.c:1110 AUDIO RTP CHANGING
> DEST TO: [32.50.63.192:14104]
> 2009-10-06 19:03:47.077291 [DEBUG] switch_core_session.c:630 Send signal
> sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
> 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3359 Channel
> sofia/internal/214 at pbx0.tssec.lan entering state [ready][200]
> 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3366 Remote SDP:
> v=0
> o=- 216899808 216899808 IN IP4 192.168.50.51
> s=-
> c=IN IP4 192.168.50.51
> t=0 0
> m=image 16474 udptl t38
> a=T38FaxVersion:0
> a=T38MaxBitRate:14400
> a=T38FaxRateManagement:transferredTCF
> a=T38FaxMaxBuffer:200
> a=T38FaxMaxDatagram:200
> a=T38FaxUdpEC:t38UDPRedundancy
>
> send 1003 bytes to udp/[38.99.70.232]:5060 at 15:06:05.605615:
> ? ------------------------------------------------------------------------
> ? SIP/2.0 200 OK
> ? Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.0
> ? Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK41d48157;rport=5062
> ? From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
> ? To: "Mihailova Ludmila"
> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
> ? Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
> ? CSeq: 102 INVITE
> ? Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
> ? Accept: application/sdp
> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
> ? Supported: timer, precondition, path, replaces
> ? Content-Type: application/sdp
> ? Content-Disposition: session
> ? Content-Length: 274
>
> ? v=0
> ? o=- 216896551 216896552 IN IP4 192.168.50.51
> ? s=-
> ? c=IN IP4 77.239.230.202
> ? t=0 0
> ? m=image 32404 udptl t38
> ? a=T38FaxVersion:0
> ? a=T38MaxBitRate:14400
> ? a=T38FaxRateManagement:transferredTCF
> ? a=T38FaxMaxBuffer:200
> ? a=T38FaxMaxDatagram:200
> ? a=T38FaxUdpEC:t38UDPRedundancy
> ? ------------------------------------------------------------------------
> 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3359 Channel
> sofia/external/*408774957558438 at sip.callwithus.com entering state
> [completed][200]
> 2009-10-06 19:03:47.285347 [INFO] switch_rtp.c:1905 Auto Changing port
> from 32.50.63.192:14104 to 4.55.17.66:14104
> recv 527 bytes from udp/[38.99.70.232]:5060 at 15:06:05.817672:
> ? ------------------------------------------------------------------------
> ? ACK sip:gw+callwithus at 192.168.50.10:5080;transport=udp SIP/2.0
> ? Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.2
> ? Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK5ffd72cd;rport=5062
> ? From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
> ? To: "Mihailova Ludmila"
> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
> ? Contact: <sip:*408774957558438 at 204.74.213.1:5062>
> ? Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
> ? CSeq: 102 ACK
> ? User-Agent: CWU SIP-GW
> ? Max-Forwards: 69
> ? Content-Length: 0
>
> ? ------------------------------------------------------------------------
> 2009-10-06 19:03:47.289349 [DEBUG] sofia.c:3359 Channel
> sofia/external/*408774957558438 at sip.callwithus.com entering state
> [ready][200]
> recv 476 bytes from udp/[38.99.70.232]:5060 at 15:06:47.588933:
> ? ------------------------------------------------------------------------
> ? BYE sip:gw+callwithus at 192.168.50.10:5080;transport=udp SIP/2.0
> ? Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKd406.f8553e14.0
> ? Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK0f2cff01;rport=5062
> ? From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
> ? To: "Mihailova Ludmila"
> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
> ? Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
> ? CSeq: 103 BYE
> ? User-Agent: CWU SIP-GW
> ? Max-Forwards: 69
> ? Content-Length: 0
>
> ? ------------------------------------------------------------------------
> 2009-10-06 19:04:29.060609 [NOTICE] sofia.c:328 Hangup
> sofia/external/*408774957558438 at sip.callwithus.com [CS_EXCHANGE_MEDIA]
> [NORMAL_CLEARING]
> 2009-10-06 19:04:29.060609 [DEBUG] switch_channel.c:1726 Send signal
> sofia/external/*408774957558438 at sip.callwithus.com [KILL]
> 2009-10-06 19:04:29.060609 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
> 2009-10-06 19:04:29.060609 [DEBUG] switch_core_state_machine.c:437
> thread mismatch skipping state handler.
> send 574 bytes to udp/[38.99.70.232]:5060 at 15:06:47.588933:
> ? ------------------------------------------------------------------------
> ? SIP/2.0 200 OK
> ? Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKd406.f8553e14.0
> ? Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK0f2cff01;rport=5062
> ? From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
> ? To: "Mihailova Ludmila"
> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
> ? Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
> ? CSeq: 103 BYE
> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
> ? Supported: timer, precondition, path, replaces
> ? Content-Length: 0
>
> ? ------------------------------------------------------------------------
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From brian at freeswitch.org  Wed Oct  7 13:09:45 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 7 Oct 2009 15:09:45 -0500
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
Message-ID: <82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>

 From what I have been told h323plus is a based/fork of OpenH323 which  
OPAL is just a continuation of OpenH323.  So why not support the  
developers of OPAL/OpenH323 ?

/b

On Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote:

> We are developing module to handle h323 proto now, we try to use  
> mod_opal and try improve it, but no luck,
> there is many issues in libopal, and finaly we now move to h323plus  
> library.




From xengelpublicx at gmail.com  Wed Oct  7 14:26:06 2009
From: xengelpublicx at gmail.com (Vladimir Elizarov)
Date: Thu, 08 Oct 2009 01:26:06 +0400
Subject: [Freeswitch-users] linksys spa-3102 fax t38 + freeswitch media
 proxy
In-Reply-To: <2d9149cd0910071235k27c1429v2bc555c0e7397fc4@mail.gmail.com>
References: <4ACB9519.9000805@gmail.com>
	<2d9149cd0910071235k27c1429v2bc555c0e7397fc4@mail.gmail.com>
Message-ID: <4ACD076E.2080107@gmail.com>

Kristian Kielhofner ?????:
> Since no one else has responded I'll chime in with some general advice.
>
> It's troubling to see that your "provider" is using Asterisk to face
> you (the customer).
>
> I've never had any luck getting T.38 to work (at all, in any mode)
> using Asterisk.  I've heard of other people making it work but
> consider the chain here:
>
> Their provider (they appear to be using DIDx, hmmm) -> Callwithus
> (using Asterisk) -> You (FreeSWITCH) -> Your device
>
> I don't see anything specific in your case but there are many, many
> places for T.38 to go wrong in this scenario.  I'd recommend
> attempting T.38 against a local gateway or known Tier 1 provider
> supporting T.38 with known equipment.
>   
Can you advise sip-providers offering t38?
> Otherwise, do you have any updates for us?
>   
No. I can not add anything.


> On Tue, Oct 6, 2009 at 3:06 PM, Vladimir Elizarov
> <xengelpublicx at gmail.com> wrote:
>   
>> Hello.
>>
>> I want to set up faxing via the gateway linksys spa-3102 (with support
>> for t38) via SIP.
>>
>> SIP-client -> linksys spa3102 -> fs -> provider
>>
>>    <gateway name="callwithus">
>>        ....
>>        <param name="inbound-late-negotiation" value="true"/>
>>        <param name="stun-enabled" value="true"/>
>>    </gateway>
>>
>> <extension name="Fax">
>>    <condition field="destination_number" expression="^(30)(\d{11})$">
>>            <action application="set" data="proxy_media=true"/>
>>            <action application="set" data="bypass_media=false"/>
>>            <action application="bridge"
>> data="sofia/gateway/callwithus/*4087$2 at sip.callwithus.com"/>
>>        </condition>
>> </extension>
>>
>> I make a call, press start on the fax. Fax not sent. Get this in the log
>> (UA freeswitch not trunk-12805):
>>
>>   ------------------------------------------------------------------------
>>   INVITE sip:*408774957558438 at sip.callwithus.com SIP/2.0
>>   Via: SIP/2.0/UDP 77.239.230.202:5080;rport;branch=z9hG4bK8yjrHeUprX2Hr
>>   Max-Forwards: 99
>>   From: "Mihailova Ludmila"
>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>   To: <sip:*408774957558438 at sip.callwithus.com>
>>   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>   CSeq: 121302430 INVITE
>>   Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
>>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY
>>   Supported: timer, precondition, path, replaces
>>   Allow-Events: talk, refer
>>   Content-Type: application/sdp
>>   Content-Disposition: session
>>   Content-Length: 250
>>   Remote-Party-ID: "Mihailova Ludmila"
>> <sip:214 at sip.callwithus.com>;party=calling;screen=yes;privacy=off
>>
>>   v=0
>>   o=- 216896551 216896551 IN IP4 192.168.50.51
>>   s=-
>>   c=IN IP4 77.239.230.202
>>   t=0 0
>>   m=audio 32404 RTP/AVP 0 100 101
>>   a=rtpmap:0 PCMU/8000
>>   a=rtpmap:100 NSE/8000
>>   a=fmtp:100 192-193
>>   a=rtpmap:101 telephone-event/8000
>>   a=fmtp:101 0-15
>>   a=ptime:30
>>
>>   ------------------------------------------------------------------------
>> send 1333 bytes to udp/[38.99.70.232]:5060 at 15:05:34.485226:
>>   ------------------------------------------------------------------------
>>   INVITE sip:*408774957558438 at sip.callwithus.com SIP/2.0
>>   Via: SIP/2.0/UDP 77.239.230.202:5080;rport;branch=z9hG4bK97BHK9BtN6r4K
>>   Max-Forwards: 99
>>   From: "Mihailova Ludmila"
>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>   To: <sip:*408774957558438 at sip.callwithus.com>
>>   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>   CSeq: 121302431 INVITE
>>   Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
>>   Expires: 3600
>>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY
>>   Supported: timer, precondition, path, replaces
>>   Allow-Events: talk, refer
>>   Content-Type: application/sdp
>>   Content-Disposition: session
>>   Content-Length: 250
>>   Remote-Party-ID: "Mihailova Ludmila"
>> <sip:214 at sip.callwithus.com>;party=calling;screen=yes;privacy=off
>>
>>   v=0
>>   o=- 216896551 216896551 IN IP4 192.168.50.51
>>   s=-
>>   c=IN IP4 77.239.230.202
>>   t=0 0
>>   m=audio 32404 RTP/AVP 0 100 101
>>   a=rtpmap:0 PCMU/8000
>>   a=rtpmap:100 NSE/8000
>>   a=fmtp:100 192-193
>>   a=rtpmap:101 telephone-event/8000
>>   a=fmtp:101 0-15
>>   a=ptime:30
>>   ------------------------------------------------------------------------
>>   SIP/2.0 100 Giving a try
>>   Via: SIP/2.0/UDP
>> 192.168.50.10:5080;rport=5080;branch=z9hG4bK97BHK9BtN6r4K
>>   From: "Mihailova Ludmila"
>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>   To: <sip:*408774957558438 at sip.callwithus.com>
>>   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>   CSeq: 121302431 INVITE
>>   Server: CWU SIP GW
>>   Content-Length: 0
>>
>>   ------------------------------------------------------------------------
>> recv 836 bytes from udp/[38.99.70.232]:5060 at 15:05:49.641312:
>>   ------------------------------------------------------------------------
>>   SIP/2.0 183 Session Progress
>>   Via: SIP/2.0/UDP
>> 192.168.50.10:5080;rport=5080;branch=z9hG4bK97BHK9BtN6r4K
>>   Record-Route:
>> <sip:38.99.70.232;lr=on;ftag=HXjrDKpHHX8ZN;did=e8e.089fc2b6>
>>   From: "Mihailova Ludmila"
>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>   To: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>   CSeq: 121302431 INVITE
>>   User-Agent: CWU SIP-GW
>>   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>>   Supported: replaces
>>   Contact: <sip:*408774957558438 at 204.74.213.1:5062>
>>   Content-Type: application/sdp
>>   Content-Length: 209
>>
>>   v=0
>>   o=root 31797 31797 IN IP4 4.55.17.66
>>   s=session
>>   c=IN IP4 4.55.17.66
>>   t=0 0
>>   m=audio 14104 RTP/AVP 0 101
>>   a=rtpmap:0 PCMU/8000
>>   a=rtpmap:101 telephone-event/8000
>>   a=fmtp:101 0-16
>>   a=ptime:20
>>   a=sendrecv
>>   ------------------------------------------------------------------------
>> v=0
>> o=root 31797 31797 IN IP4 4.55.17.66
>> s=session
>> c=IN IP4 4.55.17.66
>> t=0 0
>> m=audio 14104 RTP/AVP 0 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>>
>> 2009-10-06 19:03:31.112988 [NOTICE] sofia.c:3449 Pre-Answer
>> sofia/external/*408774957558438 at sip.callwithus.com!
>> 2009-10-06 19:03:31.112988 [DEBUG] switch_channel.c:1822 Send signal
>> sofia/internal/214 at pbx0.tssec.lan [BREAK]
>> 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2095 Set Codec
>> sofia/external/*408774957558438 at sip.callwithus.com PROXY/8000 20 ms 160
>> samples
>> 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2372 PROXY AUDIO RTP
>> [sofia/external/*408774957558438 at sip.callwithus.com]
>> 192.168.50.10:32404->4.55.17.66:14104 codec: 0 ms: 20
>> 2009-10-06 19:03:31.112988 [DEBUG] switch_rtp.c:1163 Not using a timer
>> 2009-10-06 19:03:31.112988 [DEBUG] switch_ivr_originate.c:2154
>> sofia/internal/214 at pbx0.tssec.lan receive message [PROGRESS]
>> 2009-10-06 19:03:31.112988 [INFO] switch_ivr_originate.c:2154 Sending
>> early media
>> 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:1322
>> sofia/internal/214 at pbx0.tssec.lan Patched SDP
>> ---
>> v=0
>> o=root 31797 31797 IN IP4 4.55.17.66
>> s=session
>> c=IN IP4 4.55.17.66
>> t=0 0
>> m=audio 14104 RTP/AVP 0 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>>
>> +++
>> v=0
>> o=root 31797 31797 IN IP4 4.55.17.66
>> s=session
>> c=IN IP4 192.168.50.10
>> t=0 0
>> m=audio 32698 RTP/AVP 0 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>>
>> 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2372 PROXY AUDIO RTP
>> [sofia/internal/214 at pbx0.tssec.lan]
>> 192.168.50.10:32698->192.168.50.51:16474 codec: 0 ms: 30
>> 2009-10-06 19:03:31.112988 [DEBUG] switch_rtp.c:1163 Not using a timer
>> 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2508 Set comfort noise
>> payload to 13
>> 2009-10-06 19:03:31.112988 [DEBUG] sofia.c:3462
>> sofia/internal/214 at pbx0.tssec.lan receive message [PROGRESS]
>> 2009-10-06 19:03:31.116989 [NOTICE] mod_sofia.c:1521 Pre-Answer
>> sofia/internal/214 at pbx0.tssec.lan!
>> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal
>> sofia/internal/214 at pbx0.tssec.lan [BREAK]
>> 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_originate.c:2196 Originate
>> Resulted in Success: [sofia/external/*408774957558438 at sip.callwithus.com]
>> 2009-10-06 19:03:31.116989 [DEBUG] switch_channel.c:182
>> sofia/external/*408774957558438 at sip.callwithus.com receive message
>> [AUDIO_SYNC]
>> 2009-10-06 19:03:31.116989 [DEBUG] switch_channel.c:182
>> sofia/internal/214 at pbx0.tssec.lan receive message [AUDIO_SYNC]
>> 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_bridge.c:911
>> sofia/external/*408774957558438 at sip.callwithus.com receive message [BRIDGE]
>> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal
>> sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
>> 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_bridge.c:918
>> sofia/internal/214 at pbx0.tssec.lan receive message [BRIDGE]
>> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal
>> sofia/internal/214 at pbx0.tssec.lan [BREAK]
>> 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_bridge.c:962
>> (sofia/external/*408774957558438 at sip.callwithus.com) State Change
>> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
>> 2009-10-06 19:03:31.116989 [DEBUG] sofia.c:3359 Channel
>> sofia/internal/214 at pbx0.tssec.lan entering state [early][183]
>> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal
>> sofia/internal/214 at pbx0.tssec.lan [BREAK]
>> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
>> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/*408774957558438 at sip.callwithus.com) Running State
>> Change CS_EXCHANGE_MEDIA
>> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_state_machine.c:343
>> (sofia/external/*408774957558438 at sip.callwithus.com) State EXCHANGE_MEDIA
>> 2009-10-06 19:03:31.116989 [DEBUG] mod_sofia.c:430 SOFIA LOOPBACK
>> recv 822 bytes from udp/[38.99.70.232]:5060 at 15:05:58.503701:
>>   ------------------------------------------------------------------------
>>   SIP/2.0 200 OK
>>   Via: SIP/2.0/UDP
>> 192.168.50.10:5080;rport=5080;branch=z9hG4bK97BHK9BtN6r4K
>>   Record-Route:
>> <sip:38.99.70.232;lr=on;ftag=HXjrDKpHHX8ZN;did=e8e.089fc2b6>
>>   From: "Mihailova Ludmila"
>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>   To: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>   CSeq: 121302431 INVITE
>>   User-Agent: CWU SIP-GW
>>   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>>   Supported: replaces
>>   Contact: <sip:*408774957558438 at 204.74.213.1:5062>
>>   Content-Type: application/sdp
>>   Content-Length: 209
>>
>>   v=0
>>   o=root 31797 31798 IN IP4 4.55.17.66
>>   s=session
>>   c=IN IP4 4.55.17.66
>>   t=0 0
>>   m=audio 14104 RTP/AVP 0 101
>>   a=rtpmap:0 PCMU/8000
>>   a=rtpmap:101 telephone-event/8000
>>   a=fmtp:101 0-16
>>   a=ptime:20
>>   a=sendrecv
>>   ------------------------------------------------------------------------
>> send 762 bytes to udp/[38.99.70.232]:5060 at 15:05:58.507702:
>>   ------------------------------------------------------------------------
>>   ACK sip:*408774957558438 at 204.74.213.1:5062 SIP/2.0
>>   Via: SIP/2.0/UDP 192.168.50.10:5080;rport;branch=z9hG4bKaH59m4vXjFFQF
>>   Route: <sip:38.99.70.232;lr=on;ftag=HXjrDKpHHX8ZN;did=e8e.089fc2b6>
>>   Max-Forwards: 70
>>   From: "Mihailova Ludmila"
>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>   To: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>   CSeq: 121302431 ACK
>>   Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
>>   Content-Length: 0
>>
>>   ------------------------------------------------------------------------
>> 2009-10-06 19:03:39.975377 [DEBUG] sofia.c:3359 Channel
>> sofia/external/*408774957558438 at sip.callwithus.com entering state
>> [ready][200]
>> 2009-10-06 19:03:39.975377 [DEBUG] switch_channel.c:1935 Send signal
>> sofia/internal/214 at pbx0.tssec.lan [BREAK]
>> 2009-10-06 19:03:39.975377 [NOTICE] sofia.c:3814 Channel
>> [sofia/external/*408774957558438 at sip.callwithus.com] has been answered
>> 2009-10-06 19:03:39.975377 [DEBUG] switch_channel.c:182
>> sofia/external/*408774957558438 at sip.callwithus.com receive message
>> [AUDIO_SYNC]
>> 2009-10-06 19:03:39.975377 [DEBUG] sofia.c:3819
>> sofia/internal/214 at pbx0.tssec.lan receive message [ANSWER]
>> 2009-10-06 19:03:39.975377 [DEBUG] sofia_glue.c:1322
>> sofia/internal/214 at pbx0.tssec.lan Patched SDP
>> ---
>> v=0
>> o=root 31797 31797 IN IP4 4.55.17.66
>> s=session
>> c=IN IP4 4.55.17.66
>> t=0 0
>> m=audio 14104 RTP/AVP 0 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>>
>> +++
>> v=0
>> o=root 31797 31797 IN IP4 4.55.17.66
>> s=session
>> c=IN IP4 192.168.50.10
>> t=0 0
>> m=audio 32698 RTP/AVP 0 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>>
>> 2009-10-06 19:03:39.975377 [DEBUG] switch_core_session.c:630 Send signal
>> sofia/internal/214 at pbx0.tssec.lan [BREAK]
>> 2009-10-06 19:03:39.975377 [NOTICE] sofia.c:3819 Channel
>> [sofia/internal/214 at pbx0.tssec.lan] has been answered
>> 2009-10-06 19:03:39.975377 [DEBUG] switch_channel.c:182
>> sofia/internal/214 at pbx0.tssec.lan receive message [AUDIO_SYNC]
>> 2009-10-06 19:03:39.975377 [DEBUG] sofia.c:3359 Channel
>> sofia/internal/214 at pbx0.tssec.lan entering state [completed][200]
>> 2009-10-06 19:03:39.991381 [DEBUG] sofia.c:3359 Channel
>> sofia/internal/214 at pbx0.tssec.lan entering state [ready][200]
>> recv 1054 bytes from udp/[38.99.70.232]:5060 at 15:06:05.577607:
>>   ------------------------------------------------------------------------
>>   INVITE sip:gw+callwithus at 192.168.50.10:5080;transport=udp SIP/2.0
>>   Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.0
>>   Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK41d48157;rport=5062
>>   From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>   To: "Mihailova Ludmila"
>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>   Contact: <sip:*408774957558438 at 204.74.213.1:5062>
>>   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>   CSeq: 102 INVITE
>>   User-Agent: CWU SIP-GW
>>   Max-Forwards: 69
>>   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>>   Supported: replaces
>>   X-asterisk-info: SIP re-invite (T38 switchover)
>>   Content-Type: application/sdp
>>   Content-Length: 350
>>
>>   v=0
>>   o=root 31797 31799 IN IP4 32.50.63.192
>>   s=session
>>   c=IN IP4 32.50.63.192
>>   t=0 0
>>   m=image 14104 udptl t38
>>   a=T38FaxVersion:0
>>   a=T38MaxBitRate:9600
>>   a=T38FaxFillBitRemoval:0
>>   a=T38FaxTranscodingMMR:0
>>   a=T38FaxTranscodingJBIG:0
>>   a=T38FaxRateManagement:transferredTCF
>>   a=T38FaxMaxBuffer:176
>>   a=T38FaxMaxDatagram:176
>>   a=T38FaxUdpEC:t38UDPRedundancy
>>   ------------------------------------------------------------------------
>> send 443 bytes to udp/[38.99.70.232]:5060 at 15:06:05.577607:
>>   ------------------------------------------------------------------------
>>   SIP/2.0 100 Trying
>>   Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.0
>>   Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK41d48157;rport=5062
>>   From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>   To: "Mihailova Ludmila"
>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>   CSeq: 102 INVITE
>>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
>>   Content-Length: 0
>>
>>   ------------------------------------------------------------------------
>> 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3359 Channel
>> sofia/external/*408774957558438 at sip.callwithus.com entering state
>> [received][100]
>> 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3366 Remote SDP:
>> v=0
>> o=root 31797 31799 IN IP4 32.50.63.192
>> s=session
>> c=IN IP4 32.50.63.192
>> t=0 0
>> m=image 14104 udptl t38
>> a=T38FaxVersion:0
>> a=T38MaxBitRate:9600
>> a=T38FaxFillBitRemoval:0
>> a=T38FaxTranscodingMMR:0
>> a=T38FaxTranscodingJBIG:0
>> a=T38FaxRateManagement:transferredTCF
>> a=T38FaxMaxBuffer:176
>> a=T38FaxMaxDatagram:176
>> a=T38FaxUdpEC:t38UDPRedundancy
>>
>> 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3683 Passing SDP to other leg.
>> v=0
>> o=root 31797 31799 IN IP4 32.50.63.192
>> s=session
>> c=IN IP4 32.50.63.192
>> t=0 0
>> m=image 14104 udptl t38
>> a=T38FaxVersion:0
>> a=T38MaxBitRate:9600
>> a=T38FaxFillBitRemoval:0
>> a=T38FaxTranscodingMMR:0
>> a=T38FaxTranscodingJBIG:0
>> a=T38FaxRateManagement:transferredTCF
>> a=T38FaxMaxBuffer:176
>> a=T38FaxMaxDatagram:176
>> a=T38FaxUdpEC:t38UDPRedundancy
>>
>> 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3696
>> sofia/internal/214 at pbx0.tssec.lan receive message [MEDIA_REDIRECT]
>> 2009-10-06 19:03:47.049284 [DEBUG] mod_sofia.c:1195
>> sofia/internal/214 at pbx0.tssec.lan Sending media re-direct:
>> v=0
>> o=root 31797 31799 IN IP4 32.50.63.192
>> s=session
>> c=IN IP4 32.50.63.192
>> t=0 0
>> m=image 14104 udptl t38
>> a=T38FaxVersion:0
>> a=T38MaxBitRate:9600
>> a=T38FaxFillBitRemoval:0
>> a=T38FaxTranscodingMMR:0
>> a=T38FaxTranscodingJBIG:0
>> a=T38FaxRateManagement:transferredTCF
>> a=T38FaxMaxBuffer:176
>> a=T38FaxMaxDatagram:176
>> a=T38FaxUdpEC:t38UDPRedundancy
>>
>> 2009-10-06 19:03:47.049284 [DEBUG] sofia_glue.c:1101 Remote address:port
>> [192.168.50.51:16474] has not changed.
>> 2009-10-06 19:03:47.049284 [DEBUG] sofia_glue.c:1322
>> sofia/internal/214 at pbx0.tssec.lan Patched SDP
>> ---
>> v=0
>> o=root 31797 31799 IN IP4 32.50.63.192
>> s=session
>> c=IN IP4 32.50.63.192
>> t=0 0
>> m=image 14104 udptl t38
>> a=T38FaxVersion:0
>> a=T38MaxBitRate:9600
>> a=T38FaxFillBitRemoval:0
>> a=T38FaxTranscodingMMR:0
>> a=T38FaxTranscodingJBIG:0
>> a=T38FaxRateManagement:transferredTCF
>> a=T38FaxMaxBuffer:176
>> a=T38FaxMaxDatagram:176
>> a=T38FaxUdpEC:t38UDPRedundancy
>>
>> +++
>> v=0
>> o=root 31797 31799 IN IP4 32.50.63.192
>> s=session
>> c=IN IP4 192.168.50.10
>> t=0 0
>> m=image 32698 udptl t38
>> a=T38FaxVersion:0
>> a=T38MaxBitRate:9600
>> a=T38FaxFillBitRemoval:0
>> a=T38FaxTranscodingMMR:0
>> a=T38FaxTranscodingJBIG:0
>> a=T38FaxRateManagement:transferredTCF
>> a=T38FaxMaxBuffer:176
>> a=T38FaxMaxDatagram:176
>> a=T38FaxUdpEC:t38UDPRedundancy
>>
>> 2009-10-06 19:03:47.049284 [DEBUG] switch_core_session.c:630 Send signal
>> sofia/internal/214 at pbx0.tssec.lan [BREAK]
>> 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3359 Channel
>> sofia/internal/214 at pbx0.tssec.lan entering state [calling][0]
>> 2009-10-06 19:03:47.077291 [DEBUG] sofia_glue.c:1101 Remote address:port
>> [192.168.50.51:16474] has not changed.
>> 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3158 Passing 200 OK to other leg
>> 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3170
>> sofia/external/*408774957558438 at sip.callwithus.com receive message [RESPOND]
>> 2009-10-06 19:03:47.077291 [DEBUG] mod_sofia.c:1427 Responding with 200 [OK]
>> 2009-10-06 19:03:47.077291 [DEBUG] sofia_glue.c:1322
>> sofia/external/*408774957558438 at sip.callwithus.com Patched SDP
>> ---
>> v=0
>> o=- 216899808 216899808 IN IP4 192.168.50.51
>> s=-
>> c=IN IP4 192.168.50.51
>> t=0 0
>> m=image 16474 udptl t38
>> a=T38FaxVersion:0
>> a=T38MaxBitRate:14400
>> a=T38FaxRateManagement:transferredTCF
>> a=T38FaxMaxBuffer:200
>> a=T38FaxMaxDatagram:200
>> a=T38FaxUdpEC:t38UDPRedundancy
>>
>> +++
>> v=0
>> o=- 216899808 216899808 IN IP4 192.168.50.51
>> s=-
>> c=IN IP4 77.239.230.202
>> t=0 0
>> m=image 32404 udptl t38
>> a=T38FaxVersion:0
>> a=T38MaxBitRate:14400
>> a=T38FaxRateManagement:transferredTCF
>> a=T38FaxMaxBuffer:200
>> a=T38FaxMaxDatagram:200
>> a=T38FaxUdpEC:t38UDPRedundancy
>>
>> 2009-10-06 19:03:47.077291 [DEBUG] sofia_glue.c:1110 AUDIO RTP CHANGING
>> DEST TO: [32.50.63.192:14104]
>> 2009-10-06 19:03:47.077291 [DEBUG] switch_core_session.c:630 Send signal
>> sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
>> 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3359 Channel
>> sofia/internal/214 at pbx0.tssec.lan entering state [ready][200]
>> 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3366 Remote SDP:
>> v=0
>> o=- 216899808 216899808 IN IP4 192.168.50.51
>> s=-
>> c=IN IP4 192.168.50.51
>> t=0 0
>> m=image 16474 udptl t38
>> a=T38FaxVersion:0
>> a=T38MaxBitRate:14400
>> a=T38FaxRateManagement:transferredTCF
>> a=T38FaxMaxBuffer:200
>> a=T38FaxMaxDatagram:200
>> a=T38FaxUdpEC:t38UDPRedundancy
>>
>> send 1003 bytes to udp/[38.99.70.232]:5060 at 15:06:05.605615:
>>   ------------------------------------------------------------------------
>>   SIP/2.0 200 OK
>>   Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.0
>>   Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK41d48157;rport=5062
>>   From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>   To: "Mihailova Ludmila"
>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>   CSeq: 102 INVITE
>>   Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
>>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
>>   Accept: application/sdp
>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY
>>   Supported: timer, precondition, path, replaces
>>   Content-Type: application/sdp
>>   Content-Disposition: session
>>   Content-Length: 274
>>
>>   v=0
>>   o=- 216896551 216896552 IN IP4 192.168.50.51
>>   s=-
>>   c=IN IP4 77.239.230.202
>>   t=0 0
>>   m=image 32404 udptl t38
>>   a=T38FaxVersion:0
>>   a=T38MaxBitRate:14400
>>   a=T38FaxRateManagement:transferredTCF
>>   a=T38FaxMaxBuffer:200
>>   a=T38FaxMaxDatagram:200
>>   a=T38FaxUdpEC:t38UDPRedundancy
>>   ------------------------------------------------------------------------
>> 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3359 Channel
>> sofia/external/*408774957558438 at sip.callwithus.com entering state
>> [completed][200]
>> 2009-10-06 19:03:47.285347 [INFO] switch_rtp.c:1905 Auto Changing port
>> from 32.50.63.192:14104 to 4.55.17.66:14104
>> recv 527 bytes from udp/[38.99.70.232]:5060 at 15:06:05.817672:
>>   ------------------------------------------------------------------------
>>   ACK sip:gw+callwithus at 192.168.50.10:5080;transport=udp SIP/2.0
>>   Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.2
>>   Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK5ffd72cd;rport=5062
>>   From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>   To: "Mihailova Ludmila"
>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>   Contact: <sip:*408774957558438 at 204.74.213.1:5062>
>>   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>   CSeq: 102 ACK
>>   User-Agent: CWU SIP-GW
>>   Max-Forwards: 69
>>   Content-Length: 0
>>
>>   ------------------------------------------------------------------------
>> 2009-10-06 19:03:47.289349 [DEBUG] sofia.c:3359 Channel
>> sofia/external/*408774957558438 at sip.callwithus.com entering state
>> [ready][200]
>> recv 476 bytes from udp/[38.99.70.232]:5060 at 15:06:47.588933:
>>   ------------------------------------------------------------------------
>>   BYE sip:gw+callwithus at 192.168.50.10:5080;transport=udp SIP/2.0
>>   Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKd406.f8553e14.0
>>   Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK0f2cff01;rport=5062
>>   From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>   To: "Mihailova Ludmila"
>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>   CSeq: 103 BYE
>>   User-Agent: CWU SIP-GW
>>   Max-Forwards: 69
>>   Content-Length: 0
>>
>>   ------------------------------------------------------------------------
>> 2009-10-06 19:04:29.060609 [NOTICE] sofia.c:328 Hangup
>> sofia/external/*408774957558438 at sip.callwithus.com [CS_EXCHANGE_MEDIA]
>> [NORMAL_CLEARING]
>> 2009-10-06 19:04:29.060609 [DEBUG] switch_channel.c:1726 Send signal
>> sofia/external/*408774957558438 at sip.callwithus.com [KILL]
>> 2009-10-06 19:04:29.060609 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
>> 2009-10-06 19:04:29.060609 [DEBUG] switch_core_state_machine.c:437
>> thread mismatch skipping state handler.
>> send 574 bytes to udp/[38.99.70.232]:5060 at 15:06:47.588933:
>>   ------------------------------------------------------------------------
>>   SIP/2.0 200 OK
>>   Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKd406.f8553e14.0
>>   Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK0f2cff01;rport=5062
>>   From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>   To: "Mihailova Ludmila"
>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>   Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>   CSeq: 103 BYE
>>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY
>>   Supported: timer, precondition, path, replaces
>>   Content-Length: 0
>>
>>   ------------------------------------------------------------------------
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>     
>
>
>
>   


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From rob4manhere at gmail.com  Wed Oct  7 17:11:54 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Wed, 7 Oct 2009 19:11:54 -0500
Subject: [Freeswitch-users] linksys spa-3102 fax t38 + freeswitch media
	proxy
In-Reply-To: <4ACD076E.2080107@gmail.com>
References: <4ACB9519.9000805@gmail.com>
	<2d9149cd0910071235k27c1429v2bc555c0e7397fc4@mail.gmail.com>
	<4ACD076E.2080107@gmail.com>
Message-ID: <5B417827-6CCB-4EE4-9524-5CD45AC9E28F@gmail.com>

> Can you advise sip-providers offering t38?

Gafachi has T38 fax support.


On Oct 7, 2009, at 4:26 PM, Vladimir Elizarov wrote:

> Kristian Kielhofner ?????:
>> Since no one else has responded I'll chime in with some general  
>> advice.
>>
>> It's troubling to see that your "provider" is using Asterisk to face
>> you (the customer).
>>
>> I've never had any luck getting T.38 to work (at all, in any mode)
>> using Asterisk.  I've heard of other people making it work but
>> consider the chain here:
>>
>> Their provider (they appear to be using DIDx, hmmm) -> Callwithus
>> (using Asterisk) -> You (FreeSWITCH) -> Your device
>>
>> I don't see anything specific in your case but there are many, many
>> places for T.38 to go wrong in this scenario.  I'd recommend
>> attempting T.38 against a local gateway or known Tier 1 provider
>> supporting T.38 with known equipment.
>>
> Can you advise sip-providers offering t38?
>> Otherwise, do you have any updates for us?
>>
> No. I can not add anything.
>
>
>> On Tue, Oct 6, 2009 at 3:06 PM, Vladimir Elizarov
>> <xengelpublicx at gmail.com> wrote:
>>
>>> Hello.
>>>
>>> I want to set up faxing via the gateway linksys spa-3102 (with  
>>> support
>>> for t38) via SIP.
>>>
>>> SIP-client -> linksys spa3102 -> fs -> provider
>>>
>>>   <gateway name="callwithus">
>>>       ....
>>>       <param name="inbound-late-negotiation" value="true"/>
>>>       <param name="stun-enabled" value="true"/>
>>>   </gateway>
>>>
>>> <extension name="Fax">
>>>   <condition field="destination_number" expression="^(30)(\d{11})$">
>>>           <action application="set" data="proxy_media=true"/>
>>>           <action application="set" data="bypass_media=false"/>
>>>           <action application="bridge"
>>> data="sofia/gateway/callwithus/*4087$2 at sip.callwithus.com"/>
>>>       </condition>
>>> </extension>
>>>
>>> I make a call, press start on the fax. Fax not sent. Get this in  
>>> the log
>>> (UA freeswitch not trunk-12805):
>>>
>>>   
>>> ------------------------------------------------------------------------
>>>  INVITE sip:*408774957558438 at sip.callwithus.com SIP/2.0
>>>  Via: SIP/2.0/UDP  
>>> 77.239.230.202:5080;rport;branch=z9hG4bK8yjrHeUprX2Hr
>>>  Max-Forwards: 99
>>>  From: "Mihailova Ludmila"
>>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>>  To: <sip:*408774957558438 at sip.callwithus.com>
>>>  Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>>  CSeq: 121302430 INVITE
>>>  Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
>>>  User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
>>>  Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>> REGISTER, REFER, NOTIFY
>>>  Supported: timer, precondition, path, replaces
>>>  Allow-Events: talk, refer
>>>  Content-Type: application/sdp
>>>  Content-Disposition: session
>>>  Content-Length: 250
>>>  Remote-Party-ID: "Mihailova Ludmila"
>>> <sip:214 at sip.callwithus.com>;party=calling;screen=yes;privacy=off
>>>
>>>  v=0
>>>  o=- 216896551 216896551 IN IP4 192.168.50.51
>>>  s=-
>>>  c=IN IP4 77.239.230.202
>>>  t=0 0
>>>  m=audio 32404 RTP/AVP 0 100 101
>>>  a=rtpmap:0 PCMU/8000
>>>  a=rtpmap:100 NSE/8000
>>>  a=fmtp:100 192-193
>>>  a=rtpmap:101 telephone-event/8000
>>>  a=fmtp:101 0-15
>>>  a=ptime:30
>>>
>>>   
>>> ------------------------------------------------------------------------
>>> send 1333 bytes to udp/[38.99.70.232]:5060 at 15:05:34.485226:
>>>   
>>> ------------------------------------------------------------------------
>>>  INVITE sip:*408774957558438 at sip.callwithus.com SIP/2.0
>>>  Via: SIP/2.0/UDP  
>>> 77.239.230.202:5080;rport;branch=z9hG4bK97BHK9BtN6r4K
>>>  Max-Forwards: 99
>>>  From: "Mihailova Ludmila"
>>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>>  To: <sip:*408774957558438 at sip.callwithus.com>
>>>  Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>>  CSeq: 121302431 INVITE
>>>  Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
>>>  Expires: 3600
>>>  User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
>>>  Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>> REGISTER, REFER, NOTIFY
>>>  Supported: timer, precondition, path, replaces
>>>  Allow-Events: talk, refer
>>>  Content-Type: application/sdp
>>>  Content-Disposition: session
>>>  Content-Length: 250
>>>  Remote-Party-ID: "Mihailova Ludmila"
>>> <sip:214 at sip.callwithus.com>;party=calling;screen=yes;privacy=off
>>>
>>>  v=0
>>>  o=- 216896551 216896551 IN IP4 192.168.50.51
>>>  s=-
>>>  c=IN IP4 77.239.230.202
>>>  t=0 0
>>>  m=audio 32404 RTP/AVP 0 100 101
>>>  a=rtpmap:0 PCMU/8000
>>>  a=rtpmap:100 NSE/8000
>>>  a=fmtp:100 192-193
>>>  a=rtpmap:101 telephone-event/8000
>>>  a=fmtp:101 0-15
>>>  a=ptime:30
>>>   
>>> ------------------------------------------------------------------------
>>>  SIP/2.0 100 Giving a try
>>>  Via: SIP/2.0/UDP
>>> 192.168.50.10:5080;rport=5080;branch=z9hG4bK97BHK9BtN6r4K
>>>  From: "Mihailova Ludmila"
>>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>>  To: <sip:*408774957558438 at sip.callwithus.com>
>>>  Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>>  CSeq: 121302431 INVITE
>>>  Server: CWU SIP GW
>>>  Content-Length: 0
>>>
>>>   
>>> ------------------------------------------------------------------------
>>> recv 836 bytes from udp/[38.99.70.232]:5060 at 15:05:49.641312:
>>>   
>>> ------------------------------------------------------------------------
>>>  SIP/2.0 183 Session Progress
>>>  Via: SIP/2.0/UDP
>>> 192.168.50.10:5080;rport=5080;branch=z9hG4bK97BHK9BtN6r4K
>>>  Record-Route:
>>> <sip:38.99.70.232;lr=on;ftag=HXjrDKpHHX8ZN;did=e8e.089fc2b6>
>>>  From: "Mihailova Ludmila"
>>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>>  To: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>>  Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>>  CSeq: 121302431 INVITE
>>>  User-Agent: CWU SIP-GW
>>>  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>>>  Supported: replaces
>>>  Contact: <sip:*408774957558438 at 204.74.213.1:5062>
>>>  Content-Type: application/sdp
>>>  Content-Length: 209
>>>
>>>  v=0
>>>  o=root 31797 31797 IN IP4 4.55.17.66
>>>  s=session
>>>  c=IN IP4 4.55.17.66
>>>  t=0 0
>>>  m=audio 14104 RTP/AVP 0 101
>>>  a=rtpmap:0 PCMU/8000
>>>  a=rtpmap:101 telephone-event/8000
>>>  a=fmtp:101 0-16
>>>  a=ptime:20
>>>  a=sendrecv
>>>   
>>> ------------------------------------------------------------------------
>>> v=0
>>> o=root 31797 31797 IN IP4 4.55.17.66
>>> s=session
>>> c=IN IP4 4.55.17.66
>>> t=0 0
>>> m=audio 14104 RTP/AVP 0 101
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>>
>>> 2009-10-06 19:03:31.112988 [NOTICE] sofia.c:3449 Pre-Answer
>>> sofia/external/*408774957558438 at sip.callwithus.com!
>>> 2009-10-06 19:03:31.112988 [DEBUG] switch_channel.c:1822 Send signal
>>> sofia/internal/214 at pbx0.tssec.lan [BREAK]
>>> 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2095 Set Codec
>>> sofia/external/*408774957558438 at sip.callwithus.com PROXY/8000 20  
>>> ms 160
>>> samples
>>> 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2372 PROXY AUDIO RTP
>>> [sofia/external/*408774957558438 at sip.callwithus.com]
>>> 192.168.50.10:32404->4.55.17.66:14104 codec: 0 ms: 20
>>> 2009-10-06 19:03:31.112988 [DEBUG] switch_rtp.c:1163 Not using a  
>>> timer
>>> 2009-10-06 19:03:31.112988 [DEBUG] switch_ivr_originate.c:2154
>>> sofia/internal/214 at pbx0.tssec.lan receive message [PROGRESS]
>>> 2009-10-06 19:03:31.112988 [INFO] switch_ivr_originate.c:2154  
>>> Sending
>>> early media
>>> 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:1322
>>> sofia/internal/214 at pbx0.tssec.lan Patched SDP
>>> ---
>>> v=0
>>> o=root 31797 31797 IN IP4 4.55.17.66
>>> s=session
>>> c=IN IP4 4.55.17.66
>>> t=0 0
>>> m=audio 14104 RTP/AVP 0 101
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>>
>>> +++
>>> v=0
>>> o=root 31797 31797 IN IP4 4.55.17.66
>>> s=session
>>> c=IN IP4 192.168.50.10
>>> t=0 0
>>> m=audio 32698 RTP/AVP 0 101
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>>
>>> 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2372 PROXY AUDIO RTP
>>> [sofia/internal/214 at pbx0.tssec.lan]
>>> 192.168.50.10:32698->192.168.50.51:16474 codec: 0 ms: 30
>>> 2009-10-06 19:03:31.112988 [DEBUG] switch_rtp.c:1163 Not using a  
>>> timer
>>> 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2508 Set comfort  
>>> noise
>>> payload to 13
>>> 2009-10-06 19:03:31.112988 [DEBUG] sofia.c:3462
>>> sofia/internal/214 at pbx0.tssec.lan receive message [PROGRESS]
>>> 2009-10-06 19:03:31.116989 [NOTICE] mod_sofia.c:1521 Pre-Answer
>>> sofia/internal/214 at pbx0.tssec.lan!
>>> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send  
>>> signal
>>> sofia/internal/214 at pbx0.tssec.lan [BREAK]
>>> 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_originate.c:2196  
>>> Originate
>>> Resulted in Success: [sofia/external/*408774957558438 at sip.callwithus.com 
>>> ]
>>> 2009-10-06 19:03:31.116989 [DEBUG] switch_channel.c:182
>>> sofia/external/*408774957558438 at sip.callwithus.com receive message
>>> [AUDIO_SYNC]
>>> 2009-10-06 19:03:31.116989 [DEBUG] switch_channel.c:182
>>> sofia/internal/214 at pbx0.tssec.lan receive message [AUDIO_SYNC]
>>> 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_bridge.c:911
>>> sofia/external/*408774957558438 at sip.callwithus.com receive message  
>>> [BRIDGE]
>>> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send  
>>> signal
>>> sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
>>> 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_bridge.c:918
>>> sofia/internal/214 at pbx0.tssec.lan receive message [BRIDGE]
>>> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send  
>>> signal
>>> sofia/internal/214 at pbx0.tssec.lan [BREAK]
>>> 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_bridge.c:962
>>> (sofia/external/*408774957558438 at sip.callwithus.com) State Change
>>> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
>>> 2009-10-06 19:03:31.116989 [DEBUG] sofia.c:3359 Channel
>>> sofia/internal/214 at pbx0.tssec.lan entering state [early][183]
>>> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send  
>>> signal
>>> sofia/internal/214 at pbx0.tssec.lan [BREAK]
>>> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:932 Send  
>>> signal
>>> sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
>>> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/*408774957558438 at sip.callwithus.com) Running State
>>> Change CS_EXCHANGE_MEDIA
>>> 2009-10-06 19:03:31.116989 [DEBUG] switch_core_state_machine.c:343
>>> (sofia/external/*408774957558438 at sip.callwithus.com) State  
>>> EXCHANGE_MEDIA
>>> 2009-10-06 19:03:31.116989 [DEBUG] mod_sofia.c:430 SOFIA LOOPBACK
>>> recv 822 bytes from udp/[38.99.70.232]:5060 at 15:05:58.503701:
>>>   
>>> ------------------------------------------------------------------------
>>>  SIP/2.0 200 OK
>>>  Via: SIP/2.0/UDP
>>> 192.168.50.10:5080;rport=5080;branch=z9hG4bK97BHK9BtN6r4K
>>>  Record-Route:
>>> <sip:38.99.70.232;lr=on;ftag=HXjrDKpHHX8ZN;did=e8e.089fc2b6>
>>>  From: "Mihailova Ludmila"
>>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>>  To: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>>  Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>>  CSeq: 121302431 INVITE
>>>  User-Agent: CWU SIP-GW
>>>  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>>>  Supported: replaces
>>>  Contact: <sip:*408774957558438 at 204.74.213.1:5062>
>>>  Content-Type: application/sdp
>>>  Content-Length: 209
>>>
>>>  v=0
>>>  o=root 31797 31798 IN IP4 4.55.17.66
>>>  s=session
>>>  c=IN IP4 4.55.17.66
>>>  t=0 0
>>>  m=audio 14104 RTP/AVP 0 101
>>>  a=rtpmap:0 PCMU/8000
>>>  a=rtpmap:101 telephone-event/8000
>>>  a=fmtp:101 0-16
>>>  a=ptime:20
>>>  a=sendrecv
>>>   
>>> ------------------------------------------------------------------------
>>> send 762 bytes to udp/[38.99.70.232]:5060 at 15:05:58.507702:
>>>   
>>> ------------------------------------------------------------------------
>>>  ACK sip:*408774957558438 at 204.74.213.1:5062 SIP/2.0
>>>  Via: SIP/2.0/UDP  
>>> 192.168.50.10:5080;rport;branch=z9hG4bKaH59m4vXjFFQF
>>>  Route: <sip:38.99.70.232;lr=on;ftag=HXjrDKpHHX8ZN;did=e8e.089fc2b6>
>>>  Max-Forwards: 70
>>>  From: "Mihailova Ludmila"
>>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>>  To: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>>  Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>>  CSeq: 121302431 ACK
>>>  Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
>>>  Content-Length: 0
>>>
>>>   
>>> ------------------------------------------------------------------------
>>> 2009-10-06 19:03:39.975377 [DEBUG] sofia.c:3359 Channel
>>> sofia/external/*408774957558438 at sip.callwithus.com entering state
>>> [ready][200]
>>> 2009-10-06 19:03:39.975377 [DEBUG] switch_channel.c:1935 Send signal
>>> sofia/internal/214 at pbx0.tssec.lan [BREAK]
>>> 2009-10-06 19:03:39.975377 [NOTICE] sofia.c:3814 Channel
>>> [sofia/external/*408774957558438 at sip.callwithus.com] has been  
>>> answered
>>> 2009-10-06 19:03:39.975377 [DEBUG] switch_channel.c:182
>>> sofia/external/*408774957558438 at sip.callwithus.com receive message
>>> [AUDIO_SYNC]
>>> 2009-10-06 19:03:39.975377 [DEBUG] sofia.c:3819
>>> sofia/internal/214 at pbx0.tssec.lan receive message [ANSWER]
>>> 2009-10-06 19:03:39.975377 [DEBUG] sofia_glue.c:1322
>>> sofia/internal/214 at pbx0.tssec.lan Patched SDP
>>> ---
>>> v=0
>>> o=root 31797 31797 IN IP4 4.55.17.66
>>> s=session
>>> c=IN IP4 4.55.17.66
>>> t=0 0
>>> m=audio 14104 RTP/AVP 0 101
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>>
>>> +++
>>> v=0
>>> o=root 31797 31797 IN IP4 4.55.17.66
>>> s=session
>>> c=IN IP4 192.168.50.10
>>> t=0 0
>>> m=audio 32698 RTP/AVP 0 101
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>>
>>> 2009-10-06 19:03:39.975377 [DEBUG] switch_core_session.c:630 Send  
>>> signal
>>> sofia/internal/214 at pbx0.tssec.lan [BREAK]
>>> 2009-10-06 19:03:39.975377 [NOTICE] sofia.c:3819 Channel
>>> [sofia/internal/214 at pbx0.tssec.lan] has been answered
>>> 2009-10-06 19:03:39.975377 [DEBUG] switch_channel.c:182
>>> sofia/internal/214 at pbx0.tssec.lan receive message [AUDIO_SYNC]
>>> 2009-10-06 19:03:39.975377 [DEBUG] sofia.c:3359 Channel
>>> sofia/internal/214 at pbx0.tssec.lan entering state [completed][200]
>>> 2009-10-06 19:03:39.991381 [DEBUG] sofia.c:3359 Channel
>>> sofia/internal/214 at pbx0.tssec.lan entering state [ready][200]
>>> recv 1054 bytes from udp/[38.99.70.232]:5060 at 15:06:05.577607:
>>>   
>>> ------------------------------------------------------------------------
>>>  INVITE sip:gw+callwithus at 192.168.50.10:5080;transport=udp SIP/2.0
>>>  Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.0
>>>  Via: SIP/2.0/UDP  
>>> 204.74.213.1:5062;branch=z9hG4bK41d48157;rport=5062
>>>  From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>>  To: "Mihailova Ludmila"
>>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>>  Contact: <sip:*408774957558438 at 204.74.213.1:5062>
>>>  Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>>  CSeq: 102 INVITE
>>>  User-Agent: CWU SIP-GW
>>>  Max-Forwards: 69
>>>  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>>>  Supported: replaces
>>>  X-asterisk-info: SIP re-invite (T38 switchover)
>>>  Content-Type: application/sdp
>>>  Content-Length: 350
>>>
>>>  v=0
>>>  o=root 31797 31799 IN IP4 32.50.63.192
>>>  s=session
>>>  c=IN IP4 32.50.63.192
>>>  t=0 0
>>>  m=image 14104 udptl t38
>>>  a=T38FaxVersion:0
>>>  a=T38MaxBitRate:9600
>>>  a=T38FaxFillBitRemoval:0
>>>  a=T38FaxTranscodingMMR:0
>>>  a=T38FaxTranscodingJBIG:0
>>>  a=T38FaxRateManagement:transferredTCF
>>>  a=T38FaxMaxBuffer:176
>>>  a=T38FaxMaxDatagram:176
>>>  a=T38FaxUdpEC:t38UDPRedundancy
>>>   
>>> ------------------------------------------------------------------------
>>> send 443 bytes to udp/[38.99.70.232]:5060 at 15:06:05.577607:
>>>   
>>> ------------------------------------------------------------------------
>>>  SIP/2.0 100 Trying
>>>  Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.0
>>>  Via: SIP/2.0/UDP  
>>> 204.74.213.1:5062;branch=z9hG4bK41d48157;rport=5062
>>>  From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>>  To: "Mihailova Ludmila"
>>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>>  Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>>  CSeq: 102 INVITE
>>>  User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
>>>  Content-Length: 0
>>>
>>>   
>>> ------------------------------------------------------------------------
>>> 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3359 Channel
>>> sofia/external/*408774957558438 at sip.callwithus.com entering state
>>> [received][100]
>>> 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3366 Remote SDP:
>>> v=0
>>> o=root 31797 31799 IN IP4 32.50.63.192
>>> s=session
>>> c=IN IP4 32.50.63.192
>>> t=0 0
>>> m=image 14104 udptl t38
>>> a=T38FaxVersion:0
>>> a=T38MaxBitRate:9600
>>> a=T38FaxFillBitRemoval:0
>>> a=T38FaxTranscodingMMR:0
>>> a=T38FaxTranscodingJBIG:0
>>> a=T38FaxRateManagement:transferredTCF
>>> a=T38FaxMaxBuffer:176
>>> a=T38FaxMaxDatagram:176
>>> a=T38FaxUdpEC:t38UDPRedundancy
>>>
>>> 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3683 Passing SDP to  
>>> other leg.
>>> v=0
>>> o=root 31797 31799 IN IP4 32.50.63.192
>>> s=session
>>> c=IN IP4 32.50.63.192
>>> t=0 0
>>> m=image 14104 udptl t38
>>> a=T38FaxVersion:0
>>> a=T38MaxBitRate:9600
>>> a=T38FaxFillBitRemoval:0
>>> a=T38FaxTranscodingMMR:0
>>> a=T38FaxTranscodingJBIG:0
>>> a=T38FaxRateManagement:transferredTCF
>>> a=T38FaxMaxBuffer:176
>>> a=T38FaxMaxDatagram:176
>>> a=T38FaxUdpEC:t38UDPRedundancy
>>>
>>> 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3696
>>> sofia/internal/214 at pbx0.tssec.lan receive message [MEDIA_REDIRECT]
>>> 2009-10-06 19:03:47.049284 [DEBUG] mod_sofia.c:1195
>>> sofia/internal/214 at pbx0.tssec.lan Sending media re-direct:
>>> v=0
>>> o=root 31797 31799 IN IP4 32.50.63.192
>>> s=session
>>> c=IN IP4 32.50.63.192
>>> t=0 0
>>> m=image 14104 udptl t38
>>> a=T38FaxVersion:0
>>> a=T38MaxBitRate:9600
>>> a=T38FaxFillBitRemoval:0
>>> a=T38FaxTranscodingMMR:0
>>> a=T38FaxTranscodingJBIG:0
>>> a=T38FaxRateManagement:transferredTCF
>>> a=T38FaxMaxBuffer:176
>>> a=T38FaxMaxDatagram:176
>>> a=T38FaxUdpEC:t38UDPRedundancy
>>>
>>> 2009-10-06 19:03:47.049284 [DEBUG] sofia_glue.c:1101 Remote  
>>> address:port
>>> [192.168.50.51:16474] has not changed.
>>> 2009-10-06 19:03:47.049284 [DEBUG] sofia_glue.c:1322
>>> sofia/internal/214 at pbx0.tssec.lan Patched SDP
>>> ---
>>> v=0
>>> o=root 31797 31799 IN IP4 32.50.63.192
>>> s=session
>>> c=IN IP4 32.50.63.192
>>> t=0 0
>>> m=image 14104 udptl t38
>>> a=T38FaxVersion:0
>>> a=T38MaxBitRate:9600
>>> a=T38FaxFillBitRemoval:0
>>> a=T38FaxTranscodingMMR:0
>>> a=T38FaxTranscodingJBIG:0
>>> a=T38FaxRateManagement:transferredTCF
>>> a=T38FaxMaxBuffer:176
>>> a=T38FaxMaxDatagram:176
>>> a=T38FaxUdpEC:t38UDPRedundancy
>>>
>>> +++
>>> v=0
>>> o=root 31797 31799 IN IP4 32.50.63.192
>>> s=session
>>> c=IN IP4 192.168.50.10
>>> t=0 0
>>> m=image 32698 udptl t38
>>> a=T38FaxVersion:0
>>> a=T38MaxBitRate:9600
>>> a=T38FaxFillBitRemoval:0
>>> a=T38FaxTranscodingMMR:0
>>> a=T38FaxTranscodingJBIG:0
>>> a=T38FaxRateManagement:transferredTCF
>>> a=T38FaxMaxBuffer:176
>>> a=T38FaxMaxDatagram:176
>>> a=T38FaxUdpEC:t38UDPRedundancy
>>>
>>> 2009-10-06 19:03:47.049284 [DEBUG] switch_core_session.c:630 Send  
>>> signal
>>> sofia/internal/214 at pbx0.tssec.lan [BREAK]
>>> 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3359 Channel
>>> sofia/internal/214 at pbx0.tssec.lan entering state [calling][0]
>>> 2009-10-06 19:03:47.077291 [DEBUG] sofia_glue.c:1101 Remote  
>>> address:port
>>> [192.168.50.51:16474] has not changed.
>>> 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3158 Passing 200 OK to  
>>> other leg
>>> 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3170
>>> sofia/external/*408774957558438 at sip.callwithus.com receive message  
>>> [RESPOND]
>>> 2009-10-06 19:03:47.077291 [DEBUG] mod_sofia.c:1427 Responding  
>>> with 200 [OK]
>>> 2009-10-06 19:03:47.077291 [DEBUG] sofia_glue.c:1322
>>> sofia/external/*408774957558438 at sip.callwithus.com Patched SDP
>>> ---
>>> v=0
>>> o=- 216899808 216899808 IN IP4 192.168.50.51
>>> s=-
>>> c=IN IP4 192.168.50.51
>>> t=0 0
>>> m=image 16474 udptl t38
>>> a=T38FaxVersion:0
>>> a=T38MaxBitRate:14400
>>> a=T38FaxRateManagement:transferredTCF
>>> a=T38FaxMaxBuffer:200
>>> a=T38FaxMaxDatagram:200
>>> a=T38FaxUdpEC:t38UDPRedundancy
>>>
>>> +++
>>> v=0
>>> o=- 216899808 216899808 IN IP4 192.168.50.51
>>> s=-
>>> c=IN IP4 77.239.230.202
>>> t=0 0
>>> m=image 32404 udptl t38
>>> a=T38FaxVersion:0
>>> a=T38MaxBitRate:14400
>>> a=T38FaxRateManagement:transferredTCF
>>> a=T38FaxMaxBuffer:200
>>> a=T38FaxMaxDatagram:200
>>> a=T38FaxUdpEC:t38UDPRedundancy
>>>
>>> 2009-10-06 19:03:47.077291 [DEBUG] sofia_glue.c:1110 AUDIO RTP  
>>> CHANGING
>>> DEST TO: [32.50.63.192:14104]
>>> 2009-10-06 19:03:47.077291 [DEBUG] switch_core_session.c:630 Send  
>>> signal
>>> sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
>>> 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3359 Channel
>>> sofia/internal/214 at pbx0.tssec.lan entering state [ready][200]
>>> 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3366 Remote SDP:
>>> v=0
>>> o=- 216899808 216899808 IN IP4 192.168.50.51
>>> s=-
>>> c=IN IP4 192.168.50.51
>>> t=0 0
>>> m=image 16474 udptl t38
>>> a=T38FaxVersion:0
>>> a=T38MaxBitRate:14400
>>> a=T38FaxRateManagement:transferredTCF
>>> a=T38FaxMaxBuffer:200
>>> a=T38FaxMaxDatagram:200
>>> a=T38FaxUdpEC:t38UDPRedundancy
>>>
>>> send 1003 bytes to udp/[38.99.70.232]:5060 at 15:06:05.605615:
>>>   
>>> ------------------------------------------------------------------------
>>>  SIP/2.0 200 OK
>>>  Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.0
>>>  Via: SIP/2.0/UDP  
>>> 204.74.213.1:5062;branch=z9hG4bK41d48157;rport=5062
>>>  From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>>  To: "Mihailova Ludmila"
>>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>>  Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>>  CSeq: 102 INVITE
>>>  Contact: <sip:gw+callwithus at 77.239.230.202:5080;transport=udp>
>>>  User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
>>>  Accept: application/sdp
>>>  Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>> REGISTER, REFER, NOTIFY
>>>  Supported: timer, precondition, path, replaces
>>>  Content-Type: application/sdp
>>>  Content-Disposition: session
>>>  Content-Length: 274
>>>
>>>  v=0
>>>  o=- 216896551 216896552 IN IP4 192.168.50.51
>>>  s=-
>>>  c=IN IP4 77.239.230.202
>>>  t=0 0
>>>  m=image 32404 udptl t38
>>>  a=T38FaxVersion:0
>>>  a=T38MaxBitRate:14400
>>>  a=T38FaxRateManagement:transferredTCF
>>>  a=T38FaxMaxBuffer:200
>>>  a=T38FaxMaxDatagram:200
>>>  a=T38FaxUdpEC:t38UDPRedundancy
>>>   
>>> ------------------------------------------------------------------------
>>> 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3359 Channel
>>> sofia/external/*408774957558438 at sip.callwithus.com entering state
>>> [completed][200]
>>> 2009-10-06 19:03:47.285347 [INFO] switch_rtp.c:1905 Auto Changing  
>>> port
>>> from 32.50.63.192:14104 to 4.55.17.66:14104
>>> recv 527 bytes from udp/[38.99.70.232]:5060 at 15:06:05.817672:
>>>   
>>> ------------------------------------------------------------------------
>>>  ACK sip:gw+callwithus at 192.168.50.10:5080;transport=udp SIP/2.0
>>>  Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.2
>>>  Via: SIP/2.0/UDP  
>>> 204.74.213.1:5062;branch=z9hG4bK5ffd72cd;rport=5062
>>>  From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>>  To: "Mihailova Ludmila"
>>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>>  Contact: <sip:*408774957558438 at 204.74.213.1:5062>
>>>  Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>>  CSeq: 102 ACK
>>>  User-Agent: CWU SIP-GW
>>>  Max-Forwards: 69
>>>  Content-Length: 0
>>>
>>>   
>>> ------------------------------------------------------------------------
>>> 2009-10-06 19:03:47.289349 [DEBUG] sofia.c:3359 Channel
>>> sofia/external/*408774957558438 at sip.callwithus.com entering state
>>> [ready][200]
>>> recv 476 bytes from udp/[38.99.70.232]:5060 at 15:06:47.588933:
>>>   
>>> ------------------------------------------------------------------------
>>>  BYE sip:gw+callwithus at 192.168.50.10:5080;transport=udp SIP/2.0
>>>  Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKd406.f8553e14.0
>>>  Via: SIP/2.0/UDP  
>>> 204.74.213.1:5062;branch=z9hG4bK0f2cff01;rport=5062
>>>  From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>>  To: "Mihailova Ludmila"
>>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>>  Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>>  CSeq: 103 BYE
>>>  User-Agent: CWU SIP-GW
>>>  Max-Forwards: 69
>>>  Content-Length: 0
>>>
>>>   
>>> ------------------------------------------------------------------------
>>> 2009-10-06 19:04:29.060609 [NOTICE] sofia.c:328 Hangup
>>> sofia/external/*408774957558438 at sip.callwithus.com  
>>> [CS_EXCHANGE_MEDIA]
>>> [NORMAL_CLEARING]
>>> 2009-10-06 19:04:29.060609 [DEBUG] switch_channel.c:1726 Send signal
>>> sofia/external/*408774957558438 at sip.callwithus.com [KILL]
>>> 2009-10-06 19:04:29.060609 [DEBUG] switch_core_session.c:932 Send  
>>> signal
>>> sofia/external/*408774957558438 at sip.callwithus.com [BREAK]
>>> 2009-10-06 19:04:29.060609 [DEBUG] switch_core_state_machine.c:437
>>> thread mismatch skipping state handler.
>>> send 574 bytes to udp/[38.99.70.232]:5060 at 15:06:47.588933:
>>>   
>>> ------------------------------------------------------------------------
>>>  SIP/2.0 200 OK
>>>  Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKd406.f8553e14.0
>>>  Via: SIP/2.0/UDP  
>>> 204.74.213.1:5062;branch=z9hG4bK0f2cff01;rport=5062
>>>  From: <sip:*408774957558438 at sip.callwithus.com>;tag=as22d3d3b0
>>>  To: "Mihailova Ludmila"
>>> <sip:544620048 at sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN
>>>  Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f
>>>  CSeq: 103 BYE
>>>  User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805
>>>  Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>> REGISTER, REFER, NOTIFY
>>>  Supported: timer, precondition, path, replaces
>>>  Content-Length: 0
>>>
>>>   
>>> ------------------------------------------------------------------------
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>
>>
>>
>>
>
>
> _______________________________________________
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> http://www.freeswitch.org




From Mailings at kh-dev.de  Wed Oct  7 18:46:10 2009
From: Mailings at kh-dev.de (Klaus Hochlehnert)
Date: Thu, 8 Oct 2009 03:46:10 +0200
Subject: [Freeswitch-users] Changing callerid before/after call pickup
In-Reply-To: <5B417827-6CCB-4EE4-9524-5CD45AC9E28F@gmail.com>
References: <4ACB9519.9000805@gmail.com>
	<2d9149cd0910071235k27c1429v2bc555c0e7397fc4@mail.gmail.com>
	<4ACD076E.2080107@gmail.com>
	<5B417827-6CCB-4EE4-9524-5CD45AC9E28F@gmail.com>
Message-ID: <A2F1C0E212BBBA4E8C0E7BE568361F04027F0233024A@srv02.khdev.corp>

Hi,

currently I'm playing around with call pickup and Snom phones.
I'm using the intercept function for that.

My "problem" is now that after the call pickup (which works fine) I don't see the caller id of the original call.
Instead I see the pickup code, e.g. *820

I've tried to change nearly every channel variable (before doing the call pickup), but the phone won't display the caller id.

Does anybody know how to change that?

Thanks, Klaus




From jason at jasonjgw.net  Wed Oct  7 19:05:57 2009
From: jason at jasonjgw.net (Jason White)
Date: Thu, 8 Oct 2009 13:05:57 +1100
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
Message-ID: <20091008020557.GA23391@jdc.jasonjgw.net>

Brian West <brian at freeswitch.org> wrote:
>  From what I have been told h323plus is a based/fork of OpenH323 which  
> OPAL is just a continuation of OpenH323.  

>From a quick search of gmane.org, the situation seems a little more
complicated.

http://article.gmane.org/gmane.comp.telephony.openh323.general/10930/match=h+460+opal




From srinivas.ksvreddy at gmail.com  Wed Oct  7 22:20:41 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Thu, 8 Oct 2009 10:50:41 +0530
Subject: [Freeswitch-users] sofia_reg_handle_register
Message-ID: <f8af5740910072220j384799fcob42ffcfa3dca1f1a@mail.gmail.com>

Hi,

sofia_reg_handle_register, once it got executed, actuallyl where it will
maintain the registred users adn groups data,

Thanks
Srinivasula Reddy K
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From woodydickson at gmail.com  Wed Oct  7 22:45:08 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Thu, 8 Oct 2009 13:45:08 +0800
Subject: [Freeswitch-users] unable to configure Digium TDM400P
In-Reply-To: <87f2f3b90910070956l520b2c51p7f33463128bf68b2@mail.gmail.com>
References: <a44688a0910070340p4209e7b5yec9218b8c4fc692c@mail.gmail.com>
	<87f2f3b90910070956l520b2c51p7f33463128bf68b2@mail.gmail.com>
Message-ID: <a44688a0910072245v7f496385of411552f0098ba90@mail.gmail.com>

Hi Michael

Is the ztcfg output supposed to say something like 2 channels configured?

I have not set up openzap yet because I don't know if the openzap.config is
good or not.

Can you give me any suggestion if the ztcfg output I am getting is proper or
not?

thx,
woody


-MC
>
> On Wed, Oct 7, 2009 at 3:40 AM, Woody Dickson <woodydickson at gmail.com>wrote:
>
>> Hi,
>>
>> I am trying to setup a Digium TDM400P following the instruction on the
>> wiki.
>> It is a 1 fxo and 1 fxs card, so I tried
>> loadzone=in
>> defaultzone=in
>> fxsks=2
>> fxoks=1
>>
>> and
>>
>> loadzone=in
>> defaultzone=in
>> fxsks=1
>> fxoks=2
>>
>> None works.  Does anyone know how it should be configured?
>>
>> Here is what I get by following the wiki.
>>
>> [root at localhost zaptel]# ztcfg -vv
>>
>> Zaptel Version: SVN-branch-1.4-r4629M
>> Echo Canceller: MG2
>> Configuration
>> ======================
>>
>>
>> Channel map:
>>
>> Channel 01: FXS Kewlstart (Default) (Slaves: 01)
>> Channel 02: FXO Kewlstart (Default) (Slaves: 02)
>>
>> 2 channels to configure.
>>
>> Changing signalling on channel 1 from FXO Kewlstart to FXS Kewlstart
>> Changing signalling on channel 2 from FXS Kewlstart to FXO Kewlstart
>>
>>
>> [root at localhost zaptel]# ztcfg -vv
>>
>> Zaptel Version: SVN-branch-1.4-r4629M
>> Echo Canceller: MG2
>> Configuration
>> ======================
>>
>>
>> Channel map:
>>
>> Channel 01: FXO Kewlstart (Default) (Slaves: 01)
>> Channel 02: FXS Kewlstart (Default) (Slaves: 02)
>>
>> 2 channels to configure.
>>
>> Changing signalling on channel 1 from FXS Kewlstart to FXO Kewlstart
>> Changing signalling on channel 2 from FXO Kewlstart to FXS Kewlstart
>>
>>
>> [root at localhost zaptel]# lspci
>> 00:14.0 Ethernet controller: Digium, Inc. TDM400P (rev 11)
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lakindia89 at gmail.com  Thu Oct  8 02:41:08 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Thu, 8 Oct 2009 15:11:08 +0530
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <87f2f3b90910070849t79b7cb90x4150bf1dd8501ca4@mail.gmail.com>
References: <7d79b3930910070334g53cef0c5m63037e264f1a9233@mail.gmail.com>
	<87f2f3b90910070849t79b7cb90x4150bf1dd8501ca4@mail.gmail.com>
Message-ID: <7d79b3930910080241q4eeb374an2ec34642bf450818@mail.gmail.com>

Hi I tried with the following openzap.conf
[span zt PRI_1]
trunk_type => e1
b-channel => 1:1-15,17-31
d-channel => 1:16

openzap.conf.xml
<configuration name="openzap.conf" description="OpenZAP Configuration">
        <settings>
                <param name="debug" value="7"/>
        </settings>
        <libpri_spans>
                <span name="PRI_1">
                        <param name="node" value="cpe"/>
                        <param name="switch" value="euroisdn"/>
                        <param name="dialplan" value="XML"/>
                        <param name="context" value="default"/>
                </span>
        </libpri_spans>
</configuration>

But as soon as I start the freeswitch I got the following error

2009-10-08 15:08:35.496270 [DEBUG] ozmod_libpri.c:885 Event Failure! 1
2009-10-08 15:08:37.601135 [DEBUG] ozmod_libpri.c:817 EVENT [ONHOOK][1][1:1]
STATE [DOWN]
2009-10-08 15:08:37.701176 [DEBUG] ozmod_libpri.c:885 Event Failure! 1
2009-10-08 15:08:39.802004 [DEBUG] ozmod_libpri.c:817 EVENT [DTMF][0][1:1]
STATE [DOWN]
2009-10-08 15:08:39.902034 [DEBUG] ozmod_libpri.c:885 Event Failure! 1
2009-10-08 15:08:42.010983 [DEBUG] ozmod_libpri.c:885 Event Failure! 1
2009-10-08 15:08:44.115874 [DEBUG] ozmod_libpri.c:817 EVENT [ONHOOK][1][1:1]
STATE [DOWN]
2009-10-08 15:08:44.215915 [DEBUG] ozmod_libpri.c:885 Event Failure! 1
2009-10-08 15:08:46.312748 [DEBUG] ozmod_libpri.c:885 Event Failure! 1
2009-10-08 15:08:48.417552 [DEBUG] ozmod_libpri.c:817 EVENT [DTMF][0][1:1]
STATE [DOWN]
2009-10-08 15:08:48.517587 [DEBUG] ozmod_libpri.c:885 Event Failure! 1
2009-10-08 15:08:50.626485 [DEBUG] ozmod_libpri.c:885 Event Failure! 1
2009-10-08 15:08:52.735325 [DEBUG] ozmod_libpri.c:817 EVENT [ONHOOK][1][1:1]
STATE [DOWN]
2009-10-08 15:08:52.835357 [DEBUG] ozmod_libpri.c:885 Event Failure! 1
2009-10-08 15:08:54.944190 [DEBUG] ozmod_libpri.c:885 Event Failure! 1
2009-10-08 15:08:57.049109 [DEBUG] ozmod_libpri.c:817 EVENT [DTMF][0][1:1]
STATE [DOWN]
2009-10-08 15:08:57.149148 [DEBUG] ozmod_libpri.c:885 Event Failure! 1
2009-10-08 15:08:59.253979 [DEBUG] ozmod_libpri.c:885 Event Failure! 1

I also got the below error in the middle

2009-10-08 15:09:48.992613 [CRIT] lpwrap_pri.c:154 span 1 D-WRITE FAIL! []
2009-10-08 15:09:48.992613 [ERR] ozmod_libpri.c:88 Short write: -1/5
(Invalid or incomplete multibyte or wide character)

Please help.

On Wed, Oct 7, 2009 at 9:19 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Oct 7, 2009 at 3:34 AM, lakshmanan ganapathy <lakindia89 at gmail.com
> > wrote:
>
>> Hi,
>> Again I was struck in a problem, Here is the scenario.
>>
>> On incomming call, I just call an event outboud socket. But what happens
>> is, for the first 15 call, it is working fine. But from the 16th call to
>> 30th call, it says the below error.
>>
>> 2009-10-07 15:07:48.201846 [WARNING] ozmod_libpri.c:761 --Failure opening
>> channel 1:17 (ignored)
>> 2009-10-07 15:07:55.381861 [WARNING] ozmod_libpri.c:761 --Failure opening
>> channel 1:18 (ignored)
>> 2009-10-07 15:07:58.569774 [WARNING] ozmod_libpri.c:761 --Failure opening
>> channel 1:19 (ignored)
>> 2009-10-07 15:08:01.37824 [WARNING] ozmod_libpri.c:761 --Failure opening
>> channel 1:20 (ignored)
>> 2009-10-07 15:08:03.129846 [WARNING] ozmod_libpri.c:761 --Failure opening
>> channel 1:21 (ignored)
>> 2009-10-07 15:08:04.825851 [WARNING] ozmod_libpri.c:761 --Failure opening
>> channel 1:22 (ignored)
>> 2009-10-07 15:08:06.289977 [WARNING] ozmod_libpri.c:761 --Failure opening
>> channel 1:23 (ignored)
>> 2009-10-07 15:08:07.761961 [WARNING] ozmod_libpri.c:761 --Failure opening
>> channel 1:24 (ignored)
>> 2009-10-07 15:08:09.737944 [WARNING] ozmod_libpri.c:761 --Failure opening
>> channel 1:25 (ignored)
>> 2009-10-07 15:08:11.462018 [WARNING] ozmod_libpri.c:761 --Failure opening
>> channel 1:26 (ignored)
>> 2009-10-07 15:08:13.566024 [WARNING] ozmod_libpri.c:761 --Failure opening
>> channel 1:27 (ignored)
>> 2009-10-07 15:08:15.430163 [WARNING] ozmod_libpri.c:761 --Failure opening
>> channel 1:28 (ignored)
>> 2009-10-07 15:08:17.446103 [WARNING] ozmod_libpri.c:761 --Failure opening
>> channel 1:29 (ignored)
>> 2009-10-07 15:08:19.430118 [WARNING] ozmod_libpri.c:761 --Failure opening
>> channel 1:30 (ignored)
>> 2009-10-07 15:08:21.358121 [WARNING] ozmod_libpri.c:761 --Failure opening
>> channel 1:31 (ignored)
>>
>>
>> But for the next call, when it is opening channel 1:1, it is executing my
>> dial plans.
>> I don't know why it failed when it is choosing 1:17-1:31. Any one has any
>> idea.
>>
>> Below are my configuration details:
>>
>> openzap.conf
>> [span zt PRI_1]
>> trunk_type => e1
>> b-channel => 1:1-15
>> d-channel=> 1:16
>> b-channel => 1:17-31
>>
>
> Just a hunch, but could you try this in openzap.conf:
> b-channel => 1:1-15,17-31
> d-channel => 1:16
>
> I want to see what happens. I don't have an E1 setup to test with right now
> otherwise I'd do it myself. Please report back. Thanks,
> MC
>
>
>>
>> openzap.conf.xml
>> <configuration name="openzap.conf" description="OpenZAP Configuration">
>>         <settings>
>>                 <param name="debug" value="7"/>
>>         </settings>
>>         <libpri_spans>
>>                 <span name="PRI_1">
>>                         <param name="node" value="cpe"/>
>>                         <param name="switch" value="euroisdn"/>
>>                         <param name="dialplan" value="XML"/>
>>                         <param name="context" value="default"/>
>>                 </span>
>>         </libpri_spans>
>> </configuration>
>>
>> zaptel.conf
>>
>> span=1,1,0,ccs,hdb3
>> bchan=1-15,17-31
>> dchan=16
>>
>> loadzone        = us
>> defaultzone     = us
>>
>> oz list
>>
>> span: 1 (PRI_1)
>> type: isdn
>> chan_count: 47
>> dialplan: XML
>> context: default
>> dial_regex:
>> fail_dial_regex:
>> hold_music:
>> analog_options none
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bottleman at icf.org.ru  Thu Oct  8 02:42:51 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Thu, 8 Oct 2009 13:42:51 +0400 (MSD)
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
Message-ID: <Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>

On 2009-10-07 15:09 -0500, Brian West wrote freeswitch-users at lists.freeswit...:

opal have addition abstraction layer called opalmgr, and it implementation is not so good in 
this case, for example to implemet pre_answer in mod_opal i need patch libopal, because 
there is no way to send progress inicator throuch opalmgr. and there is many another issues like
this in that layer. ftp://srv.icf.org.ru/pub/soft/f/freeswitch/ - there is my work on mod_opal before 
i start moving to h323plus, may be this help somebody there.

BW> From what I have been told h323plus is a based/fork of OpenH323 which  
BW>OPAL is just a continuation of OpenH323.  So why not support the  
BW>developers of OPAL/OpenH323 ?
BW>
BW>/b
BW>
BW>On Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote:
BW>
BW>> We are developing module to handle h323 proto now, we try to use  
BW>> mod_opal and try improve it, but no luck,
BW>> there is many issues in libopal, and finaly we now move to h323plus  
BW>> library.
BW>
BW>
BW>_______________________________________________
BW>FreeSWITCH-users mailing list
BW>FreeSWITCH-users at lists.freeswitch.org
BW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
BW>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
BW>http://www.freeswitch.org
BW>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From vhatz at kinetix.gr  Thu Oct  8 02:47:47 2009
From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI))
Date: Thu, 08 Oct 2009 12:47:47 +0300
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
Message-ID: <4ACDB543.3070906@kinetix.gr>

Brian West wrote:
>  From what I have been told h323plus is a based/fork of OpenH323 which  
> OPAL is just a continuation of OpenH323.  So why not support the  
> developers of OPAL/OpenH323 ?

H323Plus incorporates many if not all new features/additions etc from 
OPAL (for H323 that is) and a few more features of its own. It is not 
just a continuation of the antiquated OpenH323, so, IMHO it is a good 
option for H323 development.

mod_opal is mostly developed for use with H323 (SIP and IAX support is 
already there in FS in different modules), so it doesn't make much 
difference if it uses H323Plus, OPAL or any other H323 lib.

It will make a difference however if the underlying library leads to 
faster mod_opal/mod_h323 development.

So, yes it would be great to support OPAL and its developers, but 
sometimes the end result (and how fast you obtain it) is what matters 
most.

Therefore, if there are people out there who have already built a usable 
and stable mod for H323 (compared to the current state of mod_opal) I 
think we should give it a try, even if it uses a different library.

At least I'm more than willing to test it in lab and production and help 
in bug reports, or even patching if I can.

Regards,
Vlasis.




From juanbackson at gmail.com  Thu Oct  8 03:20:38 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Thu, 8 Oct 2009 18:20:38 +0800
Subject: [Freeswitch-users] directory variables not set correct with the
	latest trunk
Message-ID: <27c25bc40910080320x7b3b380eu933797197e334cb8@mail.gmail.com>

Hi,

My application fails to set the appropriate variables using directory xml
after using the latest trunk as of yesterday.

My curl looks like:



<document type="freeswitch/xml">
  <section name="directory">
    <domain name="192.168.1.102">
      <user id="200002">
        <params>
          <param name="password" value="123456"></param>
          <param name="dial-string"
value="sofia/internal/sip:200002 at 192.168.1.11:29440
;rinstance=0245b2a59ddff837"></param>
        </params>
        <variables>
          <variable name="account_id" value="20"></variable>
          <variable name="vm-code" value="1000"></variable>
        </variables>
      </user>
    </domain>
  </section>
</document>



In the info app, I am not seeing account-id and vm-code anymore.

How to fix that?

juan
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From srinivas.ksvreddy at gmail.com  Thu Oct  8 04:29:03 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Thu, 8 Oct 2009 16:59:03 +0530
Subject: [Freeswitch-users] sofia_Reg_find_url
Message-ID: <f8af5740910080429y42a9bf05wb626d806e826fc8a@mail.gmail.com>

Hi,

can any one please tell me how sofia_reg_fine_url, and
sofia_glue_execute_sql_callback will work?
which format data is storing and retriving.

Thanks
-- 
Srinivasula Reddy K
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From maciej.aniserowicz at gmail.com  Thu Oct  8 06:18:04 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Thu, 8 Oct 2009 06:18:04 -0700 (PDT)
Subject: [Freeswitch-users] gateway FS informs it's client FS about
 users hanged up with a long delay
In-Reply-To: <98F72CCB-88B9-47ED-AED3-B6BA6DE648C0@jerris.com>
References: <1A16501E57CA4727A593226A8C810308@procent>
	<98F72CCB-88B9-47ED-AED3-B6BA6DE648C0@jerris.com>
Message-ID: <1255007884165-3787956.post@n2.nabble.com>


Both of the instances are run on the same machine, i just changed the default
ports they use. Can anything else cause this strange behavior?
MA



Michael Jerris wrote:
> 
> Incorrect NAT configuration so one of the boxes is not actually  
> getting a BYE.
> 
> 
> On Oct 5, 2009, at 3:13 AM, Maciej Aniserowicz wrote:
> 
>> Hi,
>> When I use two FreeSWITCH instances ('internal' and 'external'), all  
>> users register to the 'external' instance which acts as a gateway by  
>> 'internal' instance (which in turn is controlled by my applicaiton  
>> with commands sent by socket).
>> When user hangs up, the 'hanged up' event is propagated to the  
>> 'internal' instance after a long time (~3 minutes) instead of being  
>> propagated immediately.
>> What can cause this issue?
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://n2.nabble.com/gateway-FS-informs-it-s-client-FS-about-users-hanged-up-with-a-long-delay-tp3768540p3787956.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From d.kochmashev at enforta.com  Wed Oct  7 20:50:13 2009
From: d.kochmashev at enforta.com (Denis Kochmashev "Enforta")
Date: Thu, 8 Oct 2009 09:50:13 +0600
Subject: [Freeswitch-users] [Spam] Re: mod_opal - call charged before
	H.225 connect
In-Reply-To: <20091008020557.GA23391@jdc.jasonjgw.net>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com><65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com><191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com><65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com><86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com><65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com><Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru><82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<20091008020557.GA23391@jdc.jasonjgw.net>
Message-ID: <C723CF32A20B4B61939DB5CD38DB6A06@kochmashev>

It's better to read this -
http://lists.packetizer.com/pipermail/h323plus/2007-October/000020.html 

> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On 
> Behalf Of Jason White
> Sent: Thursday, October 08, 2009 8:06 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Spam] Re: [Freeswitch-users] mod_opal - call 
> charged before H.225 connect
> 
> Brian West <brian at freeswitch.org> wrote:
> >  From what I have been told h323plus is a based/fork of 
> OpenH323 which  
> > OPAL is just a continuation of OpenH323.  
> 
> >From a quick search of gmane.org, the situation seems a little more
> complicated.
> 
> http://article.gmane.org/gmane.comp.telephony.openh323.general
> /10930/match=h+460+opal
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
> itch-users
> http://www.freeswitch.org




From maciej.aniserowicz at gmail.com  Thu Oct  8 06:19:31 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Thu, 8 Oct 2009 06:19:31 -0700 (PDT)
Subject: [Freeswitch-users] Recording creates a 388-byte long file and
 deletes it
In-Reply-To: <74D80E7B-119B-4CC1-82DC-9A370F940BA5@jerris.com>
References: <4ED3AB65AFE34242AACDE97127FE1248@procent>
	<74D80E7B-119B-4CC1-82DC-9A370F940BA5@jerris.com>
Message-ID: <1255007971021-3787968.post@n2.nabble.com>


Yes, I know that FS deletes short files. I just don't know why the file is so
small... it is always 388 bytes, no matter how long the session lasts.
MA



Michael Jerris wrote:
> 
> switch_ivr_async.c:480
> 
> On Oct 5, 2009, at 3:16 AM, Maciej Aniserowicz wrote:
> 
>> Hi,
>> When I record a call in FS, it only creates a 388-byte-long wav  
>> file. The conversation is no written there, and FS deletes the file  
>> when the session finishes.
>> What can cause this strange behavior?
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://n2.nabble.com/Recording-creates-a-388-byte-long-file-and-deletes-it-tp3768541p3787968.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From maciej.aniserowicz at gmail.com  Thu Oct  8 06:27:07 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Thu, 8 Oct 2009 06:27:07 -0700 (PDT)
Subject: [Freeswitch-users] Bad sound quality while eavesdropping
In-Reply-To: <8437F5BC-7AFF-4A74-B8CD-C5B8219021F6@jerris.com>
References: <41A44DD027064988A914974405788C2E@procent>
	<191c3a030910050731m2d74979ep4598e5a1945d58ae@mail.gmail.com>
	<1254901192035-3780245.post@n2.nabble.com>
	<8437F5BC-7AFF-4A74-B8CD-C5B8219021F6@jerris.com>
Message-ID: <1255008427639-3788019.post@n2.nabble.com>


It's the same on the trunk (the last rev I used was not so old anyway).

Codecs are the same on both legs:
read codec/read rate: PCMU	8000
write codec/write rate: PCMU	8000	

MA




Michael Jerris wrote:
> 
> What codecs are all the call legs using, also, please try current svn  
> trunk.
> 
> Mike
> 
> On Oct 7, 2009, at 3:39 AM, Maciej Aniserowicz wrote:
> 
>>
>> Sorry about posting several questions at once, I wasn't aware it's  
>> "rude".
>> Let's concentrate on this issue then.
>>
>> I use FS rev 14994. Phones on extensions:
>> 1) x-lite
>> 2) cisco sip phone
>> 3) audio played by fs to the extension being eavesdropped
>>
>> I did not change any codec configuration, I just use the standard  
>> one that
>> comes with both FS and the phones.
>> Some time ago someone on FS irc channel told me that this is just  
>> how FS
>> eavesdropping works... from your response I understand that this is  
>> not
>> entirely true?
>>
>> Maciej Aniserowicz
>>
>>
>>
>> Anthony Minessale wrote:
>>>
>>> That's is a somewhat vague position.
>>>
>>> You did not mention which version of FreeSWITCH you are running, the
>>> phones
>>> being used in your example, your configuration, the codecs in use  
>>> etc.
>>>
>>> BTW,
>>> I think you should only ask one question at a time on this list.   
>>> The list
>>> is run by volunteers and it's sort of rude to expect 3 or 4 threads  
>>> to be
>>> tended to concerning the same one individual.
>>>
>>>
>>> 2009/10/5 Maciej Aniserowicz <maciej.aniserowicz at gmail.com>
>>>
>>>> Hello,
>>>> When I use eavesdropping in FreeSWITCH, the sound quality is  
>>>> really bad.
>>>> Is
>>>> there any way to improve it? Is this a known problem?
>>>> Br/
>>>> Maciej Aniserowicz
>>>>
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3788019.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From tculjaga at gmail.com  Thu Oct  8 07:20:44 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 8 Oct 2009 16:20:44 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
Message-ID: <65d96fc80910080720x5a9e9756tab7c08ec6b67f163@mail.gmail.com>

Hi Yuriy, did you manage to do something with H323plus and FS ?

btw: have you checked Objective OpenH323
http://www.obj-sys.com/telephony-objective.shtml ?
This looks better to me as it is lighter and can be easily customized.

T.


2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-07 15:09 -0500, Brian West wrote
> freeswitch-users at lists.freeswit...:
>
> opal have addition abstraction layer called opalmgr, and it implementation
> is not so good in
> this case, for example to implemet pre_answer in mod_opal i need patch
> libopal, because
> there is no way to send progress inicator throuch opalmgr. and there is
> many another issues like
> this in that layer. ftp://srv.icf.org.ru/pub/soft/f/freeswitch/ - there is
> my work on mod_opal before
> i start moving to h323plus, may be this help somebody there.
>
> BW> From what I have been told h323plus is a based/fork of OpenH323 which
> BW>OPAL is just a continuation of OpenH323.  So why not support the
> BW>developers of OPAL/OpenH323 ?
> BW>
> BW>/b
> BW>
> BW>On Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote:
> BW>
> BW>> We are developing module to handle h323 proto now, we try to use
> BW>> mod_opal and try improve it, but no luck,
> BW>> there is many issues in libopal, and finaly we now move to h323plus
> BW>> library.
> BW>
> BW>
> BW>_______________________________________________
> BW>FreeSWITCH-users mailing list
> BW>FreeSWITCH-users at lists.freeswitch.org
> BW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> BW>UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> BW>http://www.freeswitch.org
> BW>
>
> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129-RIPE
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rupa at rupa.com  Thu Oct  8 07:40:09 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 8 Oct 2009 08:40:09 -0600
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910080720x5a9e9756tab7c08ec6b67f163@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<65d96fc80910080720x5a9e9756tab7c08ec6b67f163@mail.gmail.com>
Message-ID: <b39273530910080740u7671454eg36cab6bee8339296@mail.gmail.com>

That code is GPL, not license compatible. ;(

2009/10/8 Tihomir Culjaga <tculjaga at gmail.com>:
> Hi Yuriy, did you manage to do something with H323plus and FS ?
>
> btw: have you checked Objective OpenH323
> http://www.obj-sys.com/telephony-objective.shtml ?
> This looks better to me as it is lighter and can be easily customized.
>
> T.


-- 
-Rupa



From dome at tel.co.th  Thu Oct  8 07:39:46 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Thu, 8 Oct 2009 21:39:46 +0700
Subject: [Freeswitch-users] max-sessions change when running about 2 hr
Message-ID: <8ccbff060910080739h14143bf8lb43060a5ec6393a0@mail.gmail.com>

Dear sir,
   I'm running freeswitch with option
/opt/freeswitch/bin/freeswitch -nf -waste -nonat -nc
I'm setting up max-sessions in switch.conf to 10000

    <param name="max-sessions" value="10000"/>
    <!--Most channels to create per second -->
    <param name="sessions-per-second" value="50"/>
But when fs running about 2 hr. max session down to 299
freeswitch at internal> status
UP 0 years, 0 days, 2 hours, 11 minutes, 10 seconds, 434 milliseconds,
251 microseconds
20502 session(s) since startup
271 session(s) 0/50
299 session(s) max

Can somone tell me what's happen ?

Dome C.



From gabe at gundy.org  Thu Oct  8 07:42:22 2009
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 8 Oct 2009 08:42:22 -0600
Subject: [Freeswitch-users] FS Slide deck?
Message-ID: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>

Hey all,

The 3rd annual UTOSC (Utah Open Source Conference) [1] starts today.
It's a conference that gives Open Source users/hackers a chance gather
together to mingle and share.  It's a lot of fun.  Anyway, this year
I've offered to work with a friend of mine to lead a BoF (Birds of a
Feather) discussion on Open Source Telephony [2].

While it should mostly be group discussion, you never know when people
are going to get shy and clam up.  *If* they do, it would be nice to
have some slides to fill that time.  Rather than make some from
scratch (that are unlikely to get used), I thought I'd ask here and
see if anyone has a FS slide deck that they wouldn't mind sharing.

Thanks,
Gabe


1) http://2009.utosc.com/pages/home/
2) http://2009.utosc.com/presentation/136/



From vhatz at kinetix.gr  Thu Oct  8 07:44:55 2009
From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI))
Date: Thu, 08 Oct 2009 17:44:55 +0300
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910080720x5a9e9756tab7c08ec6b67f163@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<65d96fc80910080720x5a9e9756tab7c08ec6b67f163@mail.gmail.com>
Message-ID: <4ACDFAE7.1000409@kinetix.gr>


Tihomir Culjaga wrote:
> Hi Yuriy, did you manage to do something with H323plus and FS ?
> 
> btw: have you checked Objective OpenH323 
> http://www.obj-sys.com/telephony-objective.shtml ?
> This looks better to me as it is lighter and can be easily customized.
> 
> T.

AFAIK Objective Systems H323 library does not support fax or video, it 
does only voice (at least the last time I checked it which was many 
months ago). Using it would lead to a mod for H323 which would never be 
as flexible or feature-rich as mod_sofia...

Regards,
Vlasis.



From bottleman at icf.org.ru  Thu Oct  8 07:44:09 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Thu, 8 Oct 2009 18:44:09 +0400 (MSD)
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910080720x5a9e9756tab7c08ec6b67f163@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<65d96fc80910080720x5a9e9756tab7c08ec6b67f163@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910081830380.11841@srv.icf.org.ru>

On 2009-10-08 16:20 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>Hi Yuriy, did you manage to do something with H323plus and FS ?

i already doing it, but now it not in usable state.

TC>btw: have you checked Objective OpenH323
TC>http://www.obj-sys.com/telephony-objective.shtml ?
TC>This looks better to me as it is lighter and can be easily customized.

i see this library later in asterisk module, h323plus is a successor of opanh323, i use it many yars and 
i think it more complete mature and stable than objective systems stack, and finally h323plus not depend 
in its development from some kinde of Objective System Inc/any other xxx Inc.

TC>
TC>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
TC>
TC>> On 2009-10-07 15:09 -0500, Brian West wrote
TC>> freeswitch-users at lists.freeswit...:
TC>>
TC>> opal have addition abstraction layer called opalmgr, and it implementation
TC>> is not so good in
TC>> this case, for example to implemet pre_answer in mod_opal i need patch
TC>> libopal, because
TC>> there is no way to send progress inicator throuch opalmgr. and there is
TC>> many another issues like
TC>> this in that layer. ftp://srv.icf.org.ru/pub/soft/f/freeswitch/ - there is
TC>> my work on mod_opal before
TC>> i start moving to h323plus, may be this help somebody there.
TC>>
TC>> BW> From what I have been told h323plus is a based/fork of OpenH323 which
TC>> BW>OPAL is just a continuation of OpenH323.  So why not support the
TC>> BW>developers of OPAL/OpenH323 ?
TC>> BW>
TC>> BW>/b
TC>> BW>
TC>> BW>On Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote:
TC>> BW>
TC>> BW>> We are developing module to handle h323 proto now, we try to use
TC>> BW>> mod_opal and try improve it, but no luck,
TC>> BW>> there is many issues in libopal, and finaly we now move to h323plus
TC>> BW>> library.
TC>> BW>
TC>> BW>
TC>> BW>_______________________________________________
TC>> BW>FreeSWITCH-users mailing list
TC>> BW>FreeSWITCH-users at lists.freeswitch.org
TC>> BW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> BW>UNSUBSCRIBE:
TC>> http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> BW>http://www.freeswitch.org
TC>> BW>
TC>>
TC>> C ?????????                       With Best Regards
TC>> ???????????? ????.                Georgiewskiy Yuriy
TC>> +7 4872 711666                    +7 4872 711666
TC>> ???? +7 4872 711143               fax +7 4872 711143
TC>> ???????? ??? "?? ?? ??????"       IT Service Ltd
TC>> http://nkoort.ru                  http://nkoort.ru
TC>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
TC>> YG129-RIPE                        YG129-RIPE
TC>>
TC>> _______________________________________________
TC>> FreeSWITCH-users mailing list
TC>> FreeSWITCH-users at lists.freeswitch.org
TC>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> http://www.freeswitch.org
TC>>
TC>>
TC>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From dome at tel.co.th  Thu Oct  8 07:46:44 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Thu, 8 Oct 2009 21:46:44 +0700
Subject: [Freeswitch-users] max-sessions change when running about 2 hr
In-Reply-To: <8ccbff060910080739h14143bf8lb43060a5ec6393a0@mail.gmail.com>
References: <8ccbff060910080739h14143bf8lb43060a5ec6393a0@mail.gmail.com>
Message-ID: <8ccbff060910080746p30333c13t2a11a70d0b961e3@mail.gmail.com>

When i try
fsctl max_sessions 10000
in CLI . FS set it back to 2xx again

Dome C.
2009/10/8 Dome Charoenyost <dome at tel.co.th>:
> Dear sir,
> ? I'm running freeswitch with option
> /opt/freeswitch/bin/freeswitch -nf -waste -nonat -nc
> I'm setting up max-sessions in switch.conf to 10000
>
> ? ?<param name="max-sessions" value="10000"/>
> ? ?<!--Most channels to create per second -->
> ? ?<param name="sessions-per-second" value="50"/>
> But when fs running about 2 hr. max session down to 299
> freeswitch at internal> status
> UP 0 years, 0 days, 2 hours, 11 minutes, 10 seconds, 434 milliseconds,
> 251 microseconds
> 20502 session(s) since startup
> 271 session(s) 0/50
> 299 session(s) max
>
> Can somone tell me what's happen ?
>
> Dome C.
>



From brian at freeswitch.org  Thu Oct  8 07:59:24 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 8 Oct 2009 09:59:24 -0500
Subject: [Freeswitch-users] max-sessions change when running about 2 hr
In-Reply-To: <8ccbff060910080746p30333c13t2a11a70d0b961e3@mail.gmail.com>
References: <8ccbff060910080739h14143bf8lb43060a5ec6393a0@mail.gmail.com>
	<8ccbff060910080746p30333c13t2a11a70d0b961e3@mail.gmail.com>
Message-ID: <75EBC642-1912-4CFA-8774-40DF1FA0152E@freeswitch.org>

How much ram do you have and what distro are you running on, 32bit or  
64bit?

/b

On Oct 8, 2009, at 9:46 AM, Dome Charoenyost wrote:

> When i try
> fsctl max_sessions 10000
> in CLI . FS set it back to 2xx again
>
> Dome C.




From tculjaga at gmail.com  Thu Oct  8 08:00:30 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 8 Oct 2009 17:00:30 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910081830380.11841@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<65d96fc80910080720x5a9e9756tab7c08ec6b67f163@mail.gmail.com>
	<Pine.LNX.4.64.0910081830380.11841@srv.icf.org.ru>
Message-ID: <65d96fc80910080800t5427abc5m2012ff2a5c3326c0@mail.gmail.com>

yep, you made the point :P

T.


2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-08 16:20 +0200, Tihomir Culjaga wrote
> freeswitch-users at lists.fre...:
>
> TC>Hi Yuriy, did you manage to do something with H323plus and FS ?
>
> i already doing it, but now it not in usable state.
>
> TC>btw: have you checked Objective OpenH323
> TC>http://www.obj-sys.com/telephony-objective.shtml ?
> TC>This looks better to me as it is lighter and can be easily customized.
>
> i see this library later in asterisk module, h323plus is a successor of
> opanh323, i use it many yars and
> i think it more complete mature and stable than objective systems stack,
> and finally h323plus not depend
> in its development from some kinde of Objective System Inc/any other xxx
> Inc.
>
> TC>
> TC>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
> TC>
> TC>> On 2009-10-07 15:09 -0500, Brian West wrote
> TC>> freeswitch-users at lists.freeswit...:
> TC>>
> TC>> opal have addition abstraction layer called opalmgr, and it
> implementation
> TC>> is not so good in
> TC>> this case, for example to implemet pre_answer in mod_opal i need patch
> TC>> libopal, because
> TC>> there is no way to send progress inicator throuch opalmgr. and there
> is
> TC>> many another issues like
> TC>> this in that layer. ftp://srv.icf.org.ru/pub/soft/f/freeswitch/ -
> there is
> TC>> my work on mod_opal before
> TC>> i start moving to h323plus, may be this help somebody there.
> TC>>
> TC>> BW> From what I have been told h323plus is a based/fork of OpenH323
> which
> TC>> BW>OPAL is just a continuation of OpenH323.  So why not support the
> TC>> BW>developers of OPAL/OpenH323 ?
> TC>> BW>
> TC>> BW>/b
> TC>> BW>
> TC>> BW>On Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote:
> TC>> BW>
> TC>> BW>> We are developing module to handle h323 proto now, we try to use
> TC>> BW>> mod_opal and try improve it, but no luck,
> TC>> BW>> there is many issues in libopal, and finaly we now move to
> h323plus
> TC>> BW>> library.
> TC>> BW>
> TC>> BW>
> TC>> BW>_______________________________________________
> TC>> BW>FreeSWITCH-users mailing list
> TC>> BW>FreeSWITCH-users at lists.freeswitch.org
> TC>> BW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> TC>> BW>UNSUBSCRIBE:
> TC>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> TC>> BW>http://www.freeswitch.org
> TC>> BW>
> TC>>
> TC>> C ?????????                       With Best Regards
> TC>> ???????????? ????.                Georgiewskiy Yuriy
> TC>> +7 4872 711666                    +7 4872 711666
> TC>> ???? +7 4872 711143               fax +7 4872 711143
> TC>> ???????? ??? "?? ?? ??????"       IT Service Ltd
> TC>> http://nkoort.ru                  http://nkoort.ru
> TC>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> TC>> YG129-RIPE                        YG129-RIPE
> TC>>
> TC>> _______________________________________________
> TC>> FreeSWITCH-users mailing list
> TC>> FreeSWITCH-users at lists.freeswitch.org
> TC>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> TC>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> TC>> http://www.freeswitch.org
> TC>>
> TC>>
> TC>
>
> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129-RIPE
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From moises.silva at gmail.com  Thu Oct  8 08:02:26 2009
From: moises.silva at gmail.com (Moises Silva)
Date: Thu, 8 Oct 2009 11:02:26 -0400
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <7d79b3930910080241q4eeb374an2ec34642bf450818@mail.gmail.com>
References: <7d79b3930910070334g53cef0c5m63037e264f1a9233@mail.gmail.com>
	<87f2f3b90910070849t79b7cb90x4150bf1dd8501ca4@mail.gmail.com>
	<7d79b3930910080241q4eeb374an2ec34642bf450818@mail.gmail.com>
Message-ID: <c4d05cbe0910080802t7ebf8007m333cbd6e74efd286@mail.gmail.com>

On Thu, Oct 8, 2009 at 5:41 AM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Hi I tried with the following openzap.conf
> [span zt PRI_1]
> trunk_type => e1
> b-channel => 1:1-15,17-31
> d-channel => 1:16
>

This does not look like a healthy config to me. You are using
<span>:<channels> notation in zaptel spans, zaptel channels increment across
spans, read point number 5 of this web page
http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2 , this is for another type of
signaling, but the span/channel numbering concepts are the same.

-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From dome at tel.co.th  Thu Oct  8 08:14:43 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Thu, 8 Oct 2009 22:14:43 +0700
Subject: [Freeswitch-users] max-sessions change when running about 2 hr
In-Reply-To: <75EBC642-1912-4CFA-8774-40DF1FA0152E@freeswitch.org>
References: <8ccbff060910080739h14143bf8lb43060a5ec6393a0@mail.gmail.com>
	<8ccbff060910080746p30333c13t2a11a70d0b961e3@mail.gmail.com>
	<75EBC642-1912-4CFA-8774-40DF1FA0152E@freeswitch.org>
Message-ID: <8ccbff060910080814v56dbd129nccb1b99353a367cf@mail.gmail.com>

2009/10/8 Brian West <brian at freeswitch.org>:
> How much ram do you have and what distro are you running on, 32bit or
> 64bit?

2GB  32bit  debian squeeze
(swap 6 GB)

Dome C.


>
> /b
>
> On Oct 8, 2009, at 9:46 AM, Dome Charoenyost wrote:
>
>> When i try
>> fsctl max_sessions 10000
>> in CLI . FS set it back to 2xx again
>>
>> Dome C.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From dome at tel.co.th  Thu Oct  8 08:15:42 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Thu, 8 Oct 2009 22:15:42 +0700
Subject: [Freeswitch-users] max-sessions change when running about 2 hr
In-Reply-To: <8ccbff060910080814v56dbd129nccb1b99353a367cf@mail.gmail.com>
References: <8ccbff060910080739h14143bf8lb43060a5ec6393a0@mail.gmail.com>
	<8ccbff060910080746p30333c13t2a11a70d0b961e3@mail.gmail.com>
	<75EBC642-1912-4CFA-8774-40DF1FA0152E@freeswitch.org>
	<8ccbff060910080814v56dbd129nccb1b99353a367cf@mail.gmail.com>
Message-ID: <8ccbff060910080815h7f79f68v992270146a747cfe@mail.gmail.com>

may be -waste option effect  ?

2009/10/8 Dome Charoenyost <dome at tel.co.th>:
> 2009/10/8 Brian West <brian at freeswitch.org>:
>> How much ram do you have and what distro are you running on, 32bit or
>> 64bit?
>
> 2GB ?32bit ?debian squeeze
> (swap 6 GB)
>
> Dome C.
>
>
>>
>> /b
>>
>> On Oct 8, 2009, at 9:46 AM, Dome Charoenyost wrote:
>>
>>> When i try
>>> fsctl max_sessions 10000
>>> in CLI . FS set it back to 2xx again
>>>
>>> Dome C.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>



From anthony.minessale at gmail.com  Thu Oct  8 08:21:14 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 8 Oct 2009 10:21:14 -0500
Subject: [Freeswitch-users] directory variables not set correct with the
	latest trunk
In-Reply-To: <27c25bc40910080320x7b3b380eu933797197e334cb8@mail.gmail.com>
References: <27c25bc40910080320x7b3b380eu933797197e334cb8@mail.gmail.com>
Message-ID: <191c3a030910080821w691aa78aw272d71dee7675ff1@mail.gmail.com>

they only set if you authenticate on sip or run the set_user app


On Thu, Oct 8, 2009 at 5:20 AM, Juan Backson <juanbackson at gmail.com> wrote:

> Hi,
>
> My application fails to set the appropriate variables using directory xml
> after using the latest trunk as of yesterday.
>
> My curl looks like:
>
>
>
> <document type="freeswitch/xml">
>   <section name="directory">
>     <domain name="192.168.1.102">
>       <user id="200002">
>         <params>
>           <param name="password" value="123456"></param>
>           <param name="dial-string"
> value="sofia/internal/sip:200002 at 192.168.1.11:29440
> ;rinstance=0245b2a59ddff837"></param>
>         </params>
>         <variables>
>           <variable name="account_id" value="20"></variable>
>           <variable name="vm-code" value="1000"></variable>
>         </variables>
>       </user>
>     </domain>
>   </section>
> </document>
>
>
>
> In the info app, I am not seeing account-id and vm-code anymore.
>
> How to fix that?
>
> juan
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Thu Oct  8 08:23:13 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 8 Oct 2009 10:23:13 -0500
Subject: [Freeswitch-users] Bad sound quality while eavesdropping
In-Reply-To: <1255008427639-3788019.post@n2.nabble.com>
References: <41A44DD027064988A914974405788C2E@procent>
	<191c3a030910050731m2d74979ep4598e5a1945d58ae@mail.gmail.com>
	<1254901192035-3780245.post@n2.nabble.com>
	<8437F5BC-7AFF-4A74-B8CD-C5B8219021F6@jerris.com>
	<1255008427639-3788019.post@n2.nabble.com>
Message-ID: <191c3a030910080823g79c7c596x1cd887e1538ce2e1@mail.gmail.com>

you probably have some device lying about ptime everywhere
look at a sip trace an pay especially close attention to ptime:x param in
sdp
if you don't understand this just attach it here

execute the following at the cli
sofia profile internal siptrace on
sofila loglevel debug



On Thu, Oct 8, 2009 at 8:27 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> It's the same on the trunk (the last rev I used was not so old anyway).
>
> Codecs are the same on both legs:
> read codec/read rate: PCMU      8000
> write codec/write rate: PCMU    8000
>
> MA
>
>
>
>
> Michael Jerris wrote:
> >
> > What codecs are all the call legs using, also, please try current svn
> > trunk.
> >
> > Mike
> >
> > On Oct 7, 2009, at 3:39 AM, Maciej Aniserowicz wrote:
> >
> >>
> >> Sorry about posting several questions at once, I wasn't aware it's
> >> "rude".
> >> Let's concentrate on this issue then.
> >>
> >> I use FS rev 14994. Phones on extensions:
> >> 1) x-lite
> >> 2) cisco sip phone
> >> 3) audio played by fs to the extension being eavesdropped
> >>
> >> I did not change any codec configuration, I just use the standard
> >> one that
> >> comes with both FS and the phones.
> >> Some time ago someone on FS irc channel told me that this is just
> >> how FS
> >> eavesdropping works... from your response I understand that this is
> >> not
> >> entirely true?
> >>
> >> Maciej Aniserowicz
> >>
> >>
> >>
> >> Anthony Minessale wrote:
> >>>
> >>> That's is a somewhat vague position.
> >>>
> >>> You did not mention which version of FreeSWITCH you are running, the
> >>> phones
> >>> being used in your example, your configuration, the codecs in use
> >>> etc.
> >>>
> >>> BTW,
> >>> I think you should only ask one question at a time on this list.
> >>> The list
> >>> is run by volunteers and it's sort of rude to expect 3 or 4 threads
> >>> to be
> >>> tended to concerning the same one individual.
> >>>
> >>>
> >>> 2009/10/5 Maciej Aniserowicz <maciej.aniserowicz at gmail.com>
> >>>
> >>>> Hello,
> >>>> When I use eavesdropping in FreeSWITCH, the sound quality is
> >>>> really bad.
> >>>> Is
> >>>> there any way to improve it? Is this a known problem?
> >>>> Br/
> >>>> Maciej Aniserowicz
> >>>>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> --
> View this message in context:
> http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3788019.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Thu Oct  8 08:29:21 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 8 Oct 2009 10:29:21 -0500
Subject: [Freeswitch-users] max-sessions change when running about 2 hr
In-Reply-To: <8ccbff060910080815h7f79f68v992270146a747cfe@mail.gmail.com>
References: <8ccbff060910080739h14143bf8lb43060a5ec6393a0@mail.gmail.com>
	<8ccbff060910080746p30333c13t2a11a70d0b961e3@mail.gmail.com>
	<75EBC642-1912-4CFA-8774-40DF1FA0152E@freeswitch.org>
	<8ccbff060910080814v56dbd129nccb1b99353a367cf@mail.gmail.com>
	<8ccbff060910080815h7f79f68v992270146a747cfe@mail.gmail.com>
Message-ID: <191c3a030910080829s7886f664m73f6e504562bb38d@mail.gmail.com>

you are running out of threads and it's fixing itself. one of many reasons
you should upgrade to 64 bit OS

execute this in your shell before you start FS

ulimit -c unlimited
ulimit -d unlimited
ulimit -f unlimited
ulimit -i unlimited
ulimit -n 999999
ulimit -q unlimited
ulimit -u unlimited
ulimit -v unlimited
ulimit -x unlimited
ulimit -s 244
ulimit -l unlimited



On Thu, Oct 8, 2009 at 10:15 AM, Dome Charoenyost <dome at tel.co.th> wrote:

> may be -waste option effect  ?
>
> 2009/10/8 Dome Charoenyost <dome at tel.co.th>:
> > 2009/10/8 Brian West <brian at freeswitch.org>:
> >> How much ram do you have and what distro are you running on, 32bit or
> >> 64bit?
> >
> > 2GB  32bit  debian squeeze
> > (swap 6 GB)
> >
> > Dome C.
> >
> >
> >>
> >> /b
> >>
> >> On Oct 8, 2009, at 9:46 AM, Dome Charoenyost wrote:
> >>
> >>> When i try
> >>> fsctl max_sessions 10000
> >>> in CLI . FS set it back to 2xx again
> >>>
> >>> Dome C.
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Thu Oct  8 08:43:14 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 8 Oct 2009 10:43:14 -0500
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
Message-ID: <191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>

If you are going to make that alternate module are you going to host it in
the FS tree along side mod_opal?
also if were working on mod_opal why did you not try to involve us and the
opal team?
How far away from what is in tree are these patches you have?

2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-07 15:09 -0500, Brian West wrote
> freeswitch-users at lists.freeswit...:
>
> opal have addition abstraction layer called opalmgr, and it implementation
> is not so good in
> this case, for example to implemet pre_answer in mod_opal i need patch
> libopal, because
> there is no way to send progress inicator throuch opalmgr. and there is
> many another issues like
> this in that layer. ftp://srv.icf.org.ru/pub/soft/f/freeswitch/ - there is
> my work on mod_opal before
> i start moving to h323plus, may be this help somebody there.
>
> BW> From what I have been told h323plus is a based/fork of OpenH323 which
> BW>OPAL is just a continuation of OpenH323.  So why not support the
> BW>developers of OPAL/OpenH323 ?
> BW>
> BW>/b
> BW>
> BW>On Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote:
> BW>
> BW>> We are developing module to handle h323 proto now, we try to use
> BW>> mod_opal and try improve it, but no luck,
> BW>> there is many issues in libopal, and finaly we now move to h323plus
> BW>> library.
> BW>
> BW>
> BW>_______________________________________________
> BW>FreeSWITCH-users mailing list
> BW>FreeSWITCH-users at lists.freeswitch.org
> BW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> BW>UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> BW>http://www.freeswitch.org
> BW>
>
> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129-RIPE
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From dome at tel.co.th  Thu Oct  8 08:50:36 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Thu, 8 Oct 2009 22:50:36 +0700
Subject: [Freeswitch-users] max-sessions change when running about 2 hr
In-Reply-To: <191c3a030910080829s7886f664m73f6e504562bb38d@mail.gmail.com>
References: <8ccbff060910080739h14143bf8lb43060a5ec6393a0@mail.gmail.com>
	<8ccbff060910080746p30333c13t2a11a70d0b961e3@mail.gmail.com>
	<75EBC642-1912-4CFA-8774-40DF1FA0152E@freeswitch.org>
	<8ccbff060910080814v56dbd129nccb1b99353a367cf@mail.gmail.com>
	<8ccbff060910080815h7f79f68v992270146a747cfe@mail.gmail.com>
	<191c3a030910080829s7886f664m73f6e504562bb38d@mail.gmail.com>
Message-ID: <8ccbff060910080850k2a849842u450a303b92d27619@mail.gmail.com>

Thanks.  I'll try 64 bit :)

2009/10/8 Anthony Minessale <anthony.minessale at gmail.com>:
> you are running out of threads and it's fixing itself. one of many reasons
> you should upgrade to 64 bit OS
>
> execute this in your shell before you start FS
>
> ulimit -c unlimited
> ulimit -d unlimited
> ulimit -f unlimited
>
> ulimit -i unlimited
> ulimit -n 999999
> ulimit -q unlimited
> ulimit -u unlimited
> ulimit -v unlimited
> ulimit -x unlimited
> ulimit -s 244
> ulimit -l unlimited
>
>
> On Thu, Oct 8, 2009 at 10:15 AM, Dome Charoenyost <dome at tel.co.th> wrote:
>>
>> may be -waste option effect ??
>>
>> 2009/10/8 Dome Charoenyost <dome at tel.co.th>:
>> > 2009/10/8 Brian West <brian at freeswitch.org>:
>> >> How much ram do you have and what distro are you running on, 32bit or
>> >> 64bit?
>> >
>> > 2GB ?32bit ?debian squeeze
>> > (swap 6 GB)
>> >
>> > Dome C.
>> >
>> >
>> >>
>> >> /b
>> >>
>> >> On Oct 8, 2009, at 9:46 AM, Dome Charoenyost wrote:
>> >>
>> >>> When i try
>> >>> fsctl max_sessions 10000
>> >>> in CLI . FS set it back to 2xx again
>> >>>
>> >>> Dome C.
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From bottleman at icf.org.ru  Thu Oct  8 09:03:00 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Thu, 8 Oct 2009 20:03:00 +0400 (MSD)
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>

On 2009-10-08 10:43 -0500, Anthony Minessale wrote freeswitch-users at lists.f...:

AM>If you are going to make that alternate module are you going to host it in
AM>the FS tree along side mod_opal?

Yes, but then it be useful, now i have working only signaling part and some 
kinde of not working rtp part :)

AM>also if were working on mod_opal why did you not try to involve us and the
AM>opal team?

Because i made patches for libopal, one is a bugfix in rtp part, there is a race condition 
in inicialisation in jitter buffer, another patch implements method to send progress indicator, 
and i don't wont spent my time to incorporate this changes into libopal. without this changes
my work on mod_opal in freeswitch don't useful at all, i provide link to my work with all 
patches, if somebody wont incorporate it in libopal tree and fs - go on, but i think 
better and more elegant make new module based on h323plus.

AM>How far away from what is in tree are these patches you have?
AM>
AM>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
AM>
AM>> On 2009-10-07 15:09 -0500, Brian West wrote
AM>> freeswitch-users at lists.freeswit...:
AM>>
AM>> opal have addition abstraction layer called opalmgr, and it implementation
AM>> is not so good in
AM>> this case, for example to implemet pre_answer in mod_opal i need patch
AM>> libopal, because
AM>> there is no way to send progress inicator throuch opalmgr. and there is
AM>> many another issues like
AM>> this in that layer. ftp://srv.icf.org.ru/pub/soft/f/freeswitch/ - there is
AM>> my work on mod_opal before
AM>> i start moving to h323plus, may be this help somebody there.
AM>>
AM>> BW> From what I have been told h323plus is a based/fork of OpenH323 which
AM>> BW>OPAL is just a continuation of OpenH323.  So why not support the
AM>> BW>developers of OPAL/OpenH323 ?
AM>> BW>
AM>> BW>/b
AM>> BW>
AM>> BW>On Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote:
AM>> BW>
AM>> BW>> We are developing module to handle h323 proto now, we try to use
AM>> BW>> mod_opal and try improve it, but no luck,
AM>> BW>> there is many issues in libopal, and finaly we now move to h323plus
AM>> BW>> library.
AM>> BW>
AM>> BW>
AM>> BW>_______________________________________________
AM>> BW>FreeSWITCH-users mailing list
AM>> BW>FreeSWITCH-users at lists.freeswitch.org
AM>> BW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
AM>> BW>UNSUBSCRIBE:
AM>> http://lists.freeswitch.org/mailman/options/freeswitch-users
AM>> BW>http://www.freeswitch.org
AM>> BW>
AM>>
AM>> C ?????????                       With Best Regards
AM>> ???????????? ????.                Georgiewskiy Yuriy
AM>> +7 4872 711666                    +7 4872 711666
AM>> ???? +7 4872 711143               fax +7 4872 711143
AM>> ???????? ??? "?? ?? ??????"       IT Service Ltd
AM>> http://nkoort.ru                  http://nkoort.ru
AM>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
AM>> YG129-RIPE                        YG129-RIPE
AM>>
AM>> _______________________________________________
AM>> FreeSWITCH-users mailing list
AM>> FreeSWITCH-users at lists.freeswitch.org
AM>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
AM>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
AM>> http://www.freeswitch.org
AM>>
AM>>
AM>
AM>
AM>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From msc at freeswitch.org  Thu Oct  8 09:47:04 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Oct 2009 09:47:04 -0700
Subject: [Freeswitch-users] unable to configure Digium TDM400P
In-Reply-To: <a44688a0910072245v7f496385of411552f0098ba90@mail.gmail.com>
References: <a44688a0910070340p4209e7b5yec9218b8c4fc692c@mail.gmail.com>
	<87f2f3b90910070956l520b2c51p7f33463128bf68b2@mail.gmail.com>
	<a44688a0910072245v7f496385of411552f0098ba90@mail.gmail.com>
Message-ID: <87f2f3b90910080947h3cebf253s6e011dd8ddda8504@mail.gmail.com>

On Wed, Oct 7, 2009 at 10:45 PM, Woody Dickson <woodydickson at gmail.com>wrote:

>
> Hi Michael
>
> Is the ztcfg output supposed to say something like 2 channels configured?
>

Yes. If you had the signaling incorrect it would complain.


>
> I have not set up openzap yet because I don't know if the openzap.config is
> good or not.
>
>
You need to make sure that zaptel is set up first. Running ztcfg and getting
no errors or warnings, plus a message that says "2 channels configured" is a
pretty good sign that you can move on to the OpenZAP stuff.


> Can you give me any suggestion if the ztcfg output I am getting is proper
> or not?
>
> No obvious problems, so move on to the OpenZAP configs.
-MC

> thx,
> woody
>
>
>
> -MC
>>
>> On Wed, Oct 7, 2009 at 3:40 AM, Woody Dickson <woodydickson at gmail.com>wrote:
>>
>>>  Hi,
>>>
>>> I am trying to setup a Digium TDM400P following the instruction on the
>>> wiki.
>>> It is a 1 fxo and 1 fxs card, so I tried
>>> loadzone=in
>>> defaultzone=in
>>> fxsks=2
>>> fxoks=1
>>>
>>> and
>>>
>>> loadzone=in
>>> defaultzone=in
>>> fxsks=1
>>> fxoks=2
>>>
>>> None works.  Does anyone know how it should be configured?
>>>
>>> Here is what I get by following the wiki.
>>>
>>> [root at localhost zaptel]# ztcfg -vv
>>>
>>> Zaptel Version: SVN-branch-1.4-r4629M
>>> Echo Canceller: MG2
>>> Configuration
>>> ======================
>>>
>>>
>>> Channel map:
>>>
>>> Channel 01: FXS Kewlstart (Default) (Slaves: 01)
>>> Channel 02: FXO Kewlstart (Default) (Slaves: 02)
>>>
>>> 2 channels to configure.
>>>
>>> Changing signalling on channel 1 from FXO Kewlstart to FXS Kewlstart
>>> Changing signalling on channel 2 from FXS Kewlstart to FXO Kewlstart
>>>
>>>
>>> [root at localhost zaptel]# ztcfg -vv
>>>
>>> Zaptel Version: SVN-branch-1.4-r4629M
>>> Echo Canceller: MG2
>>> Configuration
>>> ======================
>>>
>>>
>>> Channel map:
>>>
>>> Channel 01: FXO Kewlstart (Default) (Slaves: 01)
>>> Channel 02: FXS Kewlstart (Default) (Slaves: 02)
>>>
>>> 2 channels to configure.
>>>
>>> Changing signalling on channel 1 from FXS Kewlstart to FXO Kewlstart
>>> Changing signalling on channel 2 from FXO Kewlstart to FXS Kewlstart
>>>
>>>
>>> [root at localhost zaptel]# lspci
>>> 00:14.0 Ethernet controller: Digium, Inc. TDM400P (rev 11)
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vhatz at kinetix.gr  Thu Oct  8 09:53:28 2009
From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI))
Date: Thu, 08 Oct 2009 19:53:28 +0300
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
Message-ID: <4ACE1908.9040004@kinetix.gr>

Georgiewskiy Yuriy wrote:
> On 2009-10-08 10:43 -0500, Anthony Minessale wrote freeswitch-users at lists.f...:
> 
> AM>If you are going to make that alternate module are you going to host it in
> AM>the FS tree along side mod_opal?
> 
> Yes, but then it be useful, now i have working only signaling part and some 
> kinde of not working rtp part :)


Hello Yuriy,

Since FS already has its own RTP stack (for pass-through and 
transcoding) wouldn't it be best if your H323 implementation with OPAL 
was used for signaling only and rely on FS RTP for the media? Perhaps 
that would save you time and effort, too?


Best regards,
Vlasis Hatzistavrou.



From msc at freeswitch.org  Thu Oct  8 09:54:18 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Oct 2009 09:54:18 -0700
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <c4d05cbe0910080802t7ebf8007m333cbd6e74efd286@mail.gmail.com>
References: <7d79b3930910070334g53cef0c5m63037e264f1a9233@mail.gmail.com>
	<87f2f3b90910070849t79b7cb90x4150bf1dd8501ca4@mail.gmail.com>
	<7d79b3930910080241q4eeb374an2ec34642bf450818@mail.gmail.com>
	<c4d05cbe0910080802t7ebf8007m333cbd6e74efd286@mail.gmail.com>
Message-ID: <87f2f3b90910080954l1ecb867dn8a025f5f0f701eb4@mail.gmail.com>

On Thu, Oct 8, 2009 at 8:02 AM, Moises Silva <moises.silva at gmail.com> wrote:

> On Thu, Oct 8, 2009 at 5:41 AM, lakshmanan ganapathy <lakindia89 at gmail.com
> > wrote:
>
>> Hi I tried with the following openzap.conf
>> [span zt PRI_1]
>> trunk_type => e1
>> b-channel => 1:1-15,17-31
>> d-channel => 1:16
>>
>
> This does not look like a healthy config to me. You are using
> <span>:<channels> notation in zaptel spans, zaptel channels increment across
> spans, read point number 5 of this web page
> http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2 , this is for another type
> of signaling, but the span/channel numbering concepts are the same.
>
>
Moy,

As usual you are right on the money. I'm so used to doing Sangoma configs
that I forget about the zaptel syntax. I recommend this config:

[span zt PRI_1]
trunk_type => e1
b-channel => 1-15
d-channel => 16
b-channel => 17-31


Give that a try and report back if you have issues.
-MC

> --
> Moises Silva
> Software Developer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Oct  8 09:58:12 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Oct 2009 09:58:12 -0700
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
Message-ID: <87f2f3b90910080958i1f5cfccax621fd12100a05e3e@mail.gmail.com>

On Thu, Oct 8, 2009 at 7:42 AM, Gabriel Gunderson <gabe at gundy.org> wrote:

> Hey all,
>
> The 3rd annual UTOSC (Utah Open Source Conference) [1] starts today.
> It's a conference that gives Open Source users/hackers a chance gather
> together to mingle and share.  It's a lot of fun.  Anyway, this year
> I've offered to work with a friend of mine to lead a BoF (Birds of a
> Feather) discussion on Open Source Telephony [2].
>
> While it should mostly be group discussion, you never know when people
> are going to get shy and clam up.  *If* they do, it would be nice to
> have some slides to fill that time.  Rather than make some from
> scratch (that are unlikely to get used), I thought I'd ask here and
> see if anyone has a FS slide deck that they wouldn't mind sharing.
>
> Thanks,
> Gabe
>
>
> 1) http://2009.utosc.com/pages/home/
> 2) http://2009.utosc.com/presentation/136/
>

There are numerous slides and such from ClueCon. Go to
http://www.cluecon.com and you can download both the video files as well as
the actual presentations. The presentation files are in various formats such
as PPT(X), PDF, and Keynote.
The second presentation on day 1 is Anthony Minessale and he talks a lot
about expanding one's thinking when it comes to applying FreeSWITCH in
production. For example, some people will jump through crazy hoops to solve
a problem "because that's how they've always done it." Anyway, check out
that video and presentation file for some great ideas.
-MC


>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Thu Oct  8 10:00:12 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Oct 2009 10:00:12 -0700
Subject: [Freeswitch-users] gateway FS informs it's client FS about
	users hanged up with a long delay
In-Reply-To: <1255007884165-3787956.post@n2.nabble.com>
References: <1A16501E57CA4727A593226A8C810308@procent>
	<98F72CCB-88B9-47ED-AED3-B6BA6DE648C0@jerris.com>
	<1255007884165-3787956.post@n2.nabble.com>
Message-ID: <87f2f3b90910081000v5b505f4ao53104211b620ea77@mail.gmail.com>

On Thu, Oct 8, 2009 at 6:18 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> Both of the instances are run on the same machine, i just changed the
> default
> ports they use. Can anything else cause this strange behavior?
> MA
>
Did a packet capture yield any clues? That is, were you able to confirm that
each instance sent and received all the packets that you believe they should
have sent and received? The reason I ask is so that you don't end up chasing
a ghost because you made an assumption somewhere in your troubleshooting.

-MC


>
>
>
> Michael Jerris wrote:
> >
> > Incorrect NAT configuration so one of the boxes is not actually
> > getting a BYE.
> >
> >
> > On Oct 5, 2009, at 3:13 AM, Maciej Aniserowicz wrote:
> >
> >> Hi,
> >> When I use two FreeSWITCH instances ('internal' and 'external'), all
> >> users register to the 'external' instance which acts as a gateway by
> >> 'internal' instance (which in turn is controlled by my applicaiton
> >> with commands sent by socket).
> >> When user hangs up, the 'hanged up' event is propagated to the
> >> 'internal' instance after a long time (~3 minutes) instead of being
> >> propagated immediately.
> >> What can cause this issue?
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> --
> View this message in context:
> http://n2.nabble.com/gateway-FS-informs-it-s-client-FS-about-users-hanged-up-with-a-long-delay-tp3768540p3787956.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bottleman at icf.org.ru  Thu Oct  8 10:03:25 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Thu, 8 Oct 2009 21:03:25 +0400 (MSD)
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <4ACE1908.9040004@kinetix.gr>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<4ACE1908.9040004@kinetix.gr>
Message-ID: <Pine.LNX.4.64.0910082058320.31576@srv.icf.org.ru>

On 2009-10-08 19:53 +0300, Vlasis Hatzistavrou (KTI) wrote freeswitch-users...:

VHK>Georgiewskiy Yuriy wrote:
VHK>> On 2009-10-08 10:43 -0500, Anthony Minessale wrote freeswitch-users at lists.f...:
VHK>> 
VHK>> AM>If you are going to make that alternate module are you going to host it in
VHK>> AM>the FS tree along side mod_opal?
VHK>> 
VHK>> Yes, but then it be useful, now i have working only signaling part and some 
VHK>> kinde of not working rtp part :)
VHK>
VHK>
VHK>Hello Yuriy,
VHK>
VHK>Since FS already has its own RTP stack (for pass-through and 
VHK>transcoding) wouldn't it be best if your H323 implementation with OPAL 
VHK>was used for signaling only and rely on FS RTP for the media? Perhaps 
VHK>that would save you time and effort, too?

it already use fs rtp stack, and my implementation based on h323plus, not on opal, 
but fs rtp stack not so complete, there need write some code to send/receive packets 
because fs stac only encode/decode packets, not send/receive, or i don't known how to 
do it with it. i don't find any example for this.


C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From anthony.minessale at gmail.com  Thu Oct  8 10:12:47 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 8 Oct 2009 12:12:47 -0500
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910082058320.31576@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<4ACE1908.9040004@kinetix.gr>
	<Pine.LNX.4.64.0910082058320.31576@srv.icf.org.ru>
Message-ID: <191c3a030910081012o63634e3bg1e510df510199541@mail.gmail.com>

hmm, not following that very well.
FS RTP stack only sends and receives and has no understanding of
encode/decode......

mod_opal uses OPAL rtp stack.



2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-08 19:53 +0300, Vlasis Hatzistavrou (KTI) wrote
> freeswitch-users...:
>
> VHK>Georgiewskiy Yuriy wrote:
> VHK>> On 2009-10-08 10:43 -0500, Anthony Minessale wrote
> freeswitch-users at lists.f...:
> VHK>>
> VHK>> AM>If you are going to make that alternate module are you going to
> host it in
> VHK>> AM>the FS tree along side mod_opal?
> VHK>>
> VHK>> Yes, but then it be useful, now i have working only signaling part
> and some
> VHK>> kinde of not working rtp part :)
> VHK>
> VHK>
> VHK>Hello Yuriy,
> VHK>
> VHK>Since FS already has its own RTP stack (for pass-through and
> VHK>transcoding) wouldn't it be best if your H323 implementation with OPAL
> VHK>was used for signaling only and rely on FS RTP for the media? Perhaps
> VHK>that would save you time and effort, too?
>
> it already use fs rtp stack, and my implementation based on h323plus, not
> on opal,
> but fs rtp stack not so complete, there need write some code to
> send/receive packets
> because fs stac only encode/decode packets, not send/receive, or i don't
> known how to
> do it with it. i don't find any example for this.
>
>
> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129-RIPE
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From bottleman at icf.org.ru  Thu Oct  8 10:22:43 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Thu, 8 Oct 2009 21:22:43 +0400 (MSD)
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <191c3a030910081012o63634e3bg1e510df510199541@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<4ACE1908.9040004@kinetix.gr>
	<Pine.LNX.4.64.0910082058320.31576@srv.icf.org.ru>
	<191c3a030910081012o63634e3bg1e510df510199541@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910082117120.757@srv.icf.org.ru>

On 2009-10-08 12:12 -0500, Anthony Minessale wrote freeswitch-users at lists.f...:

AM>hmm, not following that very well.
AM>FS RTP stack only sends and receives and has no understanding of
AM>encode/decode......

hmm, give me some example how to use fs stack, or link there i can read about this.

AM>
AM>mod_opal uses OPAL rtp stack.
AM>
AM>
AM>
AM>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
AM>
AM>> On 2009-10-08 19:53 +0300, Vlasis Hatzistavrou (KTI) wrote
AM>> freeswitch-users...:
AM>>
AM>> VHK>Georgiewskiy Yuriy wrote:
AM>> VHK>> On 2009-10-08 10:43 -0500, Anthony Minessale wrote
AM>> freeswitch-users at lists.f...:
AM>> VHK>>
AM>> VHK>> AM>If you are going to make that alternate module are you going to
AM>> host it in
AM>> VHK>> AM>the FS tree along side mod_opal?
AM>> VHK>>
AM>> VHK>> Yes, but then it be useful, now i have working only signaling part
AM>> and some
AM>> VHK>> kinde of not working rtp part :)
AM>> VHK>
AM>> VHK>
AM>> VHK>Hello Yuriy,
AM>> VHK>
AM>> VHK>Since FS already has its own RTP stack (for pass-through and
AM>> VHK>transcoding) wouldn't it be best if your H323 implementation with OPAL
AM>> VHK>was used for signaling only and rely on FS RTP for the media? Perhaps
AM>> VHK>that would save you time and effort, too?
AM>>
AM>> it already use fs rtp stack, and my implementation based on h323plus, not
AM>> on opal,
AM>> but fs rtp stack not so complete, there need write some code to
AM>> send/receive packets
AM>> because fs stac only encode/decode packets, not send/receive, or i don't
AM>> known how to
AM>> do it with it. i don't find any example for this.
AM>>
AM>>
AM>> C ?????????                       With Best Regards
AM>> ???????????? ????.                Georgiewskiy Yuriy
AM>> +7 4872 711666                    +7 4872 711666
AM>> ???? +7 4872 711143               fax +7 4872 711143
AM>> ???????? ??? "?? ?? ??????"       IT Service Ltd
AM>> http://nkoort.ru                  http://nkoort.ru
AM>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
AM>> YG129-RIPE                        YG129-RIPE
AM>>
AM>> _______________________________________________
AM>> FreeSWITCH-users mailing list
AM>> FreeSWITCH-users at lists.freeswitch.org
AM>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
AM>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
AM>> http://www.freeswitch.org
AM>>
AM>>
AM>
AM>
AM>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From tuyanozipek at gmail.com  Thu Oct  8 10:25:13 2009
From: tuyanozipek at gmail.com (=?ISO-8859-1?Q?Tuyan_=D6zipek?=)
Date: Thu, 8 Oct 2009 13:25:13 -0400
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
Message-ID: <881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>

Hi,

2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>:
> On 2009-10-08 10:43 -0500, Anthony Minessale wrote freeswitch-users at lists.f...:
>
> AM>If you are going to make that alternate module are you going to host it in
> AM>the FS tree along side mod_opal?
>
> Yes, but then it be useful, now i have working only signaling part and some
> kinde of not working rtp part :)

If you dont use fs rtp stack, its unlikely that it will be accepted
into the tree.

>
> AM>also if were working on mod_opal why did you not try to involve us and the
> AM>opal team?
>
> Because i made patches for libopal, one is a bugfix in rtp part, there is a race condition
> in inicialisation in jitter buffer, another patch implements method to send progress indicator,
> and i don't wont spent my time to incorporate this changes into libopal.

Thats bad.
Any bugfixes from fs, goes to upstream on any of the used libraries.
You should be doing the same.
And Opal developers, will either include or refuse your patches. If
they refuse it, they will give you the reason.

> without this changes
> my work on mod_opal in freeswitch don't useful at all, i provide link to my work with all
> patches, if somebody wont incorporate it in libopal tree and fs - go on, but i think
> better and more elegant make new module based on h323plus.

If you dont publish your changes, all those you are trying to achieve,
wont happen.

>
> AM>How far away from what is in tree are these patches you have?
> AM>
> AM>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
> AM>
> AM>> On 2009-10-07 15:09 -0500, Brian West wrote
> AM>> freeswitch-users at lists.freeswit...:
> AM>>
> AM>> opal have addition abstraction layer called opalmgr, and it implementation
> AM>> is not so good in
> AM>> this case, for example to implemet pre_answer in mod_opal i need patch
> AM>> libopal, because

The patch you have in there, adds a method to the OpalCall,  it does
not touch any parts of OpalManager
so, i dont understand why opalmanager would be the cause of your pain?

> AM>> there is no way to send progress inicator throuch opalmgr. and there is
> AM>> many another issues like
> AM>> this in that layer.

Please point me to the issues you have in opal, their bug reports , traces etc.
I dont think any of the opal people has psychic abilities to detect
-your- problems
and solve them.

>ftp://srv.icf.org.ru/pub/soft/f/freeswitch/ - there is
> AM>> my work on mod_opal before
> AM>> i start moving to h323plus, may be this help somebody there.
> AM>>
> AM>> BW> From what I have been told h323plus is a based/fork of OpenH323 which
> AM>> BW>OPAL is just a continuation of OpenH323. ?So why not support the
> AM>> BW>developers of OPAL/OpenH323 ?
> AM>> BW>
> AM>> BW>/b
> AM>> BW>
> AM>> BW>On Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote:
> AM>> BW>
> AM>> BW>> We are developing module to handle h323 proto now, we try to use
> AM>> BW>> mod_opal and try improve it, but no luck,
> AM>> BW>> there is many issues in libopal, and finaly we now move to h323plus
> AM>> BW>> library.

Did any of you try to report those issues?

Regards

/tyn

> AM>> BW>
> AM>> BW>
> AM>> BW>_______________________________________________
> AM>> BW>FreeSWITCH-users mailing list
> AM>> BW>FreeSWITCH-users at lists.freeswitch.org
> AM>> BW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> AM>> BW>UNSUBSCRIBE:
> AM>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> AM>> BW>http://www.freeswitch.org
> AM>> BW>
> AM>>
> AM>> C ????????? ? ? ? ? ? ? ? ? ? ? ? With Best Regards
> AM>> ???????????? ????. ? ? ? ? ? ? ? ?Georgiewskiy Yuriy
> AM>> +7 4872 711666 ? ? ? ? ? ? ? ? ? ?+7 4872 711666
> AM>> ???? +7 4872 711143 ? ? ? ? ? ? ? fax +7 4872 711143
> AM>> ???????? ??? "?? ?? ??????" ? ? ? IT Service Ltd
> AM>> http://nkoort.ru ? ? ? ? ? ? ? ? ?http://nkoort.ru
> AM>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> AM>> YG129-RIPE ? ? ? ? ? ? ? ? ? ? ? ?YG129-RIPE
> AM>>
> AM>> _______________________________________________
> AM>> FreeSWITCH-users mailing list
> AM>> FreeSWITCH-users at lists.freeswitch.org
> AM>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> AM>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> AM>> http://www.freeswitch.org
> AM>>
> AM>>
> AM>
> AM>
> AM>
>
> C ????????? ? ? ? ? ? ? ? ? ? ? ? With Best Regards
> ???????????? ????. ? ? ? ? ? ? ? ?Georgiewskiy Yuriy
> +7 4872 711666 ? ? ? ? ? ? ? ? ? ?+7 4872 711666
> ???? +7 4872 711143 ? ? ? ? ? ? ? fax +7 4872 711143
> ???????? ??? "?? ?? ??????" ? ? ? IT Service Ltd
> http://nkoort.ru ? ? ? ? ? ? ? ? ?http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE ? ? ? ? ? ? ? ? ? ? ? ?YG129-RIPE
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
>



From bottleman at icf.org.ru  Thu Oct  8 10:41:09 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Thu, 8 Oct 2009 21:41:09 +0400 (MSD)
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>

On 2009-10-08 13:25 -0400, Tuyan ?zipek wrote freeswitch-users at lists.freesw...:

Tz>Hi,
Tz>
Tz>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>:
Tz>> On 2009-10-08 10:43 -0500, Anthony Minessale wrote freeswitch-users at lists.f...:
Tz>>
Tz>> AM>If you are going to make that alternate module are you going to host it in
Tz>> AM>the FS tree along side mod_opal?
Tz>>
Tz>> Yes, but then it be useful, now i have working only signaling part and some
Tz>> kinde of not working rtp part :)
Tz>
Tz>If you dont use fs rtp stack, its unlikely that it will be accepted
Tz>into the tree.
Tz>
Tz>>
Tz>> AM>also if were working on mod_opal why did you not try to involve us and the
Tz>> AM>opal team?
Tz>>
Tz>> Because i made patches for libopal, one is a bugfix in rtp part, there is a race condition
Tz>> in inicialisation in jitter buffer, another patch implements method to send progress indicator,
Tz>> and i don't wont spent my time to incorporate this changes into libopal.
Tz>
Tz>Thats bad.
Tz>Any bugfixes from fs, goes to upstream on any of the used libraries.
Tz>You should be doing the same.
Tz>And Opal developers, will either include or refuse your patches. If
Tz>they refuse it, they will give you the reason.

i make this fix only to freeze my current mod_opal work on working state, while it now work for me i work on 
my new implementation of h323 proto for fs, i think opal developers will fix this rtp bug himself becouse 
it crashes and make library unuseful.

Tz>
Tz>> without this changes
Tz>> my work on mod_opal in freeswitch don't useful at all, i provide link to my work with all
Tz>> patches, if somebody wont incorporate it in libopal tree and fs - go on, but i think
Tz>> better and more elegant make new module based on h323plus.
Tz>
Tz>If you dont publish your changes, all those you are trying to achieve,
Tz>wont happen.
Tz>
Tz>>
Tz>> AM>How far away from what is in tree are these patches you have?
Tz>> AM>
Tz>> AM>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
Tz>> AM>
Tz>> AM>> On 2009-10-07 15:09 -0500, Brian West wrote
Tz>> AM>> freeswitch-users at lists.freeswit...:
Tz>> AM>>
Tz>> AM>> opal have addition abstraction layer called opalmgr, and it implementation
Tz>> AM>> is not so good in
Tz>> AM>> this case, for example to implemet pre_answer in mod_opal i need patch
Tz>> AM>> libopal, because
Tz>
Tz>The patch you have in there, adds a method to the OpalCall,  it does
Tz>not touch any parts of OpalManager
Tz>so, i dont understand why opalmanager would be the cause of your pain?
Tz>
Tz>> AM>> there is no way to send progress inicator throuch opalmgr. and there is
Tz>> AM>> many another issues like
Tz>> AM>> this in that layer.
Tz>
Tz>Please point me to the issues you have in opal, their bug reports , traces etc.
Tz>I dont think any of the opal people has psychic abilities to detect
Tz>-your- problems
Tz>and solve them.
Tz>
Tz>>ftp://srv.icf.org.ru/pub/soft/f/freeswitch/ - there is
Tz>> AM>> my work on mod_opal before
Tz>> AM>> i start moving to h323plus, may be this help somebody there.
Tz>> AM>>
Tz>> AM>> BW> From what I have been told h323plus is a based/fork of OpenH323 which
Tz>> AM>> BW>OPAL is just a continuation of OpenH323. ?So why not support the
Tz>> AM>> BW>developers of OPAL/OpenH323 ?
Tz>> AM>> BW>
Tz>> AM>> BW>/b
Tz>> AM>> BW>
Tz>> AM>> BW>On Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote:
Tz>> AM>> BW>
Tz>> AM>> BW>> We are developing module to handle h323 proto now, we try to use
Tz>> AM>> BW>> mod_opal and try improve it, but no luck,
Tz>> AM>> BW>> there is many issues in libopal, and finaly we now move to h323plus
Tz>> AM>> BW>> library.
Tz>
Tz>Did any of you try to report those issues?
Tz>
Tz>Regards
Tz>
Tz>/tyn
Tz>
Tz>> AM>> BW>
Tz>> AM>> BW>
Tz>> AM>> BW>_______________________________________________
Tz>> AM>> BW>FreeSWITCH-users mailing list
Tz>> AM>> BW>FreeSWITCH-users at lists.freeswitch.org
Tz>> AM>> BW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Tz>> AM>> BW>UNSUBSCRIBE:
Tz>> AM>> http://lists.freeswitch.org/mailman/options/freeswitch-users
Tz>> AM>> BW>http://www.freeswitch.org
Tz>> AM>> BW>
Tz>> AM>>
Tz>> AM>> C ????????? ? ? ? ? ? ? ? ? ? ? ? With Best Regards
Tz>> AM>> ???????????? ????. ? ? ? ? ? ? ? ?Georgiewskiy Yuriy
Tz>> AM>> +7 4872 711666 ? ? ? ? ? ? ? ? ? ?+7 4872 711666
Tz>> AM>> ???? +7 4872 711143 ? ? ? ? ? ? ? fax +7 4872 711143
Tz>> AM>> ???????? ??? "?? ?? ??????" ? ? ? IT Service Ltd
Tz>> AM>> http://nkoort.ru ? ? ? ? ? ? ? ? ?http://nkoort.ru
Tz>> AM>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
Tz>> AM>> YG129-RIPE ? ? ? ? ? ? ? ? ? ? ? ?YG129-RIPE
Tz>> AM>>
Tz>> AM>> _______________________________________________
Tz>> AM>> FreeSWITCH-users mailing list
Tz>> AM>> FreeSWITCH-users at lists.freeswitch.org
Tz>> AM>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Tz>> AM>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Tz>> AM>> http://www.freeswitch.org
Tz>> AM>>
Tz>> AM>>
Tz>> AM>
Tz>> AM>
Tz>> AM>
Tz>>
Tz>> C ????????? ? ? ? ? ? ? ? ? ? ? ? With Best Regards
Tz>> ???????????? ????. ? ? ? ? ? ? ? ?Georgiewskiy Yuriy
Tz>> +7 4872 711666 ? ? ? ? ? ? ? ? ? ?+7 4872 711666
Tz>> ???? +7 4872 711143 ? ? ? ? ? ? ? fax +7 4872 711143
Tz>> ???????? ??? "?? ?? ??????" ? ? ? IT Service Ltd
Tz>> http://nkoort.ru ? ? ? ? ? ? ? ? ?http://nkoort.ru
Tz>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
Tz>> YG129-RIPE ? ? ? ? ? ? ? ? ? ? ? ?YG129-RIPE
Tz>> _______________________________________________
Tz>> FreeSWITCH-users mailing list
Tz>> FreeSWITCH-users at lists.freeswitch.org
Tz>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Tz>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Tz>> http://www.freeswitch.org
Tz>>
Tz>>
Tz>
Tz>_______________________________________________
Tz>FreeSWITCH-users mailing list
Tz>FreeSWITCH-users at lists.freeswitch.org
Tz>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Tz>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Tz>http://www.freeswitch.org
Tz>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From tculjaga at gmail.com  Thu Oct  8 11:32:40 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 8 Oct 2009 20:32:40 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
Message-ID: <65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>

Hi Yuriy,

can you share what you have so far, I'm sure we can help with RTP part...

T.

2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-08 13:25 -0400, Tuyan ?zipek wrote
> freeswitch-users at lists.freesw...:
>
> Tz>Hi,
> Tz>
> Tz>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>:
> Tz>> On 2009-10-08 10:43 -0500, Anthony Minessale wrote
> freeswitch-users at lists.f...:
> Tz>>
> Tz>> AM>If you are going to make that alternate module are you going to
> host it in
> Tz>> AM>the FS tree along side mod_opal?
> Tz>>
> Tz>> Yes, but then it be useful, now i have working only signaling part and
> some
> Tz>> kinde of not working rtp part :)
> Tz>
> Tz>If you dont use fs rtp stack, its unlikely that it will be accepted
> Tz>into the tree.
> Tz>
> Tz>>
> Tz>> AM>also if were working on mod_opal why did you not try to involve us
> and the
> Tz>> AM>opal team?
> Tz>>
> Tz>> Because i made patches for libopal, one is a bugfix in rtp part, there
> is a race condition
> Tz>> in inicialisation in jitter buffer, another patch implements method to
> send progress indicator,
> Tz>> and i don't wont spent my time to incorporate this changes into
> libopal.
> Tz>
> Tz>Thats bad.
> Tz>Any bugfixes from fs, goes to upstream on any of the used libraries.
> Tz>You should be doing the same.
> Tz>And Opal developers, will either include or refuse your patches. If
> Tz>they refuse it, they will give you the reason.
>
> i make this fix only to freeze my current mod_opal work on working state,
> while it now work for me i work on
> my new implementation of h323 proto for fs, i think opal developers will
> fix this rtp bug himself becouse
> it crashes and make library unuseful.
>
> Tz>
> Tz>> without this changes
> Tz>> my work on mod_opal in freeswitch don't useful at all, i provide link
> to my work with all
> Tz>> patches, if somebody wont incorporate it in libopal tree and fs - go
> on, but i think
> Tz>> better and more elegant make new module based on h323plus.
> Tz>
> Tz>If you dont publish your changes, all those you are trying to achieve,
> Tz>wont happen.
> Tz>
> Tz>>
> Tz>> AM>How far away from what is in tree are these patches you have?
> Tz>> AM>
> Tz>> AM>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
> Tz>> AM>
> Tz>> AM>> On 2009-10-07 15:09 -0500, Brian West wrote
> Tz>> AM>> freeswitch-users at lists.freeswit...:
> Tz>> AM>>
> Tz>> AM>> opal have addition abstraction layer called opalmgr, and it
> implementation
> Tz>> AM>> is not so good in
> Tz>> AM>> this case, for example to implemet pre_answer in mod_opal i need
> patch
> Tz>> AM>> libopal, because
> Tz>
> Tz>The patch you have in there, adds a method to the OpalCall,  it does
> Tz>not touch any parts of OpalManager
> Tz>so, i dont understand why opalmanager would be the cause of your pain?
> Tz>
> Tz>> AM>> there is no way to send progress inicator throuch opalmgr. and
> there is
> Tz>> AM>> many another issues like
> Tz>> AM>> this in that layer.
> Tz>
> Tz>Please point me to the issues you have in opal, their bug reports ,
> traces etc.
> Tz>I dont think any of the opal people has psychic abilities to detect
> Tz>-your- problems
> Tz>and solve them.
> Tz>
> Tz>>ftp://srv.icf.org.ru/pub/soft/f/freeswitch/ - there is
> Tz>> AM>> my work on mod_opal before
> Tz>> AM>> i start moving to h323plus, may be this help somebody there.
> Tz>> AM>>
> Tz>> AM>> BW> From what I have been told h323plus is a based/fork of
> OpenH323 which
> Tz>> AM>> BW>OPAL is just a continuation of OpenH323.  So why not support
> the
> Tz>> AM>> BW>developers of OPAL/OpenH323 ?
> Tz>> AM>> BW>
> Tz>> AM>> BW>/b
> Tz>> AM>> BW>
> Tz>> AM>> BW>On Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote:
> Tz>> AM>> BW>
> Tz>> AM>> BW>> We are developing module to handle h323 proto now, we try to
> use
> Tz>> AM>> BW>> mod_opal and try improve it, but no luck,
> Tz>> AM>> BW>> there is many issues in libopal, and finaly we now move to
> h323plus
> Tz>> AM>> BW>> library.
> Tz>
> Tz>Did any of you try to report those issues?
> Tz>
> Tz>Regards
> Tz>
> Tz>/tyn
> Tz>
> Tz>> AM>> BW>
> Tz>> AM>> BW>
> Tz>> AM>> BW>_______________________________________________
> Tz>> AM>> BW>FreeSWITCH-users mailing list
> Tz>> AM>> BW>FreeSWITCH-users at lists.freeswitch.org
> Tz>> AM>> BW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> Tz>> AM>> BW>UNSUBSCRIBE:
> Tz>> AM>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> Tz>> AM>> BW>http://www.freeswitch.org
> Tz>> AM>> BW>
> Tz>> AM>>
> Tz>> AM>> C ?????????                       With Best Regards
> Tz>> AM>> ???????????? ????.                Georgiewskiy Yuriy
> Tz>> AM>> +7 4872 711666                    +7 4872 711666
> Tz>> AM>> ???? +7 4872 711143               fax +7 4872 711143
> Tz>> AM>> ???????? ??? "?? ?? ??????"       IT Service Ltd
> Tz>> AM>> http://nkoort.ru                  http://nkoort.ru
> Tz>> AM>> JID: GHhost at jabber.tula-ix.net.ru JID:
> GHhost at jabber.tula-ix.net.ru
> Tz>> AM>> YG129-RIPE                        YG129-RIPE
> Tz>> AM>>
> Tz>> AM>> _______________________________________________
> Tz>> AM>> FreeSWITCH-users mailing list
> Tz>> AM>> FreeSWITCH-users at lists.freeswitch.org
> Tz>> AM>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> Tz>> AM>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> Tz>> AM>> http://www.freeswitch.org
> Tz>> AM>>
> Tz>> AM>>
> Tz>> AM>
> Tz>> AM>
> Tz>> AM>
> Tz>>
> Tz>> C ?????????                       With Best Regards
> Tz>> ???????????? ????.                Georgiewskiy Yuriy
> Tz>> +7 4872 711666                    +7 4872 711666
> Tz>> ???? +7 4872 711143               fax +7 4872 711143
> Tz>> ???????? ??? "?? ?? ??????"       IT Service Ltd
> Tz>> http://nkoort.ru                  http://nkoort.ru
> Tz>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> Tz>> YG129-RIPE                        YG129-RIPE
> Tz>> _______________________________________________
> Tz>> FreeSWITCH-users mailing list
> Tz>> FreeSWITCH-users at lists.freeswitch.org
> Tz>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> Tz>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> Tz>> http://www.freeswitch.org
> Tz>>
> Tz>>
> Tz>
> Tz>_______________________________________________
> Tz>FreeSWITCH-users mailing list
> Tz>FreeSWITCH-users at lists.freeswitch.org
> Tz>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> Tz>UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> Tz>http://www.freeswitch.org
> Tz>
>
> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129-RIPE
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Oct  8 11:38:37 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Oct 2009 11:38:37 -0700
Subject: [Freeswitch-users] Problem with gateway registration
In-Reply-To: <1b46b4e80910070743h38d39e5ax362ff7944754067e@mail.gmail.com>
References: <1b46b4e80909291105m385fb287m937667f9b51beaf@mail.gmail.com>
	<87f2f3b90910050920w4d17033fl4c386a0018107bdf@mail.gmail.com>
	<1b46b4e80910052119me12a0d9y4653edd831f9555e@mail.gmail.com>
	<1b46b4e80910052142jaa09f08l72e5e4b2b24d10fc@mail.gmail.com>
	<083DBCB9-48B2-4E61-81B5-B7D00A42EAB5@freeswitch.org>
	<1b46b4e80910061422q7be2a21fg5d2caaef7e307abb@mail.gmail.com>
	<9106C8E1-498F-4AE8-8D34-7F6AAD4DC8AE@freeswitch.org>
	<1b46b4e80910070648n622c3f9haacc22c2328b44ff@mail.gmail.com>
	<8750A70D-6891-4C07-B4DF-C7F4ABA58D67@freeswitch.org>
	<1b46b4e80910070743h38d39e5ax362ff7944754067e@mail.gmail.com>
Message-ID: <87f2f3b90910081138i32d1c28dmf229344f4b5a4e84@mail.gmail.com>

On Wed, Oct 7, 2009 at 7:43 AM, Nicolas Brenner <nicolas at medularis.com>wrote:

> You are missing the point, it is only rejecting auth for FS, Asterisk and
> X-Lite work fine with the same config for that gateway.
>
> Did you figure this one out yet? If not, snag some SIP captures of working
vs. non-working REGISTRATIONs and pastebin them along with your failed reg
and your redacted gateway config.
-MC


>
>
> On Wed, Oct 7, 2009 at 10:20 AM, Brian West <brian at freeswitch.org> wrote:
>
>> I would suspect its a PEBKAC.  I mean if you could register to a
>> gateway that rejected auth... what purpose would auth serve in the
>> first place?
>>
>> /b
>>
>> On Oct 7, 2009, at 8:48 AM, Nicolas Brenner wrote:
>>
>> > Is there some way to make FS register with the gateway that is
>> > rejecting the authentication? is it FS or the SIP server at fault?
>> > Why would X-Lite work and FS not?
>> >
>> > Thanks again for your time and help.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bottleman at icf.org.ru  Thu Oct  8 11:44:38 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Thu, 8 Oct 2009 22:44:38 +0400 (MSD)
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910082242570.16063@srv.icf.org.ru>

On 2009-10-08 20:32 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>Hi Yuriy,
TC>
TC>can you share what you have so far, I'm sure we can help with RTP part...

I think there is a few days and i make it work, after this i start to test and share it.

TC>
TC>T.
TC>
TC>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
TC>
TC>> On 2009-10-08 13:25 -0400, Tuyan ?zipek wrote
TC>> freeswitch-users at lists.freesw...:
TC>>
TC>> Tz>Hi,
TC>> Tz>
TC>> Tz>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>:
TC>> Tz>> On 2009-10-08 10:43 -0500, Anthony Minessale wrote
TC>> freeswitch-users at lists.f...:
TC>> Tz>>
TC>> Tz>> AM>If you are going to make that alternate module are you going to
TC>> host it in
TC>> Tz>> AM>the FS tree along side mod_opal?
TC>> Tz>>
TC>> Tz>> Yes, but then it be useful, now i have working only signaling part and
TC>> some
TC>> Tz>> kinde of not working rtp part :)
TC>> Tz>
TC>> Tz>If you dont use fs rtp stack, its unlikely that it will be accepted
TC>> Tz>into the tree.
TC>> Tz>
TC>> Tz>>
TC>> Tz>> AM>also if were working on mod_opal why did you not try to involve us
TC>> and the
TC>> Tz>> AM>opal team?
TC>> Tz>>
TC>> Tz>> Because i made patches for libopal, one is a bugfix in rtp part, there
TC>> is a race condition
TC>> Tz>> in inicialisation in jitter buffer, another patch implements method to
TC>> send progress indicator,
TC>> Tz>> and i don't wont spent my time to incorporate this changes into
TC>> libopal.
TC>> Tz>
TC>> Tz>Thats bad.
TC>> Tz>Any bugfixes from fs, goes to upstream on any of the used libraries.
TC>> Tz>You should be doing the same.
TC>> Tz>And Opal developers, will either include or refuse your patches. If
TC>> Tz>they refuse it, they will give you the reason.
TC>>
TC>> i make this fix only to freeze my current mod_opal work on working state,
TC>> while it now work for me i work on
TC>> my new implementation of h323 proto for fs, i think opal developers will
TC>> fix this rtp bug himself becouse
TC>> it crashes and make library unuseful.
TC>>
TC>> Tz>
TC>> Tz>> without this changes
TC>> Tz>> my work on mod_opal in freeswitch don't useful at all, i provide link
TC>> to my work with all
TC>> Tz>> patches, if somebody wont incorporate it in libopal tree and fs - go
TC>> on, but i think
TC>> Tz>> better and more elegant make new module based on h323plus.
TC>> Tz>
TC>> Tz>If you dont publish your changes, all those you are trying to achieve,
TC>> Tz>wont happen.
TC>> Tz>
TC>> Tz>>
TC>> Tz>> AM>How far away from what is in tree are these patches you have?
TC>> Tz>> AM>
TC>> Tz>> AM>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
TC>> Tz>> AM>
TC>> Tz>> AM>> On 2009-10-07 15:09 -0500, Brian West wrote
TC>> Tz>> AM>> freeswitch-users at lists.freeswit...:
TC>> Tz>> AM>>
TC>> Tz>> AM>> opal have addition abstraction layer called opalmgr, and it
TC>> implementation
TC>> Tz>> AM>> is not so good in
TC>> Tz>> AM>> this case, for example to implemet pre_answer in mod_opal i need
TC>> patch
TC>> Tz>> AM>> libopal, because
TC>> Tz>
TC>> Tz>The patch you have in there, adds a method to the OpalCall,  it does
TC>> Tz>not touch any parts of OpalManager
TC>> Tz>so, i dont understand why opalmanager would be the cause of your pain?
TC>> Tz>
TC>> Tz>> AM>> there is no way to send progress inicator throuch opalmgr. and
TC>> there is
TC>> Tz>> AM>> many another issues like
TC>> Tz>> AM>> this in that layer.
TC>> Tz>
TC>> Tz>Please point me to the issues you have in opal, their bug reports ,
TC>> traces etc.
TC>> Tz>I dont think any of the opal people has psychic abilities to detect
TC>> Tz>-your- problems
TC>> Tz>and solve them.
TC>> Tz>
TC>> Tz>>ftp://srv.icf.org.ru/pub/soft/f/freeswitch/ - there is
TC>> Tz>> AM>> my work on mod_opal before
TC>> Tz>> AM>> i start moving to h323plus, may be this help somebody there.
TC>> Tz>> AM>>
TC>> Tz>> AM>> BW> From what I have been told h323plus is a based/fork of
TC>> OpenH323 which
TC>> Tz>> AM>> BW>OPAL is just a continuation of OpenH323.  So why not support
TC>> the
TC>> Tz>> AM>> BW>developers of OPAL/OpenH323 ?
TC>> Tz>> AM>> BW>
TC>> Tz>> AM>> BW>/b
TC>> Tz>> AM>> BW>
TC>> Tz>> AM>> BW>On Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote:
TC>> Tz>> AM>> BW>
TC>> Tz>> AM>> BW>> We are developing module to handle h323 proto now, we try to
TC>> use
TC>> Tz>> AM>> BW>> mod_opal and try improve it, but no luck,
TC>> Tz>> AM>> BW>> there is many issues in libopal, and finaly we now move to
TC>> h323plus
TC>> Tz>> AM>> BW>> library.
TC>> Tz>
TC>> Tz>Did any of you try to report those issues?
TC>> Tz>
TC>> Tz>Regards
TC>> Tz>
TC>> Tz>/tyn
TC>> Tz>
TC>> Tz>> AM>> BW>
TC>> Tz>> AM>> BW>
TC>> Tz>> AM>> BW>_______________________________________________
TC>> Tz>> AM>> BW>FreeSWITCH-users mailing list
TC>> Tz>> AM>> BW>FreeSWITCH-users at lists.freeswitch.org
TC>> Tz>> AM>> BW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> Tz>> AM>> BW>UNSUBSCRIBE:
TC>> Tz>> AM>> http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> Tz>> AM>> BW>http://www.freeswitch.org
TC>> Tz>> AM>> BW>
TC>> Tz>> AM>>
TC>> Tz>> AM>> C ?????????                       With Best Regards
TC>> Tz>> AM>> ???????????? ????.                Georgiewskiy Yuriy
TC>> Tz>> AM>> +7 4872 711666                    +7 4872 711666
TC>> Tz>> AM>> ???? +7 4872 711143               fax +7 4872 711143
TC>> Tz>> AM>> ???????? ??? "?? ?? ??????"       IT Service Ltd
TC>> Tz>> AM>> http://nkoort.ru                  http://nkoort.ru
TC>> Tz>> AM>> JID: GHhost at jabber.tula-ix.net.ru JID:
TC>> GHhost at jabber.tula-ix.net.ru
TC>> Tz>> AM>> YG129-RIPE                        YG129-RIPE
TC>> Tz>> AM>>
TC>> Tz>> AM>> _______________________________________________
TC>> Tz>> AM>> FreeSWITCH-users mailing list
TC>> Tz>> AM>> FreeSWITCH-users at lists.freeswitch.org
TC>> Tz>> AM>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> Tz>> AM>> UNSUBSCRIBE:
TC>> http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> Tz>> AM>> http://www.freeswitch.org
TC>> Tz>> AM>>
TC>> Tz>> AM>>
TC>> Tz>> AM>
TC>> Tz>> AM>
TC>> Tz>> AM>
TC>> Tz>>
TC>> Tz>> C ?????????                       With Best Regards
TC>> Tz>> ???????????? ????.                Georgiewskiy Yuriy
TC>> Tz>> +7 4872 711666                    +7 4872 711666
TC>> Tz>> ???? +7 4872 711143               fax +7 4872 711143
TC>> Tz>> ???????? ??? "?? ?? ??????"       IT Service Ltd
TC>> Tz>> http://nkoort.ru                  http://nkoort.ru
TC>> Tz>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
TC>> Tz>> YG129-RIPE                        YG129-RIPE
TC>> Tz>> _______________________________________________
TC>> Tz>> FreeSWITCH-users mailing list
TC>> Tz>> FreeSWITCH-users at lists.freeswitch.org
TC>> Tz>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> Tz>> UNSUBSCRIBE:
TC>> http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> Tz>> http://www.freeswitch.org
TC>> Tz>>
TC>> Tz>>
TC>> Tz>
TC>> Tz>_______________________________________________
TC>> Tz>FreeSWITCH-users mailing list
TC>> Tz>FreeSWITCH-users at lists.freeswitch.org
TC>> Tz>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> Tz>UNSUBSCRIBE:
TC>> http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> Tz>http://www.freeswitch.org
TC>> Tz>
TC>>
TC>> C ?????????                       With Best Regards
TC>> ???????????? ????.                Georgiewskiy Yuriy
TC>> +7 4872 711666                    +7 4872 711666
TC>> ???? +7 4872 711143               fax +7 4872 711143
TC>> ???????? ??? "?? ?? ??????"       IT Service Ltd
TC>> http://nkoort.ru                  http://nkoort.ru
TC>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
TC>> YG129-RIPE                        YG129-RIPE
TC>>
TC>> _______________________________________________
TC>> FreeSWITCH-users mailing list
TC>> FreeSWITCH-users at lists.freeswitch.org
TC>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> http://www.freeswitch.org
TC>>
TC>>
TC>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From gabe at gundy.org  Thu Oct  8 12:01:35 2009
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 8 Oct 2009 13:01:35 -0600
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <87f2f3b90910080958i1f5cfccax621fd12100a05e3e@mail.gmail.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<87f2f3b90910080958i1f5cfccax621fd12100a05e3e@mail.gmail.com>
Message-ID: <903da5680910081201p427a8358n768edb61a96674a4@mail.gmail.com>

Naturally the wireless is jacked at the conf.  It's no match for the
100's of geeks sporting laptops.  Thanks for the link. Looks like
we'll be shooting from the hip. :)

>From my phone,
Gabe

On Thursday, October 8, 2009, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Thu, Oct 8, 2009 at 7:42 AM, Gabriel Gunderson <gabe at gundy.org?<javascript:_e({}, 'cvml', 'gabe at gundy.org');>> wrote:
>
> Hey all,
>
> The 3rd annual UTOSC (Utah Open Source Conference) [1] starts today.
> It's a conference that gives Open Source users/hackers a chance gather
> together to mingle and share. ?It's a lot of fun. ?Anyway, this year
> I've offered to work with a friend of mine to lead a BoF (Birds of a
> Feather) discussion on Open Source Telephony [2].
>
> While it should mostly be group discussion, you never know when people
> are going to get shy and clam up. ?*If* they do, it would be nice to
> have some slides to fill that time. ?Rather than make some from
> scratch (that are unlikely to get used), I thought I'd ask here and
> see if anyone has a FS slide deck that they wouldn't mind sharing.
>
> Thanks,
> Gabe
>
>
> 1) http://2009.utosc.com/pages/home/
> 2) http://2009.utosc.com/presentation/136/
>
> There are numerous slides and such from ClueCon. Go to http://www.cluecon.com and you can download both the video files as well as the actual presentations. The presentation files are in various formats such as PPT(X), PDF, and Keynote.
> The second presentation on day 1 is Anthony Minessale and he talks a lot about expanding one's thinking when it comes to applying FreeSWITCH in production. For example, some people will jump through crazy hoops to solve a problem "because that's how they've always done it." Anyway, check out that video and presentation file for some great ideas.
> -MC
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From msc at freeswitch.org  Thu Oct  8 12:02:43 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Oct 2009 12:02:43 -0700
Subject: [Freeswitch-users] REMINDER: Weekly FreeSWITCH Conference Scheduled
	for October 9, 11AM 	CST (GMT -6)
Message-ID: <87f2f3b90910081202m5dd3f567v240541dd2009c9a8@mail.gmail.com>

Just a friendly reminder that we will be having the weekly FreeSWITCH
conference call tomorrow, October 9th, at 11AM Central. The agenda is
updated:
http://bit.ly/lzEYy

The conference lasts for six hours, so feel free to dial in at any time. We
usually start going over the agenda about 15 minutes after the conference
starts. The agenda itself only takes about an hour or two, after which we
spend most of the time discussing whatever is on the minds of the community
members. Remember, the core FreeSWITCH development team of Anthony Minessale
(primary author, lead architect), Mike Jerris (build master), and Brian West
(all around VoIP expert, FS configuration guru) are present for the whole
conference. This is a great time to ask questions about the project. Please
join us!

Call in options include SIP, Skype, PSTN, and Jabber. Please see the agenda
for more information.

Looking forward to speaking with you all tomorrow,
Michael
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From tculjaga at gmail.com  Thu Oct  8 12:03:36 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 8 Oct 2009 21:03:36 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910082242570.16063@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910082242570.16063@srv.icf.org.ru>
Message-ID: <65d96fc80910081203x5a8dfc73n57ebf78cc532b920@mail.gmail.com>

k

2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-08 20:32 +0200, Tihomir Culjaga wrote
> freeswitch-users at lists.fre...:
>
> TC>Hi Yuriy,
> TC>
> TC>can you share what you have so far, I'm sure we can help with RTP
> part...
>
> I think there is a few days and i make it work, after this i start to test
> and share it.
>
> TC>
> TC>T.
> TC>
> TC>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
> TC>
> TC>> On 2009-10-08 13:25 -0400, Tuyan ?zipek wrote
> TC>> freeswitch-users at lists.freesw...:
> TC>>
> TC>> Tz>Hi,
> TC>> Tz>
> TC>> Tz>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>:
> TC>> Tz>> On 2009-10-08 10:43 -0500, Anthony Minessale wrote
> TC>> freeswitch-users at lists.f...:
> TC>> Tz>>
> TC>> Tz>> AM>If you are going to make that alternate module are you going
> to
> TC>> host it in
> TC>> Tz>> AM>the FS tree along side mod_opal?
> TC>> Tz>>
> TC>> Tz>> Yes, but then it be useful, now i have working only signaling
> part and
> TC>> some
> TC>> Tz>> kinde of not working rtp part :)
> TC>> Tz>
> TC>> Tz>If you dont use fs rtp stack, its unlikely that it will be accepted
> TC>> Tz>into the tree.
> TC>> Tz>
> TC>> Tz>>
> TC>> Tz>> AM>also if were working on mod_opal why did you not try to
> involve us
> TC>> and the
> TC>> Tz>> AM>opal team?
> TC>> Tz>>
> TC>> Tz>> Because i made patches for libopal, one is a bugfix in rtp part,
> there
> TC>> is a race condition
> TC>> Tz>> in inicialisation in jitter buffer, another patch implements
> method to
> TC>> send progress indicator,
> TC>> Tz>> and i don't wont spent my time to incorporate this changes into
> TC>> libopal.
> TC>> Tz>
> TC>> Tz>Thats bad.
> TC>> Tz>Any bugfixes from fs, goes to upstream on any of the used
> libraries.
> TC>> Tz>You should be doing the same.
> TC>> Tz>And Opal developers, will either include or refuse your patches. If
> TC>> Tz>they refuse it, they will give you the reason.
> TC>>
> TC>> i make this fix only to freeze my current mod_opal work on working
> state,
> TC>> while it now work for me i work on
> TC>> my new implementation of h323 proto for fs, i think opal developers
> will
> TC>> fix this rtp bug himself becouse
> TC>> it crashes and make library unuseful.
> TC>>
> TC>> Tz>
> TC>> Tz>> without this changes
> TC>> Tz>> my work on mod_opal in freeswitch don't useful at all, i provide
> link
> TC>> to my work with all
> TC>> Tz>> patches, if somebody wont incorporate it in libopal tree and fs -
> go
> TC>> on, but i think
> TC>> Tz>> better and more elegant make new module based on h323plus.
> TC>> Tz>
> TC>> Tz>If you dont publish your changes, all those you are trying to
> achieve,
> TC>> Tz>wont happen.
> TC>> Tz>
> TC>> Tz>>
> TC>> Tz>> AM>How far away from what is in tree are these patches you have?
> TC>> Tz>> AM>
> TC>> Tz>> AM>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
> TC>> Tz>> AM>
> TC>> Tz>> AM>> On 2009-10-07 15:09 -0500, Brian West wrote
> TC>> Tz>> AM>> freeswitch-users at lists.freeswit...:
> TC>> Tz>> AM>>
> TC>> Tz>> AM>> opal have addition abstraction layer called opalmgr, and it
> TC>> implementation
> TC>> Tz>> AM>> is not so good in
> TC>> Tz>> AM>> this case, for example to implemet pre_answer in mod_opal i
> need
> TC>> patch
> TC>> Tz>> AM>> libopal, because
> TC>> Tz>
> TC>> Tz>The patch you have in there, adds a method to the OpalCall,  it
> does
> TC>> Tz>not touch any parts of OpalManager
> TC>> Tz>so, i dont understand why opalmanager would be the cause of your
> pain?
> TC>> Tz>
> TC>> Tz>> AM>> there is no way to send progress inicator throuch opalmgr.
> and
> TC>> there is
> TC>> Tz>> AM>> many another issues like
> TC>> Tz>> AM>> this in that layer.
> TC>> Tz>
> TC>> Tz>Please point me to the issues you have in opal, their bug reports ,
> TC>> traces etc.
> TC>> Tz>I dont think any of the opal people has psychic abilities to detect
> TC>> Tz>-your- problems
> TC>> Tz>and solve them.
> TC>> Tz>
> TC>> Tz>>ftp://srv.icf.org.ru/pub/soft/f/freeswitch/ - there is
> TC>> Tz>> AM>> my work on mod_opal before
> TC>> Tz>> AM>> i start moving to h323plus, may be this help somebody there.
> TC>> Tz>> AM>>
> TC>> Tz>> AM>> BW> From what I have been told h323plus is a based/fork of
> TC>> OpenH323 which
> TC>> Tz>> AM>> BW>OPAL is just a continuation of OpenH323.  So why not
> support
> TC>> the
> TC>> Tz>> AM>> BW>developers of OPAL/OpenH323 ?
> TC>> Tz>> AM>> BW>
> TC>> Tz>> AM>> BW>/b
> TC>> Tz>> AM>> BW>
> TC>> Tz>> AM>> BW>On Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote:
> TC>> Tz>> AM>> BW>
> TC>> Tz>> AM>> BW>> We are developing module to handle h323 proto now, we
> try to
> TC>> use
> TC>> Tz>> AM>> BW>> mod_opal and try improve it, but no luck,
> TC>> Tz>> AM>> BW>> there is many issues in libopal, and finaly we now move
> to
> TC>> h323plus
> TC>> Tz>> AM>> BW>> library.
> TC>> Tz>
> TC>> Tz>Did any of you try to report those issues?
> TC>> Tz>
> TC>> Tz>Regards
> TC>> Tz>
> TC>> Tz>/tyn
> TC>> Tz>
> TC>> Tz>> AM>> BW>
> TC>> Tz>> AM>> BW>
> TC>> Tz>> AM>> BW>_______________________________________________
> TC>> Tz>> AM>> BW>FreeSWITCH-users mailing list
> TC>> Tz>> AM>> BW>FreeSWITCH-users at lists.freeswitch.org
> TC>> Tz>> AM>> BW>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> TC>> Tz>> AM>> BW>UNSUBSCRIBE:
> TC>> Tz>> AM>>
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> TC>> Tz>> AM>> BW>http://www.freeswitch.org
> TC>> Tz>> AM>> BW>
> TC>> Tz>> AM>>
> TC>> Tz>> AM>> C ?????????                       With Best Regards
> TC>> Tz>> AM>> ???????????? ????.                Georgiewskiy Yuriy
> TC>> Tz>> AM>> +7 4872 711666                    +7 4872 711666
> TC>> Tz>> AM>> ???? +7 4872 711143               fax +7 4872 711143
> TC>> Tz>> AM>> ???????? ??? "?? ?? ??????"       IT Service Ltd
> TC>> Tz>> AM>> http://nkoort.ru                  http://nkoort.ru
> TC>> Tz>> AM>> JID: GHhost at jabber.tula-ix.net.ru JID:
> TC>> GHhost at jabber.tula-ix.net.ru
> TC>> Tz>> AM>> YG129-RIPE                        YG129-RIPE
> TC>> Tz>> AM>>
> TC>> Tz>> AM>> _______________________________________________
> TC>> Tz>> AM>> FreeSWITCH-users mailing list
> TC>> Tz>> AM>> FreeSWITCH-users at lists.freeswitch.org
> TC>> Tz>> AM>>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> TC>> Tz>> AM>> UNSUBSCRIBE:
> TC>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> TC>> Tz>> AM>> http://www.freeswitch.org
> TC>> Tz>> AM>>
> TC>> Tz>> AM>>
> TC>> Tz>> AM>
> TC>> Tz>> AM>
> TC>> Tz>> AM>
> TC>> Tz>>
> TC>> Tz>> C ?????????                       With Best Regards
> TC>> Tz>> ???????????? ????.                Georgiewskiy Yuriy
> TC>> Tz>> +7 4872 711666                    +7 4872 711666
> TC>> Tz>> ???? +7 4872 711143               fax +7 4872 711143
> TC>> Tz>> ???????? ??? "?? ?? ??????"       IT Service Ltd
> TC>> Tz>> http://nkoort.ru                  http://nkoort.ru
> TC>> Tz>> JID: GHhost at jabber.tula-ix.net.ru JID:
> GHhost at jabber.tula-ix.net.ru
> TC>> Tz>> YG129-RIPE                        YG129-RIPE
> TC>> Tz>> _______________________________________________
> TC>> Tz>> FreeSWITCH-users mailing list
> TC>> Tz>> FreeSWITCH-users at lists.freeswitch.org
> TC>> Tz>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> TC>> Tz>> UNSUBSCRIBE:
> TC>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> TC>> Tz>> http://www.freeswitch.org
> TC>> Tz>>
> TC>> Tz>>
> TC>> Tz>
> TC>> Tz>_______________________________________________
> TC>> Tz>FreeSWITCH-users mailing list
> TC>> Tz>FreeSWITCH-users at lists.freeswitch.org
> TC>> Tz>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> TC>> Tz>UNSUBSCRIBE:
> TC>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> TC>> Tz>http://www.freeswitch.org
> TC>> Tz>
> TC>>
> TC>> C ?????????                       With Best Regards
> TC>> ???????????? ????.                Georgiewskiy Yuriy
> TC>> +7 4872 711666                    +7 4872 711666
> TC>> ???? +7 4872 711143               fax +7 4872 711143
> TC>> ???????? ??? "?? ?? ??????"       IT Service Ltd
> TC>> http://nkoort.ru                  http://nkoort.ru
> TC>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> TC>> YG129-RIPE                        YG129-RIPE
> TC>>
> TC>> _______________________________________________
> TC>> FreeSWITCH-users mailing list
> TC>> FreeSWITCH-users at lists.freeswitch.org
> TC>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> TC>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> TC>> http://www.freeswitch.org
> TC>>
> TC>>
> TC>
>
> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129-RIPE
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From xengelpublicx at gmail.com  Thu Oct  8 12:50:44 2009
From: xengelpublicx at gmail.com (Vladimir Elizarov)
Date: Thu, 08 Oct 2009 23:50:44 +0400
Subject: [Freeswitch-users] freeswitch distributed
Message-ID: <4ACE4294.3030402@gmail.com>

I want to make freeswitch distributed in such a situation. Several
offices in each of fs. We need a unified dialplan, a single
entrance/exit on the SIP external gateways, in the case of
unavailability of one of the fs others work without his participation.

How to understand. It should be the following. Base presence to keep in
a separate database (pgpoll + pg). Dialplan, configure pull xml_curl.
What can you tell from the web-application (wikipbx, fs2web etc)? The
main task of which will impact dialplan, configure, directory?

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From dmitry.bely at gmail.com  Thu Oct  8 13:23:32 2009
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Fri, 9 Oct 2009 00:23:32 +0400
Subject: [Freeswitch-users] How to limit the number of incoming+outgoing
	calls via specific gateway?
In-Reply-To: <7e2ac3270910051352p7d052a5aife6f07827173767@mail.gmail.com>
References: <90823c940910011001o3698504ewc63184d0b012edc6@mail.gmail.com>
	<e403cddca1c90c83ce4fb6f85b026e63.squirrel@correo.nodo50.org>
	<65d96fc80910020532i4ba33353g7ac141624a4bb89e@mail.gmail.com>
	<C1AD67AA-F969-4BA1-8415-5E381BDF18FA@jerris.com>
	<90823c940910051257u403108afg3478c4c13ab26283@mail.gmail.com>
	<7e2ac3270910051319xfdfc9d3wa583c2da7e932339@mail.gmail.com>
	<90823c940910051339h3614be97o48a6319f68da7977@mail.gmail.com>
	<7e2ac3270910051352p7d052a5aife6f07827173767@mail.gmail.com>
Message-ID: <90823c940910081323j73e63385kdaab8cf3ac77d139@mail.gmail.com>

On Tue, Oct 6, 2009 at 12:52 AM, SP <sprice at gmail.com> wrote:
> did you use the application limit on the inbound call?
>?You'll need to in order to account for it.

Can you provide more detail? I have DID and gateway

  <extension name="public_did">
    <condition field="destination_number" expression="^(1234567)$">
      <action application="transfer" data="incoming_call XML default"/>
    </condition>
  </extension>

  <gateway name="my_provider">
    <param name="username" value="1234567"/>
(...)
  </gateway>

  <extension name="my_provider_local">
    <condition field="${toll_allow}" expression="local"/>
(...)
      <action application="bridge" data="sofia/gateway/my_provider/$1"/>
    </condition>
  </extension>

that correspond to the same provider account. Then an incoming call is
active, outgoing one is not possible (provider will reject the it).
How to generate "limit_exceeded" then?

- Dmitry Bely



From msc at freeswitch.org  Thu Oct  8 15:38:10 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Oct 2009 15:38:10 -0700
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <903da5680910081201p427a8358n768edb61a96674a4@mail.gmail.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<87f2f3b90910080958i1f5cfccax621fd12100a05e3e@mail.gmail.com>
	<903da5680910081201p427a8358n768edb61a96674a4@mail.gmail.com>
Message-ID: <87f2f3b90910081538l6e4e4b7dv3aedb2c4d1af58bc@mail.gmail.com>

On Thu, Oct 8, 2009 at 12:01 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> Naturally the wireless is jacked at the conf.  It's no match for the
> 100's of geeks sporting laptops.  Thanks for the link. Looks like
> we'll be shooting from the hip. :)
>
> We experienced something similar at ClueCon, but it was that the hotel's
DHCP server was a total wuss. :)
-MC
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From msc at freeswitch.org  Thu Oct  8 15:44:49 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Oct 2009 15:44:49 -0700
Subject: [Freeswitch-users] Changing callerid before/after call pickup
In-Reply-To: <A2F1C0E212BBBA4E8C0E7BE568361F04027F0233024A@srv02.khdev.corp>
References: <4ACB9519.9000805@gmail.com>
	<2d9149cd0910071235k27c1429v2bc555c0e7397fc4@mail.gmail.com>
	<4ACD076E.2080107@gmail.com>
	<5B417827-6CCB-4EE4-9524-5CD45AC9E28F@gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F04027F0233024A@srv02.khdev.corp>
Message-ID: <87f2f3b90910081544h3f86404fid50dfb644cbbb9f5@mail.gmail.com>

On Wed, Oct 7, 2009 at 6:46 PM, Klaus Hochlehnert <Mailings at kh-dev.de>wrote:

> Hi,
>
> currently I'm playing around with call pickup and Snom phones.
> I'm using the intercept function for that.
>
> My "problem" is now that after the call pickup (which works fine) I don't
> see the caller id of the original call.
> Instead I see the pickup code, e.g. *820
>
> I've tried to change nearly every channel variable (before doing the call
> pickup), but the phone won't display the caller id.
>
> Does anybody know how to change that?
>
Can you paste a debug log of the call flow? Also, are you using just the
default dialplan or have you made any changes? Lastly, what mods, if any,
did you make to your Snom configs? I have a bricked Snom that I'm waiting
for TFTP to fix but when it does I will see about testing this.

-MC
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From msc at freeswitch.org  Thu Oct  8 15:50:42 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Oct 2009 15:50:42 -0700
Subject: [Freeswitch-users] freeswitch distributed
In-Reply-To: <4ACE4294.3030402@gmail.com>
References: <4ACE4294.3030402@gmail.com>
Message-ID: <87f2f3b90910081550r7596fd11v71c406304e15478e@mail.gmail.com>

On Thu, Oct 8, 2009 at 12:50 PM, Vladimir Elizarov
<xengelpublicx at gmail.com>wrote:

> I want to make freeswitch distributed in such a situation. Several
> offices in each of fs. We need a unified dialplan, a single
> entrance/exit on the SIP external gateways, in the case of
> unavailability of one of the fs others work without his participation.
>
> How to understand. It should be the following. Base presence to keep in
> a separate database (pgpoll + pg). Dialplan, configure pull xml_curl.
> What can you tell from the web-application (wikipbx, fs2web etc)? The
> main task of which will impact dialplan, configure, directory?
>
>
This is quite an in depth question. It's not really something that can be
answered on a mailing list because of all of the specific questions that
need to be answered. I think your best bet is to email
consulting at freeswitch.org and let them discuss it with you. They can get you
hooked up with a FreeSWITCH expert whom you can hire to assist with this
project.
-MC
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From msc at freeswitch.org  Thu Oct  8 15:54:23 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Oct 2009 15:54:23 -0700
Subject: [Freeswitch-users] Changing callerid before/after call pickup
In-Reply-To: <A2F1C0E212BBBA4E8C0E7BE568361F04027F0233024A@srv02.khdev.corp>
References: <4ACB9519.9000805@gmail.com>
	<2d9149cd0910071235k27c1429v2bc555c0e7397fc4@mail.gmail.com>
	<4ACD076E.2080107@gmail.com>
	<5B417827-6CCB-4EE4-9524-5CD45AC9E28F@gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F04027F0233024A@srv02.khdev.corp>
Message-ID: <87f2f3b90910081554g187e1412hac62fa0a387a4c1a@mail.gmail.com>

Also,

I failed to mention two things:
#1 - Please don't hijack threads. :)
#2 - Update to the latest SVN and try again. You might be pleasantly
surprised.

-MC

On Wed, Oct 7, 2009 at 6:46 PM, Klaus Hochlehnert <Mailings at kh-dev.de>wrote:

> Hi,
>
> currently I'm playing around with call pickup and Snom phones.
> I'm using the intercept function for that.
>
> My "problem" is now that after the call pickup (which works fine) I don't
> see the caller id of the original call.
> Instead I see the pickup code, e.g. *820
>
> I've tried to change nearly every channel variable (before doing the call
> pickup), but the phone won't display the caller id.
>
> Does anybody know how to change that?
>
> Thanks, Klaus
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Mailings at kh-dev.de  Thu Oct  8 16:23:25 2009
From: Mailings at kh-dev.de (Klaus Hochlehnert)
Date: Fri, 9 Oct 2009 01:23:25 +0200
Subject: [Freeswitch-users] Changing callerid before/after call pickup
In-Reply-To: <87f2f3b90910081554g187e1412hac62fa0a387a4c1a@mail.gmail.com>
References: <4ACB9519.9000805@gmail.com>
	<2d9149cd0910071235k27c1429v2bc555c0e7397fc4@mail.gmail.com>
	<4ACD076E.2080107@gmail.com>
	<5B417827-6CCB-4EE4-9524-5CD45AC9E28F@gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F04027F0233024A@srv02.khdev.corp>
	<87f2f3b90910081554g187e1412hac62fa0a387a4c1a@mail.gmail.com>
Message-ID: <A2F1C0E212BBBA4E8C0E7BE568361F04027F0233024D@srv02.khdev.corp>

Sorry for hijacking. Didn't know that the mailing list system finds out about that... ;-)

Will do the update in the next days and try again.

Today I tried with the command  send_display.
This workaround currently works for me.

Thanks, Klaus


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Friday, October 09, 2009 12:54 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Changing callerid before/after call pickup

Also,

I failed to mention two things:
#1 - Please don't hijack threads. :)
#2 - Update to the latest SVN and try again. You might be pleasantly surprised.

-MC
On Wed, Oct 7, 2009 at 6:46 PM, Klaus Hochlehnert <Mailings at kh-dev.de<mailto:Mailings at kh-dev.de>> wrote:
Hi,

currently I'm playing around with call pickup and Snom phones.
I'm using the intercept function for that.

My "problem" is now that after the call pickup (which works fine) I don't see the caller id of the original call.
Instead I see the pickup code, e.g. *820

I've tried to change nearly every channel variable (before doing the call pickup), but the phone won't display the caller id.

Does anybody know how to change that?

Thanks, Klaus


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From brian at freeswitch.org  Thu Oct  8 16:44:14 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 8 Oct 2009 18:44:14 -0500
Subject: [Freeswitch-users] Changing callerid before/after call pickup
In-Reply-To: <A2F1C0E212BBBA4E8C0E7BE568361F04027F0233024D@srv02.khdev.corp>
References: <4ACB9519.9000805@gmail.com>
	<2d9149cd0910071235k27c1429v2bc555c0e7397fc4@mail.gmail.com>
	<4ACD076E.2080107@gmail.com>
	<5B417827-6CCB-4EE4-9524-5CD45AC9E28F@gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F04027F0233024A@srv02.khdev.corp>
	<87f2f3b90910081554g187e1412hac62fa0a387a4c1a@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F04027F0233024D@srv02.khdev.corp>
Message-ID: <9B081533-60AD-4D25-858C-D9814D2AF5FE@freeswitch.org>

If you're starting a new topic/thread please click new message.  If  
you click reply, delete the subject and body and change it.. the  
thread is now hijacked.

/b

On Oct 8, 2009, at 6:23 PM, Klaus Hochlehnert wrote:

> Sorry for hijacking. Didn?t know that the mailing list system finds  
> out about that... ;-)
>
> Will do the update in the next days and try again.
>
> Today I tried with the command  send_display.
> This workaround currently works for me.
>
> Thanks, Klaus

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From msc at freeswitch.org  Thu Oct  8 18:42:59 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Oct 2009 18:42:59 -0700
Subject: [Freeswitch-users] FreeSWITCH Update: Valet Parking
Message-ID: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>

FYI,

The FreeSWITCH devs have added valet parking! Check it out:
http://www.freeswitch.org/node/207

Let us know what you think.

Thanks,
Michael
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From diego.viola at gmail.com  Thu Oct  8 19:56:39 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 9 Oct 2009 02:56:39 +0000
Subject: [Freeswitch-users] Heartbeat question
Message-ID: <86a32abc0910081956k91b3600sb8c556e589f5b089@mail.gmail.com>

Hi everyone,

I have a question about FreeSWITCH heartbeat, I have this on my dialplan:

<extension name="test">
<condition field="destination_number" expression="^8084$">
<action application="answer"/>
<action application="enable_heartbeat" data="1"/>
</condition>
</extension>

But when I do "event plain all" I see the heartbeats are being fired every
20 seconds... what I'm doing wrong?

Thanks,

Diego
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From diego.viola at gmail.com  Thu Oct  8 20:03:43 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 9 Oct 2009 03:03:43 +0000
Subject: [Freeswitch-users] Heartbeat question
In-Reply-To: <86a32abc0910081956k91b3600sb8c556e589f5b089@mail.gmail.com>
References: <86a32abc0910081956k91b3600sb8c556e589f5b089@mail.gmail.com>
Message-ID: <86a32abc0910082003v3f5d4e6dua8e7eae18fd2f939@mail.gmail.com>

I want it to fire every 1 second...

On Fri, Oct 9, 2009 at 2:56 AM, Diego Viola <diego.viola at gmail.com> wrote:

> Hi everyone,
>
> I have a question about FreeSWITCH heartbeat, I have this on my dialplan:
>
> <extension name="test">
> <condition field="destination_number" expression="^8084$">
> <action application="answer"/>
> <action application="enable_heartbeat" data="1"/>
> </condition>
> </extension>
>
> But when I do "event plain all" I see the heartbeats are being fired every
> 20 seconds... what I'm doing wrong?
>
> Thanks,
>
> Diego
>
>
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From mcampbellsmith at gmail.com  Thu Oct  8 20:06:37 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Fri, 9 Oct 2009 14:06:37 +1100
Subject: [Freeswitch-users] mod_fax compile fails
Message-ID: <33c87fa30910082006v1e91849fjbf4282541e7ac377@mail.gmail.com>

HI all,

I just tried to update to the latest svn and I get these errors right
at the end after issuing a 'make current'.  I am using Debian Lenny.

making all mod_fax
make[5]: Entering directory `/home/mark/freeswitch/src/mod/applications/mod_fax'
make[6]: Entering directory `/home/mark/freeswitch/src/mod/applications/mod_fax'
make[7]: Entering directory `/home/mark/freeswitch/libs/tiff-3.8.2'
running /bin/sh ./configure  --prefix=/usr/local/freeswitch
--cache-file=/dev/null --srcdir=. --disable-shared --with-pic
--no-create --no-recursion
configure: error: cannot run /bin/sh config/config.sub
make[7]: *** [config.status] Error 1
make[7]: Leaving directory `/home/mark/freeswitch/libs/tiff-3.8.2'
make[6]: *** [../../../../libs/tiff-3.8.2/libtiff/libtiff.la] Error 2
make[6]: Leaving directory `/home/mark/freeswitch/src/mod/applications/mod_fax'
make[5]: *** [all] Error 1
make[5]: Leaving directory `/home/mark/freeswitch/src/mod/applications/mod_fax'
make[4]: *** [mod_fax-all] Error 1
make[4]: Leaving directory `/home/mark/freeswitch/src/mod'
make[3]: *** [all-recursive] Error 1
make[3]: Leaving directory `/home/mark/freeswitch/src'
Making all in build
make[3]: Entering directory `/home/mark/freeswitch/build'
 +-------- FreeSWITCH Build Complete -----------+
 + FreeSWITCH has been successfully built.      +
 + Install by running:                          +
 +                                              +
 +               make install                   +
 +----------------------------------------------+
make[3]: Leaving directory `/home/mark/freeswitch/build'
make[2]: *** [all-recursive] Error 1
make[2]: Leaving directory `/home/mark/freeswitch'
make[1]: *** [all] Error 2
make[1]: Leaving directory `/home/mark/freeswitch'
make: *** [current] Error 2

Also, are the 'Leaving directory / all-recursive' errors going to
cause a problem?

Thanks!
Any ideas what the cause is?



From william.suffill at gmail.com  Thu Oct  8 20:36:15 2009
From: william.suffill at gmail.com (William Suffill)
Date: Thu, 8 Oct 2009 23:36:15 -0400
Subject: [Freeswitch-users] Heartbeat question
In-Reply-To: <86a32abc0910082003v3f5d4e6dua8e7eae18fd2f939@mail.gmail.com>
References: <86a32abc0910081956k91b3600sb8c556e589f5b089@mail.gmail.com> 
	<86a32abc0910082003v3f5d4e6dua8e7eae18fd2f939@mail.gmail.com>
Message-ID: <6b65470d0910082036g204d5dcbw26e1e43ef80c7c84@mail.gmail.com>

Why do you need it every second? If you want real time channel counts
you would  be able to track each create/destroy even instead of
relying on the heartbeat summary.

-- W



From rob4manhere at gmail.com  Thu Oct  8 20:39:10 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Thu, 8 Oct 2009 22:39:10 -0500
Subject: [Freeswitch-users] mod_fax compile fails
In-Reply-To: <33c87fa30910082006v1e91849fjbf4282541e7ac377@mail.gmail.com>
References: <33c87fa30910082006v1e91849fjbf4282541e7ac377@mail.gmail.com>
Message-ID: <343C884A-9AB5-4755-9C5F-42ED705D05E7@gmail.com>

I had that issue too where make current failed on mod_fax (under libs/ 
tiff).  And yeah, it caused a problem where a bunch of modules  
wouldn't load.  You'll want to get it resolved before installing.  I  
ended up moving the existing source aside and re-checked out the  
trunk, which compiled fine.


On Oct 8, 2009, at 10:06 PM, Mark Campbell-Smith wrote:

> HI all,
>
> I just tried to update to the latest svn and I get these errors right
> at the end after issuing a 'make current'.  I am using Debian Lenny.
>
> making all mod_fax
> make[5]: Entering directory `/home/mark/freeswitch/src/mod/ 
> applications/mod_fax'
> make[6]: Entering directory `/home/mark/freeswitch/src/mod/ 
> applications/mod_fax'
> make[7]: Entering directory `/home/mark/freeswitch/libs/tiff-3.8.2'
> running /bin/sh ./configure  --prefix=/usr/local/freeswitch
> --cache-file=/dev/null --srcdir=. --disable-shared --with-pic
> --no-create --no-recursion
> configure: error: cannot run /bin/sh config/config.sub
> make[7]: *** [config.status] Error 1
> make[7]: Leaving directory `/home/mark/freeswitch/libs/tiff-3.8.2'
> make[6]: *** [../../../../libs/tiff-3.8.2/libtiff/libtiff.la] Error 2
> make[6]: Leaving directory `/home/mark/freeswitch/src/mod/ 
> applications/mod_fax'
> make[5]: *** [all] Error 1
> make[5]: Leaving directory `/home/mark/freeswitch/src/mod/ 
> applications/mod_fax'
> make[4]: *** [mod_fax-all] Error 1
> make[4]: Leaving directory `/home/mark/freeswitch/src/mod'
> make[3]: *** [all-recursive] Error 1
> make[3]: Leaving directory `/home/mark/freeswitch/src'
> Making all in build
> make[3]: Entering directory `/home/mark/freeswitch/build'
> +-------- FreeSWITCH Build Complete -----------+
> + FreeSWITCH has been successfully built.      +
> + Install by running:                          +
> +                                              +
> +               make install                   +
> +----------------------------------------------+
> make[3]: Leaving directory `/home/mark/freeswitch/build'
> make[2]: *** [all-recursive] Error 1
> make[2]: Leaving directory `/home/mark/freeswitch'
> make[1]: *** [all] Error 2
> make[1]: Leaving directory `/home/mark/freeswitch'
> make: *** [current] Error 2
>
> Also, are the 'Leaving directory / all-recursive' errors going to
> cause a problem?
>
> Thanks!
> Any ideas what the cause is?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From ahmedmunir007 at gmail.com  Thu Oct  8 20:39:24 2009
From: ahmedmunir007 at gmail.com (Ahmed Munir)
Date: Fri, 9 Oct 2009 08:39:24 +0500
Subject: [Freeswitch-users] Questions regarding to mod_nibble
Message-ID: <a60fed5e0910082039yc8b7ef8x24260041a7c11d31@mail.gmail.com>

 I want to ask three questions related to mod_nibble bills, as I'm listing
down below;

1- Can we select/use dynamic tables for billing using nibble bill?
2- Can we define more than two tables and attributes in nibblebill.conf.xml?
3- As Nibble bill is use to deduct amount of user account, Can we deduct
minutes instead of cash? Because my case is, if a user buy a package and I
only want to deducts his/her minutes. How we can resolve it by nibble bill?
/ What other way we can resolve it?

Kindly advise soon.

-- 
Regards,

Ahmed Munir
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From hads at nice.net.nz  Thu Oct  8 21:54:41 2009
From: hads at nice.net.nz (Hadley Rich)
Date: Fri, 09 Oct 2009 17:54:41 +1300
Subject: [Freeswitch-users] FreeSWITCH Update: Valet Parking
In-Reply-To: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>
References: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>
Message-ID: <1255064081.2669.1.camel@sodium>

On Thu, 2009-10-08 at 18:42 -0700, Michael Collins wrote:
> FYI,
> 
> The FreeSWITCH devs have added valet parking! Check it out:
> http://www.freeswitch.org/node/207
> 
> Let us know what you think.

I think it's awesome. Love your work as always guys.

hads

-- 
http://nicegear.co.nz
New Zealand's Open Source Hardware Supplier




From msc at freeswitch.org  Thu Oct  8 22:13:37 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Oct 2009 22:13:37 -0700
Subject: [Freeswitch-users] Heartbeat question
In-Reply-To: <86a32abc0910081956k91b3600sb8c556e589f5b089@mail.gmail.com>
References: <86a32abc0910081956k91b3600sb8c556e589f5b089@mail.gmail.com>
Message-ID: <87f2f3b90910082213j1fb95654mf482c1ca8cf4cdf@mail.gmail.com>

On Thu, Oct 8, 2009 at 7:56 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Hi everyone,
>
> I have a question about FreeSWITCH heartbeat, I have this on my dialplan:
>
> <extension name="test">
> <condition field="destination_number" expression="^8084$">
> <action application="answer"/>
> <action application="enable_heartbeat" data="1"/>
> </condition>
> </extension>
>
> But when I do "event plain all" I see the heartbeats are being fired every
> 20 seconds... what I'm doing wrong?
>
>
On the console do you see the call to switch_core_session_enable_heartbeat?
switch_core_session.c:1041:
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "%s
setting session heartbeat to %u second(s).\n",

Confirm that it is actually calling that.

BTW, it would sure be nice if this app were documented on the wiki, hint
hint. ;)
-MC


> Thanks,
>
> Diego
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gabe at gundy.org  Thu Oct  8 22:38:56 2009
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 8 Oct 2009 23:38:56 -0600
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <87f2f3b90910081538l6e4e4b7dv3aedb2c4d1af58bc@mail.gmail.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<87f2f3b90910080958i1f5cfccax621fd12100a05e3e@mail.gmail.com>
	<903da5680910081201p427a8358n768edb61a96674a4@mail.gmail.com>
	<87f2f3b90910081538l6e4e4b7dv3aedb2c4d1af58bc@mail.gmail.com>
Message-ID: <903da5680910082238x27bff476s8bea0e74ab516a8d@mail.gmail.com>

On Thu, Oct 8, 2009 at 4:38 PM, Michael Collins <msc at freeswitch.org> wrote:
> On Thu, Oct 8, 2009 at 12:01 PM, Gabriel Gunderson <gabe at gundy.org> wrote:
>> Naturally the wireless is jacked at the conf. ?It's no match for the
>> 100's of geeks sporting laptops. ?Thanks for the link. Looks like
>> we'll be shooting from the hip. :)
>>
> We experienced something similar at ClueCon, but it was that the hotel's
> DHCP server was a total wuss. :)

So, I just got back from the first day of the conference and there was
plenty of interest in FreeSWITCH from all the Asterisk users.  We had
a good time and answered lots of questions.  It was interesting to
hear what their perceptions about FS were.

Gabe



From diego.viola at gmail.com  Thu Oct  8 22:39:46 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 9 Oct 2009 05:39:46 +0000
Subject: [Freeswitch-users] Heartbeat question
In-Reply-To: <87f2f3b90910082213j1fb95654mf482c1ca8cf4cdf@mail.gmail.com>
References: <86a32abc0910081956k91b3600sb8c556e589f5b089@mail.gmail.com>
	<87f2f3b90910082213j1fb95654mf482c1ca8cf4cdf@mail.gmail.com>
Message-ID: <86a32abc0910082239s68f9df39ocf1548b57f79fe3f@mail.gmail.com>

Yes, I have called to my extension and saw this:

2009-10-09 01:25:44.820677 [INFO] switch_core_session.c:1041 sofia/internal/
1000 at 192.168.0.2 setting session heartbeat to 1 second(s).

But I still see the heartbeat events being fired after 20 seconds...

I have made this script that tells me the seconds when a heartbeat is being
fired...

#!/usr/bin/env ruby

require 'rubygems'
require 'fsr'
require "fsr/listener/inbound"

def custom_channel_heartbeat_handler(event)
  puts Time.now.strftime('%S')
end
FSL::Inbound.add_event_hook(:HEARTBEAT) {|event|
custom_channel_heartbeat_handler(event) }
FSR.start_ies!(FSL::Inbound, :host => "localhost", :port => 8021)


This is the output:

[diego at myhost ~]$ ruby test.rb
No log4r found, falling back to standard ruby library Logger
I, [2009-10-09T01:25:51.797012 #3292]  INFO -- : *** FreeSWITCHer Inbound
EventSocket Listener connected to localhost:8021 ***
I, [2009-10-09T01:25:51.797291 #3292]  INFO -- : ***
http://code.rubyists.com/projects/fs
59
19
39
59
19
39
59


On Fri, Oct 9, 2009 at 5:13 AM, Michael Collins <msc at freeswitch.org> wrote:

>
> On Thu, Oct 8, 2009 at 7:56 PM, Diego Viola <diego.viola at gmail.com> wrote:
>
>> Hi everyone,
>>
>> I have a question about FreeSWITCH heartbeat, I have this on my dialplan:
>>
>> <extension name="test">
>> <condition field="destination_number" expression="^8084$">
>> <action application="answer"/>
>> <action application="enable_heartbeat" data="1"/>
>> </condition>
>> </extension>
>>
>> But when I do "event plain all" I see the heartbeats are being fired every
>> 20 seconds... what I'm doing wrong?
>>
>>
> On the console do you see the call to switch_core_session_enable_heartbeat?
>
> switch_core_session.c:1041:
> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "%s
> setting session heartbeat to %u second(s).\n",
>
> Confirm that it is actually calling that.
>
> BTW, it would sure be nice if this app were documented on the wiki, hint
> hint. ;)
> -MC
>
>
>> Thanks,
>>
>> Diego
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mcampbellsmith at gmail.com  Thu Oct  8 22:43:39 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Fri, 9 Oct 2009 16:43:39 +1100
Subject: [Freeswitch-users] mod_fax compile fails
In-Reply-To: <343C884A-9AB5-4755-9C5F-42ED705D05E7@gmail.com>
References: <33c87fa30910082006v1e91849fjbf4282541e7ac377@mail.gmail.com>
	<343C884A-9AB5-4755-9C5F-42ED705D05E7@gmail.com>
Message-ID: <33c87fa30910082243h6c689816i23368bedfdfd14c0@mail.gmail.com>

Thanks Rob,

Is this a fault in the svn update process?  if so, should/has it been
bug reported?


On Fri, Oct 9, 2009 at 2:39 PM, Rob Forman <rob4manhere at gmail.com> wrote:
> I had that issue too where make current failed on mod_fax (under libs/
> tiff). ?And yeah, it caused a problem where a bunch of modules
> wouldn't load. ?You'll want to get it resolved before installing. ?I
> ended up moving the existing source aside and re-checked out the
> trunk, which compiled fine.
>
>
> On Oct 8, 2009, at 10:06 PM, Mark Campbell-Smith wrote:
>
>> HI all,
>>
>> I just tried to update to the latest svn and I get these errors right
>> at the end after issuing a 'make current'. ?I am using Debian Lenny.
>>
>> making all mod_fax
>> make[5]: Entering directory `/home/mark/freeswitch/src/mod/
>> applications/mod_fax'
>> make[6]: Entering directory `/home/mark/freeswitch/src/mod/
>> applications/mod_fax'
>> make[7]: Entering directory `/home/mark/freeswitch/libs/tiff-3.8.2'
>> running /bin/sh ./configure ?--prefix=/usr/local/freeswitch
>> --cache-file=/dev/null --srcdir=. --disable-shared --with-pic
>> --no-create --no-recursion
>> configure: error: cannot run /bin/sh config/config.sub
>> make[7]: *** [config.status] Error 1
>> make[7]: Leaving directory `/home/mark/freeswitch/libs/tiff-3.8.2'
>> make[6]: *** [../../../../libs/tiff-3.8.2/libtiff/libtiff.la] Error 2
>> make[6]: Leaving directory `/home/mark/freeswitch/src/mod/
>> applications/mod_fax'
>> make[5]: *** [all] Error 1
>> make[5]: Leaving directory `/home/mark/freeswitch/src/mod/
>> applications/mod_fax'
>> make[4]: *** [mod_fax-all] Error 1
>> make[4]: Leaving directory `/home/mark/freeswitch/src/mod'
>> make[3]: *** [all-recursive] Error 1
>> make[3]: Leaving directory `/home/mark/freeswitch/src'
>> Making all in build
>> make[3]: Entering directory `/home/mark/freeswitch/build'
>> +-------- FreeSWITCH Build Complete -----------+
>> + FreeSWITCH has been successfully built. ? ? ?+
>> + Install by running: ? ? ? ? ? ? ? ? ? ? ? ? ?+
>> + ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?+
>> + ? ? ? ? ? ? ? make install ? ? ? ? ? ? ? ? ? +
>> +----------------------------------------------+
>> make[3]: Leaving directory `/home/mark/freeswitch/build'
>> make[2]: *** [all-recursive] Error 1
>> make[2]: Leaving directory `/home/mark/freeswitch'
>> make[1]: *** [all] Error 2
>> make[1]: Leaving directory `/home/mark/freeswitch'
>> make: *** [current] Error 2
>>
>> Also, are the 'Leaving directory / all-recursive' errors going to
>> cause a problem?
>>
>> Thanks!
>> Any ideas what the cause is?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From diego.viola at gmail.com  Thu Oct  8 22:45:47 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 9 Oct 2009 05:45:47 +0000
Subject: [Freeswitch-users] Heartbeat question
In-Reply-To: <6b65470d0910082036g204d5dcbw26e1e43ef80c7c84@mail.gmail.com>
References: <86a32abc0910081956k91b3600sb8c556e589f5b089@mail.gmail.com>
	<86a32abc0910082003v3f5d4e6dua8e7eae18fd2f939@mail.gmail.com>
	<6b65470d0910082036g204d5dcbw26e1e43ef80c7c84@mail.gmail.com>
Message-ID: <86a32abc0910082245j36b8b3e9ncaf1cdd358fce319@mail.gmail.com>

Nope, I was just wondering why it didn't work at 1 second exactly...

On Fri, Oct 9, 2009 at 3:36 AM, William Suffill
<william.suffill at gmail.com>wrote:

> Why do you need it every second? If you want real time channel counts
> you would  be able to track each create/destroy even instead of
> relying on the heartbeat summary.
>
> -- W
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Thu Oct  8 23:08:19 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Oct 2009 23:08:19 -0700
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <903da5680910082238x27bff476s8bea0e74ab516a8d@mail.gmail.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<87f2f3b90910080958i1f5cfccax621fd12100a05e3e@mail.gmail.com>
	<903da5680910081201p427a8358n768edb61a96674a4@mail.gmail.com>
	<87f2f3b90910081538l6e4e4b7dv3aedb2c4d1af58bc@mail.gmail.com>
	<903da5680910082238x27bff476s8bea0e74ab516a8d@mail.gmail.com>
Message-ID: <87f2f3b90910082308i7f6144d9kc887596be54dd609@mail.gmail.com>

On Thu, Oct 8, 2009 at 10:38 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Thu, Oct 8, 2009 at 4:38 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> > On Thu, Oct 8, 2009 at 12:01 PM, Gabriel Gunderson <gabe at gundy.org>
> wrote:
> >> Naturally the wireless is jacked at the conf.  It's no match for the
> >> 100's of geeks sporting laptops.  Thanks for the link. Looks like
> >> we'll be shooting from the hip. :)
> >>
> > We experienced something similar at ClueCon, but it was that the hotel's
> > DHCP server was a total wuss. :)
>
> So, I just got back from the first day of the conference and there was
> plenty of interest in FreeSWITCH from all the Asterisk users.  We had
> a good time and answered lots of questions.  It was interesting to
> hear what their perceptions about FS were.
>

Thanks for reporting back. Please let all the Asterisk users know that they
are welcome to join us in #freeswitch on irc.freenode.net and that they will
not be abused like people do in other less friendly IRC channels.

-MC


>
> Gabe
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jason at jasonjgw.net  Thu Oct  8 23:11:30 2009
From: jason at jasonjgw.net (Jason White)
Date: Fri, 9 Oct 2009 17:11:30 +1100
Subject: [Freeswitch-users] mod_fax compile fails
In-Reply-To: <33c87fa30910082243h6c689816i23368bedfdfd14c0@mail.gmail.com>
References: <33c87fa30910082006v1e91849fjbf4282541e7ac377@mail.gmail.com>
	<343C884A-9AB5-4755-9C5F-42ED705D05E7@gmail.com>
	<33c87fa30910082243h6c689816i23368bedfdfd14c0@mail.gmail.com>
Message-ID: <20091009061130.GA7509@jdc.jasonjgw.net>

Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
> Thanks Rob,
> 
> Is this a fault in the svn update process?  

Usually, no, it's rather that if you build in a directory that you've checked
out and then run svn update, not everything gets cleaned up properly the next
time you build.

I find it's faster to build in a tmpfs file system anyway, e.g.,

mkdir /tmp/fs && sudo mount -t tmpfs tmpfs /tmpfs && svn export . /tmp/fs
then go into /tmp/fs and run the compilation process, or the package building
command, or whatever you need.

This way, my original svn tree is never altered and it will always update
cleanly.




From gabe at gundy.org  Thu Oct  8 23:41:48 2009
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 9 Oct 2009 00:41:48 -0600
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <87f2f3b90910082308i7f6144d9kc887596be54dd609@mail.gmail.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<87f2f3b90910080958i1f5cfccax621fd12100a05e3e@mail.gmail.com>
	<903da5680910081201p427a8358n768edb61a96674a4@mail.gmail.com>
	<87f2f3b90910081538l6e4e4b7dv3aedb2c4d1af58bc@mail.gmail.com>
	<903da5680910082238x27bff476s8bea0e74ab516a8d@mail.gmail.com>
	<87f2f3b90910082308i7f6144d9kc887596be54dd609@mail.gmail.com>
Message-ID: <903da5680910082341r485092vb0cd10c43c4a2534@mail.gmail.com>

On Fri, Oct 9, 2009 at 12:08 AM, Michael Collins <msc at freeswitch.org> wrote:
> Thanks for reporting back. Please let all the Asterisk users know that they
> are welcome to join us in #freeswitch on irc.freenode.net and that they will
> not be abused like people do in other less friendly IRC channels.

Funny you mention this.  Many people report that the way the FS
community refers to Asterisk in docs/wikis/irc/whatever makes the FS
camp seem *less* welcoming to them.  After all, they identify as
Asterisk Users and take the criticism as being kinda harsh.  Most of
them acknowledge the shortcomings of Asterisk but are put off when
someone else points them out.  It's crazy, I know.  The thing is, I
remember thinking that too.  After getting to know FS better, I didn't
notice it as much.  Nobody likes to hear their baby is ugly --even if
they know it is.

At our session, and in general, I've noticed people are more
interested in hearing about FS when you don't make direct comparisons
to Asterisk.  Besides, FS stands on it's own merit.

Just what I've observed *and* my 2 additional cents.

Gabe



From gabe at gundy.org  Thu Oct  8 23:45:53 2009
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 9 Oct 2009 00:45:53 -0600
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910082242570.16063@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910082242570.16063@srv.icf.org.ru>
Message-ID: <903da5680910082345y12c11fa8h65c559ffdbf3df23@mail.gmail.com>

Doesn't anyone trim on this list?

Gabe

P.S.  Sorry about the top post, but I was worried that if I bottom
posted, nobody would find it.


2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>:
> On 2009-10-08 20:32 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:
>
> TC>Hi Yuriy,
> TC>
> TC>can you share what you have so far, I'm sure we can help with RTP part...
>
> I think there is a few days and i make it work, after this i start to test and share it.
>
> TC>
> TC>T.
> TC>
> TC>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
> TC>
> TC>> On 2009-10-08 13:25 -0400, Tuyan ?zipek wrote
> TC>> freeswitch-users at lists.freesw...:
> TC>>
> TC>> Tz>Hi,
> TC>> Tz>
> TC>> Tz>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>:
> TC>> Tz>> On 2009-10-08 10:43 -0500, Anthony Minessale wrote
> TC>> freeswitch-users at lists.f...:
> TC>> Tz>>
> TC>> Tz>> AM>If you are going to make that alternate module are you going to
> TC>> host it in
> TC>> Tz>> AM>the FS tree along side mod_opal?
> TC>> Tz>>
> TC>> Tz>> Yes, but then it be useful, now i have working only signaling part and
> TC>> some
> TC>> Tz>> kinde of not working rtp part :)
> TC>> Tz>
> TC>> Tz>If you dont use fs rtp stack, its unlikely that it will be accepted
> TC>> Tz>into the tree.
> TC>> Tz>
> TC>> Tz>>
> TC>> Tz>> AM>also if were working on mod_opal why did you not try to involve us
> TC>> and the
> TC>> Tz>> AM>opal team?
> TC>> Tz>>
> TC>> Tz>> Because i made patches for libopal, one is a bugfix in rtp part, there
> TC>> is a race condition
> TC>> Tz>> in inicialisation in jitter buffer, another patch implements method to
> TC>> send progress indicator,
> TC>> Tz>> and i don't wont spent my time to incorporate this changes into
> TC>> libopal.
> TC>> Tz>
> TC>> Tz>Thats bad.
> TC>> Tz>Any bugfixes from fs, goes to upstream on any of the used libraries.
> TC>> Tz>You should be doing the same.
> TC>> Tz>And Opal developers, will either include or refuse your patches. If
> TC>> Tz>they refuse it, they will give you the reason.
> TC>>
> TC>> i make this fix only to freeze my current mod_opal work on working state,
> TC>> while it now work for me i work on
> TC>> my new implementation of h323 proto for fs, i think opal developers will
> TC>> fix this rtp bug himself becouse
> TC>> it crashes and make library unuseful.
> TC>>
> TC>> Tz>
> TC>> Tz>> without this changes
> TC>> Tz>> my work on mod_opal in freeswitch don't useful at all, i provide link
> TC>> to my work with all
> TC>> Tz>> patches, if somebody wont incorporate it in libopal tree and fs - go
> TC>> on, but i think
> TC>> Tz>> better and more elegant make new module based on h323plus.
> TC>> Tz>
> TC>> Tz>If you dont publish your changes, all those you are trying to achieve,
> TC>> Tz>wont happen.
> TC>> Tz>
> TC>> Tz>>
> TC>> Tz>> AM>How far away from what is in tree are these patches you have?
> TC>> Tz>> AM>
> TC>> Tz>> AM>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
> TC>> Tz>> AM>
> TC>> Tz>> AM>> On 2009-10-07 15:09 -0500, Brian West wrote
> TC>> Tz>> AM>> freeswitch-users at lists.freeswit...:
> TC>> Tz>> AM>>
> TC>> Tz>> AM>> opal have addition abstraction layer called opalmgr, and it
> TC>> implementation
> TC>> Tz>> AM>> is not so good in
> TC>> Tz>> AM>> this case, for example to implemet pre_answer in mod_opal i need
> TC>> patch
> TC>> Tz>> AM>> libopal, because
> TC>> Tz>
> TC>> Tz>The patch you have in there, adds a method to the OpalCall, ?it does
> TC>> Tz>not touch any parts of OpalManager
> TC>> Tz>so, i dont understand why opalmanager would be the cause of your pain?
> TC>> Tz>
> TC>> Tz>> AM>> there is no way to send progress inicator throuch opalmgr. and
> TC>> there is
> TC>> Tz>> AM>> many another issues like
> TC>> Tz>> AM>> this in that layer.
> TC>> Tz>
> TC>> Tz>Please point me to the issues you have in opal, their bug reports ,
> TC>> traces etc.
> TC>> Tz>I dont think any of the opal people has psychic abilities to detect
> TC>> Tz>-your- problems
> TC>> Tz>and solve them.
> TC>> Tz>
> TC>> Tz>>ftp://srv.icf.org.ru/pub/soft/f/freeswitch/ - there is
> TC>> Tz>> AM>> my work on mod_opal before
> TC>> Tz>> AM>> i start moving to h323plus, may be this help somebody there.
> TC>> Tz>> AM>>
> TC>> Tz>> AM>> BW> From what I have been told h323plus is a based/fork of
> TC>> OpenH323 which
> TC>> Tz>> AM>> BW>OPAL is just a continuation of OpenH323. ?So why not support
> TC>> the
> TC>> Tz>> AM>> BW>developers of OPAL/OpenH323 ?
> TC>> Tz>> AM>> BW>
> TC>> Tz>> AM>> BW>/b
> TC>> Tz>> AM>> BW>
> TC>> Tz>> AM>> BW>On Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote:
> TC>> Tz>> AM>> BW>
> TC>> Tz>> AM>> BW>> We are developing module to handle h323 proto now, we try to
> TC>> use
> TC>> Tz>> AM>> BW>> mod_opal and try improve it, but no luck,
> TC>> Tz>> AM>> BW>> there is many issues in libopal, and finaly we now move to
> TC>> h323plus
> TC>> Tz>> AM>> BW>> library.
> TC>> Tz>
> TC>> Tz>Did any of you try to report those issues?
> TC>> Tz>
> TC>> Tz>Regards
> TC>> Tz>
> TC>> Tz>/tyn
> TC>> Tz>
> TC>> Tz>> AM>> BW>
> TC>> Tz>> AM>> BW>
> TC>> Tz>> AM>> BW>_______________________________________________
> TC>> Tz>> AM>> BW>FreeSWITCH-users mailing list
> TC>> Tz>> AM>> BW>FreeSWITCH-users at lists.freeswitch.org
> TC>> Tz>> AM>> BW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> TC>> Tz>> AM>> BW>UNSUBSCRIBE:
> TC>> Tz>> AM>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> TC>> Tz>> AM>> BW>http://www.freeswitch.org
> TC>> Tz>> AM>> BW>
> TC>> Tz>> AM>>
> TC>> Tz>> AM>> C ????????? ? ? ? ? ? ? ? ? ? ? ? With Best Regards
> TC>> Tz>> AM>> ???????????? ????. ? ? ? ? ? ? ? ?Georgiewskiy Yuriy
> TC>> Tz>> AM>> +7 4872 711666 ? ? ? ? ? ? ? ? ? ?+7 4872 711666
> TC>> Tz>> AM>> ???? +7 4872 711143 ? ? ? ? ? ? ? fax +7 4872 711143
> TC>> Tz>> AM>> ???????? ??? "?? ?? ??????" ? ? ? IT Service Ltd
> TC>> Tz>> AM>> http://nkoort.ru ? ? ? ? ? ? ? ? ?http://nkoort.ru
> TC>> Tz>> AM>> JID: GHhost at jabber.tula-ix.net.ru JID:
> TC>> GHhost at jabber.tula-ix.net.ru
> TC>> Tz>> AM>> YG129-RIPE ? ? ? ? ? ? ? ? ? ? ? ?YG129-RIPE
> TC>> Tz>> AM>>
> TC>> Tz>> AM>> _______________________________________________
> TC>> Tz>> AM>> FreeSWITCH-users mailing list
> TC>> Tz>> AM>> FreeSWITCH-users at lists.freeswitch.org
> TC>> Tz>> AM>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> TC>> Tz>> AM>> UNSUBSCRIBE:
> TC>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> TC>> Tz>> AM>> http://www.freeswitch.org
> TC>> Tz>> AM>>
> TC>> Tz>> AM>>
> TC>> Tz>> AM>
> TC>> Tz>> AM>
> TC>> Tz>> AM>
> TC>> Tz>>
> TC>> Tz>> C ????????? ? ? ? ? ? ? ? ? ? ? ? With Best Regards
> TC>> Tz>> ???????????? ????. ? ? ? ? ? ? ? ?Georgiewskiy Yuriy
> TC>> Tz>> +7 4872 711666 ? ? ? ? ? ? ? ? ? ?+7 4872 711666
> TC>> Tz>> ???? +7 4872 711143 ? ? ? ? ? ? ? fax +7 4872 711143
> TC>> Tz>> ???????? ??? "?? ?? ??????" ? ? ? IT Service Ltd
> TC>> Tz>> http://nkoort.ru ? ? ? ? ? ? ? ? ?http://nkoort.ru
> TC>> Tz>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> TC>> Tz>> YG129-RIPE ? ? ? ? ? ? ? ? ? ? ? ?YG129-RIPE
> TC>> Tz>> _______________________________________________
> TC>> Tz>> FreeSWITCH-users mailing list
> TC>> Tz>> FreeSWITCH-users at lists.freeswitch.org
> TC>> Tz>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> TC>> Tz>> UNSUBSCRIBE:
> TC>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> TC>> Tz>> http://www.freeswitch.org
> TC>> Tz>>
> TC>> Tz>>
> TC>> Tz>
> TC>> Tz>_______________________________________________
> TC>> Tz>FreeSWITCH-users mailing list
> TC>> Tz>FreeSWITCH-users at lists.freeswitch.org
> TC>> Tz>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> TC>> Tz>UNSUBSCRIBE:
> TC>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> TC>> Tz>http://www.freeswitch.org
> TC>> Tz>
> TC>>
> TC>> C ????????? ? ? ? ? ? ? ? ? ? ? ? With Best Regards
> TC>> ???????????? ????. ? ? ? ? ? ? ? ?Georgiewskiy Yuriy
> TC>> +7 4872 711666 ? ? ? ? ? ? ? ? ? ?+7 4872 711666
> TC>> ???? +7 4872 711143 ? ? ? ? ? ? ? fax +7 4872 711143
> TC>> ???????? ??? "?? ?? ??????" ? ? ? IT Service Ltd
> TC>> http://nkoort.ru ? ? ? ? ? ? ? ? ?http://nkoort.ru
> TC>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> TC>> YG129-RIPE ? ? ? ? ? ? ? ? ? ? ? ?YG129-RIPE
> TC>>
> TC>> _______________________________________________
> TC>> FreeSWITCH-users mailing list
> TC>> FreeSWITCH-users at lists.freeswitch.org
> TC>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> TC>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> TC>> http://www.freeswitch.org
> TC>>
> TC>>
> TC>
>
> C ????????? ? ? ? ? ? ? ? ? ? ? ? With Best Regards
> ???????????? ????. ? ? ? ? ? ? ? ?Georgiewskiy Yuriy
> +7 4872 711666 ? ? ? ? ? ? ? ? ? ?+7 4872 711666
> ???? +7 4872 711143 ? ? ? ? ? ? ? fax +7 4872 711143
> ???????? ??? "?? ?? ??????" ? ? ? IT Service Ltd
> http://nkoort.ru ? ? ? ? ? ? ? ? ?http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE ? ? ? ? ? ? ? ? ? ? ? ?YG129-RIPE
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From msc at freeswitch.org  Fri Oct  9 00:05:39 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Oct 2009 00:05:39 -0700
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <903da5680910082341r485092vb0cd10c43c4a2534@mail.gmail.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<87f2f3b90910080958i1f5cfccax621fd12100a05e3e@mail.gmail.com>
	<903da5680910081201p427a8358n768edb61a96674a4@mail.gmail.com>
	<87f2f3b90910081538l6e4e4b7dv3aedb2c4d1af58bc@mail.gmail.com>
	<903da5680910082238x27bff476s8bea0e74ab516a8d@mail.gmail.com>
	<87f2f3b90910082308i7f6144d9kc887596be54dd609@mail.gmail.com>
	<903da5680910082341r485092vb0cd10c43c4a2534@mail.gmail.com>
Message-ID: <87f2f3b90910090005i51dbfc60t20dd77789872bc9e@mail.gmail.com>

On Thu, Oct 8, 2009 at 11:41 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Fri, Oct 9, 2009 at 12:08 AM, Michael Collins <msc at freeswitch.org>
> wrote:
> > Thanks for reporting back. Please let all the Asterisk users know that
> they
> > are welcome to join us in #freeswitch on irc.freenode.net and that they
> will
> > not be abused like people do in other less friendly IRC channels.
>
> Funny you mention this.  Many people report that the way the FS
> community refers to Asterisk in docs/wikis/irc/whatever makes the FS
> camp seem *less* welcoming to them.  After all, they identify as
> Asterisk Users and take the criticism as being kinda harsh.  Most of
> them acknowledge the shortcomings of Asterisk but are put off when
> someone else points them out.  It's crazy, I know.  The thing is, I
> remember thinking that too.  After getting to know FS better, I didn't
> notice it as much.  Nobody likes to hear their baby is ugly --even if
> they know it is.
>

Interesting. I'd like to know what specifically turned them off. It could be
node 117, but that was written by Anthony in response to dozens of questions
about the subject. In any case, if the Asterisk guys look at the first
edition of the Starfish/TFOT book they'll see glowing praise of Anthony
Minessale and Brian West, so maybe they'll be willing to listen to those
guys since they've got so much Asterisk knowledge. The other thing is that
many people in the FreeSWITCH community are Asterisk refugees. They left
Asterisk for FreeSWITCH because they had some sort of trouble with Asterisk.
If that comes across in the docs then we'll have to see about cleaning that
up.

>
> At our session, and in general, I've noticed people are more
> interested in hearing about FS when you don't make direct comparisons
> to Asterisk.  Besides, FS stands on it's own merit.
>

Agreed. Direct comparisons can always be slanted in any direction. Besides,
FreeSWITCH stacks up very well against a lot of modern (i.e. expensive)
equipment put out by the big boys (Cisco, Avaya, NEC, ShoreTel, etc.).

>
> Just what I've observed *and* my 2 additional cents.
>
> Gabe
>
Thanks,
MC
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From mcampbellsmith at gmail.com  Fri Oct  9 03:38:58 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Fri, 9 Oct 2009 21:38:58 +1100
Subject: [Freeswitch-users] [ERR] mod_dingaling.c:980 Stun Failed!
In-Reply-To: <b16156850910020300x6cd7dd9cm2348942b04c79030@mail.gmail.com>
References: <33c87fa30909290415y178b6506x120a6e5c39839ce0@mail.gmail.com>
	<33c87fa30910020158q7589e18fj53bd4cc95ff3926f@mail.gmail.com>
	<b16156850910020300x6cd7dd9cm2348942b04c79030@mail.gmail.com>
Message-ID: <33c87fa30910090338w7d81b8dehc2f47059e1bf32f@mail.gmail.com>

It was the svn revision I was using... I have updated now and it is
working again.

On Fri, Oct 2, 2009 at 9:00 PM, Muhammad Shahzad
<shaheryarkh at googlemail.com> wrote:
> Yes, i had same problem, then i changed stun server to one of our own
> servers. You may try some of public stun servers listed on below link,
>
> http://www.voip-info.org/wiki/view/STUN
>
> Thank you.
>
>
> On Fri, Oct 2, 2009 at 2:58 PM, Mark Campbell-Smith
> <mcampbellsmith at gmail.com> wrote:
>>
>> Anyone have this issue?
>>
>> On Tue, Sep 29, 2009 at 9:15 PM, Mark Campbell-Smith
>> <mcampbellsmith at gmail.com> wrote:
>> > Hi!
>> >
>> > I have just started to use dingaling again, and noticed I constantly
>> > get a stun error.
>> >
>> > 2009-09-29 21:11:03.175002 [ERR] mod_dingaling.c:980 Stun Failed!
>> > stun.fwdnet.net:3478 [Remote Address Error!]
>> >
>> > I have tried with stun.freeswitch.org and stun.fwdnet.net stun servers
>> > and keep getting this error with dingaling. ?I have no problems with
>> > inbound sip calls, so I don't think ?its the actual stun server.
>> >
>> > Has anyone else seen this? ?I am using: FreeSWITCH Version 1.0.trunk
>> > (14952)
>> >
>> > Thanks!
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> ________________________________________________________
> |
> ? ? ? ? ? ? ? ? ? ? ?|
> | FATAL ERROR ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ---
> O X |
> |_______________________________________________________|
> | ? ? ? ? ? ? ? ? ? ? ? ?You have moved the mouse.
> ?|
> | Windows must be restarted for the changes to take effect. ? |
> | ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?<OK>
> ? ? ? ? ? ? ? ?|
> ####################################/
>
>
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From lakindia89 at gmail.com  Fri Oct  9 04:07:35 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Fri, 9 Oct 2009 16:37:35 +0530
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <87f2f3b90910080954l1ecb867dn8a025f5f0f701eb4@mail.gmail.com>
References: <7d79b3930910070334g53cef0c5m63037e264f1a9233@mail.gmail.com>
	<87f2f3b90910070849t79b7cb90x4150bf1dd8501ca4@mail.gmail.com>
	<7d79b3930910080241q4eeb374an2ec34642bf450818@mail.gmail.com>
	<c4d05cbe0910080802t7ebf8007m333cbd6e74efd286@mail.gmail.com>
	<87f2f3b90910080954l1ecb867dn8a025f5f0f701eb4@mail.gmail.com>
Message-ID: <7d79b3930910090407m5dc81d12pf2b0a47f6017804@mail.gmail.com>

Thanks all u guys. It got worked when I replaced with the following
configuration and the incoming has no problem with that.
[span zt PRI_1]
trunk_type => e1
b-channel => 1-15
d-channel => 16
b-channel => 17-31

But still I'm facing problem with the outgoing call. It says
INVALID_IE_CONTENTS.
What might be the issue? Even I tried the following dialplan to call by
using bridge.
<extension name="outgoing-fxo">
                  <condition field="destination_number"
expression="^(\d{10})$">
                          <action application="answer"/>
                          <action application="bridge"
data="openzap/1/1/${dialed_ext}"/>
                  </condition>
          </extension>
But it prints the same error.
What might be the issue?


On Thu, Oct 8, 2009 at 10:24 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Thu, Oct 8, 2009 at 8:02 AM, Moises Silva <moises.silva at gmail.com>wrote:
>
>> On Thu, Oct 8, 2009 at 5:41 AM, lakshmanan ganapathy <
>> lakindia89 at gmail.com> wrote:
>>
>>> Hi I tried with the following openzap.conf
>>> [span zt PRI_1]
>>> trunk_type => e1
>>> b-channel => 1:1-15,17-31
>>> d-channel => 1:16
>>>
>>
>> This does not look like a healthy config to me. You are using
>> <span>:<channels> notation in zaptel spans, zaptel channels increment across
>> spans, read point number 5 of this web page
>> http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2 , this is for another type
>> of signaling, but the span/channel numbering concepts are the same.
>>
>>
> Moy,
>
> As usual you are right on the money. I'm so used to doing Sangoma configs
> that I forget about the zaptel syntax. I recommend this config:
>
> [span zt PRI_1]
> trunk_type => e1
> b-channel => 1-15
> d-channel => 16
> b-channel => 17-31
>
>
> Give that a try and report back if you have issues.
> -MC
>
>>  --
>> Moises Silva
>> Software Developer
>> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
>> 9T3 Canada
>> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From srinivas.ksvreddy at gmail.com  Fri Oct  9 04:58:28 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Fri, 9 Oct 2009 17:28:28 +0530
Subject: [Freeswitch-users] apr_queue
Message-ID: <f8af5740910090458w6540db9fg972ad1ba8a691c30@mail.gmail.com>

Hi all,

does any know about How apr_queue is maintaing and retriving all registered
and all stuff....


-- 
Srinivasula Reddy K
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From lakindia89 at gmail.com  Fri Oct  9 04:59:50 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Fri, 9 Oct 2009 17:29:50 +0530
Subject: [Freeswitch-users] libpri_span vs prispans
Message-ID: <7d79b3930910090459v2b07e3aapd715c7618f11ada0@mail.gmail.com>

Hi all,

What difference it will make, if I use pri_span configuration and lib_pri
span configuration in openzap.conf.xml.
I'm preety much confused on the difference between this.
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From rob4manhere at gmail.com  Fri Oct  9 05:22:35 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Fri, 9 Oct 2009 07:22:35 -0500
Subject: [Freeswitch-users] mod_fax compile fails
In-Reply-To: <20091009061130.GA7509@jdc.jasonjgw.net>
References: <33c87fa30910082006v1e91849fjbf4282541e7ac377@mail.gmail.com>
	<343C884A-9AB5-4755-9C5F-42ED705D05E7@gmail.com>
	<33c87fa30910082243h6c689816i23368bedfdfd14c0@mail.gmail.com>
	<20091009061130.GA7509@jdc.jasonjgw.net>
Message-ID: <826997E7-F76D-440C-BF70-ECE21B9F1E63@gmail.com>

I was in a production window so when the latest trunk worked I moved  
on.  I went back to troubleshoot later when I saw your email but I  
couldn't reproduce it.

I like the tmpfs build- thanks for the tip Jason.


On Oct 9, 2009, at 1:11 AM, Jason White wrote:

> Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
>> Thanks Rob,
>>
>> Is this a fault in the svn update process?
>
> Usually, no, it's rather that if you build in a directory that  
> you've checked
> out and then run svn update, not everything gets cleaned up properly  
> the next
> time you build.
>
> I find it's faster to build in a tmpfs file system anyway, e.g.,
>
> mkdir /tmp/fs && sudo mount -t tmpfs tmpfs /tmpfs && svn export . / 
> tmp/fs
> then go into /tmp/fs and run the compilation process, or the package  
> building
> command, or whatever you need.
>
> This way, my original svn tree is never altered and it will always  
> update
> cleanly.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From Russell.Mosemann at cune.org  Fri Oct  9 05:36:15 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Fri, 9 Oct 2009 12:36:15 -0000
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <7d79b3930910090407m5dc81d12pf2b0a47f6017804@mail.gmail.com>
Message-ID: <20091009123615.BC50A3FA6E9@mail.cune.org>

lakshmanan ganapathy <lakindia89 at gmail.com> said:

> But still I'm facing problem with the outgoing call. It says
> INVALID_IE_CONTENTS.
> What might be the issue? Even I tried the following dialplan to call by
> using bridge.
> <extension name="outgoing-fxo">
>                   <condition field="destination_number"
> expression="^(\d{10})$">
>                           <action application="answer"/>
>                           <action application="bridge"
> data="openzap/1/1/${dialed_ext}"/>

Does "answer" need to be called here? I haven't used an fxo. So, I don't
know. What value does $dialed_ext have? If you want to use the number
matched in the condition, then it should be

openzap/1/1/$1

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!




From dujinfang at gmail.com  Fri Oct  9 05:43:45 2009
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 9 Oct 2009 20:43:45 +0800
Subject: [Freeswitch-users] FreeSWITCH Update: Valet Parking
In-Reply-To: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>
References: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>
Message-ID: <23f91030910090543o1f70c997t7d860d4d65b2a175@mail.gmail.com>

It's very cool. But think about the following scenario:

Alice answers a call from Bob.
Alice transfers Bob to 6001 and hangs up.
Bob is now in parking stall 6001, hearing MOH.
Alice calls Charlie and tells him that he has a call parked in 6001.

  Dee call in and answered by Alice  # at that time, Alice don't know the
lot is empty or not
  If Alice transfer Dee to 6001 # then Dee will talk to Bob, that's not
expected

Charlie calls 6000, waits for prompt, then dials 6001#.
Charlie Listening to music.............

Event it's rarely happen, I do think Alice would like to now the if lot is
empty or not before transfer. Make sense?

2009/10/9 Michael Collins <msc at freeswitch.org>

> FYI,
>
> The FreeSWITCH devs have added valet parking! Check it out:
> http://www.freeswitch.org/node/207
>
> Let us know what you think.
>
> Thanks,
> Michael
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Russell.Mosemann at cune.org  Fri Oct  9 05:57:14 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Fri, 9 Oct 2009 12:57:14 -0000
Subject: [Freeswitch-users] libpri_span vs prispans
In-Reply-To: <7d79b3930910090459v2b07e3aapd715c7618f11ada0@mail.gmail.com>
Message-ID: <20091009125714.E0FF73043D7@mail.cune.org>

lakshmanan ganapathy <lakindia89 at gmail.com> said:
> What difference it will make, if I use pri_span configuration and lib_pri
> span configuration in openzap.conf.xml.
> I'm preety much confused on the difference between this.

Openzap can handle a T1/E1 itself, or openzap can use libpri to do that.
libpri is more mature than openzap. In my case openzap was not able to
handle everything invovling a T1 here, and using libpri fixed it.

A libpri_span configuration is handled by libpri. A pri_span
configuration is handled natively by openzap. Use the one that works for you.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!




From rupa at rupa.com  Fri Oct  9 06:24:39 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 9 Oct 2009 07:24:39 -0600
Subject: [Freeswitch-users] FreeSWITCH Update: Valet Parking
In-Reply-To: <23f91030910090543o1f70c997t7d860d4d65b2a175@mail.gmail.com>
References: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>
	<23f91030910090543o1f70c997t7d860d4d65b2a175@mail.gmail.com>
Message-ID: <b39273530910090624l5aa8a507s2be37f833e19a5a1@mail.gmail.com>

Perhaps a "put in next empty slot" mode and then a say of the slot so
Alice and make a note of the lot #?

On Fri, Oct 9, 2009 at 6:43 AM, Seven Du <dujinfang at gmail.com> wrote:
> It's very cool. But think about the following scenario:
>
> Alice answers a call from Bob.
> Alice transfers Bob to 6001 and hangs up.
> Bob is now in parking stall 6001, hearing MOH.
> Alice calls Charlie and tells him that he has a call parked in 6001.
>
> ? Dee call in and answered by Alice? # at that time, Alice don't know the
> lot is empty or not
> ? If Alice transfer Dee to 6001 # then Dee will talk to Bob, that's not
> expected
>
> Charlie calls 6000, waits for prompt, then dials 6001#.
> Charlie Listening to music.............
>
> Event it's rarely happen, I do think Alice would like to now the if lot is
> empty or not before transfer. Make sense?
>
> 2009/10/9 Michael Collins <msc at freeswitch.org>
>>
>> FYI,
>>
>> The FreeSWITCH devs have added valet parking! Check it out:
>> http://www.freeswitch.org/node/207
>>
>> Let us know what you think.
>>
>> Thanks,
>> Michael
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa



From andy at fabulous4.co.uk  Fri Oct  9 07:12:51 2009
From: andy at fabulous4.co.uk (Andy)
Date: Fri, 9 Oct 2009 15:12:51 +0100
Subject: [Freeswitch-users] NAT problems - sorry
In-Reply-To: <0A2A7A27-611E-40FA-8268-D2C451AA8B77@freeswitch.org>
References: <5BE289D2AE22492BA71A6919C8A440B3@D810><5A70A0D4-70FA-43B2-BAF5-F8D6ECC64838@freeswitch.org><60E56B6B355C45BCA0D31F4425B9173A@D810><6B2B7FFD-75BC-44CE-A11C-2B26447DA889@freeswitch.org><703EEB9AC9AD43898C6C46F82904E6A9@D810>
	<0A2A7A27-611E-40FA-8268-D2C451AA8B77@freeswitch.org>
Message-ID: <9B8EDDA782464750A4AAC1F947B56A4C@D810>

Hi folks,
 
Just thought I'd post the solution to this problem. In my case the problem
was that my config was based on an older version and so was missing a
crucial parameter that exists in the default config wirth a new install.
 
The external.xml profile needed to contain the parameter:
 
<param name="local-network-acl" value="localnet.auto"/>
 
With this in place, DTMF and hangup messages traverse the nat firewall
correctly. Without it they don't. I searched the Wiki and couldn't find any
info on this parameter. Can anyone provide a description of what it does and
why it's significant that can be added to the WIKI?
 
Cheers
Andy

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: 07 October 2009 19:07
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] NAT problems - sorry


s/auto-nat/$realip/ then forward the rtp ports and sip ports. 

/b
PS chances are you have to ENABLE upnp.

On Oct 7, 2009, at 12:58 PM, Andy wrote:


Many thanks Brian, the firewall docs assure me it is uPnp but is probably
lying or a poor implementation. Could you point me to the right section of
the Wiki to tell me how to do this manually as I've been scouting for some
time and can;t seem to find the right thing. sorry if I'm being blind.



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From anthony.minessale at gmail.com  Fri Oct  9 07:25:33 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 9 Oct 2009 09:25:33 -0500
Subject: [Freeswitch-users] FreeSWITCH Update: Valet Parking
In-Reply-To: <23f91030910090543o1f70c997t7d860d4d65b2a175@mail.gmail.com>
References: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>
	<23f91030910090543o1f70c997t7d860d4d65b2a175@mail.gmail.com>
Message-ID: <191c3a030910090725o4150a17x56b93303bf0b8c2b@mail.gmail.com>

or twist that into a feature... =D

alice is bob's secretary so she calls him on his cell when he has an appt
and parks him.
Then she calls the other guy and says "I have bob on line for you" and xfers
him to the same parking space =p


On Fri, Oct 9, 2009 at 7:43 AM, Seven Du <dujinfang at gmail.com> wrote:

> It's very cool. But think about the following scenario:
>
> Alice answers a call from Bob.
> Alice transfers Bob to 6001 and hangs up.
> Bob is now in parking stall 6001, hearing MOH.
> Alice calls Charlie and tells him that he has a call parked in 6001.
>
>   Dee call in and answered by Alice  # at that time, Alice don't know the
> lot is empty or not
>   If Alice transfer Dee to 6001 # then Dee will talk to Bob, that's not
> expected
>
> Charlie calls 6000, waits for prompt, then dials 6001#.
> Charlie Listening to music.............
>
> Event it's rarely happen, I do think Alice would like to now the if lot is
> empty or not before transfer. Make sense?
>
> 2009/10/9 Michael Collins <msc at freeswitch.org>
>
>> FYI,
>>
>> The FreeSWITCH devs have added valet parking! Check it out:
>> http://www.freeswitch.org/node/207
>>
>> Let us know what you think.
>>
>> Thanks,
>> Michael
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From maciej.aniserowicz at gmail.com  Fri Oct  9 07:40:32 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Fri, 9 Oct 2009 07:40:32 -0700 (PDT)
Subject: [Freeswitch-users] gateway FS informs it's client FS about
 users hanged up with a long delay
In-Reply-To: <87f2f3b90910081000v5b505f4ao53104211b620ea77@mail.gmail.com>
References: <1A16501E57CA4727A593226A8C810308@procent>
	<98F72CCB-88B9-47ED-AED3-B6BA6DE648C0@jerris.com>
	<1255007884165-3787956.post@n2.nabble.com>
	<87f2f3b90910081000v5b505f4ao53104211b620ea77@mail.gmail.com>
Message-ID: <1255099232783-3795011.post@n2.nabble.com>


Hello,
The issue is resolved. I feel stupid, because Michael Jerris was right the first time. Setting external_rtp_ip and external_sip_ip to $${local_ip_v4} made it work.
But the strange thing is: it SOMETIMES worked before without any delay, which 'should not be possible', because the original IP was my external ip and the BYE message was sent straight to it. And there is no way it could reach the target 'internal' FS, because it runs on virtual machine, and no ports are forwarded on my router.
Any thoughts? Why this could (rarely) work even with the previous config?

Thanks to both of you for your answers.

MA


  ----- Original Message ----- 
  From: mercutioviz (via Nabble) 
  To: Maciej Aniserowicz 
  Sent: Thursday, October 08, 2009 7:06 PM
  Subject: Re: [Freeswitch-users] gateway FS informs it's client FS about users hanged up with a long delay





  On Thu, Oct 8, 2009 at 6:18 AM, Maciej Aniserowicz <[hidden email]> wrote:


    Both of the instances are run on the same machine, i just changed the default
    ports they use. Can anything else cause this strange behavior?
    MA

  Did a packet capture yield any clues? That is, were you able to confirm that each instance sent and received all the packets that you believe they should have sent and received? The reason I ask is so that you don't end up chasing a ghost because you made an assumption somewhere in your troubleshooting.

  -MC
  ? 




    Michael Jerris wrote:
    >
    > Incorrect NAT configuration so one of the boxes is not actually
    > getting a BYE.
    >
    >
    > On Oct 5, 2009, at 3:13 AM, Maciej Aniserowicz wrote:
    >
    >> Hi,
    >> When I use two FreeSWITCH instances ('internal' and 'external'), all
    >> users register to the 'external' instance which acts as a gateway by
    >> 'internal' instance (which in turn is controlled by my applicaiton
    >> with commands sent by socket).
    >> When user hangs up, the 'hanged up' event is propagated to the
    >> 'internal' instance after a long time (~3 minutes) instead of being
    >> propagated immediately.
    >> What can cause this issue?
    >
    >
    > _______________________________________________
    > FreeSWITCH-users mailing list
    > [hidden email]
    > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    > http://www.freeswitch.org
    >
    >

    --
    View this message in context: http://n2.nabble.com/gateway-FS-informs-it-s-client-FS-about-users-hanged-up-with-a-long-delay-tp3768540p3787956.html
    Sent from the freeswitch-users mailing list archive at Nabble.com.

    _______________________________________________
    FreeSWITCH-users mailing list
    [hidden email]
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org



  _______________________________________________ 
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  http://www.freeswitch.org



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-- 
View this message in context: http://n2.nabble.com/gateway-FS-informs-it-s-client-FS-about-users-hanged-up-with-a-long-delay-tp3768540p3795011.html
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From rupa at rupa.com  Fri Oct  9 07:50:33 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 9 Oct 2009 08:50:33 -0600
Subject: [Freeswitch-users] FreeSWITCH Update: Valet Parking
In-Reply-To: <191c3a030910090725o4150a17x56b93303bf0b8c2b@mail.gmail.com>
References: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>
	<23f91030910090543o1f70c997t7d860d4d65b2a175@mail.gmail.com>
	<191c3a030910090725o4150a17x56b93303bf0b8c2b@mail.gmail.com>
Message-ID: <b39273530910090750ga9debecp783d374560e8c577@mail.gmail.com>

Oh, definitely.  I think both use cases (existing, and park in next
avail slot and read back slot #) are useful.

On Fri, Oct 9, 2009 at 8:25 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> or twist that into a feature... =D
>
> alice is bob's secretary so she calls him on his cell when he has an appt
> and parks him.
> Then she calls the other guy and says "I have bob on line for you" and xfers
> him to the same parking space =p



-- 
-Rupa



From leo.zibi at gmail.com  Fri Oct  9 00:35:23 2009
From: leo.zibi at gmail.com (leo.zibi at gmail.com)
Date: Fri, 09 Oct 2009 09:35:23 +0200
Subject: [Freeswitch-users] mod_fax compile fails
In-Reply-To: <33c87fa30910082006v1e91849fjbf4282541e7ac377@mail.gmail.com>
References: <33c87fa30910082006v1e91849fjbf4282541e7ac377@mail.gmail.com>
Message-ID: <4ACEE7BB.1040606@gmail.com>

Hi,

./bootstrap.sh
./configure
make
make install

--
Regards,
Leo



Mark Campbell-Smith wrote:
> HI all,
>
> I just tried to update to the latest svn and I get these errors right
> at the end after issuing a 'make current'.  I am using Debian Lenny.
>
> making all mod_fax
> make[5]: Entering directory `/home/mark/freeswitch/src/mod/applications/mod_fax'
> make[6]: Entering directory `/home/mark/freeswitch/src/mod/applications/mod_fax'
> make[7]: Entering directory `/home/mark/freeswitch/libs/tiff-3.8.2'
> running /bin/sh ./configure  --prefix=/usr/local/freeswitch
> --cache-file=/dev/null --srcdir=. --disable-shared --with-pic
> --no-create --no-recursion
> configure: error: cannot run /bin/sh config/config.sub
> make[7]: *** [config.status] Error 1
> make[7]: Leaving directory `/home/mark/freeswitch/libs/tiff-3.8.2'
> make[6]: *** [../../../../libs/tiff-3.8.2/libtiff/libtiff.la] Error 2
> make[6]: Leaving directory `/home/mark/freeswitch/src/mod/applications/mod_fax'
> make[5]: *** [all] Error 1
> make[5]: Leaving directory `/home/mark/freeswitch/src/mod/applications/mod_fax'
> make[4]: *** [mod_fax-all] Error 1
> make[4]: Leaving directory `/home/mark/freeswitch/src/mod'
> make[3]: *** [all-recursive] Error 1
> make[3]: Leaving directory `/home/mark/freeswitch/src'
> Making all in build
> make[3]: Entering directory `/home/mark/freeswitch/build'
>  +-------- FreeSWITCH Build Complete -----------+
>  + FreeSWITCH has been successfully built.      +
>  + Install by running:                          +
>  +                                              +
>  +               make install                   +
>  +----------------------------------------------+
> make[3]: Leaving directory `/home/mark/freeswitch/build'
> make[2]: *** [all-recursive] Error 1
> make[2]: Leaving directory `/home/mark/freeswitch'
> make[1]: *** [all] Error 2
> make[1]: Leaving directory `/home/mark/freeswitch'
> make: *** [current] Error 2
>
> Also, are the 'Leaving directory / all-recursive' errors going to
> cause a problem?
>
> Thanks!
> Any ideas what the cause is?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   




From jerry.richards at teotech.com  Fri Oct  9 08:23:27 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 9 Oct 2009 08:23:27 -0700
Subject: [Freeswitch-users] SLAs and BLAs
References: <5B5F50E0B1D34BC0BCBCC05E1FE05C8A@greyhawk.tonecommander.com>
	<3F7AC380-CACB-432F-8B94-80FBA18628C0@freeswitch.org> 
Message-ID: <5CC2E91025314615ACBB65E029D5CB12@greyhawk.tonecommander.com>


I gather from the mailing archive that BLAs are implemented using the
draft-anil-sipping-bla-04.txt document.  According to the draft, the
Appearance Agent is supposed to initiate a SUBSCRIBE request, but I don't
see FS doing this.

What phone types/models are known to work with the FS BLA implementation?

Best Regards,
Jerry
 

-----Original Message-----
From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Monday, October 05, 2009 3:24 PM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: [Freeswitch-users] SLAs and BLAs

We are building our own in-house developed Teo phones.  I also have
CounterPath's Bria Professional phone.  For test purposes, I have one snom
phone and a couple Polycomm phones.

Jerry
 

-----Original Message-----
From: Brian West [mailto:brian at freeswitch.org]
Sent: Monday, October 05, 2009 11:02 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] SLAs and BLAs

First off what phones are you going to be using?

/b

On Oct 5, 2009, at 12:58 PM, Jerry Richards wrote:

>
> I can see how BLFs and Presence are managed, however I haven't found 
> much documentation on SLAs and BLAs.  What is the RFC(s) that 
> Freeswitch used to implement SLAs and BLAs?  Do they differ from BLFs?
>
> Best Regards,
> Jerry







From anthony.minessale at gmail.com  Fri Oct  9 09:02:24 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 9 Oct 2009 11:02:24 -0500
Subject: [Freeswitch-users] Heartbeat question
In-Reply-To: <86a32abc0910082245j36b8b3e9ncaf1cdd358fce319@mail.gmail.com>
References: <86a32abc0910081956k91b3600sb8c556e589f5b089@mail.gmail.com>
	<86a32abc0910082003v3f5d4e6dua8e7eae18fd2f939@mail.gmail.com>
	<6b65470d0910082036g204d5dcbw26e1e43ef80c7c84@mail.gmail.com>
	<86a32abc0910082245j36b8b3e9ncaf1cdd358fce319@mail.gmail.com>
Message-ID: <191c3a030910090902t38282f72l6b6d1733a758836@mail.gmail.com>

Update to trunk and try it with fs_cli it for sure will let you do every 1
second

in fs_cli type

/events plain all

if you make that call you will see one every 1 second


On Fri, Oct 9, 2009 at 12:45 AM, Diego Viola <diego.viola at gmail.com> wrote:

> Nope, I was just wondering why it didn't work at 1 second exactly...
>
>
> On Fri, Oct 9, 2009 at 3:36 AM, William Suffill <william.suffill at gmail.com
> > wrote:
>
>> Why do you need it every second? If you want real time channel counts
>> you would  be able to track each create/destroy even instead of
>> relying on the heartbeat summary.
>>
>> -- W
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Fri Oct  9 09:07:56 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Oct 2009 09:07:56 -0700
Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Starting,
	Please Join Us!
Message-ID: <87f2f3b90910090907v6a01bae2rd74f577d482dddf6@mail.gmail.com>

Agenda and call-in info: http://bit.ly/lzEYy
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From mike at jerris.com  Fri Oct  9 10:25:40 2009
From: mike at jerris.com (Michael Jerris)
Date: Fri, 9 Oct 2009 13:25:40 -0400
Subject: [Freeswitch-users] apr_queue
In-Reply-To: <f8af5740910090458w6540db9fg972ad1ba8a691c30@mail.gmail.com>
References: <f8af5740910090458w6540db9fg972ad1ba8a691c30@mail.gmail.com>
Message-ID: <91049B87-E571-4181-B436-6DAD065A6516@jerris.com>

On Oct 9, 2009, at 7:58 AM, srinivasula reddy wrote:

> Hi all,
>
> does any know about How apr_queue is maintaing and retriving all  
> registered and all stuff....
>

parse error




From msc at freeswitch.org  Fri Oct  9 10:57:04 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Oct 2009 10:57:04 -0700
Subject: [Freeswitch-users] libpri_span vs prispans
In-Reply-To: <20091009125714.E0FF73043D7@mail.cune.org>
References: <7d79b3930910090459v2b07e3aapd715c7618f11ada0@mail.gmail.com>
	<20091009125714.E0FF73043D7@mail.cune.org>
Message-ID: <87f2f3b90910091057q3a033548j462446cf424ad257@mail.gmail.com>

On Fri, Oct 9, 2009 at 5:57 AM, <Russell.Mosemann at cune.org> wrote:

> lakshmanan ganapathy <lakindia89 at gmail.com> said:
> > What difference it will make, if I use pri_span configuration and lib_pri
> > span configuration in openzap.conf.xml.
> > I'm preety much confused on the difference between this.
>
> Openzap can handle a T1/E1 itself, or openzap can use libpri to do that.
> libpri is more mature than openzap. In my case openzap was not able to
> handle everything invovling a T1 here, and using libpri fixed it.
>
> A libpri_span configuration is handled by libpri. A pri_span
> configuration is handled natively by openzap. Use the one that works for
> you.
>

Right now the libpri stack is more mature than the ozmod_isdn stuff. See the
OpenZAP page on the wiki for instructions on how to set up ozmod_libpri.
-MC
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From msc at freeswitch.org  Fri Oct  9 11:01:34 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Oct 2009 11:01:34 -0700
Subject: [Freeswitch-users] FreeSWITCH Update: Valet Parking
In-Reply-To: <b39273530910090750ga9debecp783d374560e8c577@mail.gmail.com>
References: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>
	<23f91030910090543o1f70c997t7d860d4d65b2a175@mail.gmail.com>
	<191c3a030910090725o4150a17x56b93303bf0b8c2b@mail.gmail.com>
	<b39273530910090750ga9debecp783d374560e8c577@mail.gmail.com>
Message-ID: <87f2f3b90910091101x33bfbd8vf504b8bb3186da08@mail.gmail.com>

On Fri, Oct 9, 2009 at 7:50 AM, Rupa Schomaker <rupa at rupa.com> wrote:

> Oh, definitely.  I think both use cases (existing, and park in next
> avail slot and read back slot #) are useful.
>
>
They do have some usefulness. The most important reason for this feature,
though, is for the person who handles calls all day and moves them around.
That person will most likely have a pretty good idea of which parking stalls
are available. However, I do like the idea of a webby interface showing what
stalls are in use...
-MC


> On Fri, Oct 9, 2009 at 8:25 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > or twist that into a feature... =D
> >
> > alice is bob's secretary so she calls him on his cell when he has an appt
> > and parks him.
> > Then she calls the other guy and says "I have bob on line for you" and
> xfers
> > him to the same parking space =p
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kristian.kielhofner at gmail.com  Fri Oct  9 12:10:36 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 9 Oct 2009 15:10:36 -0400
Subject: [Freeswitch-users] On the handling of SIP headers
Message-ID: <2d9149cd0910091210s33fb2bb9oc0c94a64aa87e0be@mail.gmail.com>

Hello everyone,

  In using FS for various scenarios I've noticed some behavior that
I'm not sure is completely "proper".  Given that this probably lives
in mod_sofia who knows what's really "proper".  It is SIP after all...

  So the issue comes up when using FreeSWITCH as a B2BUA and bridging
between endpoints (very common).  Should FreeSWITCH copy the X-
headers (possibly others) as it does now?  I'd like to think it
shouldn't by default and the behavior should be one of:

1)  Don't pass X-* (or anything else, really) from one leg to another.
 If you want to pass specific X- headers (or anything else), set them
explicitly on the outbound leg.
2)  Make the behavior configurable with a channel variable and/or
sofia config option:

{sip_pass_headers=all|none|X-MyCustomHeaderByName}

  Thoughts?

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From mattdfong at gmail.com  Fri Oct  9 12:34:01 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Sat, 10 Oct 2009 02:34:01 +0700
Subject: [Freeswitch-users] Sending an Event to a Session for onInput
Message-ID: <4256bf830910091234t11f4f4d0w28283f341bd1fb06@mail.gmail.com>

I'm used to using the onInput callbacks inside lua and javascript to listen
for dtmf and other events and perform a task accordingly. I'm wondering if
there is a way to send an event to a session or channel that can be caught
using the setInputCallback inside lua from outside the session program.
Maybe an API command that can generate an event for a specific UUID. Does a
mechanism exist to do this that I'm over looking? Thanks.
--matt
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From jerry.richards at teotech.com  Fri Oct  9 13:06:43 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 9 Oct 2009 13:06:43 -0700
Subject: [Freeswitch-users] FW: FS Does Not Relay
	PresencePUBLISHToSubscribing Phones
Message-ID: <6E290A3B014040CFA97E8BE97BC36BAC@greyhawk.tonecommander.com>

I put the sqlite3 select query in the paste bin again, and prior to that, I
entered the .dump command.  The select command came back with the sqlite3
prompt, which I guess means it didn't find an entry.  How do I go about
isolating this problem?  I'm using CounterPath's Bria Professional
softphone.  They are the same company that make the Eyebeam.
 
Any ideas?
 
Best Regards,
Jerry


  _____  

From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Friday, October 02, 2009 11:28 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: [Freeswitch-users] FS Does Not Relay
PresencePUBLISHToSubscribing Phones


I put the sqlite3 select query in the paste bin, and prior to that, I
entered the .dump command.  The select command came back with a "...>"
prompt which I don't understand.  I don't know enough about sqlite3 to know
what that means?
 
Best Regards,
Jerry


  _____  

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Friday, October 02, 2009 10:52 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Does Not Relay
PresencePUBLISHToSubscribing Phones


connect to sqlite directly with sqlite3 app and try that sql stmt and see
why it doesn't match anything.

sqlite3 /usr/local/freeswitch/db/sofia_reg_internal.db

select
sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_hos
t,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscripti
ons.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscripti
ons.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subsc
riptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,
'Away','away','192.168.72.38',sip_presence.status,sip_presence.rpid from
sip_subscriptions left join sip_presence on
(sip_subscriptions.sub_to_user=sip_presence.sip_user and
sip_subscriptions.sub_to_host=sip_presence.sip_host and
sip_subscriptions.profile_name=sip_presence.profile_name) where
(event='presence' or event='presence') and sub_to_user='1001' and
(sub_to_host='192.168.72.38' or presence_hosts like '%192.168.72.38%') and
(sip_subscriptions.profile_name = 'external' or
sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host)



On Fri, Oct 2, 2009 at 12:12 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:


Okay, I put a log up on the pastebin that shows the PUBLISH event coming
from a CounterPath Bria Professional phone.  For some reason, FS is getting
an error and not relaying the presence status to the subscriber.
 
Best Regards,
Jerry


  _____  

From: Jo?o Mesquita [mailto:jmesquita at freeswitch.org] 
Sent: Thursday, October 01, 2009 8:14 PM 

To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Does Not Relay Presence
PUBLISHToSubscribing Phones


Piece of advice, don't ask, just do it. ;)

jmesquita


On Thu, Oct 1, 2009 at 3:29 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:


If you have time to take a look, I could put a trace in the pastebin?
 
Jerry


  _____  


From: Jerry Richards [mailto:jerry.richards at teotech.com] 

Sent: Thursday, October 01, 2009 10:29 AM 

To: 'freeswitch-users at lists.freeswitch.org'

Subject: RE: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
ToSubscribing Phones


I am using two Bria Professional Version 2.5.4 Build 54835 softphones.
 
Thanks,
Jerry


  _____  


From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 

Sent: Thursday, October 01, 2009 9:36 AM 

To: freeswitch-users at lists.freeswitch.org

Subject: Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
ToSubscribing Phones


which phone is it,
we tested it with eyebeam and it appears to work for us.



On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards <jerry.richards at teotech.com>
wrote:



By the way, I see the following lines at the FS console, which might be a
clue as to why this is happening.  Could someone point me toward what might
cause this?  I set the "manage-presence" parameter to "true" in each XML
file where I saw it defined.

[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[WARNING] sofia_presence.c:565  192.168.72.38 is an alias, skipping


Best Regards,
Jerry



-----Original Message-----
From: Jerry Richards [mailto:jerry.richards at teotech.com]
Sent: Wednesday, September 30, 2009 9:12 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones

I have two phones configured to subscribe to each other's presence status.
When I change the presence status in one phone, I see the SIP PUBLISH
message going to FS, but I don't see FS relaying that presence status to the
subscribing phone.  Does anyone know why?

Best Regards,
Jerry


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:213-799-1400



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org






-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:213-799-1400


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From andrew at hijacked.us  Fri Oct  9 13:24:10 2009
From: andrew at hijacked.us (Andrew Thompson)
Date: Fri, 9 Oct 2009 16:24:10 -0400
Subject: [Freeswitch-users] FreeSWITCH Update: Valet Parking
In-Reply-To: <87f2f3b90910091101x33bfbd8vf504b8bb3186da08@mail.gmail.com>
References: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>
	<23f91030910090543o1f70c997t7d860d4d65b2a175@mail.gmail.com>
	<191c3a030910090725o4150a17x56b93303bf0b8c2b@mail.gmail.com>
	<b39273530910090750ga9debecp783d374560e8c577@mail.gmail.com>
	<87f2f3b90910091101x33bfbd8vf504b8bb3186da08@mail.gmail.com>
Message-ID: <20091009202409.GQ4572@hijacked.us>

On Fri, Oct 09, 2009 at 11:01:34AM -0700, Michael Collins wrote:
> On Fri, Oct 9, 2009 at 7:50 AM, Rupa Schomaker <rupa at rupa.com> wrote:
> 
> > Oh, definitely.  I think both use cases (existing, and park in next
> > avail slot and read back slot #) are useful.
> >
> >
> They do have some usefulness. The most important reason for this feature,
> though, is for the person who handles calls all day and moves them around.
> That person will most likely have a pretty good idea of which parking stalls
> are available. However, I do like the idea of a webby interface showing what
> stalls are in use...

mod_snom and the programmable LEDs would be a handy way to do it too. I
implemented a poor-man's version of a key system using that, but I think
I'll replace the parking portion with the valet parking stuff when I
get a chance.

Andrew



From diego.viola at gmail.com  Fri Oct  9 14:27:20 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 9 Oct 2009 21:27:20 +0000
Subject: [Freeswitch-users] Heartbeat question
In-Reply-To: <191c3a030910090902t38282f72l6b6d1733a758836@mail.gmail.com>
References: <86a32abc0910081956k91b3600sb8c556e589f5b089@mail.gmail.com>
	<86a32abc0910082003v3f5d4e6dua8e7eae18fd2f939@mail.gmail.com>
	<6b65470d0910082036g204d5dcbw26e1e43ef80c7c84@mail.gmail.com>
	<86a32abc0910082245j36b8b3e9ncaf1cdd358fce319@mail.gmail.com>
	<191c3a030910090902t38282f72l6b6d1733a758836@mail.gmail.com>
Message-ID: <86a32abc0910091427i2395fd80t3bc557186ece3972@mail.gmail.com>

Thanks Anthony, this solved it. You rock :)

My program now outputs:

Got a SESSION_HEARTBEAT at 17:14:59
Got a SESSION_HEARTBEAT at 17:15:00
Got a SESSION_HEARTBEAT at 17:15:02
Got a SESSION_HEARTBEAT at 17:15:03
Got a SESSION_HEARTBEAT at 17:15:04
Got a SESSION_HEARTBEAT at 17:15:05
Got a SESSION_HEARTBEAT at 17:15:06
Got a SESSION_HEARTBEAT at 17:15:07
Got a SESSION_HEARTBEAT at 17:15:08
Got a SESSION_HEARTBEAT at 17:15:09
Got a SESSION_HEARTBEAT at 17:15:10
Got a SESSION_HEARTBEAT at 17:15:11
Got a SESSION_HEARTBEAT at 17:15:12
Got a SESSION_HEARTBEAT at 17:15:13
Got a SESSION_HEARTBEAT at 17:15:14
Got a SESSION_HEARTBEAT at 17:15:15
Got a SESSION_HEARTBEAT at 17:15:16
Got a SESSION_HEARTBEAT at 17:15:17
Got a SESSION_HEARTBEAT at 17:15:18
Got a SESSION_HEARTBEAT at 17:15:19
Got a SESSION_HEARTBEAT at 17:15:20
Got a SESSION_HEARTBEAT at 17:15:21
Got a SESSION_HEARTBEAT at 17:15:22
Got a SESSION_HEARTBEAT at 17:15:23
Got a SESSION_HEARTBEAT at 17:15:24
Got a SESSION_HEARTBEAT at 17:15:25
Got a SESSION_HEARTBEAT at 17:15:26
Got a SESSION_HEARTBEAT at 17:15:27
Got a SESSION_HEARTBEAT at 17:15:28
Got a SESSION_HEARTBEAT at 17:15:29
Got a SESSION_HEARTBEAT at 17:15:30




On Fri, Oct 9, 2009 at 4:02 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Update to trunk and try it with fs_cli it for sure will let you do every 1
> second
>
> in fs_cli type
>
> /events plain all
>
> if you make that call you will see one every 1 second
>
>
>
> On Fri, Oct 9, 2009 at 12:45 AM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> Nope, I was just wondering why it didn't work at 1 second exactly...
>>
>>
>> On Fri, Oct 9, 2009 at 3:36 AM, William Suffill <
>> william.suffill at gmail.com> wrote:
>>
>>> Why do you need it every second? If you want real time channel counts
>>> you would  be able to track each create/destroy even instead of
>>> relying on the heartbeat summary.
>>>
>>> -- W
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From diego.viola at gmail.com  Fri Oct  9 14:30:12 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 9 Oct 2009 21:30:12 +0000
Subject: [Freeswitch-users] Heartbeat question
In-Reply-To: <86a32abc0910091427i2395fd80t3bc557186ece3972@mail.gmail.com>
References: <86a32abc0910081956k91b3600sb8c556e589f5b089@mail.gmail.com>
	<86a32abc0910082003v3f5d4e6dua8e7eae18fd2f939@mail.gmail.com>
	<6b65470d0910082036g204d5dcbw26e1e43ef80c7c84@mail.gmail.com>
	<86a32abc0910082245j36b8b3e9ncaf1cdd358fce319@mail.gmail.com>
	<191c3a030910090902t38282f72l6b6d1733a758836@mail.gmail.com>
	<86a32abc0910091427i2395fd80t3bc557186ece3972@mail.gmail.com>
Message-ID: <86a32abc0910091430hc151ed8of085fcc3af2f1de4@mail.gmail.com>

Here is on two seconds ;)

Got a SESSION_HEARTBEAT at 17:17:13
Got a SESSION_HEARTBEAT at 17:17:15
Got a SESSION_HEARTBEAT at 17:17:17
Got a SESSION_HEARTBEAT at 17:17:19
Got a SESSION_HEARTBEAT at 17:17:21
Got a SESSION_HEARTBEAT at 17:17:23
Got a SESSION_HEARTBEAT at 17:17:25
Got a SESSION_HEARTBEAT at 17:17:27
Got a SESSION_HEARTBEAT at 17:17:29
Got a SESSION_HEARTBEAT at 17:17:31
Got a SESSION_HEARTBEAT at 17:17:33
Got a SESSION_HEARTBEAT at 17:17:35
Got a SESSION_HEARTBEAT at 17:17:37
Got a SESSION_HEARTBEAT at 17:17:39
Got a SESSION_HEARTBEAT at 17:17:41
Got a SESSION_HEARTBEAT at 17:17:43
Got a SESSION_HEARTBEAT at 17:17:45
Got a SESSION_HEARTBEAT at 17:17:47
Got a SESSION_HEARTBEAT at 17:17:49
Got a SESSION_HEARTBEAT at 17:17:51
Got a SESSION_HEARTBEAT at 17:17:53
Got a SESSION_HEARTBEAT at 17:17:55
Got a SESSION_HEARTBEAT at 17:17:57
Got a SESSION_HEARTBEAT at 17:17:59
Got a SESSION_HEARTBEAT at 17:18:01
Got a SESSION_HEARTBEAT at 17:18:03
Got a SESSION_HEARTBEAT at 17:18:05
Got a SESSION_HEARTBEAT at 17:18:07
Got a SESSION_HEARTBEAT at 17:18:09
Got a SESSION_HEARTBEAT at 17:18:11
Got a SESSION_HEARTBEAT at 17:18:13
Got a SESSION_HEARTBEAT at 17:18:15
Got a SESSION_HEARTBEAT at 17:18:17


On Fri, Oct 9, 2009 at 9:27 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Thanks Anthony, this solved it. You rock :)
>
> My program now outputs:
>
> Got a SESSION_HEARTBEAT at 17:14:59
> Got a SESSION_HEARTBEAT at 17:15:00
> Got a SESSION_HEARTBEAT at 17:15:02
> Got a SESSION_HEARTBEAT at 17:15:03
> Got a SESSION_HEARTBEAT at 17:15:04
> Got a SESSION_HEARTBEAT at 17:15:05
> Got a SESSION_HEARTBEAT at 17:15:06
> Got a SESSION_HEARTBEAT at 17:15:07
> Got a SESSION_HEARTBEAT at 17:15:08
> Got a SESSION_HEARTBEAT at 17:15:09
> Got a SESSION_HEARTBEAT at 17:15:10
> Got a SESSION_HEARTBEAT at 17:15:11
> Got a SESSION_HEARTBEAT at 17:15:12
> Got a SESSION_HEARTBEAT at 17:15:13
> Got a SESSION_HEARTBEAT at 17:15:14
> Got a SESSION_HEARTBEAT at 17:15:15
> Got a SESSION_HEARTBEAT at 17:15:16
> Got a SESSION_HEARTBEAT at 17:15:17
> Got a SESSION_HEARTBEAT at 17:15:18
> Got a SESSION_HEARTBEAT at 17:15:19
> Got a SESSION_HEARTBEAT at 17:15:20
> Got a SESSION_HEARTBEAT at 17:15:21
> Got a SESSION_HEARTBEAT at 17:15:22
> Got a SESSION_HEARTBEAT at 17:15:23
> Got a SESSION_HEARTBEAT at 17:15:24
> Got a SESSION_HEARTBEAT at 17:15:25
> Got a SESSION_HEARTBEAT at 17:15:26
> Got a SESSION_HEARTBEAT at 17:15:27
> Got a SESSION_HEARTBEAT at 17:15:28
> Got a SESSION_HEARTBEAT at 17:15:29
> Got a SESSION_HEARTBEAT at 17:15:30
>
>
>
>
>
> On Fri, Oct 9, 2009 at 4:02 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Update to trunk and try it with fs_cli it for sure will let you do every 1
>> second
>>
>> in fs_cli type
>>
>> /events plain all
>>
>> if you make that call you will see one every 1 second
>>
>>
>>
>> On Fri, Oct 9, 2009 at 12:45 AM, Diego Viola <diego.viola at gmail.com>wrote:
>>
>>> Nope, I was just wondering why it didn't work at 1 second exactly...
>>>
>>>
>>> On Fri, Oct 9, 2009 at 3:36 AM, William Suffill <
>>> william.suffill at gmail.com> wrote:
>>>
>>>> Why do you need it every second? If you want real time channel counts
>>>> you would  be able to track each create/destroy even instead of
>>>> relying on the heartbeat summary.
>>>>
>>>> -- W
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From diego.viola at gmail.com  Fri Oct  9 15:01:46 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 9 Oct 2009 22:01:46 +0000
Subject: [Freeswitch-users] Heartbeat question
In-Reply-To: <86a32abc0910091430hc151ed8of085fcc3af2f1de4@mail.gmail.com>
References: <86a32abc0910081956k91b3600sb8c556e589f5b089@mail.gmail.com>
	<86a32abc0910082003v3f5d4e6dua8e7eae18fd2f939@mail.gmail.com>
	<6b65470d0910082036g204d5dcbw26e1e43ef80c7c84@mail.gmail.com>
	<86a32abc0910082245j36b8b3e9ncaf1cdd358fce319@mail.gmail.com>
	<191c3a030910090902t38282f72l6b6d1733a758836@mail.gmail.com>
	<86a32abc0910091427i2395fd80t3bc557186ece3972@mail.gmail.com>
	<86a32abc0910091430hc151ed8of085fcc3af2f1de4@mail.gmail.com>
Message-ID: <86a32abc0910091501y40dd34c4r1bca3eeb2b6fa693@mail.gmail.com>

Here's my heartbeat script now.

#!/usr/bin/env ruby

require 'rubygems'
require 'fsr'
require "fsr/listener/inbound"

def custom_channel_heartbeat_handler(event)
  puts "Got a SESSION_HEARTBEAT at #{Time.now.strftime('%H:%M:%S')}"
end
FSL::Inbound.add_event_hook(:SESSION_HEARTBEAT) {|event|
custom_channel_heartbeat_handler(event) }
FSR.start_ies!(FSL::Inbound, :host => "localhost", :port => 8021)

Thanks again.

Diego

On Fri, Oct 9, 2009 at 9:30 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Here is on two seconds ;)
>
> Got a SESSION_HEARTBEAT at 17:17:13
> Got a SESSION_HEARTBEAT at 17:17:15
> Got a SESSION_HEARTBEAT at 17:17:17
> Got a SESSION_HEARTBEAT at 17:17:19
> Got a SESSION_HEARTBEAT at 17:17:21
> Got a SESSION_HEARTBEAT at 17:17:23
> Got a SESSION_HEARTBEAT at 17:17:25
> Got a SESSION_HEARTBEAT at 17:17:27
> Got a SESSION_HEARTBEAT at 17:17:29
> Got a SESSION_HEARTBEAT at 17:17:31
> Got a SESSION_HEARTBEAT at 17:17:33
> Got a SESSION_HEARTBEAT at 17:17:35
> Got a SESSION_HEARTBEAT at 17:17:37
> Got a SESSION_HEARTBEAT at 17:17:39
> Got a SESSION_HEARTBEAT at 17:17:41
> Got a SESSION_HEARTBEAT at 17:17:43
> Got a SESSION_HEARTBEAT at 17:17:45
> Got a SESSION_HEARTBEAT at 17:17:47
> Got a SESSION_HEARTBEAT at 17:17:49
> Got a SESSION_HEARTBEAT at 17:17:51
> Got a SESSION_HEARTBEAT at 17:17:53
> Got a SESSION_HEARTBEAT at 17:17:55
> Got a SESSION_HEARTBEAT at 17:17:57
> Got a SESSION_HEARTBEAT at 17:17:59
> Got a SESSION_HEARTBEAT at 17:18:01
> Got a SESSION_HEARTBEAT at 17:18:03
> Got a SESSION_HEARTBEAT at 17:18:05
> Got a SESSION_HEARTBEAT at 17:18:07
> Got a SESSION_HEARTBEAT at 17:18:09
> Got a SESSION_HEARTBEAT at 17:18:11
> Got a SESSION_HEARTBEAT at 17:18:13
> Got a SESSION_HEARTBEAT at 17:18:15
> Got a SESSION_HEARTBEAT at 17:18:17
>
>
>
> On Fri, Oct 9, 2009 at 9:27 PM, Diego Viola <diego.viola at gmail.com> wrote:
>
>> Thanks Anthony, this solved it. You rock :)
>>
>> My program now outputs:
>>
>> Got a SESSION_HEARTBEAT at 17:14:59
>> Got a SESSION_HEARTBEAT at 17:15:00
>> Got a SESSION_HEARTBEAT at 17:15:02
>> Got a SESSION_HEARTBEAT at 17:15:03
>> Got a SESSION_HEARTBEAT at 17:15:04
>> Got a SESSION_HEARTBEAT at 17:15:05
>> Got a SESSION_HEARTBEAT at 17:15:06
>> Got a SESSION_HEARTBEAT at 17:15:07
>> Got a SESSION_HEARTBEAT at 17:15:08
>> Got a SESSION_HEARTBEAT at 17:15:09
>> Got a SESSION_HEARTBEAT at 17:15:10
>> Got a SESSION_HEARTBEAT at 17:15:11
>> Got a SESSION_HEARTBEAT at 17:15:12
>> Got a SESSION_HEARTBEAT at 17:15:13
>> Got a SESSION_HEARTBEAT at 17:15:14
>> Got a SESSION_HEARTBEAT at 17:15:15
>> Got a SESSION_HEARTBEAT at 17:15:16
>> Got a SESSION_HEARTBEAT at 17:15:17
>> Got a SESSION_HEARTBEAT at 17:15:18
>> Got a SESSION_HEARTBEAT at 17:15:19
>> Got a SESSION_HEARTBEAT at 17:15:20
>> Got a SESSION_HEARTBEAT at 17:15:21
>> Got a SESSION_HEARTBEAT at 17:15:22
>> Got a SESSION_HEARTBEAT at 17:15:23
>> Got a SESSION_HEARTBEAT at 17:15:24
>> Got a SESSION_HEARTBEAT at 17:15:25
>> Got a SESSION_HEARTBEAT at 17:15:26
>> Got a SESSION_HEARTBEAT at 17:15:27
>> Got a SESSION_HEARTBEAT at 17:15:28
>> Got a SESSION_HEARTBEAT at 17:15:29
>> Got a SESSION_HEARTBEAT at 17:15:30
>>
>>
>>
>>
>>
>> On Fri, Oct 9, 2009 at 4:02 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> Update to trunk and try it with fs_cli it for sure will let you do every
>>> 1 second
>>>
>>> in fs_cli type
>>>
>>> /events plain all
>>>
>>> if you make that call you will see one every 1 second
>>>
>>>
>>>
>>> On Fri, Oct 9, 2009 at 12:45 AM, Diego Viola <diego.viola at gmail.com>wrote:
>>>
>>>> Nope, I was just wondering why it didn't work at 1 second exactly...
>>>>
>>>>
>>>> On Fri, Oct 9, 2009 at 3:36 AM, William Suffill <
>>>> william.suffill at gmail.com> wrote:
>>>>
>>>>> Why do you need it every second? If you want real time channel counts
>>>>> you would  be able to track each create/destroy even instead of
>>>>> relying on the heartbeat summary.
>>>>>
>>>>> -- W
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:213-799-1400
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
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From anthony.minessale at gmail.com  Fri Oct  9 15:53:13 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 9 Oct 2009 17:53:13 -0500
Subject: [Freeswitch-users] FW: FS Does Not Relay
	PresencePUBLISHToSubscribing Phones
In-Reply-To: <6E290A3B014040CFA97E8BE97BC36BAC@greyhawk.tonecommander.com>
References: <6E290A3B014040CFA97E8BE97BC36BAC@greyhawk.tonecommander.com>
Message-ID: <191c3a030910091553s3d16a992x2b0116abe686408a@mail.gmail.com>

yah what i was getting at was between the sql statement and what was
actually in the table you should be able to tell what's wrong.


On Fri, Oct 9, 2009 at 3:06 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>  I put the sqlite3 select query in the paste bin again, and prior to that,
> I entered the .dump command.  The select command came back with the
> sqlite3 prompt, which I guess means it didn't find an entry.  How do I go
> about isolating this problem?  I'm using CounterPath's Bria Professional
> softphone.  They are the same company that make the Eyebeam.
>
> Any ideas?
>
> Best Regards,
> Jerry
>
>
>  ------------------------------
> *From:* Jerry Richards [mailto:jerry.richards at teotech.com]
> *Sent:* Friday, October 02, 2009 11:28 AM
> *To:* 'freeswitch-users at lists.freeswitch.org'
> *Subject:* RE: [Freeswitch-users] FS Does Not Relay
> PresencePUBLISHToSubscribing Phones
>
>  I put the sqlite3 select query in the paste bin, and prior to that, I
> entered the .dump command.  The select command came back with a "...>"
> prompt which I don't understand.  I don't know enough about sqlite3 to know
> what that means?
>
> Best Regards,
> Jerry
>
>  ------------------------------
> *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com]
> *Sent:* Friday, October 02, 2009 10:52 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] FS Does Not Relay
> PresencePUBLISHToSubscribing Phones
>
> connect to sqlite directly with sqlite3 app and try that sql stmt and see
> why it doesn't match anything.
>
> sqlite3 /usr/local/freeswitch/db/sofia_reg_internal.db
>
> select
> sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Away','away','
> 192.168.72.38',sip_presence.status,sip_presence.rpid from
> sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user
> and sip_subscriptions.sub_to_host=sip_presence.sip_host and
> sip_subscriptions.profile_name=sip_presence.profile_name) where (event='presence'
> or event='presence') and sub_to_user='1001' and (sub_to_host='192.168.72
> .38' or presence_hosts like '%192.168.72.38%') and (sip_subscriptions.profile_name
> = 'external' or sip_subscriptions.presence_hosts !=
> sip_subscriptions.sub_to_host)
>
>
> On Fri, Oct 2, 2009 at 12:12 PM, Jerry Richards <
> jerry.richards at teotech.com> wrote:
>
>>  Okay, I put a log up on the pastebin that shows the PUBLISH event coming
>> from a CounterPath Bria Professional phone.  For some reason, FS is getting
>> an error and not relaying the presence status to the subscriber.
>>
>> Best Regards,
>> Jerry
>>
>>  ------------------------------
>> *From:* Jo?o Mesquita [mailto:jmesquita at freeswitch.org]
>> *Sent:* Thursday, October 01, 2009 8:14 PM
>>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* Re: [Freeswitch-users] FS Does Not Relay Presence
>> PUBLISHToSubscribing Phones
>>
>>   Piece of advice, don't ask, just do it. ;)
>>
>> jmesquita
>>
>> On Thu, Oct 1, 2009 at 3:29 PM, Jerry Richards <
>> jerry.richards at teotech.com> wrote:
>>
>>>  If you have time to take a look, I could put a trace in the pastebin?
>>>
>>> Jerry
>>>
>>>  ------------------------------
>>>  *From:* Jerry Richards [mailto:jerry.richards at teotech.com]
>>> *Sent:* Thursday, October 01, 2009 10:29 AM
>>> *To:* 'freeswitch-users at lists.freeswitch.org'
>>> *Subject:* RE: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
>>> ToSubscribing Phones
>>>
>>>  I am using two Bria Professional Version 2.5.4 Build 54835 softphones.
>>>
>>> Thanks,
>>> Jerry
>>>
>>>  ------------------------------
>>>  *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com]
>>> *Sent:* Thursday, October 01, 2009 9:36 AM
>>> *To:* freeswitch-users at lists.freeswitch.org
>>> *Subject:* Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
>>> ToSubscribing Phones
>>>
>>>   which phone is it,
>>> we tested it with eyebeam and it appears to work for us.
>>>
>>>
>>> On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards <
>>> jerry.richards at teotech.com> wrote:
>>>
>>>>
>>>> By the way, I see the following lines at the FS console, which might be
>>>> a
>>>> clue as to why this is happening.  Could someone point me toward what
>>>> might
>>>> cause this?  I set the "manage-presence" parameter to "true" in each XML
>>>> file where I saw it defined.
>>>>
>>>> [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal)
>>>> [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
>>>>   ...
>>>> [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6)
>>>> [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
>>>>   ...
>>>> [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external)
>>>> [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
>>>>   ...
>>>> [WARNING] sofia_presence.c:565  192.168.72.38 is an alias, skipping
>>>>
>>>>
>>>> Best Regards,
>>>> Jerry
>>>>
>>>>
>>>> -----Original Message-----
>>>> From: Jerry Richards [mailto:jerry.richards at teotech.com]
>>>> Sent: Wednesday, September 30, 2009 9:12 AM
>>>> To: 'freeswitch-users at lists.freeswitch.org'
>>>> Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones
>>>>
>>>> I have two phones configured to subscribe to each other's presence
>>>> status.
>>>> When I change the presence status in one phone, I see the SIP PUBLISH
>>>> message going to FS, but I don't see FS relaying that presence status to
>>>> the
>>>> subscribing phone.  Does anyone know why?
>>>>
>>>> Best Regards,
>>>> Jerry
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:213-799-1400
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From nagalenoj at gmail.com  Fri Oct  9 20:50:28 2009
From: nagalenoj at gmail.com (Nagalenoj)
Date: Fri, 9 Oct 2009 20:50:28 -0700 (PDT)
Subject: [Freeswitch-users] Re  corded file as voicemail.
In-Reply-To: <BC536E8C-6813-4A69-9B80-EA11E41A79FC@jerris.com>
References: <25751158.post@talk.nabble.com>
	<BC536E8C-6813-4A69-9B80-EA11E41A79FC@jerris.com>
Message-ID: <25830785.post@talk.nabble.com>


No, When I do voicemail_inject it is reporting the following error.

'Stereo is currently not supported, please downsample to mono.'

Any other way to do this.??


Michael Jerris wrote:
> 
> http://wiki.freeswitch.org/wiki/Mod_voicemail#voicemail_inject
> 
> On Oct 5, 2009, at 9:46 AM, Nagalenoj wrote:
> 
>>
>> Is it possible to treat a recorded voice as voice mail?
>>
>> Assume that, I've recorded a conversation and I want this recorded  
>> file to
>> be treated like voicemail. So, I could check it like voicemail!!
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/Recorded-file-as-voicemail.-tp25751158p25830785.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From nagalenoj at gmail.com  Fri Oct  9 20:51:30 2009
From: nagalenoj at gmail.com (Nagalenoj)
Date: Fri, 9 Oct 2009 20:51:30 -0700 (PDT)
Subject: [Freeswitch-users] Re  corded file as voicemail.
Message-ID: <25830785.post@talk.nabble.com>


No, When I do voicemail_inject and check through voicemail, it is not playing
the file instead reporting the following error.

'Stereo is currently not supported, please downsample to mono.'

Any other way to do this.??


Michael Jerris wrote:
> 
> http://wiki.freeswitch.org/wiki/Mod_voicemail#voicemail_inject
> 
> On Oct 5, 2009, at 9:46 AM, Nagalenoj wrote:
> 
>>
>> Is it possible to treat a recorded voice as voice mail?
>>
>> Assume that, I've recorded a conversation and I want this recorded  
>> file to
>> be treated like voicemail. So, I could check it like voicemail!!
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/Recorded-file-as-voicemail.-tp25751158p25830785.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From jason at jasonjgw.net  Fri Oct  9 21:05:39 2009
From: jason at jasonjgw.net (Jason White)
Date: Sat, 10 Oct 2009 15:05:39 +1100
Subject: [Freeswitch-users] Re  corded file as voicemail.
In-Reply-To: <25830785.post@talk.nabble.com>
References: <25830785.post@talk.nabble.com>
Message-ID: <20091010040539.GB23188@jdc.jasonjgw.net>

Nagalenoj <nagalenoj at gmail.com> wrote:
> 
> No, When I do voicemail_inject and check through voicemail, it is not playing
> the file instead reporting the following error.
> 
> 'Stereo is currently not supported, please downsample to mono.'

This has been discussed on the list before. I think the solution was to use
sox to convert the files to mono. You might need a script to do this before
injecting them into the voicemail.




From lakindia89 at gmail.com  Fri Oct  9 21:39:07 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Sat, 10 Oct 2009 10:09:07 +0530
Subject: [Freeswitch-users] libpri_span vs prispans
In-Reply-To: <87f2f3b90910091057q3a033548j462446cf424ad257@mail.gmail.com>
References: <7d79b3930910090459v2b07e3aapd715c7618f11ada0@mail.gmail.com>
	<20091009125714.E0FF73043D7@mail.cune.org>
	<87f2f3b90910091057q3a033548j462446cf424ad257@mail.gmail.com>
Message-ID: <7d79b3930910092139v3589b761oab0bae46598118f@mail.gmail.com>

Where can I found entire configurations for a libpri spans. Right now I've
only 4 things in my libpri configuration. They are Switchtype, Node,
Contect, Dialplan, which is given as example in wiki.

But I think libprispan has more configuration. Where can I find those
things?



On Fri, Oct 9, 2009 at 11:27 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Oct 9, 2009 at 5:57 AM, <Russell.Mosemann at cune.org> wrote:
>
>> lakshmanan ganapathy <lakindia89 at gmail.com> said:
>> > What difference it will make, if I use pri_span configuration and
>> lib_pri
>> > span configuration in openzap.conf.xml.
>> > I'm preety much confused on the difference between this.
>>
>> Openzap can handle a T1/E1 itself, or openzap can use libpri to do that.
>> libpri is more mature than openzap. In my case openzap was not able to
>> handle everything invovling a T1 here, and using libpri fixed it.
>>
>> A libpri_span configuration is handled by libpri. A pri_span
>> configuration is handled natively by openzap. Use the one that works for
>> you.
>>
>
> Right now the libpri stack is more mature than the ozmod_isdn stuff. See
> the OpenZAP page on the wiki for instructions on how to set up ozmod_libpri.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Fri Oct  9 22:40:18 2009
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 10 Oct 2009 13:40:18 +0800
Subject: [Freeswitch-users] Re corded file as voicemail.
In-Reply-To: <20091010040539.GB23188@jdc.jasonjgw.net>
References: <25830785.post@talk.nabble.com>
	<20091010040539.GB23188@jdc.jasonjgw.net>
Message-ID: <23f91030910092240u53b0cc51jd5b1670901f20e21@mail.gmail.com>

Yes, it's discussed before.

http://wiki.freeswitch.org/wiki/Channel_Variables#RECORD_STEREO

set that var to false before you record.


2009/10/10 Jason White <jason at jasonjgw.net>

> Nagalenoj <nagalenoj at gmail.com> wrote:
> >
> > No, When I do voicemail_inject and check through voicemail, it is not
> playing
> > the file instead reporting the following error.
> >
> > 'Stereo is currently not supported, please downsample to mono.'
>
> This has been discussed on the list before. I think the solution was to use
> sox to convert the files to mono. You might need a script to do this before
> injecting them into the voicemail.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From velu.technical at gmail.com  Fri Oct  9 23:51:43 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Sat, 10 Oct 2009 12:21:43 +0530
Subject: [Freeswitch-users] Difference between park and valet_park
Message-ID: <1452e2980910092351v766cad5bpcb55b59ad5fee6d0@mail.gmail.com>

Dear All,
     Could you please any one explain the difference between parking and
valet parking?
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From jason at jasonjgw.net  Sat Oct 10 00:00:03 2009
From: jason at jasonjgw.net (Jason White)
Date: Sat, 10 Oct 2009 18:00:03 +1100
Subject: [Freeswitch-users] Status of ubuntu/debian packages.
In-Reply-To: <4ABA47FB.2050100@gmail.com>
References: <4ABA47FB.2050100@gmail.com>
Message-ID: <20091010070003.GA16076@jdc.jasonjgw.net>

for those who are working on the Debian packages, the following ITP bug may be
of interest.
http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=513606
which also refers to an older entry.

Thanks are due to all who are contributing to the packaging effort.




From dujinfang at gmail.com  Sat Oct 10 00:32:37 2009
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 10 Oct 2009 15:32:37 +0800
Subject: [Freeswitch-users] Difference between park and valet_park
In-Reply-To: <1452e2980910092351v766cad5bpcb55b59ad5fee6d0@mail.gmail.com>
References: <1452e2980910092351v766cad5bpcb55b59ad5fee6d0@mail.gmail.com>
Message-ID: <23f91030910100032k41513eecp2db3d32a79dfe724@mail.gmail.com>

search this list, just has been discussed.

2009/10/10 velusamy velu <velu.technical at gmail.com>

> Dear All,
>      Could you please any one explain the difference between parking and
> valet parking?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From velu.technical at gmail.com  Sat Oct 10 02:37:20 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Sat, 10 Oct 2009 15:07:20 +0530
Subject: [Freeswitch-users] Play music on hold after parking the call
Message-ID: <1452e2980910100237u317d6138w1d13bd3cfb250353@mail.gmail.com>

Dear All,
      I am using ESL.pm module to control the dial plan application. I want
to play some music while executing the some external scripts. I executed
park after then I executed the playback the music didn't play.

    Could any one please explain how can I solve this problem without using
async mode in socket application?

Thanks,
Velusamy.
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From wangdq.no1 at gmail.com  Sat Oct 10 02:53:19 2009
From: wangdq.no1 at gmail.com (daqiang wang)
Date: Sat, 10 Oct 2009 17:53:19 +0800
Subject: [Freeswitch-users] how to match '#' in XML dialplan ?
Message-ID: <ecd05ce30910100253q1424cd4an9be93ab434912b3f@mail.gmail.com>

hello every one :
   I want to match the # in XML dialplan , how to do  ?
   example :
      1#5555#6666 . how to do ?
     I do this :
     <condition field="destination_number" expression="^1#(d+)#(d+)$"/>
    but it's not work
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From maciej.aniserowicz at gmail.com  Sat Oct 10 03:04:04 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Sat, 10 Oct 2009 03:04:04 -0700 (PDT)
Subject: [Freeswitch-users] Bad sound quality while eavesdropping
In-Reply-To: <191c3a030910080823g79c7c596x1cd887e1538ce2e1@mail.gmail.com>
References: <41A44DD027064988A914974405788C2E@procent>
	<191c3a030910050731m2d74979ep4598e5a1945d58ae@mail.gmail.com>
	<1254901192035-3780245.post@n2.nabble.com>
	<8437F5BC-7AFF-4A74-B8CD-C5B8219021F6@jerris.com>
	<1255008427639-3788019.post@n2.nabble.com>
	<191c3a030910080823g79c7c596x1cd887e1538ce2e1@mail.gmail.com>
Message-ID: <1255169044209-3799274.post@n2.nabble.com>


Hi,
Here are the messages with a:ptime parameter. All the calls are started by
commands sent through socket.
I'm not sure if this is all information you need, please let me know if
something is missing here and I'll post that.

1) starting connection with x-lite (number 2003, the eavesdropper):

   INVITE sip:2003 at 192.168.3.100:60188;rinstance=80b8f8d92af87cd2 SIP/2.0
   Via: SIP/2.0/UDP 192.168.3.159:15060;rport;branch=z9hG4bKD9meHQN5XQ88K
   Max-Forwards: 69
   From: "MyApp" <sip:0000000000 at 192.168.3.159>;tag=jpQ6D7D2jUXvF
   To: <sip:2003 at 192.168.3.100:60188;rinstance=80b8f8d92af87cd2>
   Call-ID: 663c4500-3024-122d-57a2-9be8d85a4cff
   CSeq: 121465610 INVITE
   Contact: <sip:mod_sofia at 192.168.3.159:15060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 447
   Remote-Party-ID: "MyApp"
<sip:0000000000 at 192.168.3.159>;party=calling;screen=yes;privacy=off
   
   v=0
   o=FreeSWITCH 2223565947735016740 3096553520713245589 IN IP4 192.168.3.159
   s=FreeSWITCH
   c=IN IP4 192.168.3.159
   t=0 0
   m=audio 29966 RTP/AVP 0 115 107 9 8 3 101 13
   a=rtpmap:0 PCMU/8000
   a=rtpmap:115 G7221/32000
   a=fmtp:115 bitrate=48000
   a=rtpmap:107 G7221/16000
   a=fmtp:107 bitrate=32000
   a=rtpmap:9 G722/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:3 GSM/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:13 CN/8000
   a=ptime:20


2) starting connection with cisco ip phone (number 2006, first leg of
eavesdropped session):

   INVITE sip:2006 at 192.168.2.106:5060;user=phone SIP/2.0
   Via: SIP/2.0/UDP 192.168.3.159:15060;rport;branch=z9hG4bKg40rp87FNjB1p
   Max-Forwards: 69
   From: "MyApp" <sip:0000000000 at 192.168.3.159>;tag=Q3N2pe2K47ctS
   To: <sip:2006 at 192.168.2.106:5060;user=phone>
   Call-ID: 6e6fd8a0-3024-122d-57a2-9be8d85a4cff
   CSeq: 121465616 INVITE
   Contact: <sip:mod_sofia at 192.168.3.159:15060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 447
   Remote-Party-ID: "MyApp"
<sip:0000000000 at 192.168.3.159>;party=calling;screen=yes;privacy=off
   
   v=0
   o=FreeSWITCH 8453497903781974949 2294419114567885490 IN IP4 192.168.3.159
   s=FreeSWITCH
   c=IN IP4 192.168.3.159
   t=0 0
   m=audio 17670 RTP/AVP 0 115 107 9 8 3 101 13
   a=rtpmap:0 PCMU/8000
   a=rtpmap:115 G7221/32000
   a=fmtp:115 bitrate=48000
   a=rtpmap:107 G7221/16000
   a=fmtp:107 bitrate=32000
   a=rtpmap:9 G722/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:3 GSM/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:13 CN/8000
   a=ptime:20


3) starting connection with extension playing a file (number 9999, second
leg of eavesdropped session):

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP
192.168.3.159:5080;rport=5080;branch=z9hG4bKv6Dg7myj4tvBS
   From: "FreeSWITCH" <sip:myuser at mydomain;transport=udp>;tag=091j2Q0Fre8vp
   To: <sip:9999 at 192.168.3.159:15060>;tag=U7t5Xt51rB64Q
   Call-ID: 7551b7d0-3024-122d-b3a6-156c4939c4f8
   CSeq: 121465623 INVITE
   Contact: <sip:mod_sofia at 192.168.3.159:15060;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 263
   
   v=0
   o=FreeSWITCH 5614997529598779838 636912243381649698 IN IP4 192.168.3.159
   s=FreeSWITCH
   c=IN IP4 192.168.3.159
   t=0 0
   m=audio 30086 RTP/AVP 0 101 13
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:13 CN/8000
   a=ptime:20




Anthony Minessale wrote:
> 
> you probably have some device lying about ptime everywhere
> look at a sip trace an pay especially close attention to ptime:x param in
> sdp
> if you don't understand this just attach it here
> 
> execute the following at the cli
> sofia profile internal siptrace on
> sofila loglevel debug
> 
> 
> 
> On Thu, Oct 8, 2009 at 8:27 AM, Maciej Aniserowicz <
> maciej.aniserowicz at gmail.com> wrote:
> 
>>
>> It's the same on the trunk (the last rev I used was not so old anyway).
>>
>> Codecs are the same on both legs:
>> read codec/read rate: PCMU      8000
>> write codec/write rate: PCMU    8000
>>
>> MA
>>
>>
>>
>>
>> Michael Jerris wrote:
>> >
>> > What codecs are all the call legs using, also, please try current svn
>> > trunk.
>> >
>> > Mike
>> >
>> > On Oct 7, 2009, at 3:39 AM, Maciej Aniserowicz wrote:
>> >
>> >>
>> >> Sorry about posting several questions at once, I wasn't aware it's
>> >> "rude".
>> >> Let's concentrate on this issue then.
>> >>
>> >> I use FS rev 14994. Phones on extensions:
>> >> 1) x-lite
>> >> 2) cisco sip phone
>> >> 3) audio played by fs to the extension being eavesdropped
>> >>
>> >> I did not change any codec configuration, I just use the standard
>> >> one that
>> >> comes with both FS and the phones.
>> >> Some time ago someone on FS irc channel told me that this is just
>> >> how FS
>> >> eavesdropping works... from your response I understand that this is
>> >> not
>> >> entirely true?
>> >>
>> >> Maciej Aniserowicz
>> >>
>> >>
>> >>
>> >> Anthony Minessale wrote:
>> >>>
>> >>> That's is a somewhat vague position.
>> >>>
>> >>> You did not mention which version of FreeSWITCH you are running, the
>> >>> phones
>> >>> being used in your example, your configuration, the codecs in use
>> >>> etc.
>> >>>
>> >>> BTW,
>> >>> I think you should only ask one question at a time on this list.
>> >>> The list
>> >>> is run by volunteers and it's sort of rude to expect 3 or 4 threads
>> >>> to be
>> >>> tended to concerning the same one individual.
>> >>>
>> >>>
>> >>> 2009/10/5 Maciej Aniserowicz <maciej.aniserowicz at gmail.com>
>> >>>
>> >>>> Hello,
>> >>>> When I use eavesdropping in FreeSWITCH, the sound quality is
>> >>>> really bad.
>> >>>> Is
>> >>>> there any way to improve it? Is this a known problem?
>> >>>> Br/
>> >>>> Maciej Aniserowicz
>> >>>>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> --
>> View this message in context:
>> http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3788019.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3799274.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From juanbackson at gmail.com  Sat Oct 10 03:05:21 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Sat, 10 Oct 2009 18:05:21 +0800
Subject: [Freeswitch-users] directory variables not set correct with the
	latest trunk
In-Reply-To: <191c3a030910080821w691aa78aw272d71dee7675ff1@mail.gmail.com>
References: <27c25bc40910080320x7b3b380eu933797197e334cb8@mail.gmail.com>
	<191c3a030910080821w691aa78aw272d71dee7675ff1@mail.gmail.com>
Message-ID: <27c25bc40910100305q5ae87ddbjc9999852724470d8@mail.gmail.com>

Hi,

My sip phone did authentication ( sending in SIP REGISTER etc ), but it is
still not showing up.  How come?

It was working before.

thanks,
juan

On Thu, Oct 8, 2009 at 11:21 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> they only set if you authenticate on sip or run the set_user app
>
>
> On Thu, Oct 8, 2009 at 5:20 AM, Juan Backson <juanbackson at gmail.com>wrote:
>
>> Hi,
>>
>> My application fails to set the appropriate variables using directory xml
>> after using the latest trunk as of yesterday.
>>
>> My curl looks like:
>>
>>
>>
>> <document type="freeswitch/xml">
>>   <section name="directory">
>>     <domain name="192.168.1.102">
>>       <user id="200002">
>>         <params>
>>           <param name="password" value="123456"></param>
>>           <param name="dial-string"
>> value="sofia/internal/sip:200002 at 192.168.1.11:29440
>> ;rinstance=0245b2a59ddff837"></param>
>>         </params>
>>         <variables>
>>           <variable name="account_id" value="20"></variable>
>>           <variable name="vm-code" value="1000"></variable>
>>         </variables>
>>       </user>
>>     </domain>
>>   </section>
>> </document>
>>
>>
>>
>> In the info app, I am not seeing account-id and vm-code anymore.
>>
>> How to fix that?
>>
>> juan
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tayeb.meftah at gmail.com  Sat Oct 10 05:32:11 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sat, 10 Oct 2009 12:32:11 +0000
Subject: [Freeswitch-users] Anyone intairaisted?
Message-ID: <4AD07ECB.5070602@gmail.com>

dear all
anyone intairaisted in porting/integrating some billing engine/software 
to freeswitch?
we are building a new Community for that
no for billing only but billing is the primary focus
if anyone intairaisted please mail me to:
tayeb.meftah at gmail.com
i'm waiting for you to join!
the web site is not yet ready, is:
http://www.awesomevoip.org
we have also a hosted PBX to do conferencing... but we need a conference 
bridge
thanks!


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4494 (20091009) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com





From testeador01 at gmail.com  Sat Oct 10 06:24:20 2009
From: testeador01 at gmail.com (Milena)
Date: Sat, 10 Oct 2009 08:24:20 -0500
Subject: [Freeswitch-users] how to match '#' in XML dialplan ?
In-Reply-To: <ecd05ce30910100253q1424cd4an9be93ab434912b3f@mail.gmail.com>
References: <ecd05ce30910100253q1424cd4an9be93ab434912b3f@mail.gmail.com>
Message-ID: <c29b930b0910100624n7d5dd257h526d2815705703d3@mail.gmail.com>

escape character is '\'try
  <condition field="destination_number" expression="^1\#(d+)\#(d+)$"/>
2009/10/10 daqiang wang <wangdq.no1 at gmail.com>

> hello every one :
>    I want to match the # in XML dialplan , how to do  ?
>    example :
>       1#5555#6666 . how to do ?
>      I do this :
>      <condition field="destination_number" expression="^1#(d+)#(d+)$"/>
>     but it's not work
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lakindia89 at gmail.com  Sat Oct 10 06:47:00 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Sat, 10 Oct 2009 19:17:00 +0530
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <20091009123615.BC50A3FA6E9@mail.cune.org>
References: <7d79b3930910090407m5dc81d12pf2b0a47f6017804@mail.gmail.com>
	<20091009123615.BC50A3FA6E9@mail.cune.org>
Message-ID: <7d79b3930910100647u7fd11413ucec0a535d68b3e3a@mail.gmail.com>

Hi,
I've tried with the following dialplan(After making the changes that are
recommended). But still I got INVALID_IE_CONTENTS error.

Another thing is, if I use prispan configuration, I'm able to make outgoing
calls.
But if I use the libpri span configuration I'm not able to make outgoing
calls, which says INVALID_IE_CONTENTS.

I've been struck with this problem for the past 1 week.
Any solution to this???


On Fri, Oct 9, 2009 at 6:06 PM, <Russell.Mosemann at cune.org> wrote:

> lakshmanan ganapathy <lakindia89 at gmail.com> said:
>
> > But still I'm facing problem with the outgoing call. It says
> > INVALID_IE_CONTENTS.
> > What might be the issue? Even I tried the following dialplan to call by
> > using bridge.
> > <extension name="outgoing-fxo">
> >                   <condition field="destination_number"
> > expression="^(\d{10})$">
> >                           <action application="answer"/>
> >                           <action application="bridge"
> > data="openzap/1/1/${dialed_ext}"/>
>
> Does "answer" need to be called here? I haven't used an fxo. So, I don't
> know. What value does $dialed_ext have? If you want to use the number
> matched in the condition, then it should be
>
> openzap/1/1/$1
>
> --
> Russell Mosemann
>
>
>
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jmesquita at freeswitch.org  Sat Oct 10 07:30:05 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 10 Oct 2009 11:30:05 -0300
Subject: [Freeswitch-users] Questions regarding to mod_nibble
In-Reply-To: <a60fed5e0910082039yc8b7ef8x24260041a7c11d31@mail.gmail.com>
References: <a60fed5e0910082039yc8b7ef8x24260041a7c11d31@mail.gmail.com>
Message-ID: <ebbc2d550910100730o6abae5b5p254b72cf1c3869c4@mail.gmail.com>

I am testing the latest version of nibblebill so let me see if I can help
you with your questions.

jm

On Fri, Oct 9, 2009 at 12:39 AM, Ahmed Munir <ahmedmunir007 at gmail.com>wrote:

> I want to ask three questions related to mod_nibble bills, as I'm listing
> down below;
>
> 1- Can we select/use dynamic tables for billing using nibble bill?
>
What do you mean for dynamic tables? Like LCR does that you can specify your
own SQL statement to be executed? If that's what you are asking, no, but it
would be a nice todo.


> 2- Can we define more than two tables and attributes in
> nibblebill.conf.xml?
>
What else do you want to define and how do you imagine it to behave?


> 3- As Nibble bill is use to deduct amount of user account, Can we deduct
> minutes instead of cash? Because my case is, if a user buy a package and I
> only want to deducts his/her minutes. How we can resolve it by nibble bill?
> / What other way we can resolve it?
>
When we say cash on the column, we are really saying just a number that is
being deducted, that's it. If you deduct 1 every 60 seconds, you will have
your "cash" converted to minutes, won't you?


>
> Kindly advise soon.
>
> --
> Regards,
>
> Ahmed Munir
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dome at tel.co.th  Sat Oct 10 08:02:42 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Sat, 10 Oct 2009 22:02:42 +0700
Subject: [Freeswitch-users] Questions regarding to mod_nibble
In-Reply-To: <a60fed5e0910082039yc8b7ef8x24260041a7c11d31@mail.gmail.com>
References: <a60fed5e0910082039yc8b7ef8x24260041a7c11d31@mail.gmail.com>
Message-ID: <8ccbff060910100802t71a606aeo42c75008c3054794@mail.gmail.com>

2009/10/9 Ahmed Munir <ahmedmunir007 at gmail.com>:
> I want to ask three questions related to mod_nibble bills, as I'm listing
> down below;
>
> 1- Can we select/use dynamic tables for billing using nibble bill?
> 2- Can we define more than two tables and attributes in nibblebill.conf.xml?
> 3- As Nibble bill is use to deduct amount of user account, Can we deduct
> minutes instead of cash? Because my case is, if a user buy a package and I
> only want to deducts his/her minutes. How we can resolve it by nibble bill?
> / What other way we can resolve it?
My solution (Postgresql 8.4 with postgresql-prefix).
1. modify mod_nibblebill.c  change sqlupdate command to update
duration in my cdr table (insert blank data when start billing)
2. create trigger to deduct balance.


Dome C.
>
> Kindly advise soon.
>
> --
> Regards,
>
> Ahmed Munir
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From msc at freeswitch.org  Sat Oct 10 10:09:04 2009
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 10 Oct 2009 10:09:04 -0700
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <7d79b3930910100647u7fd11413ucec0a535d68b3e3a@mail.gmail.com>
References: <7d79b3930910090407m5dc81d12pf2b0a47f6017804@mail.gmail.com>
	<20091009123615.BC50A3FA6E9@mail.cune.org>
	<7d79b3930910100647u7fd11413ucec0a535d68b3e3a@mail.gmail.com>
Message-ID: <87f2f3b90910101009j1295d812jd98cae91e30b1eff@mail.gmail.com>

Okay, please go to pastebin.freeswitch.org and paste your openzap.conf.xml
file. Also, paste the ENTIRE debug log from a call from start to finish.
Telling us that you see INVALID_IE_CONTENTS doesn't help if we don't know
what the information element contains. Finally, turn on PRI debugging and
make another test call and pastebin that debug log as well. The debug will
show details about the communications between your machine and the carrier.
Instructions for turning on debugging for libpri are found in the OpenZAP
wiki page in the same place where the libpri instructions are located.

Put the pastebin number in this email thread and then we'll go have a look.

Thanks,
MC

On Sat, Oct 10, 2009 at 6:47 AM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Hi,
> I've tried with the following dialplan(After making the changes that are
> recommended). But still I got INVALID_IE_CONTENTS error.
>
> Another thing is, if I use prispan configuration, I'm able to make outgoing
> calls.
> But if I use the libpri span configuration I'm not able to make outgoing
> calls, which says INVALID_IE_CONTENTS.
>
> I've been struck with this problem for the past 1 week.
> Any solution to this???
>
>
>
> On Fri, Oct 9, 2009 at 6:06 PM, <Russell.Mosemann at cune.org> wrote:
>
>> lakshmanan ganapathy <lakindia89 at gmail.com> said:
>>
>> > But still I'm facing problem with the outgoing call. It says
>> > INVALID_IE_CONTENTS.
>> > What might be the issue? Even I tried the following dialplan to call by
>> > using bridge.
>> > <extension name="outgoing-fxo">
>> >                   <condition field="destination_number"
>> > expression="^(\d{10})$">
>> >                           <action application="answer"/>
>> >                           <action application="bridge"
>> > data="openzap/1/1/${dialed_ext}"/>
>>
>> Does "answer" need to be called here? I haven't used an fxo. So, I don't
>> know. What value does $dialed_ext have? If you want to use the number
>> matched in the condition, then it should be
>>
>> openzap/1/1/$1
>>
>> --
>> Russell Mosemann
>>
>>
>>
>> ________________________________________________________
>> Concordia University, Nebraska
>> See http://www.cune.edu/ for the latest news and events!
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Sat Oct 10 10:22:08 2009
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 10 Oct 2009 10:22:08 -0700
Subject: [Freeswitch-users] Difference between park and valet_park
In-Reply-To: <1452e2980910092351v766cad5bpcb55b59ad5fee6d0@mail.gmail.com>
References: <1452e2980910092351v766cad5bpcb55b59ad5fee6d0@mail.gmail.com>
Message-ID: <87f2f3b90910101022v52153d0dhd1ed774af84ecd22@mail.gmail.com>

On Fri, Oct 9, 2009 at 11:51 PM, velusamy velu <velu.technical at gmail.com>wrote:

> Dear All,
>      Could you please any one explain the difference between parking and
> valet parking?
>

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_valet_park

Like the wiki says, valet_park puts calls into numbered parking stalls. A
call that is parked in a numbered stall is retrieved by calling (or
transferring) a second call into that numbered stall. That makes it easy to
pass calls around. The regular park app puts calls in a special park state
that is like being on hold, but the only way to get them out of park is to
uuid_bridge a second leg to the uuid of the parked call leg.

Read the wiki entry on valet_park and then read the one on park and you'll
see that they are similar apps but the valet_park app is a bit easier for
phone users to use for moving calls in and out of the park state.

-MC


> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Sat Oct 10 10:30:50 2009
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 10 Oct 2009 10:30:50 -0700
Subject: [Freeswitch-users] how to match '#' in XML dialplan ?
In-Reply-To: <c29b930b0910100624n7d5dd257h526d2815705703d3@mail.gmail.com>
References: <ecd05ce30910100253q1424cd4an9be93ab434912b3f@mail.gmail.com>
	<c29b930b0910100624n7d5dd257h526d2815705703d3@mail.gmail.com>
Message-ID: <87f2f3b90910101030y7d1142b5kbdac5ffdbc12334b@mail.gmail.com>

Some characters need a backslash to match in a regular expression. However,
# is not one of them. I think your regex is wrong:
<condition field="destination_number" expression="^1#(d+)#(d+)$"/>

It should probably be:
<condition field="destination_number" expression="^1#(\d+)#(\d+)$"/>

Note the backslashes in front of the d+ entries. \d means "match a digit"
whereas a bare d means "make a lowercase d character".

Hope that helps.
-MC

P.S. - The * character does need to be escaped in regexes. See the
default.xml dialplan file for some obvious examples.

On Sat, Oct 10, 2009 at 6:24 AM, Milena <testeador01 at gmail.com> wrote:

> escape character is '\'try
>   <condition field="destination_number" expression="^1\#(d+)\#(d+)$"/>
> 2009/10/10 daqiang wang <wangdq.no1 at gmail.com>
>
>> hello every one :
>>    I want to match the # in XML dialplan , how to do  ?
>>    example :
>>       1#5555#6666 . how to do ?
>>      I do this :
>>      <condition field="destination_number" expression="^1#(d+)#(d+)$"/>
>>     but it's not work
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Sat Oct 10 10:36:09 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 10 Oct 2009 12:36:09 -0500
Subject: [Freeswitch-users] Difference between park and valet_park
In-Reply-To: <87f2f3b90910101022v52153d0dhd1ed774af84ecd22@mail.gmail.com>
References: <1452e2980910092351v766cad5bpcb55b59ad5fee6d0@mail.gmail.com>
	<87f2f3b90910101022v52153d0dhd1ed774af84ecd22@mail.gmail.com>
Message-ID: <191c3a030910101036y64c68a37i5a65bc73ba8aee69@mail.gmail.com>

park is an app you can never return from that just sits idle so you can
remote control the channel.
valet_park is an app to be used from the phones as a way to park and
retrieve calls.



On Sat, Oct 10, 2009 at 12:22 PM, Michael Collins <msc at freeswitch.org>wrote:

>
>
> On Fri, Oct 9, 2009 at 11:51 PM, velusamy velu <velu.technical at gmail.com>wrote:
>
>> Dear All,
>>      Could you please any one explain the difference between parking and
>> valet parking?
>>
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_valet_park
>
> Like the wiki says, valet_park puts calls into numbered parking stalls. A
> call that is parked in a numbered stall is retrieved by calling (or
> transferring) a second call into that numbered stall. That makes it easy to
> pass calls around. The regular park app puts calls in a special park state
> that is like being on hold, but the only way to get them out of park is to
> uuid_bridge a second leg to the uuid of the parked call leg.
>
> Read the wiki entry on valet_park and then read the one on park and you'll
> see that they are similar apps but the valet_park app is a bit easier for
> phone users to use for moving calls in and out of the park state.
>
> -MC
>
>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From moises.silva at gmail.com  Sat Oct 10 10:40:16 2009
From: moises.silva at gmail.com (Moises Silva)
Date: Sat, 10 Oct 2009 13:40:16 -0400
Subject: [Freeswitch-users] libpri_span vs prispans
In-Reply-To: <7d79b3930910092139v3589b761oab0bae46598118f@mail.gmail.com>
References: <7d79b3930910090459v2b07e3aapd715c7618f11ada0@mail.gmail.com>
	<20091009125714.E0FF73043D7@mail.cune.org>
	<87f2f3b90910091057q3a033548j462446cf424ad257@mail.gmail.com>
	<7d79b3930910092139v3589b761oab0bae46598118f@mail.gmail.com>
Message-ID: <c4d05cbe0910101040m4dea6a3bx7e7e2c2c983141ef@mail.gmail.com>

>From libs/openzap/mod_openzap/mod_openzap.c and
libs/openzap/src/ozmod_libpri/ozmod_libpri.c

node=cpe|net
switch=ni1|ni2|dms100|lucent5|5ess|att4ess|4ess|euroisdn|gr303eoc|gr303tmc
dp=international|national|local|isdn|private|unknown
l1=alaw|ulaw <-- this recently is set based on the trunk type, but old
openzap in E1 mode needs to explicitly set it to alaw
debug=q921_raw,q921_dump,q921_state,config,q931_dump,q931_state,q931_anomaly,apdu,acc
none or all
opts=suggest_channel,omit_display,omit_redirecting_number

The following values are commonly known.
context=
dialplan=

On Sat, Oct 10, 2009 at 12:39 AM, lakshmanan ganapathy <lakindia89 at gmail.com
> wrote:

> Where can I found entire configurations for a libpri spans. Right now I've
> only 4 things in my libpri configuration. They are Switchtype, Node,
> Contect, Dialplan, which is given as example in wiki.
>
> But I think libprispan has more configuration. Where can I find those
> things?
>
>
>
> On Fri, Oct 9, 2009 at 11:27 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Fri, Oct 9, 2009 at 5:57 AM, <Russell.Mosemann at cune.org> wrote:
>>
>>> lakshmanan ganapathy <lakindia89 at gmail.com> said:
>>> > What difference it will make, if I use pri_span configuration and
>>> lib_pri
>>> > span configuration in openzap.conf.xml.
>>> > I'm preety much confused on the difference between this.
>>>
>>> Openzap can handle a T1/E1 itself, or openzap can use libpri to do that.
>>> libpri is more mature than openzap. In my case openzap was not able to
>>> handle everything invovling a T1 here, and using libpri fixed it.
>>>
>>> A libpri_span configuration is handled by libpri. A pri_span
>>> configuration is handled natively by openzap. Use the one that works for
>>> you.
>>>
>>
>> Right now the libpri stack is more mature than the ozmod_isdn stuff. See
>> the OpenZAP page on the wiki for instructions on how to set up ozmod_libpri.
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From shaheryarkh at googlemail.com  Sat Oct 10 12:01:04 2009
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Sun, 11 Oct 2009 01:01:04 +0600
Subject: [Freeswitch-users] Questions regarding to mod_nibble
In-Reply-To: <ebbc2d550910100730o6abae5b5p254b72cf1c3869c4@mail.gmail.com>
References: <a60fed5e0910082039yc8b7ef8x24260041a7c11d31@mail.gmail.com>
	<ebbc2d550910100730o6abae5b5p254b72cf1c3869c4@mail.gmail.com>
Message-ID: <b16156850910101201v23b30ffcxec887a7681066079@mail.gmail.com>

I haven't don't any nibble bill project so i am not aware of its
configuration details but here are some of the feature of mod_nibble
(reference to wiki url http://wiki.freeswitch.org/wiki/Mod_nibblebill),

   - Debit credit/cash from accounts real-time
   - Allow for billing at different rates during a single call
   - Allow for warning callers when their balance is low (via audio,
   in-channel)
   - Allow for disconnecting or re-routing calls when balance is depleted
   - Allow billing functions listed above to operate with multiple
   concurrent calls

This means you can have packages where you offer users say X amount of free
minutes and when they are all used up, you can start charging cost Y WITHOUT
dropping user's call. All you need to do is (reference to JM's reply below),

1. Set X in cash column, and deduct 1 per 60 seconds. If you are billing on
different time scale then per minute charges then cash value would be,

Cash = X * 1/T

Where,
X = number of free minutes in package
T = billing interval in minutes

So, if you have 30 free minutes and 30 second billing interval then cash
would be 30 x 1/0.5 = 60.

2. After X minutes have passed during a call (i.e. when cash value becomes
zero), you can change billing rate to whatever you want to charge user on
per minute basis (or whatever your billing interval may be) without dropping
the call (but it would be good to warn user about it before you start
charging him money).

Lastly using different tables schema for mod_nibble may not be a good idea.
I recommend using a standard billing table like mod_nibble already has. You
can of course extend this schema to store addition information of your
choice OR you can use SQL Views to combine different tables and behave as a
single table for any pre-existing reports etc. But do remember that SQL
Views are normally Read Only in nearly all modren DBMS.

Thank you.


2009/10/10 Jo?o Mesquita <jmesquita at freeswitch.org>

> I am testing the latest version of nibblebill so let me see if I can help
> you with your questions.
>
> jm
>
> On Fri, Oct 9, 2009 at 12:39 AM, Ahmed Munir <ahmedmunir007 at gmail.com>wrote:
>
>> I want to ask three questions related to mod_nibble bills, as I'm listing
>> down below;
>>
>> 1- Can we select/use dynamic tables for billing using nibble bill?
>>
> What do you mean for dynamic tables? Like LCR does that you can specify
> your own SQL statement to be executed? If that's what you are asking, no,
> but it would be a nice todo.
>
>
>> 2- Can we define more than two tables and attributes in
>> nibblebill.conf.xml?
>>
> What else do you want to define and how do you imagine it to behave?
>
>
>> 3- As Nibble bill is use to deduct amount of user account, Can we deduct
>> minutes instead of cash? Because my case is, if a user buy a package and I
>> only want to deducts his/her minutes. How we can resolve it by nibble bill?
>> / What other way we can resolve it?
>>
> When we say cash on the column, we are really saying just a number that is
> being deducted, that's it. If you deduct 1 every 60 seconds, you will have
> your "cash" converted to minutes, won't you?
>
>
>>
>> Kindly advise soon.
>>
>> --
>> Regards,
>>
>> Ahmed Munir
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
________________________________________________________
|
                     |
| FATAL ERROR                                                           ---
O X |
|_______________________________________________________|
|                        You have moved the mouse.
 |
| Windows must be restarted for the changes to take effect.   |
|                                        <OK>
               |
####################################/


Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From lon at kickasspixels.com  Sat Oct 10 12:31:58 2009
From: lon at kickasspixels.com (Lon Baker)
Date: Sat, 10 Oct 2009 12:31:58 -0700
Subject: [Freeswitch-users] Music on hold volume?
Message-ID: <5d3e0dc60910101231x5dfb02a0w1e56068dc5ccdd56@mail.gmail.com>

Is there a way to adjust the volume of music on hold?

Lon



From brian at freeswitch.org  Sat Oct 10 12:47:28 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 10 Oct 2009 14:47:28 -0500
Subject: [Freeswitch-users] Music on hold volume?
In-Reply-To: <5d3e0dc60910101231x5dfb02a0w1e56068dc5ccdd56@mail.gmail.com>
References: <5d3e0dc60910101231x5dfb02a0w1e56068dc5ccdd56@mail.gmail.com>
Message-ID: <A1B37588-4AE5-43C6-9BCA-56A5F4F3D91C@freeswitch.org>

adjust the files themselves.

/b

On Oct 10, 2009, at 2:31 PM, Lon Baker wrote:

> Is there a way to adjust the volume of music on hold?
>
> Lon




From eric at rf.com  Sat Oct 10 13:33:28 2009
From: eric at rf.com (Eric Chamberlain)
Date: Sat, 10 Oct 2009 13:33:28 -0700
Subject: [Freeswitch-users] Gateways,
	configuration or directory with mod_xml_curl
Message-ID: <5232A6B3-8104-4AA5-918F-E951DF023E22@rf.com>

Hello,

We are looking at Freeswitch to solve a problem of ours, we have  
thousands of users with individual gateway information.  The user's  
will not register with us, but we need to register with their gateways  
on their behalf.

Since users will be constantly adding/changing/deleting gateways, we  
figure mox_xml_curl is the best way to maintain the configuration.

What we are not sure of is if we should be putting the gateway  
information in the directory entry for each user (gateways are not  
shared between users) or if the gateway should go in the configuration.

If the gateway configuration is in the directory, how does sofia know  
when the gateway configuration changes?  The reloadxml documentation  
doesn't talk about gateways in the directory, does it work with the  
directory.  If so, is it one curl call or one call per user?  Is there  
a way to tell sofia that only one user's gateway configuration has  
changed?

Likewise with gateways in the configuration.  Does reloadxml generate  
one curl call or many?

In both cases, is there a way to minimize parsing the data that has  
not changed?

--
Eric Chamberlain







From juanbackson at gmail.com  Sat Oct 10 16:02:38 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Sun, 11 Oct 2009 07:02:38 +0800
Subject: [Freeswitch-users] directory variables not set correct with the
	latest trunk
In-Reply-To: <27c25bc40910100305q5ae87ddbjc9999852724470d8@mail.gmail.com>
References: <27c25bc40910080320x7b3b380eu933797197e334cb8@mail.gmail.com>
	<191c3a030910080821w691aa78aw272d71dee7675ff1@mail.gmail.com>
	<27c25bc40910100305q5ae87ddbjc9999852724470d8@mail.gmail.com>
Message-ID: <27c25bc40910101602p523f472fp390d2cc8dbc23a47@mail.gmail.com>

Hi,

I am still stuck in trying to get curl directory variables to show up in
channel even after the user has registered.

Is this something that gets changed in the latest trunk or is it just my
mis-configuration?

please help.

jb

On Sat, Oct 10, 2009 at 6:05 PM, Juan Backson <juanbackson at gmail.com> wrote:

> Hi,
>
> My sip phone did authentication ( sending in SIP REGISTER etc ), but it is
> still not showing up.  How come?
>
> It was working before.
>
> thanks,
> juan
>
>
> On Thu, Oct 8, 2009 at 11:21 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> they only set if you authenticate on sip or run the set_user app
>>
>>
>>   On Thu, Oct 8, 2009 at 5:20 AM, Juan Backson <juanbackson at gmail.com>wrote:
>>
>>>  Hi,
>>>
>>> My application fails to set the appropriate variables using directory xml
>>> after using the latest trunk as of yesterday.
>>>
>>> My curl looks like:
>>>
>>>
>>>
>>> <document type="freeswitch/xml">
>>>   <section name="directory">
>>>     <domain name="192.168.1.102">
>>>       <user id="200002">
>>>         <params>
>>>           <param name="password" value="123456"></param>
>>>           <param name="dial-string"
>>> value="sofia/internal/sip:200002 at 192.168.1.11:29440
>>> ;rinstance=0245b2a59ddff837"></param>
>>>         </params>
>>>         <variables>
>>>           <variable name="account_id" value="20"></variable>
>>>           <variable name="vm-code" value="1000"></variable>
>>>         </variables>
>>>       </user>
>>>     </domain>
>>>   </section>
>>> </document>
>>>
>>>
>>>
>>> In the info app, I am not seeing account-id and vm-code anymore.
>>>
>>> How to fix that?
>>>
>>> juan
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From brian at freeswitch.org  Sat Oct 10 17:02:57 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 10 Oct 2009 19:02:57 -0500
Subject: [Freeswitch-users] directory variables not set correct with the
	latest trunk
In-Reply-To: <27c25bc40910101602p523f472fp390d2cc8dbc23a47@mail.gmail.com>
References: <27c25bc40910080320x7b3b380eu933797197e334cb8@mail.gmail.com>
	<191c3a030910080821w691aa78aw272d71dee7675ff1@mail.gmail.com>
	<27c25bc40910100305q5ae87ddbjc9999852724470d8@mail.gmail.com>
	<27c25bc40910101602p523f472fp390d2cc8dbc23a47@mail.gmail.com>
Message-ID: <675B4EDC-AC25-40C0-A88D-A83AAE4B4BCF@freeswitch.org>

It happens on invite during a call not on REGISTER... I can 100%  
confirm it works so I do not know what the heck you're doing.

/b

On Oct 10, 2009, at 6:02 PM, Juan Backson wrote:

> Hi,
>
> I am still stuck in trying to get curl directory variables to show  
> up in channel even after the user has registered.
>
> Is this something that gets changed in the latest trunk or is it  
> just my mis-configuration?
>
> please help.
>
> jb




From brian at freeswitch.org  Sat Oct 10 17:07:52 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 10 Oct 2009 19:07:52 -0500
Subject: [Freeswitch-users] directory variables not set correct with the
	latest trunk
In-Reply-To: <27c25bc40910101602p523f472fp390d2cc8dbc23a47@mail.gmail.com>
References: <27c25bc40910080320x7b3b380eu933797197e334cb8@mail.gmail.com>
	<191c3a030910080821w691aa78aw272d71dee7675ff1@mail.gmail.com>
	<27c25bc40910100305q5ae87ddbjc9999852724470d8@mail.gmail.com>
	<27c25bc40910101602p523f472fp390d2cc8dbc23a47@mail.gmail.com>
Message-ID: <137DBF84-F0E5-41C6-B327-4C81A20FB4F6@freeswitch.org>

Also check to make sure you're returning the correct format as per the  
XML curl page.

/b

On Oct 10, 2009, at 6:02 PM, Juan Backson wrote:

> Hi,
>
> I am still stuck in trying to get curl directory variables to show  
> up in channel even after the user has registered.
>
> Is this something that gets changed in the latest trunk or is it  
> just my mis-configuration?
>
> please help.
>
> jb




From wangdq.no1 at gmail.com  Sat Oct 10 19:45:17 2009
From: wangdq.no1 at gmail.com (daqiang wang)
Date: Sun, 11 Oct 2009 10:45:17 +0800
Subject: [Freeswitch-users] how to match '#' in XML dialplan ?
In-Reply-To: <87f2f3b90910101030y7d1142b5kbdac5ffdbc12334b@mail.gmail.com>
References: <ecd05ce30910100253q1424cd4an9be93ab434912b3f@mail.gmail.com>
	<c29b930b0910100624n7d5dd257h526d2815705703d3@mail.gmail.com>
	<87f2f3b90910101030y7d1142b5kbdac5ffdbc12334b@mail.gmail.com>
Message-ID: <ecd05ce30910101945k4399256cp9bfb800424eb1654@mail.gmail.com>

it's work . Thank you very much .

2009/10/11 Michael Collins <msc at freeswitch.org>

> Some characters need a backslash to match in a regular expression. However,
> # is not one of them. I think your regex is wrong:
> <condition field="destination_number" expression="^1#(d+)#(d+)$"/>
>
> It should probably be:
> <condition field="destination_number" expression="^1#(\d+)#(\d+)$"/>
>
> Note the backslashes in front of the d+ entries. \d means "match a digit"
> whereas a bare d means "make a lowercase d character".
>
> Hope that helps.
> -MC
>
> P.S. - The * character does need to be escaped in regexes. See the
> default.xml dialplan file for some obvious examples.
>
>
> On Sat, Oct 10, 2009 at 6:24 AM, Milena <testeador01 at gmail.com> wrote:
>
>> escape character is '\'try
>>   <condition field="destination_number" expression="^1\#(d+)\#(d+)$"/>
>> 2009/10/10 daqiang wang <wangdq.no1 at gmail.com>
>>
>>> hello every one :
>>>    I want to match the # in XML dialplan , how to do  ?
>>>    example :
>>>       1#5555#6666 . how to do ?
>>>      I do this :
>>>      <condition field="destination_number" expression="^1#(d+)#(d+)$"/>
>>>     but it's not work
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mayamatakeshi at gmail.com  Sat Oct 10 20:18:39 2009
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Sun, 11 Oct 2009 12:18:39 +0900
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Update: Valet
	Parking
In-Reply-To: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>
References: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>
Message-ID: <15b9404e0910102018oda8834ah8cc87787ee00c22b@mail.gmail.com>

On Fri, Oct 9, 2009 at 10:42 AM, Michael Collins <msc at freeswitch.org> wrote:

> FYI,
>
> The FreeSWITCH devs have added valet parking! Check it out:
> http://www.freeswitch.org/node/207
>
> Let us know what you think.
>

Very nice.

But I think a valet_unpark app is missing.
If the intention of the person sent to the valet lot is to retrieve a call
there, the person can assume the call was already retrieved by someone else
or that the caller hung up if he/she hears MOH. But it would be nicer to
have a valet_unpark app to fail and let the dialplan play a message.

regards,
takeshi
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From mayamatakeshi at gmail.com  Sat Oct 10 21:26:43 2009
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Sun, 11 Oct 2009 13:26:43 +0900
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Update: Valet
	Parking
In-Reply-To: <15b9404e0910102018oda8834ah8cc87787ee00c22b@mail.gmail.com>
References: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>
	<15b9404e0910102018oda8834ah8cc87787ee00c22b@mail.gmail.com>
Message-ID: <15b9404e0910102126g67697544ie6091ed3d326de4f@mail.gmail.com>

On Sun, Oct 11, 2009 at 12:18 PM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:

>
>
> On Fri, Oct 9, 2009 at 10:42 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> FYI,
>>
>> The FreeSWITCH devs have added valet parking! Check it out:
>> http://www.freeswitch.org/node/207
>>
>> Let us know what you think.
>>
>
> Very nice.
>
> But I think a valet_unpark app is missing.
> If the intention of the person sent to the valet lot
>

I meant "parking stall"


> is to retrieve a call there, the person can assume the call was already
> retrieved by someone else or that the caller hung up if he/she hears MOH.
> But it would be nicer to have a valet_unpark app to fail and let the
> dialplan play a message.
>
> regards,
> takeshi
>
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From red.rain.seven at gmail.com  Sun Oct 11 03:10:07 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Sun, 11 Oct 2009 18:10:07 +0800
Subject: [Freeswitch-users] how to match '#' in XML dialplan ?
In-Reply-To: <ecd05ce30910101945k4399256cp9bfb800424eb1654@mail.gmail.com>
References: <ecd05ce30910100253q1424cd4an9be93ab434912b3f@mail.gmail.com>
	<c29b930b0910100624n7d5dd257h526d2815705703d3@mail.gmail.com>
	<87f2f3b90910101030y7d1142b5kbdac5ffdbc12334b@mail.gmail.com>
	<ecd05ce30910101945k4399256cp9bfb800424eb1654@mail.gmail.com>
Message-ID: <59ad9ca10910110310hc271115ja11626bc119d3e25@mail.gmail.com>

Daqiang:


How do you make your IP phone not dial right after you press "#"? Usually
the IP phone will dial the number already once you pushed "#"




On Sun, Oct 11, 2009 at 10:45 AM, daqiang wang <wangdq.no1 at gmail.com> wrote:

> it's work . Thank you very much .
>
> 2009/10/11 Michael Collins <msc at freeswitch.org>
>
>> Some characters need a backslash to match in a regular expression.
>> However, # is not one of them. I think your regex is wrong:
>> <condition field="destination_number" expression="^1#(d+)#(d+)$"/>
>>
>> It should probably be:
>> <condition field="destination_number" expression="^1#(\d+)#(\d+)$"/>
>>
>> Note the backslashes in front of the d+ entries. \d means "match a digit"
>> whereas a bare d means "make a lowercase d character".
>>
>> Hope that helps.
>> -MC
>>
>> P.S. - The * character does need to be escaped in regexes. See the
>> default.xml dialplan file for some obvious examples.
>>
>>
>> On Sat, Oct 10, 2009 at 6:24 AM, Milena <testeador01 at gmail.com> wrote:
>>
>>> escape character is '\'try
>>>   <condition field="destination_number" expression="^1\#(d+)\#(d+)$"/>
>>> 2009/10/10 daqiang wang <wangdq.no1 at gmail.com>
>>>
>>>> hello every one :
>>>>    I want to match the # in XML dialplan , how to do  ?
>>>>    example :
>>>>       1#5555#6666 . how to do ?
>>>>      I do this :
>>>>      <condition field="destination_number" expression="^1#(d+)#(d+)$"/>
>>>>     but it's not work
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From ivdreg at gmail.com  Sun Oct 11 05:05:45 2009
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Sun, 11 Oct 2009 15:05:45 +0300
Subject: [Freeswitch-users] More complex hunting question
Message-ID: <b437a0e0910110505s20a176b0vf78603aa9d7e9d78@mail.gmail.com>

Dear All,

I'm trying to use FS with xml_curl as routing server for otgoing calls to my
provides. Here is simplified setup:

[REGISTRARs] ----> [APP SERVERs] ----> [ROUTING SERVER/FreeSwitch] ---->
[SBC/no Transcoder]----> [Terminating GWs/ITSPs]

Because of simplicity I do not use hunting functionality in SBC but only in
FS.
I try to avoid if possible any trascoding but there are cases that it is
necessary. I want ROUTING SERVER to be
configured in proxy_media mode or bypass_media but on some errors such as
"415 Unsupported Media" that supposed are codec
negotiation issue to hunt again via trascoder before SBC.

Can someone help to find some elegant way functionality described below to
be achieved via dialplan_xml ?
Call to +XXXXXXXXXXXXXXXX for example:
available routes by priority are via: ITSP1,ITSP5,ITSP2,ITSP7
outgoing leg to ITSP1 - (returns unallocated number)
outgoing leg to ITSP5 - (returns "415 Unsupported Media")
    new outgoing leg to ITSP5 again but via transcoder - (If call is
connected OK, If other error hunt again to next GW - ITSP2)
outgoing leg to ITSP2 - (returns "415 Unsupported Media")
    new outgoing leg to ITSP2 again but via transcoder - (If call is
connected OK, If other error hunt again to next GW - ITSP7)
outgoing leg to ITSP7
Remarks: Codec errors may be more that one (415 Unsupported Media/488 Not
Acceptable Here ...) and number it ITSP/GW in not fixed.


Thanks to all.
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From tculjaga at gmail.com  Sun Oct 11 09:50:00 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Sun, 11 Oct 2009 18:50:00 +0200
Subject: [Freeswitch-users] how to match '#' in XML dialplan ?
In-Reply-To: <59ad9ca10910110310hc271115ja11626bc119d3e25@mail.gmail.com>
References: <ecd05ce30910100253q1424cd4an9be93ab434912b3f@mail.gmail.com>
	<c29b930b0910100624n7d5dd257h526d2815705703d3@mail.gmail.com>
	<87f2f3b90910101030y7d1142b5kbdac5ffdbc12334b@mail.gmail.com>
	<ecd05ce30910101945k4399256cp9bfb800424eb1654@mail.gmail.com>
	<59ad9ca10910110310hc271115ja11626bc119d3e25@mail.gmail.com>
Message-ID: <65d96fc80910110950n61738c06jb59ca4b91d98c4bb@mail.gmail.com>

this is up to your phone.... # means address complete and you phone sends
the number you dialed into an INVITE message.

if you want to support FAC with # you should modify the phone's dialplan and
make it expect more digits... for certain prefixes.

T.


On Sun, Oct 11, 2009 at 12:10 PM, Henry Huang <red.rain.seven at gmail.com>wrote:

> Daqiang:
>
>
> How do you make your IP phone not dial right after you press "#"? Usually
> the IP phone will dial the number already once you pushed "#"
>
>
>
>
>
> On Sun, Oct 11, 2009 at 10:45 AM, daqiang wang <wangdq.no1 at gmail.com>wrote:
>
>> it's work . Thank you very much .
>>
>> 2009/10/11 Michael Collins <msc at freeswitch.org>
>>
>>> Some characters need a backslash to match in a regular expression.
>>> However, # is not one of them. I think your regex is wrong:
>>> <condition field="destination_number" expression="^1#(d+)#(d+)$"/>
>>>
>>> It should probably be:
>>> <condition field="destination_number" expression="^1#(\d+)#(\d+)$"/>
>>>
>>> Note the backslashes in front of the d+ entries. \d means "match a digit"
>>> whereas a bare d means "make a lowercase d character".
>>>
>>> Hope that helps.
>>> -MC
>>>
>>> P.S. - The * character does need to be escaped in regexes. See the
>>> default.xml dialplan file for some obvious examples.
>>>
>>>
>>> On Sat, Oct 10, 2009 at 6:24 AM, Milena <testeador01 at gmail.com> wrote:
>>>
>>>> escape character is '\'try
>>>>   <condition field="destination_number" expression="^1\#(d+)\#(d+)$"/>
>>>> 2009/10/10 daqiang wang <wangdq.no1 at gmail.com>
>>>>
>>>>> hello every one :
>>>>>    I want to match the # in XML dialplan , how to do  ?
>>>>>    example :
>>>>>       1#5555#6666 . how to do ?
>>>>>      I do this :
>>>>>      <condition field="destination_number" expression="^1#(d+)#(d+)$"/>
>>>>>     but it's not work
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Henry Huang
> UniC Solution - Communication Unified
> VoIP & Open Source software Consultant
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From shiyanov at gmail.com  Sun Oct 11 09:59:09 2009
From: shiyanov at gmail.com (Artem Shiyanov)
Date: Sun, 11 Oct 2009 20:59:09 +0400
Subject: [Freeswitch-users] mod_socket: custom timeout for HEARTBEAT event
Message-ID: <d211f54a0910110959y3d3deed7s43201a3e39756585@mail.gmail.com>

Hi all!
As it stays in wiki:
...
HEARTBEAT
Status information for freeswitch trigerred by freeswitch's heartbeat every
20 seconds.
...

Is there any way to customize timeout of HEARTBEAT events?


Thanks in advance,
Artem
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From edpimentl at gmail.com  Sun Oct 11 10:48:47 2009
From: edpimentl at gmail.com (EdPimentl)
Date: Sun, 11 Oct 2009 13:48:47 -0400
Subject: [Freeswitch-users] Is assistivetech.net confusing the market place
	with their use of FreeSwitch product name?
Message-ID: <9dc4a1670910111048y2802b155oc7086b975193c9d7@mail.gmail.com>

http://www.assistivetech.net/search/productDisplay.php?product_id=18854
-E
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From herman.griffin at gmail.com  Sun Oct 11 12:12:08 2009
From: herman.griffin at gmail.com (frek818)
Date: Sun, 11 Oct 2009 12:12:08 -0700 (PDT)
Subject: [Freeswitch-users] problem compiling esl for use with freepbx v3
In-Reply-To: <20090823185152.D17845FE@sinclaire.sibble.net>
References: <20090823185152.D17845FE@sinclaire.sibble.net>
Message-ID: <25846572.post@talk.nabble.com>


Did anyone find a solution to this problem? I too would like to install the
esl module for PHP.

Herman

Harondel J. Sibble wrote:
> 
> Okay, just finished installing 1.0.4 from tarball on Ubuntu 9.0.4 server, 
> then went to install FreePBX v3, I've gotten all the prerequisities in the 
> wizard fixed except for ESL
> 
> As per
> 
> http://wiki.freeswitch.org/wiki/Event_Socket_Library
> http://wiki.freeswitch.org/wiki/Event_Socket
> 
> I go into my FS source dir
> 
> /home/sibbleh/freeswitch-1.0.4/libs/esl
> 
> Run make and then "sudo make phpmod-install" 
> 
> and I get 
> 
> 
> $ sudo make phpmod-install
> make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-
> I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g
> -ggdb 
> -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-
> variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes"
> CXXFLAGS="-
> I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g
> -ggdb 
> -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" 
> CXX_CFLAGS="" -C php
> make[1]: Entering directory `/home/sibbleh/freeswitch-1.0.4/libs/esl/php'
> g++  -I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE
> -g 
> -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable -
> I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM -
> I/usr/include/php5/Zend -I/usr/include/php5/ext -
> I/usr/include/php5/ext/date/lib -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64
> -
> Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o
> cc1plus: warnings being treated as errors
> esl_wrap.cpp: In function 'void _wrap_ESLevent_event_set(int, zval*,
> zval**, 
> zval*, int)':
> esl_wrap.cpp:1047: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_event_get(int, zval*,
> zval**, 
> zval*, int)':
> esl_wrap.cpp:1073: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_serialized_string_set(int, 
> zval*, zval**, zval*, int)':
> esl_wrap.cpp:1111: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_serialized_string_get(int, 
> zval*, zval**, zval*, int)':
> esl_wrap.cpp:1141: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_mine_set(int, zval*,
> zval**, 
> zval*, int)':
> esl_wrap.cpp:1172: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_mine_get(int, zval*,
> zval**, 
> zval*, int)':
> esl_wrap.cpp:1198: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_0(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1234: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_1(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1269: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_2(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1294: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_new_ESLevent(int, zval*, zval**,
> zval*, 
> int)':
> esl_wrap.cpp:1346: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_serialize(int, zval*,
> zval**, 
> zval*, int)':
> esl_wrap.cpp:1403: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_setPriority(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1441: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_getHeader(int, zval*,
> zval**, 
> zval*, int)':
> esl_wrap.cpp:1478: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_getBody(int, zval*, zval**, 
> zval*, int)':
> esl_wrap.cpp:1508: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_getType(int, zval*, zval**, 
> zval*, int)':
> esl_wrap.cpp:1538: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_addBody(int, zval*, zval**, 
> zval*, int)':
> esl_wrap.cpp:1571: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_addHeader(int, zval*,
> zval**, 
> zval*, int)':
> esl_wrap.cpp:1611: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_delHeader(int, zval*,
> zval**, 
> zval*, int)':
> esl_wrap.cpp:1644: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_firstHeader(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1674: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_nextHeader(int, zval*,
> zval**, 
> zval*, int)':
> esl_wrap.cpp:1704: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_new_ESLconnection__SWIG_0(int,
> zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1744: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_new_ESLconnection__SWIG_1(int,
> zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1770: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_new_ESLconnection(int, zval*,
> zval**, 
> zval*, int)':
> esl_wrap.cpp:1803: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_socketDescriptor(int, 
> zval*, zval**, zval*, int)':
> esl_wrap.cpp:1846: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_connected(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1872: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_getInfo(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1898: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_send(int, zval*,
> zval**, 
> zval*, int)':
> esl_wrap.cpp:1931: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_sendRecv(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1964: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_api(int, zval*,
> zval**, 
> zval*, int)':
> esl_wrap.cpp:2007: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_bgapi(int, zval*,
> zval**, 
> zval*, int)':
> esl_wrap.cpp:2050: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_sendEvent(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:2082: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_recvEvent(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:2108: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_recvEventTimed(int, 
> zval*, zval**, zval*, int)':
> esl_wrap.cpp:2141: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_filter(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:2181: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_events(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:2221: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_execute(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:2272: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_executeAsync(int,
> zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:2323: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_setAsyncExecute(int, 
> zval*, zval**, zval*, int)':
> esl_wrap.cpp:2356: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_setEventLock(int,
> zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:2389: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_disconnect(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:2415: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_eslSetLogLevel(int, zval*, zval**, 
> zval*, int)':
> esl_wrap.cpp:2438: error: format not a string literal and no format
> arguments
> make[1]: *** [esl_wrap.o] Error 1
> make[1]: Leaving directory `/home/sibbleh/freeswitch-1.0.4/libs/esl/php'
> make: *** [phpmod] Error 2
> 
> Same thing happens if I try sudo make everymod
> 
> Checking the list archives I found this thread
> 
> http://www.nabble.com/ESL-Wrapper-td22209991.html#a22222338
> 
> I've made sure that the php-dev packages are installed. Any suggestions on 
> what to do next?
> -- 
> Harondel J. Sibble 
> Sibble Computer Consulting
> Creating Solutions for the small and medium business computer user.
> help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
> (604) 739-3709 (voice)
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/problem-compiling-esl-for-use-with-freepbx-v3-tp25106337p25846572.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From dave at 3c.co.uk  Sun Oct 11 12:33:54 2009
From: dave at 3c.co.uk (David Knell)
Date: Sun, 11 Oct 2009 12:33:54 -0700
Subject: [Freeswitch-users] Gateways,
	configuration or directory with	mod_xml_curl
In-Reply-To: <5232A6B3-8104-4AA5-918F-E951DF023E22@rf.com>
References: <5232A6B3-8104-4AA5-918F-E951DF023E22@rf.com>
Message-ID: <1255289634.5451.19.camel@localhost.localdomain>

Hi Eric -

The way that we do it is to keep each gateway in its own Sofia profile.
Issuing 
api sofia profile <profile> start reloadxml
does one call to the web server for that profile's XML, which can be
pretty compact if it just contains one gateway.

--Dave

> Hello,
> 
> We are looking at Freeswitch to solve a problem of ours, we have  
> thousands of users with individual gateway information.  The user's  
> will not register with us, but we need to register with their gateways  
> on their behalf.
> 
> Since users will be constantly adding/changing/deleting gateways, we  
> figure mox_xml_curl is the best way to maintain the configuration.
> 
> What we are not sure of is if we should be putting the gateway  
> information in the directory entry for each user (gateways are not  
> shared between users) or if the gateway should go in the configuration.
> 
> If the gateway configuration is in the directory, how does sofia know  
> when the gateway configuration changes?  The reloadxml documentation  
> doesn't talk about gateways in the directory, does it work with the  
> directory.  If so, is it one curl call or one call per user?  Is there  
> a way to tell sofia that only one user's gateway configuration has  
> changed?
> 
> Likewise with gateways in the configuration.  Does reloadxml generate  
> one curl call or many?
> 
> In both cases, is there a way to minimize parsing the data that has  
> not changed?
> 
> --
> Eric Chamberlain
> 
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 




From mike at jerris.com  Sun Oct 11 14:34:30 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Oct 2009 17:34:30 -0400
Subject: [Freeswitch-users] mod_sofia.c registered calls how to know
In-Reply-To: <f8af5740910070748g1efee71ft5841b6fa4d2412c8@mail.gmail.com>
References: <f8af5740910070748g1efee71ft5841b6fa4d2412c8@mail.gmail.com>
Message-ID: <14A1216A-61C3-4C03-B710-57D2E7F4CD7A@jerris.com>


On Oct 7, 2009, at 10:48 AM, srinivasula reddy wrote:

>
> Hi
>
> can any please tell me where registered calls are stored, so when  
> incoming call came to mod_sofia.c how it will check it is registered  
> or not?\\
>


Calls are not registered and calls have nothing to do with  
registration.  Users are registered so that you may send calls to  
them.  Registration data is stored either in a sqlite database, or  
optionally if you setup odbc, in another database of your choice.  If  
you try to send a call to an unregistered user in the dialplan using  
the proper syntax to send calls to registered users (see the wiki for  
more details), and that user is not registered, the bridge app will  
fail, optionally letting you continue on in the dialplan based on  
variables such as continue_on_fail and hangup_after_bridge.  You can  
use the sofia_contact function to see if there is anyone registered to  
a specific user.

Mike

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From herman.griffin at gmail.com  Sun Oct 11 14:36:21 2009
From: herman.griffin at gmail.com (Herman Griffin)
Date: Sun, 11 Oct 2009 14:36:21 -0700
Subject: [Freeswitch-users] problem compiling esl for use with freepbx v3
In-Reply-To: <25846572.post@talk.nabble.com>
References: <20090823185152.D17845FE@sinclaire.sibble.net>
	<25846572.post@talk.nabble.com>
Message-ID: <4d6f26b0910111436h416121b7s2a4b5c79ffc01db0@mail.gmail.com>

Although probably not the best solution, I figured out a way to make it
compile and install:

I removed all of the -Werror instances in
PATH_TO_FREESWITCH_SOURCE/libs/esl/Makefile

If I was a hardcore c/c++ programmer, I'd figure out the real problem.

Herman aka frek818

On Sun, Oct 11, 2009 at 12:12 PM, frek818 <herman.griffin at gmail.com> wrote:

>
> Did anyone find a solution to this problem? I too would like to install the
> esl module for PHP.
>
> Herman
>
> Harondel J. Sibble wrote:
> >
> > Okay, just finished installing 1.0.4 from tarball on Ubuntu 9.0.4 server,
> > then went to install FreePBX v3, I've gotten all the prerequisities in
> the
> > wizard fixed except for ESL
> >
> > As per
> >
> > http://wiki.freeswitch.org/wiki/Event_Socket_Library
> > http://wiki.freeswitch.org/wiki/Event_Socket
> >
> > I go into my FS source dir
> >
> > /home/sibbleh/freeswitch-1.0.4/libs/esl
> >
> > Run make and then "sudo make phpmod-install"
> >
> > and I get
> >
> >
> > $ sudo make phpmod-install
> > make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-
> > I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g
> > -ggdb
> > -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-
> > variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes"
> > CXXFLAGS="-
> > I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g
> > -ggdb
> > -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable"
> > CXX_CFLAGS="" -C php
> > make[1]: Entering directory `/home/sibbleh/freeswitch-1.0.4/libs/esl/php'
> > g++  -I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include
> -DHAVE_EDITLINE
> > -g
> > -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable
> -
> > I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM -
> > I/usr/include/php5/Zend -I/usr/include/php5/ext -
> > I/usr/include/php5/ext/date/lib -D_LARGEFILE_SOURCE
> -D_FILE_OFFSET_BITS=64
> > -
> > Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o
> > cc1plus: warnings being treated as errors
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_event_set(int, zval*,
> > zval**,
> > zval*, int)':
> > esl_wrap.cpp:1047: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_event_get(int, zval*,
> > zval**,
> > zval*, int)':
> > esl_wrap.cpp:1073: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_serialized_string_set(int,
> > zval*, zval**, zval*, int)':
> > esl_wrap.cpp:1111: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_serialized_string_get(int,
> > zval*, zval**, zval*, int)':
> > esl_wrap.cpp:1141: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_mine_set(int, zval*,
> > zval**,
> > zval*, int)':
> > esl_wrap.cpp:1172: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_mine_get(int, zval*,
> > zval**,
> > zval*, int)':
> > esl_wrap.cpp:1198: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_0(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1234: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_1(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1269: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_2(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1294: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_new_ESLevent(int, zval*, zval**,
> > zval*,
> > int)':
> > esl_wrap.cpp:1346: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_serialize(int, zval*,
> > zval**,
> > zval*, int)':
> > esl_wrap.cpp:1403: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_setPriority(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1441: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_getHeader(int, zval*,
> > zval**,
> > zval*, int)':
> > esl_wrap.cpp:1478: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_getBody(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1508: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_getType(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1538: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_addBody(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1571: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_addHeader(int, zval*,
> > zval**,
> > zval*, int)':
> > esl_wrap.cpp:1611: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_delHeader(int, zval*,
> > zval**,
> > zval*, int)':
> > esl_wrap.cpp:1644: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_firstHeader(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1674: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_nextHeader(int, zval*,
> > zval**,
> > zval*, int)':
> > esl_wrap.cpp:1704: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_new_ESLconnection__SWIG_0(int,
> > zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1744: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_new_ESLconnection__SWIG_1(int,
> > zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1770: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_new_ESLconnection(int, zval*,
> > zval**,
> > zval*, int)':
> > esl_wrap.cpp:1803: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_socketDescriptor(int,
> > zval*, zval**, zval*, int)':
> > esl_wrap.cpp:1846: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_connected(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1872: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_getInfo(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1898: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_send(int, zval*,
> > zval**,
> > zval*, int)':
> > esl_wrap.cpp:1931: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_sendRecv(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1964: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_api(int, zval*,
> > zval**,
> > zval*, int)':
> > esl_wrap.cpp:2007: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_bgapi(int, zval*,
> > zval**,
> > zval*, int)':
> > esl_wrap.cpp:2050: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_sendEvent(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2082: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_recvEvent(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2108: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_recvEventTimed(int,
> > zval*, zval**, zval*, int)':
> > esl_wrap.cpp:2141: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_filter(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2181: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_events(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2221: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_execute(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2272: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_executeAsync(int,
> > zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2323: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_setAsyncExecute(int,
> > zval*, zval**, zval*, int)':
> > esl_wrap.cpp:2356: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_setEventLock(int,
> > zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2389: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_disconnect(int,
> zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2415: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_eslSetLogLevel(int, zval*, zval**,
> > zval*, int)':
> > esl_wrap.cpp:2438: error: format not a string literal and no format
> > arguments
> > make[1]: *** [esl_wrap.o] Error 1
> > make[1]: Leaving directory `/home/sibbleh/freeswitch-1.0.4/libs/esl/php'
> > make: *** [phpmod] Error 2
> >
> > Same thing happens if I try sudo make everymod
> >
> > Checking the list archives I found this thread
> >
> > http://www.nabble.com/ESL-Wrapper-td22209991.html#a22222338
> >
> > I've made sure that the php-dev packages are installed. Any suggestions
> on
> > what to do next?
> > --
> > Harondel J. Sibble
> > Sibble Computer Consulting
> > Creating Solutions for the small and medium business computer user.
> > help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
> > (604) 739-3709 (voice)
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> --
> View this message in context:
> http://www.nabble.com/problem-compiling-esl-for-use-with-freepbx-v3-tp25106337p25846572.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From diego.viola at gmail.com  Sun Oct 11 14:40:15 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 11 Oct 2009 21:40:15 +0000
Subject: [Freeswitch-users] REMINDER: Weekly FreeSWITCH Conference
	Scheduled for October 9, 11AM CST (GMT -6)
In-Reply-To: <87f2f3b90910081202m5dd3f567v240541dd2009c9a8@mail.gmail.com>
References: <87f2f3b90910081202m5dd3f567v240541dd2009c9a8@mail.gmail.com>
Message-ID: <86a32abc0910111440n184b882fwee29736bb15850ef@mail.gmail.com>

I'd like to add this for the next weekly conference.

I have added a few events to the event list, as you can see here:

http://wiki.freeswitch.org/wiki/Event_list

But I need more help from the community to complete that and add content to
the events, etc. So if you can add that for the next weekly conference it
would be really nice :).

Thanks,

Diego

On Thu, Oct 8, 2009 at 7:02 PM, Michael Collins <msc at freeswitch.org> wrote:

> Just a friendly reminder that we will be having the weekly FreeSWITCH
> conference call tomorrow, October 9th, at 11AM Central. The agenda is
> updated:
> http://bit.ly/lzEYy
>
> The conference lasts for six hours, so feel free to dial in at any time. We
> usually start going over the agenda about 15 minutes after the conference
> starts. The agenda itself only takes about an hour or two, after which we
> spend most of the time discussing whatever is on the minds of the community
> members. Remember, the core FreeSWITCH development team of Anthony Minessale
> (primary author, lead architect), Mike Jerris (build master), and Brian West
> (all around VoIP expert, FS configuration guru) are present for the whole
> conference. This is a great time to ask questions about the project. Please
> join us!
>
> Call in options include SIP, Skype, PSTN, and Jabber. Please see the agenda
> for more information.
>
> Looking forward to speaking with you all tomorrow,
> Michael
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From diego.viola at gmail.com  Sun Oct 11 14:44:59 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 11 Oct 2009 21:44:59 +0000
Subject: [Freeswitch-users] mod_socket: custom timeout for HEARTBEAT
	event
In-Reply-To: <d211f54a0910110959y3d3deed7s43201a3e39756585@mail.gmail.com>
References: <d211f54a0910110959y3d3deed7s43201a3e39756585@mail.gmail.com>
Message-ID: <86a32abc0910111444j1ab2e446p3cb94b0ae04126e3@mail.gmail.com>

Maybe you want enable_heartbeat or sched_heartbeat from mod_dptools?

You can pass your parameters in second to these two.

Example:

<action application="enable_heartbeat" data="1"/>
<action application="sched_heartbeat" data="1"/>

Where 1 in this case is the number of heartbeats per seconds.

You can use that example on the Dialplan XML but you can also use it on
mod_event_socket outbound, etc.

Best regards,

Diego



On Sun, Oct 11, 2009 at 4:59 PM, Artem Shiyanov <shiyanov at gmail.com> wrote:

> Hi all!
> As it stays in wiki:
> ...
> HEARTBEAT
> Status information for freeswitch trigerred by freeswitch's heartbeat every
> 20 seconds.
> ...
>
> Is there any way to customize timeout of HEARTBEAT events?
>
>
> Thanks in advance,
> Artem
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Sun Oct 11 14:52:08 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Oct 2009 17:52:08 -0400
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <903da5680910082341r485092vb0cd10c43c4a2534@mail.gmail.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<87f2f3b90910080958i1f5cfccax621fd12100a05e3e@mail.gmail.com>
	<903da5680910081201p427a8358n768edb61a96674a4@mail.gmail.com>
	<87f2f3b90910081538l6e4e4b7dv3aedb2c4d1af58bc@mail.gmail.com>
	<903da5680910082238x27bff476s8bea0e74ab516a8d@mail.gmail.com>
	<87f2f3b90910082308i7f6144d9kc887596be54dd609@mail.gmail.com>
	<903da5680910082341r485092vb0cd10c43c4a2534@mail.gmail.com>
Message-ID: <219716B3-C355-4128-A333-1459AC63C364@jerris.com>


On Oct 9, 2009, at 2:41 AM, Gabriel Gunderson wrote:

> On Fri, Oct 9, 2009 at 12:08 AM, Michael Collins  
> <msc at freeswitch.org> wrote:
>> Thanks for reporting back. Please let all the Asterisk users know  
>> that they
>> are welcome to join us in #freeswitch on irc.freenode.net and that  
>> they will
>> not be abused like people do in other less friendly IRC channels.
>
> Funny you mention this.  Many people report that the way the FS
> community refers to Asterisk in docs/wikis/irc/whatever makes the FS
> camp seem *less* welcoming to them.  After all, they identify as
> Asterisk Users and take the criticism as being kinda harsh.  Most of
> them acknowledge the shortcomings of Asterisk but are put off when
> someone else points them out.  It's crazy, I know.  The thing is, I
> remember thinking that too.  After getting to know FS better, I didn't
> notice it as much.  Nobody likes to hear their baby is ugly --even if
> they know it is.
>
> At our session, and in general, I've noticed people are more
> interested in hearing about FS when you don't make direct comparisons
> to Asterisk.  Besides, FS stands on it's own merit.
>
> Just what I've observed *and* my 2 additional cents.


I have certainly seen this on irc in the past and we should do our  
best to avoid this, I have not seen this in the docs or wiki, do you  
know of any specifics you can point me to so we can correct this issue.

Mike




From mike at jerris.com  Sun Oct 11 14:55:02 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Oct 2009 17:55:02 -0400
Subject: [Freeswitch-users] gateway FS informs it's client FS about
	users hanged up with a long delay
In-Reply-To: <1255099232783-3795011.post@n2.nabble.com>
References: <1A16501E57CA4727A593226A8C810308@procent>
	<98F72CCB-88B9-47ED-AED3-B6BA6DE648C0@jerris.com>
	<1255007884165-3787956.post@n2.nabble.com>
	<87f2f3b90910081000v5b505f4ao53104211b620ea77@mail.gmail.com>
	<1255099232783-3795011.post@n2.nabble.com>
Message-ID: <4EB2CA6F-F7A6-4206-9E68-0457704F5418@jerris.com>


On Oct 9, 2009, at 10:40 AM, Maciej Aniserowicz wrote:

> Hello,
> The issue is resolved. I feel stupid, because Michael Jerris was  
> right the first time. Setting external_rtp_ip and external_sip_ip to  
> $${local_ip_v4} made it work.
> But the strange thing is: it SOMETIMES worked before without any  
> delay, which 'should not be possible', because the original IP was  
> my external ip and the BYE message was sent straight to it. And  
> there is no way it could reach the target 'internal' FS, because it  
> runs on virtual machine, and no ports are forwarded on my router.
> Any thoughts? Why this could (rarely) work even with the previous  
> config?
>
> Thanks to both of you for your answers.
>
> MA
>
>
It could work at least in regards to rtp if you had something else on  
the other side that adjusted to incorrect rtp ip like freeswitch does,  
the bye probably never really worked.

Mike

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From mike at jerris.com  Sun Oct 11 14:59:38 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Oct 2009 17:59:38 -0400
Subject: [Freeswitch-users] On the handling of SIP headers
In-Reply-To: <2d9149cd0910091210s33fb2bb9oc0c94a64aa87e0be@mail.gmail.com>
References: <2d9149cd0910091210s33fb2bb9oc0c94a64aa87e0be@mail.gmail.com>
Message-ID: <85095FC9-4DC9-414D-A43F-8A7583D66DFF@jerris.com>

There is this endless push and pull on this topic, those who want them  
assume it should be default, those who don't assume that should be  
default.  This probably needs a configuration option defaulting to  
pass them (those who don't want to pass them are usually a bit more  
educated and would find the option better than the other way around).

Mike

On Oct 9, 2009, at 3:10 PM, Kristian Kielhofner wrote:

> Hello everyone,
>
>  In using FS for various scenarios I've noticed some behavior that
> I'm not sure is completely "proper".  Given that this probably lives
> in mod_sofia who knows what's really "proper".  It is SIP after all...
>
>  So the issue comes up when using FreeSWITCH as a B2BUA and bridging
> between endpoints (very common).  Should FreeSWITCH copy the X-
> headers (possibly others) as it does now?  I'd like to think it
> shouldn't by default and the behavior should be one of:
>
> 1)  Don't pass X-* (or anything else, really) from one leg to another.
> If you want to pass specific X- headers (or anything else), set them
> explicitly on the outbound leg.
> 2)  Make the behavior configurable with a channel variable and/or
> sofia config option:
>
> {sip_pass_headers=all|none|X-MyCustomHeaderByName}
>
>  Thoughts?




From mike at jerris.com  Sun Oct 11 15:04:41 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Oct 2009 18:04:41 -0400
Subject: [Freeswitch-users] Sending an Event to a Session for onInput
In-Reply-To: <4256bf830910091234t11f4f4d0w28283f341bd1fb06@mail.gmail.com>
References: <4256bf830910091234t11f4f4d0w28283f341bd1fb06@mail.gmail.com>
Message-ID: <ACA9A602-0925-4D6F-8B11-0B214A462500@jerris.com>

We don't have session messages directly exposed, except for things  
like display, respond, and deflect.  What specifically are you trying  
to send ?

Mike

On Oct 9, 2009, at 3:34 PM, Matthew Fong wrote:

> I'm used to using the onInput callbacks inside lua and javascript to  
> listen for dtmf and other events and perform a task accordingly. I'm  
> wondering if there is a way to send an event to a session or channel  
> that can be caught using the setInputCallback inside lua from  
> outside the session program. Maybe an API command that can generate  
> an event for a specific UUID. Does a mechanism exist to do this that  
> I'm over looking? Thanks.




From diego.viola at gmail.com  Sun Oct 11 15:06:25 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 11 Oct 2009 22:06:25 +0000
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <219716B3-C355-4128-A333-1459AC63C364@jerris.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<87f2f3b90910080958i1f5cfccax621fd12100a05e3e@mail.gmail.com>
	<903da5680910081201p427a8358n768edb61a96674a4@mail.gmail.com>
	<87f2f3b90910081538l6e4e4b7dv3aedb2c4d1af58bc@mail.gmail.com>
	<903da5680910082238x27bff476s8bea0e74ab516a8d@mail.gmail.com>
	<87f2f3b90910082308i7f6144d9kc887596be54dd609@mail.gmail.com>
	<903da5680910082341r485092vb0cd10c43c4a2534@mail.gmail.com>
	<219716B3-C355-4128-A333-1459AC63C364@jerris.com>
Message-ID: <86a32abc0910111506m4e6d2740tbd3ac5fe0f59178@mail.gmail.com>

I have seen this on the wiki too, for example:

Q: Does it require hardware, kernel modules, ztdumshit, etc?
>
> Nope! :D You must be thinking of [http://sofaswitch.org/eg/aac.jpg|Something
> Else]
>


I know that's just a joke and we might make one or two jokes, but we don't
really hate Asterisk and we welcome them, there are some people who might
get offended by the jokes (specially Asterisk users) but I have already
corrected these from the wiki.

Thanks for the feedback Gabriel.

Diego

On Sun, Oct 11, 2009 at 9:52 PM, Michael Jerris <mike at jerris.com> wrote:

>
> On Oct 9, 2009, at 2:41 AM, Gabriel Gunderson wrote:
>
> > On Fri, Oct 9, 2009 at 12:08 AM, Michael Collins
> > <msc at freeswitch.org> wrote:
> >> Thanks for reporting back. Please let all the Asterisk users know
> >> that they
> >> are welcome to join us in #freeswitch on irc.freenode.net and that
> >> they will
> >> not be abused like people do in other less friendly IRC channels.
> >
> > Funny you mention this.  Many people report that the way the FS
> > community refers to Asterisk in docs/wikis/irc/whatever makes the FS
> > camp seem *less* welcoming to them.  After all, they identify as
> > Asterisk Users and take the criticism as being kinda harsh.  Most of
> > them acknowledge the shortcomings of Asterisk but are put off when
> > someone else points them out.  It's crazy, I know.  The thing is, I
> > remember thinking that too.  After getting to know FS better, I didn't
> > notice it as much.  Nobody likes to hear their baby is ugly --even if
> > they know it is.
> >
> > At our session, and in general, I've noticed people are more
> > interested in hearing about FS when you don't make direct comparisons
> > to Asterisk.  Besides, FS stands on it's own merit.
> >
> > Just what I've observed *and* my 2 additional cents.
>
>
> I have certainly seen this on irc in the past and we should do our
> best to avoid this, I have not seen this in the docs or wiki, do you
> know of any specifics you can point me to so we can correct this issue.
>
> Mike
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Sun Oct 11 15:14:50 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Oct 2009 18:14:50 -0400
Subject: [Freeswitch-users] Bad sound quality while eavesdropping
In-Reply-To: <1255169044209-3799274.post@n2.nabble.com>
References: <41A44DD027064988A914974405788C2E@procent>
	<191c3a030910050731m2d74979ep4598e5a1945d58ae@mail.gmail.com>
	<1254901192035-3780245.post@n2.nabble.com>
	<8437F5BC-7AFF-4A74-B8CD-C5B8219021F6@jerris.com>
	<1255008427639-3788019.post@n2.nabble.com>
	<191c3a030910080823g79c7c596x1cd887e1538ce2e1@mail.gmail.com>
	<1255169044209-3799274.post@n2.nabble.com>
Message-ID: <CF8C1791-24CD-4132-83B5-7F83B5BDCCC3@jerris.com>

can you confirm from an rtp packet trace that they are all really  
sending 20ms?

Mike

On Oct 10, 2009, at 6:04 AM, Maciej Aniserowicz wrote:

>
> Hi,
> Here are the messages with a:ptime parameter. All the calls are  
> started by
> commands sent through socket.
> I'm not sure if this is all information you need, please let me know  
> if
> something is missing here and I'll post that.
>
> 1) starting connection with x-lite (number 2003, the eavesdropper):
>
>   INVITE sip:2003 at 192.168.3.100:60188;rinstance=80b8f8d92af87cd2 SIP/ 
> 2.0
>   Via: SIP/2.0/UDP  
> 192.168.3.159:15060;rport;branch=z9hG4bKD9meHQN5XQ88K
>   Max-Forwards: 69
>   From: "MyApp" <sip:0000000000 at 192.168.3.159>;tag=jpQ6D7D2jUXvF
>   To: <sip:2003 at 192.168.3.100:60188;rinstance=80b8f8d92af87cd2>
>   Call-ID: 663c4500-3024-122d-57a2-9be8d85a4cff
>   CSeq: 121465610 INVITE
>   Contact: <sip:mod_sofia at 192.168.3.159:15060>
>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>   Supported: timer, precondition, path, replaces
>   Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>   Content-Type: application/sdp
>   Content-Disposition: session
>   Content-Length: 447
>   Remote-Party-ID: "MyApp"
> <sip:0000000000 at 192.168.3.159>;party=calling;screen=yes;privacy=off
>
>   v=0
>   o=FreeSWITCH 2223565947735016740 3096553520713245589 IN IP4  
> 192.168.3.159
>   s=FreeSWITCH
>   c=IN IP4 192.168.3.159
>   t=0 0
>   m=audio 29966 RTP/AVP 0 115 107 9 8 3 101 13
>   a=rtpmap:0 PCMU/8000
>   a=rtpmap:115 G7221/32000
>   a=fmtp:115 bitrate=48000
>   a=rtpmap:107 G7221/16000
>   a=fmtp:107 bitrate=32000
>   a=rtpmap:9 G722/8000
>   a=rtpmap:8 PCMA/8000
>   a=rtpmap:3 GSM/8000
>   a=rtpmap:101 telephone-event/8000
>   a=fmtp:101 0-16
>   a=rtpmap:13 CN/8000
>   a=ptime:20
>
>
> 2) starting connection with cisco ip phone (number 2006, first leg of
> eavesdropped session):
>
>   INVITE sip:2006 at 192.168.2.106:5060;user=phone SIP/2.0
>   Via: SIP/2.0/UDP  
> 192.168.3.159:15060;rport;branch=z9hG4bKg40rp87FNjB1p
>   Max-Forwards: 69
>   From: "MyApp" <sip:0000000000 at 192.168.3.159>;tag=Q3N2pe2K47ctS
>   To: <sip:2006 at 192.168.2.106:5060;user=phone>
>   Call-ID: 6e6fd8a0-3024-122d-57a2-9be8d85a4cff
>   CSeq: 121465616 INVITE
>   Contact: <sip:mod_sofia at 192.168.3.159:15060>
>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>   Supported: timer, precondition, path, replaces
>   Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>   Content-Type: application/sdp
>   Content-Disposition: session
>   Content-Length: 447
>   Remote-Party-ID: "MyApp"
> <sip:0000000000 at 192.168.3.159>;party=calling;screen=yes;privacy=off
>
>   v=0
>   o=FreeSWITCH 8453497903781974949 2294419114567885490 IN IP4  
> 192.168.3.159
>   s=FreeSWITCH
>   c=IN IP4 192.168.3.159
>   t=0 0
>   m=audio 17670 RTP/AVP 0 115 107 9 8 3 101 13
>   a=rtpmap:0 PCMU/8000
>   a=rtpmap:115 G7221/32000
>   a=fmtp:115 bitrate=48000
>   a=rtpmap:107 G7221/16000
>   a=fmtp:107 bitrate=32000
>   a=rtpmap:9 G722/8000
>   a=rtpmap:8 PCMA/8000
>   a=rtpmap:3 GSM/8000
>   a=rtpmap:101 telephone-event/8000
>   a=fmtp:101 0-16
>   a=rtpmap:13 CN/8000
>   a=ptime:20
>
>
> 3) starting connection with extension playing a file (number 9999,  
> second
> leg of eavesdropped session):
>
>   SIP/2.0 200 OK
>   Via: SIP/2.0/UDP
> 192.168.3.159:5080;rport=5080;branch=z9hG4bKv6Dg7myj4tvBS
>   From: "FreeSWITCH"  
> <sip:myuser at mydomain;transport=udp>;tag=091j2Q0Fre8vp
>   To: <sip:9999 at 192.168.3.159:15060>;tag=U7t5Xt51rB64Q
>   Call-ID: 7551b7d0-3024-122d-b3a6-156c4939c4f8
>   CSeq: 121465623 INVITE
>   Contact: <sip:mod_sofia at 192.168.3.159:15060;transport=udp>
>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN
>   Accept: application/sdp
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>   Supported: timer, precondition, path, replaces
>   Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>   Content-Type: application/sdp
>   Content-Disposition: session
>   Content-Length: 263
>
>   v=0
>   o=FreeSWITCH 5614997529598779838 636912243381649698 IN IP4  
> 192.168.3.159
>   s=FreeSWITCH
>   c=IN IP4 192.168.3.159
>   t=0 0
>   m=audio 30086 RTP/AVP 0 101 13
>   a=rtpmap:0 PCMU/8000
>   a=rtpmap:101 telephone-event/8000
>   a=fmtp:101 0-16
>   a=rtpmap:13 CN/8000
>   a=ptime:20
>
>
>
>
> Anthony Minessale wrote:
>>
>> you probably have some device lying about ptime everywhere
>> look at a sip trace an pay especially close attention to ptime:x  
>> param in
>> sdp
>> if you don't understand this just attach it here
>>
>> execute the following at the cli
>> sofia profile internal siptrace on
>> sofila loglevel debug
>>
>>
>>
>> On Thu, Oct 8, 2009 at 8:27 AM, Maciej Aniserowicz <
>> maciej.aniserowicz at gmail.com> wrote:
>>
>>>
>>> It's the same on the trunk (the last rev I used was not so old  
>>> anyway).
>>>
>>> Codecs are the same on both legs:
>>> read codec/read rate: PCMU      8000
>>> write codec/write rate: PCMU    8000
>>>
>>> MA
>>>
>>>
>>>
>>>
>>> Michael Jerris wrote:
>>>>
>>>> What codecs are all the call legs using, also, please try current  
>>>> svn
>>>> trunk.
>>>>
>>>> Mike
>>>>
>>>> On Oct 7, 2009, at 3:39 AM, Maciej Aniserowicz wrote:
>>>>
>>>>>
>>>>> Sorry about posting several questions at once, I wasn't aware it's
>>>>> "rude".
>>>>> Let's concentrate on this issue then.
>>>>>
>>>>> I use FS rev 14994. Phones on extensions:
>>>>> 1) x-lite
>>>>> 2) cisco sip phone
>>>>> 3) audio played by fs to the extension being eavesdropped
>>>>>
>>>>> I did not change any codec configuration, I just use the standard
>>>>> one that
>>>>> comes with both FS and the phones.
>>>>> Some time ago someone on FS irc channel told me that this is just
>>>>> how FS
>>>>> eavesdropping works... from your response I understand that this  
>>>>> is
>>>>> not
>>>>> entirely true?
>>>>>
>>>>> Maciej Aniserowicz
>>>>>
>>>>>
>>>>>
>>>>> Anthony Minessale wrote:
>>>>>>
>>>>>> That's is a somewhat vague position.
>>>>>>
>>>>>> You did not mention which version of FreeSWITCH you are  
>>>>>> running, the
>>>>>> phones
>>>>>> being used in your example, your configuration, the codecs in use
>>>>>> etc.
>>>>>>
>>>>>> BTW,
>>>>>> I think you should only ask one question at a time on this list.
>>>>>> The list
>>>>>> is run by volunteers and it's sort of rude to expect 3 or 4  
>>>>>> threads
>>>>>> to be
>>>>>> tended to concerning the same one individual.
>>>>>>
>>>>>>
>>>>>> 2009/10/5 Maciej Aniserowicz <maciej.aniserowicz at gmail.com>
>>>>>>
>>>>>>> Hello,
>>>>>>> When I use eavesdropping in FreeSWITCH, the sound quality is
>>>>>>> really bad.
>>>>>>> Is
>>>>>>> there any way to improve it? Is this a known problem?
>>>>>>> Br/
>>>>>>> Maciej Aniserowicz
>>>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>>> users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> --
>>> View this message in context:
>>> http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3788019.html
>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>>> users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> -- 
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN 
>> %3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL 
>> %3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip 
>> %3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf 
>> %2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>>
>
> -- 
> View this message in context: http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3799274.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From mike at jerris.com  Sun Oct 11 15:52:32 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Oct 2009 18:52:32 -0400
Subject: [Freeswitch-users] mod_socket: custom timeout for HEARTBEAT
	event
In-Reply-To: <86a32abc0910111444j1ab2e446p3cb94b0ae04126e3@mail.gmail.com>
References: <d211f54a0910110959y3d3deed7s43201a3e39756585@mail.gmail.com>
	<86a32abc0910111444j1ab2e446p3cb94b0ae04126e3@mail.gmail.com>
Message-ID: <97B4BDC9-5B11-4A98-B1BF-CC00A82ECBD9@jerris.com>


On Oct 11, 2009, at 5:44 PM, Diego Viola wrote:

> Maybe you want enable_heartbeat or sched_heartbeat from mod_dptools?
>
> You can pass your parameters in second to these two.
>
> Example:
>
> <action application="enable_heartbeat" data="1"/>
> <action application="sched_heartbeat" data="1"/>
>
> Where 1 in this case is the number of heartbeats per seconds.
>

Number of seconds between hearbeats, not hearbeats per second.


> You can use that example on the Dialplan XML but you can also use it  
> on mod_event_socket outbound, etc.




From mike at jerris.com  Sun Oct 11 16:01:02 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 11 Oct 2009 19:01:02 -0400
Subject: [Freeswitch-users] problem compiling esl for use with freepbx v3
In-Reply-To: <4d6f26b0910111436h416121b7s2a4b5c79ffc01db0@mail.gmail.com>
References: <20090823185152.D17845FE@sinclaire.sibble.net>
	<25846572.post@talk.nabble.com>
	<4d6f26b0910111436h416121b7s2a4b5c79ffc01db0@mail.gmail.com>
Message-ID: <0E66A94D-2F5E-421A-B997-A8705715B0AD@jerris.com>

I am still working on the new build system for esl, stay tuned for  
more info soon, it should be in 1.0.5.

Mike

On Oct 11, 2009, at 5:36 PM, Herman Griffin wrote:

> Although probably not the best solution, I figured out a way to make  
> it compile and install:
>
> I removed all of the -Werror instances in PATH_TO_FREESWITCH_SOURCE/ 
> libs/esl/Makefile
>
> If I was a hardcore c/c++ programmer, I'd figure out the real problem.
>
> Herman aka frek818
>
> On Sun, Oct 11, 2009 at 12:12 PM, frek818 <herman.griffin at gmail.com>  
> wrote:
>
> Did anyone find a solution to this problem? I too would like to  
> install the
> esl module for PHP.
>

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From kristian.kielhofner at gmail.com  Sun Oct 11 16:12:41 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Sun, 11 Oct 2009 19:12:41 -0400
Subject: [Freeswitch-users] On the handling of SIP headers
In-Reply-To: <85095FC9-4DC9-414D-A43F-8A7583D66DFF@jerris.com>
References: <2d9149cd0910091210s33fb2bb9oc0c94a64aa87e0be@mail.gmail.com>
	<85095FC9-4DC9-414D-A43F-8A7583D66DFF@jerris.com>
Message-ID: <2d9149cd0910111612w668abce4m7562a8fbda3ba13f@mail.gmail.com>

Mike,

  Thanks for getting back to me.  I agree.

  I'm willing to throw down on a bounty for this.  Any idea how much
work we're talking about here?

On Sun, Oct 11, 2009 at 5:59 PM, Michael Jerris <mike at jerris.com> wrote:
> There is this endless push and pull on this topic, those who want them
> assume it should be default, those who don't assume that should be
> default. ?This probably needs a configuration option defaulting to
> pass them (those who don't want to pass them are usually a bit more
> educated and would find the option better than the other way around).
>
> Mike
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From brian at freeswitch.org  Sun Oct 11 16:39:27 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 11 Oct 2009 18:39:27 -0500
Subject: [Freeswitch-users] SLAs and BLAs
In-Reply-To: <5CC2E91025314615ACBB65E029D5CB12@greyhawk.tonecommander.com>
References: <5B5F50E0B1D34BC0BCBCC05E1FE05C8A@greyhawk.tonecommander.com>
	<3F7AC380-CACB-432F-8B94-80FBA18628C0@freeswitch.org>
	<5CC2E91025314615ACBB65E029D5CB12@greyhawk.tonecommander.com>
Message-ID: <FCD5284F-8F0D-4D69-91E1-B25145F35F13@freeswitch.org>

No we do not issue a subscribe OUTBOUND.  We work with Polycom, Snom  
and a few other that outbound subscribe is something we should do if  
we want to know you took the phone off hook.

/b

On Oct 9, 2009, at 10:23 AM, Jerry Richards wrote:

>
> I gather from the mailing archive that BLAs are implemented using the
> draft-anil-sipping-bla-04.txt document.  According to the draft, the
> Appearance Agent is supposed to initiate a SUBSCRIBE request, but I  
> don't
> see FS doing this.
>
> What phone types/models are known to work with the FS BLA  
> implementation?
>
> Best Regards,
> Jerry




From brian at freeswitch.org  Sun Oct 11 16:43:37 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 11 Oct 2009 18:43:37 -0500
Subject: [Freeswitch-users] SLAs and BLAs
In-Reply-To: <5CC2E91025314615ACBB65E029D5CB12@greyhawk.tonecommander.com>
References: <5B5F50E0B1D34BC0BCBCC05E1FE05C8A@greyhawk.tonecommander.com>
	<3F7AC380-CACB-432F-8B94-80FBA18628C0@freeswitch.org>
	<5CC2E91025314615ACBB65E029D5CB12@greyhawk.tonecommander.com>
Message-ID: <422F970E-7708-4124-AD41-A51D22A3EDEF@freeswitch.org>

btw the polycom it will do the sub like you expect but the rest will  
only know when the phone is actually on a call or receiving a call...  
we won't know if the handset is taken off hook...I would like to  
rework the functionality similar to how sofia_sla.c handles it.    
Again if you want to work on it and patch it then I'm sure we can  
provide you guidance.


/b

On Oct 9, 2009, at 10:23 AM, Jerry Richards wrote:

>
> I gather from the mailing archive that BLAs are implemented using the
> draft-anil-sipping-bla-04.txt document.  According to the draft, the
> Appearance Agent is supposed to initiate a SUBSCRIBE request, but I  
> don't
> see FS doing this.
>
> What phone types/models are known to work with the FS BLA  
> implementation?
>
> Best Regards,
> Jerry




From brian at freeswitch.org  Sun Oct 11 16:44:35 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 11 Oct 2009 18:44:35 -0500
Subject: [Freeswitch-users] On the handling of SIP headers
In-Reply-To: <2d9149cd0910111612w668abce4m7562a8fbda3ba13f@mail.gmail.com>
References: <2d9149cd0910091210s33fb2bb9oc0c94a64aa87e0be@mail.gmail.com>
	<85095FC9-4DC9-414D-A43F-8A7583D66DFF@jerris.com>
	<2d9149cd0910111612w668abce4m7562a8fbda3ba13f@mail.gmail.com>
Message-ID: <62E38F6E-C929-423D-AF45-6FEBB58F6CD3@freeswitch.org>

Well since we aren't a proxy you shouldn't default to passing them  
right?

/b

On Oct 11, 2009, at 6:12 PM, Kristian Kielhofner wrote:

> Mike,
>
>  Thanks for getting back to me.  I agree.
>
>  I'm willing to throw down on a bounty for this.  Any idea how much
> work we're talking about here?




From brian at freeswitch.org  Sun Oct 11 16:49:58 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 11 Oct 2009 18:49:58 -0500
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <86a32abc0910111506m4e6d2740tbd3ac5fe0f59178@mail.gmail.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<87f2f3b90910080958i1f5cfccax621fd12100a05e3e@mail.gmail.com>
	<903da5680910081201p427a8358n768edb61a96674a4@mail.gmail.com>
	<87f2f3b90910081538l6e4e4b7dv3aedb2c4d1af58bc@mail.gmail.com>
	<903da5680910082238x27bff476s8bea0e74ab516a8d@mail.gmail.com>
	<87f2f3b90910082308i7f6144d9kc887596be54dd609@mail.gmail.com>
	<903da5680910082341r485092vb0cd10c43c4a2534@mail.gmail.com>
	<219716B3-C355-4128-A333-1459AC63C364@jerris.com>
	<86a32abc0910111506m4e6d2740tbd3ac5fe0f59178@mail.gmail.com>
Message-ID: <5162D0B7-357C-4D0E-9913-46564A8C0B1D@freeswitch.org>

I have tried to police the wiki when things like this appear.. its one  
thing to crack a joke in fun from time to time... but to put stuff  
like that on the wiki isn't acceptable.

/b

On Oct 11, 2009, at 5:06 PM, Diego Viola wrote:

>
> I know that's just a joke and we might make one or two jokes, but we  
> don't really hate Asterisk and we welcome them, there are some  
> people who might get offended by the jokes (specially Asterisk  
> users) but I have already corrected these from the wiki.
>
> Thanks for the feedback Gabriel.
>
> Diego




From brian at freeswitch.org  Sun Oct 11 16:53:03 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 11 Oct 2009 18:53:03 -0500
Subject: [Freeswitch-users] Re corded file as voicemail.
In-Reply-To: <23f91030910092240u53b0cc51jd5b1670901f20e21@mail.gmail.com>
References: <25830785.post@talk.nabble.com>
	<20091010040539.GB23188@jdc.jasonjgw.net>
	<23f91030910092240u53b0cc51jd5b1670901f20e21@mail.gmail.com>
Message-ID: <89F518C6-B470-482D-8C27-2812BDC20C39@freeswitch.org>

FreeSWITCH can play back stereo files it'll just mux them down to mono  
before playing... can you elaborate on the error you're getting?

/b

On Oct 10, 2009, at 12:40 AM, Seven Du wrote:

> Yes, it's discussed before.
>
> http://wiki.freeswitch.org/wiki/Channel_Variables#RECORD_STEREO
>
> set that var to false before you record.

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From dujinfang at gmail.com  Sun Oct 11 18:45:46 2009
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 12 Oct 2009 09:45:46 +0800
Subject: [Freeswitch-users] Re corded file as voicemail.
In-Reply-To: <89F518C6-B470-482D-8C27-2812BDC20C39@freeswitch.org>
References: <25830785.post@talk.nabble.com>
	<20091010040539.GB23188@jdc.jasonjgw.net>
	<23f91030910092240u53b0cc51jd5b1670901f20e21@mail.gmail.com>
	<89F518C6-B470-482D-8C27-2812BDC20C39@freeswitch.org>
Message-ID: <23f91030910111845yacd54ees40f246ef9152e306@mail.gmail.com>

got

2009-10-12 01:41:30.349961 [WARNING] switch_core_file.c:133 File has 2
channels, muxing to mono will occur.
2009-10-12 01:41:30.349961 [ERR] switch_core_codec.c:431 Stereo is currently
unsupported. please downsample audio source to mono.

freeswitch at internal>
freeswitch at internal> version
FreeSWITCH Version 1.0.trunk (14696)


originate {ignore_early_media=true,RECORD_STEREO=true}sofia/gateway/xx/xx
&bridge(sofia/gateway/yy/yy)

uuid_record uuid start /tmp/a.wav



play by sox:

Input File     : 'a.wav'
Sample Size    : 16-bit (2 bytes)
Sample Encoding: signed (2's complement)
Channels       : 2
Sample Rate    : 8000


full logs here:

freeswitch at internal> originate
{ignore_early_media=true}sofia/gateway/skype/seven1240 &playback(/tmp/a.wav)
+OK dcbd971e-c570-481d-843a-0c5582669c69

2009-10-12 01:41:26.89775 [DEBUG] switch_ivr_originate.c:1043 variable
string 0 = [ignore_early_media=true]
freeswitch at internal> 2009-10-12 01:41:26.89775 [NOTICE] switch_channel.c:602
New Channel sofia/eqenglish/seven1240 [dcbd971e-c570-481d-843a-0c5582669c69]
2009-10-12 01:41:26.89775 [DEBUG] mod_sofia.c:2867
(sofia/eqenglish/seven1240) State Change CS_NEW -> CS_INIT
2009-10-12 01:41:26.89775 [DEBUG] switch_core_session.c:932 Send signal
sofia/eqenglish/seven1240 [BREAK]
2009-10-12 01:41:26.89775 [DEBUG] switch_core_state_machine.c:398
(sofia/eqenglish/seven1240) Running State Change CS_INIT
2009-10-12 01:41:26.89775 [DEBUG] switch_core_state_machine.c:481
(sofia/eqenglish/seven1240) State INIT
2009-10-12 01:41:26.89775 [DEBUG] mod_sofia.c:83 sofia/eqenglish/seven1240
SOFIA INIT
2009-10-12 01:41:26.89775 [DEBUG] mod_sofia.c:111
(sofia/eqenglish/seven1240) State Change CS_INIT -> CS_ROUTING
2009-10-12 01:41:26.89775 [DEBUG] switch_core_session.c:932 Send signal
sofia/eqenglish/seven1240 [BREAK]
2009-10-12 01:41:26.89775 [DEBUG] switch_core_state_machine.c:481
(sofia/eqenglish/seven1240) State INIT going to sleep
2009-10-12 01:41:26.89775 [DEBUG] switch_core_state_machine.c:398
(sofia/eqenglish/seven1240) Running State Change CS_ROUTING
2009-10-12 01:41:26.89775 [DEBUG] switch_core_state_machine.c:484
(sofia/eqenglish/seven1240) State ROUTING
2009-10-12 01:41:26.89775 [DEBUG] mod_sofia.c:130 sofia/eqenglish/seven1240
SOFIA ROUTING
2009-10-12 01:41:26.89775 [DEBUG] switch_ivr_originate.c:63
(sofia/eqenglish/seven1240) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2009-10-12 01:41:26.89775 [DEBUG] switch_core_session.c:932 Send signal
sofia/eqenglish/seven1240 [BREAK]
2009-10-12 01:41:26.89775 [DEBUG] switch_core_state_machine.c:484
(sofia/eqenglish/seven1240) State ROUTING going to sleep
2009-10-12 01:41:26.89775 [DEBUG] switch_core_state_machine.c:398
(sofia/eqenglish/seven1240) Running State Change CS_CONSUME_MEDIA
2009-10-12 01:41:26.89775 [DEBUG] switch_core_state_machine.c:503
(sofia/eqenglish/seven1240) State CONSUME_MEDIA
2009-10-12 01:41:26.89775 [DEBUG] sofia.c:3302 Channel
sofia/eqenglish/seven1240 entering state [calling][0]
2009-10-12 01:41:26.169828 [DEBUG] sofia.c:3302 Channel
sofia/eqenglish/seven1240 entering state [calling][0]
2009-10-12 01:41:28.700782 [DEBUG] sofia.c:3302 Channel
sofia/eqenglish/seven1240 entering state [proceeding][180]
2009-10-12 01:41:28.700782 [NOTICE] sofia.c:3366 Ring-Ready
sofia/eqenglish/seven1240!
2009-10-12 01:41:30.330095 [DEBUG] sofia.c:3302 Channel
sofia/eqenglish/seven1240 entering state [ready][200]
2009-10-12 01:41:30.330095 [DEBUG] sofia.c:3309 Remote SDP:
v=0
o=FreeSWITCH 1255290734 1255290735 IN IP4 67.228.224.149
s=FreeSWITCH
c=IN IP4 127.0.0.1
t=0 0
m=audio 20956 RTP/AVP 0 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20

2009-10-12 01:41:30.330095 [DEBUG] sofia_glue.c:3132 Audio Codec Compare
[PCMU:0:8000:20]/[PCMU:0:8000:20]
2009-10-12 01:41:30.330095 [DEBUG] sofia_glue.c:2090 Set Codec
sofia/eqenglish/seven1240 PCMU/8000 20 ms 160 samples
2009-10-12 01:41:30.330095 [DEBUG] sofia_glue.c:3092 Set 2833 dtmf payload
to 101
2009-10-12 01:41:30.330095 [DEBUG] sofia_glue.c:2324 AUDIO RTP
[sofia/eqenglish/seven1240] 67.228.224.149 port 24862 -> 127.0.0.1 port
20956 codec: 0 ms: 20
2009-10-12 01:41:30.330095 [DEBUG] switch_rtp.c:1139 Starting timer [soft]
160 bytes per 20ms
2009-10-12 01:41:30.339941 [DEBUG] sofia_glue.c:2503 Set comfort noise
payload to 13
2009-10-12 01:41:30.339941 [NOTICE] sofia.c:3807 Channel
[sofia/eqenglish/seven1240] has been answered
2009-10-12 01:41:30.339941 [DEBUG] switch_channel.c:182
sofia/eqenglish/seven1240 receive message [AUDIO_SYNC]
2009-10-12 01:41:30.339941 [NOTICE] switch_cpp.cpp:1130 ==DebugVar==
gateway_name: skype
2009-10-12 01:41:30.349961 [DEBUG] switch_ivr_originate.c:2089 Originate
Resulted in Success: [sofia/eqenglish/seven1240]
2009-10-12 01:41:30.349961 [DEBUG] switch_channel.c:182
sofia/eqenglish/seven1240 receive message [AUDIO_SYNC]
2009-10-12 01:41:30.349961 [DEBUG] mod_commands.c:2334
(sofia/eqenglish/seven1240) State Change CS_CONSUME_MEDIA -> CS_EXECUTE
2009-10-12 01:41:30.349961 [DEBUG] switch_core_session.c:932 Send signal
sofia/eqenglish/seven1240 [BREAK]
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:503
(sofia/eqenglish/seven1240) State CONSUME_MEDIA going to sleep
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:398
(sofia/eqenglish/seven1240) Running State Change CS_EXECUTE
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:491
(sofia/eqenglish/seven1240) State EXECUTE
2009-10-12 01:41:30.349961 [DEBUG] mod_sofia.c:173 sofia/eqenglish/seven1240
SOFIA EXECUTE
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:151
sofia/eqenglish/seven1240 Standard EXECUTE
EXECUTE sofia/eqenglish/seven1240 playback(/tmp/a.wav)
2009-10-12 01:41:30.349961 [WARNING] switch_core_file.c:133 File has 2
channels, muxing to mono will occur.
2009-10-12 01:41:30.349961 [ERR] switch_core_codec.c:431 Stereo is currently
unsupported. please downsample audio source to mono.
2009-10-12 01:41:30.349961 [DEBUG] switch_ivr_play_say.c:1113 Raw Codec
Activation Failed L16 at 8000hz 2 channels 20ms
2009-10-12 01:41:30.349961 [NOTICE] switch_core_state_machine.c:179 Hangup
sofia/eqenglish/seven1240 [CS_EXECUTE] [NORMAL_CLEARING]
2009-10-12 01:41:30.349961 [DEBUG] switch_channel.c:1715 Send signal
sofia/eqenglish/seven1240 [KILL]
2009-10-12 01:41:30.349961 [DEBUG] switch_core_session.c:932 Send signal
sofia/eqenglish/seven1240 [BREAK]
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:491
(sofia/eqenglish/seven1240) State EXECUTE going to sleep
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:398
(sofia/eqenglish/seven1240) Running State Change CS_HANGUP
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:434
(sofia/eqenglish/seven1240) State HANGUP
2009-10-12 01:41:30.349961 [DEBUG] mod_sofia.c:338 Channel
sofia/eqenglish/seven1240 hanging up, cause: NORMAL_CLEARING
2009-10-12 01:41:30.349961 [DEBUG] mod_sofia.c:376 Sending BYE to
sofia/eqenglish/seven1240
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:46
sofia/eqenglish/seven1240 Standard HANGUP, cause: NORMAL_CLEARING
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:434
(sofia/eqenglish/seven1240) State HANGUP going to sleep
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:476
(sofia/eqenglish/seven1240) State Change CS_HANGUP -> CS_REPORTING
2009-10-12 01:41:30.349961 [DEBUG] switch_core_session.c:932 Send signal
sofia/eqenglish/seven1240 [BREAK]
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:398
(sofia/eqenglish/seven1240) Running State Change CS_REPORTING
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:613
(sofia/eqenglish/seven1240) State REPORTING
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:53
sofia/eqenglish/seven1240 Standard REPORTING, cause: NORMAL_CLEARING
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:613
(sofia/eqenglish/seven1240) State REPORTING going to sleep
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:411
(sofia/eqenglish/seven1240) State Change CS_REPORTING -> CS_DESTROY
2009-10-12 01:41:30.349961 [DEBUG] switch_core_session.c:932 Send signal
sofia/eqenglish/seven1240 [BREAK]
2009-10-12 01:41:30.349961 [DEBUG] switch_core_session.c:1068 Session 2178
(sofia/eqenglish/seven1240) Locked, Waiting on external entities
2009-10-12 01:41:30.349961 [NOTICE] switch_core_session.c:1086 Session 2178
(sofia/eqenglish/seven1240) Ended
2009-10-12 01:41:30.349961 [NOTICE] switch_core_session.c:1088 Close Channel
sofia/eqenglish/seven1240 [CS_DESTROY]
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:556
(sofia/eqenglish/seven1240) Running State Change CS_DESTROY
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:565
(sofia/eqenglish/seven1240) State DESTROY
2009-10-12 01:41:30.349961 [DEBUG] mod_sofia.c:255 sofia/eqenglish/seven1240
SOFIA DESTROY
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:60
sofia/eqenglish/seven1240 Standard DESTROY
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:565
(sofia/eqenglish/seven1240) State DESTROY going to sleep
2009-10-12 01:41:30.349961 [NOTICE] switch_cpp.cpp:1130 ==DebugVar==
gateway_name: skype


2009/10/12 Brian West <brian at freeswitch.org>

> FreeSWITCH can play back stereo files it'll just mux them down to mono
> before playing... can you elaborate on the error you're getting?
> /b
>
> On Oct 10, 2009, at 12:40 AM, Seven Du wrote:
>
> Yes, it's discussed before.
>
> http://wiki.freeswitch.org/wiki/Channel_Variables#RECORD_STEREO
>
> set that var to false before you record.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jason at jasonjgw.net  Sun Oct 11 19:07:16 2009
From: jason at jasonjgw.net (Jason White)
Date: Mon, 12 Oct 2009 13:07:16 +1100
Subject: [Freeswitch-users] Re corded file as voicemail.
In-Reply-To: <23f91030910111845yacd54ees40f246ef9152e306@mail.gmail.com>
References: <25830785.post@talk.nabble.com>
	<20091010040539.GB23188@jdc.jasonjgw.net>
	<23f91030910092240u53b0cc51jd5b1670901f20e21@mail.gmail.com>
	<89F518C6-B470-482D-8C27-2812BDC20C39@freeswitch.org>
	<23f91030910111845yacd54ees40f246ef9152e306@mail.gmail.com>
Message-ID: <20091012020716.GA8518@jdc.jasonjgw.net>

Seven Du <dujinfang at gmail.com> wrote:
> originate {ignore_early_media=true,RECORD_STEREO=true}sofia/gateway/xx/xx
> &bridge(sofia/gateway/yy/yy)

Shouldn't that be record_stereo=false for mono recording?




From kristian.kielhofner at gmail.com  Sun Oct 11 19:17:08 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Sun, 11 Oct 2009 22:17:08 -0400
Subject: [Freeswitch-users] On the handling of SIP headers
In-Reply-To: <62E38F6E-C929-423D-AF45-6FEBB58F6CD3@freeswitch.org>
References: <2d9149cd0910091210s33fb2bb9oc0c94a64aa87e0be@mail.gmail.com>
	<85095FC9-4DC9-414D-A43F-8A7583D66DFF@jerris.com>
	<2d9149cd0910111612w668abce4m7562a8fbda3ba13f@mail.gmail.com>
	<62E38F6E-C929-423D-AF45-6FEBB58F6CD3@freeswitch.org>
Message-ID: <2d9149cd0910111917n53b59be5m1ab839fc959c7dbd@mail.gmail.com>

Brian,

  You are correct, they *probably* shouldn't be passed by default.

On Sun, Oct 11, 2009 at 7:44 PM, Brian West <brian at freeswitch.org> wrote:
> Well since we aren't a proxy you shouldn't default to passing them
> right?
>
> /b
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From dujinfang at gmail.com  Sun Oct 11 19:27:11 2009
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 12 Oct 2009 10:27:11 +0800
Subject: [Freeswitch-users] Re corded file as voicemail.
In-Reply-To: <20091012020716.GA8518@jdc.jasonjgw.net>
References: <25830785.post@talk.nabble.com>
	<20091010040539.GB23188@jdc.jasonjgw.net>
	<23f91030910092240u53b0cc51jd5b1670901f20e21@mail.gmail.com>
	<89F518C6-B470-482D-8C27-2812BDC20C39@freeswitch.org>
	<23f91030910111845yacd54ees40f246ef9152e306@mail.gmail.com>
	<20091012020716.GA8518@jdc.jasonjgw.net>
Message-ID: <23f91030910111927v704b2eb9tfd9558068b063153@mail.gmail.com>

I set to true because brian said it can play stereo files but no lucky for
me.

2009/10/12 Jason White <jason at jasonjgw.net>

> Seven Du <dujinfang at gmail.com> wrote:
> > originate {ignore_early_media=true,RECORD_STEREO=true}sofia/gateway/xx/xx
> > &bridge(sofia/gateway/yy/yy)
>
> Shouldn't that be record_stereo=false for mono recording?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sun Oct 11 19:52:59 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 11 Oct 2009 21:52:59 -0500
Subject: [Freeswitch-users] Re corded file as voicemail.
In-Reply-To: <23f91030910111927v704b2eb9tfd9558068b063153@mail.gmail.com>
References: <25830785.post@talk.nabble.com>
	<20091010040539.GB23188@jdc.jasonjgw.net>
	<23f91030910092240u53b0cc51jd5b1670901f20e21@mail.gmail.com>
	<89F518C6-B470-482D-8C27-2812BDC20C39@freeswitch.org>
	<23f91030910111845yacd54ees40f246ef9152e306@mail.gmail.com>
	<20091012020716.GA8518@jdc.jasonjgw.net>
	<23f91030910111927v704b2eb9tfd9558068b063153@mail.gmail.com>
Message-ID: <5F170ABD-E654-4511-889B-5893FC33C03E@freeswitch.org>

Where are you playing the files?

/b

On Oct 11, 2009, at 9:27 PM, Seven Du wrote:

> I set to true because brian said it can play stereo files but no  
> lucky for me.




From brian at freeswitch.org  Sun Oct 11 20:01:23 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 11 Oct 2009 22:01:23 -0500
Subject: [Freeswitch-users] Re corded file as voicemail.
In-Reply-To: <23f91030910111927v704b2eb9tfd9558068b063153@mail.gmail.com>
References: <25830785.post@talk.nabble.com>
	<20091010040539.GB23188@jdc.jasonjgw.net>
	<23f91030910092240u53b0cc51jd5b1670901f20e21@mail.gmail.com>
	<89F518C6-B470-482D-8C27-2812BDC20C39@freeswitch.org>
	<23f91030910111845yacd54ees40f246ef9152e306@mail.gmail.com>
	<20091012020716.GA8518@jdc.jasonjgw.net>
	<23f91030910111927v704b2eb9tfd9558068b063153@mail.gmail.com>
Message-ID: <32DFC62F-D799-4C23-8F99-580CA6DBF6DD@freeswitch.org>

Please open a jira please this did work but a recent change in  
switch_core_codec caused this to appear I usually test this regularly  
but haven't run thru a full run of tests lately.

/b

On Oct 11, 2009, at 9:27 PM, Seven Du wrote:

> I set to true because brian said it can play stereo files but no  
> lucky for me.




From brian at freeswitch.org  Sun Oct 11 20:37:40 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 11 Oct 2009 22:37:40 -0500
Subject: [Freeswitch-users] Re corded file as voicemail.
In-Reply-To: <23f91030910111927v704b2eb9tfd9558068b063153@mail.gmail.com>
References: <25830785.post@talk.nabble.com>
	<20091010040539.GB23188@jdc.jasonjgw.net>
	<23f91030910092240u53b0cc51jd5b1670901f20e21@mail.gmail.com>
	<89F518C6-B470-482D-8C27-2812BDC20C39@freeswitch.org>
	<23f91030910111845yacd54ees40f246ef9152e306@mail.gmail.com>
	<20091012020716.GA8518@jdc.jasonjgw.net>
	<23f91030910111927v704b2eb9tfd9558068b063153@mail.gmail.com>
Message-ID: <D5F3C41A-36B9-4172-B719-8BA824EBFC95@freeswitch.org>

not sure I said this but open a jira... its a bug thats recent...  
usually it will mux them to mono and the codec engine is trying to  
open L16 with two channels so something has changed to cause this  
regression.

Expect a fix sometime tomorrow.

/b

On Oct 11, 2009, at 9:27 PM, Seven Du wrote:

> I set to true because brian said it can play stereo files but no  
> lucky for me.
>
> 2009/10/12 Jason White <jason at jasonjgw.net>
> Seven Du <dujinfang at gmail.com> wrote:
> > originate {ignore_early_media=true,RECORD_STEREO=true}sofia/ 
> gateway/xx/xx
> > &bridge(sofia/gateway/yy/yy)
>
> Shouldn't that be record_stereo=false for mono recording?
>

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From nagalenoj at gmail.com  Sun Oct 11 20:44:02 2009
From: nagalenoj at gmail.com (Nagalenoj)
Date: Sun, 11 Oct 2009 20:44:02 -0700 (PDT)
Subject: [Freeswitch-users] Re  corded file as voicemail.
In-Reply-To: <32DFC62F-D799-4C23-8F99-580CA6DBF6DD@freeswitch.org>
References: <25751158.post@talk.nabble.com>
	<BC536E8C-6813-4A69-9B80-EA11E41A79FC@jerris.com>
	<25830785.post@talk.nabble.com>
	<20091010040539.GB23188@jdc.jasonjgw.net>
	<23f91030910092240u53b0cc51jd5b1670901f20e21@mail.gmail.com>
	<89F518C6-B470-482D-8C27-2812BDC20C39@freeswitch.org>
	<23f91030910111845yacd54ees40f246ef9152e306@mail.gmail.com>
	<20091012020716.GA8518@jdc.jasonjgw.net>
	<23f91030910111927v704b2eb9tfd9558068b063153@mail.gmail.com>
	<32DFC62F-D799-4C23-8F99-580CA6DBF6DD@freeswitch.org>
Message-ID: <25850181.post@talk.nabble.com>


Is it possible to play a stereo file ?!!! whats the conclusion.?!


Brian West-3 wrote:
> 
> Please open a jira please this did work but a recent change in  
> switch_core_codec caused this to appear I usually test this regularly  
> but haven't run thru a full run of tests lately.
> /b
> 
> On Oct 11, 2009, at 9:27 PM, Seven Du wrote:
> 
>> I set to true because brian said it can play stereo files but no  
>> lucky for me.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/Recorded-file-as-voicemail.-tp25751158p25850181.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From nagalenoj at gmail.com  Sun Oct 11 20:46:02 2009
From: nagalenoj at gmail.com (Nagalenoj)
Date: Sun, 11 Oct 2009 20:46:02 -0700 (PDT)
Subject: [Freeswitch-users] Re  corded file as voicemail.
Message-ID: <25850181.post@talk.nabble.com>


Whats the conclusion.?! 


Brian West-3 wrote:
> 
> Please open a jira please this did work but a recent change in  
> switch_core_codec caused this to appear I usually test this regularly  
> but haven't run thru a full run of tests lately.
> /b
> 
> On Oct 11, 2009, at 9:27 PM, Seven Du wrote:
> 
>> I set to true because brian said it can play stereo files but no  
>> lucky for me.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/Recorded-file-as-voicemail.-tp25751158p25850181.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From brian at freeswitch.org  Sun Oct 11 21:11:43 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 11 Oct 2009 23:11:43 -0500
Subject: [Freeswitch-users] Re  corded file as voicemail.
In-Reply-To: <25850181.post@talk.nabble.com>
References: <25850181.post@talk.nabble.com>
Message-ID: <BEDBBF51-2D48-4CDE-A467-11D5BBB458E1@freeswitch.org>

It was possible but we have a regression in the code that isn't  
letting that happen right now... hence the reason i said Open a jira  
so we could fix it.

IS THAT not clear?

/b

On Oct 11, 2009, at 10:46 PM, Nagalenoj wrote:

>
> Whats the conclusion.?!




From srinivas.ksvreddy at gmail.com  Sun Oct 11 21:14:36 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Mon, 12 Oct 2009 09:44:36 +0530
Subject: [Freeswitch-users] mod_sofia.c registered calls how to know
In-Reply-To: <14A1216A-61C3-4C03-B710-57D2E7F4CD7A@jerris.com>
References: <f8af5740910070748g1efee71ft5841b6fa4d2412c8@mail.gmail.com>
	<14A1216A-61C3-4C03-B710-57D2E7F4CD7A@jerris.com>
Message-ID: <f8af5740910112114k3206c3cbq5757e2ff7de81162@mail.gmail.com>

Hi Mike,

Thanks for your valuable reply,
when i install freeswitch1.0.2 in my machine(Windows xp operation system) i
dont have any databasae installed in my system, then from sqllite will come
into picture, and how can i see the registered users data from sqllite.


Thanks
Srinivas

On Mon, Oct 12, 2009 at 3:04 AM, Michael Jerris <mike at jerris.com> wrote:

>
> On Oct 7, 2009, at 10:48 AM, srinivasula reddy wrote:
>
>
> Hi
>
> can any please tell me where registered calls are stored, so when incoming
> call came to mod_sofia.c how it will check it is registered or not?\\
>
>
> Calls are not registered and calls have nothing to do with registration.
>  Users are registered so that you may send calls to them.  Registration data
> is stored either in a sqlite database, or optionally if you setup odbc, in
> another database of your choice.  If you try to send a call to an
> unregistered user in the dialplan using the proper syntax to send calls to
> registered users (see the wiki for more details), and that user is not
> registered, the bridge app will fail, optionally letting you continue on in
> the dialplan based on variables such as continue_on_fail and
> hangup_after_bridge.  You can use the sofia_contact function to see if there
> is anyone registered to a specific user.
>
> Mike
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Srinivasula Reddy K
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From shiyanov at gmail.com  Sun Oct 11 22:38:08 2009
From: shiyanov at gmail.com (Artem Shiyanov)
Date: Mon, 12 Oct 2009 09:38:08 +0400
Subject: [Freeswitch-users] mod_socket: custom timeout for HEARTBEAT
	event
In-Reply-To: <97B4BDC9-5B11-4A98-B1BF-CC00A82ECBD9@jerris.com>
References: <d211f54a0910110959y3d3deed7s43201a3e39756585@mail.gmail.com>
	<86a32abc0910111444j1ab2e446p3cb94b0ae04126e3@mail.gmail.com>
	<97B4BDC9-5B11-4A98-B1BF-CC00A82ECBD9@jerris.com>
Message-ID: <d211f54a0910112238i5f69275ey7cad437e47e6b567@mail.gmail.com>

Michael, Diego,
thanks for the rapid answers!

As far as I understand, "enable_heartbeat" app is launching
SESSION_HEARTBEAT events that will stop when the call will be cleared. Also
I "heard" that "enable_heartbeat" works only for calls with proxied media.

What I want is to monitor FreeSwitch status: is it alive and what is the
system status message. This info is provided in HEARTBEAT event gracefully
but in constant time period = 20 sec. So the main question is- how to
customize this period?


Artem



On Mon, Oct 12, 2009 at 2:52 AM, Michael Jerris <mike at jerris.com> wrote:

>
> On Oct 11, 2009, at 5:44 PM, Diego Viola wrote:
>
> > Maybe you want enable_heartbeat or sched_heartbeat from mod_dptools?
> >
> > You can pass your parameters in second to these two.
> >
> > Example:
> >
> > <action application="enable_heartbeat" data="1"/>
> > <action application="sched_heartbeat" data="1"/>
> >
> > Where 1 in this case is the number of heartbeats per seconds.
> >
>
> Number of seconds between hearbeats, not hearbeats per second.
>
>
> > You can use that example on the Dialplan XML but you can also use it
> > on mod_event_socket outbound, etc.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mattdfong at gmail.com  Sun Oct 11 23:11:32 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Mon, 12 Oct 2009 13:11:32 +0700
Subject: [Freeswitch-users] Sending an Event to a Session for onInput
In-Reply-To: <ACA9A602-0925-4D6F-8B11-0B214A462500@jerris.com>
References: <4256bf830910091234t11f4f4d0w28283f341bd1fb06@mail.gmail.com>
	<ACA9A602-0925-4D6F-8B11-0B214A462500@jerris.com>
Message-ID: <4256bf830910112311t409c45a3q2f26cef3464e49e7@mail.gmail.com>

Hi Mike,
I'm just trying to send it an event with some custom event headers, just so
an external program can communicate with a session without having to
transfer the session to a different program.  I'm curious what uuid_display
does...the wiki only gives a brief description and my Google'ing could not
find any examples. Thanks for the help.

--matt
http://www.hellohunter.com

On Mon, Oct 12, 2009 at 5:04 AM, Michael Jerris <mike at jerris.com> wrote:

> We don't have session messages directly exposed, except for things
> like display, respond, and deflect.  What specifically are you trying
> to send ?
>
> Mike
>
> On Oct 9, 2009, at 3:34 PM, Matthew Fong wrote:
>
> > I'm used to using the onInput callbacks inside lua and javascript to
> > listen for dtmf and other events and perform a task accordingly. I'm
> > wondering if there is a way to send an event to a session or channel
> > that can be caught using the setInputCallback inside lua from
> > outside the session program. Maybe an API command that can generate
> > an event for a specific UUID. Does a mechanism exist to do this that
> > I'm over looking? Thanks.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dujinfang at gmail.com  Sun Oct 11 23:15:30 2009
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 12 Oct 2009 14:15:30 +0800
Subject: [Freeswitch-users] Re corded file as voicemail.
In-Reply-To: <BEDBBF51-2D48-4CDE-A467-11D5BBB458E1@freeswitch.org>
References: <25850181.post@talk.nabble.com>
	<BEDBBF51-2D48-4CDE-A467-11D5BBB458E1@freeswitch.org>
Message-ID: <23f91030910112315h5ea79b81ue55381a3caff0173@mail.gmail.com>

http://jira.freeswitch.org/browse/MODCODEC-15

Is it ok I assigned to you ?

Thanks.

2009/10/12 Brian West <brian at freeswitch.org>

> It was possible but we have a regression in the code that isn't
> letting that happen right now... hence the reason i said Open a jira
> so we could fix it.
>
> IS THAT not clear?
>
> /b
>
> On Oct 11, 2009, at 10:46 PM, Nagalenoj wrote:
>
> >
> > Whats the conclusion.?!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From diego.viola at gmail.com  Sun Oct 11 23:52:07 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Mon, 12 Oct 2009 06:52:07 +0000
Subject: [Freeswitch-users] mod_sofia.c registered calls how to know
In-Reply-To: <f8af5740910112114k3206c3cbq5757e2ff7de81162@mail.gmail.com>
References: <f8af5740910070748g1efee71ft5841b6fa4d2412c8@mail.gmail.com>
	<14A1216A-61C3-4C03-B710-57D2E7F4CD7A@jerris.com>
	<f8af5740910112114k3206c3cbq5757e2ff7de81162@mail.gmail.com>
Message-ID: <86a32abc0910112352q2e5cfac2u23a87f9f92a5431b@mail.gmail.com>

FreeSWITCH 1.0.2?

That's more than a year old I think, you should really update to 1.0.4 or
latest SVN trunk.

Diego

On Mon, Oct 12, 2009 at 4:14 AM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

> Hi Mike,
>
> Thanks for your valuable reply,
> when i install freeswitch1.0.2 in my machine(Windows xp operation system) i
> dont have any databasae installed in my system, then from sqllite will come
> into picture, and how can i see the registered users data from sqllite.
>
>
> Thanks
> Srinivas
>
> On Mon, Oct 12, 2009 at 3:04 AM, Michael Jerris <mike at jerris.com> wrote:
>
>>
>> On Oct 7, 2009, at 10:48 AM, srinivasula reddy wrote:
>>
>>
>> Hi
>>
>> can any please tell me where registered calls are stored, so when incoming
>> call came to mod_sofia.c how it will check it is registered or not?\\
>>
>>
>> Calls are not registered and calls have nothing to do with registration.
>>  Users are registered so that you may send calls to them.  Registration data
>> is stored either in a sqlite database, or optionally if you setup odbc, in
>> another database of your choice.  If you try to send a call to an
>> unregistered user in the dialplan using the proper syntax to send calls to
>> registered users (see the wiki for more details), and that user is not
>> registered, the bridge app will fail, optionally letting you continue on in
>> the dialplan based on variables such as continue_on_fail and
>> hangup_after_bridge.  You can use the sofia_contact function to see if there
>> is anyone registered to a specific user.
>>
>> Mike
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Srinivasula Reddy K
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From velu.technical at gmail.com  Mon Oct 12 00:36:06 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Mon, 12 Oct 2009 13:06:06 +0530
Subject: [Freeswitch-users] Play music on hold after parking the call
In-Reply-To: <1452e2980910100237u317d6138w1d13bd3cfb250353@mail.gmail.com>
References: <1452e2980910100237u317d6138w1d13bd3cfb250353@mail.gmail.com>
Message-ID: <1452e2980910120036k5ab2fb36vbec031c0b0280116@mail.gmail.com>

 Any one please help me to solve the mentioned problem........

On Sat, Oct 10, 2009 at 3:07 PM, velusamy velu <velu.technical at gmail.com>wrote:

> Dear All,
>       I am using ESL.pm module to control the dial plan application. I want
> to play some music while executing the some external scripts. I executed
> park after then I executed the playback the music didn't play.
>
>     Could any one please explain how can I solve this problem without using
> async mode in socket application?
>
> Thanks,
> Velusamy.
>
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From maciej.aniserowicz at gmail.com  Mon Oct 12 01:47:36 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Mon, 12 Oct 2009 01:47:36 -0700 (PDT)
Subject: [Freeswitch-users] Bad sound quality while eavesdropping
In-Reply-To: <CF8C1791-24CD-4132-83B5-7F83B5BDCCC3@jerris.com>
References: <41A44DD027064988A914974405788C2E@procent>
	<191c3a030910050731m2d74979ep4598e5a1945d58ae@mail.gmail.com>
	<1254901192035-3780245.post@n2.nabble.com>
	<8437F5BC-7AFF-4A74-B8CD-C5B8219021F6@jerris.com>
	<1255008427639-3788019.post@n2.nabble.com>
	<191c3a030910080823g79c7c596x1cd887e1538ce2e1@mail.gmail.com>
	<1255169044209-3799274.post@n2.nabble.com>
	<CF8C1791-24CD-4132-83B5-7F83B5BDCCC3@jerris.com>
Message-ID: <1255337256919-3806786.post@n2.nabble.com>


Yes, I confirmed that with Wireshark (filter "rtp and ip.src == <device ip>). RTP packets are sent every 20ms.

MAniserowicz

  ----- Original Message ----- 
  From: Michael Jerris (via Nabble) 
  To: Maciej Aniserowicz 
  Sent: Monday, October 12, 2009 12:21 AM
  Subject: Re: [Freeswitch-users] Bad sound quality while eavesdropping


  can you confirm from an rtp packet trace that they are all really   
  sending 20ms? 

  Mike 

  On Oct 10, 2009, at 6:04 AM, Maciej Aniserowicz wrote: 


  > 
  > Hi, 
  > Here are the messages with a:ptime parameter. All the calls are   
  > started by 
  > commands sent through socket. 
  > I'm not sure if this is all information you need, please let me know   
  > if 
  > something is missing here and I'll post that. 
  > 
  > 1) starting connection with x-lite (number 2003, the eavesdropper): 
  > 
  >   INVITE sip:[hidden email]:60188;rinstance=80b8f8d92af87cd2 SIP/ 
  > 2.0 
  >   Via: SIP/2.0/UDP   
  > 192.168.3.159:15060;rport;branch=z9hG4bKD9meHQN5XQ88K 
  >   Max-Forwards: 69 
  >   From: "MyApp" <sip:[hidden email]>;tag=jpQ6D7D2jUXvF 
  >   To: <sip:[hidden email]:60188;rinstance=80b8f8d92af87cd2> 
  >   Call-ID: 663c4500-3024-122d-57a2-9be8d85a4cff 
  >   CSeq: 121465610 INVITE 
  >   Contact: <sip:[hidden email]:15060> 
  >   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN 
  >   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, 
  > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH 
  >   Supported: timer, precondition, path, replaces 
  >   Allow-Events: talk, presence, dialog, call-info, sla, 
  > include-session-description, presence.winfo, message-summary, refer 
  >   Content-Type: application/sdp 
  >   Content-Disposition: session 
  >   Content-Length: 447 
  >   Remote-Party-ID: "MyApp" 
  > <sip:[hidden email]>;party=calling;screen=yes;privacy=off 
  > 
  >   v=0 
  >   o=FreeSWITCH 2223565947735016740 3096553520713245589 IN IP4   
  > 192.168.3.159 
  >   s=FreeSWITCH 
  >   c=IN IP4 192.168.3.159 
  >   t=0 0 
  >   m=audio 29966 RTP/AVP 0 115 107 9 8 3 101 13 
  >   a=rtpmap:0 PCMU/8000 
  >   a=rtpmap:115 G7221/32000 
  >   a=fmtp:115 bitrate=48000 
  >   a=rtpmap:107 G7221/16000 
  >   a=fmtp:107 bitrate=32000 
  >   a=rtpmap:9 G722/8000 
  >   a=rtpmap:8 PCMA/8000 
  >   a=rtpmap:3 GSM/8000 
  >   a=rtpmap:101 telephone-event/8000 
  >   a=fmtp:101 0-16 
  >   a=rtpmap:13 CN/8000 
  >   a=ptime:20 
  > 
  > 
  > 2) starting connection with cisco ip phone (number 2006, first leg of 
  > eavesdropped session): 
  > 
  >   INVITE sip:[hidden email]:5060;user=phone SIP/2.0 
  >   Via: SIP/2.0/UDP   
  > 192.168.3.159:15060;rport;branch=z9hG4bKg40rp87FNjB1p 
  >   Max-Forwards: 69 
  >   From: "MyApp" <sip:[hidden email]>;tag=Q3N2pe2K47ctS 
  >   To: <sip:[hidden email]:5060;user=phone> 
  >   Call-ID: 6e6fd8a0-3024-122d-57a2-9be8d85a4cff 
  >   CSeq: 121465616 INVITE 
  >   Contact: <sip:[hidden email]:15060> 
  >   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN 
  >   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, 
  > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH 
  >   Supported: timer, precondition, path, replaces 
  >   Allow-Events: talk, presence, dialog, call-info, sla, 
  > include-session-description, presence.winfo, message-summary, refer 
  >   Content-Type: application/sdp 
  >   Content-Disposition: session 
  >   Content-Length: 447 
  >   Remote-Party-ID: "MyApp" 
  > <sip:[hidden email]>;party=calling;screen=yes;privacy=off 
  > 
  >   v=0 
  >   o=FreeSWITCH 8453497903781974949 2294419114567885490 IN IP4   
  > 192.168.3.159 
  >   s=FreeSWITCH 
  >   c=IN IP4 192.168.3.159 
  >   t=0 0 
  >   m=audio 17670 RTP/AVP 0 115 107 9 8 3 101 13 
  >   a=rtpmap:0 PCMU/8000 
  >   a=rtpmap:115 G7221/32000 
  >   a=fmtp:115 bitrate=48000 
  >   a=rtpmap:107 G7221/16000 
  >   a=fmtp:107 bitrate=32000 
  >   a=rtpmap:9 G722/8000 
  >   a=rtpmap:8 PCMA/8000 
  >   a=rtpmap:3 GSM/8000 
  >   a=rtpmap:101 telephone-event/8000 
  >   a=fmtp:101 0-16 
  >   a=rtpmap:13 CN/8000 
  >   a=ptime:20 
  > 
  > 
  > 3) starting connection with extension playing a file (number 9999,   
  > second 
  > leg of eavesdropped session): 
  > 
  >   SIP/2.0 200 OK 
  >   Via: SIP/2.0/UDP 
  > 192.168.3.159:5080;rport=5080;branch=z9hG4bKv6Dg7myj4tvBS 
  >   From: "FreeSWITCH"   
  > <sip:myuser at mydomain;transport=udp>;tag=091j2Q0Fre8vp 
  >   To: <sip:[hidden email]:15060>;tag=U7t5Xt51rB64Q 
  >   Call-ID: 7551b7d0-3024-122d-b3a6-156c4939c4f8 
  >   CSeq: 121465623 INVITE 
  >   Contact: <sip:[hidden email]:15060;transport=udp> 
  >   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN 
  >   Accept: application/sdp 
  >   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, 
  > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH 
  >   Supported: timer, precondition, path, replaces 
  >   Allow-Events: talk, presence, dialog, call-info, sla, 
  > include-session-description, presence.winfo, message-summary, refer 
  >   Content-Type: application/sdp 
  >   Content-Disposition: session 
  >   Content-Length: 263 
  > 
  >   v=0 
  >   o=FreeSWITCH 5614997529598779838 636912243381649698 IN IP4   
  > 192.168.3.159 
  >   s=FreeSWITCH 
  >   c=IN IP4 192.168.3.159 
  >   t=0 0 
  >   m=audio 30086 RTP/AVP 0 101 13 
  >   a=rtpmap:0 PCMU/8000 
  >   a=rtpmap:101 telephone-event/8000 
  >   a=fmtp:101 0-16 
  >   a=rtpmap:13 CN/8000 
  >   a=ptime:20 
  > 
  > 
  > 
  > 
  > Anthony Minessale wrote: 
  >> 
  >> you probably have some device lying about ptime everywhere 
  >> look at a sip trace an pay especially close attention to ptime:x   
  >> param in 
  >> sdp 
  >> if you don't understand this just attach it here 
  >> 
  >> execute the following at the cli 
  >> sofia profile internal siptrace on 
  >> sofila loglevel debug 
  >> 
  >> 
  >> 
  >> On Thu, Oct 8, 2009 at 8:27 AM, Maciej Aniserowicz < 
  >> [hidden email]> wrote: 
  >> 
  >>> 
  >>> It's the same on the trunk (the last rev I used was not so old   
  >>> anyway). 
  >>> 
  >>> Codecs are the same on both legs: 
  >>> read codec/read rate: PCMU      8000 
  >>> write codec/write rate: PCMU    8000 
  >>> 
  >>> MA 
  >>> 
  >>> 
  >>> 
  >>> 
  >>> Michael Jerris wrote: 
  >>>> 
  >>>> What codecs are all the call legs using, also, please try current   
  >>>> svn 
  >>>> trunk. 
  >>>> 
  >>>> Mike 
  >>>> 
  >>>> On Oct 7, 2009, at 3:39 AM, Maciej Aniserowicz wrote: 
  >>>> 
  >>>>> 
  >>>>> Sorry about posting several questions at once, I wasn't aware it's 
  >>>>> "rude". 
  >>>>> Let's concentrate on this issue then. 
  >>>>> 
  >>>>> I use FS rev 14994. Phones on extensions: 
  >>>>> 1) x-lite 
  >>>>> 2) cisco sip phone 
  >>>>> 3) audio played by fs to the extension being eavesdropped 
  >>>>> 
  >>>>> I did not change any codec configuration, I just use the standard 
  >>>>> one that 
  >>>>> comes with both FS and the phones. 
  >>>>> Some time ago someone on FS irc channel told me that this is just 
  >>>>> how FS 
  >>>>> eavesdropping works... from your response I understand that this   
  >>>>> is 
  >>>>> not 
  >>>>> entirely true? 
  >>>>> 
  >>>>> Maciej Aniserowicz 
  >>>>> 
  >>>>> 
  >>>>> 
  >>>>> Anthony Minessale wrote: 
  >>>>>> 
  >>>>>> That's is a somewhat vague position. 
  >>>>>> 
  >>>>>> You did not mention which version of FreeSWITCH you are   
  >>>>>> running, the 
  >>>>>> phones 
  >>>>>> being used in your example, your configuration, the codecs in use 
  >>>>>> etc. 
  >>>>>> 
  >>>>>> BTW, 
  >>>>>> I think you should only ask one question at a time on this list. 
  >>>>>> The list 
  >>>>>> is run by volunteers and it's sort of rude to expect 3 or 4   
  >>>>>> threads 
  >>>>>> to be 
  >>>>>> tended to concerning the same one individual. 
  >>>>>> 
  >>>>>> 
  >>>>>> 2009/10/5 Maciej Aniserowicz <[hidden email]> 
  >>>>>> 
  >>>>>>> Hello, 
  >>>>>>> When I use eavesdropping in FreeSWITCH, the sound quality is 
  >>>>>>> really bad. 
  >>>>>>> Is 
  >>>>>>> there any way to improve it? Is this a known problem? 
  >>>>>>> Br/ 
  >>>>>>> Maciej Aniserowicz 
  >>>>>>> 
  >>>> 
  >>>> 
  >>>> _______________________________________________ 
  >>>> FreeSWITCH-users mailing list 
  >>>> [hidden email] 
  >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  >>>> 
  >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-  
  >>> users 
  >>>> http://www.freeswitch.org
  >>>> 
  >>>> 
  >>> 
  >>> -- 
  >>> View this message in context: 
  >>> http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3788019.html
  >>> Sent from the freeswitch-users mailing list archive at Nabble.com. 
  >>> 
  >>> _______________________________________________ 
  >>> FreeSWITCH-users mailing list 
  >>> [hidden email] 
  >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-  
  >>> users 
  >>> http://www.freeswitch.org
  >>> 
  >> 
  >> 
  >> 
  >> -- 
  >> Anthony Minessale II 
  >> 
  >> FreeSWITCH http://www.freeswitch.org/
  >> ClueCon http://www.cluecon.com/
  >> Twitter: http://twitter.com/FreeSWITCH_wire
  >> 
  >> AIM: anthm 
  >> MSN:[hidden email] <MSN 
  >> %[hidden email]> 
  >> GTALK/JABBER/PAYPAL:[hidden email]<PAYPAL 
  >> %[hidden email]> 
  >> IRC: irc.freenode.net #freeswitch 
  >> 
  >> FreeSWITCH Developer Conference 
  >> sip:[hidden email] <sip 
  >> %[hidden email]> 
  >> iax:[hidden email]/888 
  >> googletalk:[hidden email]<googletalk%3Aconf 
  >> %[hidden email]> 
  >> pstn:213-799-1400 
  >> 
  >> _______________________________________________ 
  >> FreeSWITCH-users mailing list 
  >> [hidden email] 
  >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-  
  >> users 
  >> http://www.freeswitch.org
  >> 
  >> 
  > 
  > -- 
  > View this message in context: http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3799274.html
  > Sent from the freeswitch-users mailing list archive at Nabble.com. 
  > 
  > _______________________________________________ 
  > FreeSWITCH-users mailing list 
  > [hidden email] 
  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-  
  > users 
  > http://www.freeswitch.org


  _______________________________________________ 
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  [hidden email] 
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org



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-- 
View this message in context: http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3806786.html
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From dujinfang at gmail.com  Mon Oct 12 04:19:53 2009
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 12 Oct 2009 19:19:53 +0800
Subject: [Freeswitch-users] mod_sofia.c registered calls how to know
In-Reply-To: <f8af5740910112114k3206c3cbq5757e2ff7de81162@mail.gmail.com>
References: <f8af5740910070748g1efee71ft5841b6fa4d2412c8@mail.gmail.com>
	<14A1216A-61C3-4C03-B710-57D2E7F4CD7A@jerris.com>
	<f8af5740910112114k3206c3cbq5757e2ff7de81162@mail.gmail.com>
Message-ID: <23f91030910120419o74f53279v5f061988ff585d56@mail.gmail.com>

try open YOUR_FreeSWITCH_INSTALL_DIR/db/*.db, you need sqlite3 to open them.
not sure how to do that on windows, but on linux:


# sqlite3 xx.db
sqlite> select * from sip_registration;

2009/10/12 srinivasula reddy <srinivas.ksvreddy at gmail.com>

> Hi Mike,
>
> Thanks for your valuable reply,
> when i install freeswitch1.0.2 in my machine(Windows xp operation system) i
> dont have any databasae installed in my system, then from sqllite will come
> into picture, and how can i see the registered users data from sqllite.
>
>
> Thanks
> Srinivas
>
> On Mon, Oct 12, 2009 at 3:04 AM, Michael Jerris <mike at jerris.com> wrote:
>
>>
>> On Oct 7, 2009, at 10:48 AM, srinivasula reddy wrote:
>>
>>
>> Hi
>>
>> can any please tell me where registered calls are stored, so when incoming
>> call came to mod_sofia.c how it will check it is registered or not?\\
>>
>>
>> Calls are not registered and calls have nothing to do with registration.
>>  Users are registered so that you may send calls to them.  Registration data
>> is stored either in a sqlite database, or optionally if you setup odbc, in
>> another database of your choice.  If you try to send a call to an
>> unregistered user in the dialplan using the proper syntax to send calls to
>> registered users (see the wiki for more details), and that user is not
>> registered, the bridge app will fail, optionally letting you continue on in
>> the dialplan based on variables such as continue_on_fail and
>> hangup_after_bridge.  You can use the sofia_contact function to see if there
>> is anyone registered to a specific user.
>>
>> Mike
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Srinivasula Reddy K
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From elihay at savion.huji.ac.il  Mon Oct 12 04:37:12 2009
From: elihay at savion.huji.ac.il (Eli Hayun)
Date: Mon, 12 Oct 2009 13:37:12 +0200
Subject: [Freeswitch-users] mod_sofia.c registered calls how to know
In-Reply-To: <23f91030910120419o74f53279v5f061988ff585d56@mail.gmail.com>
References: <f8af5740910070748g1efee71ft5841b6fa4d2412c8@mail.gmail.com>
	<14A1216A-61C3-4C03-B710-57D2E7F4CD7A@jerris.com>
	<f8af5740910112114k3206c3cbq5757e2ff7de81162@mail.gmail.com>
	<23f91030910120419o74f53279v5f061988ff585d56@mail.gmail.com>
Message-ID: <1255347432.3724.4.camel@eli-desktop>

Is it possible to keep a list of registered phone, and when FS will
start it will register them all automatically?


On Mon, 2009-10-12 at 13:19 +0200, Seven Du wrote:

> try open YOUR_FreeSWITCH_INSTALL_DIR/db/*.db, you need sqlite3 to open
> them. not sure how to do that on windows, but on linux:
> 
> 
> # sqlite3 xx.db
> sqlite> select * from sip_registration;
> 
> 
> 2009/10/12 srinivasula reddy <srinivas.ksvreddy at gmail.com>
> 
>         Hi Mike,
>         
>         Thanks for your valuable reply,
>         when i install freeswitch1.0.2 in my machine(Windows xp
>         operation system) i dont have any databasae installed in my
>         system, then from sqllite will come into picture, and how can
>         i see the registered users data from sqllite.
>         
>         
>         Thanks
>         Srinivas
>         
>         
>         
>         On Mon, Oct 12, 2009 at 3:04 AM, Michael Jerris
>         <mike at jerris.com> wrote:
>         
>                 
>                 
>                 
>                 On Oct 7, 2009, at 10:48 AM, srinivasula reddy wrote:
>                 
>                 
>                 
>                 > 
>                 > Hi
>                 > 
>                 > can any please tell me where registered calls are
>                 > stored, so when incoming call came to mod_sofia.c
>                 > how it will check it is registered or not?\\
>                 > 
>                 
>                 
>                 Calls are not registered and calls have nothing to do
>                 with registration.  Users are registered so that you
>                 may send calls to them.  Registration data is stored
>                 either in a sqlite database, or optionally if you
>                 setup odbc, in another database of your choice.  If
>                 you try to send a call to an unregistered user in the
>                 dialplan using the proper syntax to send calls to
>                 registered users (see the wiki for more details), and
>                 that user is not registered, the bridge app will fail,
>                 optionally letting you continue on in the dialplan
>                 based on variables such as continue_on_fail and
>                 hangup_after_bridge.  You can use the sofia_contact
>                 function to see if there is anyone registered to a
>                 specific user.
>                 
>                 
>                 Mike
>                 
>                 
>                 
>                 
>                 
>                 _______________________________________________
>                 FreeSWITCH-users mailing list
>                 FreeSWITCH-users at lists.freeswitch.org
>                 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>                 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>                 http://www.freeswitch.org
>                 
>                 
>         
>         
>         
>         
>         -- 
>         Srinivasula Reddy K
>         
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>         
> 
> 
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From lakindia89 at gmail.com  Mon Oct 12 04:54:18 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Mon, 12 Oct 2009 17:24:18 +0530
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <87f2f3b90910101009j1295d812jd98cae91e30b1eff@mail.gmail.com>
References: <7d79b3930910090407m5dc81d12pf2b0a47f6017804@mail.gmail.com>
	<20091009123615.BC50A3FA6E9@mail.cune.org>
	<7d79b3930910100647u7fd11413ucec0a535d68b3e3a@mail.gmail.com>
	<87f2f3b90910101009j1295d812jd98cae91e30b1eff@mail.gmail.com>
Message-ID: <7d79b3930910120454m1ffc6dd4u42b91a4349597166@mail.gmail.com>

Ya ok. Here is the required stuff
Configuration: (PRI span)
<configuration name="openzap.conf" description="OpenZAP Configuration">
<settings>
<param name="debug" value="7"/>
</settings>
<pri_spans>
<span id="1" name="PRI_1">
<param name="q921loglevel" value="alert"/>
<param name="q931loglevel" value="alert"/>
<param name="mode" value="user"/>
<param name="dialect" value="q931"/>
<param name="dialplan" value="XML"/>
<param name="context" value="default"/>
</span>
</pri_spans>
</configuration>

Log while starting freeswitch:
http://pastebin.freeswitch.org/10646

Log when making a call:
http://pastebin.freeswitch.org/10647


Configuration: (LIBPRI SPAN)
<configuration name="openzap.conf" description="OpenZAP Configuration">
<settings>
<param name="debug" value="7"/>
</settings>
<libpri_spans>
<span name="PRI_1">
<param name="node" value="cpe"/>
<param name="switch" value="euroisdn"/>
<param name="dialplan" value="XML"/>
<param name="context" value="default"/>
</span>
</libpri_spans>
</configuration>

Log while starting freeswitch:
http://pastebin.freeswitch.org/10648

Log while making a call:
http://pastebin.freeswitch.org/10649



On Sat, Oct 10, 2009 at 10:39 PM, Michael Collins <msc at freeswitch.org>wrote:

> Okay, please go to pastebin.freeswitch.org and paste your openzap.conf.xml
> file. Also, paste the ENTIRE debug log from a call from start to finish.
> Telling us that you see INVALID_IE_CONTENTS doesn't help if we don't know
> what the information element contains. Finally, turn on PRI debugging and
> make another test call and pastebin that debug log as well. The debug will
> show details about the communications between your machine and the carrier.
> Instructions for turning on debugging for libpri are found in the OpenZAP
> wiki page in the same place where the libpri instructions are located.
>
> Put the pastebin number in this email thread and then we'll go have a look.
>
> Thanks,
> MC
>
>
> On Sat, Oct 10, 2009 at 6:47 AM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> Hi,
>> I've tried with the following dialplan(After making the changes that are
>> recommended). But still I got INVALID_IE_CONTENTS error.
>>
>> Another thing is, if I use prispan configuration, I'm able to make
>> outgoing calls.
>> But if I use the libpri span configuration I'm not able to make outgoing
>> calls, which says INVALID_IE_CONTENTS.
>>
>> I've been struck with this problem for the past 1 week.
>> Any solution to this???
>>
>>
>>
>> On Fri, Oct 9, 2009 at 6:06 PM, <Russell.Mosemann at cune.org> wrote:
>>
>>> lakshmanan ganapathy <lakindia89 at gmail.com> said:
>>>
>>> > But still I'm facing problem with the outgoing call. It says
>>> > INVALID_IE_CONTENTS.
>>> > What might be the issue? Even I tried the following dialplan to call by
>>> > using bridge.
>>> > <extension name="outgoing-fxo">
>>> >                   <condition field="destination_number"
>>> > expression="^(\d{10})$">
>>> >                           <action application="answer"/>
>>> >                           <action application="bridge"
>>> > data="openzap/1/1/${dialed_ext}"/>
>>>
>>> Does "answer" need to be called here? I haven't used an fxo. So, I don't
>>> know. What value does $dialed_ext have? If you want to use the number
>>> matched in the condition, then it should be
>>>
>>> openzap/1/1/$1
>>>
>>> --
>>> Russell Mosemann
>>>
>>>
>>>
>>> ________________________________________________________
>>> Concordia University, Nebraska
>>> See http://www.cune.edu/ for the latest news and events!
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Mon Oct 12 06:16:27 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Oct 2009 08:16:27 -0500
Subject: [Freeswitch-users] Re corded file as voicemail.
In-Reply-To: <23f91030910112315h5ea79b81ue55381a3caff0173@mail.gmail.com>
References: <25850181.post@talk.nabble.com>
	<BEDBBF51-2D48-4CDE-A467-11D5BBB458E1@freeswitch.org>
	<23f91030910112315h5ea79b81ue55381a3caff0173@mail.gmail.com>
Message-ID: <5CCE4643-063D-4F2D-B98F-35058F465C10@freeswitch.org>

Perfect! Thanks.

/b

On Oct 12, 2009, at 1:15 AM, Seven Du wrote:

> http://jira.freeswitch.org/browse/MODCODEC-15
>
> Is it ok I assigned to you ?
>
> Thanks.

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From brian at freeswitch.org  Mon Oct 12 06:23:20 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Oct 2009 08:23:20 -0500
Subject: [Freeswitch-users] Bad sound quality while eavesdropping
In-Reply-To: <1255337256919-3806786.post@n2.nabble.com>
References: <41A44DD027064988A914974405788C2E@procent>
	<191c3a030910050731m2d74979ep4598e5a1945d58ae@mail.gmail.com>
	<1254901192035-3780245.post@n2.nabble.com>
	<8437F5BC-7AFF-4A74-B8CD-C5B8219021F6@jerris.com>
	<1255008427639-3788019.post@n2.nabble.com>
	<191c3a030910080823g79c7c596x1cd887e1538ce2e1@mail.gmail.com>
	<1255169044209-3799274.post@n2.nabble.com>
	<CF8C1791-24CD-4132-83B5-7F83B5BDCCC3@jerris.com>
	<1255337256919-3806786.post@n2.nabble.com>
Message-ID: <59F3CD44-5FEA-403C-98BE-EEE49EC3815B@freeswitch.org>

Did you open a jira and attach all the info?

/b

On Oct 12, 2009, at 3:47 AM, Maciej Aniserowicz wrote:

> Yes, I confirmed that with Wireshark (filter "rtp and ip.src ==  
> <device ip>). RTP packets are sent every 20ms.
>
> MAniserowicz
>

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From bottleman at icf.org.ru  Mon Oct 12 06:31:03 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Mon, 12 Oct 2009 17:31:03 +0400 (MSD)
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>

On 2009-10-08 20:32 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>Hi Yuriy,
TC>
TC>can you share what you have so far, I'm sure we can help with RTP part...

ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/ alfa code, but seems it work, but should be buggy, 
to build need libpt 2.6.5 and h323plus cvs version, i test it now on fs 1.0.4.

TC>
TC>T.
TC>
TC>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
TC>
TC>> On 2009-10-08 13:25 -0400, Tuyan ?zipek wrote
TC>> freeswitch-users at lists.freesw...:
TC>>
TC>> Tz>Hi,
TC>> Tz>
TC>> Tz>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>:
TC>> Tz>> On 2009-10-08 10:43 -0500, Anthony Minessale wrote
TC>> freeswitch-users at lists.f...:
TC>> Tz>>
TC>> Tz>> AM>If you are going to make that alternate module are you going to
TC>> host it in
TC>> Tz>> AM>the FS tree along side mod_opal?
TC>> Tz>>
TC>> Tz>> Yes, but then it be useful, now i have working only signaling part and
TC>> some
TC>> Tz>> kinde of not working rtp part :)
TC>> Tz>
TC>> Tz>If you dont use fs rtp stack, its unlikely that it will be accepted
TC>> Tz>into the tree.
TC>> Tz>
TC>> Tz>>
TC>> Tz>> AM>also if were working on mod_opal why did you not try to involve us
TC>> and the
TC>> Tz>> AM>opal team?
TC>> Tz>>
TC>> Tz>> Because i made patches for libopal, one is a bugfix in rtp part, there
TC>> is a race condition
TC>> Tz>> in inicialisation in jitter buffer, another patch implements method to
TC>> send progress indicator,
TC>> Tz>> and i don't wont spent my time to incorporate this changes into
TC>> libopal.
TC>> Tz>
TC>> Tz>Thats bad.
TC>> Tz>Any bugfixes from fs, goes to upstream on any of the used libraries.
TC>> Tz>You should be doing the same.
TC>> Tz>And Opal developers, will either include or refuse your patches. If
TC>> Tz>they refuse it, they will give you the reason.
TC>>
TC>> i make this fix only to freeze my current mod_opal work on working state,
TC>> while it now work for me i work on
TC>> my new implementation of h323 proto for fs, i think opal developers will
TC>> fix this rtp bug himself becouse
TC>> it crashes and make library unuseful.
TC>>
TC>> Tz>
TC>> Tz>> without this changes
TC>> Tz>> my work on mod_opal in freeswitch don't useful at all, i provide link
TC>> to my work with all
TC>> Tz>> patches, if somebody wont incorporate it in libopal tree and fs - go
TC>> on, but i think
TC>> Tz>> better and more elegant make new module based on h323plus.
TC>> Tz>
TC>> Tz>If you dont publish your changes, all those you are trying to achieve,
TC>> Tz>wont happen.
TC>> Tz>
TC>> Tz>>
TC>> Tz>> AM>How far away from what is in tree are these patches you have?
TC>> Tz>> AM>
TC>> Tz>> AM>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
TC>> Tz>> AM>
TC>> Tz>> AM>> On 2009-10-07 15:09 -0500, Brian West wrote
TC>> Tz>> AM>> freeswitch-users at lists.freeswit...:
TC>> Tz>> AM>>
TC>> Tz>> AM>> opal have addition abstraction layer called opalmgr, and it
TC>> implementation
TC>> Tz>> AM>> is not so good in
TC>> Tz>> AM>> this case, for example to implemet pre_answer in mod_opal i need
TC>> patch
TC>> Tz>> AM>> libopal, because
TC>> Tz>
TC>> Tz>The patch you have in there, adds a method to the OpalCall,  it does
TC>> Tz>not touch any parts of OpalManager
TC>> Tz>so, i dont understand why opalmanager would be the cause of your pain?
TC>> Tz>
TC>> Tz>> AM>> there is no way to send progress inicator throuch opalmgr. and
TC>> there is
TC>> Tz>> AM>> many another issues like
TC>> Tz>> AM>> this in that layer.
TC>> Tz>
TC>> Tz>Please point me to the issues you have in opal, their bug reports ,
TC>> traces etc.
TC>> Tz>I dont think any of the opal people has psychic abilities to detect
TC>> Tz>-your- problems
TC>> Tz>and solve them.
TC>> Tz>
TC>> Tz>>ftp://srv.icf.org.ru/pub/soft/f/freeswitch/ - there is
TC>> Tz>> AM>> my work on mod_opal before
TC>> Tz>> AM>> i start moving to h323plus, may be this help somebody there.
TC>> Tz>> AM>>
TC>> Tz>> AM>> BW> From what I have been told h323plus is a based/fork of
TC>> OpenH323 which
TC>> Tz>> AM>> BW>OPAL is just a continuation of OpenH323.  So why not support
TC>> the
TC>> Tz>> AM>> BW>developers of OPAL/OpenH323 ?
TC>> Tz>> AM>> BW>
TC>> Tz>> AM>> BW>/b
TC>> Tz>> AM>> BW>
TC>> Tz>> AM>> BW>On Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote:
TC>> Tz>> AM>> BW>
TC>> Tz>> AM>> BW>> We are developing module to handle h323 proto now, we try to
TC>> use
TC>> Tz>> AM>> BW>> mod_opal and try improve it, but no luck,
TC>> Tz>> AM>> BW>> there is many issues in libopal, and finaly we now move to
TC>> h323plus
TC>> Tz>> AM>> BW>> library.
TC>> Tz>
TC>> Tz>Did any of you try to report those issues?
TC>> Tz>
TC>> Tz>Regards
TC>> Tz>
TC>> Tz>/tyn
TC>> Tz>
TC>> Tz>> AM>> BW>
TC>> Tz>> AM>> BW>
TC>> Tz>> AM>> BW>_______________________________________________
TC>> Tz>> AM>> BW>FreeSWITCH-users mailing list
TC>> Tz>> AM>> BW>FreeSWITCH-users at lists.freeswitch.org
TC>> Tz>> AM>> BW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> Tz>> AM>> BW>UNSUBSCRIBE:
TC>> Tz>> AM>> http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> Tz>> AM>> BW>http://www.freeswitch.org
TC>> Tz>> AM>> BW>
TC>> Tz>> AM>>
TC>> Tz>> AM>> C ?????????                       With Best Regards
TC>> Tz>> AM>> ???????????? ????.                Georgiewskiy Yuriy
TC>> Tz>> AM>> +7 4872 711666                    +7 4872 711666
TC>> Tz>> AM>> ???? +7 4872 711143               fax +7 4872 711143
TC>> Tz>> AM>> ???????? ??? "?? ?? ??????"       IT Service Ltd
TC>> Tz>> AM>> http://nkoort.ru                  http://nkoort.ru
TC>> Tz>> AM>> JID: GHhost at jabber.tula-ix.net.ru JID:
TC>> GHhost at jabber.tula-ix.net.ru
TC>> Tz>> AM>> YG129-RIPE                        YG129-RIPE
TC>> Tz>> AM>>
TC>> Tz>> AM>> _______________________________________________
TC>> Tz>> AM>> FreeSWITCH-users mailing list
TC>> Tz>> AM>> FreeSWITCH-users at lists.freeswitch.org
TC>> Tz>> AM>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> Tz>> AM>> UNSUBSCRIBE:
TC>> http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> Tz>> AM>> http://www.freeswitch.org
TC>> Tz>> AM>>
TC>> Tz>> AM>>
TC>> Tz>> AM>
TC>> Tz>> AM>
TC>> Tz>> AM>
TC>> Tz>>
TC>> Tz>> C ?????????                       With Best Regards
TC>> Tz>> ???????????? ????.                Georgiewskiy Yuriy
TC>> Tz>> +7 4872 711666                    +7 4872 711666
TC>> Tz>> ???? +7 4872 711143               fax +7 4872 711143
TC>> Tz>> ???????? ??? "?? ?? ??????"       IT Service Ltd
TC>> Tz>> http://nkoort.ru                  http://nkoort.ru
TC>> Tz>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
TC>> Tz>> YG129-RIPE                        YG129-RIPE
TC>> Tz>> _______________________________________________
TC>> Tz>> FreeSWITCH-users mailing list
TC>> Tz>> FreeSWITCH-users at lists.freeswitch.org
TC>> Tz>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> Tz>> UNSUBSCRIBE:
TC>> http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> Tz>> http://www.freeswitch.org
TC>> Tz>>
TC>> Tz>>
TC>> Tz>
TC>> Tz>_______________________________________________
TC>> Tz>FreeSWITCH-users mailing list
TC>> Tz>FreeSWITCH-users at lists.freeswitch.org
TC>> Tz>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> Tz>UNSUBSCRIBE:
TC>> http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> Tz>http://www.freeswitch.org
TC>> Tz>
TC>>
TC>> C ?????????                       With Best Regards
TC>> ???????????? ????.                Georgiewskiy Yuriy
TC>> +7 4872 711666                    +7 4872 711666
TC>> ???? +7 4872 711143               fax +7 4872 711143
TC>> ???????? ??? "?? ?? ??????"       IT Service Ltd
TC>> http://nkoort.ru                  http://nkoort.ru
TC>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
TC>> YG129-RIPE                        YG129-RIPE
TC>>
TC>> _______________________________________________
TC>> FreeSWITCH-users mailing list
TC>> FreeSWITCH-users at lists.freeswitch.org
TC>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> http://www.freeswitch.org
TC>>
TC>>
TC>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From anthony.minessale at gmail.com  Mon Oct 12 07:26:18 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 12 Oct 2009 09:26:18 -0500
Subject: [Freeswitch-users] mod_socket: custom timeout for HEARTBEAT
	event
In-Reply-To: <d211f54a0910112238i5f69275ey7cad437e47e6b567@mail.gmail.com>
References: <d211f54a0910110959y3d3deed7s43201a3e39756585@mail.gmail.com>
	<86a32abc0910111444j1ab2e446p3cb94b0ae04126e3@mail.gmail.com>
	<97B4BDC9-5B11-4A98-B1BF-CC00A82ECBD9@jerris.com>
	<d211f54a0910112238i5f69275ey7cad437e47e6b567@mail.gmail.com>
Message-ID: <191c3a030910120726p46d6410ejd94b61612686fbbc@mail.gmail.com>

it works in either case with or without media
the syntax for setting the frequency was answered above.


On Mon, Oct 12, 2009 at 12:38 AM, Artem Shiyanov <shiyanov at gmail.com> wrote:

> Michael, Diego,
> thanks for the rapid answers!
>
> As far as I understand, "enable_heartbeat" app is launching
> SESSION_HEARTBEAT events that will stop when the call will be cleared. Also
> I "heard" that "enable_heartbeat" works only for calls with proxied media.
>
> What I want is to monitor FreeSwitch status: is it alive and what is the
> system status message. This info is provided in HEARTBEAT event gracefully
> but in constant time period = 20 sec. So the main question is- how to
> customize this period?
>
>
> Artem
>
>
>
>
> On Mon, Oct 12, 2009 at 2:52 AM, Michael Jerris <mike at jerris.com> wrote:
>
>>
>> On Oct 11, 2009, at 5:44 PM, Diego Viola wrote:
>>
>> > Maybe you want enable_heartbeat or sched_heartbeat from mod_dptools?
>> >
>> > You can pass your parameters in second to these two.
>> >
>> > Example:
>> >
>> > <action application="enable_heartbeat" data="1"/>
>> > <action application="sched_heartbeat" data="1"/>
>> >
>> > Where 1 in this case is the number of heartbeats per seconds.
>> >
>>
>> Number of seconds between hearbeats, not hearbeats per second.
>>
>>
>> > You can use that example on the Dialplan XML but you can also use it
>> > on mod_event_socket outbound, etc.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From tculjaga at gmail.com  Mon Oct 12 07:27:25 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 12 Oct 2009 16:27:25 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
Message-ID: <65d96fc80910120727q6d35b51ard78005312079e021@mail.gmail.com>

2009/10/12 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-08 20:32 +0200, Tihomir Culjaga wrote
> freeswitch-users at lists.fre...:
>
> TC>Hi Yuriy,
> TC>
> TC>can you share what you have so far, I'm sure we can help with RTP
> part...
>
> ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/ alfa code, but seems
> it work, but should be buggy,
> to build need libpt 2.6.5 and h323plus cvs version, i test it now on fs
> 1.0.4.
>
> TC>
> TC>T.
> TC>
> TC>2009/10/8 Georgiewskiy Yuriy <bottleman at icf.org.ru>
>

got it and building it right now...

T.
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From ngay01042005 at gmail.com  Mon Oct 12 04:01:10 2009
From: ngay01042005 at gmail.com (homqua)
Date: Mon, 12 Oct 2009 04:01:10 -0700 (PDT)
Subject: [Freeswitch-users]  Question about fax tone detection
Message-ID: <1255345270255-3807298.post@n2.nabble.com>


Hi,
I have implemented the solution for tone detection in wiki, and also answer
the channel before detecting the tone:
<condition field="destination_number" expression="^(055138419992)$">
    <action application="answer"/>
    <action application="tone_detect" data="fax 1100 r +15000 transfer fax
XML default"/>


<extension name="fax">
      <condition field="destination_number" expression="^fax$">
        <action application="answer" />
        <action application="sleep" data="1000"/>
        <action application="rxfax"
data="/usr/local/freeswitch/storage/fax/${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}.tiff"/>
    <action application="set" data="fax_mode=recv"/>
        <action application="hangup"/>
      </condition>
    </extension>


But FS cannot recognize the tone, and therefore cannot move to fax
extension.  Below are the error in FS:

2009-10-12 10:57:16.702287 [NOTICE] switch_channel.c:602 New Channel
sofia/external/anonymous at anonymous.invalid
[c431f0a3-9231-4724-ba39-9e4ef7edfca2]
2009-10-12 10:57:16.703413 [INFO] mod_dialplan_xml.c:315 Processing
Anonymous->055138419992 in context public
2009-10-12 10:57:16.719288 [NOTICE] switch_ivr.c:1349 Transfer
sofia/external/anonymous at anonymous.invalid to XML[055138419992 at default]
2009-10-12 10:57:16.719288 [INFO] mod_dialplan_xml.c:315 Processing
Anonymous->055138419992 in context default
2009-10-12 10:57:16.722289 [NOTICE] mod_dptools.c:649 Channel
[sofia/external/anonymous at anonymous.invalid] has been answered
2009-10-12 10:57:16.722289 [NOTICE] mod_dptools.c:1324 Enabling tone
detection 'fax' '1100'
2009-10-12 10:57:16.723302 [NOTICE] switch_core_state_machine.c:179 Hangup
sofia/external/anonymous at anonymous.invalid [CS_EXECUTE] [NORMAL_CLEARING]
2009-10-12 10:57:16.740285 [NOTICE] switch_core_session.c:1086 Session 1
(sofia/external/anonymous at anonymous.invalid) Ended
2009-10-12 10:57:16.740285 [NOTICE] switch_core_session.c:1088 Close Channel
sofia/external/anonymous at anonymous.invalid [CS_DESTROY]

And the trace for SIP messages:  http://pastebin.com/m4e47e7d9 

If anyone has any idea, tell me please.
Thanks.
-- 
View this message in context: http://n2.nabble.com/Question-about-fax-tone-detection-tp3807298p3807298.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From mattdfong at gmail.com  Mon Oct 12 07:41:30 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Mon, 12 Oct 2009 21:41:30 +0700
Subject: [Freeswitch-users] hangup_after_bridge=false not continueing after
	failed bridge
Message-ID: <4256bf830910120741g5facaa6cr7ed92303dfafe74b@mail.gmail.com>

I think think this might be a bug, but wanted to post here instead of Jira
in-case I'm overlooking a configuration variable
Dialplan

    <extension name="1920"><!--init agent for manual and power dial mode-->
      <condition field="destination_number" expression="^1920$">
        <action application="set" data="hangup_after_bridge=false"/>
        <action application="bridge" data="sofia/gateway/
debug.com/14159927717"/>
        <action application="transfer" data="1999"/><!-- send to unable to
reach any contacts-->
      </condition>
    </extension>

API Command
originate sofia/internal/sip_1%192.168.1.10 1920

When the bridge to 14159927717 fails (NO_ANSWER) both calls are terminated
instead of continuing on in the dial plan to exten 1999 (which in my
dialplan parks the call). hangup_after_bridge however seems to work OK if
someone picks up in the bridge. Is this the correct behavior? How else can I
prevent the call from hanging up if a bridge fails? Thanks.

I'm using 15135M

--matt
http://www.hellohunter.com - Predictive Dialer
http://www.hellohunter.com/voice_broadcast.php - Voice Broadcasting
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From brian at freeswitch.org  Mon Oct 12 07:43:46 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Oct 2009 09:43:46 -0500
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
Message-ID: <39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>

We can host this in our SVN if you wish?

/b

On Oct 12, 2009, at 8:31 AM, Georgiewskiy Yuriy wrote:

> ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/ alfa code, but  
> seems it work, but should be buggy,
> to build need libpt 2.6.5 and h323plus cvs version, i test it now on  
> fs 1.0.4.

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From bottleman at icf.org.ru  Mon Oct 12 07:53:18 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Mon, 12 Oct 2009 18:53:18 +0400 (MSD)
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<86a32abc0910061445w44ad9c60jb5aca1b7b7c29681@mail.gmail.com>
	<65d96fc80910061641u5870f081o8d49d2e89546f740@mail.gmail.com>
	<Pine.LNX.4.64.0910071645500.32499@srv.icf.org.ru>
	<82C4D161-9CD8-4D33-B05A-CDC701EF23EC@freeswitch.org>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
Message-ID: <Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>

On 2009-10-12 09:43 -0500, Brian West wrote freeswitch-users at lists.freeswit...:

BW>We can host this in our SVN if you wish?

If in fs svn i think yes. But i think may be little time later? 
i don't known is it builds on trunk because i develop it on 1.0.4.

BW>/b
BW>
BW>On Oct 12, 2009, at 8:31 AM, Georgiewskiy Yuriy wrote:
BW>
BW>> ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/ alfa code, but seems
BW>> it work, but should be buggy,
BW>> to build need libpt 2.6.5 and h323plus cvs version, i test it now on fs
BW>> 1.0.4.
BW>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From anthony.minessale at gmail.com  Mon Oct 12 08:06:54 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 12 Oct 2009 10:06:54 -0500
Subject: [Freeswitch-users] hangup_after_bridge=false not continueing
	after failed bridge
In-Reply-To: <4256bf830910120741g5facaa6cr7ed92303dfafe74b@mail.gmail.com>
References: <4256bf830910120741g5facaa6cr7ed92303dfafe74b@mail.gmail.com>
Message-ID: <191c3a030910120806o30911131o743aa58420b41ae4@mail.gmail.com>

which line is hanging up your A (inbound) leg?

look for a blue line that says "Hangup xyz...." that matches it so i can
see.

I think what is happening is you are getting early media so the bridge is
actually working then when nobody answers it dies but technically the bridge
worked.

On Mon, Oct 12, 2009 at 9:41 AM, Matthew Fong <mattdfong at gmail.com> wrote:

> I think think this might be a bug, but wanted to post here instead of Jira
> in-case I'm overlooking a configuration variable
> Dialplan
>
>     <extension name="1920"><!--init agent for manual and power dial mode-->
>       <condition field="destination_number" expression="^1920$">
>         <action application="set" data="hangup_after_bridge=false"/>
>         <action application="bridge" data="sofia/gateway/
> debug.com/14159927717"/>
>         <action application="transfer" data="1999"/><!-- send to unable to
> reach any contacts-->
>       </condition>
>     </extension>
>
> API Command
> originate sofia/internal/sip_1%192.168.1.10 1920
>
> When the bridge to 14159927717 fails (NO_ANSWER) both calls are terminated
> instead of continuing on in the dial plan to exten 1999 (which in my
> dialplan parks the call). hangup_after_bridge however seems to work OK if
> someone picks up in the bridge. Is this the correct behavior? How else can I
> prevent the call from hanging up if a bridge fails? Thanks.
>
> I'm using 15135M
>
> --matt
> http://www.hellohunter.com - Predictive Dialer
> http://www.hellohunter.com/voice_broadcast.php - Voice Broadcasting
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mattdfong at gmail.com  Mon Oct 12 08:23:54 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Mon, 12 Oct 2009 22:23:54 +0700
Subject: [Freeswitch-users] hangup_after_bridge=false not continueing
	after failed bridge
In-Reply-To: <191c3a030910120806o30911131o743aa58420b41ae4@mail.gmail.com>
References: <4256bf830910120741g5facaa6cr7ed92303dfafe74b@mail.gmail.com>
	<191c3a030910120806o30911131o743aa58420b41ae4@mail.gmail.com>
Message-ID: <4256bf830910120823w5fcbc3eja4afb76b60b68fda@mail.gmail.com>

2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179 Hangup
sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]

might be the line..or the entire output is below....

freeswitch at matthew-laptop> originate sofia/internal/sip_1%192.168.1.10 1920
2009-10-12 15:21:44.029517 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/sip_1 [1e722934-7e94-46aa-9d62-e6ec7e7449cf]
2009-10-12 15:21:44.121484 [NOTICE] sofia.c:3552 Ring-Ready
sofia/internal/sip_1!
2009-10-12 15:21:47.285531 [NOTICE] sofia.c:3998 Channel
[sofia/internal/sip_1] has been answered
2009-10-12 15:21:47.290996 [INFO] mod_dialplan_xml.c:391 Processing
FreeSWITCH->1920 in context default
2009-10-12 15:21:47.293452 [NOTICE] switch_channel.c:613 New Channel
sofia/external/14159927717 [6b6cc440-e1d6-415a-b84b-494117e7361d]
2009-10-12 15:21:47.293452 [NOTICE] switch_ivr.c:1367 Transfer
sofia/internal/sip_1 to XML[1920 at default]
API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
+OK 1e722934-7e94-46aa-9d62-e6ec7e7449cf

freeswitch at matthew-laptop> 2009-10-12 15:21:47.369855 [NOTICE] sofia.c:3552
Ring-Ready sofia/external/14159927717!
2009-10-12 15:22:47.009474 [NOTICE] switch_ivr_originate.c:2336 Hangup
sofia/external/14159927717 [CS_CONSUME_MEDIA] [NO_ANSWER]
2009-10-12 15:22:47.009474 [INFO] switch_cpp.cpp:1116 PCHangup
gw:debug.comhc:NO_ANSWER du:0 cn:sofia/external/14159927717
2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1087 Session 47
(sofia/external/14159927717) Ended
2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1089 Close Channel
sofia/external/14159927717 [CS_DESTROY]
2009-10-12 15:22:47.009474 [INFO] mod_dptools.c:2133 Originate Failed.
 Cause: NO_ANSWER
2009-10-12 15:22:47.009474 [NOTICE] switch_ivr.c:1367 Transfer
sofia/internal/sip_1 to XML[1999 at default]
2009-10-12 15:22:47.009474 [INFO] mod_dialplan_xml.c:391 Processing
FreeSWITCH->1999 in context default
2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179 Hangup
sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1087 Session 46
(sofia/internal/sip_1) Ended
2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1089 Close Channel
sofia/internal/sip_1 [CS_DESTROY]


thanks for looking at this.

On Mon, Oct 12, 2009 at 10:06 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> which line is hanging up your A (inbound) leg?
>
> look for a blue line that says "Hangup xyz...." that matches it so i can
> see.
>
> I think what is happening is you are getting early media so the bridge is
> actually working then when nobody answers it dies but technically the bridge
> worked.
>
> On Mon, Oct 12, 2009 at 9:41 AM, Matthew Fong <mattdfong at gmail.com> wrote:
>
>> I think think this might be a bug, but wanted to post here instead of Jira
>> in-case I'm overlooking a configuration variable
>> Dialplan
>>
>>     <extension name="1920"><!--init agent for manual and power dial
>> mode-->
>>       <condition field="destination_number" expression="^1920$">
>>         <action application="set" data="hangup_after_bridge=false"/>
>>         <action application="bridge" data="sofia/gateway/
>> debug.com/14159927717"/>
>>         <action application="transfer" data="1999"/><!-- send to unable to
>> reach any contacts-->
>>       </condition>
>>     </extension>
>>
>> API Command
>> originate sofia/internal/sip_1%192.168.1.10 1920
>>
>> When the bridge to 14159927717 fails (NO_ANSWER) both calls are terminated
>> instead of continuing on in the dial plan to exten 1999 (which in my
>> dialplan parks the call). hangup_after_bridge however seems to work OK if
>> someone picks up in the bridge. Is this the correct behavior? How else can I
>> prevent the call from hanging up if a bridge fails? Thanks.
>>
>> I'm using 15135M
>>
>> --matt
>> http://www.hellohunter.com - Predictive Dialer
>> http://www.hellohunter.com/voice_broadcast.php - Voice Broadcasting
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mattdfong at gmail.com  Mon Oct 12 08:25:53 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Mon, 12 Oct 2009 22:25:53 +0700
Subject: [Freeswitch-users] hangup_after_bridge=false not continueing
	after failed bridge
In-Reply-To: <4256bf830910120823w5fcbc3eja4afb76b60b68fda@mail.gmail.com>
References: <4256bf830910120741g5facaa6cr7ed92303dfafe74b@mail.gmail.com>
	<191c3a030910120806o30911131o743aa58420b41ae4@mail.gmail.com>
	<4256bf830910120823w5fcbc3eja4afb76b60b68fda@mail.gmail.com>
Message-ID: <4256bf830910120825u70e988p4d7a041c520048a@mail.gmail.com>

    <extension name="1999"><!--DIRECT POWER-->
      <condition field="destination_number" expression="^1997$">
        <action application="playback"
data="hh/hh-unable_to_connect_contact.wav"/>
        <action application="park"/>
      </condition>
    </extension>

my extn 1999... since it looks from the output like it's transferring, just
don't know why it's disconnecting the call instead of playing the .wav and
parking.

On Mon, Oct 12, 2009 at 10:23 PM, Matthew Fong <mattdfong at gmail.com> wrote:

> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179 Hangup
> sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
>
> might be the line..or the entire output is below....
>
> freeswitch at matthew-laptop> originate sofia/internal/sip_1%192.168.1.10
> 1920
> 2009-10-12 15:21:44.029517 [NOTICE] switch_channel.c:613 New Channel
> sofia/internal/sip_1 [1e722934-7e94-46aa-9d62-e6ec7e7449cf]
> 2009-10-12 15:21:44.121484 [NOTICE] sofia.c:3552 Ring-Ready
> sofia/internal/sip_1!
> 2009-10-12 15:21:47.285531 [NOTICE] sofia.c:3998 Channel
> [sofia/internal/sip_1] has been answered
> 2009-10-12 15:21:47.290996 [INFO] mod_dialplan_xml.c:391 Processing
> FreeSWITCH->1920 in context default
> 2009-10-12 15:21:47.293452 [NOTICE] switch_channel.c:613 New Channel
> sofia/external/14159927717 [6b6cc440-e1d6-415a-b84b-494117e7361d]
> 2009-10-12 15:21:47.293452 [NOTICE] switch_ivr.c:1367 Transfer
> sofia/internal/sip_1 to XML[1920 at default]
> API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
> +OK 1e722934-7e94-46aa-9d62-e6ec7e7449cf
>
> freeswitch at matthew-laptop> 2009-10-12 15:21:47.369855 [NOTICE]
> sofia.c:3552 Ring-Ready sofia/external/14159927717!
> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr_originate.c:2336 Hangup
> sofia/external/14159927717 [CS_CONSUME_MEDIA] [NO_ANSWER]
> 2009-10-12 15:22:47.009474 [INFO] switch_cpp.cpp:1116 PCHangup gw:
> debug.com hc:NO_ANSWER du:0 cn:sofia/external/14159927717
> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1087 Session 47
> (sofia/external/14159927717) Ended
> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1089 Close
> Channel sofia/external/14159927717 [CS_DESTROY]
> 2009-10-12 15:22:47.009474 [INFO] mod_dptools.c:2133 Originate Failed.
>  Cause: NO_ANSWER
> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr.c:1367 Transfer
> sofia/internal/sip_1 to XML[1999 at default]
> 2009-10-12 15:22:47.009474 [INFO] mod_dialplan_xml.c:391 Processing
> FreeSWITCH->1999 in context default
> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179 Hangup
> sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1087 Session 46
> (sofia/internal/sip_1) Ended
> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1089 Close
> Channel sofia/internal/sip_1 [CS_DESTROY]
>
>
> thanks for looking at this.
>
> On Mon, Oct 12, 2009 at 10:06 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> which line is hanging up your A (inbound) leg?
>>
>> look for a blue line that says "Hangup xyz...." that matches it so i can
>> see.
>>
>> I think what is happening is you are getting early media so the bridge is
>> actually working then when nobody answers it dies but technically the bridge
>> worked.
>>
>> On Mon, Oct 12, 2009 at 9:41 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>>
>>> I think think this might be a bug, but wanted to post here instead of
>>> Jira in-case I'm overlooking a configuration variable
>>> Dialplan
>>>
>>>     <extension name="1920"><!--init agent for manual and power dial
>>> mode-->
>>>       <condition field="destination_number" expression="^1920$">
>>>         <action application="set" data="hangup_after_bridge=false"/>
>>>         <action application="bridge" data="sofia/gateway/
>>> debug.com/14159927717"/>
>>>         <action application="transfer" data="1999"/><!-- send to unable
>>> to reach any contacts-->
>>>       </condition>
>>>     </extension>
>>>
>>> API Command
>>> originate sofia/internal/sip_1%192.168.1.10 1920
>>>
>>> When the bridge to 14159927717 fails (NO_ANSWER) both calls are
>>> terminated instead of continuing on in the dial plan to exten 1999 (which in
>>> my dialplan parks the call). hangup_after_bridge however seems to work OK if
>>> someone picks up in the bridge. Is this the correct behavior? How else can I
>>> prevent the call from hanging up if a bridge fails? Thanks.
>>>
>>> I'm using 15135M
>>>
>>> --matt
>>> http://www.hellohunter.com - Predictive Dialer
>>> http://www.hellohunter.com/voice_broadcast.php - Voice Broadcasting
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From anthony.minessale at gmail.com  Mon Oct 12 08:33:01 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 12 Oct 2009 10:33:01 -0500
Subject: [Freeswitch-users] hangup_after_bridge=false not continueing
	after failed bridge
In-Reply-To: <4256bf830910120825u70e988p4d7a041c520048a@mail.gmail.com>
References: <4256bf830910120741g5facaa6cr7ed92303dfafe74b@mail.gmail.com>
	<191c3a030910120806o30911131o743aa58420b41ae4@mail.gmail.com>
	<4256bf830910120823w5fcbc3eja4afb76b60b68fda@mail.gmail.com>
	<4256bf830910120825u70e988p4d7a041c520048a@mail.gmail.com>
Message-ID: <191c3a030910120833i691b6fbfk3eecbf40e5496479@mail.gmail.com>

because the regex is on 1997 not 1999


On Mon, Oct 12, 2009 at 10:25 AM, Matthew Fong <mattdfong at gmail.com> wrote:

>     <extension name="1999"><!--DIRECT POWER-->
>       <condition field="destination_number" expression="^1997$">
>         <action application="playback"
> data="hh/hh-unable_to_connect_contact.wav"/>
>         <action application="park"/>
>       </condition>
>     </extension>
>
> my extn 1999... since it looks from the output like it's transferring, just
> don't know why it's disconnecting the call instead of playing the .wav and
> parking.
>
> On Mon, Oct 12, 2009 at 10:23 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>
>> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179 Hangup
>> sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
>>
>> might be the line..or the entire output is below....
>>
>> freeswitch at matthew-laptop> originate sofia/internal/sip_1%192.168.1.10
>> 1920
>> 2009-10-12 15:21:44.029517 [NOTICE] switch_channel.c:613 New Channel
>> sofia/internal/sip_1 [1e722934-7e94-46aa-9d62-e6ec7e7449cf]
>> 2009-10-12 15:21:44.121484 [NOTICE] sofia.c:3552 Ring-Ready
>> sofia/internal/sip_1!
>> 2009-10-12 15:21:47.285531 [NOTICE] sofia.c:3998 Channel
>> [sofia/internal/sip_1] has been answered
>> 2009-10-12 15:21:47.290996 [INFO] mod_dialplan_xml.c:391 Processing
>> FreeSWITCH->1920 in context default
>> 2009-10-12 15:21:47.293452 [NOTICE] switch_channel.c:613 New Channel
>> sofia/external/14159927717 [6b6cc440-e1d6-415a-b84b-494117e7361d]
>> 2009-10-12 15:21:47.293452 [NOTICE] switch_ivr.c:1367 Transfer
>> sofia/internal/sip_1 to XML[1920 at default]
>> API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
>> +OK 1e722934-7e94-46aa-9d62-e6ec7e7449cf
>>
>> freeswitch at matthew-laptop> 2009-10-12 15:21:47.369855 [NOTICE]
>> sofia.c:3552 Ring-Ready sofia/external/14159927717!
>> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr_originate.c:2336 Hangup
>> sofia/external/14159927717 [CS_CONSUME_MEDIA] [NO_ANSWER]
>> 2009-10-12 15:22:47.009474 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>> debug.com hc:NO_ANSWER du:0 cn:sofia/external/14159927717
>> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1087 Session 47
>> (sofia/external/14159927717) Ended
>> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1089 Close
>> Channel sofia/external/14159927717 [CS_DESTROY]
>> 2009-10-12 15:22:47.009474 [INFO] mod_dptools.c:2133 Originate Failed.
>>  Cause: NO_ANSWER
>> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr.c:1367 Transfer
>> sofia/internal/sip_1 to XML[1999 at default]
>> 2009-10-12 15:22:47.009474 [INFO] mod_dialplan_xml.c:391 Processing
>> FreeSWITCH->1999 in context default
>> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179 Hangup
>> sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
>> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1087 Session 46
>> (sofia/internal/sip_1) Ended
>> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1089 Close
>> Channel sofia/internal/sip_1 [CS_DESTROY]
>>
>>
>> thanks for looking at this.
>>
>> On Mon, Oct 12, 2009 at 10:06 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> which line is hanging up your A (inbound) leg?
>>>
>>> look for a blue line that says "Hangup xyz...." that matches it so i can
>>> see.
>>>
>>> I think what is happening is you are getting early media so the bridge is
>>> actually working then when nobody answers it dies but technically the bridge
>>> worked.
>>>
>>> On Mon, Oct 12, 2009 at 9:41 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>
>>>> I think think this might be a bug, but wanted to post here instead of
>>>> Jira in-case I'm overlooking a configuration variable
>>>> Dialplan
>>>>
>>>>     <extension name="1920"><!--init agent for manual and power dial
>>>> mode-->
>>>>       <condition field="destination_number" expression="^1920$">
>>>>         <action application="set" data="hangup_after_bridge=false"/>
>>>>         <action application="bridge" data="sofia/gateway/
>>>> debug.com/14159927717"/>
>>>>         <action application="transfer" data="1999"/><!-- send to unable
>>>> to reach any contacts-->
>>>>       </condition>
>>>>     </extension>
>>>>
>>>> API Command
>>>> originate sofia/internal/sip_1%192.168.1.10 1920
>>>>
>>>> When the bridge to 14159927717 fails (NO_ANSWER) both calls are
>>>> terminated instead of continuing on in the dial plan to exten 1999 (which in
>>>> my dialplan parks the call). hangup_after_bridge however seems to work OK if
>>>> someone picks up in the bridge. Is this the correct behavior? How else can I
>>>> prevent the call from hanging up if a bridge fails? Thanks.
>>>>
>>>> I'm using 15135M
>>>>
>>>> --matt
>>>> http://www.hellohunter.com - Predictive Dialer
>>>> http://www.hellohunter.com/voice_broadcast.php - Voice Broadcasting
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:213-799-1400
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Mon Oct 12 08:38:11 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 12 Oct 2009 10:38:11 -0500
Subject: [Freeswitch-users] Play music on hold after parking the call
In-Reply-To: <1452e2980910120036k5ab2fb36vbec031c0b0280116@mail.gmail.com>
References: <1452e2980910100237u317d6138w1d13bd3cfb250353@mail.gmail.com>
	<1452e2980910120036k5ab2fb36vbec031c0b0280116@mail.gmail.com>
Message-ID: <191c3a030910120838w2efa7168o19dc070af231b524@mail.gmail.com>

You are asking how you can do asynchronous actions park then play a sound
before park exits without enabling async mode. Think about that.

you could use new valet_parking that parks with music i guess
or tell the channel execute playback instead of park since playback an park
are almost the same thing only one is silent and one is not.


On Mon, Oct 12, 2009 at 2:36 AM, velusamy velu <velu.technical at gmail.com>wrote:

>  Any one please help me to solve the mentioned problem........
>
>
> On Sat, Oct 10, 2009 at 3:07 PM, velusamy velu <velu.technical at gmail.com>wrote:
>
>> Dear All,
>>       I am using ESL.pm module to control the dial plan application. I
>> want to play some music while executing the some external scripts. I
>> executed park after then I executed the playback the music didn't play.
>>
>>     Could any one please explain how can I solve this problem without
>> using async mode in socket application?
>>
>> Thanks,
>> Velusamy.
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mattdfong at gmail.com  Mon Oct 12 08:45:10 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Mon, 12 Oct 2009 22:45:10 +0700
Subject: [Freeswitch-users] hangup_after_bridge=false not continueing
	after failed bridge
In-Reply-To: <191c3a030910120833i691b6fbfk3eecbf40e5496479@mail.gmail.com>
References: <4256bf830910120741g5facaa6cr7ed92303dfafe74b@mail.gmail.com>
	<191c3a030910120806o30911131o743aa58420b41ae4@mail.gmail.com>
	<4256bf830910120823w5fcbc3eja4afb76b60b68fda@mail.gmail.com>
	<4256bf830910120825u70e988p4d7a041c520048a@mail.gmail.com>
	<191c3a030910120833i691b6fbfk3eecbf40e5496479@mail.gmail.com>
Message-ID: <4256bf830910120845g4f991a55m8753c772dcc8cc31@mail.gmail.com>

doh! thanks!

On Mon, Oct 12, 2009 at 10:33 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> because the regex is on 1997 not 1999
>
>
>
> On Mon, Oct 12, 2009 at 10:25 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>
>>     <extension name="1999"><!--DIRECT POWER-->
>>       <condition field="destination_number" expression="^1997$">
>>         <action application="playback"
>> data="hh/hh-unable_to_connect_contact.wav"/>
>>         <action application="park"/>
>>       </condition>
>>     </extension>
>>
>> my extn 1999... since it looks from the output like it's transferring,
>> just don't know why it's disconnecting the call instead of playing the .wav
>> and parking.
>>
>> On Mon, Oct 12, 2009 at 10:23 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>>
>>> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179
>>> Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
>>>
>>> might be the line..or the entire output is below....
>>>
>>> freeswitch at matthew-laptop> originate sofia/internal/sip_1%192.168.1.10
>>> 1920
>>> 2009-10-12 15:21:44.029517 [NOTICE] switch_channel.c:613 New Channel
>>> sofia/internal/sip_1 [1e722934-7e94-46aa-9d62-e6ec7e7449cf]
>>> 2009-10-12 15:21:44.121484 [NOTICE] sofia.c:3552 Ring-Ready
>>> sofia/internal/sip_1!
>>> 2009-10-12 15:21:47.285531 [NOTICE] sofia.c:3998 Channel
>>> [sofia/internal/sip_1] has been answered
>>> 2009-10-12 15:21:47.290996 [INFO] mod_dialplan_xml.c:391 Processing
>>> FreeSWITCH->1920 in context default
>>> 2009-10-12 15:21:47.293452 [NOTICE] switch_channel.c:613 New Channel
>>> sofia/external/14159927717 [6b6cc440-e1d6-415a-b84b-494117e7361d]
>>> 2009-10-12 15:21:47.293452 [NOTICE] switch_ivr.c:1367 Transfer
>>> sofia/internal/sip_1 to XML[1920 at default]
>>> API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
>>> +OK 1e722934-7e94-46aa-9d62-e6ec7e7449cf
>>>
>>> freeswitch at matthew-laptop> 2009-10-12 15:21:47.369855 [NOTICE]
>>> sofia.c:3552 Ring-Ready sofia/external/14159927717!
>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr_originate.c:2336 Hangup
>>> sofia/external/14159927717 [CS_CONSUME_MEDIA] [NO_ANSWER]
>>> 2009-10-12 15:22:47.009474 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>>> debug.com hc:NO_ANSWER du:0 cn:sofia/external/14159927717
>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1087 Session 47
>>> (sofia/external/14159927717) Ended
>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1089 Close
>>> Channel sofia/external/14159927717 [CS_DESTROY]
>>> 2009-10-12 15:22:47.009474 [INFO] mod_dptools.c:2133 Originate Failed.
>>>  Cause: NO_ANSWER
>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr.c:1367 Transfer
>>> sofia/internal/sip_1 to XML[1999 at default]
>>> 2009-10-12 15:22:47.009474 [INFO] mod_dialplan_xml.c:391 Processing
>>> FreeSWITCH->1999 in context default
>>> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179
>>> Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
>>> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1087 Session 46
>>> (sofia/internal/sip_1) Ended
>>> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1089 Close
>>> Channel sofia/internal/sip_1 [CS_DESTROY]
>>>
>>>
>>> thanks for looking at this.
>>>
>>> On Mon, Oct 12, 2009 at 10:06 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> which line is hanging up your A (inbound) leg?
>>>>
>>>> look for a blue line that says "Hangup xyz...." that matches it so i can
>>>> see.
>>>>
>>>> I think what is happening is you are getting early media so the bridge
>>>> is actually working then when nobody answers it dies but technically the
>>>> bridge worked.
>>>>
>>>> On Mon, Oct 12, 2009 at 9:41 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>>
>>>>> I think think this might be a bug, but wanted to post here instead of
>>>>> Jira in-case I'm overlooking a configuration variable
>>>>> Dialplan
>>>>>
>>>>>     <extension name="1920"><!--init agent for manual and power dial
>>>>> mode-->
>>>>>       <condition field="destination_number" expression="^1920$">
>>>>>         <action application="set" data="hangup_after_bridge=false"/>
>>>>>         <action application="bridge" data="sofia/gateway/
>>>>> debug.com/14159927717"/>
>>>>>         <action application="transfer" data="1999"/><!-- send to unable
>>>>> to reach any contacts-->
>>>>>       </condition>
>>>>>     </extension>
>>>>>
>>>>> API Command
>>>>> originate sofia/internal/sip_1%192.168.1.10 1920
>>>>>
>>>>> When the bridge to 14159927717 fails (NO_ANSWER) both calls are
>>>>> terminated instead of continuing on in the dial plan to exten 1999 (which in
>>>>> my dialplan parks the call). hangup_after_bridge however seems to work OK if
>>>>> someone picks up in the bridge. Is this the correct behavior? How else can I
>>>>> prevent the call from hanging up if a bridge fails? Thanks.
>>>>>
>>>>> I'm using 15135M
>>>>>
>>>>> --matt
>>>>> http://www.hellohunter.com - Predictive Dialer
>>>>> http://www.hellohunter.com/voice_broadcast.php - Voice Broadcasting
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:213-799-1400
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Mon Oct 12 09:03:50 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 12 Oct 2009 18:03:50 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
Message-ID: <65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>

hi,

can't make it...

subZero:~/freeswitch-trunk$ make mod_h323

making all mod_h323
Compiling mod_h323.cpp...
quiet_libtool: compile:  g++ -g -ggdb -I/usr/local/include/ptlib
-I/usr/local/include/openh323 -I. -DPTRACING=1 -D_REENTRANT -fno-exceptions
-I/home/tculjaga/freeswitch-trunk/src/include
-I/home/tculjaga/freeswitch-trunk/libs/libteletone/src -fPIC
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2
-D_GNU_SOURCE -DHAVE_CONFIG_H -c mod_h323.cpp  -fPIC -DPIC -o
.libs/mod_h323.o
In file included from /usr/local/include/openh323/h323.h:493,
                 from mod_h323.h:8,
                 from mod_h323.cpp:3:
/usr/local/include/openh323/h323ep.h: In member function ???virtual void
NATFactoryStartup::OnShutdown()???:
/usr/local/include/openh323/h323ep.h:2731: error: ???NatFactory??? has not
been declared
make[4]: *** [mod_h323.lo] Error 1
make[3]: *** [all] Error 1
make[2]: *** [mod_h323-all] Error 1
make[1]: *** [mod_h323] Error 2
make: *** [mod_h323] Error 2



what exact ptlib and h323plus versions did you use? .. can you send us a
link so we can use the exact ones.


T.

2009/10/12 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-12 09:43 -0500, Brian West wrote
> freeswitch-users at lists.freeswit...:
>
> BW>We can host this in our SVN if you wish?
>
> If in fs svn i think yes. But i think may be little time later?
> i don't known is it builds on trunk because i develop it on 1.0.4.
>
> BW>/b
> BW>
> BW>On Oct 12, 2009, at 8:31 AM, Georgiewskiy Yuriy wrote:
> BW>
> BW>> ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/ alfa code, but
> seems
> BW>> it work, but should be buggy,
> BW>> to build need libpt 2.6.5 and h323plus cvs version, i test it now on
> fs
> BW>> 1.0.4.
> BW>
>
> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129-RIPE
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From maciej.aniserowicz at gmail.com  Mon Oct 12 09:04:52 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Mon, 12 Oct 2009 09:04:52 -0700 (PDT)
Subject: [Freeswitch-users] Bad sound quality while eavesdropping
In-Reply-To: <59F3CD44-5FEA-403C-98BE-EEE49EC3815B@freeswitch.org>
References: <41A44DD027064988A914974405788C2E@procent>
	<191c3a030910050731m2d74979ep4598e5a1945d58ae@mail.gmail.com>
	<1254901192035-3780245.post@n2.nabble.com>
	<8437F5BC-7AFF-4A74-B8CD-C5B8219021F6@jerris.com>
	<1255008427639-3788019.post@n2.nabble.com>
	<191c3a030910080823g79c7c596x1cd887e1538ce2e1@mail.gmail.com>
	<1255169044209-3799274.post@n2.nabble.com>
	<CF8C1791-24CD-4132-83B5-7F83B5BDCCC3@jerris.com>
	<1255337256919-3806786.post@n2.nabble.com>
	<59F3CD44-5FEA-403C-98BE-EEE49EC3815B@freeswitch.org>
Message-ID: <1255363492193-3808860.post@n2.nabble.com>


Nope, I wanted to make sure that this is indeed a bug. I opened an issue in
JIRA before regarding some other matter and it turned out to be my mistake,
so I decided to try mailing list first this time.
MA



Brian West wrote:
> 
> Did you open a jira and attach all the info?
> 
> /b
> 
> On Oct 12, 2009, at 3:47 AM, Maciej Aniserowicz wrote:
> 
>> Yes, I confirmed that with Wireshark (filter "rtp and ip.src ==  
>> <device ip>). RTP packets are sent every 20ms.
>>
>> MAniserowicz
>>
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3808860.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From brian at freeswitch.org  Mon Oct 12 09:44:21 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Oct 2009 11:44:21 -0500
Subject: [Freeswitch-users] Re corded file as voicemail.
In-Reply-To: <23f91030910112315h5ea79b81ue55381a3caff0173@mail.gmail.com>
References: <25850181.post@talk.nabble.com>
	<BEDBBF51-2D48-4CDE-A467-11D5BBB458E1@freeswitch.org>
	<23f91030910112315h5ea79b81ue55381a3caff0173@mail.gmail.com>
Message-ID: <BA9A12F7-EB74-485D-8612-12E448621257@freeswitch.org>

Fixed... svn up.

/b

On Oct 12, 2009, at 1:15 AM, Seven Du wrote:

> http://jira.freeswitch.org/browse/MODCODEC-15
>
> Is it ok I assigned to you ?
>
> Thanks.

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From ryannyl at gmail.com  Mon Oct 12 10:18:51 2009
From: ryannyl at gmail.com (Ryanny Lin)
Date: Tue, 13 Oct 2009 01:18:51 +0800
Subject: [Freeswitch-users] Can freeswitch forward an un-implemented SIP
	method?(like a SIP proxy server)
Message-ID: <4bfcac7e0910121018m39e1f820qb39b2eb13348da9c@mail.gmail.com>

Hi, all

I used sipp to test to forward an un-implemented SIP method, but I got 501
Not Implemented.
Can freeswitch be a SIP proxy server?

SIP/2.0 501 Not Implemented
Via: SIP/2.0/UDP 10.10.59.161:5061
From: <sip:1000 at 10.10.59.69:5060>
To: <sip:1001 at 10.10.59.69:5060>;tag=gBpay50SD2SUD
Call-ID: 10-27119 at 10.10.59.161
CSeq: 3 TESTA
User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Content-Length: 0


It looks like freeswitch always processes SIP messages.
-- 
Sincerely regards,
Ryanny
Oct. 13
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From msc at freeswitch.org  Mon Oct 12 10:21:44 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 10:21:44 -0700
Subject: [Freeswitch-users] REMINDER: Weekly FreeSWITCH Conference
	Scheduled for October 9, 11AM CST (GMT -6)
In-Reply-To: <86a32abc0910111440n184b882fwee29736bb15850ef@mail.gmail.com>
References: <87f2f3b90910081202m5dd3f567v240541dd2009c9a8@mail.gmail.com>
	<86a32abc0910111440n184b882fwee29736bb15850ef@mail.gmail.com>
Message-ID: <87f2f3b90910121021q59a99c5cgafa3f718436e83e@mail.gmail.com>

On Sun, Oct 11, 2009 at 2:40 PM, Diego Viola <diego.viola at gmail.com> wrote:

> I'd like to add this for the next weekly conference.
>
> I have added a few events to the event list, as you can see here:
>
> http://wiki.freeswitch.org/wiki/Event_list
>
> But I need more help from the community to complete that and add content to
> the events, etc. So if you can add that for the next weekly conference it
> would be really nice :).
>
> I added this to the Oct 16th agenda. I'll be sending out the reminder email
today.
-MC
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From bottleman at icf.org.ru  Mon Oct 12 10:29:46 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Mon, 12 Oct 2009 21:29:46 +0400 (MSD)
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910081333310.25302@srv.icf.org.ru>
	<191c3a030910080843i3b8d43f4ga0f6b9927f96d2e2@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>

On 2009-10-12 18:03 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

i already write this, ptlib 2.6.5, tou can find link to it on oplalvoip.org, h32plus latest CVS version,
you can find it on www.h323plus.org.

TC>hi,
TC>
TC>can't make it...
TC>
TC>subZero:~/freeswitch-trunk$ make mod_h323
TC>
TC>making all mod_h323
TC>Compiling mod_h323.cpp...
TC>quiet_libtool: compile:  g++ -g -ggdb -I/usr/local/include/ptlib
TC>-I/usr/local/include/openh323 -I. -DPTRACING=1 -D_REENTRANT -fno-exceptions
TC>-I/home/tculjaga/freeswitch-trunk/src/include
TC>-I/home/tculjaga/freeswitch-trunk/libs/libteletone/src -fPIC
TC>-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2
TC>-D_GNU_SOURCE -DHAVE_CONFIG_H -c mod_h323.cpp  -fPIC -DPIC -o
TC>.libs/mod_h323.o
TC>In file included from /usr/local/include/openh323/h323.h:493,
TC>                 from mod_h323.h:8,
TC>                 from mod_h323.cpp:3:
TC>/usr/local/include/openh323/h323ep.h: In member function ??????virtual void
TC>NATFactoryStartup::OnShutdown()??????:
TC>/usr/local/include/openh323/h323ep.h:2731: error: ??????NatFactory?????? has not
TC>been declared
TC>make[4]: *** [mod_h323.lo] Error 1
TC>make[3]: *** [all] Error 1
TC>make[2]: *** [mod_h323-all] Error 1
TC>make[1]: *** [mod_h323] Error 2
TC>make: *** [mod_h323] Error 2
TC>
TC>
TC>
TC>what exact ptlib and h323plus versions did you use? .. can you send us a
TC>link so we can use the exact ones.
TC>
TC>
TC>T.
TC>
TC>2009/10/12 Georgiewskiy Yuriy <bottleman at icf.org.ru>
TC>
TC>> On 2009-10-12 09:43 -0500, Brian West wrote
TC>> freeswitch-users at lists.freeswit...:
TC>>
TC>> BW>We can host this in our SVN if you wish?
TC>>
TC>> If in fs svn i think yes. But i think may be little time later?
TC>> i don't known is it builds on trunk because i develop it on 1.0.4.
TC>>
TC>> BW>/b
TC>> BW>
TC>> BW>On Oct 12, 2009, at 8:31 AM, Georgiewskiy Yuriy wrote:
TC>> BW>
TC>> BW>> ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/ alfa code, but
TC>> seems
TC>> BW>> it work, but should be buggy,
TC>> BW>> to build need libpt 2.6.5 and h323plus cvs version, i test it now on
TC>> fs
TC>> BW>> 1.0.4.
TC>> BW>
TC>>
TC>> C ?????????                       With Best Regards
TC>> ???????????? ????.                Georgiewskiy Yuriy
TC>> +7 4872 711666                    +7 4872 711666
TC>> ???? +7 4872 711143               fax +7 4872 711143
TC>> ???????? ??? "?? ?? ??????"       IT Service Ltd
TC>> http://nkoort.ru                  http://nkoort.ru
TC>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
TC>> YG129-RIPE                        YG129-RIPE
TC>>
TC>> _______________________________________________
TC>> FreeSWITCH-users mailing list
TC>> FreeSWITCH-users at lists.freeswitch.org
TC>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> http://www.freeswitch.org
TC>>
TC>>
TC>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From sicfslist at gmail.com  Mon Oct 12 10:31:24 2009
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Mon, 12 Oct 2009 12:31:24 -0500
Subject: [Freeswitch-users] Can freeswitch forward an un-implemented SIP
	method?(like a SIP proxy server)
In-Reply-To: <4bfcac7e0910121018m39e1f820qb39b2eb13348da9c@mail.gmail.com>
References: <4bfcac7e0910121018m39e1f820qb39b2eb13348da9c@mail.gmail.com>
Message-ID: <35b355e90910121031k2a8aa6ah2ea5bc063e6a59de@mail.gmail.com>

This is because FS is a B2BUA ... not a proxy.  You should consider
OpenSER/SIPS/Kaemillio for this type of application.

SDR
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From msc at freeswitch.org  Mon Oct 12 10:35:08 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 10:35:08 -0700
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <5162D0B7-357C-4D0E-9913-46564A8C0B1D@freeswitch.org>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<87f2f3b90910080958i1f5cfccax621fd12100a05e3e@mail.gmail.com>
	<903da5680910081201p427a8358n768edb61a96674a4@mail.gmail.com>
	<87f2f3b90910081538l6e4e4b7dv3aedb2c4d1af58bc@mail.gmail.com>
	<903da5680910082238x27bff476s8bea0e74ab516a8d@mail.gmail.com>
	<87f2f3b90910082308i7f6144d9kc887596be54dd609@mail.gmail.com>
	<903da5680910082341r485092vb0cd10c43c4a2534@mail.gmail.com>
	<219716B3-C355-4128-A333-1459AC63C364@jerris.com>
	<86a32abc0910111506m4e6d2740tbd3ac5fe0f59178@mail.gmail.com>
	<5162D0B7-357C-4D0E-9913-46564A8C0B1D@freeswitch.org>
Message-ID: <87f2f3b90910121035v5cfce2bel7d32da1671e325b6@mail.gmail.com>

On Sun, Oct 11, 2009 at 4:49 PM, Brian West <brian at freeswitch.org> wrote:

> I have tried to police the wiki when things like this appear.. its one
> thing to crack a joke in fun from time to time... but to put stuff
> like that on the wiki isn't acceptable.
>
> /b
>
>
It seemed appropriate to do so, therefore I added a small snippet on the
documentation guidelines:
http://wiki.freeswitch.org/wiki/Documentation_guidelines#Keep_It_Professional

-MC
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From mattdfong at gmail.com  Mon Oct 12 10:42:15 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Tue, 13 Oct 2009 00:42:15 +0700
Subject: [Freeswitch-users] hangup_after_bridge=false not continueing
	after failed bridge
In-Reply-To: <4256bf830910120845g4f991a55m8753c772dcc8cc31@mail.gmail.com>
References: <4256bf830910120741g5facaa6cr7ed92303dfafe74b@mail.gmail.com>
	<191c3a030910120806o30911131o743aa58420b41ae4@mail.gmail.com>
	<4256bf830910120823w5fcbc3eja4afb76b60b68fda@mail.gmail.com>
	<4256bf830910120825u70e988p4d7a041c520048a@mail.gmail.com>
	<191c3a030910120833i691b6fbfk3eecbf40e5496479@mail.gmail.com>
	<4256bf830910120845g4f991a55m8753c772dcc8cc31@mail.gmail.com>
Message-ID: <4256bf830910121042o50ee79edife2243682bc9d3e2@mail.gmail.com>

when I add a leg_timeout, I get an ALLOTTED_TIMEOUT from my failed bridge...
when an ALLOTTED_TIMEOUT is received, the hangup_after_bridge=false is not
recognized (I think). Is there anyway to get an alloted_timeout to continue
after bridge (failure)?

revised dialplan & cmd output

    <extension name="1920"><!--DEBUG-->
      <condition field="destination_number" expression="^1920$">
        <action application="set" data="hangup_after_bridge=false"/>
        <action application="bridge" data="{leg_timeout=10}sofia/gateway/
debug.com/14159927717"/>
        <action application="transfer" data="1999"/><!-- send to unable to
reach any contacts-->
      </condition>
    </extension>

freeswitch at matthew-laptop> originate sofia/internal/sip_1%192.168.1.10 1920
2009-10-12 17:39:22.237622 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/sip_1 [1d58fb59-c7f9-4908-b612-5bd1c12083cc]
2009-10-12 17:39:22.313524 [NOTICE] sofia.c:3552 Ring-Ready
sofia/internal/sip_1!
2009-10-12 17:39:25.142400 [NOTICE] sofia.c:3998 Channel
[sofia/internal/sip_1] has been answered
2009-10-12 17:39:25.146259 [INFO] mod_dialplan_xml.c:391 Processing
FreeSWITCH->1920 in context default
2009-10-12 17:39:25.146259 [NOTICE] switch_channel.c:613 New Channel
sofia/external/14159927717 [38cb7046-0c0d-47ef-94b5-bf8ccf35d185]
2009-10-12 17:39:25.152026 [NOTICE] switch_ivr.c:1367 Transfer
sofia/internal/sip_1 to XML[1920 at default]
API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
+OK 1d58fb59-c7f9-4908-b612-5bd1c12083cc

freeswitch at matthew-laptop> 2009-10-12 17:39:25.217629 [NOTICE] sofia.c:3552
Ring-Ready sofia/external/14159927717!
2009-10-12 17:39:36.017513 [NOTICE] switch_ivr_originate.c:297 Hangup
sofia/external/14159927717 [CS_CONSUME_MEDIA] [ALLOTTED_TIMEOUT]
2009-10-12 17:39:36.017513 [INFO] switch_cpp.cpp:1116 PCHangup
gw:debug.comhc:ALLOTTED_TIMEOUT du:0 cn:sofia/external/14159927717
2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1087 Session 12
(sofia/external/14159927717) Ended
2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1089 Close Channel
sofia/external/14159927717 [CS_DESTROY]
2009-10-12 17:39:36.037520 [INFO] mod_dptools.c:2133 Originate Failed.
 Cause: ALLOTTED_TIMEOUT
2009-10-12 17:39:36.037520 [NOTICE] mod_dptools.c:2166 Hangup
sofia/internal/sip_1 [CS_EXECUTE] [ALLOTTED_TIMEOUT]
2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1087 Session 11
(sofia/internal/sip_1) Ended
2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1089 Close Channel
sofia/internal/sip_1 [CS_DESTROY]

thanks.

--matt


On Mon, Oct 12, 2009 at 10:45 PM, Matthew Fong <mattdfong at gmail.com> wrote:

> doh! thanks!
>
>
> On Mon, Oct 12, 2009 at 10:33 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> because the regex is on 1997 not 1999
>>
>>
>>
>> On Mon, Oct 12, 2009 at 10:25 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>>
>>>     <extension name="1999"><!--DIRECT POWER-->
>>>       <condition field="destination_number" expression="^1997$">
>>>         <action application="playback"
>>> data="hh/hh-unable_to_connect_contact.wav"/>
>>>         <action application="park"/>
>>>       </condition>
>>>     </extension>
>>>
>>> my extn 1999... since it looks from the output like it's transferring,
>>> just don't know why it's disconnecting the call instead of playing the .wav
>>> and parking.
>>>
>>> On Mon, Oct 12, 2009 at 10:23 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>
>>>> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179
>>>> Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
>>>>
>>>> might be the line..or the entire output is below....
>>>>
>>>> freeswitch at matthew-laptop> originate sofia/internal/sip_1%192.168.1.10
>>>> 1920
>>>> 2009-10-12 15:21:44.029517 [NOTICE] switch_channel.c:613 New Channel
>>>> sofia/internal/sip_1 [1e722934-7e94-46aa-9d62-e6ec7e7449cf]
>>>> 2009-10-12 15:21:44.121484 [NOTICE] sofia.c:3552 Ring-Ready
>>>> sofia/internal/sip_1!
>>>> 2009-10-12 15:21:47.285531 [NOTICE] sofia.c:3998 Channel
>>>> [sofia/internal/sip_1] has been answered
>>>> 2009-10-12 15:21:47.290996 [INFO] mod_dialplan_xml.c:391 Processing
>>>> FreeSWITCH->1920 in context default
>>>> 2009-10-12 15:21:47.293452 [NOTICE] switch_channel.c:613 New Channel
>>>> sofia/external/14159927717 [6b6cc440-e1d6-415a-b84b-494117e7361d]
>>>> 2009-10-12 15:21:47.293452 [NOTICE] switch_ivr.c:1367 Transfer
>>>> sofia/internal/sip_1 to XML[1920 at default]
>>>> API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
>>>> +OK 1e722934-7e94-46aa-9d62-e6ec7e7449cf
>>>>
>>>> freeswitch at matthew-laptop> 2009-10-12 15:21:47.369855 [NOTICE]
>>>> sofia.c:3552 Ring-Ready sofia/external/14159927717!
>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr_originate.c:2336 Hangup
>>>> sofia/external/14159927717 [CS_CONSUME_MEDIA] [NO_ANSWER]
>>>> 2009-10-12 15:22:47.009474 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>>>> debug.com hc:NO_ANSWER du:0 cn:sofia/external/14159927717
>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1087 Session
>>>> 47 (sofia/external/14159927717) Ended
>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1089 Close
>>>> Channel sofia/external/14159927717 [CS_DESTROY]
>>>> 2009-10-12 15:22:47.009474 [INFO] mod_dptools.c:2133 Originate Failed.
>>>>  Cause: NO_ANSWER
>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr.c:1367 Transfer
>>>> sofia/internal/sip_1 to XML[1999 at default]
>>>> 2009-10-12 15:22:47.009474 [INFO] mod_dialplan_xml.c:391 Processing
>>>> FreeSWITCH->1999 in context default
>>>> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179
>>>> Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
>>>> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1087 Session
>>>> 46 (sofia/internal/sip_1) Ended
>>>> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1089 Close
>>>> Channel sofia/internal/sip_1 [CS_DESTROY]
>>>>
>>>>
>>>> thanks for looking at this.
>>>>
>>>> On Mon, Oct 12, 2009 at 10:06 PM, Anthony Minessale <
>>>> anthony.minessale at gmail.com> wrote:
>>>>
>>>>> which line is hanging up your A (inbound) leg?
>>>>>
>>>>> look for a blue line that says "Hangup xyz...." that matches it so i
>>>>> can see.
>>>>>
>>>>> I think what is happening is you are getting early media so the bridge
>>>>> is actually working then when nobody answers it dies but technically the
>>>>> bridge worked.
>>>>>
>>>>> On Mon, Oct 12, 2009 at 9:41 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>>>
>>>>>> I think think this might be a bug, but wanted to post here instead of
>>>>>> Jira in-case I'm overlooking a configuration variable
>>>>>> Dialplan
>>>>>>
>>>>>>     <extension name="1920"><!--init agent for manual and power dial
>>>>>> mode-->
>>>>>>       <condition field="destination_number" expression="^1920$">
>>>>>>         <action application="set" data="hangup_after_bridge=false"/>
>>>>>>         <action application="bridge" data="sofia/gateway/
>>>>>> debug.com/14159927717"/>
>>>>>>         <action application="transfer" data="1999"/><!-- send to
>>>>>> unable to reach any contacts-->
>>>>>>       </condition>
>>>>>>     </extension>
>>>>>>
>>>>>> API Command
>>>>>> originate sofia/internal/sip_1%192.168.1.10 1920
>>>>>>
>>>>>> When the bridge to 14159927717 fails (NO_ANSWER) both calls are
>>>>>> terminated instead of continuing on in the dial plan to exten 1999 (which in
>>>>>> my dialplan parks the call). hangup_after_bridge however seems to work OK if
>>>>>> someone picks up in the bridge. Is this the correct behavior? How else can I
>>>>>> prevent the call from hanging up if a bridge fails? Thanks.
>>>>>>
>>>>>> I'm using 15135M
>>>>>>
>>>>>> --matt
>>>>>> http://www.hellohunter.com - Predictive Dialer
>>>>>> http://www.hellohunter.com/voice_broadcast.php - Voice Broadcasting
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>> iax:guest at conference.freeswitch.org/888
>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>> pstn:213-799-1400
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From msc at freeswitch.org  Mon Oct 12 11:00:18 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 11:00:18 -0700
Subject: [Freeswitch-users] Question about fax tone detection
In-Reply-To: <1255345270255-3807298.post@n2.nabble.com>
References: <1255345270255-3807298.post@n2.nabble.com>
Message-ID: <87f2f3b90910121100l40a1b22hff33df032411cf31@mail.gmail.com>

On Mon, Oct 12, 2009 at 4:01 AM, homqua <ngay01042005 at gmail.com> wrote:

>
> Hi,
> I have implemented the solution for tone detection in wiki, and also answer
> the channel before detecting the tone:
> <condition field="destination_number" expression="^(055138419992)$">
>    <action application="answer"/>
>    <action application="tone_detect" data="fax 1100 r +15000 transfer fax
> XML default"/>
>
>
> <extension name="fax">
>      <condition field="destination_number" expression="^fax$">
>        <action application="answer" />
>        <action application="sleep" data="1000"/>
>        <action application="rxfax"
>
> data="/usr/local/freeswitch/storage/fax/${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}.tiff"/>
>    <action application="set" data="fax_mode=recv"/>
>        <action application="hangup"/>
>      </condition>
>    </extension>
>
>
> But FS cannot recognize the tone, and therefore cannot move to fax
> extension.  Below are the error in FS:
>
> 2009-10-12 10:57:16.702287 [NOTICE] switch_channel.c:602 New Channel
> sofia/external/anonymous at anonymous.invalid
> [c431f0a3-9231-4724-ba39-9e4ef7edfca2]
> 2009-10-12 10:57:16.703413 [INFO] mod_dialplan_xml.c:315 Processing
> Anonymous->055138419992 in context public
> 2009-10-12 10:57:16.719288 [NOTICE] switch_ivr.c:1349 Transfer
> sofia/external/anonymous at anonymous.invalid to XML[055138419992 at default]
> 2009-10-12 10:57:16.719288 [INFO] mod_dialplan_xml.c:315 Processing
> Anonymous->055138419992 in context default
> 2009-10-12 10:57:16.722289 [NOTICE] mod_dptools.c:649 Channel
> [sofia/external/anonymous at anonymous.invalid] has been answered
> 2009-10-12 10:57:16.722289 [NOTICE] mod_dptools.c:1324 Enabling tone
> detection 'fax' '1100'
> 2009-10-12 10:57:16.723302 [NOTICE] switch_core_state_machine.c:179 Hangup
> sofia/external/anonymous at anonymous.invalid [CS_EXECUTE] [NORMAL_CLEARING]
> 2009-10-12 10:57:16.740285 [NOTICE] switch_core_session.c:1086 Session 1
> (sofia/external/anonymous at anonymous.invalid) Ended
> 2009-10-12 10:57:16.740285 [NOTICE] switch_core_session.c:1088 Close
> Channel
> sofia/external/anonymous at anonymous.invalid [CS_DESTROY]
>
> And the trace for SIP messages:  http://pastebin.com/m4e47e7d9
>
> If anyone has any idea, tell me please.
> Thanks.
>

I think the trouble here is that you don't have anything else in the
dialplan after the tone_detect. The tone_detect app is non-block, which
means that it doesn't sit there and wait for a tone. If you want the
dialplan to sit and wait then do a sleep app after your tone_detect. The
other question I would have is this: what happens if the incoming call is
not a fax? What do you want to do then?

-MC
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From msc at freeswitch.org  Mon Oct 12 11:11:24 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 11:11:24 -0700
Subject: [Freeswitch-users] hangup_after_bridge=false not continueing
	after failed bridge
In-Reply-To: <4256bf830910121042o50ee79edife2243682bc9d3e2@mail.gmail.com>
References: <4256bf830910120741g5facaa6cr7ed92303dfafe74b@mail.gmail.com>
	<191c3a030910120806o30911131o743aa58420b41ae4@mail.gmail.com>
	<4256bf830910120823w5fcbc3eja4afb76b60b68fda@mail.gmail.com>
	<4256bf830910120825u70e988p4d7a041c520048a@mail.gmail.com>
	<191c3a030910120833i691b6fbfk3eecbf40e5496479@mail.gmail.com>
	<4256bf830910120845g4f991a55m8753c772dcc8cc31@mail.gmail.com>
	<4256bf830910121042o50ee79edife2243682bc9d3e2@mail.gmail.com>
Message-ID: <87f2f3b90910121111m54cf2586pfca0d867307b1e86@mail.gmail.com>

On Mon, Oct 12, 2009 at 10:42 AM, Matthew Fong <mattdfong at gmail.com> wrote:

> when I add a leg_timeout, I get an ALLOTTED_TIMEOUT from my failed
> bridge...
> when an ALLOTTED_TIMEOUT is received, the hangup_after_bridge=false is not
> recognized (I think). Is there anyway to get an alloted_timeout to continue
> after bridge (failure)?
>

Try it with ignore_early_media=true and see if it's the early media that's
tripping you up.
-MC


>
> revised dialplan & cmd output
>
>     <extension name="1920"><!--DEBUG-->
>       <condition field="destination_number" expression="^1920$">
>         <action application="set" data="hangup_after_bridge=false"/>
>         <action application="bridge" data="{leg_timeout=10}sofia/gateway/
> debug.com/14159927717"/>
>         <action application="transfer" data="1999"/><!-- send to unable to
> reach any contacts-->
>       </condition>
>     </extension>
>
> freeswitch at matthew-laptop> originate sofia/internal/sip_1%192.168.1.10
> 1920
> 2009-10-12 17:39:22.237622 [NOTICE] switch_channel.c:613 New Channel
> sofia/internal/sip_1 [1d58fb59-c7f9-4908-b612-5bd1c12083cc]
> 2009-10-12 17:39:22.313524 [NOTICE] sofia.c:3552 Ring-Ready
> sofia/internal/sip_1!
> 2009-10-12 17:39:25.142400 [NOTICE] sofia.c:3998 Channel
> [sofia/internal/sip_1] has been answered
> 2009-10-12 17:39:25.146259 [INFO] mod_dialplan_xml.c:391 Processing
> FreeSWITCH->1920 in context default
> 2009-10-12 17:39:25.146259 [NOTICE] switch_channel.c:613 New Channel
> sofia/external/14159927717 [38cb7046-0c0d-47ef-94b5-bf8ccf35d185]
> 2009-10-12 17:39:25.152026 [NOTICE] switch_ivr.c:1367 Transfer
> sofia/internal/sip_1 to XML[1920 at default]
> API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
> +OK 1d58fb59-c7f9-4908-b612-5bd1c12083cc
>
> freeswitch at matthew-laptop> 2009-10-12 17:39:25.217629 [NOTICE]
> sofia.c:3552 Ring-Ready sofia/external/14159927717!
> 2009-10-12 17:39:36.017513 [NOTICE] switch_ivr_originate.c:297 Hangup
> sofia/external/14159927717 [CS_CONSUME_MEDIA] [ALLOTTED_TIMEOUT]
> 2009-10-12 17:39:36.017513 [INFO] switch_cpp.cpp:1116 PCHangup gw:
> debug.com hc:ALLOTTED_TIMEOUT du:0 cn:sofia/external/14159927717
> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1087 Session 12
> (sofia/external/14159927717) Ended
> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1089 Close
> Channel sofia/external/14159927717 [CS_DESTROY]
> 2009-10-12 17:39:36.037520 [INFO] mod_dptools.c:2133 Originate Failed.
>  Cause: ALLOTTED_TIMEOUT
> 2009-10-12 17:39:36.037520 [NOTICE] mod_dptools.c:2166 Hangup
> sofia/internal/sip_1 [CS_EXECUTE] [ALLOTTED_TIMEOUT]
> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1087 Session 11
> (sofia/internal/sip_1) Ended
> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1089 Close
> Channel sofia/internal/sip_1 [CS_DESTROY]
>
> thanks.
>
> --matt
>
>
> On Mon, Oct 12, 2009 at 10:45 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>
>> doh! thanks!
>>
>>
>> On Mon, Oct 12, 2009 at 10:33 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> because the regex is on 1997 not 1999
>>>
>>>
>>>
>>> On Mon, Oct 12, 2009 at 10:25 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>
>>>>     <extension name="1999"><!--DIRECT POWER-->
>>>>       <condition field="destination_number" expression="^1997$">
>>>>         <action application="playback"
>>>> data="hh/hh-unable_to_connect_contact.wav"/>
>>>>         <action application="park"/>
>>>>       </condition>
>>>>     </extension>
>>>>
>>>> my extn 1999... since it looks from the output like it's transferring,
>>>> just don't know why it's disconnecting the call instead of playing the .wav
>>>> and parking.
>>>>
>>>> On Mon, Oct 12, 2009 at 10:23 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>>
>>>>> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179
>>>>> Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
>>>>>
>>>>> might be the line..or the entire output is below....
>>>>>
>>>>> freeswitch at matthew-laptop> originate sofia/internal/sip_1%192.168.1.10
>>>>> 1920
>>>>> 2009-10-12 15:21:44.029517 [NOTICE] switch_channel.c:613 New Channel
>>>>> sofia/internal/sip_1 [1e722934-7e94-46aa-9d62-e6ec7e7449cf]
>>>>> 2009-10-12 15:21:44.121484 [NOTICE] sofia.c:3552 Ring-Ready
>>>>> sofia/internal/sip_1!
>>>>> 2009-10-12 15:21:47.285531 [NOTICE] sofia.c:3998 Channel
>>>>> [sofia/internal/sip_1] has been answered
>>>>> 2009-10-12 15:21:47.290996 [INFO] mod_dialplan_xml.c:391 Processing
>>>>> FreeSWITCH->1920 in context default
>>>>> 2009-10-12 15:21:47.293452 [NOTICE] switch_channel.c:613 New Channel
>>>>> sofia/external/14159927717 [6b6cc440-e1d6-415a-b84b-494117e7361d]
>>>>> 2009-10-12 15:21:47.293452 [NOTICE] switch_ivr.c:1367 Transfer
>>>>> sofia/internal/sip_1 to XML[1920 at default]
>>>>> API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
>>>>> +OK 1e722934-7e94-46aa-9d62-e6ec7e7449cf
>>>>>
>>>>> freeswitch at matthew-laptop> 2009-10-12 15:21:47.369855 [NOTICE]
>>>>> sofia.c:3552 Ring-Ready sofia/external/14159927717!
>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr_originate.c:2336 Hangup
>>>>> sofia/external/14159927717 [CS_CONSUME_MEDIA] [NO_ANSWER]
>>>>> 2009-10-12 15:22:47.009474 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>>>>> debug.com hc:NO_ANSWER du:0 cn:sofia/external/14159927717
>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1087 Session
>>>>> 47 (sofia/external/14159927717) Ended
>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1089 Close
>>>>> Channel sofia/external/14159927717 [CS_DESTROY]
>>>>> 2009-10-12 15:22:47.009474 [INFO] mod_dptools.c:2133 Originate Failed.
>>>>>  Cause: NO_ANSWER
>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr.c:1367 Transfer
>>>>> sofia/internal/sip_1 to XML[1999 at default]
>>>>> 2009-10-12 15:22:47.009474 [INFO] mod_dialplan_xml.c:391 Processing
>>>>> FreeSWITCH->1999 in context default
>>>>> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179
>>>>> Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
>>>>> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1087 Session
>>>>> 46 (sofia/internal/sip_1) Ended
>>>>> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1089 Close
>>>>> Channel sofia/internal/sip_1 [CS_DESTROY]
>>>>>
>>>>>
>>>>> thanks for looking at this.
>>>>>
>>>>> On Mon, Oct 12, 2009 at 10:06 PM, Anthony Minessale <
>>>>> anthony.minessale at gmail.com> wrote:
>>>>>
>>>>>> which line is hanging up your A (inbound) leg?
>>>>>>
>>>>>> look for a blue line that says "Hangup xyz...." that matches it so i
>>>>>> can see.
>>>>>>
>>>>>> I think what is happening is you are getting early media so the bridge
>>>>>> is actually working then when nobody answers it dies but technically the
>>>>>> bridge worked.
>>>>>>
>>>>>> On Mon, Oct 12, 2009 at 9:41 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>>>>
>>>>>>> I think think this might be a bug, but wanted to post here instead of
>>>>>>> Jira in-case I'm overlooking a configuration variable
>>>>>>> Dialplan
>>>>>>>
>>>>>>>     <extension name="1920"><!--init agent for manual and power dial
>>>>>>> mode-->
>>>>>>>       <condition field="destination_number" expression="^1920$">
>>>>>>>         <action application="set" data="hangup_after_bridge=false"/>
>>>>>>>         <action application="bridge" data="sofia/gateway/
>>>>>>> debug.com/14159927717"/>
>>>>>>>         <action application="transfer" data="1999"/><!-- send to
>>>>>>> unable to reach any contacts-->
>>>>>>>       </condition>
>>>>>>>     </extension>
>>>>>>>
>>>>>>> API Command
>>>>>>> originate sofia/internal/sip_1%192.168.1.10 1920
>>>>>>>
>>>>>>> When the bridge to 14159927717 fails (NO_ANSWER) both calls are
>>>>>>> terminated instead of continuing on in the dial plan to exten 1999 (which in
>>>>>>> my dialplan parks the call). hangup_after_bridge however seems to work OK if
>>>>>>> someone picks up in the bridge. Is this the correct behavior? How else can I
>>>>>>> prevent the call from hanging up if a bridge fails? Thanks.
>>>>>>>
>>>>>>> I'm using 15135M
>>>>>>>
>>>>>>> --matt
>>>>>>> http://www.hellohunter.com - Predictive Dialer
>>>>>>> http://www.hellohunter.com/voice_broadcast.php - Voice Broadcasting
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Anthony Minessale II
>>>>>>
>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>> ClueCon http://www.cluecon.com/
>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>
>>>>>> AIM: anthm
>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>
>>>>>> FreeSWITCH Developer Conference
>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>> pstn:213-799-1400
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:213-799-1400
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mattdfong at gmail.com  Mon Oct 12 11:26:32 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Tue, 13 Oct 2009 01:26:32 +0700
Subject: [Freeswitch-users] hangup_after_bridge=false not continueing
	after failed bridge
In-Reply-To: <87f2f3b90910121111m54cf2586pfca0d867307b1e86@mail.gmail.com>
References: <4256bf830910120741g5facaa6cr7ed92303dfafe74b@mail.gmail.com>
	<191c3a030910120806o30911131o743aa58420b41ae4@mail.gmail.com>
	<4256bf830910120823w5fcbc3eja4afb76b60b68fda@mail.gmail.com>
	<4256bf830910120825u70e988p4d7a041c520048a@mail.gmail.com>
	<191c3a030910120833i691b6fbfk3eecbf40e5496479@mail.gmail.com>
	<4256bf830910120845g4f991a55m8753c772dcc8cc31@mail.gmail.com>
	<4256bf830910121042o50ee79edife2243682bc9d3e2@mail.gmail.com>
	<87f2f3b90910121111m54cf2586pfca0d867307b1e86@mail.gmail.com>
Message-ID: <4256bf830910121126l6f4326fco9ee3a83f51820742@mail.gmail.com>

still no luck...

    <extension name="1920"><!--DEBUG-->
      <condition field="destination_number" expression="^1920$">
        <action application="set" data="hangup_after_bridge=false"/>
        <action application="bridge"
data="{leg_timeout=10,ignore_early_media=true}sofia/gateway/
debug.com/14159927717"/>
        <action application="transfer" data="1999"/><!-- send to unable to
reach any contacts-->
      </condition>
    </extension>


freeswitch at matthew-laptop> originate sofia/internal/sip_1%192.168.1.10 1920
2009-10-12 18:25:44.345480 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/sip_1 [3fc6efb2-e4fa-454a-abb7-ebe39da748f5]
2009-10-12 18:25:44.489480 [NOTICE] sofia.c:3552 Ring-Ready
sofia/internal/sip_1!
2009-10-12 18:25:46.601509 [NOTICE] sofia.c:3998 Channel
[sofia/internal/sip_1] has been answered
2009-10-12 18:25:46.601509 [INFO] mod_dialplan_xml.c:391 Processing
FreeSWITCH->1920 in context default
2009-10-12 18:25:46.601509 [NOTICE] switch_channel.c:613 New Channel
sofia/external/14159927717 [1976e3c2-187c-4f05-98f5-36742ab8248f]
2009-10-12 18:25:46.601509 [NOTICE] switch_ivr.c:1367 Transfer
sofia/internal/sip_1 to XML[1920 at default]
API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
+OK 3fc6efb2-e4fa-454a-abb7-ebe39da748f5

freeswitch at matthew-laptop> 2009-10-12 18:25:46.677650 [NOTICE] sofia.c:3552
Ring-Ready sofia/external/14159927717!
2009-10-12 18:25:57.017477 [NOTICE] switch_ivr_originate.c:297 Hangup
sofia/external/14159927717 [CS_CONSUME_MEDIA] [ALLOTTED_TIMEOUT]
2009-10-12 18:25:57.017477 [INFO] switch_cpp.cpp:1116 PCHangup
gw:debug.comhc:ALLOTTED_TIMEOUT du:0 cn:sofia/external/14159927717
2009-10-12 18:25:57.037695 [NOTICE] switch_core_session.c:1087 Session 4
(sofia/external/14159927717) Ended
2009-10-12 18:25:57.037695 [NOTICE] switch_core_session.c:1089 Close Channel
sofia/external/14159927717 [CS_DESTROY]
2009-10-12 18:25:57.037695 [INFO] mod_dptools.c:2133 Originate Failed.
 Cause: ALLOTTED_TIMEOUT
2009-10-12 18:25:57.037695 [NOTICE] mod_dptools.c:2166 Hangup
sofia/internal/sip_1 [CS_EXECUTE] [ALLOTTED_TIMEOUT]
2009-10-12 18:25:57.037695 [NOTICE] switch_core_session.c:1087 Session 3
(sofia/internal/sip_1) Ended
2009-10-12 18:25:57.037695 [NOTICE] switch_core_session.c:1089 Close Channel
sofia/internal/sip_1 [CS_DESTROY]

--matt

On Tue, Oct 13, 2009 at 1:11 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Oct 12, 2009 at 10:42 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>
>> when I add a leg_timeout, I get an ALLOTTED_TIMEOUT from my failed
>> bridge...
>> when an ALLOTTED_TIMEOUT is received, the hangup_after_bridge=false is not
>> recognized (I think). Is there anyway to get an alloted_timeout to continue
>> after bridge (failure)?
>>
>
> Try it with ignore_early_media=true and see if it's the early media that's
> tripping you up.
> -MC
>
>
>>
>> revised dialplan & cmd output
>>
>>     <extension name="1920"><!--DEBUG-->
>>       <condition field="destination_number" expression="^1920$">
>>         <action application="set" data="hangup_after_bridge=false"/>
>>         <action application="bridge" data="{leg_timeout=10}sofia/gateway/
>> debug.com/14159927717"/>
>>          <action application="transfer" data="1999"/><!-- send to unable
>> to reach any contacts-->
>>       </condition>
>>     </extension>
>>
>> freeswitch at matthew-laptop> originate sofia/internal/sip_1%192.168.1.10
>> 1920
>> 2009-10-12 17:39:22.237622 [NOTICE] switch_channel.c:613 New Channel
>> sofia/internal/sip_1 [1d58fb59-c7f9-4908-b612-5bd1c12083cc]
>> 2009-10-12 17:39:22.313524 [NOTICE] sofia.c:3552 Ring-Ready
>> sofia/internal/sip_1!
>> 2009-10-12 17:39:25.142400 [NOTICE] sofia.c:3998 Channel
>> [sofia/internal/sip_1] has been answered
>> 2009-10-12 17:39:25.146259 [INFO] mod_dialplan_xml.c:391 Processing
>> FreeSWITCH->1920 in context default
>> 2009-10-12 17:39:25.146259 [NOTICE] switch_channel.c:613 New Channel
>> sofia/external/14159927717 [38cb7046-0c0d-47ef-94b5-bf8ccf35d185]
>> 2009-10-12 17:39:25.152026 [NOTICE] switch_ivr.c:1367 Transfer
>> sofia/internal/sip_1 to XML[1920 at default]
>>  API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
>> +OK 1d58fb59-c7f9-4908-b612-5bd1c12083cc
>>
>> freeswitch at matthew-laptop> 2009-10-12 17:39:25.217629 [NOTICE]
>> sofia.c:3552 Ring-Ready sofia/external/14159927717!
>> 2009-10-12 17:39:36.017513 [NOTICE] switch_ivr_originate.c:297 Hangup
>> sofia/external/14159927717 [CS_CONSUME_MEDIA] [ALLOTTED_TIMEOUT]
>> 2009-10-12 17:39:36.017513 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>> debug.com hc:ALLOTTED_TIMEOUT du:0 cn:sofia/external/14159927717
>> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1087 Session 12
>> (sofia/external/14159927717) Ended
>> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1089 Close
>> Channel sofia/external/14159927717 [CS_DESTROY]
>> 2009-10-12 17:39:36.037520 [INFO] mod_dptools.c:2133 Originate Failed.
>>  Cause: ALLOTTED_TIMEOUT
>> 2009-10-12 17:39:36.037520 [NOTICE] mod_dptools.c:2166 Hangup
>> sofia/internal/sip_1 [CS_EXECUTE] [ALLOTTED_TIMEOUT]
>> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1087 Session 11
>> (sofia/internal/sip_1) Ended
>> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1089 Close
>> Channel sofia/internal/sip_1 [CS_DESTROY]
>>
>> thanks.
>>
>> --matt
>>
>>
>> On Mon, Oct 12, 2009 at 10:45 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>>
>>> doh! thanks!
>>>
>>>
>>> On Mon, Oct 12, 2009 at 10:33 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> because the regex is on 1997 not 1999
>>>>
>>>>
>>>>
>>>> On Mon, Oct 12, 2009 at 10:25 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>>
>>>>>     <extension name="1999"><!--DIRECT POWER-->
>>>>>       <condition field="destination_number" expression="^1997$">
>>>>>         <action application="playback"
>>>>> data="hh/hh-unable_to_connect_contact.wav"/>
>>>>>         <action application="park"/>
>>>>>       </condition>
>>>>>     </extension>
>>>>>
>>>>> my extn 1999... since it looks from the output like it's transferring,
>>>>> just don't know why it's disconnecting the call instead of playing the .wav
>>>>> and parking.
>>>>>
>>>>> On Mon, Oct 12, 2009 at 10:23 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>>>
>>>>>> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179
>>>>>> Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
>>>>>>
>>>>>> might be the line..or the entire output is below....
>>>>>>
>>>>>> freeswitch at matthew-laptop> originate
>>>>>> sofia/internal/sip_1%192.168.1.10 1920
>>>>>> 2009-10-12 15:21:44.029517 [NOTICE] switch_channel.c:613 New Channel
>>>>>> sofia/internal/sip_1 [1e722934-7e94-46aa-9d62-e6ec7e7449cf]
>>>>>> 2009-10-12 15:21:44.121484 [NOTICE] sofia.c:3552 Ring-Ready
>>>>>> sofia/internal/sip_1!
>>>>>> 2009-10-12 15:21:47.285531 [NOTICE] sofia.c:3998 Channel
>>>>>> [sofia/internal/sip_1] has been answered
>>>>>> 2009-10-12 15:21:47.290996 [INFO] mod_dialplan_xml.c:391 Processing
>>>>>> FreeSWITCH->1920 in context default
>>>>>> 2009-10-12 15:21:47.293452 [NOTICE] switch_channel.c:613 New Channel
>>>>>> sofia/external/14159927717 [6b6cc440-e1d6-415a-b84b-494117e7361d]
>>>>>> 2009-10-12 15:21:47.293452 [NOTICE] switch_ivr.c:1367 Transfer
>>>>>> sofia/internal/sip_1 to XML[1920 at default]
>>>>>> API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
>>>>>> +OK 1e722934-7e94-46aa-9d62-e6ec7e7449cf
>>>>>>
>>>>>> freeswitch at matthew-laptop> 2009-10-12 15:21:47.369855 [NOTICE]
>>>>>> sofia.c:3552 Ring-Ready sofia/external/14159927717!
>>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr_originate.c:2336 Hangup
>>>>>> sofia/external/14159927717 [CS_CONSUME_MEDIA] [NO_ANSWER]
>>>>>> 2009-10-12 15:22:47.009474 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>>>>>> debug.com hc:NO_ANSWER du:0 cn:sofia/external/14159927717
>>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1087 Session
>>>>>> 47 (sofia/external/14159927717) Ended
>>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1089 Close
>>>>>> Channel sofia/external/14159927717 [CS_DESTROY]
>>>>>> 2009-10-12 15:22:47.009474 [INFO] mod_dptools.c:2133 Originate Failed.
>>>>>>  Cause: NO_ANSWER
>>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr.c:1367 Transfer
>>>>>> sofia/internal/sip_1 to XML[1999 at default]
>>>>>> 2009-10-12 15:22:47.009474 [INFO] mod_dialplan_xml.c:391 Processing
>>>>>> FreeSWITCH->1999 in context default
>>>>>> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179
>>>>>> Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
>>>>>> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1087 Session
>>>>>> 46 (sofia/internal/sip_1) Ended
>>>>>> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1089 Close
>>>>>> Channel sofia/internal/sip_1 [CS_DESTROY]
>>>>>>
>>>>>>
>>>>>> thanks for looking at this.
>>>>>>
>>>>>> On Mon, Oct 12, 2009 at 10:06 PM, Anthony Minessale <
>>>>>> anthony.minessale at gmail.com> wrote:
>>>>>>
>>>>>>> which line is hanging up your A (inbound) leg?
>>>>>>>
>>>>>>> look for a blue line that says "Hangup xyz...." that matches it so i
>>>>>>> can see.
>>>>>>>
>>>>>>> I think what is happening is you are getting early media so the
>>>>>>> bridge is actually working then when nobody answers it dies but technically
>>>>>>> the bridge worked.
>>>>>>>
>>>>>>> On Mon, Oct 12, 2009 at 9:41 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>>>>>
>>>>>>>> I think think this might be a bug, but wanted to post here instead
>>>>>>>> of Jira in-case I'm overlooking a configuration variable
>>>>>>>> Dialplan
>>>>>>>>
>>>>>>>>     <extension name="1920"><!--init agent for manual and power dial
>>>>>>>> mode-->
>>>>>>>>       <condition field="destination_number" expression="^1920$">
>>>>>>>>         <action application="set" data="hangup_after_bridge=false"/>
>>>>>>>>         <action application="bridge" data="sofia/gateway/
>>>>>>>> debug.com/14159927717"/>
>>>>>>>>         <action application="transfer" data="1999"/><!-- send to
>>>>>>>> unable to reach any contacts-->
>>>>>>>>       </condition>
>>>>>>>>     </extension>
>>>>>>>>
>>>>>>>> API Command
>>>>>>>> originate sofia/internal/sip_1%192.168.1.10 1920
>>>>>>>>
>>>>>>>> When the bridge to 14159927717 fails (NO_ANSWER) both calls are
>>>>>>>> terminated instead of continuing on in the dial plan to exten 1999 (which in
>>>>>>>> my dialplan parks the call). hangup_after_bridge however seems to work OK if
>>>>>>>> someone picks up in the bridge. Is this the correct behavior? How else can I
>>>>>>>> prevent the call from hanging up if a bridge fails? Thanks.
>>>>>>>>
>>>>>>>> I'm using 15135M
>>>>>>>>
>>>>>>>> --matt
>>>>>>>> http://www.hellohunter.com - Predictive Dialer
>>>>>>>> http://www.hellohunter.com/voice_broadcast.php - Voice Broadcasting
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Anthony Minessale II
>>>>>>>
>>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>>> ClueCon http://www.cluecon.com/
>>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>>
>>>>>>> AIM: anthm
>>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>>
>>>>>>> FreeSWITCH Developer Conference
>>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>>> pstn:213-799-1400
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:213-799-1400
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Oct 12 11:34:27 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 11:34:27 -0700
Subject: [Freeswitch-users] hangup_after_bridge=false not continueing
	after failed bridge
In-Reply-To: <87f2f3b90910121111m54cf2586pfca0d867307b1e86@mail.gmail.com>
References: <4256bf830910120741g5facaa6cr7ed92303dfafe74b@mail.gmail.com>
	<191c3a030910120806o30911131o743aa58420b41ae4@mail.gmail.com>
	<4256bf830910120823w5fcbc3eja4afb76b60b68fda@mail.gmail.com>
	<4256bf830910120825u70e988p4d7a041c520048a@mail.gmail.com>
	<191c3a030910120833i691b6fbfk3eecbf40e5496479@mail.gmail.com>
	<4256bf830910120845g4f991a55m8753c772dcc8cc31@mail.gmail.com>
	<4256bf830910121042o50ee79edife2243682bc9d3e2@mail.gmail.com>
	<87f2f3b90910121111m54cf2586pfca0d867307b1e86@mail.gmail.com>
Message-ID: <87f2f3b90910121134j25c728f5tb831cde418a14de7@mail.gmail.com>

Turn on debug, make another test call, and pastebin the output.
-MC

On Mon, Oct 12, 2009 at 11:11 AM, Michael Collins <msc at freeswitch.org>wrote:

>
>
> On Mon, Oct 12, 2009 at 10:42 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>
>> when I add a leg_timeout, I get an ALLOTTED_TIMEOUT from my failed
>> bridge...
>> when an ALLOTTED_TIMEOUT is received, the hangup_after_bridge=false is not
>> recognized (I think). Is there anyway to get an alloted_timeout to continue
>> after bridge (failure)?
>>
>
> Try it with ignore_early_media=true and see if it's the early media that's
> tripping you up.
> -MC
>
>
>>
>> revised dialplan & cmd output
>>
>>     <extension name="1920"><!--DEBUG-->
>>       <condition field="destination_number" expression="^1920$">
>>         <action application="set" data="hangup_after_bridge=false"/>
>>         <action application="bridge" data="{leg_timeout=10}sofia/gateway/
>> debug.com/14159927717"/>
>>          <action application="transfer" data="1999"/><!-- send to unable
>> to reach any contacts-->
>>       </condition>
>>     </extension>
>>
>> freeswitch at matthew-laptop> originate sofia/internal/sip_1%192.168.1.10
>> 1920
>> 2009-10-12 17:39:22.237622 [NOTICE] switch_channel.c:613 New Channel
>> sofia/internal/sip_1 [1d58fb59-c7f9-4908-b612-5bd1c12083cc]
>> 2009-10-12 17:39:22.313524 [NOTICE] sofia.c:3552 Ring-Ready
>> sofia/internal/sip_1!
>> 2009-10-12 17:39:25.142400 [NOTICE] sofia.c:3998 Channel
>> [sofia/internal/sip_1] has been answered
>> 2009-10-12 17:39:25.146259 [INFO] mod_dialplan_xml.c:391 Processing
>> FreeSWITCH->1920 in context default
>> 2009-10-12 17:39:25.146259 [NOTICE] switch_channel.c:613 New Channel
>> sofia/external/14159927717 [38cb7046-0c0d-47ef-94b5-bf8ccf35d185]
>> 2009-10-12 17:39:25.152026 [NOTICE] switch_ivr.c:1367 Transfer
>> sofia/internal/sip_1 to XML[1920 at default]
>>  API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
>> +OK 1d58fb59-c7f9-4908-b612-5bd1c12083cc
>>
>> freeswitch at matthew-laptop> 2009-10-12 17:39:25.217629 [NOTICE]
>> sofia.c:3552 Ring-Ready sofia/external/14159927717!
>> 2009-10-12 17:39:36.017513 [NOTICE] switch_ivr_originate.c:297 Hangup
>> sofia/external/14159927717 [CS_CONSUME_MEDIA] [ALLOTTED_TIMEOUT]
>> 2009-10-12 17:39:36.017513 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>> debug.com hc:ALLOTTED_TIMEOUT du:0 cn:sofia/external/14159927717
>> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1087 Session 12
>> (sofia/external/14159927717) Ended
>> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1089 Close
>> Channel sofia/external/14159927717 [CS_DESTROY]
>> 2009-10-12 17:39:36.037520 [INFO] mod_dptools.c:2133 Originate Failed.
>>  Cause: ALLOTTED_TIMEOUT
>> 2009-10-12 17:39:36.037520 [NOTICE] mod_dptools.c:2166 Hangup
>> sofia/internal/sip_1 [CS_EXECUTE] [ALLOTTED_TIMEOUT]
>> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1087 Session 11
>> (sofia/internal/sip_1) Ended
>> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1089 Close
>> Channel sofia/internal/sip_1 [CS_DESTROY]
>>
>> thanks.
>>
>> --matt
>>
>>
>> On Mon, Oct 12, 2009 at 10:45 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>>
>>> doh! thanks!
>>>
>>>
>>> On Mon, Oct 12, 2009 at 10:33 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> because the regex is on 1997 not 1999
>>>>
>>>>
>>>>
>>>> On Mon, Oct 12, 2009 at 10:25 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>>
>>>>>     <extension name="1999"><!--DIRECT POWER-->
>>>>>       <condition field="destination_number" expression="^1997$">
>>>>>         <action application="playback"
>>>>> data="hh/hh-unable_to_connect_contact.wav"/>
>>>>>         <action application="park"/>
>>>>>       </condition>
>>>>>     </extension>
>>>>>
>>>>> my extn 1999... since it looks from the output like it's transferring,
>>>>> just don't know why it's disconnecting the call instead of playing the .wav
>>>>> and parking.
>>>>>
>>>>> On Mon, Oct 12, 2009 at 10:23 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>>>
>>>>>> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179
>>>>>> Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
>>>>>>
>>>>>> might be the line..or the entire output is below....
>>>>>>
>>>>>> freeswitch at matthew-laptop> originate
>>>>>> sofia/internal/sip_1%192.168.1.10 1920
>>>>>> 2009-10-12 15:21:44.029517 [NOTICE] switch_channel.c:613 New Channel
>>>>>> sofia/internal/sip_1 [1e722934-7e94-46aa-9d62-e6ec7e7449cf]
>>>>>> 2009-10-12 15:21:44.121484 [NOTICE] sofia.c:3552 Ring-Ready
>>>>>> sofia/internal/sip_1!
>>>>>> 2009-10-12 15:21:47.285531 [NOTICE] sofia.c:3998 Channel
>>>>>> [sofia/internal/sip_1] has been answered
>>>>>> 2009-10-12 15:21:47.290996 [INFO] mod_dialplan_xml.c:391 Processing
>>>>>> FreeSWITCH->1920 in context default
>>>>>> 2009-10-12 15:21:47.293452 [NOTICE] switch_channel.c:613 New Channel
>>>>>> sofia/external/14159927717 [6b6cc440-e1d6-415a-b84b-494117e7361d]
>>>>>> 2009-10-12 15:21:47.293452 [NOTICE] switch_ivr.c:1367 Transfer
>>>>>> sofia/internal/sip_1 to XML[1920 at default]
>>>>>> API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
>>>>>> +OK 1e722934-7e94-46aa-9d62-e6ec7e7449cf
>>>>>>
>>>>>> freeswitch at matthew-laptop> 2009-10-12 15:21:47.369855 [NOTICE]
>>>>>> sofia.c:3552 Ring-Ready sofia/external/14159927717!
>>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr_originate.c:2336 Hangup
>>>>>> sofia/external/14159927717 [CS_CONSUME_MEDIA] [NO_ANSWER]
>>>>>> 2009-10-12 15:22:47.009474 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>>>>>> debug.com hc:NO_ANSWER du:0 cn:sofia/external/14159927717
>>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1087 Session
>>>>>> 47 (sofia/external/14159927717) Ended
>>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1089 Close
>>>>>> Channel sofia/external/14159927717 [CS_DESTROY]
>>>>>> 2009-10-12 15:22:47.009474 [INFO] mod_dptools.c:2133 Originate Failed.
>>>>>>  Cause: NO_ANSWER
>>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr.c:1367 Transfer
>>>>>> sofia/internal/sip_1 to XML[1999 at default]
>>>>>> 2009-10-12 15:22:47.009474 [INFO] mod_dialplan_xml.c:391 Processing
>>>>>> FreeSWITCH->1999 in context default
>>>>>> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179
>>>>>> Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
>>>>>> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1087 Session
>>>>>> 46 (sofia/internal/sip_1) Ended
>>>>>> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1089 Close
>>>>>> Channel sofia/internal/sip_1 [CS_DESTROY]
>>>>>>
>>>>>>
>>>>>> thanks for looking at this.
>>>>>>
>>>>>> On Mon, Oct 12, 2009 at 10:06 PM, Anthony Minessale <
>>>>>> anthony.minessale at gmail.com> wrote:
>>>>>>
>>>>>>> which line is hanging up your A (inbound) leg?
>>>>>>>
>>>>>>> look for a blue line that says "Hangup xyz...." that matches it so i
>>>>>>> can see.
>>>>>>>
>>>>>>> I think what is happening is you are getting early media so the
>>>>>>> bridge is actually working then when nobody answers it dies but technically
>>>>>>> the bridge worked.
>>>>>>>
>>>>>>> On Mon, Oct 12, 2009 at 9:41 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>>>>>
>>>>>>>> I think think this might be a bug, but wanted to post here instead
>>>>>>>> of Jira in-case I'm overlooking a configuration variable
>>>>>>>> Dialplan
>>>>>>>>
>>>>>>>>     <extension name="1920"><!--init agent for manual and power dial
>>>>>>>> mode-->
>>>>>>>>       <condition field="destination_number" expression="^1920$">
>>>>>>>>         <action application="set" data="hangup_after_bridge=false"/>
>>>>>>>>         <action application="bridge" data="sofia/gateway/
>>>>>>>> debug.com/14159927717"/>
>>>>>>>>         <action application="transfer" data="1999"/><!-- send to
>>>>>>>> unable to reach any contacts-->
>>>>>>>>       </condition>
>>>>>>>>     </extension>
>>>>>>>>
>>>>>>>> API Command
>>>>>>>> originate sofia/internal/sip_1%192.168.1.10 1920
>>>>>>>>
>>>>>>>> When the bridge to 14159927717 fails (NO_ANSWER) both calls are
>>>>>>>> terminated instead of continuing on in the dial plan to exten 1999 (which in
>>>>>>>> my dialplan parks the call). hangup_after_bridge however seems to work OK if
>>>>>>>> someone picks up in the bridge. Is this the correct behavior? How else can I
>>>>>>>> prevent the call from hanging up if a bridge fails? Thanks.
>>>>>>>>
>>>>>>>> I'm using 15135M
>>>>>>>>
>>>>>>>> --matt
>>>>>>>> http://www.hellohunter.com - Predictive Dialer
>>>>>>>> http://www.hellohunter.com/voice_broadcast.php - Voice Broadcasting
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Anthony Minessale II
>>>>>>>
>>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>>> ClueCon http://www.cluecon.com/
>>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>>
>>>>>>> AIM: anthm
>>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>>
>>>>>>> FreeSWITCH Developer Conference
>>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>>> pstn:213-799-1400
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:213-799-1400
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From msc at freeswitch.org  Mon Oct 12 11:35:29 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 11:35:29 -0700
Subject: [Freeswitch-users] Is assistivetech.net confusing the market
	place with their use of FreeSwitch product name?
In-Reply-To: <9dc4a1670910111048y2802b155oc7086b975193c9d7@mail.gmail.com>
References: <9dc4a1670910111048y2802b155oc7086b975193c9d7@mail.gmail.com>
Message-ID: <87f2f3b90910121135v89c348eoc033cfea692fa041@mail.gmail.com>

On Sun, Oct 11, 2009 at 10:48 AM, EdPimentl <edpimentl at gmail.com> wrote:

> http://www.assistivetech.net/search/productDisplay.php?product_id=18854
> -E
>
> Let's keep an eye on this but I doubt that an assistive technology
application is going to create any real confusion.
-MC

>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Oct 12 11:49:54 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 11:49:54 -0700
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <7d79b3930910120454m1ffc6dd4u42b91a4349597166@mail.gmail.com>
References: <7d79b3930910090407m5dc81d12pf2b0a47f6017804@mail.gmail.com>
	<20091009123615.BC50A3FA6E9@mail.cune.org>
	<7d79b3930910100647u7fd11413ucec0a535d68b3e3a@mail.gmail.com>
	<87f2f3b90910101009j1295d812jd98cae91e30b1eff@mail.gmail.com>
	<7d79b3930910120454m1ffc6dd4u42b91a4349597166@mail.gmail.com>
Message-ID: <87f2f3b90910121149x6e2114f0q996a262ad272a1f6@mail.gmail.com>

Lak,

Okay I will need a little bit of time to dig into the IE's and what they
contain. In the meantime can you tell me who the carrier is? I'd like to
find out if they have some specific requirements. The fact that it doesn't
work with libpri surprises me because that would mean that Asterisk systems
would probably not work with this carrier as well.

BTW, thanks for the very complete pastebin entries. :)
-MC
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From msc at freeswitch.org  Mon Oct 12 11:51:56 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 11:51:56 -0700
Subject: [Freeswitch-users] Upgrading causes no answer
In-Reply-To: <00ea01ca469b$7929faf0$6b7df0d0$@com>
References: <00ea01ca469b$7929faf0$6b7df0d0$@com>
Message-ID: <87f2f3b90910121151v229fb65fy1fd60ae1db2604c9@mail.gmail.com>

On Tue, Oct 6, 2009 at 8:41 AM, Lars Zeb <larclap at yahoo.com> wrote:

>  http://pastebin.freeswitch.org/10612
>
>
>
> I having been running v14996 OK for a while. I have upgraded a couple of
> times after, but every time, an inbound call is hung up on. The only thing
> that has changed is the upgrade. This morning I upgraded to v15098 and the
> problem persists.
>
>
>
> I believe it has to do with a lua script I use for inbound calls. Reading
> from the log, just after the script is launched, the following two lines
> appear:
>
>
>
> switch_cpp.cpp:1116  session not ready
>
> switch_cpp.cpp:925 destroy/unlink session from object
>
>
>
> Has something changed recently with lua processing? Is there something in
> the lua script which is causing the problem?
>
> Just following up. Have you upgraded to the most recent SVN and tried
again? If so, is the problem still here? Please let us know.
-MC
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From mattdfong at gmail.com  Mon Oct 12 12:00:29 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Tue, 13 Oct 2009 02:00:29 +0700
Subject: [Freeswitch-users] hangup_after_bridge=false not continueing
	after failed bridge
In-Reply-To: <87f2f3b90910121134j25c728f5tb831cde418a14de7@mail.gmail.com>
References: <4256bf830910120741g5facaa6cr7ed92303dfafe74b@mail.gmail.com>
	<191c3a030910120806o30911131o743aa58420b41ae4@mail.gmail.com>
	<4256bf830910120823w5fcbc3eja4afb76b60b68fda@mail.gmail.com>
	<4256bf830910120825u70e988p4d7a041c520048a@mail.gmail.com>
	<191c3a030910120833i691b6fbfk3eecbf40e5496479@mail.gmail.com>
	<4256bf830910120845g4f991a55m8753c772dcc8cc31@mail.gmail.com>
	<4256bf830910121042o50ee79edife2243682bc9d3e2@mail.gmail.com>
	<87f2f3b90910121111m54cf2586pfca0d867307b1e86@mail.gmail.com>
	<87f2f3b90910121134j25c728f5tb831cde418a14de7@mail.gmail.com>
Message-ID: <4256bf830910121200x7c0c0f30lfdab7ea5fbb4abc5@mail.gmail.com>

http://pastebin.freeswitch.org/10656

On Tue, Oct 13, 2009 at 1:34 AM, Michael Collins <msc at freeswitch.org> wrote:

> Turn on debug, make another test call, and pastebin the output.
> -MC
>
>
> On Mon, Oct 12, 2009 at 11:11 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Mon, Oct 12, 2009 at 10:42 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>>
>>> when I add a leg_timeout, I get an ALLOTTED_TIMEOUT from my failed
>>> bridge...
>>> when an ALLOTTED_TIMEOUT is received, the hangup_after_bridge=false is
>>> not recognized (I think). Is there anyway to get an alloted_timeout to
>>> continue after bridge (failure)?
>>>
>>
>> Try it with ignore_early_media=true and see if it's the early media that's
>> tripping you up.
>> -MC
>>
>>
>>>
>>> revised dialplan & cmd output
>>>
>>>     <extension name="1920"><!--DEBUG-->
>>>       <condition field="destination_number" expression="^1920$">
>>>         <action application="set" data="hangup_after_bridge=false"/>
>>>         <action application="bridge" data="{leg_timeout=10}sofia/gateway/
>>> debug.com/14159927717"/>
>>>          <action application="transfer" data="1999"/><!-- send to unable
>>> to reach any contacts-->
>>>       </condition>
>>>     </extension>
>>>
>>> freeswitch at matthew-laptop> originate sofia/internal/sip_1%192.168.1.10
>>> 1920
>>> 2009-10-12 17:39:22.237622 [NOTICE] switch_channel.c:613 New Channel
>>> sofia/internal/sip_1 [1d58fb59-c7f9-4908-b612-5bd1c12083cc]
>>> 2009-10-12 17:39:22.313524 [NOTICE] sofia.c:3552 Ring-Ready
>>> sofia/internal/sip_1!
>>> 2009-10-12 17:39:25.142400 [NOTICE] sofia.c:3998 Channel
>>> [sofia/internal/sip_1] has been answered
>>> 2009-10-12 17:39:25.146259 [INFO] mod_dialplan_xml.c:391 Processing
>>> FreeSWITCH->1920 in context default
>>> 2009-10-12 17:39:25.146259 [NOTICE] switch_channel.c:613 New Channel
>>> sofia/external/14159927717 [38cb7046-0c0d-47ef-94b5-bf8ccf35d185]
>>> 2009-10-12 17:39:25.152026 [NOTICE] switch_ivr.c:1367 Transfer
>>> sofia/internal/sip_1 to XML[1920 at default]
>>>  API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
>>> +OK 1d58fb59-c7f9-4908-b612-5bd1c12083cc
>>>
>>> freeswitch at matthew-laptop> 2009-10-12 17:39:25.217629 [NOTICE]
>>> sofia.c:3552 Ring-Ready sofia/external/14159927717!
>>> 2009-10-12 17:39:36.017513 [NOTICE] switch_ivr_originate.c:297 Hangup
>>> sofia/external/14159927717 [CS_CONSUME_MEDIA] [ALLOTTED_TIMEOUT]
>>> 2009-10-12 17:39:36.017513 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>>> debug.com hc:ALLOTTED_TIMEOUT du:0 cn:sofia/external/14159927717
>>> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1087 Session 12
>>> (sofia/external/14159927717) Ended
>>> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1089 Close
>>> Channel sofia/external/14159927717 [CS_DESTROY]
>>> 2009-10-12 17:39:36.037520 [INFO] mod_dptools.c:2133 Originate Failed.
>>>  Cause: ALLOTTED_TIMEOUT
>>> 2009-10-12 17:39:36.037520 [NOTICE] mod_dptools.c:2166 Hangup
>>> sofia/internal/sip_1 [CS_EXECUTE] [ALLOTTED_TIMEOUT]
>>> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1087 Session 11
>>> (sofia/internal/sip_1) Ended
>>> 2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1089 Close
>>> Channel sofia/internal/sip_1 [CS_DESTROY]
>>>
>>> thanks.
>>>
>>> --matt
>>>
>>>
>>> On Mon, Oct 12, 2009 at 10:45 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>
>>>> doh! thanks!
>>>>
>>>>
>>>> On Mon, Oct 12, 2009 at 10:33 PM, Anthony Minessale <
>>>> anthony.minessale at gmail.com> wrote:
>>>>
>>>>> because the regex is on 1997 not 1999
>>>>>
>>>>>
>>>>>
>>>>> On Mon, Oct 12, 2009 at 10:25 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>>>
>>>>>>     <extension name="1999"><!--DIRECT POWER-->
>>>>>>       <condition field="destination_number" expression="^1997$">
>>>>>>         <action application="playback"
>>>>>> data="hh/hh-unable_to_connect_contact.wav"/>
>>>>>>         <action application="park"/>
>>>>>>       </condition>
>>>>>>     </extension>
>>>>>>
>>>>>> my extn 1999... since it looks from the output like it's transferring,
>>>>>> just don't know why it's disconnecting the call instead of playing the .wav
>>>>>> and parking.
>>>>>>
>>>>>> On Mon, Oct 12, 2009 at 10:23 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>>>>
>>>>>>> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179
>>>>>>> Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
>>>>>>>
>>>>>>> might be the line..or the entire output is below....
>>>>>>>
>>>>>>> freeswitch at matthew-laptop> originate
>>>>>>> sofia/internal/sip_1%192.168.1.10 1920
>>>>>>> 2009-10-12 15:21:44.029517 [NOTICE] switch_channel.c:613 New Channel
>>>>>>> sofia/internal/sip_1 [1e722934-7e94-46aa-9d62-e6ec7e7449cf]
>>>>>>> 2009-10-12 15:21:44.121484 [NOTICE] sofia.c:3552 Ring-Ready
>>>>>>> sofia/internal/sip_1!
>>>>>>> 2009-10-12 15:21:47.285531 [NOTICE] sofia.c:3998 Channel
>>>>>>> [sofia/internal/sip_1] has been answered
>>>>>>> 2009-10-12 15:21:47.290996 [INFO] mod_dialplan_xml.c:391 Processing
>>>>>>> FreeSWITCH->1920 in context default
>>>>>>> 2009-10-12 15:21:47.293452 [NOTICE] switch_channel.c:613 New Channel
>>>>>>> sofia/external/14159927717 [6b6cc440-e1d6-415a-b84b-494117e7361d]
>>>>>>> 2009-10-12 15:21:47.293452 [NOTICE] switch_ivr.c:1367 Transfer
>>>>>>> sofia/internal/sip_1 to XML[1920 at default]
>>>>>>> API CALL [originate(sofia/internal/sip_1%192.168.1.10 1920)] output:
>>>>>>> +OK 1e722934-7e94-46aa-9d62-e6ec7e7449cf
>>>>>>>
>>>>>>> freeswitch at matthew-laptop> 2009-10-12 15:21:47.369855 [NOTICE]
>>>>>>> sofia.c:3552 Ring-Ready sofia/external/14159927717!
>>>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr_originate.c:2336
>>>>>>> Hangup sofia/external/14159927717 [CS_CONSUME_MEDIA] [NO_ANSWER]
>>>>>>> 2009-10-12 15:22:47.009474 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>>>>>>> debug.com hc:NO_ANSWER du:0 cn:sofia/external/14159927717
>>>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1087
>>>>>>> Session 47 (sofia/external/14159927717) Ended
>>>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_core_session.c:1089 Close
>>>>>>> Channel sofia/external/14159927717 [CS_DESTROY]
>>>>>>> 2009-10-12 15:22:47.009474 [INFO] mod_dptools.c:2133 Originate
>>>>>>> Failed.  Cause: NO_ANSWER
>>>>>>> 2009-10-12 15:22:47.009474 [NOTICE] switch_ivr.c:1367 Transfer
>>>>>>> sofia/internal/sip_1 to XML[1999 at default]
>>>>>>> 2009-10-12 15:22:47.009474 [INFO] mod_dialplan_xml.c:391 Processing
>>>>>>> FreeSWITCH->1999 in context default
>>>>>>> 2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179
>>>>>>> Hangup sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
>>>>>>> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1087
>>>>>>> Session 46 (sofia/internal/sip_1) Ended
>>>>>>> 2009-10-12 15:22:47.017768 [NOTICE] switch_core_session.c:1089 Close
>>>>>>> Channel sofia/internal/sip_1 [CS_DESTROY]
>>>>>>>
>>>>>>>
>>>>>>> thanks for looking at this.
>>>>>>>
>>>>>>> On Mon, Oct 12, 2009 at 10:06 PM, Anthony Minessale <
>>>>>>> anthony.minessale at gmail.com> wrote:
>>>>>>>
>>>>>>>> which line is hanging up your A (inbound) leg?
>>>>>>>>
>>>>>>>> look for a blue line that says "Hangup xyz...." that matches it so i
>>>>>>>> can see.
>>>>>>>>
>>>>>>>> I think what is happening is you are getting early media so the
>>>>>>>> bridge is actually working then when nobody answers it dies but technically
>>>>>>>> the bridge worked.
>>>>>>>>
>>>>>>>> On Mon, Oct 12, 2009 at 9:41 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>>>>>>
>>>>>>>>> I think think this might be a bug, but wanted to post here instead
>>>>>>>>> of Jira in-case I'm overlooking a configuration variable
>>>>>>>>> Dialplan
>>>>>>>>>
>>>>>>>>>     <extension name="1920"><!--init agent for manual and power dial
>>>>>>>>> mode-->
>>>>>>>>>       <condition field="destination_number" expression="^1920$">
>>>>>>>>>         <action application="set"
>>>>>>>>> data="hangup_after_bridge=false"/>
>>>>>>>>>         <action application="bridge" data="sofia/gateway/
>>>>>>>>> debug.com/14159927717"/>
>>>>>>>>>         <action application="transfer" data="1999"/><!-- send to
>>>>>>>>> unable to reach any contacts-->
>>>>>>>>>       </condition>
>>>>>>>>>     </extension>
>>>>>>>>>
>>>>>>>>> API Command
>>>>>>>>> originate sofia/internal/sip_1%192.168.1.10 1920
>>>>>>>>>
>>>>>>>>> When the bridge to 14159927717 fails (NO_ANSWER) both calls are
>>>>>>>>> terminated instead of continuing on in the dial plan to exten 1999 (which in
>>>>>>>>> my dialplan parks the call). hangup_after_bridge however seems to work OK if
>>>>>>>>> someone picks up in the bridge. Is this the correct behavior? How else can I
>>>>>>>>> prevent the call from hanging up if a bridge fails? Thanks.
>>>>>>>>>
>>>>>>>>> I'm using 15135M
>>>>>>>>>
>>>>>>>>> --matt
>>>>>>>>> http://www.hellohunter.com - Predictive Dialer
>>>>>>>>> http://www.hellohunter.com/voice_broadcast.php - Voice
>>>>>>>>> Broadcasting
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> Anthony Minessale II
>>>>>>>>
>>>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>>>> ClueCon http://www.cluecon.com/
>>>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>>>
>>>>>>>> AIM: anthm
>>>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>>>
>>>>>>>> FreeSWITCH Developer Conference
>>>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>>>> pstn:213-799-1400
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>> iax:guest at conference.freeswitch.org/888
>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>> pstn:213-799-1400
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Oct 12 12:01:05 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 12:01:05 -0700
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Update: Valet
	Parking
In-Reply-To: <15b9404e0910102018oda8834ah8cc87787ee00c22b@mail.gmail.com>
References: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>
	<15b9404e0910102018oda8834ah8cc87787ee00c22b@mail.gmail.com>
Message-ID: <87f2f3b90910121201o422c7891vd30ee3761aef0eb7@mail.gmail.com>

On Sat, Oct 10, 2009 at 8:18 PM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:

>
>
> On Fri, Oct 9, 2009 at 10:42 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> FYI,
>>
>> The FreeSWITCH devs have added valet parking! Check it out:
>> http://www.freeswitch.org/node/207
>>
>> Let us know what you think.
>>
>
> Very nice.
>
> But I think a valet_unpark app is missing.
> If the intention of the person sent to the valet lot is to retrieve a call
> there, the person can assume the call was already retrieved by someone else
> or that the caller hung up if he/she hears MOH. But it would be nicer to
> have a valet_unpark app to fail and let the dialplan play a message.
>
> I understand what you are saying. I'm not sure I agree, but we'll kick the
idea around when we have a few minutes and let you know what we decide.
-MC
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From Russell.Mosemann at cune.org  Mon Oct 12 12:20:07 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Mon, 12 Oct 2009 19:20:07 -0000
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Update:
	Valet	Parking
In-Reply-To: <87f2f3b90910121201o422c7891vd30ee3761aef0eb7@mail.gmail.com>
Message-ID: <20091012192008.05FE3301FB6@mail.cune.org>

Michael Collins <msc at freeswitch.org> said:

> On Sat, Oct 10, 2009 at 8:18 PM, mayamatakeshi
<mayamatakeshi at gmail.com>wrote:
> 
> >
> >
> > On Fri, Oct 9, 2009 at 10:42 AM, Michael Collins
<msc at freeswitch.org>wrote:
> >
> >> FYI,
> >>
> >> The FreeSWITCH devs have added valet parking! Check it out:
> >> http://www.freeswitch.org/node/207
> >>
> >> Let us know what you think.
> >>
> >
> > Very nice.
> >
> > But I think a valet_unpark app is missing.
> > If the intention of the person sent to the valet lot is to retrieve a
call
> > there, the person can assume the call was already retrieved by
someone else
> > or that the caller hung up if he/she hears MOH. But it would be nicer to
> > have a valet_unpark app to fail and let the dialplan play a message.
> >
> > I understand what you are saying. I'm not sure I agree, but we'll
kick the
> idea around when we have a few minutes and let you know what we decide.
> -MC

If you do decide to implement something, I would encourage that it be
flexible so that when the parking meter runs out :-), it could either
play a message or forward the call to an extension (default to the
extension that parked it).

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!




From jmesquita at freeswitch.org  Mon Oct 12 12:37:26 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 12 Oct 2009 16:37:26 -0300
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Update: Valet
	Parking
In-Reply-To: <20091012192008.05FE3301FB6@mail.cune.org>
References: <87f2f3b90910121201o422c7891vd30ee3761aef0eb7@mail.gmail.com>
	<20091012192008.05FE3301FB6@mail.cune.org>
Message-ID: <ebbc2d550910121237rbca507fpa76880b0b99d03f4@mail.gmail.com>

I would say that the parking meter is a good idea and it is the default
behavior of parking on legacy PBXs. Since we always do _more_, what do you
think about having the option to transfer to any extension instead of just
the one that transfered the call?

Regards,

jm

On Mon, Oct 12, 2009 at 4:20 PM, <Russell.Mosemann at cune.org> wrote:

> Michael Collins <msc at freeswitch.org> said:
>
> > On Sat, Oct 10, 2009 at 8:18 PM, mayamatakeshi
> <mayamatakeshi at gmail.com>wrote:
> >
> > >
> > >
> > > On Fri, Oct 9, 2009 at 10:42 AM, Michael Collins
> <msc at freeswitch.org>wrote:
> > >
> > >> FYI,
> > >>
> > >> The FreeSWITCH devs have added valet parking! Check it out:
> > >> http://www.freeswitch.org/node/207
> > >>
> > >> Let us know what you think.
> > >>
> > >
> > > Very nice.
> > >
> > > But I think a valet_unpark app is missing.
> > > If the intention of the person sent to the valet lot is to retrieve a
> call
> > > there, the person can assume the call was already retrieved by
> someone else
> > > or that the caller hung up if he/she hears MOH. But it would be nicer
> to
> > > have a valet_unpark app to fail and let the dialplan play a message.
> > >
> > > I understand what you are saying. I'm not sure I agree, but we'll
> kick the
> > idea around when we have a few minutes and let you know what we decide.
> > -MC
>
> If you do decide to implement something, I would encourage that it be
> flexible so that when the parking meter runs out :-), it could either
> play a message or forward the call to an extension (default to the
> extension that parked it).
>
> --
> Russell Mosemann
>
>
>
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From quentusrex at gmail.com  Mon Oct 12 12:55:20 2009
From: quentusrex at gmail.com (William King)
Date: Mon, 12 Oct 2009 12:55:20 -0700
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Update: Valet
 Parking
In-Reply-To: <87f2f3b90910121201o422c7891vd30ee3761aef0eb7@mail.gmail.com>
References: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>	<15b9404e0910102018oda8834ah8cc87787ee00c22b@mail.gmail.com>
	<87f2f3b90910121201o422c7891vd30ee3761aef0eb7@mail.gmail.com>
Message-ID: <4AD389A8.1080603@gmail.com>

I don't know if this was mentioned yet. It would be useful to have a way
to have the parking lot automatically find the next available spot and
tts it to the person parking the call.

Then the auto unpark would pop off the lowest numbered lot, or return
fail if there is nobody in the parking lots etc.

-William King

Michael Collins wrote:
>
>
> On Sat, Oct 10, 2009 at 8:18 PM, mayamatakeshi
> <mayamatakeshi at gmail.com <mailto:mayamatakeshi at gmail.com>> wrote:
>
>
>
>     On Fri, Oct 9, 2009 at 10:42 AM, Michael Collins
>     <msc at freeswitch.org <mailto:msc at freeswitch.org>> wrote:
>
>         FYI,
>
>         The FreeSWITCH devs have added valet parking! Check it out:
>         http://www.freeswitch.org/node/207
>
>         Let us know what you think.
>
>
>     Very nice.
>
>     But I think a valet_unpark app is missing.
>     If the intention of the person sent to the valet lot is to
>     retrieve a call there, the person can assume the call was already
>     retrieved by someone else or that the caller hung up if he/she
>     hears MOH. But it would be nicer to have a valet_unpark app to
>     fail and let the dialplan play a message.
>
> I understand what you are saying. I'm not sure I agree, but we'll kick
> the idea around when we have a few minutes and let you know what we
> decide.
> -MC
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From vinuth.madinur at gmail.com  Mon Oct 12 12:58:32 2009
From: vinuth.madinur at gmail.com (Vinuth Madinur)
Date: Tue, 13 Oct 2009 01:28:32 +0530
Subject: [Freeswitch-users] SIT tones and SIP Trunk provider.
Message-ID: <910309030910121258r4afe6b96g9db4756e9f0e090b@mail.gmail.com>

Hi,
Does Freeswitch detect all of these hangup cases mentioned here [
http://wiki.freeswitch.org/wiki/Hangup_causes] when using it through a SIP
Trunk provider?

If not, should I put in tone_detect application in the dialplan for
detecting the SITs?

Won't freeswitch have to depend on the SIP status sent from SIP trunk to
know the hangup status? So, I'm wondering if tone_detect will work at all?

Please provide your advice.

Thanks,
Vinuth.
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From msc at freeswitch.org  Mon Oct 12 13:31:19 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 13:31:19 -0700
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Update: Valet
	Parking
In-Reply-To: <4AD389A8.1080603@gmail.com>
References: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>
	<15b9404e0910102018oda8834ah8cc87787ee00c22b@mail.gmail.com>
	<87f2f3b90910121201o422c7891vd30ee3761aef0eb7@mail.gmail.com>
	<4AD389A8.1080603@gmail.com>
Message-ID: <87f2f3b90910121331y23a11983vd40f75f861249623@mail.gmail.com>

On Mon, Oct 12, 2009 at 12:55 PM, William King <quentusrex at gmail.com> wrote:

> I don't know if this was mentioned yet. It would be useful to have a way
> to have the parking lot automatically find the next available spot and
> tts it to the person parking the call.
>
> Then the auto unpark would pop off the lowest numbered lot, or return
> fail if there is nobody in the parking lots etc.
>

Of course, if you're gonna do this then you might as well use FIFO queues.
;)
-MC
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From msc at freeswitch.org  Mon Oct 12 13:39:09 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 13:39:09 -0700
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Update: Valet
	Parking
In-Reply-To: <ebbc2d550910121237rbca507fpa76880b0b99d03f4@mail.gmail.com>
References: <87f2f3b90910121201o422c7891vd30ee3761aef0eb7@mail.gmail.com>
	<20091012192008.05FE3301FB6@mail.cune.org>
	<ebbc2d550910121237rbca507fpa76880b0b99d03f4@mail.gmail.com>
Message-ID: <87f2f3b90910121339m3e08484enbc2d9eab746bf03d@mail.gmail.com>

2009/10/12 Jo?o Mesquita <jmesquita at freeswitch.org>

> I would say that the parking meter is a good idea and it is the default
> behavior of parking on legacy PBXs. Since we always do _more_, what do you
> think about having the option to transfer to any extension instead of just
> the one that transfered the call?
>
>
Gents,

Thanks for all the feedback. I think the proper thing to do is to make sure
that we have, as Brian would put it, all the Lego(tm) bricks necessary to
build the functionality that you wish to implement. A lot of what I'm
hearing that you guys want can be handled with FIFO. Before we do any more
programming with valet_park let's make sure that the existing Legos don't
already give you the tools that you need.

-MC
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From msc at freeswitch.org  Mon Oct 12 13:45:13 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 13:45:13 -0700
Subject: [Freeswitch-users] SIT tones and SIP Trunk provider.
In-Reply-To: <910309030910121258r4afe6b96g9db4756e9f0e090b@mail.gmail.com>
References: <910309030910121258r4afe6b96g9db4756e9f0e090b@mail.gmail.com>
Message-ID: <87f2f3b90910121345wef2a3f6u760fde22df0aba81@mail.gmail.com>

On Mon, Oct 12, 2009 at 12:58 PM, Vinuth Madinur
<vinuth.madinur at gmail.com>wrote:

> Hi,
> Does Freeswitch detect all of these hangup cases mentioned here [
> http://wiki.freeswitch.org/wiki/Hangup_causes] when using it through a SIP
> Trunk provider?
>
> If not, should I put in tone_detect application in the dialplan for
> detecting the SITs?
>
> Won't freeswitch have to depend on the SIP status sent from SIP trunk to
> know the hangup status? So, I'm wondering if tone_detect will work at all?
>
>
Vinuth,

As usual, "it depends." Your provider is the key to this whole operation. If
the SIP provider sends the information inband then you will definitely need
to use tone_detect to look for the SIT tones. However, if the information
comes back with the normal SIP messages then you're good to go. I've seen
more than a few SIP providers do both, which means that you have to prepare
for both cases.

My advice to you is to get pcaps of failed calls and analyze them with
Wireshark. If you need help analyzing them then put your pcaps on a web
server and post a link so that others can download them. The wiki has some
information on grabbing pcaps:

http://wiki.freeswitch.org/wiki/Packet_Capture

If you haven't already done so, go to cluecon.com and download the torrent
file that has the ClueCon speaker presentations. The last presentation on
Day 3 is Jason Garland and he walks you through using Wireshark for
analyzing a SIP call, including both the signaling (SIP) and the media (RTP)
parts of the call. BTW, if you have a copy of "VoIP Deployment For Dummies"
it has a small section on using Wireshark for call analysis.

-MC
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From vinuth.madinur at gmail.com  Mon Oct 12 14:19:22 2009
From: vinuth.madinur at gmail.com (Vinuth Madinur)
Date: Tue, 13 Oct 2009 02:49:22 +0530
Subject: [Freeswitch-users] SIT tones and SIP Trunk provider.
In-Reply-To: <87f2f3b90910121345wef2a3f6u760fde22df0aba81@mail.gmail.com>
References: <910309030910121258r4afe6b96g9db4756e9f0e090b@mail.gmail.com>
	<87f2f3b90910121345wef2a3f6u760fde22df0aba81@mail.gmail.com>
Message-ID: <910309030910121419o56a547bft83e5218946a9a0f4@mail.gmail.com>

Thanks Michael. I'll go through the resources you mentioned.
Thanks,
Vinuth.


On Tue, Oct 13, 2009 at 2:15 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Oct 12, 2009 at 12:58 PM, Vinuth Madinur <vinuth.madinur at gmail.com
> > wrote:
>
>> Hi,
>> Does Freeswitch detect all of these hangup cases mentioned here [
>> http://wiki.freeswitch.org/wiki/Hangup_causes] when using it through a
>> SIP Trunk provider?
>>
>> If not, should I put in tone_detect application in the dialplan for
>> detecting the SITs?
>>
>> Won't freeswitch have to depend on the SIP status sent from SIP trunk to
>> know the hangup status? So, I'm wondering if tone_detect will work at all?
>>
>>
> Vinuth,
>
> As usual, "it depends." Your provider is the key to this whole operation.
> If the SIP provider sends the information inband then you will definitely
> need to use tone_detect to look for the SIT tones. However, if the
> information comes back with the normal SIP messages then you're good to go.
> I've seen more than a few SIP providers do both, which means that you have
> to prepare for both cases.
>
> My advice to you is to get pcaps of failed calls and analyze them with
> Wireshark. If you need help analyzing them then put your pcaps on a web
> server and post a link so that others can download them. The wiki has some
> information on grabbing pcaps:
>
> http://wiki.freeswitch.org/wiki/Packet_Capture
>
> If you haven't already done so, go to cluecon.com and download the torrent
> file that has the ClueCon speaker presentations. The last presentation on
> Day 3 is Jason Garland and he walks you through using Wireshark for
> analyzing a SIP call, including both the signaling (SIP) and the media (RTP)
> parts of the call. BTW, if you have a copy of "VoIP Deployment For Dummies"
> it has a small section on using Wireshark for call analysis.
>
> -MC
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Mon Oct 12 14:43:47 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 12 Oct 2009 23:43:47 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
Message-ID: <65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>

hi,

finally i compiled it right ... had a stupid issue with ekiga and wrong
ptlib in place...

anyhow, i loaded the module and will continue the tests tomorrow ...first
thing i arrive in my office :P



freeswitch at subZero>
freeswitch at subZero>
API CALL [console(loglevel 7)] output:
+OK console log level set to DEBUG

freeswitch at subZero> load mod_h323
2009-10-12 23:38:09.509279 [CONSOLE] mod_h323.cpp:93 Starting loading
mod_h323
2009-10-12 23:38:09.511413 [DEBUG] mod_h323.cpp:461 Created Listener
'default'
2009-10-12 23:38:09.511413 [DEBUG] mod_h323.cpp:305 Config capabilliti
PCMA,GSM,G729,G726
2009-10-12 23:38:09.511413 [DEBUG] mod_h323.cpp:309 Find capabilliti PCMU to
PCMA,GSM,G729,G726
2009-10-12 23:38:09.511413 [DEBUG] mod_h323.cpp:309 Find capabilliti PCMA to
PCMA,GSM,G729,G726
2009-10-12 23:38:09.511413 [DEBUG] h323caps.cxx:3264 FindCapability:
"G.711-uLaw-64k{sw}"
2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:3176 Added capability:
G.711-uLaw-64k <1>
2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:3264 FindCapability:
"G.711-uLaw-64k{sw}"
2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:3264 FindCapability:
"G.711-uLaw-64k{sw}"
2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:3264 FindCapability:
"G.711-uLaw-64k{sw}"
2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:3264 FindCapability:
"G.711-uLaw-64k{sw}"
2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:1989 No Extended
Capabilities found to load
2009-10-12 23:38:09.512480 [DEBUG] mod_h323.cpp:325 H.323 added 1
capabilities 'G.711-uLaw-64k*{sw}'
2009-10-12 23:38:09.512480 [DEBUG] mod_h323.cpp:309 Find capabilliti gsm to
PCMA,GSM,G729,G726
2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti msgsm
to PCMA,GSM,G729,G726
2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti speex
to PCMA,GSM,G729,G726
2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti lpc10
to PCMA,GSM,G729,G726
2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti ilbc20
to PCMA,GSM,G729,G726
2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti ilbc30
to PCMA,GSM,G729,G726
2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g723 to
PCMA,GSM,G729,G726
2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g726 to
PCMA,GSM,G729,G726
2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g728 to
PCMA,GSM,G729,G726
2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g729b
to PCMA,GSM,G729,G726
2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g729 to
PCMA,GSM,G729,G726
2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti slin to
PCMA,GSM,G729,G726
2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g729a
to PCMA,GSM,G729,G726
2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g729ab
to PCMA,GSM,G729,G726
2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti g723.1
to PCMA,GSM,G729,G726
2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
g723.1-5k3 to PCMA,GSM,G729,G726
2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
g723.1a-5k3 to PCMA,GSM,G729,G726
2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
g723.1a-6k3 to PCMA,GSM,G729,G726
2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
g723.1a-6k3-cisco to PCMA,GSM,G729,G726
2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti g726-16
to PCMA,GSM,G729,G726
2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti g726-24
to PCMA,GSM,G729,G726
2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti g726-32
to PCMA,GSM,G729,G726
2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti g726-40
to PCMA,GSM,G729,G726
2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti ilbc to
PCMA,GSM,G729,G726
2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
speex-18k2 to PCMA,GSM,G729,G726
2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
speex-15k to PCMA,GSM,G729,G726
2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
speex-11k to PCMA,GSM,G729,G726
2009-10-12 23:38:09.515708 [DEBUG] mod_h323.cpp:309 Find capabilliti
speex-8k to PCMA,GSM,G729,G726
2009-10-12 23:38:09.515708 [DEBUG] mod_h323.cpp:309 Find capabilliti
speex-5k95 to PCMA,GSM,G729,G726
2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
UserInput/hookflash <2>
2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
UserInput/basicString <3>
2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
UserInput/dtmf <4>
2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
UserInput/RFC2833 <5>
2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
UserInput/Navigation <6>
2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
UserInput/Softkey <7>
2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
UserInput/PointDevice <8>
2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
UserInput/Modal <9>
2009-10-12 23:38:09.516760 [DEBUG] osutil.cxx:188 File handle high water
mark set: 44 PTCPSocket
2009-10-12 23:38:09.516760 [DEBUG] h323ep.cxx:1995 Started listener
Listener[ip$10.4.62.7:1720]
2009-10-12 23:38:09.516760 [DEBUG] tlibthrd.cxx:547 Thread high water mark
set: 3
2009-10-12 23:38:09.516760 [CONSOLE] mod_h323.cpp:108 Opal manager
initialized and running
2009-10-12 23:38:09.516760 [DEBUG] tlibthrd.cxx:454 Started thread
0xb75e0f18 H323 Listener:b5c98b90
2009-10-12 23:38:09.516760 [CONSOLE] switch_loadable_module.c:889
Successfully Loaded [mod_h323]
2009-10-12 23:38:09.516760 [NOTICE] switch_loadable_module.c:142 Adding
Endpoint 'h323'
API CALL [load(mod_h323)] output:
+OK

2009-10-12 23:38:09.516760 [DEBUG] transports.cxx:1521 Awaiting TCP
connections on port 1720
2009-10-12 23:38:09.517832 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.4.62.7:1720
2009-10-12 23:38:09.517832 [DEBUG] tlibthrd.cxx:1023
PThread::PXBlockOnIO(44,2)
freeswitch at subZero>


BTW: to compile everytihng,  I used FS trunk, ptlib 2.7.1 and h323plus
1.22.0

Will let you know the results asap.


T.



2009/10/12 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-12 18:03 +0200, Tihomir Culjaga wrote
> freeswitch-users at lists.fre...:
>
> i already write this, ptlib 2.6.5, tou can find link to it on
> oplalvoip.org, h32plus latest CVS version,
> you can find it on www.h323plus.org.
>
> TC>hi,
> TC>
> TC>can't make it...
> TC>
> TC>subZero:~/freeswitch-trunk$ make mod_h323
> TC>
> TC>making all mod_h323
> TC>Compiling mod_h323.cpp...
> TC>quiet_libtool: compile:  g++ -g -ggdb -I/usr/local/include/ptlib
> TC>-I/usr/local/include/openh323 -I. -DPTRACING=1 -D_REENTRANT
> -fno-exceptions
> TC>-I/home/tculjaga/freeswitch-trunk/src/include
> TC>-I/home/tculjaga/freeswitch-trunk/libs/libteletone/src -fPIC
> TC>-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2
> TC>-D_GNU_SOURCE -DHAVE_CONFIG_H -c mod_h323.cpp  -fPIC -DPIC -o
> TC>.libs/mod_h323.o
> TC>In file included from /usr/local/include/openh323/h323.h:493,
> TC>                 from mod_h323.h:8,
> TC>                 from mod_h323.cpp:3:
> TC>/usr/local/include/openh323/h323ep.h: In member function ??????virtual
> void
> TC>NATFactoryStartup::OnShutdown()??????:
> TC>/usr/local/include/openh323/h323ep.h:2731: error: ??????NatFactory??????
> has not
> TC>been declared
> TC>make[4]: *** [mod_h323.lo] Error 1
> TC>make[3]: *** [all] Error 1
> TC>make[2]: *** [mod_h323-all] Error 1
> TC>make[1]: *** [mod_h323] Error 2
> TC>make: *** [mod_h323] Error 2
> TC>
> TC>
> TC>
> TC>what exact ptlib and h323plus versions did you use? .. can you send us a
> TC>link so we can use the exact ones.
> TC>
> TC>
> TC>T.
> TC>
> TC>2009/10/12 Georgiewskiy Yuriy <bottleman at icf.org.ru>
> TC>
> TC>> On 2009-10-12 09:43 -0500, Brian West wrote
> TC>> freeswitch-users at lists.freeswit...:
> TC>>
> TC>> BW>We can host this in our SVN if you wish?
> TC>>
> TC>> If in fs svn i think yes. But i think may be little time later?
> TC>> i don't known is it builds on trunk because i develop it on 1.0.4.
> TC>>
> TC>> BW>/b
> TC>> BW>
> TC>> BW>On Oct 12, 2009, at 8:31 AM, Georgiewskiy Yuriy wrote:
> TC>> BW>
> TC>> BW>> ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/ alfa code,
> but
> TC>> seems
> TC>> BW>> it work, but should be buggy,
> TC>> BW>> to build need libpt 2.6.5 and h323plus cvs version, i test it now
> on
> TC>> fs
> TC>> BW>> 1.0.4.
> TC>> BW>
> TC>>
> TC>> C ?????????                       With Best Regards
> TC>> ???????????? ????.                Georgiewskiy Yuriy
> TC>> +7 4872 711666                    +7 4872 711666
> TC>> ???? +7 4872 711143               fax +7 4872 711143
> TC>> ???????? ??? "?? ?? ??????"       IT Service Ltd
> TC>> http://nkoort.ru                  http://nkoort.ru
> TC>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> TC>> YG129-RIPE                        YG129-RIPE
> TC>>
> TC>> _______________________________________________
> TC>> FreeSWITCH-users mailing list
> TC>> FreeSWITCH-users at lists.freeswitch.org
> TC>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> TC>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> TC>> http://www.freeswitch.org
> TC>>
> TC>>
> TC>
>
> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129-RIPE
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Oct 12 14:48:57 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 14:48:57 -0700
Subject: [Freeswitch-users] hangup_after_bridge=false not continueing
	after failed bridge
In-Reply-To: <4256bf830910121200x7c0c0f30lfdab7ea5fbb4abc5@mail.gmail.com>
References: <4256bf830910120741g5facaa6cr7ed92303dfafe74b@mail.gmail.com>
	<191c3a030910120806o30911131o743aa58420b41ae4@mail.gmail.com>
	<4256bf830910120823w5fcbc3eja4afb76b60b68fda@mail.gmail.com>
	<4256bf830910120825u70e988p4d7a041c520048a@mail.gmail.com>
	<191c3a030910120833i691b6fbfk3eecbf40e5496479@mail.gmail.com>
	<4256bf830910120845g4f991a55m8753c772dcc8cc31@mail.gmail.com>
	<4256bf830910121042o50ee79edife2243682bc9d3e2@mail.gmail.com>
	<87f2f3b90910121111m54cf2586pfca0d867307b1e86@mail.gmail.com>
	<87f2f3b90910121134j25c728f5tb831cde418a14de7@mail.gmail.com>
	<4256bf830910121200x7c0c0f30lfdab7ea5fbb4abc5@mail.gmail.com>
Message-ID: <87f2f3b90910121448w581854d7tf878671430cce133@mail.gmail.com>

On Mon, Oct 12, 2009 at 12:00 PM, Matthew Fong <mattdfong at gmail.com> wrote:

> http://pastebin.freeswitch.org/10656
>
>
> Matthew,

Try continue_on_fail=true instead of hangup_after_bridge=false.
http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail

I think it will do what you want.
-MC
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From diego.viola at gmail.com  Mon Oct 12 15:11:51 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Mon, 12 Oct 2009 22:11:51 +0000
Subject: [Freeswitch-users] Some documentation thoughts
Message-ID: <86a32abc0910121511m4241faefg4b8d1caf3b0f7a87@mail.gmail.com>

Hello,

I have been doing some work recently on the FreeSWITCH wiki, to improve
things.

You can see some of my work here:

http://wiki.freeswitch.org/index.php?title=Special:Contributions&limit=500&target=Diego.viola

I am trying to polish the wiki and give it a more professional and clean
look, this means that we need to enforce some guidelines and strive to make
the wiki even better.

How can we do this?

Well, the first thing is to review everything we have, right now we have too
many separated and little pages here and there that no one cares or read, we
should avoid doing little or small pages. We need to correct typos, I
suggest that people click on "Random page" link and start correcting typos,
install an English (US) dictionary in your browser and enable Spell Checking
and start correcting these typos if you want to help, also, make acronyms
and initialisms all-uppercase when you make one.

When you are adding documentation make sure you don't do a separate page for
an existing page, etc. and we also need to define a size for titles, body,
etc, so different pages don't look different from each other, I will make
sure I add all this into our guidelines and that all pages follow this.

My goal is to have great documentation, similar to the Apache documentation:
http://httpd.apache.org/docs/2.2/

Or even better, I hope you like this idea.

Diego
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From diego.viola at gmail.com  Mon Oct 12 15:21:59 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Mon, 12 Oct 2009 22:21:59 +0000
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <86a32abc0910121511m4241faefg4b8d1caf3b0f7a87@mail.gmail.com>
References: <86a32abc0910121511m4241faefg4b8d1caf3b0f7a87@mail.gmail.com>
Message-ID: <86a32abc0910121521h46c6e0a8r19718c020b6ded5e@mail.gmail.com>

Is this page still necessary:

http://wiki.freeswitch.org/wiki/Old_mod_python

I'd like to have the mod_python page only:

http://wiki.freeswitch.org/wiki/Mod_python

If there is something in the old one please let me know so we can move to
the new one and then get rid of the older one.

Thanks,

Diego

On Mon, Oct 12, 2009 at 10:11 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Hello,
>
> I have been doing some work recently on the FreeSWITCH wiki, to improve
> things.
>
> You can see some of my work here:
>
>
> http://wiki.freeswitch.org/index.php?title=Special:Contributions&limit=500&target=Diego.viola
>
> I am trying to polish the wiki and give it a more professional and clean
> look, this means that we need to enforce some guidelines and strive to make
> the wiki even better.
>
> How can we do this?
>
> Well, the first thing is to review everything we have, right now we have
> too many separated and little pages here and there that no one cares or
> read, we should avoid doing little or small pages. We need to correct typos,
> I suggest that people click on "Random page" link and start correcting
> typos, install an English (US) dictionary in your browser and enable Spell
> Checking and start correcting these typos if you want to help, also, make
> acronyms and initialisms all-uppercase when you make one.
>
> When you are adding documentation make sure you don't do a separate page
> for an existing page, etc. and we also need to define a size for titles,
> body, etc, so different pages don't look different from each other, I will
> make sure I add all this into our guidelines and that all pages follow this.
>
> My goal is to have great documentation, similar to the Apache
> documentation:
> http://httpd.apache.org/docs/2.2/
>
> Or even better, I hope you like this idea.
>
> Diego
>
>
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From anthony.minessale at gmail.com  Mon Oct 12 15:28:11 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 12 Oct 2009 17:28:11 -0500
Subject: [Freeswitch-users] Upgrading causes no answer
In-Reply-To: <00ea01ca469b$7929faf0$6b7df0d0$@com>
References: <00ea01ca469b$7929faf0$6b7df0d0$@com>
Message-ID: <191c3a030910121528w1adb6779xb9923d2a8cdd5943@mail.gmail.com>

if you are using ringback variable you now must also use
ignore_early_media=true either exported from A leg or in {} on the b leg
dial string to get the original behavior.



On Tue, Oct 6, 2009 at 10:41 AM, Lars Zeb <larclap at yahoo.com> wrote:

>  http://pastebin.freeswitch.org/10612
>
>
>
> I having been running v14996 OK for a while. I have upgraded a couple of
> times after, but every time, an inbound call is hung up on. The only thing
> that has changed is the upgrade. This morning I upgraded to v15098 and the
> problem persists.
>
>
>
> I believe it has to do with a lua script I use for inbound calls. Reading
> from the log, just after the script is launched, the following two lines
> appear:
>
>
>
> switch_cpp.cpp:1116  session not ready
>
> switch_cpp.cpp:925 destroy/unlink session from object
>
>
>
> Has something changed recently with lua processing? Is there something in
> the lua script which is causing the problem?
>
>
>
> I would appreciate any help.
>
>
>
> Thanks, Lars
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From gabe at gundy.org  Mon Oct 12 15:33:33 2009
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 12 Oct 2009 16:33:33 -0600
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <87f2f3b90910121035v5cfce2bel7d32da1671e325b6@mail.gmail.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<903da5680910081201p427a8358n768edb61a96674a4@mail.gmail.com>
	<87f2f3b90910081538l6e4e4b7dv3aedb2c4d1af58bc@mail.gmail.com>
	<903da5680910082238x27bff476s8bea0e74ab516a8d@mail.gmail.com>
	<87f2f3b90910082308i7f6144d9kc887596be54dd609@mail.gmail.com>
	<903da5680910082341r485092vb0cd10c43c4a2534@mail.gmail.com>
	<219716B3-C355-4128-A333-1459AC63C364@jerris.com>
	<86a32abc0910111506m4e6d2740tbd3ac5fe0f59178@mail.gmail.com>
	<5162D0B7-357C-4D0E-9913-46564A8C0B1D@freeswitch.org>
	<87f2f3b90910121035v5cfce2bel7d32da1671e325b6@mail.gmail.com>
Message-ID: <903da5680910121533wed3544ek6929ee9191a90848@mail.gmail.com>

On Mon, Oct 12, 2009 at 11:35 AM, Michael Collins <msc at freeswitch.org> wrote:
> It seemed appropriate to do so, therefore I added a small snippet on the
> documentation guidelines:
> http://wiki.freeswitch.org/wiki/Documentation_guidelines#Keep_It_Professional


Good guidelines...


Just in time to help the author of this lovely bit:
'I would love some brief descriptions of what you guys are doing with
FreeSWITCH so I can write an article to shove down the throats of
idiots who keep saying that "FreeSWITCH is an Asterisk alternative
that offers basic PBX functionality."'


Link:
http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_09


Best,
Gabe



From msc at freeswitch.org  Mon Oct 12 15:38:27 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 15:38:27 -0700
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <86a32abc0910121511m4241faefg4b8d1caf3b0f7a87@mail.gmail.com>
References: <86a32abc0910121511m4241faefg4b8d1caf3b0f7a87@mail.gmail.com>
Message-ID: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>

On Mon, Oct 12, 2009 at 3:11 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Hello,
>
> I have been doing some work recently on the FreeSWITCH wiki, to improve
> things.
>
> You can see some of my work here:
>
>
> http://wiki.freeswitch.org/index.php?title=Special:Contributions&limit=500&target=Diego.viola
>
> I am trying to polish the wiki and give it a more professional and clean
> look, this means that we need to enforce some guidelines and strive to make
> the wiki even better.
>
> How can we do this?
>
> Well, the first thing is to review everything we have, right now we have
> too many separated and little pages here and there that no one cares or
> read, we should avoid doing little or small pages. We need to correct typos,
> I suggest that people click on "Random page" link and start correcting
> typos, install an English (US) dictionary in your browser and enable Spell
> Checking and start correcting these typos if you want to help, also, make
> acronyms and initialisms all-uppercase when you make one.
>
> When you are adding documentation make sure you don't do a separate page
> for an existing page, etc. and we also need to define a size for titles,
> body, etc, so different pages don't look different from each other, I will
> make sure I add all this into our guidelines and that all pages follow this.
>
> My goal is to have great documentation, similar to the Apache
> documentation:
> http://httpd.apache.org/docs/2.2/
>
> Or even better, I hope you like this idea.
>

Diego,

Thank you so much for all of your help on this. Many in the community have
seen your work - it has not gone unnoticed and it most truly is appreciated.
I like the idea of improving the documentation. One thing we need to do is
re-think the organization. In many cases the issue with the docs isn't that
they aren't complete, but rather that they are hard to find. It's all about
organization. I'm definitely open to ideas.

For those who are interested in helping out with the wiki please let me know
of your availability and skill set. I am maintaining a list of volunteers.

Lastly, if you want to talk about documentation in real time please join us
in #freeswitch-docs.

Thanks,
MC
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From msc at freeswitch.org  Mon Oct 12 15:41:48 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 15:41:48 -0700
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <903da5680910121533wed3544ek6929ee9191a90848@mail.gmail.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<87f2f3b90910081538l6e4e4b7dv3aedb2c4d1af58bc@mail.gmail.com>
	<903da5680910082238x27bff476s8bea0e74ab516a8d@mail.gmail.com>
	<87f2f3b90910082308i7f6144d9kc887596be54dd609@mail.gmail.com>
	<903da5680910082341r485092vb0cd10c43c4a2534@mail.gmail.com>
	<219716B3-C355-4128-A333-1459AC63C364@jerris.com>
	<86a32abc0910111506m4e6d2740tbd3ac5fe0f59178@mail.gmail.com>
	<5162D0B7-357C-4D0E-9913-46564A8C0B1D@freeswitch.org>
	<87f2f3b90910121035v5cfce2bel7d32da1671e325b6@mail.gmail.com>
	<903da5680910121533wed3544ek6929ee9191a90848@mail.gmail.com>
Message-ID: <87f2f3b90910121541u180f10fdj18f97f0d8981b097@mail.gmail.com>

On Mon, Oct 12, 2009 at 3:33 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Mon, Oct 12, 2009 at 11:35 AM, Michael Collins <msc at freeswitch.org>
> wrote:
> > It seemed appropriate to do so, therefore I added a small snippet on the
> > documentation guidelines:
> >
> http://wiki.freeswitch.org/wiki/Documentation_guidelines#Keep_It_Professional
>
>
> Good guidelines...
>
>
> Just in time to help the author of this lovely bit:
> 'I would love some brief descriptions of what you guys are doing with
> FreeSWITCH so I can write an article to shove down the throats of
> idiots who keep saying that "FreeSWITCH is an Asterisk alternative
> that offers basic PBX functionality."'
>

Touch?, my good man, touch?. I'll fix that right away. :)
-MC
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From frank at carmickle.com  Mon Oct 12 16:02:47 2009
From: frank at carmickle.com (Frank Carmickle)
Date: Mon, 12 Oct 2009 19:02:47 -0400
Subject: [Freeswitch-users] torrents
Message-ID: <20091012230246.GF25698@base.carmickle.com>

Hello 
I would like to watch these files.  Can someone seed please.  Thank you.

--FC



From quentusrex at gmail.com  Mon Oct 12 16:15:44 2009
From: quentusrex at gmail.com (William King)
Date: Mon, 12 Oct 2009 16:15:44 -0700
Subject: [Freeswitch-users] torrents
In-Reply-To: <20091012230246.GF25698@base.carmickle.com>
References: <20091012230246.GF25698@base.carmickle.com>
Message-ID: <4AD3B8A0.5030501@gmail.com>

Opps. didn't realize it was closed. Here you go. Moving it to my seedbox
for permanent seeding.

-William King

Frank Carmickle wrote:
> Hello 
> I would like to watch these files.  Can someone seed please.  Thank you.
>
> --FC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From djbinter at yahoo.com  Mon Oct 12 16:45:27 2009
From: djbinter at yahoo.com (DJB)
Date: Mon, 12 Oct 2009 16:45:27 -0700 (PDT)
Subject: [Freeswitch-users] Problem with Luasql on x86_64
Message-ID: <626589.60602.qm@web37505.mail.mud.yahoo.com>

Hello,

I am trying to use luasql with freeswitch but having a hard time compile it with x86_64 Server running CentOS 5.3

Linux  2.6.18-128.el5 #1 SMP Wed Jan 21 10:41:14 EST 2009 x86_64 x86_64 x86_64 GNU/Linux

[root at fsx1 luasql-2.1.1]# make
export MACOSX_DEPLOYMENT_TARGET="10.3"; gcc -O2 -Wall -Wmissing-prototypes -Wmissing-declarations -ansi -pedantic -I../compat/src -I/opt/local/include -I/usr/local/include  -o src/sqlite3.so -shared  src/luasql.o src/ls_sqlite3.o  -L/opt/local/lib -lsqlite3
/usr/bin/ld: src/luasql.o: relocation R_X86_64_32 against `a local symbol' can not be used when making a shared object; recompile with -fPIC
src/luasql.o: could not read symbols: Bad value
collect2: ld returned 1 exit status
make: *** [src/sqlite3.so] Error 1


Any suggestions.

Thank you.


      



From stkn at freeswitch.org  Mon Oct 12 16:56:01 2009
From: stkn at freeswitch.org (Stefan Knoblich)
Date: Tue, 13 Oct 2009 01:56:01 +0200
Subject: [Freeswitch-users] Problem with Luasql on x86_64
In-Reply-To: <626589.60602.qm@web37505.mail.mud.yahoo.com>
References: <626589.60602.qm@web37505.mail.mud.yahoo.com>
Message-ID: <4AD3C211.5070305@freeswitch.org>

DJB wrote:
> Hello,
> 
> I am trying to use luasql with freeswitch but having a hard time compile it with x86_64 Server running CentOS 5.3
> 
> Linux  2.6.18-128.el5 #1 SMP Wed Jan 21 10:41:14 EST 2009 x86_64 x86_64 x86_64 GNU/Linux
> 
> [root at fsx1 luasql-2.1.1]# make
> export MACOSX_DEPLOYMENT_TARGET="10.3"; gcc -O2 -Wall -Wmissing-prototypes -Wmissing-declarations -ansi -pedantic -I../compat/src -I/opt/local/include -I/usr/local/include  -o src/sqlite3.so -shared  src/luasql.o src/ls_sqlite3.o  -L/opt/local/lib -lsqlite3
> /usr/bin/ld: src/luasql.o: relocation R_X86_64_32 against `a local symbol' can not be used when making a shared object; recompile with -fPIC
> src/luasql.o: could not read symbols: Bad value
> collect2: ld returned 1 exit status
> make: *** [src/sqlite3.so] Error 1
> 
> 
> Any suggestions.
> 
> Thank you.


Well, i'm sure people could've helped you on IRC if you stay for more than two minutes,
after asking your question...



From Mailings at kh-dev.de  Mon Oct 12 17:01:30 2009
From: Mailings at kh-dev.de (Klaus Hochlehnert)
Date: Tue, 13 Oct 2009 02:01:30 +0200
Subject: [Freeswitch-users] Problem with transfer and Snom phones...
Message-ID: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330251@srv02.khdev.corp>

Hi,

I'm facing an issue and I don't know why this happens or what I can do to solve this.

Here's the scenario:
- FS call timeout set to 30 sec
- Setting continue_on_fail=true and hangup_after_bridge=false
- 1 Snom phone and an external SIP/ISDN gateway (Lancom) connected to FS established a call and talk to each other (dir. from GW to Phone)
- The called phone transfers the call to another internal Snom using the transfer button on the phone
- The third phone doesn't answer and after 30 sec the call gets hung up with status NORMAL_CLEARING

Actually I want the call back to the initiator of the transfer if the 3rd phone doesn't answer.

Can anyone tell me why this happens and how I can set it up to do what I want?

Thanks, Klaus

P.S.: I'm using FS trunk from last week and newest Snom Firmware

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From gabe at gundy.org  Mon Oct 12 17:26:42 2009
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 12 Oct 2009 18:26:42 -0600
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <87f2f3b90910121541u180f10fdj18f97f0d8981b097@mail.gmail.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<903da5680910082238x27bff476s8bea0e74ab516a8d@mail.gmail.com>
	<87f2f3b90910082308i7f6144d9kc887596be54dd609@mail.gmail.com>
	<903da5680910082341r485092vb0cd10c43c4a2534@mail.gmail.com>
	<219716B3-C355-4128-A333-1459AC63C364@jerris.com>
	<86a32abc0910111506m4e6d2740tbd3ac5fe0f59178@mail.gmail.com>
	<5162D0B7-357C-4D0E-9913-46564A8C0B1D@freeswitch.org>
	<87f2f3b90910121035v5cfce2bel7d32da1671e325b6@mail.gmail.com>
	<903da5680910121533wed3544ek6929ee9191a90848@mail.gmail.com>
	<87f2f3b90910121541u180f10fdj18f97f0d8981b097@mail.gmail.com>
Message-ID: <903da5680910121726s17e4dd39u306c2255cd7479ac@mail.gmail.com>

On Mon, Oct 12, 2009 at 4:41 PM, Michael Collins <msc at freeswitch.org> wrote:
> On Mon, Oct 12, 2009 at 3:33 PM, Gabriel Gunderson <gabe at gundy.org> wrote:
>> On Mon, Oct 12, 2009 at 11:35 AM, Michael Collins <msc at freeswitch.org>
>> wrote:
>> > It seemed appropriate to do so, therefore I added a small snippet on the
>> > documentation guidelines:
>> >
>> > http://wiki.freeswitch.org/wiki/Documentation_guidelines#Keep_It_Professional
>>
>> Good guidelines...
>>
>> Just in time to help the author of this lovely bit:
>> 'I would love some brief descriptions of what you guys are doing with
>> FreeSWITCH so I can write an article to shove down the throats of
>> idiots who keep saying that "FreeSWITCH is an Asterisk alternative
>> that offers basic PBX functionality."'
>
> Touch?, my good man, touch?. I'll fix that right away. :)

Never let it be said that you're not a good sport.

:)

Gabe



From djbinter at yahoo.com  Mon Oct 12 17:28:13 2009
From: djbinter at yahoo.com (DJB)
Date: Mon, 12 Oct 2009 17:28:13 -0700 (PDT)
Subject: [Freeswitch-users] Problem with Luasql on x86_64
In-Reply-To: <626589.60602.qm@web37505.mail.mud.yahoo.com>
References: <626589.60602.qm@web37505.mail.mud.yahoo.com>
Message-ID: <536709.63053.qm@web37502.mail.mud.yahoo.com>

If anyone ran into the same problem, here is the fix that I got from [stkn] on IRC:

make CFLAGS="-O2 -fPIC" 

I hope it might help someone in the future.

Regards.



----- Original Message ----
From: DJB <djbinter at yahoo.com>
To: FREESWITCH-USERS MAILING LIST <freeswitch-users at lists.freeswitch.org>
Sent: Mon, October 12, 2009 4:45:27 PM
Subject: [Freeswitch-users] Problem with Luasql on x86_64

Hello,

I am trying to use luasql with freeswitch but having a hard time compile it with x86_64 Server running CentOS 5.3

Linux  2.6.18-128.el5 #1 SMP Wed Jan 21 10:41:14 EST 2009 x86_64 x86_64 x86_64 GNU/Linux

[root at fsx1 luasql-2.1.1]# make
export MACOSX_DEPLOYMENT_TARGET="10.3"; gcc -O2 -Wall -Wmissing-prototypes -Wmissing-declarations -ansi -pedantic -I../compat/src -I/opt/local/include -I/usr/local/include  -o src/sqlite3.so -shared  src/luasql.o src/ls_sqlite3.o  -L/opt/local/lib -lsqlite3
/usr/bin/ld: src/luasql.o: relocation R_X86_64_32 against `a local symbol' can not be used when making a shared object; recompile with -fPIC
src/luasql.o: could not read symbols: Bad value
collect2: ld returned 1 exit status
make: *** [src/sqlite3.so] Error 1


Any suggestions.

Thank you.


      

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



      



From mcampbellsmith at gmail.com  Mon Oct 12 19:31:14 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 13 Oct 2009 13:31:14 +1100
Subject: [Freeswitch-users] Question about fax tone detection
In-Reply-To: <87f2f3b90910121100l40a1b22hff33df032411cf31@mail.gmail.com>
References: <1255345270255-3807298.post@n2.nabble.com>
	<87f2f3b90910121100l40a1b22hff33df032411cf31@mail.gmail.com>
Message-ID: <33c87fa30910121931y48c57a94sb8fce42cc946c552@mail.gmail.com>

This is what I have in my dialplan and the fax is detected
beautifully.  Note that in my case, extension 1000 will ring for a
second or two before the fax is detected.  So in your example, the fax
does not have time to be detected, the dialplan exists and the call is
hungup.

When the fax is detected, the call is transferred to the receivefax
extension in context features.  The extension 1000 does not have to be
answered for the transfer to occur.

    <extension name="Local_1000s">
      <condition field="destination_number" expression="^(10[01][0-9])$">
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
        <action application="set" data="ringback=${au-ring}"/>
        <action application="tone_detect" data="fax 1100 r +5000
transfer 'receivefax XML features' 1" />
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="bridge" data="user/${dialed_extension}@${domain}"/>
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="voicemail" data="default ${domain_name}
${dialed_extension}"/>
      </condition>
   </extension>

and in the features context I have
    <extension name="receivefax">
      <condition field="destination_number" expression="^receivefax$">
        <action application="answer" />
        <action application="playback" data="silence_stream://2000"/>
        <action application="rxfax"
data="//usr//local//freeswitch//storage//${caller_id_number}-${uuid}.rxfax.tiff"/>
        <action application="system"
data="/usr/local/freeswitch/scripts/emailfax.sh
/usr/local/freeswitch/storage/${caller_id_number}-${uuid}.rxfax.tiff"/>
        <action application="hangup"/>
      </condition>
    </extension>
On Tue, Oct 13, 2009 at 5:00 AM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Mon, Oct 12, 2009 at 4:01 AM, homqua <ngay01042005 at gmail.com> wrote:
>>
>> Hi,
>> I have implemented the solution for tone detection in wiki, and also
>> answer
>> the channel before detecting the tone:
>> <condition field="destination_number" expression="^(055138419992)$">
>> ? ?<action application="answer"/>
>> ? ?<action application="tone_detect" data="fax 1100 r +15000 transfer fax
>> XML default"/>
>>
>>
>> <extension name="fax">
>> ? ? ?<condition field="destination_number" expression="^fax$">
>> ? ? ? ?<action application="answer" />
>> ? ? ? ?<action application="sleep" data="1000"/>
>> ? ? ? ?<action application="rxfax"
>>
>> data="/usr/local/freeswitch/storage/fax/${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}.tiff"/>
>> ? ?<action application="set" data="fax_mode=recv"/>
>> ? ? ? ?<action application="hangup"/>
>> ? ? ?</condition>
>> ? ?</extension>
>>
>>
>> But FS cannot recognize the tone, and therefore cannot move to fax
>> extension. ?Below are the error in FS:
>>
>> 2009-10-12 10:57:16.702287 [NOTICE] switch_channel.c:602 New Channel
>> sofia/external/anonymous at anonymous.invalid
>> [c431f0a3-9231-4724-ba39-9e4ef7edfca2]
>> 2009-10-12 10:57:16.703413 [INFO] mod_dialplan_xml.c:315 Processing
>> Anonymous->055138419992 in context public
>> 2009-10-12 10:57:16.719288 [NOTICE] switch_ivr.c:1349 Transfer
>> sofia/external/anonymous at anonymous.invalid to XML[055138419992 at default]
>> 2009-10-12 10:57:16.719288 [INFO] mod_dialplan_xml.c:315 Processing
>> Anonymous->055138419992 in context default
>> 2009-10-12 10:57:16.722289 [NOTICE] mod_dptools.c:649 Channel
>> [sofia/external/anonymous at anonymous.invalid] has been answered
>> 2009-10-12 10:57:16.722289 [NOTICE] mod_dptools.c:1324 Enabling tone
>> detection 'fax' '1100'
>> 2009-10-12 10:57:16.723302 [NOTICE] switch_core_state_machine.c:179 Hangup
>> sofia/external/anonymous at anonymous.invalid [CS_EXECUTE] [NORMAL_CLEARING]
>> 2009-10-12 10:57:16.740285 [NOTICE] switch_core_session.c:1086 Session 1
>> (sofia/external/anonymous at anonymous.invalid) Ended
>> 2009-10-12 10:57:16.740285 [NOTICE] switch_core_session.c:1088 Close
>> Channel
>> sofia/external/anonymous at anonymous.invalid [CS_DESTROY]
>>
>> And the trace for SIP messages: ?http://pastebin.com/m4e47e7d9
>>
>> If anyone has any idea, tell me please.
>> Thanks.
>
> I think the trouble here is that you don't have anything else in the
> dialplan after the tone_detect. The tone_detect app is non-block, which
> means that it doesn't sit there and wait for a tone. If you want the
> dialplan to sit and wait then do a sleep app after your tone_detect. The
> other question I would have is this: what happens if the incoming call is
> not a fax? What do you want to do then?
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From diego.viola at gmail.com  Mon Oct 12 19:37:45 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 13 Oct 2009 02:37:45 +0000
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
References: <86a32abc0910121511m4241faefg4b8d1caf3b0f7a87@mail.gmail.com>
	<87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
Message-ID: <86a32abc0910121937l5fe537fcu48089d1c5e1bfc1@mail.gmail.com>

Sure, I'm happy to put my little two cents to help the project :).

Diego

On Mon, Oct 12, 2009 at 10:38 PM, Michael Collins <msc at freeswitch.org>wrote:

>
> On Mon, Oct 12, 2009 at 3:11 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> Hello,
>>
>> I have been doing some work recently on the FreeSWITCH wiki, to improve
>> things.
>>
>> You can see some of my work here:
>>
>>
>> http://wiki.freeswitch.org/index.php?title=Special:Contributions&limit=500&target=Diego.viola
>>
>> I am trying to polish the wiki and give it a more professional and clean
>> look, this means that we need to enforce some guidelines and strive to make
>> the wiki even better.
>>
>> How can we do this?
>>
>> Well, the first thing is to review everything we have, right now we have
>> too many separated and little pages here and there that no one cares or
>> read, we should avoid doing little or small pages. We need to correct typos,
>> I suggest that people click on "Random page" link and start correcting
>> typos, install an English (US) dictionary in your browser and enable Spell
>> Checking and start correcting these typos if you want to help, also, make
>> acronyms and initialisms all-uppercase when you make one.
>>
>> When you are adding documentation make sure you don't do a separate page
>> for an existing page, etc. and we also need to define a size for titles,
>> body, etc, so different pages don't look different from each other, I will
>> make sure I add all this into our guidelines and that all pages follow this.
>>
>> My goal is to have great documentation, similar to the Apache
>> documentation:
>> http://httpd.apache.org/docs/2.2/
>>
>> Or even better, I hope you like this idea.
>>
>
> Diego,
>
> Thank you so much for all of your help on this. Many in the community have
> seen your work - it has not gone unnoticed and it most truly is appreciated.
> I like the idea of improving the documentation. One thing we need to do is
> re-think the organization. In many cases the issue with the docs isn't that
> they aren't complete, but rather that they are hard to find. It's all about
> organization. I'm definitely open to ideas.
>
> For those who are interested in helping out with the wiki please let me
> know of your availability and skill set. I am maintaining a list of
> volunteers.
>
> Lastly, if you want to talk about documentation in real time please join us
> in #freeswitch-docs.
>
> Thanks,
> MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Mon Oct 12 20:16:38 2009
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 13 Oct 2009 11:16:38 +0800
Subject: [Freeswitch-users] Re corded file as voicemail.
In-Reply-To: <25850181.post@talk.nabble.com>
References: <25850181.post@talk.nabble.com>
Message-ID: <23f91030910122016q33012754n791ab9629e59c7a0@mail.gmail.com>

It was a problem and has been fixed in the last trunk. Just update to the
latest code should be ok.

btw, the developers using jira to track bugs, so feel free to report one (as
you see http://jira.freeswitch.org/browse/FSCORE-463) if you think it's a
bug next time.

2009/10/12 Nagalenoj <nagalenoj at gmail.com>

>
> Whats the conclusion.?!
>
>
> Brian West-3 wrote:
> >
> > Please open a jira please this did work but a recent change in
> > switch_core_codec caused this to appear I usually test this regularly
> > but haven't run thru a full run of tests lately.
> > /b
> >
>
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From dujinfang at gmail.com  Mon Oct 12 20:17:37 2009
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 13 Oct 2009 11:17:37 +0800
Subject: [Freeswitch-users] Re corded file as voicemail.
In-Reply-To: <BA9A12F7-EB74-485D-8612-12E448621257@freeswitch.org>
References: <25850181.post@talk.nabble.com>
	<BEDBBF51-2D48-4CDE-A467-11D5BBB458E1@freeswitch.org>
	<23f91030910112315h5ea79b81ue55381a3caff0173@mail.gmail.com>
	<BA9A12F7-EB74-485D-8612-12E448621257@freeswitch.org>
Message-ID: <23f91030910122017j6efdfa26l958704ce8d67e53e@mail.gmail.com>

I won't try until I need that, but I believe it works. Thanks Brian.

2009/10/13 Brian West <brian at freeswitch.org>

> Fixed... svn up.
> /b
>
> On Oct 12, 2009, at 1:15 AM, Seven Du wrote:
>
> http://jira.freeswitch.org/browse/MODCODEC-15
>
> Is it ok I assigned to you ?
>
> Thanks.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From srinivas.ksvreddy at gmail.com  Mon Oct 12 21:44:01 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 13 Oct 2009 10:14:01 +0530
Subject: [Freeswitch-users] mod_sofia.c registered calls how to know
In-Reply-To: <1255347432.3724.4.camel@eli-desktop>
References: <f8af5740910070748g1efee71ft5841b6fa4d2412c8@mail.gmail.com>
	<14A1216A-61C3-4C03-B710-57D2E7F4CD7A@jerris.com>
	<f8af5740910112114k3206c3cbq5757e2ff7de81162@mail.gmail.com>
	<23f91030910120419o74f53279v5f061988ff585d56@mail.gmail.com>
	<1255347432.3724.4.camel@eli-desktop>
Message-ID: <f8af5740910122144g6a46eabbxd498ba2a6044a0f7@mail.gmail.com>

exactly that is what i am trying to do now, if any help that would be great.

On Mon, Oct 12, 2009 at 5:07 PM, Eli Hayun <elihay at savion.huji.ac.il> wrote:

>  Is it possible to keep a list of registered phone, and when FS will start
> it will register them all automatically?
>
>
>
> On Mon, 2009-10-12 at 13:19 +0200, Seven Du wrote:
>
> try open YOUR_FreeSWITCH_INSTALL_DIR/db/*.db, you need sqlite3 to open
> them. not sure how to do that on windows, but on linux:
>
>
> # sqlite3 xx.db
> sqlite> select * from sip_registration;
>
>  2009/10/12 srinivasula reddy <srinivas.ksvreddy at gmail.com>
>
> Hi Mike,
>
> Thanks for your valuable reply,
> when i install freeswitch1.0.2 in my machine(Windows xp operation system) i
> dont have any databasae installed in my system, then from sqllite will come
> into picture, and how can i see the registered users data from sqllite.
>
>
> Thanks
> Srinivas
>
>
>   On Mon, Oct 12, 2009 at 3:04 AM, Michael Jerris <mike at jerris.com> wrote:
>
>
>
>    On Oct 7, 2009, at 10:48 AM, srinivasula reddy wrote:
>
>
>
> Hi
>
> can any please tell me where registered calls are stored, so when incoming
> call came to mod_sofia.c how it will check it is registered or not?\\
>
>
>
>    Calls are not registered and calls have nothing to do with
> registration.  Users are registered so that you may send calls to them.
>  Registration data is stored either in a sqlite database, or optionally if
> you setup odbc, in another database of your choice.  If you try to send a
> call to an unregistered user in the dialplan using the proper syntax to send
> calls to registered users (see the wiki for more details), and that user is
> not registered, the bridge app will fail, optionally letting you continue on
> in the dialplan based on variables such as continue_on_fail and
> hangup_after_bridge.  You can use the sofia_contact function to see if there
> is anyone registered to a specific user.
>
>
>
>    Mike
>
>
>
>
>
>    _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
> --
> Srinivasula Reddy K
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Srinivasula Reddy K
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From msc at freeswitch.org  Mon Oct 12 21:51:48 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Oct 2009 21:51:48 -0700
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <903da5680910121726s17e4dd39u306c2255cd7479ac@mail.gmail.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<87f2f3b90910082308i7f6144d9kc887596be54dd609@mail.gmail.com>
	<903da5680910082341r485092vb0cd10c43c4a2534@mail.gmail.com>
	<219716B3-C355-4128-A333-1459AC63C364@jerris.com>
	<86a32abc0910111506m4e6d2740tbd3ac5fe0f59178@mail.gmail.com>
	<5162D0B7-357C-4D0E-9913-46564A8C0B1D@freeswitch.org>
	<87f2f3b90910121035v5cfce2bel7d32da1671e325b6@mail.gmail.com>
	<903da5680910121533wed3544ek6929ee9191a90848@mail.gmail.com>
	<87f2f3b90910121541u180f10fdj18f97f0d8981b097@mail.gmail.com>
	<903da5680910121726s17e4dd39u306c2255cd7479ac@mail.gmail.com>
Message-ID: <87f2f3b90910122151s324d5dbfpa62901ac653b5c84@mail.gmail.com>

On Mon, Oct 12, 2009 at 5:26 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Mon, Oct 12, 2009 at 4:41 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> > On Mon, Oct 12, 2009 at 3:33 PM, Gabriel Gunderson <gabe at gundy.org>
> wrote:
> >> On Mon, Oct 12, 2009 at 11:35 AM, Michael Collins <msc at freeswitch.org>
> >> wrote:
> >> > It seemed appropriate to do so, therefore I added a small snippet on
> the
> >> > documentation guidelines:
> >> >
> >> >
> http://wiki.freeswitch.org/wiki/Documentation_guidelines#Keep_It_Professional
> >>
> >> Good guidelines...
> >>
> >> Just in time to help the author of this lovely bit:
> >> 'I would love some brief descriptions of what you guys are doing with
> >> FreeSWITCH so I can write an article to shove down the throats of
> >> idiots who keep saying that "FreeSWITCH is an Asterisk alternative
> >> that offers basic PBX functionality."'
> >
> > Touch?, my good man, touch?. I'll fix that right away. :)
>
> Never let it be said that you're not a good sport.
>
> :)
>
> Gabe
>
> Well, I'm occasionally a good sport. ;)
-MC
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From mctch at yahoo.com  Mon Oct 12 22:52:03 2009
From: mctch at yahoo.com (Mark Crane)
Date: Mon, 12 Oct 2009 22:52:03 -0700 (PDT)
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
Message-ID: <891080.87141.qm@web56406.mail.re3.yahoo.com>

"In many cases the issue with the docs isn't that they aren't complete but rather that they are hard to find."

Agreed! The biggest problem with the wiki is that it is hard to find things. 

How do books solve this problem they use an index. It is quite surprising that the wiki software doesn't come with ability to do this basic task automatically and instead relies solely upon a manually created index. It would be extremely beneficial to have a index of all pages and topics that are available on the wiki. The average person may only see the main links in the documentation and not realize there are actually hundreds of pages.

There used to be a pdf that was auto generated from the wiki and it gave a much easier view of all the pages on the wiki.

Mark J Crane



--- On Mon, 10/12/09, Michael Collins <msc at freeswitch.org> wrote:

From: Michael Collins <msc at freeswitch.org>
Subject: Re: [Freeswitch-users] Some documentation thoughts
To: freeswitch-users at lists.freeswitch.org
Date: Monday, October 12, 2009, 4:38 PM


On Mon, Oct 12, 2009 at 3:11 PM, Diego Viola <diego.viola at gmail.com> wrote:

Hello,

I have been doing some work recently on the FreeSWITCH wiki, to improve things.

You can see some of my work here:

http://wiki.freeswitch.org/index.php?title=Special:Contributions&limit=500&target=Diego.viola



I am trying to polish the wiki and give it a more professional and clean look, this means that we need to enforce some guidelines and strive to make the wiki even better.

How can we do this?

Well, the first thing is to review everything we have, right now we have too many separated and little pages here and there that no one cares or read, we should avoid doing little or small pages. We need to correct typos, I suggest that people click on "Random page" link and start correcting typos, install an English (US) dictionary in your browser and enable Spell Checking and start correcting these typos if you want to help, also, make acronyms and initialisms all-uppercase when you make one.



When you are adding documentation make sure you don't do a separate page for an existing page, etc. and we also need to define a size for titles, body, etc, so different pages don't look different from each other, I will make sure I add all this into our guidelines and that all pages follow this.



My goal is to have great documentation, similar to the Apache documentation:
http://httpd.apache.org/docs/2.2/

Or even better, I hope you like this idea.


Diego,

Thank you so much for all of your help on this. Many in the community have seen your work - it has not gone unnoticed and it most truly is appreciated. I like the idea of improving the documentation. One thing we need to do is re-think the organization. In many cases the issue with the docs isn't that they aren't complete, but rather that they are hard to find. It's all about organization. I'm definitely open to ideas.


For those who are interested in helping out with the wiki please let me know of your availability and skill set. I am maintaining a list of volunteers.

Lastly, if you want to talk about documentation in real time please join us in #freeswitch-docs.


Thanks,
MC



-----Inline Attachment Follows-----

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



      
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From velu.technical at gmail.com  Mon Oct 12 22:59:49 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Tue, 13 Oct 2009 11:29:49 +0530
Subject: [Freeswitch-users] Asynchronous execution in ESL.pm
Message-ID: <1452e2980910122259w3e007870m4bda8ce673d584b2@mail.gmail.com>

Dear All,
         I have set my socket mode as full. I have used ESL.pm to develop an
IVR. I encounter the situation that I need play some music file while
executing some external application. So, I used executeAsync method  to play
the music file. But the executeAsync application didn't work.

       What is the problem? Please help me....

Thanks,
Velusamy.
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From brian at freeswitch.org  Mon Oct 12 23:00:11 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Oct 2009 23:00:11 -0700
Subject: [Freeswitch-users] Problem with transfer and Snom phones...
In-Reply-To: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330251@srv02.khdev.corp>
References: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330251@srv02.khdev.corp>
Message-ID: <BD3BCAC6-5CA0-4DBD-AF75-5DC5B106F3C1@freeswitch.org>

Update to latest trunk and try again... then if it persists then  
collect all the sip traces and debug logs as per the wiki on reporting  
bugs.

Thanks,
Brian

On Oct 12, 2009, at 5:01 PM, Klaus Hochlehnert wrote:

> P.S.: I?m using FS trunk from last week and newest Snom Firmware

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From srinivas.ksvreddy at gmail.com  Mon Oct 12 23:01:55 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 13 Oct 2009 11:31:55 +0530
Subject: [Freeswitch-users] Groups information in sqllite
Message-ID: <f8af5740910122301u338ca128macfc0b769ea0e961@mail.gmail.com>

Hi,

can any know where group information is exactly stored in sqllite database,
i have seen sip_registration here i can find the registered users,
in the same way how i can i find the group information, and which user
belongs to which user?
any help would be great.

thanks
-- 
Srinivasula Reddy K
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From brian at freeswitch.org  Mon Oct 12 23:02:14 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Oct 2009 23:02:14 -0700
Subject: [Freeswitch-users] mod_sofia.c registered calls how to know
In-Reply-To: <1255347432.3724.4.camel@eli-desktop>
References: <f8af5740910070748g1efee71ft5841b6fa4d2412c8@mail.gmail.com>
	<14A1216A-61C3-4C03-B710-57D2E7F4CD7A@jerris.com>
	<f8af5740910112114k3206c3cbq5757e2ff7de81162@mail.gmail.com>
	<23f91030910120419o74f53279v5f061988ff585d56@mail.gmail.com>
	<1255347432.3724.4.camel@eli-desktop>
Message-ID: <FE9BBD84-64D4-4219-AA90-A6A5FF842446@freeswitch.org>

Eli,
	Well FreeSWITCH already keeps a list... but its the phone's job to  
register to FreeSWITCH not the other way around.  Their are various  
ways to accomplish your goals but not sure how well each will work.   
Check out the wiki for sip-force-contact.

/b

On Oct 12, 2009, at 4:37 AM, Eli Hayun wrote:

> Is it possible to keep a list of registered phone, and when FS will  
> start it will register them all automatically?
>




From srinivas.ksvreddy at gmail.com  Mon Oct 12 23:02:28 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 13 Oct 2009 11:32:28 +0530
Subject: [Freeswitch-users] Fwd: Groups information in sqllite
In-Reply-To: <f8af5740910122301u338ca128macfc0b769ea0e961@mail.gmail.com>
References: <f8af5740910122301u338ca128macfc0b769ea0e961@mail.gmail.com>
Message-ID: <f8af5740910122302i47ccde5bo9df2bae965408394@mail.gmail.com>

Hi,

can any know where group information is exactly stored in sqllite database,
i have seen sip_registration here i can find the registered users,
in the same way how i can i find the group information, and which user
belongs to which user?
any help would be great.

thanks
-- 
Srinivasula Reddy K
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From brian at freeswitch.org  Mon Oct 12 23:04:35 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Oct 2009 23:04:35 -0700
Subject: [Freeswitch-users] Asynchronous execution in ESL.pm
In-Reply-To: <1452e2980910122259w3e007870m4bda8ce673d584b2@mail.gmail.com>
References: <1452e2980910122259w3e007870m4bda8ce673d584b2@mail.gmail.com>
Message-ID: <994D92AC-0760-453C-AD7E-4D6B706347AE@freeswitch.org>

You need "async full"

/b

On Oct 12, 2009, at 10:59 PM, velusamy velu wrote:

> What is the problem? Please help me....




From brian at freeswitch.org  Mon Oct 12 23:05:26 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Oct 2009 23:05:26 -0700
Subject: [Freeswitch-users] Groups information in sqllite
In-Reply-To: <f8af5740910122301u338ca128macfc0b769ea0e961@mail.gmail.com>
References: <f8af5740910122301u338ca128macfc0b769ea0e961@mail.gmail.com>
Message-ID: <FD1AC150-0636-4861-8341-58553D3F43A0@freeswitch.org>

The sip_registration table contains the contacts for each registered  
endpoint.  Its not for the directory from a database per se... If you  
wish to serve up your users and groups from a database check out the  
XML Curl wiki page.

/b

On Oct 12, 2009, at 11:01 PM, srinivasula reddy wrote:

> Hi,
>
> can any know where group information is exactly stored in sqllite  
> database, i have seen sip_registration here i can find the  
> registered users,
> in the same way how i can i find the group information, and which  
> user belongs to which user?
> any help would be great.
>
> thanks
> -- 
> Srinivasula Reddy K




From brian at freeswitch.org  Mon Oct 12 23:05:47 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Oct 2009 23:05:47 -0700
Subject: [Freeswitch-users] Fwd: Groups information in sqllite
In-Reply-To: <f8af5740910122302i47ccde5bo9df2bae965408394@mail.gmail.com>
References: <f8af5740910122301u338ca128macfc0b769ea0e961@mail.gmail.com>
	<f8af5740910122302i47ccde5bo9df2bae965408394@mail.gmail.com>
Message-ID: <3AEBF9B1-E366-44FA-8EE0-1265CED486ED@freeswitch.org>

Please DO NOT cross post.

/b

On Oct 12, 2009, at 11:02 PM, srinivasula reddy wrote:

>
>
>
>
> Hi,
>
> can any know where group information is exactly stored in sqllite  
> database, i have seen sip_registration here i can find the  
> registered users,
> in the same way how i can i find the group information, and which  
> user belongs to which user?
> any help would be great.
>
> thanks
> -- 
> Srinivasula Reddy K
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Mon Oct 12 23:08:13 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Oct 2009 23:08:13 -0700
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>
	<881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
Message-ID: <FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>

Does anyone see a problem with hosting mod_h323 in our SVN?  I would  
like to centralize everything we can to reuse our issue tracking  
resources and not fragment the community if possible.

/b

On Oct 12, 2009, at 2:43 PM, Tihomir Culjaga wrote:

> hi,
>
> finally i compiled it right ... had a stupid issue with ekiga and  
> wrong ptlib in place...
>
> anyhow, i loaded the module and will continue the tests  
> tomorrow ...first thing i arrive in my office :P
>
>




From brian at freeswitch.org  Mon Oct 12 23:09:16 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Oct 2009 23:09:16 -0700
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Update: Valet
	Parking
In-Reply-To: <4AD389A8.1080603@gmail.com>
References: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>	<15b9404e0910102018oda8834ah8cc87787ee00c22b@mail.gmail.com>
	<87f2f3b90910121201o422c7891vd30ee3761aef0eb7@mail.gmail.com>
	<4AD389A8.1080603@gmail.com>
Message-ID: <719ED4DA-A7C6-4256-8349-09C874505930@freeswitch.org>

Thats called mod_fifo.

/b

On Oct 12, 2009, at 12:55 PM, William King wrote:

> I don't know if this was mentioned yet. It would be useful to have a  
> way
> to have the parking lot automatically find the next available spot and
> tts it to the person parking the call.
>
> Then the auto unpark would pop off the lowest numbered lot, or return
> fail if there is nobody in the parking lots etc.
>
> -William King




From srinivas.ksvreddy at gmail.com  Mon Oct 12 23:16:55 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 13 Oct 2009 11:46:55 +0530
Subject: [Freeswitch-users] Groups information in sqllite
In-Reply-To: <FD1AC150-0636-4861-8341-58553D3F43A0@freeswitch.org>
References: <f8af5740910122301u338ca128macfc0b769ea0e961@mail.gmail.com>
	<FD1AC150-0636-4861-8341-58553D3F43A0@freeswitch.org>
Message-ID: <f8af5740910122316g7db0ae7ci1f79e3aede07d112@mail.gmail.com>

HI Brain,

thank u very much for valuable reply,

you are right, sip_Registration table contains the registered endpoint
details, in the same way there is any table for groups, how many groups are
there? and information about groups(directory/default.xml this file having
the group configuration).

Thanks
Srinivas

On Tue, Oct 13, 2009 at 11:35 AM, Brian West <brian at freeswitch.org> wrote:

> The sip_registration table contains the contacts for each registered
> endpoint.  Its not for the directory from a database per se... If you
> wish to serve up your users and groups from a database check out the
> XML Curl wiki page.
>
> /b
>
> On Oct 12, 2009, at 11:01 PM, srinivasula reddy wrote:
>
> > Hi,
> >
> > can any know where group information is exactly stored in sqllite
> > database, i have seen sip_registration here i can find the
> > registered users,
> > in the same way how i can i find the group information, and which
> > user belongs to which user?
> > any help would be great.
> >
> > thanks
> > --
> > Srinivasula Reddy K
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Srinivasula Reddy K
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From brian at freeswitch.org  Mon Oct 12 23:25:05 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Oct 2009 23:25:05 -0700
Subject: [Freeswitch-users] Groups information in sqllite
In-Reply-To: <f8af5740910122316g7db0ae7ci1f79e3aede07d112@mail.gmail.com>
References: <f8af5740910122301u338ca128macfc0b769ea0e961@mail.gmail.com>
	<FD1AC150-0636-4861-8341-58553D3F43A0@freeswitch.org>
	<f8af5740910122316g7db0ae7ci1f79e3aede07d112@mail.gmail.com>
Message-ID: <25EFC4E3-88E6-4286-8432-8118CC9E899C@freeswitch.org>

NO this is in the XML... not in the db table.

/b

On Oct 12, 2009, at 11:16 PM, srinivasula reddy wrote:

>  in the same way there is any table for groups, how many groups are   
> there? and information about groups(directory/default.xml this file  
> having the group configuration).




From tculjaga at gmail.com  Mon Oct 12 23:27:39 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 13 Oct 2009 08:27:39 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>
Message-ID: <65d96fc80910122327u54505b44r4ff53e89293bcf9a@mail.gmail.com>

this will be perfect ... but it is up to Yuriy if he is willing to donate
his work...

T.


On Tue, Oct 13, 2009 at 8:08 AM, Brian West <brian at freeswitch.org> wrote:

> Does anyone see a problem with hosting mod_h323 in our SVN?  I would
> like to centralize everything we can to reuse our issue tracking
> resources and not fragment the community if possible.
>
> /b
>
> On Oct 12, 2009, at 2:43 PM, Tihomir Culjaga wrote:
>
> > hi,
> >
> > finally i compiled it right ... had a stupid issue with ekiga and
> > wrong ptlib in place...
> >
> > anyhow, i loaded the module and will continue the tests
> > tomorrow ...first thing i arrive in my office :P
> >
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Mon Oct 12 23:31:39 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Oct 2009 23:31:39 -0700
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910122327u54505b44r4ff53e89293bcf9a@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>
	<65d96fc80910122327u54505b44r4ff53e89293bcf9a@mail.gmail.com>
Message-ID: <5F65174C-2CE6-4007-B288-C81227776281@freeswitch.org>

I wouldn't call it donating per se... Its just giving it a place to  
live with easy access for end users without having to do anything  
extra go get it!  ;)

/b

On Oct 12, 2009, at 11:27 PM, Tihomir Culjaga wrote:

> this will be perfect ... but it is up to Yuriy if he is willing to  
> donate his work...
>
> T.




From srinivas.ksvreddy at gmail.com  Mon Oct 12 23:32:19 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 13 Oct 2009 12:02:19 +0530
Subject: [Freeswitch-users] Groups information in sqllite
In-Reply-To: <25EFC4E3-88E6-4286-8432-8118CC9E899C@freeswitch.org>
References: <f8af5740910122301u338ca128macfc0b769ea0e961@mail.gmail.com>
	<FD1AC150-0636-4861-8341-58553D3F43A0@freeswitch.org>
	<f8af5740910122316g7db0ae7ci1f79e3aede07d112@mail.gmail.com>
	<25EFC4E3-88E6-4286-8432-8118CC9E899C@freeswitch.org>
Message-ID: <f8af5740910122332i6c266250q7a550fb05439ae53@mail.gmail.com>

OK, then when i call to group number(911) how it will call to all the
registered members in group?


On Tue, Oct 13, 2009 at 11:55 AM, Brian West <brian at freeswitch.org> wrote:

> NO this is in the XML... not in the db table.
>
> /b
>
> On Oct 12, 2009, at 11:16 PM, srinivasula reddy wrote:
>
> >  in the same way there is any table for groups, how many groups are
> > there? and information about groups(directory/default.xml this file
> > having the group configuration).
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Srinivasula Reddy K
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From brian at freeswitch.org  Mon Oct 12 23:48:01 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 12 Oct 2009 23:48:01 -0700
Subject: [Freeswitch-users] Groups information in sqllite
In-Reply-To: <f8af5740910122332i6c266250q7a550fb05439ae53@mail.gmail.com>
References: <f8af5740910122301u338ca128macfc0b769ea0e961@mail.gmail.com>
	<FD1AC150-0636-4861-8341-58553D3F43A0@freeswitch.org>
	<f8af5740910122316g7db0ae7ci1f79e3aede07d112@mail.gmail.com>
	<25EFC4E3-88E6-4286-8432-8118CC9E899C@freeswitch.org>
	<f8af5740910122332i6c266250q7a550fb05439ae53@mail.gmail.com>
Message-ID: <5FBD4715-DB4A-481A-AAAF-09D7541EC3F4@freeswitch.org>

Its based on the directory and who is in the group... check out the  
defaults it does exactly this on the 2000 range if I recall correctly.

/b

On Oct 12, 2009, at 11:32 PM, srinivasula reddy wrote:

> OK, then when i call to group number(911) how it will call to all  
> the registered members in group?




From nandy1925 at gmail.com  Tue Oct 13 00:00:43 2009
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Tue, 13 Oct 2009 15:00:43 +0800
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Update: Valet
	Parking
In-Reply-To: <4AD389A8.1080603@gmail.com>
References: <87f2f3b90910081842h6df329c2q74c6853278d3a102@mail.gmail.com>
	<15b9404e0910102018oda8834ah8cc87787ee00c22b@mail.gmail.com>
	<87f2f3b90910121201o422c7891vd30ee3761aef0eb7@mail.gmail.com>
	<4AD389A8.1080603@gmail.com>
Message-ID: <7d0bfd8c0910130000i25012d75ldc7f873214d30e12@mail.gmail.com>

i come across the valet_park application when i just finished an
improved-version of the call parking (using mod_fifo)  such that it is
parked to different ext'n numbers when the caller is att_xfer'd to ext 777.
i used  strftime(%s) to generate 700~759 parking numbers. i also added
feature that if the parked call is not picked up after xx secs timeout, it's
returned back to the transferer. the only glitch here is - when another
parking is done exactly 1 minute later (unless we'll limit the timeout to 59
secs). i hv to use att_xfer so that the transferer can hear the parking
number.

i'll submit the dialplan if it's worth seeing it.

On Tue, Oct 13, 2009 at 3:55 AM, William King <quentusrex at gmail.com> wrote:

> I don't know if this was mentioned yet. It would be useful to have a way
> to have the parking lot automatically find the next available spot and
> tts it to the person parking the call.
>
> Then the auto unpark would pop off the lowest numbered lot, or return
> fail if there is nobody in the parking lots etc.
>
> -William King
>
> Michael Collins wrote:
> >
> >
> > On Sat, Oct 10, 2009 at 8:18 PM, mayamatakeshi
> > <mayamatakeshi at gmail.com <mailto:mayamatakeshi at gmail.com>> wrote:
> >
> >
> >
> >     On Fri, Oct 9, 2009 at 10:42 AM, Michael Collins
> >     <msc at freeswitch.org <mailto:msc at freeswitch.org>> wrote:
> >
> >         FYI,
> >
> >         The FreeSWITCH devs have added valet parking! Check it out:
> >         http://www.freeswitch.org/node/207
> >
> >         Let us know what you think.
> >
> >
> >     Very nice.
> >
> >     But I think a valet_unpark app is missing.
> >     If the intention of the person sent to the valet lot is to
> >     retrieve a call there, the person can assume the call was already
> >     retrieved by someone else or that the caller hung up if he/she
> >     hears MOH. But it would be nicer to have a valet_unpark app to
> >     fail and let the dialplan play a message.
> >
> > I understand what you are saying. I'm not sure I agree, but we'll kick
> > the idea around when we have a few minutes and let you know what we
> > decide.
> > -MC
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dujinfang at gmail.com  Tue Oct 13 00:05:24 2009
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 13 Oct 2009 15:05:24 +0800
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>
Message-ID: <23f91030910130005m1dc40f90k8f5b047baf1df144@mail.gmail.com>

that will make life easier.

2009/10/13 Brian West <brian at freeswitch.org>

> Does anyone see a problem with hosting mod_h323 in our SVN?  I would
> like to centralize everything we can to reuse our issue tracking
> resources and not fragment the community if possible.
>
> /b
>
> On Oct 12, 2009, at 2:43 PM, Tihomir Culjaga wrote:
>
> > hi,
> >
> > finally i compiled it right ... had a stupid issue with ekiga and
> > wrong ptlib in place...
> >
> > anyhow, i loaded the module and will continue the tests
> > tomorrow ...first thing i arrive in my office :P
> >
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tculjaga at gmail.com  Tue Oct 13 00:17:54 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 13 Oct 2009 09:17:54 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <5F65174C-2CE6-4007-B288-C81227776281@freeswitch.org>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>
	<65d96fc80910122327u54505b44r4ff53e89293bcf9a@mail.gmail.com>
	<5F65174C-2CE6-4007-B288-C81227776281@freeswitch.org>
Message-ID: <65d96fc80910130017m5cafef68m89efa99493b0b16d@mail.gmail.com>

On Tue, Oct 13, 2009 at 8:31 AM, Brian West <brian at freeswitch.org> wrote:

> I wouldn't call it donating per se... Its just giving it a place to
> live with easy access for end users without having to do anything
> extra go get it!  ;)
>
> /b
>
>
I agree with you Brian.


>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tayeb.meftah at gmail.com  Tue Oct 13 01:33:49 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Tue, 13 Oct 2009 08:33:49 +0000
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>	<881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>
Message-ID: <4AD43B6D.7000204@gmail.com>

hello,
yes, host please to let users test it and report bug
YATE/Asterisk fully support but freeswitch no fully support it
Brian West a ?crit :
> Does anyone see a problem with hosting mod_h323 in our SVN?  I would  
> like to centralize everything we can to reuse our issue tracking  
> resources and not fragment the community if possible.
>
> /b
>
> On Oct 12, 2009, at 2:43 PM, Tihomir Culjaga wrote:
>
>   
>> hi,
>>
>> finally i compiled it right ... had a stupid issue with ekiga and  
>> wrong ptlib in place...
>>
>> anyhow, i loaded the module and will continue the tests  
>> tomorrow ...first thing i arrive in my office :P
>>
>>
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4501 (20091012) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
>   




__________ Information from ESET NOD32 Antivirus, version of virus signature database 4501 (20091012) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From srinivas.ksvreddy at gmail.com  Tue Oct 13 01:30:24 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 13 Oct 2009 14:00:24 +0530
Subject: [Freeswitch-users] Groups information in sqllite
In-Reply-To: <5FBD4715-DB4A-481A-AAAF-09D7541EC3F4@freeswitch.org>
References: <f8af5740910122301u338ca128macfc0b769ea0e961@mail.gmail.com>
	<FD1AC150-0636-4861-8341-58553D3F43A0@freeswitch.org>
	<f8af5740910122316g7db0ae7ci1f79e3aede07d112@mail.gmail.com>
	<25EFC4E3-88E6-4286-8432-8118CC9E899C@freeswitch.org>
	<f8af5740910122332i6c266250q7a550fb05439ae53@mail.gmail.com>
	<5FBD4715-DB4A-481A-AAAF-09D7541EC3F4@freeswitch.org>
Message-ID: <f8af5740910130130u6f16745h965cefdfb32c6441@mail.gmail.com>

hi brain,

thank u very much for your valuable time,
can u please tell me where the groups data will maintain thought the
session,

Thanks
Srinivas

On Tue, Oct 13, 2009 at 12:18 PM, Brian West <brian at freeswitch.org> wrote:

> Its based on the directory and who is in the group... check out the
> defaults it does exactly this on the 2000 range if I recall correctly.
>
> /b
>
> On Oct 12, 2009, at 11:32 PM, srinivasula reddy wrote:
>
> > OK, then when i call to group number(911) how it will call to all
> > the registered members in group?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Srinivasula Reddy K
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From srinivas.ksvreddy at gmail.com  Tue Oct 13 01:56:19 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 13 Oct 2009 14:26:19 +0530
Subject: [Freeswitch-users] 606 error
Message-ID: <f8af5740910130156r1145eabcl3c59f83779cb864f@mail.gmail.com>

Hi,

two users are registered in freeswitch, when i making call to another user i
am getting 606 error,
any help

-- 
Srinivasula Reddy K
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From tculjaga at gmail.com  Tue Oct 13 02:05:18 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 13 Oct 2009 11:05:18 +0200
Subject: [Freeswitch-users] 606 error
In-Reply-To: <f8af5740910130156r1145eabcl3c59f83779cb864f@mail.gmail.com>
References: <f8af5740910130156r1145eabcl3c59f83779cb864f@mail.gmail.com>
Message-ID: <65d96fc80910130205t5ce571ecu7746e4886b9362a3@mail.gmail.com>

what about some console logs & sip traces ?

T.

On Tue, Oct 13, 2009 at 10:56 AM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

> Hi,
>
> two users are registered in freeswitch, when i making call to another user
> i am getting 606 error,
> any help
>
> --
> Srinivasula Reddy K
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From srinivas.ksvreddy at gmail.com  Tue Oct 13 02:13:06 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 13 Oct 2009 14:43:06 +0530
Subject: [Freeswitch-users] 606 error
In-Reply-To: <65d96fc80910130205t5ce571ecu7746e4886b9362a3@mail.gmail.com>
References: <f8af5740910130156r1145eabcl3c59f83779cb864f@mail.gmail.com>
	<65d96fc80910130205t5ce571ecu7746e4886b9362a3@mail.gmail.com>
Message-ID: <f8af5740910130213p73d15225qbddce09854440b84@mail.gmail.com>

Hi,



Console user1181 attempted to call console user1171 resulted in failure. Sip
server returned "Temporarily unavailable" with reason header cause=606;

text="user-not-registered".  This also happened with other consoles.

Thanks
SRINIVAS


On Tue, Oct 13, 2009 at 2:35 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> what about some console logs & sip traces ?
>
> T.
>
> On Tue, Oct 13, 2009 at 10:56 AM, srinivasula reddy <
> srinivas.ksvreddy at gmail.com> wrote:
>
>> Hi,
>>
>> two users are registered in freeswitch, when i making call to another user
>> i am getting 606 error,
>> any help
>>
>> --
>> Srinivasula Reddy K
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Srinivasula Reddy K
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From vhatz at kinetix.gr  Tue Oct 13 02:40:58 2009
From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI))
Date: Tue, 13 Oct 2009 12:40:58 +0300
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <4AD43B6D.7000204@gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>	<Pine.LNX.4.64.0910081951170.23646@srv.icf.org.ru>	<881082ae0910081025o47c6057er675a10ad12019885@mail.gmail.com>	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>	<FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>
	<4AD43B6D.7000204@gmail.com>
Message-ID: <4AD44B2A.1040109@kinetix.gr>

Meftah Tayeb wrote:
> hello,
> yes, host please to let users test it and report bug
> YATE/Asterisk fully support but freeswitch no fully support it

I would disagree about YATE & Asterisk fully supporting H323.

:)

They both have some support for H323 for years now, but only for voice 
calls. None of the two platforms can do fax or video calls in H323 for 
example.

YATE especially, cannot even pass DTMF between H323 and SIP unless it is 
in RTP (RFC 2833).

I think it is a good opportunity for FS to make a difference if proper 
H323 support is built for it.


Best regards,
Vlasis Hatzistavrou.



From mcampbellsmith at gmail.com  Tue Oct 13 02:56:16 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 13 Oct 2009 20:56:16 +1100
Subject: [Freeswitch-users] dingaling: Destination out of order
Message-ID: <33c87fa30910130256j285c5f52l7a50d8ce19e11f80@mail.gmail.com>

Hi!

I am trying to call from FS to gtalk.  This used to work, so not sure
if there is a problem with my build (FreeSWITCH Version 1.0.trunk
(15126))

freeswitch at internal> dingaling status
--DingaLing status--
login   |       connected
mygmailid at gmail.com/gtalk    |       AUTHORIZED

It looks okay and I also see FS registered and online in the GTALK client.

When I dial 9999 (which is to call my gtalk user), I get the following
in the console:

2009-10-13 20:49:13.458712 [INFO] mod_dialplan_xml.c:391 Processing
1000->9999 in context default
2009-10-13 20:49:13.490719 [NOTICE] mod_dingaling.c:712 Close Channel
N/A [CS_NEW]
2009-10-13 20:49:13.498706 [ERR] switch_ivr_originate.c:1667 Cannot
create outgoing channel of type [dingaling] cause:
[DESTINATION_OUT_OF_ORDER

In dingaling.conf.xml, I only have the PCMU codec specified, and the
ATA is requesting PCMU/8000.

Any ideas why I am seeing this?

Thanks!



From fdelawarde at wirelessmundi.com  Tue Oct 13 03:09:45 2009
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Tue, 13 Oct 2009 12:09:45 +0200
Subject: [Freeswitch-users] sofia gateways and linux multipath routing
Message-ID: <1255428585.11786.90.camel@francois.tc.commsmundi.com>

Hello all,

I'm interested in using mod_sofia with multiple Internet connections
(configured as a unique load-balancing route using multipath).

One solution would be to define a different profile for each connection,
but it would be more practical having a unique external profile that
would automatically handle everything (detecting multiple public IPs and
selecting the right one for a call, being able to select the router for
gateway registration...).

Routing table:
192.168.10.0/24 dev eth0  proto kernel  scope link  src 192.168.10.1
192.168.1.0/24 dev eth1  proto kernel  scope link  src 192.168.1.2
192.168.2.0/24 dev eth2  proto kernel  scope link  src 192.168.2.2
default  proto static
	nexthop via 192.168.1.1  dev eth1 weight 1
	nexthop via 192.168.2.1  dev eth2 weight 1

Both default routers (192.168.1.1 and 192.168.2.1) would have a distinct
public IP.

Several questions cross my mind:
- Can a unique sofia profile be bound to multiple IPs (not 0.0.0.0)?

- How would FS behave with a unique external profile in that situation?
   * Would FS reply to an incoming call using the same router it came
from forcing packet source address?
   * Would FS stick to a unique router for all flows of an outgoing call
(SIP, RTP, UDPTL)?
   * Can I force a gateway to use a given router (for calls,
registration, ...)?
   * Would the NAT system (using stun or auto-nat) work in that
situation, or does it assume only one default router (and a unique
public IP) exists per profile?

- Knowing the above, would it be necessary to use a different profile
for each router/interface, and define the same gateway in each of these?

- Tricky question: What if multiple routers are on the same
network/interface (192.168.1.1, 192.168.1.2, ...)?


Thanks in advance,
Fran?ois.




From b_ball_henry at hotmail.com  Tue Oct 13 03:14:22 2009
From: b_ball_henry at hotmail.com (Henry Huang)
Date: Tue, 13 Oct 2009 18:14:22 +0800
Subject: [Freeswitch-users] sched_api doesn't get launched
Message-ID: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>

Hi:
I am using mod_java. And in my script I was able to achieve using:

execute("sched_hangup", "+300 alloted_timeout");

However, when I try to run sched_api in the same way, system log returns
that it's an invalid application. I have also tried to trigger it with many
conditional channel variable api calls , but non of them seemed to execute
the api command (because I turned on the highest level of debugging and see
no where the sched_api is being called.

The closest thing I got was by using "api_after_bridge" like the following,
but it only launches when the bridge is teared down(which is not what I
want). I originally thought after bridge means right after the 2 party is
connected. All I want is to be able to play some message to leg A at certain
time.

setVariable("api_after_bridge", "sched_api +10 none uuid_displace ${uuid}
start /path/to/some.wav 20 mux");

I have been struggling with different combination for a week now.. Please
shed some light if you know something.

Thanks,

-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From jason at jasonjgw.net  Tue Oct 13 03:16:02 2009
From: jason at jasonjgw.net (Jason White)
Date: Tue, 13 Oct 2009 21:16:02 +1100
Subject: [Freeswitch-users] dingaling: Destination out of order
In-Reply-To: <33c87fa30910130256j285c5f52l7a50d8ce19e11f80@mail.gmail.com>
References: <33c87fa30910130256j285c5f52l7a50d8ce19e11f80@mail.gmail.com>
Message-ID: <20091013101602.GA19589@jdc.jasonjgw.net>

Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
> When I dial 9999 (which is to call my gtalk user), I get the following
> in the console:

[snip]

Could you turn on debug logging in the console and post the output?




From shaheryarkh at googlemail.com  Tue Oct 13 03:15:45 2009
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Tue, 13 Oct 2009 16:15:45 +0600
Subject: [Freeswitch-users] Registering a large number of SIP users
Message-ID: <b16156850910130315k4f4fdfebw96955d1b5d8e79fa@mail.gmail.com>

Hi,

I am creating a load test setup for FreeSWITCH using Sofia SIP. I have two
machines both configured with mod_xml_curl, one machine (lets call it SIP
Server) has 100 SIP accounts. Now i want to register second machine (lets
call it SIP Client) to all these 100 SIP accounts on first machine. How can
i do that?

One approach that i can think of is that to create a new profile (or use
existing external profile) on SIP Client and add all accounts to it as
gateways having,

*<param name="register" value="true">*

define in their configuration.

Is my approach correct? Are there any better ways to do this?

Thank you.


-- 
________________________________________________________
|
                     |
| FATAL ERROR                                                           ---
O X |
|_______________________________________________________|
|                        You have moved the mouse.
 |
| Windows must be restarted for the changes to take effect.   |
|                                        <OK>
               |
####################################/


Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From Prometheus001 at gmx.net  Tue Oct 13 03:17:04 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Tue, 13 Oct 2009 12:17:04 +0200
Subject: [Freeswitch-users] Mod_fifo posision in queue
In-Reply-To: <87f2f3b90909101322v68e49f74jbc4cf2052e1811c3@mail.gmail.com>
References: <8ccbff060909010737v4a8f17ep4d6175c2a2a2f866@mail.gmail.com>	<E09986E3-1177-4260-AFFA-9018F92D81A2@jerris.com>	<8ccbff060909092332q1d06e74m8d5ca3a39ca4f572@mail.gmail.com>	<87f2f3b90909100037m45b93a9ei7ec62b4d1f84e986@mail.gmail.com>	<191c3a030909101123k6ccbf949q74fa22197f44517e@mail.gmail.com>	<8ccbff060909101203m5625a991y32138e32913959c5@mail.gmail.com>	<86a32abc0909101232k656be844qfa19bb7a679a2e02@mail.gmail.com>
	<87f2f3b90909101322v68e49f74jbc4cf2052e1811c3@mail.gmail.com>
Message-ID: <4AD453A0.4070207@gmx.net>

Has anybody managed to get this to work already?
How do you play the announcements dependent on the variable in the dialplan?

Best regards
Peter


Michael Collins schrieb:
>
>
> On Thu, Sep 10, 2009 at 12:32 PM, Diego Viola <diego.viola at gmail.com
> <mailto:diego.viola at gmail.com>> wrote:
>
>     Lets make sure we add it on the wiki too =D.
>
> Yep, as soon as we verify its functionality we'll wikify it. :)
> -MC
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From tculjaga at gmail.com  Tue Oct 13 03:32:34 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 13 Oct 2009 12:32:34 +0200
Subject: [Freeswitch-users] 606 error
In-Reply-To: <f8af5740910130213p73d15225qbddce09854440b84@mail.gmail.com>
References: <f8af5740910130156r1145eabcl3c59f83779cb864f@mail.gmail.com>
	<65d96fc80910130205t5ce571ecu7746e4886b9362a3@mail.gmail.com>
	<f8af5740910130213p73d15225qbddce09854440b84@mail.gmail.com>
Message-ID: <65d96fc80910130332hc94075bq182887b5f6760f49@mail.gmail.com>

and you are sure both users are registered to the same context and your
dialplan is correct ?

T.

On Tue, Oct 13, 2009 at 11:13 AM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

>
> Hi,
>
>
>
> Console user1181 attempted to call console user1171 resulted in failure.
> Sip server returned "Temporarily unavailable" with reason header cause=606;
>
> text="user-not-registered".  This also happened with other consoles.
>
> Thanks
> SRINIVAS
>
>
>
> On Tue, Oct 13, 2009 at 2:35 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> what about some console logs & sip traces ?
>>
>> T.
>>
>> On Tue, Oct 13, 2009 at 10:56 AM, srinivasula reddy <
>> srinivas.ksvreddy at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> two users are registered in freeswitch, when i making call to another
>>> user i am getting 606 error,
>>> any help
>>>
>>> --
>>> Srinivasula Reddy K
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Srinivasula Reddy K
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mcampbellsmith at gmail.com  Tue Oct 13 03:37:07 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 13 Oct 2009 21:37:07 +1100
Subject: [Freeswitch-users] dingaling: Destination out of order
In-Reply-To: <20091013101602.GA19589@jdc.jasonjgw.net>
References: <33c87fa30910130256j285c5f52l7a50d8ce19e11f80@mail.gmail.com>
	<20091013101602.GA19589@jdc.jasonjgw.net>
Message-ID: <33c87fa30910130337y631ffc41l8064bfc0d9125d1e@mail.gmail.com>

This is all I see:

console loglevel 9
+OK console log level set to DEBUG

freeswitch at internal> 2009-10-13 21:33:05.578863 [NOTICE]
switch_channel.c:613 New Channel sofia/internal_nat/1000 at 192.168.1.120
[cad049fe-b7e3-11de-94a7-1dd4d003eac8]
2009-10-13 21:33:05.634924 [INFO] mod_dialplan_xml.c:391 Processing
10000->9999 in context default
2009-10-13 21:33:05.666835 [NOTICE] mod_dingaling.c:712 Close Channel
N/A [CS_NEW]
2009-10-13 21:33:05.674929 [ERR] switch_ivr_originate.c:1667 Cannot
create outgoing channel of type [dingaling] cause:
[DESTINATION_OUT_OF_ORDER]
2009-10-13 21:33:05.674929 [INFO] mod_dptools.c:2133 Originate Failed.
 Cause: DESTINATION_OUT_OF_ORDER
2009-10-13 21:33:05.674929 [NOTICE] mod_dptools.c:2166 Hangup
sofia/internal_nat/1000 at 192.168.1.120 [CS_EXECUTE]
[DESTINATION_OUT_OF_ORDER]
2009-10-13 21:33:05.802716 [NOTICE] switch_core_session.c:1087 Session
16 (sofia/internal_nat/1000 at 192.168.1.120) Ended
2009-10-13 21:33:05.807529 [NOTICE] switch_core_session.c:1089 Close
Channel sofia/internal_nat/1000 at 192.168.1.120 [CS_DESTROY]


On Tue, Oct 13, 2009 at 9:16 PM, Jason White <jason at jasonjgw.net> wrote:
> Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
>> When I dial 9999 (which is to call my gtalk user), I get the following
>> in the console:
>
> [snip]
>
> Could you turn on debug logging in the console and post the output?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From bottleman at icf.org.ru  Tue Oct 13 03:44:59 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Tue, 13 Oct 2009 14:44:59 +0400 (MSD)
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <23f91030910130005m1dc40f90k8f5b047baf1df144@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>
	<23f91030910130005m1dc40f90k8f5b047baf1df144@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910131435550.8815@srv.icf.org.ru>

On 2009-10-13 15:05 +0800, Seven Du wrote freeswitch-users at lists.freeswitch.org:

hm, host it if you wont, i has nothing against it.

SD>that will make life easier.
SD>
SD>2009/10/13 Brian West <brian at freeswitch.org>
SD>
SD>> Does anyone see a problem with hosting mod_h323 in our SVN?  I would
SD>> like to centralize everything we can to reuse our issue tracking
SD>> resources and not fragment the community if possible.
SD>>
SD>> /b
SD>>
SD>> On Oct 12, 2009, at 2:43 PM, Tihomir Culjaga wrote:
SD>>
SD>> > hi,
SD>> >
SD>> > finally i compiled it right ... had a stupid issue with ekiga and
SD>> > wrong ptlib in place...
SD>> >
SD>> > anyhow, i loaded the module and will continue the tests
SD>> > tomorrow ...first thing i arrive in my office :P
SD>> >
SD>> >
SD>>
SD>>
SD>> _______________________________________________
SD>> FreeSWITCH-users mailing list
SD>> FreeSWITCH-users at lists.freeswitch.org
SD>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
SD>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
SD>> http://www.freeswitch.org
SD>>
SD>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From mcampbellsmith at gmail.com  Tue Oct 13 03:52:36 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 13 Oct 2009 21:52:36 +1100
Subject: [Freeswitch-users] dingaling: Destination out of order
In-Reply-To: <33c87fa30910130256j285c5f52l7a50d8ce19e11f80@mail.gmail.com>
References: <33c87fa30910130256j285c5f52l7a50d8ce19e11f80@mail.gmail.com>
Message-ID: <33c87fa30910130352i1f3991d9x5ab1bf9eedd5b6df@mail.gmail.com>

I've fixed the problem.

My dialplan for outbound calling had a typo:

    <action application="bridge"
data="dingaling/gtalk/mygmailid at gmail.com"/>

The gtalk was gtallk somehow .....

On Tue, Oct 13, 2009 at 8:56 PM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> Hi!
>
> I am trying to call from FS to gtalk. ?This used to work, so not sure
> if there is a problem with my build (FreeSWITCH Version 1.0.trunk
> (15126))
>
> freeswitch at internal> dingaling status
> --DingaLing status--
> login ? | ? ? ? connected
> mygmailid at gmail.com/gtalk ? ?| ? ? ? AUTHORIZED
>
> It looks okay and I also see FS registered and online in the GTALK client.
>
> When I dial 9999 (which is to call my gtalk user), I get the following
> in the console:
>
> 2009-10-13 20:49:13.458712 [INFO] mod_dialplan_xml.c:391 Processing
> 1000->9999 in context default
> 2009-10-13 20:49:13.490719 [NOTICE] mod_dingaling.c:712 Close Channel
> N/A [CS_NEW]
> 2009-10-13 20:49:13.498706 [ERR] switch_ivr_originate.c:1667 Cannot
> create outgoing channel of type [dingaling] cause:
> [DESTINATION_OUT_OF_ORDER
>
> In dingaling.conf.xml, I only have the PCMU codec specified, and the
> ATA is requesting PCMU/8000.
>
> Any ideas why I am seeing this?
>
> Thanks!
>



From srinivas.ksvreddy at gmail.com  Tue Oct 13 04:03:28 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 13 Oct 2009 16:33:28 +0530
Subject: [Freeswitch-users] 606 error
In-Reply-To: <65d96fc80910130332hc94075bq182887b5f6760f49@mail.gmail.com>
References: <f8af5740910130156r1145eabcl3c59f83779cb864f@mail.gmail.com>
	<65d96fc80910130205t5ce571ecu7746e4886b9362a3@mail.gmail.com>
	<f8af5740910130213p73d15225qbddce09854440b84@mail.gmail.com>
	<65d96fc80910130332hc94075bq182887b5f6760f49@mail.gmail.com>
Message-ID: <f8af5740910130403v714651a8x31eff11b53097ee9@mail.gmail.com>

Hi,

thank u very much for your valuable time,
s am sure they are both in same it is not occur continuously, i dont know
the reason,
i am having the wireshark file, any help?

thanks
srinivas

On Tue, Oct 13, 2009 at 4:02 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> and you are sure both users are registered to the same context and your
> dialplan is correct ?
>
> T.
>
>
> On Tue, Oct 13, 2009 at 11:13 AM, srinivasula reddy <
> srinivas.ksvreddy at gmail.com> wrote:
>
>>
>> Hi,
>>
>>
>>
>> Console user1181 attempted to call console user1171 resulted in failure.
>> Sip server returned "Temporarily unavailable" with reason header cause=606;
>>
>> text="user-not-registered".  This also happened with other consoles.
>>
>> Thanks
>> SRINIVAS
>>
>>
>>
>> On Tue, Oct 13, 2009 at 2:35 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>
>>> what about some console logs & sip traces ?
>>>
>>> T.
>>>
>>> On Tue, Oct 13, 2009 at 10:56 AM, srinivasula reddy <
>>> srinivas.ksvreddy at gmail.com> wrote:
>>>
>>>> Hi,
>>>>
>>>> two users are registered in freeswitch, when i making call to another
>>>> user i am getting 606 error,
>>>> any help
>>>>
>>>> --
>>>> Srinivasula Reddy K
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Srinivasula Reddy K
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Srinivasula Reddy K
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From srinivas.ksvreddy at gmail.com  Tue Oct 13 04:23:36 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 13 Oct 2009 16:53:36 +0530
Subject: [Freeswitch-users] 606 error
In-Reply-To: <f8af5740910130403v714651a8x31eff11b53097ee9@mail.gmail.com>
References: <f8af5740910130156r1145eabcl3c59f83779cb864f@mail.gmail.com>
	<65d96fc80910130205t5ce571ecu7746e4886b9362a3@mail.gmail.com>
	<f8af5740910130213p73d15225qbddce09854440b84@mail.gmail.com>
	<65d96fc80910130332hc94075bq182887b5f6760f49@mail.gmail.com>
	<f8af5740910130403v714651a8x31eff11b53097ee9@mail.gmail.com>
Message-ID: <f8af5740910130423o6ac1ae61g7babc49a9c4eddfa@mail.gmail.com>

wireshark image

thanks
srinivas

On Tue, Oct 13, 2009 at 4:33 PM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

> Hi,
>
> thank u very much for your valuable time,
> s am sure they are both in same it is not occur continuously, i dont know
> the reason,
> i am having the wireshark file, any help?
>
> thanks
> srinivas
>
>
> On Tue, Oct 13, 2009 at 4:02 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> and you are sure both users are registered to the same context and your
>> dialplan is correct ?
>>
>> T.
>>
>>
>> On Tue, Oct 13, 2009 at 11:13 AM, srinivasula reddy <
>> srinivas.ksvreddy at gmail.com> wrote:
>>
>>>
>>> Hi,
>>>
>>>
>>>
>>> Console user1181 attempted to call console user1171 resulted in failure.
>>> Sip server returned "Temporarily unavailable" with reason header cause=606;
>>>
>>> text="user-not-registered".  This also happened with other consoles.
>>>
>>> Thanks
>>> SRINIVAS
>>>
>>>
>>>
>>> On Tue, Oct 13, 2009 at 2:35 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>>
>>>> what about some console logs & sip traces ?
>>>>
>>>> T.
>>>>
>>>> On Tue, Oct 13, 2009 at 10:56 AM, srinivasula reddy <
>>>> srinivas.ksvreddy at gmail.com> wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> two users are registered in freeswitch, when i making call to another
>>>>> user i am getting 606 error,
>>>>> any help
>>>>>
>>>>> --
>>>>> Srinivasula Reddy K
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Srinivasula Reddy K
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Srinivasula Reddy K
>



-- 
Srinivasula Reddy K
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From tculjaga at gmail.com  Tue Oct 13 04:35:45 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 13 Oct 2009 13:35:45 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910131435550.8815@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>
	<23f91030910130005m1dc40f90k8f5b047baf1df144@mail.gmail.com>
	<Pine.LNX.4.64.0910131435550.8815@srv.icf.org.ru>
Message-ID: <65d96fc80910130435x67405db8s4d0927ad68be05d6@mail.gmail.com>

static const char* h323_formats[] = {
    "G.711-*A*Law-64k", "PCM*U*",
    "G.711-*u*Law-64k", "PCM*A*",
    "GSM-06.10", "gsm",
    "MS-GSM", "msgsm",



I've changed this to meed desired caps ... need more tests ...


2009/10/13 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-13 15:05 +0800, Seven Du wrote
> freeswitch-users at lists.freeswitch.org:
>
> hm, host it if you wont, i has nothing against it.
>
> SD>that will make life easier.
> SD>
> SD>2009/10/13 Brian West <brian at freeswitch.org>
> SD>
> SD>> Does anyone see a problem with hosting mod_h323 in our SVN?  I would
> SD>> like to centralize everything we can to reuse our issue tracking
> SD>> resources and not fragment the community if possible.
> SD>>
> SD>> /b
> SD>>
> SD>> On Oct 12, 2009, at 2:43 PM, Tihomir Culjaga wrote:
> SD>>
> SD>> > hi,
> SD>> >
> SD>> > finally i compiled it right ... had a stupid issue with ekiga and
> SD>> > wrong ptlib in place...
> SD>> >
> SD>> > anyhow, i loaded the module and will continue the tests
> SD>> > tomorrow ...first thing i arrive in my office :P
> SD>> >
> SD>> >
> SD>>
> SD>>
> SD>> _______________________________________________
> SD>> FreeSWITCH-users mailing list
> SD>> FreeSWITCH-users at lists.freeswitch.org
> SD>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> SD>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> SD>> http://www.freeswitch.org
> SD>>
> SD>
>
> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129-RIPE
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bottleman at icf.org.ru  Tue Oct 13 04:50:13 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Tue, 13 Oct 2009 15:50:13 +0400 (MSD)
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910130435x67405db8s4d0927ad68be05d6@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>
	<23f91030910130005m1dc40f90k8f5b047baf1df144@mail.gmail.com>
	<Pine.LNX.4.64.0910131435550.8815@srv.icf.org.ru>
	<65d96fc80910130435x67405db8s4d0927ad68be05d6@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910131542350.28673@srv.icf.org.ru>

On 2009-10-13 13:35 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

this morning me bring in hospital, and now i cannot make much work, 
i think return to the ranks in 1-2 week.

TC>static const char* h323_formats[] = {
TC>    "G.711-*A*Law-64k", "PCM*U*",
TC>    "G.711-*u*Law-64k", "PCM*A*",
TC>    "GSM-06.10", "gsm",
TC>    "MS-GSM", "msgsm",
TC>
TC>
TC>
TC>I've changed this to meed desired caps ... need more tests ...
TC>
TC>
TC>2009/10/13 Georgiewskiy Yuriy <bottleman at icf.org.ru>
TC>
TC>> On 2009-10-13 15:05 +0800, Seven Du wrote
TC>> freeswitch-users at lists.freeswitch.org:
TC>>
TC>> hm, host it if you wont, i has nothing against it.
TC>>
TC>> SD>that will make life easier.
TC>> SD>
TC>> SD>2009/10/13 Brian West <brian at freeswitch.org>
TC>> SD>
TC>> SD>> Does anyone see a problem with hosting mod_h323 in our SVN?  I would
TC>> SD>> like to centralize everything we can to reuse our issue tracking
TC>> SD>> resources and not fragment the community if possible.
TC>> SD>>
TC>> SD>> /b
TC>> SD>>
TC>> SD>> On Oct 12, 2009, at 2:43 PM, Tihomir Culjaga wrote:
TC>> SD>>
TC>> SD>> > hi,
TC>> SD>> >
TC>> SD>> > finally i compiled it right ... had a stupid issue with ekiga and
TC>> SD>> > wrong ptlib in place...
TC>> SD>> >
TC>> SD>> > anyhow, i loaded the module and will continue the tests
TC>> SD>> > tomorrow ...first thing i arrive in my office :P
TC>> SD>> >
TC>> SD>> >
TC>> SD>>
TC>> SD>>
TC>> SD>> _______________________________________________
TC>> SD>> FreeSWITCH-users mailing list
TC>> SD>> FreeSWITCH-users at lists.freeswitch.org
TC>> SD>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> SD>> UNSUBSCRIBE:
TC>> http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> SD>> http://www.freeswitch.org
TC>> SD>>
TC>> SD>
TC>>
TC>> C ?????????                       With Best Regards
TC>> ???????????? ????.                Georgiewskiy Yuriy
TC>> +7 4872 711666                    +7 4872 711666
TC>> ???? +7 4872 711143               fax +7 4872 711143
TC>> ???????? ??? "?? ?? ??????"       IT Service Ltd
TC>> http://nkoort.ru                  http://nkoort.ru
TC>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
TC>> YG129-RIPE                        YG129-RIPE
TC>>
TC>> _______________________________________________
TC>> FreeSWITCH-users mailing list
TC>> FreeSWITCH-users at lists.freeswitch.org
TC>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> http://www.freeswitch.org
TC>>
TC>>
TC>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From tculjaga at gmail.com  Tue Oct 13 04:54:34 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 13 Oct 2009 13:54:34 +0200
Subject: [Freeswitch-users] 606 error
In-Reply-To: <f8af5740910130403v714651a8x31eff11b53097ee9@mail.gmail.com>
References: <f8af5740910130156r1145eabcl3c59f83779cb864f@mail.gmail.com>
	<65d96fc80910130205t5ce571ecu7746e4886b9362a3@mail.gmail.com>
	<f8af5740910130213p73d15225qbddce09854440b84@mail.gmail.com>
	<65d96fc80910130332hc94075bq182887b5f6760f49@mail.gmail.com>
	<f8af5740910130403v714651a8x31eff11b53097ee9@mail.gmail.com>
Message-ID: <65d96fc80910130454n1860dc09l968454d32b4cd361@mail.gmail.com>

of course, if you can send it thi will be great...

T.

On Tue, Oct 13, 2009 at 1:03 PM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

> Hi,
>
> thank u very much for your valuable time,
> s am sure they are both in same it is not occur continuously, i dont know
> the reason,
> i am having the wireshark file, any help?
>
> thanks
> srinivas
>
>
> On Tue, Oct 13, 2009 at 4:02 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> and you are sure both users are registered to the same context and your
>> dialplan is correct ?
>>
>> T.
>>
>>
>> On Tue, Oct 13, 2009 at 11:13 AM, srinivasula reddy <
>> srinivas.ksvreddy at gmail.com> wrote:
>>
>>>
>>> Hi,
>>>
>>>
>>>
>>> Console user1181 attempted to call console user1171 resulted in failure.
>>> Sip server returned "Temporarily unavailable" with reason header cause=606;
>>>
>>> text="user-not-registered".  This also happened with other consoles.
>>>
>>> Thanks
>>> SRINIVAS
>>>
>>>
>>>
>>> On Tue, Oct 13, 2009 at 2:35 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>>
>>>> what about some console logs & sip traces ?
>>>>
>>>> T.
>>>>
>>>> On Tue, Oct 13, 2009 at 10:56 AM, srinivasula reddy <
>>>> srinivas.ksvreddy at gmail.com> wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> two users are registered in freeswitch, when i making call to another
>>>>> user i am getting 606 error,
>>>>> any help
>>>>>
>>>>> --
>>>>> Srinivasula Reddy K
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Srinivasula Reddy K
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Srinivasula Reddy K
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Tue Oct 13 05:02:56 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 13 Oct 2009 14:02:56 +0200
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910131542350.28673@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>
	<23f91030910130005m1dc40f90k8f5b047baf1df144@mail.gmail.com>
	<Pine.LNX.4.64.0910131435550.8815@srv.icf.org.ru>
	<65d96fc80910130435x67405db8s4d0927ad68be05d6@mail.gmail.com>
	<Pine.LNX.4.64.0910131542350.28673@srv.icf.org.ru>
Message-ID: <65d96fc80910130502k7a3fedf5y1c4d6fb5ad8f7bd7@mail.gmail.com>

2009/10/13 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-13 13:35 +0200, Tihomir Culjaga wrote
> freeswitch-users at lists.fre...:
>
> this morning me bring in hospital, and now i cannot make much work,
> i think return to the ranks in 1-2 week.
>
>
damn, hope you will recover soon... take it easy.

T.
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From vhatz at kinetix.gr  Tue Oct 13 05:13:48 2009
From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI))
Date: Tue, 13 Oct 2009 15:13:48 +0300
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <Pine.LNX.4.64.0910131542350.28673@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>	<FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>	<23f91030910130005m1dc40f90k8f5b047baf1df144@mail.gmail.com>	<Pine.LNX.4.64.0910131435550.8815@srv.icf.org.ru>	<65d96fc80910130435x67405db8s4d0927ad68be05d6@mail.gmail.com>
	<Pine.LNX.4.64.0910131542350.28673@srv.icf.org.ru>
Message-ID: <4AD46EFC.7060306@kinetix.gr>


Georgiewskiy Yuriy wrote:
> On 2009-10-13 13:35 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:
> 
> this morning me bring in hospital, and now i cannot make much work, 
> i think return to the ranks in 1-2 week.

I wish you a speedy recovery, Yuriy.

Regards,
Vlasis.



From ryannyl at gmail.com  Tue Oct 13 05:21:06 2009
From: ryannyl at gmail.com (Ryanny Lin)
Date: Tue, 13 Oct 2009 20:21:06 +0800
Subject: [Freeswitch-users] Registering a large number of SIP users
In-Reply-To: <b16156850910130315k4f4fdfebw96955d1b5d8e79fa@mail.gmail.com>
References: <b16156850910130315k4f4fdfebw96955d1b5d8e79fa@mail.gmail.com>
Message-ID: <4bfcac7e0910130521s44a042c0m25eb5c62032d4764@mail.gmail.com>

You may try this tool, "sipp", to execute a load test.
http://sipp.sourceforge.net/

2009/10/13 Muhammad Shahzad <shaheryarkh at googlemail.com>

> Hi,
>
> I am creating a load test setup for FreeSWITCH using Sofia SIP. I have two
> machines both configured with mod_xml_curl, one machine (lets call it SIP
> Server) has 100 SIP accounts. Now i want to register second machine (lets
> call it SIP Client) to all these 100 SIP accounts on first machine. How can
> i do that?
>
> One approach that i can think of is that to create a new profile (or use
> existing external profile) on SIP Client and add all accounts to it as
> gateways having,
>
> *<param name="register" value="true">*
>
> define in their configuration.
>
> Is my approach correct? Are there any better ways to do this?
>
> Thank you.
>
>
> --
> ________________________________________________________
> |
>                      |
> | FATAL ERROR                                                           ---
> O X |
> |_______________________________________________________|
> |                        You have moved the mouse.
>    |
> | Windows must be restarted for the changes to take effect.   |
> |                                        <OK>
>                |
> ####################################/
>
>
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sincerely regards,
Wen-Jen
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From shaheryarkh at googlemail.com  Tue Oct 13 05:31:00 2009
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Tue, 13 Oct 2009 18:31:00 +0600
Subject: [Freeswitch-users] Registering a large number of SIP users
In-Reply-To: <4bfcac7e0910130521s44a042c0m25eb5c62032d4764@mail.gmail.com>
References: <b16156850910130315k4f4fdfebw96955d1b5d8e79fa@mail.gmail.com>
	<4bfcac7e0910130521s44a042c0m25eb5c62032d4764@mail.gmail.com>
Message-ID: <b16156850910130531i27fa8695j3f83806b27d9ab45@mail.gmail.com>

Yes i have used this tool before, but sip registration is just first part of
my load test, i will be make SIP call load testing too. Also, i want to use
FreeSWITCH against FreeSWITCH to test its capability both as SIP Server and
SIP Client.

Thank you.


On Tue, Oct 13, 2009 at 6:21 PM, Ryanny Lin <ryannyl at gmail.com> wrote:

> You may try this tool, "sipp", to execute a load test.
> http://sipp.sourceforge.net/
>
> 2009/10/13 Muhammad Shahzad <shaheryarkh at googlemail.com>
>
>> Hi,
>>
>> I am creating a load test setup for FreeSWITCH using Sofia SIP. I have two
>> machines both configured with mod_xml_curl, one machine (lets call it SIP
>> Server) has 100 SIP accounts. Now i want to register second machine (lets
>> call it SIP Client) to all these 100 SIP accounts on first machine. How can
>> i do that?
>>
>> One approach that i can think of is that to create a new profile (or use
>> existing external profile) on SIP Client and add all accounts to it as
>> gateways having,
>>
>> *<param name="register" value="true">*
>>
>> define in their configuration.
>>
>> Is my approach correct? Are there any better ways to do this?
>>
>> Thank you.
>>
>>
>> --
>> ________________________________________________________
>> |
>>                        |
>> | FATAL ERROR
>> --- O X |
>> |_______________________________________________________|
>> |                        You have moved the mouse.
>>    |
>> | Windows must be restarted for the changes to take effect.   |
>> |                                        <OK>
>>                  |
>> ####################################/
>>
>>
>> Muhammad Shahzad
>> -----------------------------------
>> CISCO Rich Media Communication Specialist (CRMCS)
>> CISCO Certified Network Associate (CCNA)
>> Cell: +92 334 422 40 88
>> MSN: shari_786pk at hotmail.com
>> Email: shaheryarkh at googlemail.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Sincerely regards,
> Wen-Jen
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
________________________________________________________
|
                     |
| FATAL ERROR                                                           ---
O X |
|_______________________________________________________|
|                        You have moved the mouse.
 |
| Windows must be restarted for the changes to take effect.   |
|                                        <OK>
               |
####################################/


Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From lakindia89 at gmail.com  Tue Oct 13 06:45:04 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Tue, 13 Oct 2009 19:15:04 +0530
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <87f2f3b90910121149x6e2114f0q996a262ad272a1f6@mail.gmail.com>
References: <7d79b3930910090407m5dc81d12pf2b0a47f6017804@mail.gmail.com>
	<20091009123615.BC50A3FA6E9@mail.cune.org>
	<7d79b3930910100647u7fd11413ucec0a535d68b3e3a@mail.gmail.com>
	<87f2f3b90910101009j1295d812jd98cae91e30b1eff@mail.gmail.com>
	<7d79b3930910120454m1ffc6dd4u42b91a4349597166@mail.gmail.com>
	<87f2f3b90910121149x6e2114f0q996a262ad272a1f6@mail.gmail.com>
Message-ID: <7d79b3930910130645y658ae041v8219c4c881fa5698@mail.gmail.com>

We are using Reliance as the Carrier.
I think, with this same Reliance carrier, in my office, they are able to
make outgoing calls through asterisk+libpri.

On Tue, Oct 13, 2009 at 12:19 AM, Michael Collins <msc at freeswitch.org>wrote:

> Lak,
>
> Okay I will need a little bit of time to dig into the IE's and what they
> contain. In the meantime can you tell me who the carrier is? I'd like to
> find out if they have some specific requirements. The fact that it doesn't
> work with libpri surprises me because that would mean that Asterisk systems
> would probably not work with this carrier as well.
>
> BTW, thanks for the very complete pastebin entries. :)
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From odermann at googlemail.com  Tue Oct 13 06:51:11 2009
From: odermann at googlemail.com (Dennis)
Date: Tue, 13 Oct 2009 15:51:11 +0200
Subject: [Freeswitch-users] SIP Overlap support?
Message-ID: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>

hi there,

i would like to ask, if fs has support for something like "SIP Overlap"?

instead of receiving the phonenumber from our carrier in a block, we
want to receive the phonenumber digit-by-digit and we want to tell fs
when the number is complete. our carrier could send us the phonenumber
digit-by-digit, but what about the fs-side?


thanks and kind regards
dennis



From jerry.richards at teotech.com  Tue Oct 13 07:54:30 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 13 Oct 2009 07:54:30 -0700
Subject: [Freeswitch-users] FS Extension Groups Documentation
Message-ID: <E8FDE7D1E87C4DDC98BD76DAF7DC50A5@greyhawk.tonecommander.com>


Is there a link to the documentation of FS groups and what can be done with
a group (i.e. capabilities)?

Thanks And Best Regards,
Jerry




From anthony.minessale at gmail.com  Tue Oct 13 08:03:38 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 13 Oct 2009 10:03:38 -0500
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
Message-ID: <191c3a030910130803k321f39e3n265e64ab3c84a5e6@mail.gmail.com>

have you tried it?
I *think* either we did support it or we would with a small patch to sofia
lib that I cannot recall if we ever got committed.


On Tue, Oct 13, 2009 at 8:51 AM, Dennis <odermann at googlemail.com> wrote:

> hi there,
>
> i would like to ask, if fs has support for something like "SIP Overlap"?
>
> instead of receiving the phonenumber from our carrier in a block, we
> want to receive the phonenumber digit-by-digit and we want to tell fs
> when the number is complete. our carrier could send us the phonenumber
> digit-by-digit, but what about the fs-side?
>
>
> thanks and kind regards
> dennis
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From odermann at googlemail.com  Tue Oct 13 08:26:56 2009
From: odermann at googlemail.com (Dennis)
Date: Tue, 13 Oct 2009 17:26:56 +0200
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <191c3a030910130803k321f39e3n265e64ab3c84a5e6@mail.gmail.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
	<191c3a030910130803k321f39e3n265e64ab3c84a5e6@mail.gmail.com>
Message-ID: <5e414ed0910130826j11d45a66q86d785f900e4196e@mail.gmail.com>

how could we try? we played arround with a snom phone (snom seems to
support something in this direction, but are not shure, how we can
test it and how we can see if it is supported or not.

any hint?


2009/10/13 Anthony Minessale <anthony.minessale at gmail.com>:
> have you tried it?
> I *think* either we did support it or we would with a small patch to sofia
> lib that I cannot recall if we ever got committed.
>
>
> On Tue, Oct 13, 2009 at 8:51 AM, Dennis <odermann at googlemail.com> wrote:
>>
>> hi there,
>>
>> i would like to ask, if fs has support for something like "SIP Overlap"?
>>
>> instead of receiving the phonenumber from our carrier in a block, we
>> want to receive the phonenumber digit-by-digit and we want to tell fs
>> when the number is complete. our carrier could send us the phonenumber
>> digit-by-digit, but what about the fs-side?
>>
>>
>> thanks and kind regards
>> dennis
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From shiyanov at gmail.com  Tue Oct 13 08:32:12 2009
From: shiyanov at gmail.com (Artem Shiyanov)
Date: Tue, 13 Oct 2009 19:32:12 +0400
Subject: [Freeswitch-users] mod_socket: custom timeout for HEARTBEAT
	event
In-Reply-To: <191c3a030910120726p46d6410ejd94b61612686fbbc@mail.gmail.com>
References: <d211f54a0910110959y3d3deed7s43201a3e39756585@mail.gmail.com>
	<86a32abc0910111444j1ab2e446p3cb94b0ae04126e3@mail.gmail.com>
	<97B4BDC9-5B11-4A98-B1BF-CC00A82ECBD9@jerris.com>
	<d211f54a0910112238i5f69275ey7cad437e47e6b567@mail.gmail.com>
	<191c3a030910120726p46d6410ejd94b61612686fbbc@mail.gmail.com>
Message-ID: <d211f54a0910130832m25aaadd3t4d6e938d1c71d701@mail.gmail.com>

Sorry for my foolishness but I stil can't grasp it.
I'm developing app based on inbound mod_event_socket and I don't know how to
run "enable_heartbeat" or " sched_heartbeat" without specifying any alive
session uuid. I tried to use "create_uuid" and send mentioned  commands to
the created uuid  but this approach doesn't work. I do need to monitor the
FS state itself rather then any particular call.

Please, enlighten me!
Artem



On Mon, Oct 12, 2009 at 6:26 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> it works in either case with or without media
> the syntax for setting the frequency was answered above.
>
>
>
> On Mon, Oct 12, 2009 at 12:38 AM, Artem Shiyanov <shiyanov at gmail.com>wrote:
>
>> Michael, Diego,
>> thanks for the rapid answers!
>>
>> As far as I understand, "enable_heartbeat" app is launching
>> SESSION_HEARTBEAT events that will stop when the call will be cleared. Also
>> I "heard" that "enable_heartbeat" works only for calls with proxied media.
>>
>> What I want is to monitor FreeSwitch status: is it alive and what is the
>> system status message. This info is provided in HEARTBEAT event gracefully
>> but in constant time period = 20 sec. So the main question is- how to
>> customize this period?
>>
>>
>> Artem
>>
>>
>>
>>
>> On Mon, Oct 12, 2009 at 2:52 AM, Michael Jerris <mike at jerris.com> wrote:
>>
>>>
>>> On Oct 11, 2009, at 5:44 PM, Diego Viola wrote:
>>>
>>> > Maybe you want enable_heartbeat or sched_heartbeat from mod_dptools?
>>> >
>>> > You can pass your parameters in second to these two.
>>> >
>>> > Example:
>>> >
>>> > <action application="enable_heartbeat" data="1"/>
>>> > <action application="sched_heartbeat" data="1"/>
>>> >
>>> > Where 1 in this case is the number of heartbeats per seconds.
>>> >
>>>
>>> Number of seconds between hearbeats, not hearbeats per second.
>>>
>>>
>>> > You can use that example on the Dialplan XML but you can also use it
>>> > on mod_event_socket outbound, etc.
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Tue Oct 13 08:42:46 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 13 Oct 2009 11:42:46 -0400
Subject: [Freeswitch-users] Fwd: Groups information in sqllite
In-Reply-To: <f8af5740910122302i47ccde5bo9df2bae965408394@mail.gmail.com>
References: <f8af5740910122301u338ca128macfc0b769ea0e961@mail.gmail.com>
	<f8af5740910122302i47ccde5bo9df2bae965408394@mail.gmail.com>
Message-ID: <B3C83D64-BC34-4F67-B72E-9EAE6F3A362E@jerris.com>

Group information is not stored in sqlite, it is pulled from the xml  
registry (switch_xml_locate_group function can find them) .  Also,  
please do not cross post between lists.

http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups
http://wiki.freeswitch.org/wiki/Mod_commands#in_group
http://wiki.freeswitch.org/wiki/Mod_commands#group_call

Mike


On Oct 13, 2009, at 2:02 AM, srinivasula reddy wrote:
> can any know where group information is exactly stored in sqllite  
> database, i have seen sip_registration here i can find the  
> registered users,
> in the same way how i can i find the group information, and which  
> user belongs to which user?
> any help would be great.




From pippyduck1127 at hotmail.com  Tue Oct 13 06:58:56 2009
From: pippyduck1127 at hotmail.com (Pajongjit Buntaokit)
Date: Tue, 13 Oct 2009 13:58:56 +0000
Subject: [Freeswitch-users] Database for Audio Data
Message-ID: <BAY119-W52BD912544FAA1DE624B88B3C70@phx.gbl>


Hi,

 

Does anyone know whether FreeSWITCH has a function to automatically record every call as an audio file in a server 
or forward them to be stored in a database with additional parameters such as caller ID, date, starting time and ending time?

 

So that these recorded audio data can be queried and retrieved with the caller ID, date and time.

 

Any suggestion or guidance, please advise.

 

Thank you very much!
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
http://clk.atdmt.com/GBL/go/177141664/direct/01/
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From msc at freeswitch.org  Tue Oct 13 08:46:10 2009
From: msc at freeswitch.org (Michael S Collins)
Date: Tue, 13 Oct 2009 08:46:10 -0700
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <7d79b3930910130645y658ae041v8219c4c881fa5698@mail.gmail.com>
References: <7d79b3930910090407m5dc81d12pf2b0a47f6017804@mail.gmail.com>
	<20091009123615.BC50A3FA6E9@mail.cune.org>
	<7d79b3930910100647u7fd11413ucec0a535d68b3e3a@mail.gmail.com>
	<87f2f3b90910101009j1295d812jd98cae91e30b1eff@mail.gmail.com>
	<7d79b3930910120454m1ffc6dd4u42b91a4349597166@mail.gmail.com>
	<87f2f3b90910121149x6e2114f0q996a262ad272a1f6@mail.gmail.com>
	<7d79b3930910130645y658ae041v8219c4c881fa5698@mail.gmail.com>
Message-ID: <5E7A8451-6264-4822-8152-1B94240BDD48@freeswitch.org>


On Oct 13, 2009, at 6:45 AM, lakshmanan ganapathy  
<lakindia89 at gmail.com> wrote:

> We are using Reliance as the Carrier.
> I think, with this same Reliance carrier, in my office, they are  
> able to make outgoing calls through asterisk+libpri.

If that's the case I would be very interested in seeing a pri debug  
from a working call on an asterisk box. It might give us a clue as to  
what is not working.
-MC
>
>
> On Tue, Oct 13, 2009 at 12:19 AM, Michael Collins  
> <msc at freeswitch.org> wrote:
> Lak,
>
> Okay I will need a little bit of time to dig into the IE's and what  
> they contain. In the meantime can you tell me who the carrier is?  
> I'd like to find out if they have some specific requirements. The  
> fact that it doesn't work with libpri surprises me because that  
> would mean that Asterisk systems would probably not work with this  
> carrier as well.
>
> BTW, thanks for the very complete pastebin entries. :)
> -MC
>
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From rupa at rupa.com  Tue Oct 13 08:57:05 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 13 Oct 2009 09:57:05 -0600
Subject: [Freeswitch-users] Database for Audio Data
In-Reply-To: <BAY119-W52BD912544FAA1DE624B88B3C70@phx.gbl>
References: <BAY119-W52BD912544FAA1DE624B88B3C70@phx.gbl>
Message-ID: <b39273530910130857jb7dec2eidd71e950d3902b25@mail.gmail.com>

What I do is record all calls and store the call with the UUID as the
filename.  Then when the call is hung up a CDR entry is sent to my web
server.  This CDR contains callerid and other info I might want to
query by.  The service on the web server inserts appropriate record(s)
into the database.  The recordings are available to the webserver.
When one clicks on the "listen" link, the web server serves up the
recording by UUID in the recording directory.  I have a process that
periodically removes old recordings from that dir.  I don't purge the
CDRs, though that is certainly possible.

On Tue, Oct 13, 2009 at 7:58 AM, Pajongjit Buntaokit
<pippyduck1127 at hotmail.com> wrote:
> Hi,
>
> Does anyone know whether FreeSWITCH has a function to automatically record
> every call as an audio file in a server
> or forward them to be stored in a database with additional parameters such
> as caller ID, date, starting time and ending time?
>
> So that these recorded audio data can be queried and retrieved with the
> caller ID, date and time.
>
> Any suggestion or guidance, please advise.
>
> Thank you very much!
>
> ________________________________
> Hotmail: Trusted email with Microsoft?s powerful SPAM protection. Sign up
> now.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa



From msc at freeswitch.org  Tue Oct 13 09:52:35 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Oct 2009 09:52:35 -0700
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <891080.87141.qm@web56406.mail.re3.yahoo.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
	<891080.87141.qm@web56406.mail.re3.yahoo.com>
Message-ID: <87f2f3b90910130952r5657c8e2g6e11c22140da919c@mail.gmail.com>

I will add this thought to the weekly discussion. Perhaps we can crowdsource
this one.
-MC

On Mon, Oct 12, 2009 at 10:52 PM, Mark Crane <mctch at yahoo.com> wrote:

> "In many cases the issue with the docs isn't that they aren't complete but
> rather that they are hard to find."
>
> Agreed! The biggest problem with the wiki is that it is hard to find
> things.
>
> How do books solve this problem they use an index. It is quite surprising
> that the wiki software doesn't come with ability to do this basic task
> automatically and instead relies solely upon a manually created index. It
> would be extremely beneficial to have a index of all pages and topics that
> are available on the wiki. The average person may only see the main links in
> the documentation and not realize there are actually hundreds of pages.
>
> There used to be a pdf that was auto generated from the wiki and it gave a
> much easier view of all the pages on the wiki.
>
> Mark J Crane
>
>
>
> --- On *Mon, 10/12/09, Michael Collins <msc at freeswitch.org>* wrote:
>
>
> From: Michael Collins <msc at freeswitch.org>
> Subject: Re: [Freeswitch-users] Some documentation thoughts
> To: freeswitch-users at lists.freeswitch.org
> Date: Monday, October 12, 2009, 4:38 PM
>
>
> On Mon, Oct 12, 2009 at 3:11 PM, Diego Viola <diego.viola at gmail.com<http://mc/compose?to=diego.viola at gmail.com>
> > wrote:
>
>> Hello,
>>
>> I have been doing some work recently on the FreeSWITCH wiki, to improve
>> things.
>>
>> You can see some of my work here:
>>
>>
>> http://wiki.freeswitch.org/index.php?title=Special:Contributions&limit=500&target=Diego.viola
>>
>> I am trying to polish the wiki and give it a more professional and clean
>> look, this means that we need to enforce some guidelines and strive to make
>> the wiki even better.
>>
>> How can we do this?
>>
>> Well, the first thing is to review everything we have, right now we have
>> too many separated and little pages here and there that no one cares or
>> read, we should avoid doing little or small pages. We need to correct typos,
>> I suggest that people click on "Random page" link and start correcting
>> typos, install an English (US) dictionary in your browser and enable Spell
>> Checking and start correcting these typos if you want to help, also, make
>> acronyms and initialisms all-uppercase when you make one.
>>
>> When you are adding documentation make sure you don't do a separate page
>> for an existing page, etc. and we also need to define a size for titles,
>> body, etc, so different pages don't look different from each other, I will
>> make sure I add all this into our guidelines and that all pages follow this.
>>
>> My goal is to have great documentation, similar to the Apache
>> documentation:
>> http://httpd.apache.org/docs/2.2/
>>
>> Or even better, I hope you like this idea.
>>
>
> Diego,
>
> Thank you so much for all of your help on this. Many in the community have
> seen your work - it has not gone unnoticed and it most truly is appreciated.
> I like the idea of improving the documentation. One thing we need to do is
> re-think the organization. In many cases the issue with the docs isn't that
> they aren't complete, but rather that they are hard to find. It's all about
> organization. I'm definitely open to ideas.
>
> For those who are interested in helping out with the wiki please let me
> know of your availability and skill set. I am maintaining a list of
> volunteers.
>
> Lastly, if you want to talk about documentation in real time please join us
> in #freeswitch-docs.
>
> Thanks,
> MC
>
>
> -----Inline Attachment Follows-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From william.suffill at gmail.com  Tue Oct 13 10:09:11 2009
From: william.suffill at gmail.com (William Suffill)
Date: Tue, 13 Oct 2009 13:09:11 -0400
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <87f2f3b90910130952r5657c8e2g6e11c22140da919c@mail.gmail.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com> 
	<891080.87141.qm@web56406.mail.re3.yahoo.com>
	<87f2f3b90910130952r5657c8e2g6e11c22140da919c@mail.gmail.com>
Message-ID: <6b65470d0910131009y1cf136c3q8c0932375c19f272@mail.gmail.com>

Under special pages there is ways to get a list of all the wiki pages. (
http://wiki.freeswitch.org/wiki/Special:SpecialPages)

http://wiki.freeswitch.org/wiki/Special:AllPages

Due to the number of pages it's broken into sub pages based in alphabetical
order of the page names.

-- W
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From tculjaga at gmail.com  Tue Oct 13 10:12:11 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 13 Oct 2009 19:12:11 +0200
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <5e414ed0910130826j11d45a66q86d785f900e4196e@mail.gmail.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
	<191c3a030910130803k321f39e3n265e64ab3c84a5e6@mail.gmail.com>
	<5e414ed0910130826j11d45a66q86d785f900e4196e@mail.gmail.com>
Message-ID: <65d96fc80910131012v1f72ac49o6bfe9f2267e541c5@mail.gmail.com>

you need a softswitch.... i'm afraid a SIP phone is not designed for
overlap...

T.

On Tue, Oct 13, 2009 at 5:26 PM, Dennis <odermann at googlemail.com> wrote:

> how could we try? we played arround with a snom phone (snom seems to
> support something in this direction, but are not shure, how we can
> test it and how we can see if it is supported or not.
>
> any hint?
>
>
> 2009/10/13 Anthony Minessale <anthony.minessale at gmail.com>:
> > have you tried it?
> > I *think* either we did support it or we would with a small patch to
> sofia
> > lib that I cannot recall if we ever got committed.
> >
> >
> > On Tue, Oct 13, 2009 at 8:51 AM, Dennis <odermann at googlemail.com> wrote:
> >>
> >> hi there,
> >>
> >> i would like to ask, if fs has support for something like "SIP Overlap"?
> >>
> >> instead of receiving the phonenumber from our carrier in a block, we
> >> want to receive the phonenumber digit-by-digit and we want to tell fs
> >> when the number is complete. our carrier could send us the phonenumber
> >> digit-by-digit, but what about the fs-side?
> >>
> >>
> >> thanks and kind regards
> >> dennis
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:213-799-1400
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Tue Oct 13 11:01:29 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 13 Oct 2009 13:01:29 -0500
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <65d96fc80910131012v1f72ac49o6bfe9f2267e541c5@mail.gmail.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
	<191c3a030910130803k321f39e3n265e64ab3c84a5e6@mail.gmail.com>
	<5e414ed0910130826j11d45a66q86d785f900e4196e@mail.gmail.com>
	<65d96fc80910131012v1f72ac49o6bfe9f2267e541c5@mail.gmail.com>
Message-ID: <191c3a030910131101p418e0961t373db68285248ff6@mail.gmail.com>

i do think some softphone can do it but i forgot which one it was either
snom or grandstream


On Tue, Oct 13, 2009 at 12:12 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:

> you need a softswitch.... i'm afraid a SIP phone is not designed for
> overlap...
>
> T.
>
>
> On Tue, Oct 13, 2009 at 5:26 PM, Dennis <odermann at googlemail.com> wrote:
>
>> how could we try? we played arround with a snom phone (snom seems to
>> support something in this direction, but are not shure, how we can
>> test it and how we can see if it is supported or not.
>>
>> any hint?
>>
>>
>> 2009/10/13 Anthony Minessale <anthony.minessale at gmail.com>:
>> > have you tried it?
>> > I *think* either we did support it or we would with a small patch to
>> sofia
>> > lib that I cannot recall if we ever got committed.
>> >
>> >
>> > On Tue, Oct 13, 2009 at 8:51 AM, Dennis <odermann at googlemail.com>
>> wrote:
>> >>
>> >> hi there,
>> >>
>> >> i would like to ask, if fs has support for something like "SIP
>> Overlap"?
>> >>
>> >> instead of receiving the phonenumber from our carrier in a block, we
>> >> want to receive the phonenumber digit-by-digit and we want to tell fs
>> >> when the number is complete. our carrier could send us the phonenumber
>> >> digit-by-digit, but what about the fs-side?
>> >>
>> >>
>> >> thanks and kind regards
>> >> dennis
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> > pstn:213-799-1400
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From diego.viola at gmail.com  Tue Oct 13 11:45:38 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 13 Oct 2009 18:45:38 +0000
Subject: [Freeswitch-users] sched_api doesn't get launched
In-Reply-To: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
References: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
Message-ID: <86a32abc0910131145t730d0bc0m842ed7ed0891d425@mail.gmail.com>

You need to pass the UUID to sched_hangup.

Usage: sched_hangup [+]<time> <uuid> [<cause>]

http://wiki.freeswitch.org/wiki/Mod_commands#sched_hangup


On Tue, Oct 13, 2009 at 10:14 AM, Henry Huang <b_ball_henry at hotmail.com>wrote:

> Hi:
> I am using mod_java. And in my script I was able to achieve using:
>
> execute("sched_hangup", "+300 alloted_timeout");
>
> However, when I try to run sched_api in the same way, system log returns
> that it's an invalid application. I have also tried to trigger it with many
> conditional channel variable api calls , but non of them seemed to execute
> the api command (because I turned on the highest level of debugging and see
> no where the sched_api is being called.
>
> The closest thing I got was by using "api_after_bridge" like the following,
> but it only launches when the bridge is teared down(which is not what I
> want). I originally thought after bridge means right after the 2 party is
> connected. All I want is to be able to play some message to leg A at certain
> time.
>
> setVariable("api_after_bridge", "sched_api +10 none uuid_displace ${uuid}
> start /path/to/some.wav 20 mux");
>
> I have been struggling with different combination for a week now.. Please
> shed some light if you know something.
>
> Thanks,
>
> --
> Henry Huang
> UniC Solution - Communication Unified
> VoIP & Open Source software Consultant
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Oct 13 11:48:21 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Oct 2009 11:48:21 -0700
Subject: [Freeswitch-users] mod_socket: custom timeout for HEARTBEAT
	event
In-Reply-To: <d211f54a0910130832m25aaadd3t4d6e938d1c71d701@mail.gmail.com>
References: <d211f54a0910110959y3d3deed7s43201a3e39756585@mail.gmail.com>
	<86a32abc0910111444j1ab2e446p3cb94b0ae04126e3@mail.gmail.com>
	<97B4BDC9-5B11-4A98-B1BF-CC00A82ECBD9@jerris.com>
	<d211f54a0910112238i5f69275ey7cad437e47e6b567@mail.gmail.com>
	<191c3a030910120726p46d6410ejd94b61612686fbbc@mail.gmail.com>
	<d211f54a0910130832m25aaadd3t4d6e938d1c71d701@mail.gmail.com>
Message-ID: <87f2f3b90910131148m1853a219md8788f55e40ef4c3@mail.gmail.com>

On Tue, Oct 13, 2009 at 8:32 AM, Artem Shiyanov <shiyanov at gmail.com> wrote:

> Sorry for my foolishness but I stil can't grasp it.
> I'm developing app based on inbound mod_event_socket and I don't know how
> to run "enable_heartbeat" or " sched_heartbeat" without specifying any alive
> session uuid. I tried to use "create_uuid" and send mentioned  commands to
> the created uuid  but this approach doesn't work. I do need to monitor the
> FS state itself rather then any particular call.
>
> Please, enlighten me!
> Artem
>

So if I understand correctly, you want to see the system heartbeat more
frequently than every 20 seconds? What, exactly, is in the heartbeat that
you need? You can get the same information by sending the API "status" and
reading the results.

If I read the source code correctly it looks like the system heartbeat is
hard-coded at 20 seconds. See switch_core.c, line 155. However, before you
go messing with a core file I would make sure that you can't already use the
tools that are available, like sending a "status" API every 5 seconds (or
whatever interval you need).

-MC
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From freeswitch-users-list at metik.com  Tue Oct 13 12:03:50 2009
From: freeswitch-users-list at metik.com (Metik)
Date: Tue, 13 Oct 2009 15:03:50 -0400
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <191c3a030910131101p418e0961t373db68285248ff6@mail.gmail.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com><191c3a030910130803k321f39e3n265e64ab3c84a5e6@mail.gmail.com><5e414ed0910130826j11d45a66q86d785f900e4196e@mail.gmail.com><65d96fc80910131012v1f72ac49o6bfe9f2267e541c5@mail.gmail.com>
	<191c3a030910131101p418e0961t373db68285248ff6@mail.gmail.com>
Message-ID: <538F4CF43B294D0EACABA281E1913B4C@chaos>

Both support it.  In the Grandstream, I believe it is called Early Dial (vs. SNOM's Overlap Dialing).  It can be problematic if you have a device somewhere in the middle that doesn't support 484s.

-metik
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, October 13, 2009 2:01 PM
  Subject: Re: [Freeswitch-users] SIP Overlap support?


  i do think some softphone can do it but i forgot which one it was either snom or grandstream



  On Tue, Oct 13, 2009 at 12:12 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

    you need a softswitch.... i'm afraid a SIP phone is not designed for overlap...

    T.



    On Tue, Oct 13, 2009 at 5:26 PM, Dennis <odermann at googlemail.com> wrote:

      how could we try? we played arround with a snom phone (snom seems to
      support something in this direction, but are not shure, how we can
      test it and how we can see if it is supported or not.

      any hint?


      2009/10/13 Anthony Minessale <anthony.minessale at gmail.com>:

      > have you tried it?
      > I *think* either we did support it or we would with a small patch to sofia
      > lib that I cannot recall if we ever got committed.
      >
      >
      > On Tue, Oct 13, 2009 at 8:51 AM, Dennis <odermann at googlemail.com> wrote:
      >>
      >> hi there,
      >>
      >> i would like to ask, if fs has support for something like "SIP Overlap"?
      >>
      >> instead of receiving the phonenumber from our carrier in a block, we
      >> want to receive the phonenumber digit-by-digit and we want to tell fs
      >> when the number is complete. our carrier could send us the phonenumber
      >> digit-by-digit, but what about the fs-side?
      >>
      >>
      >> thanks and kind regards
      >> dennis
      >>
      >> _______________________________________________
      >> FreeSWITCH-users mailing list
      >> FreeSWITCH-users at lists.freeswitch.org
      >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      >> http://www.freeswitch.org
      >
      >
      >
      > --
      > Anthony Minessale II
      >
      > FreeSWITCH http://www.freeswitch.org/
      > ClueCon http://www.cluecon.com/
      > Twitter: http://twitter.com/FreeSWITCH_wire
      >
      > AIM: anthm
      > MSN:anthony_minessale at hotmail.com
      > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
      > IRC: irc.freenode.net #freeswitch
      >
      > FreeSWITCH Developer Conference
      > sip:888 at conference.freeswitch.org
      > iax:guest at conference.freeswitch.org/888
      > googletalk:conf+888 at conference.freeswitch.org
      > pstn:213-799-1400
      >
      > _______________________________________________
      > FreeSWITCH-users mailing list
      > FreeSWITCH-users at lists.freeswitch.org
      > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      > http://www.freeswitch.org
      >
      >

      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org




    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





  -- 
  Anthony Minessale II

  FreeSWITCH http://www.freeswitch.org/
  ClueCon http://www.cluecon.com/
  Twitter: http://twitter.com/FreeSWITCH_wire

  AIM: anthm
  MSN:anthony_minessale at hotmail.com
  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
  IRC: irc.freenode.net #freeswitch

  FreeSWITCH Developer Conference
  sip:888 at conference.freeswitch.org
  iax:guest at conference.freeswitch.org/888
  googletalk:conf+888 at conference.freeswitch.org
  pstn:213-799-1400



------------------------------------------------------------------------------


  _______________________________________________
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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From tculjaga at gmail.com  Tue Oct 13 12:24:55 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 13 Oct 2009 21:24:55 +0200
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <538F4CF43B294D0EACABA281E1913B4C@chaos>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
	<191c3a030910130803k321f39e3n265e64ab3c84a5e6@mail.gmail.com>
	<5e414ed0910130826j11d45a66q86d785f900e4196e@mail.gmail.com>
	<65d96fc80910131012v1f72ac49o6bfe9f2267e541c5@mail.gmail.com>
	<191c3a030910131101p418e0961t373db68285248ff6@mail.gmail.com>
	<538F4CF43B294D0EACABA281E1913B4C@chaos>
Message-ID: <65d96fc80910131224m73f35b87g355be818ad3c0caf@mail.gmail.com>

i never found it working properly... i always had some interoperability
issues and i finished having a "dialplan" on my phones being delivered
through a config file via tftp or http .. depending of the phone capability.

BTW: using overlap can lead to a greater system load... be careful when
setting the minimum number of digits you will send in 1st message. I wish
you luck...

T.

On Tue, Oct 13, 2009 at 9:03 PM, Metik <freeswitch-users-list at metik.com>wrote:

>  Both support it.  In the Grandstream, I believe it is called Early Dial
> (vs. SNOM's Overlap Dialing).  It can be problematic if you have a device
> somewhere in the middle that doesn't support 484s.
>
> -metik
>
> ----- Original Message -----
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, October 13, 2009 2:01 PM
> *Subject:* Re: [Freeswitch-users] SIP Overlap support?
>
> i do think some softphone can do it but i forgot which one it was either
> snom or grandstream
>
>
> On Tue, Oct 13, 2009 at 12:12 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> you need a softswitch.... i'm afraid a SIP phone is not designed for
>> overlap...
>>
>> T.
>>
>>
>> On Tue, Oct 13, 2009 at 5:26 PM, Dennis <odermann at googlemail.com> wrote:
>>
>>> how could we try? we played arround with a snom phone (snom seems to
>>> support something in this direction, but are not shure, how we can
>>> test it and how we can see if it is supported or not.
>>>
>>> any hint?
>>>
>>>
>>> 2009/10/13 Anthony Minessale <anthony.minessale at gmail.com>:
>>>  > have you tried it?
>>> > I *think* either we did support it or we would with a small patch to
>>> sofia
>>> > lib that I cannot recall if we ever got committed.
>>> >
>>> >
>>> > On Tue, Oct 13, 2009 at 8:51 AM, Dennis <odermann at googlemail.com>
>>> wrote:
>>> >>
>>> >> hi there,
>>> >>
>>> >> i would like to ask, if fs has support for something like "SIP
>>> Overlap"?
>>> >>
>>> >> instead of receiving the phonenumber from our carrier in a block, we
>>> >> want to receive the phonenumber digit-by-digit and we want to tell fs
>>> >> when the number is complete. our carrier could send us the phonenumber
>>> >> digit-by-digit, but what about the fs-side?
>>> >>
>>> >>
>>> >> thanks and kind regards
>>> >> dennis
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> >
>>> > --
>>> > Anthony Minessale II
>>> >
>>> > FreeSWITCH http://www.freeswitch.org/
>>> > ClueCon http://www.cluecon.com/
>>> > Twitter: http://twitter.com/FreeSWITCH_wire
>>> >
>>> > AIM: anthm
>>> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> > IRC: irc.freenode.net #freeswitch
>>> >
>>> > FreeSWITCH Developer Conference
>>> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>> > iax:guest at conference.freeswitch.org/888
>>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> > pstn:213-799-1400
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From larclap at yahoo.com  Tue Oct 13 12:38:35 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Tue, 13 Oct 2009 12:38:35 -0700
Subject: [Freeswitch-users] lua script causing FreeSwitch to crash?
Message-ID: <005a01ca4c3c$c183ceb0$448b6c10$@com>

http://pastebin.freeswitch.org/10686

 

I am trying to use a lua script for inbound calls. The caller hears a busy
signal. After this call fails to go through, FreeSwitch is no longer in
memory - it does not appear in  'ps -ef' output.

 

I don't have a clue what I might be doing incorrectly in the script in order
to cause this to happen. There is no indication in the log that FreeSwitch
has terminated. But it is not resident in memory.

 

I am using v15134. I had been using this same script without incident as
late as v14929. Until I can resolve the issue, I have removed the call to
lua from the dialplan and done all I could in it instead.

 

I have included the dialplan, the script and the cli log in the pastebin. I
would appreciate any help or suggestions.

 

Thanks, Lars

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From msc at freeswitch.org  Tue Oct 13 13:12:21 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Oct 2009 13:12:21 -0700
Subject: [Freeswitch-users] lua script causing FreeSwitch to crash?
In-Reply-To: <005a01ca4c3c$c183ceb0$448b6c10$@com>
References: <005a01ca4c3c$c183ceb0$448b6c10$@com>
Message-ID: <87f2f3b90910131312k7dac8980h70ce8be9c07aa7b2@mail.gmail.com>

On Tue, Oct 13, 2009 at 12:38 PM, Lars Zeb <larclap at yahoo.com> wrote:

>  http://pastebin.freeswitch.org/10686
>
>
>
> I am trying to use a lua script for inbound calls. The caller hears a busy
> signal. After this call fails to go through, FreeSwitch is no longer in
> memory ? it does not appear in  ?ps ?ef? output.
>
>
>
> I don?t have a clue what I might be doing incorrectly in the script in
> order to cause this to happen. There is no indication in the log that
> FreeSwitch has terminated. But it is not resident in memory.
>
>
>
> I am using v15134. I had been using this same script without incident as
> late as v14929. Until I can resolve the issue, I have removed the call to
> lua from the dialplan and done all I could in it instead.
>
We sent out a notice about a bug in event socket between revs 15119 and
15134. Update to latest trunk. (FYI, the fix was in r15135, so you just
barely missed it.)

After you update, let us know if the problem persists.
-MC
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From freeswitch-users-list at metik.com  Tue Oct 13 13:22:00 2009
From: freeswitch-users-list at metik.com (Metik)
Date: Tue, 13 Oct 2009 16:22:00 -0400
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <65d96fc80910131224m73f35b87g355be818ad3c0caf@mail.gmail.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com><191c3a030910130803k321f39e3n265e64ab3c84a5e6@mail.gmail.com><5e414ed0910130826j11d45a66q86d785f900e4196e@mail.gmail.com><65d96fc80910131012v1f72ac49o6bfe9f2267e541c5@mail.gmail.com><191c3a030910131101p418e0961t373db68285248ff6@mail.gmail.com><538F4CF43B294D0EACABA281E1913B4C@chaos>
	<65d96fc80910131224m73f35b87g355be818ad3c0caf@mail.gmail.com>
Message-ID: <48B637E0506F4A809E4C0CAF39BD08BB@chaos>

As evidenced by various DTMF interop issues (with RFC2833, inband, etc) over the years, I would avoid it if at all possible.  

What does it particularly do that can not accomplished by using RFC 2833 or (less ideal) inband DTMF?  Or are you attempting to use it as a band-aid to address some sort of interop issue with the carrier involved that is wrecking havoc with your particular application?

-metik
  ----- Original Message ----- 
  From: Tihomir Culjaga 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, October 13, 2009 3:24 PM
  Subject: Re: [Freeswitch-users] SIP Overlap support?


  i never found it working properly... i always had some interoperability issues and i finished having a "dialplan" on my phones being delivered through a config file via tftp or http .. depending of the phone capability.

  BTW: using overlap can lead to a greater system load... be careful when setting the minimum number of digits you will send in 1st message. I wish you luck...

  T.


  On Tue, Oct 13, 2009 at 9:03 PM, Metik <freeswitch-users-list at metik.com> wrote:

    Both support it.  In the Grandstream, I believe it is called Early Dial (vs. SNOM's Overlap Dialing).  It can be problematic if you have a device somewhere in the middle that doesn't support 484s.

    -metik
      ----- Original Message ----- 
      From: Anthony Minessale 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Tuesday, October 13, 2009 2:01 PM
      Subject: Re: [Freeswitch-users] SIP Overlap support?


      i do think some softphone can do it but i forgot which one it was either snom or grandstream



      On Tue, Oct 13, 2009 at 12:12 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

        you need a softswitch.... i'm afraid a SIP phone is not designed for overlap...

        T. 



        On Tue, Oct 13, 2009 at 5:26 PM, Dennis <odermann at googlemail.com> wrote:

          how could we try? we played arround with a snom phone (snom seems to
          support something in this direction, but are not shure, how we can
          test it and how we can see if it is supported or not.

          any hint?


          2009/10/13 Anthony Minessale <anthony.minessale at gmail.com>:

          > have you tried it?
          > I *think* either we did support it or we would with a small patch to sofia
          > lib that I cannot recall if we ever got committed.
          >
          >
          > On Tue, Oct 13, 2009 at 8:51 AM, Dennis <odermann at googlemail.com> wrote:
          >>
          >> hi there,
          >>
          >> i would like to ask, if fs has support for something like "SIP Overlap"?
          >>
          >> instead of receiving the phonenumber from our carrier in a block, we
          >> want to receive the phonenumber digit-by-digit and we want to tell fs
          >> when the number is complete. our carrier could send us the phonenumber
          >> digit-by-digit, but what about the fs-side?
          >>
          >>
          >> thanks and kind regards
          >> dennis
          >>
          >> _______________________________________________
          >> FreeSWITCH-users mailing list
          >> FreeSWITCH-users at lists.freeswitch.org
          >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
          >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
          >> http://www.freeswitch.org
          >
          >
          >
          > --
          > Anthony Minessale II
          >
          > FreeSWITCH http://www.freeswitch.org/
          > ClueCon http://www.cluecon.com/
          > Twitter: http://twitter.com/FreeSWITCH_wire
          >
          > AIM: anthm
          > MSN:anthony_minessale at hotmail.com
          > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
          > IRC: irc.freenode.net #freeswitch
          >
          > FreeSWITCH Developer Conference
          > sip:888 at conference.freeswitch.org
          > iax:guest at conference.freeswitch.org/888
          > googletalk:conf+888 at conference.freeswitch.org
          > pstn:213-799-1400
          >
          > _______________________________________________
          > FreeSWITCH-users mailing list
          > FreeSWITCH-users at lists.freeswitch.org
          > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
          > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
          > http://www.freeswitch.org
          >
          >

          _______________________________________________
          FreeSWITCH-users mailing list
          FreeSWITCH-users at lists.freeswitch.org
          http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
          UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
          http://www.freeswitch.org




        _______________________________________________
        FreeSWITCH-users mailing list
        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org





      -- 
      Anthony Minessale II

      FreeSWITCH http://www.freeswitch.org/
      ClueCon http://www.cluecon.com/
      Twitter: http://twitter.com/FreeSWITCH_wire

      AIM: anthm
      MSN:anthony_minessale at hotmail.com
      GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
      IRC: irc.freenode.net #freeswitch

      FreeSWITCH Developer Conference
      sip:888 at conference.freeswitch.org
      iax:guest at conference.freeswitch.org/888
      googletalk:conf+888 at conference.freeswitch.org
      pstn:213-799-1400



--------------------------------------------------------------------------


      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org


    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org






------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From msc at freeswitch.org  Tue Oct 13 13:38:05 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Oct 2009 13:38:05 -0700
Subject: [Freeswitch-users] FS Extension Groups Documentation
In-Reply-To: <E8FDE7D1E87C4DDC98BD76DAF7DC50A5@greyhawk.tonecommander.com>
References: <E8FDE7D1E87C4DDC98BD76DAF7DC50A5@greyhawk.tonecommander.com>
Message-ID: <87f2f3b90910131338o2d161d1w397295efefd4138f@mail.gmail.com>

On Tue, Oct 13, 2009 at 7:54 AM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>
> Is there a link to the documentation of FS groups and what can be done with
> a group (i.e. capabilities)?
>
> Thanks And Best Regards,
> Jerry
>

http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups
http://wiki.freeswitch.org/wiki/Mod_commands#group_call

And this intercept example from the default.xml dialplan file:
    <extension name="group-intercept">
      <condition field="destination_number" expression="^\*8$">
    <action application="answer"/>
    <action application="intercept"
data="${hash(select/${domain_name}-last_dial/${callgroup})}"/>
    <action application="sleep" data="2000"/>
      </condition>
    </extension>

Note: groups in the dialplan and directory are not related to the mod_limit
group insert/delete/call stuff.
-MC
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From msc at freeswitch.org  Tue Oct 13 13:47:13 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Oct 2009 13:47:13 -0700
Subject: [Freeswitch-users] FreeSWITCH Conf Call Agenda For October 16th
Message-ID: <87f2f3b90910131347g23bd5f11p8e17a91374ac2a76@mail.gmail.com>

FYI,

The agenda for this week's call is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_16

Please feel free to add agenda items and updates as you see fit.

-Michael
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From diego.viola at gmail.com  Tue Oct 13 14:53:02 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 13 Oct 2009 21:53:02 +0000
Subject: [Freeswitch-users] Some help with my post-paid billing project
Message-ID: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>

Hello,

I'm trying to write a post-paid billing script, I have the CDR on my
database and also a "rates" table, the CDR contains fields like
caller_destination_number, variable_duration, etc. and the rates table
contains: destination, prefix, rate (cost).

The problem is that I can't just strip the destination number to take the
prefix from it because I have to deal with destination numbers from
different countries and they all have different prefix lengths... so I need
to find another way to take the prefix from the destination number.

Any ideas how to do this?

Thanks,

Diego
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From diego.viola at gmail.com  Tue Oct 13 14:58:47 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 13 Oct 2009 21:58:47 +0000
Subject: [Freeswitch-users] Some help with my post-paid billing project
In-Reply-To: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
References: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
Message-ID: <86a32abc0910131458i207c59d1x1c4f0dce4dfa4ccb@mail.gmail.com>

Should I try to pass the whole DNIS and compare with teh rates list/table
and when a prefix from the DNIS matches with the rates list I get hte rate
that way?


On Tue, Oct 13, 2009 at 9:53 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Hello,
>
> I'm trying to write a post-paid billing script, I have the CDR on my
> database and also a "rates" table, the CDR contains fields like
> caller_destination_number, variable_duration, etc. and the rates table
> contains: destination, prefix, rate (cost).
>
> The problem is that I can't just strip the destination number to take the
> prefix from it because I have to deal with destination numbers from
> different countries and they all have different prefix lengths... so I need
> to find another way to take the prefix from the destination number.
>
> Any ideas how to do this?
>
> Thanks,
>
> Diego
>
>
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From mcampbellsmith at gmail.com  Tue Oct 13 15:02:50 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Wed, 14 Oct 2009 09:02:50 +1100
Subject: [Freeswitch-users] Dingaling: using a hostname instead of stun for
	rtp
Message-ID: <33c87fa30910131502t2af24801i1c3319518b5db50b@mail.gmail.com>

Hi!

I have a hostname set in vars.conf.xml for the parameters
external_rtp_ip and the external_sip_ip instead of the usual stun.  I
found that stun was timing out and was causing some problems.  And as
I have a hostname, it makes sense to use that instead of relying on
stun.

However, when I use Dingaling, I see the following error message in
the dl_debug tracing:

<iq to="<FS_EMAIL>@gmail.com/gtalk77BBCB94" id="344" type="error"
from="<email1>@gmail.com/Talk.v10488E8134A">
  <ses:session type="candidates" id="3697758817"
initiator=""<email1>@gmail.com/Talk.v10488D9134A"
xmlns:ses="http://www.google.com/session">
    <ses:candidate name="rtp" address="host:<hostname>" port="30710"
username="nVv7FqwNyWbiMuTt" password="nVv7BqwQyVbiMuTt"
preference="1.0" protocol="udp" type="local" network="0"
generation="0"></ses:candidate>
  </ses:session>
  <error type="modify">
    <sta:bad-request
xmlns:sta="urn:ietf:params:xml:ns:xmpp-stanzas"></sta:bad-request>
    <sta:text xml:lang="en"
xmlns:sta="urn:ietf:params:xml:ns:xmpp-stanzas">candidate has address
of zero</sta:text>
  </error>
</iq>

Is Gtalk expecting an address in this field '<ses:candidate name="rtp"
address="host:<hostname>"' instead of a hostname?  Or should FS be
sending the hostname only and not host:hostname?

Thanks!



From grevenx at me.com  Tue Oct 13 15:45:11 2009
From: grevenx at me.com (=?iso-8859-1?Q?Even_Andr=E9_Fiskvik?=)
Date: Wed, 14 Oct 2009 00:45:11 +0200
Subject: [Freeswitch-users] Some help with my post-paid billing project
In-Reply-To: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
References: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
Message-ID: <74730C11-75D0-4BC0-9F63-8B6C9F0F499F@me.com>

What database are you using?
You could do this with regular SQL, but it would by a costly operation,
for PostgreSQL we're using the prefix module: http://pgfoundry.org/projects/prefix/

You can then match the closest prefix by using something like
"WHERE myprefix_col @> caller_destination_number ORDER BY LENGTH 
(myprefix_col::text) LIMIT 1;"


Best regards,
Even Andr?


On 13. okt. 2009, at 23.53, Diego Viola wrote:

> Hello,
>
> I'm trying to write a post-paid billing script, I have the CDR on my  
> database and also a "rates" table, the CDR contains fields like  
> caller_destination_number, variable_duration, etc. and the rates  
> table contains: destination, prefix, rate (cost).
>
> The problem is that I can't just strip the destination number to  
> take the prefix from it because I have to deal with destination  
> numbers from different countries and they all have different prefix  
> lengths... so I need to find another way to take the prefix from the  
> destination number.
>
> Any ideas how to do this?
>
> Thanks,
>
> Diego
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From kjv at ken-ton.com  Tue Oct 13 15:59:36 2009
From: kjv at ken-ton.com (Karl Vesterling)
Date: Tue, 13 Oct 2009 18:59:36 -0400
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <87f2f3b90910090005i51dbfc60t20dd77789872bc9e@mail.gmail.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<87f2f3b90910080958i1f5cfccax621fd12100a05e3e@mail.gmail.com>
	<903da5680910081201p427a8358n768edb61a96674a4@mail.gmail.com>
	<87f2f3b90910081538l6e4e4b7dv3aedb2c4d1af58bc@mail.gmail.com>
	<903da5680910082238x27bff476s8bea0e74ab516a8d@mail.gmail.com>
	<87f2f3b90910082308i7f6144d9kc887596be54dd609@mail.gmail.com>
	<903da5680910082341r485092vb0cd10c43c4a2534@mail.gmail.com>
	<87f2f3b90910090005i51dbfc60t20dd77789872bc9e@mail.gmail.com>
Message-ID: <BC5F7F95-767B-4A1D-95F5-FB393E31E6D3@ken-ton.com>


If that's the case, does this also apply to VoIP gateway services?

If so, I'll consider phrasing this differently:

http://wiki.freeswitch.org/wiki/Voicepulse.xml
--begin--
Voicepulse Warning

WARNING!!! Recent changes (Sep 2008) in VoicePulse have limited  
connect03 to only IAX termination. Please remove any entries that you  
may have reflecting the gateway below from your dialplan as it will  
most likely answer the call and play a message saying something to the  
effect of "Your call could not be completed as dialed because we  
changed our production environment in an effort to demonstrate what  
bad engineering is all about. BAD is an acronymn for Broken As  
Designed. Please hang up and make changes on all your switches. Even  
though we shouldn't have answered the call and simply ignored it  
entirely so that another gateway could facilitate your call, we FELT  
it wise to seriously inconvenience you by answering the call  
immediately giving no other gateway a chance to place the call. Thanks  
for using our service and we look forward to abusing you again in the  
near future.
UPDATE - Problem getting worse, not better 3/30/2009. I switched DID  
providers because of problems w/ Voicepulse, such as echo, dropped  
calls, and consistent registration problems hence calls don't get  
placed and inbound don't get routed to me. Incidentally, they will not  
port DID's, they'll tell you they will, but they don't.
--end--

Perhaps just saying, "stay away, far far away. Recommend this service  
only to people and/or organizations you don't like or are in direct  
competition with" would suffice eh?

Suggestions? Recommendations?

Best Regards,
Karl J. Vesterling
kjv at ken-ton.com
202-461-3231 x0

On Oct 9, 2009, at 3:05 AM, Michael Collins wrote:

>
>
> On Thu, Oct 8, 2009 at 11:41 PM, Gabriel Gunderson <gabe at gundy.org>  
> wrote:
> On Fri, Oct 9, 2009 at 12:08 AM, Michael Collins  
> <msc at freeswitch.org> wrote:
> > Thanks for reporting back. Please let all the Asterisk users know  
> that they
> > are welcome to join us in #freeswitch on irc.freenode.net and that  
> they will
> > not be abused like people do in other less friendly IRC channels.
>
> Funny you mention this.  Many people report that the way the FS
> community refers to Asterisk in docs/wikis/irc/whatever makes the FS
> camp seem *less* welcoming to them.  After all, they identify as
> Asterisk Users and take the criticism as being kinda harsh.  Most of
> them acknowledge the shortcomings of Asterisk but are put off when
> someone else points them out.  It's crazy, I know.  The thing is, I
> remember thinking that too.  After getting to know FS better, I didn't
> notice it as much.  Nobody likes to hear their baby is ugly --even if
> they know it is.
>
> Interesting. I'd like to know what specifically turned them off. It  
> could be node 117, but that was written by Anthony in response to  
> dozens of questions about the subject. In any case, if the Asterisk  
> guys look at the first edition of the Starfish/TFOT book they'll see  
> glowing praise of Anthony Minessale and Brian West, so maybe they'll  
> be willing to listen to those guys since they've got so much  
> Asterisk knowledge. The other thing is that many people in the  
> FreeSWITCH community are Asterisk refugees. They left Asterisk for  
> FreeSWITCH because they had some sort of trouble with Asterisk. If  
> that comes across in the docs then we'll have to see about cleaning  
> that up.
>
> At our session, and in general, I've noticed people are more
> interested in hearing about FS when you don't make direct comparisons
> to Asterisk.  Besides, FS stands on it's own merit.
>
> Agreed. Direct comparisons can always be slanted in any direction.  
> Besides, FreeSWITCH stacks up very well against a lot of modern  
> (i.e. expensive) equipment put out by the big boys (Cisco, Avaya,  
> NEC, ShoreTel, etc.).
>
> Just what I've observed *and* my 2 additional cents.
>
> Gabe
> Thanks,
> MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From rupa at rupa.com  Tue Oct 13 16:07:48 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 13 Oct 2009 18:07:48 -0500
Subject: [Freeswitch-users] Some help with my post-paid billing project
In-Reply-To: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
References: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
Message-ID: <b39273530910131607s2ad3306fp63e715e33a266c9c@mail.gmail.com>

Why not use mod_lcr to determine your rates?  It already does all this
work for you.  I even added a feature recently that allows one to
calculate/lookup end user rates (in addition to the carrier rate you
are paying for a given route).

On Tue, Oct 13, 2009 at 4:53 PM, Diego Viola <diego.viola at gmail.com> wrote:
> Hello,
>
> I'm trying to write a post-paid billing script, I have the CDR on my
> database and also a "rates" table, the CDR contains fields like
> caller_destination_number, variable_duration, etc. and the rates table
> contains: destination, prefix, rate (cost).
>
> The problem is that I can't just strip the destination number to take the
> prefix from it because I have to deal with destination numbers from
> different countries and they all have different prefix lengths... so I need
> to find another way to take the prefix from the destination number.
>
> Any ideas how to do this?
>
> Thanks,
>



-- 
-Rupa



From sjmudd at pobox.com  Tue Oct 13 13:51:14 2009
From: sjmudd at pobox.com (Simon J Mudd)
Date: Tue, 13 Oct 2009 22:51:14 +0200
Subject: [Freeswitch-users] Preannouncing a message before making a call
Message-ID: <20091013205114.GA4927@mad06.wl0.org>

I'm looking to migrate to Freeswitch from Asterisk (SOHO usage).
One thing I have configured at the moment which I'm unsure how to do in
FreeSwitch is the following.

When the call's dial pattern is recognised I've been getting Asterisk
to say: "Dialing <destination> via <gateway>" where <destination> might
be a country, or area and <gateway> may be the name of one of several
SIP gateways I have configured.  Each phrase has been a prerecorded wav
file played in sequence.

[macro-XXXXX]
exten => s,1,Playback(/etc/asterisk/wav/dialing)
exten => s,2,Playback(/etc/asterisk/wav/${ARG2})    # Name of destination
exten => s,3,Playback(/etc/asterisk/wav/via)
exten => s,4,Playback(/etc/asterisk/wav/XXXXX)      # Gateway name
exten => s,5,Dial(SIP/XXXXX/${ARG1},60,tT)          # ARG1 = Destination number


# actual usage
exten => _0044.,1,Macro(XXXXX,${EXTEN},touk)

How would I approach doing something like this in FreeSwtich?

Thanks for any pointers.

Simon



From sjmudd at pobox.com  Tue Oct 13 13:59:07 2009
From: sjmudd at pobox.com (Simon J Mudd)
Date: Tue, 13 Oct 2009 22:59:07 +0200
Subject: [Freeswitch-users] Building Freeswitch with VPATH (src and obj
	directories are different)?
Message-ID: <20091013205907.GA7545@mad06.wl0.org>

I was hoping to be able to build Freeswitch using VPATH, that is have the src and object directories in different locations.
This ensures that the src tree is unspoilt and I don't have to reclean it if I do updates.

However this does not seem to work:

[sjmudd at mad07 src]$ pwd
/Users/sjmudd/src
[sjmudd at mad07 src]$ ls -l | grep freeswitch
drwxr-xr-x   3 sjmudd  staff       102 13 Oct 07:52 freeswitch-obj
drwxr-xr-x  56 sjmudd  staff      1904 13 Oct 14:29 freeswitch-svn
[sjmudd at mad07 src]$ cd freeswitch-obj
[sjmudd at mad07 freeswitch-obj]$ ../freeswitch-svn/configure
checking for a BSD-compatible install... /usr/bin/install -c
checking whether build environment is sane... yes
checking for a thread-safe mkdir -p... ../freeswitch-svn/build/config/install-sh -c -d
checking for gawk... no
checking for mawk... no
checking for nawk... no
checking for awk... awk
checking whether make sets $(MAKE)... yes
configure: error: source directory already configured; run "make distclean" there first
[sjmudd at mad07 freeswitch-obj]$ cd ../freeswitch-svn
[sjmudd at mad07 freeswitch-svn]$ make distclean
Making clean in build
Making clean in src
Making clean in mod

making clean mod_amr

making clean mod_cdr_csv

making clean mod_cluechoo

making clean mod_commands

making clean mod_conference

making clean mod_console

making clean mod_dialplan_asterisk

making clean mod_dialplan_xml

making clean mod_dptools

making clean mod_enum

making clean mod_esf

making clean mod_event_socket

making clean mod_expr

making clean mod_fifo

making clean mod_file_string

making clean mod_fsv

making clean mod_g723_1

making clean mod_g729

making clean mod_h26x

making clean mod_iax

making clean mod_ilbc

making clean mod_limit

making clean mod_local_stream

making clean mod_logfile

making clean mod_loopback

making clean mod_lua

making clean mod_native_file

making clean mod_say_en

making clean mod_siren

making clean mod_sndfile

making clean mod_sofia
rm -f "./so_locations"

making clean mod_speex

making clean mod_spidermonkey

making clean mod_spidermonkey_core_db

making clean mod_spidermonkey_socket

making clean mod_spidermonkey_teletone

making clean mod_syslog

making clean mod_tone_stream

making clean mod_voicemail

making clean mod_voipcodecs

making clean mod_xml_cdr

making clean mod_xml_rpc

making clean mod_alsa

making clean mod_amrwb

making clean mod_celt

making clean mod_cepstral

making clean mod_dahdi_codec

making clean mod_dialplan_directory

making clean mod_dingaling

making clean mod_easyroute

making clean mod_erlang_event

making clean mod_event_multicast

making clean mod_fax

making clean mod_flite

WARNING mod_http is not a valid FreeSWITCH module dir, skipping it.....

making clean mod_lcr

making clean mod_ldap

making clean mod_memcache

making clean mod_nibblebill

making clean mod_opal
Package opal was not found in the pkg-config search path.
Perhaps you should add the directory containing `opal.pc'
to the PKG_CONFIG_PATH environment variable
No package 'opal' found
Package opal was not found in the pkg-config search path.
Perhaps you should add the directory containing `opal.pc'
to the PKG_CONFIG_PATH environment variable
No package 'opal' found

making clean mod_openzap
rm -f "./so_locations"
rm -f "./so_locations"
rm -f "./so_locations"
rm -f "./so_locations"
rm -f "./so_locations"
rm -f "./so_locations"
rm -f "./so_locations"
 rm -f testtones testtones
 rm -f detect_tones detect_tones
 rm -f detect_dtmf detect_dtmf
 rm -f testisdn testisdn
 rm -f testpri testpri
 rm -f testr2 testr2
 rm -f testboost testboost
 rm -f testanalog testanalog
 rm -f testapp testapp
 rm -f testcid testcid

making clean mod_perl

making clean mod_pocketsphinx

making clean mod_portaudio

making clean mod_python

making clean mod_radius_cdr

making clean mod_rss

making clean mod_say_de

making clean mod_say_es

making clean mod_say_fr

making clean mod_say_it

making clean mod_say_nl

making clean mod_say_ru

making clean mod_say_zh

making clean mod_shout

making clean mod_skypiax

making clean mod_snom

making clean mod_soundtouch

making clean mod_spidermonkey_odbc

making clean mod_spy

making clean mod_vmd

making clean mod_xml_curl

making clean mod_xml_ldap

making clean mod_yaml
Making clean in .
Making clean in .
 rm -f freeswitch freeswitch
 rm -f fs_cli fs_cli
 rm -f fs_ivrd fs_ivrd
rm -f "./so_locations"
[sjmudd at mad07 freeswitch-svn]$ cd ../freeswitch-obj
[sjmudd at mad07 freeswitch-obj]$ ../freeswitch-svn/configure
checking for a BSD-compatible install... /usr/bin/install -c
checking whether build environment is sane... yes
checking for a thread-safe mkdir -p... ../freeswitch-svn/build/config/install-sh -c -d
checking for gawk... no
checking for mawk... no
checking for nawk... no
checking for awk... awk
checking whether make sets $(MAKE)... yes
configure: error: source directory already configured; run "make distclean" there first
[sjmudd at mad07 freeswitch-obj]$ 

This is with 

[sjmudd at mad07 freeswitch-svn]$ svn info
Path: .
URL: http://svn.freeswitch.org/svn/freeswitch/trunk
Repository Root: http://svn.freeswitch.org/svn
Repository UUID: d0543943-73ff-0310-b7d9-9358b9ac24b2
Revision: 15138
Node Kind: directory
Schedule: normal
Last Changed Author: anthm
Last Changed Rev: 15138
Last Changed Date: 2009-10-12 16:58:09 +0200 (Mon, 12 Oct 2009)

Simon




From frank at carmickle.com  Tue Oct 13 16:32:20 2009
From: frank at carmickle.com (Frank Carmickle)
Date: Tue, 13 Oct 2009 19:32:20 -0400
Subject: [Freeswitch-users] Building Freeswitch with VPATH (src and
	obj	directories are different)?
In-Reply-To: <20091013205907.GA7545@mad06.wl0.org>
References: <20091013205907.GA7545@mad06.wl0.org>
Message-ID: <20091013233220.GM25698@base.carmickle.com>

On Tue, Oct 13, Simon J Mudd wrote:
> I was hoping to be able to build Freeswitch using VPATH, that is have the src and object directories in different locations.
> This ensures that the src tree is unspoilt and I don't have to reclean it if I do updates.

You can use svn export.  Not sure if this is what you want.

--FC



From msc at freeswitch.org  Tue Oct 13 17:09:46 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Oct 2009 17:09:46 -0700
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <BC5F7F95-767B-4A1D-95F5-FB393E31E6D3@ken-ton.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<87f2f3b90910080958i1f5cfccax621fd12100a05e3e@mail.gmail.com>
	<903da5680910081201p427a8358n768edb61a96674a4@mail.gmail.com>
	<87f2f3b90910081538l6e4e4b7dv3aedb2c4d1af58bc@mail.gmail.com>
	<903da5680910082238x27bff476s8bea0e74ab516a8d@mail.gmail.com>
	<87f2f3b90910082308i7f6144d9kc887596be54dd609@mail.gmail.com>
	<903da5680910082341r485092vb0cd10c43c4a2534@mail.gmail.com>
	<87f2f3b90910090005i51dbfc60t20dd77789872bc9e@mail.gmail.com>
	<BC5F7F95-767B-4A1D-95F5-FB393E31E6D3@ken-ton.com>
Message-ID: <87f2f3b90910131709v368ccee9o81cc2892a332f1a4@mail.gmail.com>

Karl,

Yeah, this post is wrong. You've got "acronym" misspelled. Once you fix that
then I think it's all good! ;)

Seriously, it's okay to put critical information into the wiki, but perhaps
you could remove the tongue-in-cheek comments.

-MC

On Tue, Oct 13, 2009 at 3:59 PM, Karl Vesterling <kjv at ken-ton.com> wrote:

>
> If that's the case, does this also apply to VoIP gateway services?
>
> If so, I'll consider phrasing this differently:
>
> http://wiki.freeswitch.org/wiki/Voicepulse.xml
> --begin--
> Voicepulse Warning*WARNING!!!* Recent changes (Sep 2008) in VoicePulse
> have limited connect03 to only IAX termination. Please remove any entries
> that you may have reflecting the gateway below from your dialplan as it will
> most likely answer the call and play a message saying something to the
> effect of "Your call could not be completed as dialed because we changed our
> production environment in an effort to demonstrate what bad engineering is
> all about. BAD is an acronymn for Broken As Designed. Please hang up and
> make changes on all your switches. Even though we shouldn't have answered
> the call and simply ignored it entirely so that another gateway could
> facilitate your call, we FELT it wise to seriously inconvenience you by
> answering the call immediately giving no other gateway a chance to place the
> call. Thanks for using our service and we look forward to abusing you again
> in the near future.
> *UPDATE* - Problem getting worse, not better *3/30/2009*. I switched DID
> providers because of problems w/ Voicepulse, such as echo, dropped calls,
> and consistent registration problems hence calls don't get placed and
> inbound don't get routed to me. Incidentally, they will not port DID's,
> they'll tell you they will, but they don't.
> --end--
>
> Perhaps just saying, "stay away, far far away. Recommend this service only
> to people and/or organizations you don't like or are in direct competition
> with" would suffice eh?
>
> Suggestions? Recommendations?
>
> Best Regards,
> Karl J. Vesterling
> kjv at ken-ton.com
> 202-461-3231 x0
>
> On Oct 9, 2009, at 3:05 AM, Michael Collins wrote:
>
>
>
> On Thu, Oct 8, 2009 at 11:41 PM, Gabriel Gunderson <gabe at gundy.org> wrote:
>
>> On Fri, Oct 9, 2009 at 12:08 AM, Michael Collins <msc at freeswitch.org>
>> wrote:
>> > Thanks for reporting back. Please let all the Asterisk users know that
>> they
>> > are welcome to join us in #freeswitch on irc.freenode.net and that they
>> will
>> > not be abused like people do in other less friendly IRC channels.
>>
>> Funny you mention this.  Many people report that the way the FS
>> community refers to Asterisk in docs/wikis/irc/whatever makes the FS
>> camp seem *less* welcoming to them.  After all, they identify as
>> Asterisk Users and take the criticism as being kinda harsh.  Most of
>> them acknowledge the shortcomings of Asterisk but are put off when
>> someone else points them out.  It's crazy, I know.  The thing is, I
>> remember thinking that too.  After getting to know FS better, I didn't
>> notice it as much.  Nobody likes to hear their baby is ugly --even if
>> they know it is.
>>
>
> Interesting. I'd like to know what specifically turned them off. It could
> be node 117, but that was written by Anthony in response to dozens of
> questions about the subject. In any case, if the Asterisk guys look at the
> first edition of the Starfish/TFOT book they'll see glowing praise of
> Anthony Minessale and Brian West, so maybe they'll be willing to listen to
> those guys since they've got so much Asterisk knowledge. The other thing is
> that many people in the FreeSWITCH community are Asterisk refugees. They
> left Asterisk for FreeSWITCH because they had some sort of trouble with
> Asterisk. If that comes across in the docs then we'll have to see about
> cleaning that up.
>
>>
>> At our session, and in general, I've noticed people are more
>> interested in hearing about FS when you don't make direct comparisons
>> to Asterisk.  Besides, FS stands on it's own merit.
>>
>
> Agreed. Direct comparisons can always be slanted in any direction. Besides,
> FreeSWITCH stacks up very well against a lot of modern (i.e. expensive)
> equipment put out by the big boys (Cisco, Avaya, NEC, ShoreTel, etc.).
>
>>
>> Just what I've observed *and* my 2 additional cents.
>>
>> Gabe
>>
> Thanks,
> MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Tue Oct 13 17:33:14 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 13 Oct 2009 17:33:14 -0700
Subject: [Freeswitch-users] Dingaling: using a hostname instead of stun
	for rtp
In-Reply-To: <33c87fa30910131502t2af24801i1c3319518b5db50b@mail.gmail.com>
References: <33c87fa30910131502t2af24801i1c3319518b5db50b@mail.gmail.com>
Message-ID: <C874E25D-6D73-49AA-8DD3-0A0022617FD2@freeswitch.org>

I don't think mod_dingaling will do a lookup for host: like sofia will  
as it doesn't have the code for that last I checked... I could be  
wrong but I don't recall it doing that.

/b

On Oct 13, 2009, at 3:02 PM, Mark Campbell-Smith wrote:

> I have a hostname set in vars.conf.xml for the parameters
> external_rtp_ip and the external_sip_ip instead of the usual stun.  I
> found that stun was timing out and was causing some problems.  And as
> I have a hostname, it makes sense to use that instead of relying on
> stun.




From brian at freeswitch.org  Tue Oct 13 17:36:16 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 13 Oct 2009 17:36:16 -0700
Subject: [Freeswitch-users] Building Freeswitch with VPATH (src and obj
	directories are different)?
In-Reply-To: <20091013205907.GA7545@mad06.wl0.org>
References: <20091013205907.GA7545@mad06.wl0.org>
Message-ID: <D8EB1A6C-38A6-4BC2-AD8C-46CF11993366@freeswitch.org>

You shouldn't have to make clean usually ... doing so might break your  
tree... You can usually get by with "make current" that will ensure  
the critical things are cleaned and built correctly... every now and  
they you'll hit a snafu but we'll usually tell you about it.

/b

On Oct 13, 2009, at 1:59 PM, Simon J Mudd wrote:

> I was hoping to be able to build Freeswitch using VPATH, that is  
> have the src and object directories in different locations.
> This ensures that the src tree is unspoilt and I don't have to  
> reclean it if I do updates.




From diego.viola at gmail.com  Tue Oct 13 18:36:04 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Wed, 14 Oct 2009 01:36:04 +0000
Subject: [Freeswitch-users] Some help with my post-paid billing project
In-Reply-To: <74730C11-75D0-4BC0-9F63-8B6C9F0F499F@me.com>
References: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
	<74730C11-75D0-4BC0-9F63-8B6C9F0F499F@me.com>
Message-ID: <86a32abc0910131836w5f113a16s247caf5eeb0f8fe4@mail.gmail.com>

I'm using MySQL now but I will try PostgreSQL with the prefix module, is
there a way to do that without the prefix module and with regular SQL?

Any examples?

Diego

On Tue, Oct 13, 2009 at 10:45 PM, Even Andr? Fiskvik <grevenx at me.com> wrote:

> What database are you using?
> You could do this with regular SQL, but it would by a costly operation,
> for PostgreSQL we're using the prefix module:
> http://pgfoundry.org/projects/prefix/
>
> You can then match the closest prefix by using something like
> "WHERE myprefix_col @> caller_destination_number ORDER BY LENGTH
> (myprefix_col::text) LIMIT 1;"
>
>
> Best regards,
> Even Andr?
>
>
> On 13. okt. 2009, at 23.53, Diego Viola wrote:
>
> > Hello,
> >
> > I'm trying to write a post-paid billing script, I have the CDR on my
> > database and also a "rates" table, the CDR contains fields like
> > caller_destination_number, variable_duration, etc. and the rates
> > table contains: destination, prefix, rate (cost).
> >
> > The problem is that I can't just strip the destination number to
> > take the prefix from it because I have to deal with destination
> > numbers from different countries and they all have different prefix
> > lengths... so I need to find another way to take the prefix from the
> > destination number.
> >
> > Any ideas how to do this?
> >
> > Thanks,
> >
> > Diego
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kjv at ken-ton.com  Tue Oct 13 18:38:47 2009
From: kjv at ken-ton.com (Karl Vesterling)
Date: Tue, 13 Oct 2009 21:38:47 -0400
Subject: [Freeswitch-users] FS Slide deck?
In-Reply-To: <87f2f3b90910131709v368ccee9o81cc2892a332f1a4@mail.gmail.com>
References: <903da5680910080742p1a722fa2yd96cf4c89533c1c5@mail.gmail.com>
	<87f2f3b90910080958i1f5cfccax621fd12100a05e3e@mail.gmail.com>
	<903da5680910081201p427a8358n768edb61a96674a4@mail.gmail.com>
	<87f2f3b90910081538l6e4e4b7dv3aedb2c4d1af58bc@mail.gmail.com>
	<903da5680910082238x27bff476s8bea0e74ab516a8d@mail.gmail.com>
	<87f2f3b90910082308i7f6144d9kc887596be54dd609@mail.gmail.com>
	<903da5680910082341r485092vb0cd10c43c4a2534@mail.gmail.com>
	<87f2f3b90910090005i51dbfc60t20dd77789872bc9e@mail.gmail.com>
	<BC5F7F95-767B-4A1D-95F5-FB393E31E6D3@ken-ton.com>
	<87f2f3b90910131709v368ccee9o81cc2892a332f1a4@mail.gmail.com>
Message-ID: <302A8ECD-A1C8-41FD-87D0-0DB7F61E5E76@ken-ton.com>

Michael;

Roger that.  I'll check my schedule to see when I might have some time  
available to make the change.
It seems as if I have some time available the tuesday after (bleep)  
freezes over.

Acronym corrected. ;-)

Best Regards,
Karl J. Vesterling
kjv at ken-ton.com
202-461-3231 x0

On Oct 13, 2009, at 8:09 PM, Michael Collins wrote:

> Karl,
>
> Yeah, this post is wrong. You've got "acronym" misspelled. Once you  
> fix that then I think it's all good! ;)
>
> Seriously, it's okay to put critical information into the wiki, but  
> perhaps you could remove the tongue-in-cheek comments.
>
> -MC
>
> On Tue, Oct 13, 2009 at 3:59 PM, Karl Vesterling <kjv at ken-ton.com>  
> wrote:
>
> If that's the case, does this also apply to VoIP gateway services?
>
> If so, I'll consider phrasing this differently:
>
> http://wiki.freeswitch.org/wiki/Voicepulse.xml
> --begin--
> Voicepulse Warning
>
> WARNING!!! Recent changes (Sep 2008) in VoicePulse have limited  
> connect03 to only IAX termination. Please remove any entries that  
> you may have reflecting the gateway below from your dialplan as it  
> will most likely answer the call and play a message saying something  
> to the effect of "Your call could not be completed as dialed because  
> we changed our production environment in an effort to demonstrate  
> what bad engineering is all about. BAD is an acronymn for Broken As  
> Designed. Please hang up and make changes on all your switches. Even  
> though we shouldn't have answered the call and simply ignored it  
> entirely so that another gateway could facilitate your call, we FELT  
> it wise to seriously inconvenience you by answering the call  
> immediately giving no other gateway a chance to place the call.  
> Thanks for using our service and we look forward to abusing you  
> again in the near future.
> UPDATE - Problem getting worse, not better 3/30/2009. I switched DID  
> providers because of problems w/ Voicepulse, such as echo, dropped  
> calls, and consistent registration problems hence calls don't get  
> placed and inbound don't get routed to me. Incidentally, they will  
> not port DID's, they'll tell you they will, but they don't.
> --end--
>
> Perhaps just saying, "stay away, far far away. Recommend this  
> service only to people and/or organizations you don't like or are in  
> direct competition with" would suffice eh?
>
> Suggestions? Recommendations?
>
> Best Regards,
> Karl J. Vesterling
> kjv at ken-ton.com
> 202-461-3231 x0
>
> On Oct 9, 2009, at 3:05 AM, Michael Collins wrote:
>
>>
>>
>> On Thu, Oct 8, 2009 at 11:41 PM, Gabriel Gunderson <gabe at gundy.org>  
>> wrote:
>> On Fri, Oct 9, 2009 at 12:08 AM, Michael Collins  
>> <msc at freeswitch.org> wrote:
>> > Thanks for reporting back. Please let all the Asterisk users know  
>> that they
>> > are welcome to join us in #freeswitch on irc.freenode.net and  
>> that they will
>> > not be abused like people do in other less friendly IRC channels.
>>
>> Funny you mention this.  Many people report that the way the FS
>> community refers to Asterisk in docs/wikis/irc/whatever makes the FS
>> camp seem *less* welcoming to them.  After all, they identify as
>> Asterisk Users and take the criticism as being kinda harsh.  Most of
>> them acknowledge the shortcomings of Asterisk but are put off when
>> someone else points them out.  It's crazy, I know.  The thing is, I
>> remember thinking that too.  After getting to know FS better, I  
>> didn't
>> notice it as much.  Nobody likes to hear their baby is ugly --even if
>> they know it is.
>>
>> Interesting. I'd like to know what specifically turned them off. It  
>> could be node 117, but that was written by Anthony in response to  
>> dozens of questions about the subject. In any case, if the Asterisk  
>> guys look at the first edition of the Starfish/TFOT book they'll  
>> see glowing praise of Anthony Minessale and Brian West, so maybe  
>> they'll be willing to listen to those guys since they've got so  
>> much Asterisk knowledge. The other thing is that many people in the  
>> FreeSWITCH community are Asterisk refugees. They left Asterisk for  
>> FreeSWITCH because they had some sort of trouble with Asterisk. If  
>> that comes across in the docs then we'll have to see about cleaning  
>> that up.
>>
>> At our session, and in general, I've noticed people are more
>> interested in hearing about FS when you don't make direct comparisons
>> to Asterisk.  Besides, FS stands on it's own merit.
>>
>> Agreed. Direct comparisons can always be slanted in any direction.  
>> Besides, FreeSWITCH stacks up very well against a lot of modern  
>> (i.e. expensive) equipment put out by the big boys (Cisco, Avaya,  
>> NEC, ShoreTel, etc.).
>>
>> Just what I've observed *and* my 2 additional cents.
>>
>> Gabe
>> Thanks,
>> MC
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From diego.viola at gmail.com  Tue Oct 13 18:53:54 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Wed, 14 Oct 2009 01:53:54 +0000
Subject: [Freeswitch-users] Some help with my post-paid billing project
In-Reply-To: <86a32abc0910131836w5f113a16s247caf5eeb0f8fe4@mail.gmail.com>
References: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
	<74730C11-75D0-4BC0-9F63-8B6C9F0F499F@me.com>
	<86a32abc0910131836w5f113a16s247caf5eeb0f8fe4@mail.gmail.com>
Message-ID: <86a32abc0910131853v244ed61bof9c24f345a0944b2@mail.gmail.com>

Wrong question.

Is there a way to compare numbers with prefixes without using the prefix
module?

Diego

On Wed, Oct 14, 2009 at 1:36 AM, Diego Viola <diego.viola at gmail.com> wrote:

> I'm using MySQL now but I will try PostgreSQL with the prefix module, is
> there a way to do that without the prefix module and with regular SQL?
>
> Any examples?
>
> Diego
>
>
> On Tue, Oct 13, 2009 at 10:45 PM, Even Andr? Fiskvik <grevenx at me.com>wrote:
>
>> What database are you using?
>> You could do this with regular SQL, but it would by a costly operation,
>> for PostgreSQL we're using the prefix module:
>> http://pgfoundry.org/projects/prefix/
>>
>> You can then match the closest prefix by using something like
>> "WHERE myprefix_col @> caller_destination_number ORDER BY LENGTH
>> (myprefix_col::text) LIMIT 1;"
>>
>>
>> Best regards,
>> Even Andr?
>>
>>
>> On 13. okt. 2009, at 23.53, Diego Viola wrote:
>>
>> > Hello,
>> >
>> > I'm trying to write a post-paid billing script, I have the CDR on my
>> > database and also a "rates" table, the CDR contains fields like
>> > caller_destination_number, variable_duration, etc. and the rates
>> > table contains: destination, prefix, rate (cost).
>> >
>> > The problem is that I can't just strip the destination number to
>> > take the prefix from it because I have to deal with destination
>> > numbers from different countries and they all have different prefix
>> > lengths... so I need to find another way to take the prefix from the
>> > destination number.
>> >
>> > Any ideas how to do this?
>> >
>> > Thanks,
>> >
>> > Diego
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> > users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From technical at ttnc.co.uk  Tue Oct 13 19:07:11 2009
From: technical at ttnc.co.uk (TTNC - Adnan Barakat)
Date: Wed, 14 Oct 2009 03:07:11 +0100
Subject: [Freeswitch-users] Some help with my post-paid billing project
In-Reply-To: <86a32abc0910131836w5f113a16s247caf5eeb0f8fe4@mail.gmail.com>
References: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>	<74730C11-75D0-4BC0-9F63-8B6C9F0F499F@me.com>
	<86a32abc0910131836w5f113a16s247caf5eeb0f8fe4@mail.gmail.com>
Message-ID: <4AD5324F.2030000@ttnc.co.uk>

Diego Viola wrote:
> I'm using MySQL now but I will try PostgreSQL with the prefix module, is
> there a way to do that without the prefix module and with regular SQL?
> 
> Any examples?
SELECT * FROM rates WHERE prefix = SUBSTRING('$NUMBER$', 1,
LENGTH(prefix)) LIMIT 1

Adnan

> Diego
> 
> On Tue, Oct 13, 2009 at 10:45 PM, Even Andr? Fiskvik <grevenx at me.com
> <mailto:grevenx at me.com>> wrote:
> 
>     What database are you using?
>     You could do this with regular SQL, but it would by a costly operation,
>     for PostgreSQL we're using the prefix module:
>     http://pgfoundry.org/projects/prefix/
> 
>     You can then match the closest prefix by using something like
>     "WHERE myprefix_col @> caller_destination_number ORDER BY LENGTH
>     (myprefix_col::text) LIMIT 1;"
> 
> 
>     Best regards,
>     Even Andr?
> 
> 
>     On 13. okt. 2009, at 23.53, Diego Viola wrote:
> 
>     > Hello,
>     >
>     > I'm trying to write a post-paid billing script, I have the CDR on my
>     > database and also a "rates" table, the CDR contains fields like
>     > caller_destination_number, variable_duration, etc. and the rates
>     > table contains: destination, prefix, rate (cost).
>     >
>     > The problem is that I can't just strip the destination number to
>     > take the prefix from it because I have to deal with destination
>     > numbers from different countries and they all have different prefix
>     > lengths... so I need to find another way to take the prefix from the
>     > destination number.
>     >
>     > Any ideas how to do this?
>     >
>     > Thanks,
>     >
>     > Diego
>     >
>     > _______________________________________________
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>     > users
>     > http://www.freeswitch.org
> 
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From dome at tel.co.th  Tue Oct 13 19:39:13 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Wed, 14 Oct 2009 09:39:13 +0700
Subject: [Freeswitch-users] T.38 in sip header
Message-ID: <8ccbff060910131939g75b12346x2e9156fe17b61aa6@mail.gmail.com>

Dear all,

             Is posible to check T.38 in sip header? i want to
disconnect call when T.38 not support in gateway and try next gateway

BG

Dome C.



From red.rain.seven at gmail.com  Tue Oct 13 20:30:27 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Wed, 14 Oct 2009 11:30:27 +0800
Subject: [Freeswitch-users] sched_api doesn't get launched
In-Reply-To: <86a32abc0910131145t730d0bc0m842ed7ed0891d425@mail.gmail.com>
References: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
	<86a32abc0910131145t730d0bc0m842ed7ed0891d425@mail.gmail.com>
Message-ID: <59ad9ca10910132030l1fb55edbvb9fc4107dd15f61c@mail.gmail.com>

Diego:
You probably miss understood me. I said I was able to make "sched_hangup"
work, but not the "sched_api" in the same way I script for "sched_hangup"

The problem was on the second paragraph.

thanks,

On Wed, Oct 14, 2009 at 2:45 AM, Diego Viola <diego.viola at gmail.com> wrote:

> You need to pass the UUID to sched_hangup.
>
> Usage: sched_hangup [+]<time> <uuid> [<cause>]
>
> http://wiki.freeswitch.org/wiki/Mod_commands#sched_hangup
>
>
> On Tue, Oct 13, 2009 at 10:14 AM, Henry Huang <b_ball_henry at hotmail.com>wrote:
>
>> Hi:
>> I am using mod_java. And in my script I was able to achieve using:
>>
>> execute("sched_hangup", "+300 alloted_timeout");
>>
>> However, when I try to run sched_api in the same way, system log returns
>> that it's an invalid application. I have also tried to trigger it with many
>> conditional channel variable api calls , but non of them seemed to execute
>> the api command (because I turned on the highest level of debugging and see
>> no where the sched_api is being called.
>>
>> The closest thing I got was by using "api_after_bridge" like the
>> following, but it only launches when the bridge is teared down(which is not
>> what I want). I originally thought after bridge means right after the 2
>> party is connected. All I want is to be able to play some message to leg A
>> at certain time.
>>
>> setVariable("api_after_bridge", "sched_api +10 none uuid_displace ${uuid}
>> start /path/to/some.wav 20 mux");
>>
>> I have been struggling with different combination for a week now.. Please
>> shed some light if you know something.
>>
>> Thanks,
>>
>> --
>> Henry Huang
>> UniC Solution - Communication Unified
>> VoIP & Open Source software Consultant
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From red.rain.seven at gmail.com  Tue Oct 13 20:33:23 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Wed, 14 Oct 2009 11:33:23 +0800
Subject: [Freeswitch-users] Some help with my post-paid billing project
In-Reply-To: <4AD5324F.2030000@ttnc.co.uk>
References: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
	<74730C11-75D0-4BC0-9F63-8B6C9F0F499F@me.com>
	<86a32abc0910131836w5f113a16s247caf5eeb0f8fe4@mail.gmail.com>
	<4AD5324F.2030000@ttnc.co.uk>
Message-ID: <59ad9ca10910132033k28fb7f8ckdd5ba1f772ba2a7e@mail.gmail.com>

Diego:
Didn't wrote a prepaid system already? how did you compare the prefix in
that system then?

On Wed, Oct 14, 2009 at 10:07 AM, TTNC - Adnan Barakat <technical at ttnc.co.uk
> wrote:

> Diego Viola wrote:
> > I'm using MySQL now but I will try PostgreSQL with the prefix module, is
> > there a way to do that without the prefix module and with regular SQL?
> >
> > Any examples?
> SELECT * FROM rates WHERE prefix = SUBSTRING('$NUMBER$', 1,
> LENGTH(prefix)) LIMIT 1
>
> Adnan
>
> > Diego
> >
> > On Tue, Oct 13, 2009 at 10:45 PM, Even Andr? Fiskvik <grevenx at me.com
> > <mailto:grevenx at me.com>> wrote:
> >
> >     What database are you using?
> >     You could do this with regular SQL, but it would by a costly
> operation,
> >     for PostgreSQL we're using the prefix module:
> >     http://pgfoundry.org/projects/prefix/
> >
> >     You can then match the closest prefix by using something like
> >     "WHERE myprefix_col @> caller_destination_number ORDER BY LENGTH
> >     (myprefix_col::text) LIMIT 1;"
> >
> >
> >     Best regards,
> >     Even Andr?
> >
> >
> >     On 13. okt. 2009, at 23.53, Diego Viola wrote:
> >
> >     > Hello,
> >     >
> >     > I'm trying to write a post-paid billing script, I have the CDR on
> my
> >     > database and also a "rates" table, the CDR contains fields like
> >     > caller_destination_number, variable_duration, etc. and the rates
> >     > table contains: destination, prefix, rate (cost).
> >     >
> >     > The problem is that I can't just strip the destination number to
> >     > take the prefix from it because I have to deal with destination
> >     > numbers from different countries and they all have different prefix
> >     > lengths... so I need to find another way to take the prefix from
> the
> >     > destination number.
> >     >
> >     > Any ideas how to do this?
> >     >
> >     > Thanks,
> >     >
> >     > Diego
> >     >
> >     > _______________________________________________
> >     > FreeSWITCH-users mailing list
> >     > FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-
> >     > users
> >     > http://www.freeswitch.org
> >
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From msc at freeswitch.org  Tue Oct 13 21:16:07 2009
From: msc at freeswitch.org (Michael S Collins)
Date: Tue, 13 Oct 2009 21:16:07 -0700
Subject: [Freeswitch-users] sched_api doesn't get launched
In-Reply-To: <59ad9ca10910132030l1fb55edbvb9fc4107dd15f61c@mail.gmail.com>
References: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
	<86a32abc0910131145t730d0bc0m842ed7ed0891d425@mail.gmail.com>
	<59ad9ca10910132030l1fb55edbvb9fc4107dd15f61c@mail.gmail.com>
Message-ID: <31EC1285-3EF7-4B9F-A39D-CA3E8DB34A46@freeswitch.org>


On Oct 13, 2009, at 8:30 PM, Henry Huang <red.rain.seven at gmail.com>  
wrote:

> Diego:
>
> You probably miss understood me. I said I was able to make  
> "sched_hangup" work, but not the "sched_api" in the same way I  
> script for "sched_hangup"
>
> The problem was on the second paragraph.

Henry, can you capture the debug output of a call and put it in  
pastebin?That should yield a clue as to what is going on.

-MC



From brian at freeswitch.org  Tue Oct 13 21:15:25 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 13 Oct 2009 21:15:25 -0700
Subject: [Freeswitch-users] T.38 in sip header
In-Reply-To: <8ccbff060910131939g75b12346x2e9156fe17b61aa6@mail.gmail.com>
References: <8ccbff060910131939g75b12346x2e9156fe17b61aa6@mail.gmail.com>
Message-ID: <75A104CA-BE97-4912-BA7D-F091A6759EFC@freeswitch.org>

Well if you look at the SDP you can look for "image" usually but thats  
not something I think you can see PRE call setup.

/b

On Oct 13, 2009, at 7:39 PM, Dome Charoenyost wrote:

> Dear all,
>
>             Is posible to check T.38 in sip header? i want to
> disconnect call when T.38 not support in gateway and try next gateway
>
> BG
>
> Dome C.




From mgg at giagnocavo.net  Tue Oct 13 21:15:22 2009
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Wed, 14 Oct 2009 00:15:22 -0400
Subject: [Freeswitch-users] Some help with my post-paid billing project
In-Reply-To: <86a32abc0910131853v244ed61bof9c24f345a0944b2@mail.gmail.com>
References: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
	<74730C11-75D0-4BC0-9F63-8B6C9F0F499F@me.com>
	<86a32abc0910131836w5f113a16s247caf5eeb0f8fe4@mail.gmail.com>
	<86a32abc0910131853v244ed61bof9c24f345a0944b2@mail.gmail.com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C6703241D1E52@mse17be1.mse17.exchange.ms>

In our testing with SQL Server, we found that executing several queries for direct matches yielded far better performance than one query trying to check prefixes. (The column was also part of the clustered index, but AFAIK MySQL doesn't support defining your own clustered indexes; you get the PK always.)

-Michael

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Diego Viola
Sent: Tuesday, October 13, 2009 7:54 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Some help with my post-paid billing project

Wrong question.

Is there a way to compare numbers with prefixes without using the prefix module?

Diego
On Wed, Oct 14, 2009 at 1:36 AM, Diego Viola <diego.viola at gmail.com<mailto:diego.viola at gmail.com>> wrote:
I'm using MySQL now but I will try PostgreSQL with the prefix module, is there a way to do that without the prefix module and with regular SQL?

Any examples?

Diego

On Tue, Oct 13, 2009 at 10:45 PM, Even Andr? Fiskvik <grevenx at me.com<mailto:grevenx at me.com>> wrote:
What database are you using?
You could do this with regular SQL, but it would by a costly operation,
for PostgreSQL we're using the prefix module: http://pgfoundry.org/projects/prefix/

You can then match the closest prefix by using something like
"WHERE myprefix_col @> caller_destination_number ORDER BY LENGTH
(myprefix_col::text) LIMIT 1;"


Best regards,
Even Andr?


On 13. okt. 2009, at 23.53, Diego Viola wrote:

> Hello,
>
> I'm trying to write a post-paid billing script, I have the CDR on my
> database and also a "rates" table, the CDR contains fields like
> caller_destination_number, variable_duration, etc. and the rates
> table contains: destination, prefix, rate (cost).
>
> The problem is that I can't just strip the destination number to
> take the prefix from it because I have to deal with destination
> numbers from different countries and they all have different prefix
> lengths... so I need to find another way to take the prefix from the
> destination number.
>
> Any ideas how to do this?
>
> Thanks,
>
> Diego
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> users
> http://www.freeswitch.org


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


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From lakindia89 at gmail.com  Tue Oct 13 21:29:30 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Wed, 14 Oct 2009 09:59:30 +0530
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <5E7A8451-6264-4822-8152-1B94240BDD48@freeswitch.org>
References: <7d79b3930910090407m5dc81d12pf2b0a47f6017804@mail.gmail.com>
	<20091009123615.BC50A3FA6E9@mail.cune.org>
	<7d79b3930910100647u7fd11413ucec0a535d68b3e3a@mail.gmail.com>
	<87f2f3b90910101009j1295d812jd98cae91e30b1eff@mail.gmail.com>
	<7d79b3930910120454m1ffc6dd4u42b91a4349597166@mail.gmail.com>
	<87f2f3b90910121149x6e2114f0q996a262ad272a1f6@mail.gmail.com>
	<7d79b3930910130645y658ae041v8219c4c881fa5698@mail.gmail.com>
	<5E7A8451-6264-4822-8152-1B94240BDD48@freeswitch.org>
Message-ID: <7d79b3930910132129t30e25479j12cb596b2ce0a57d@mail.gmail.com>

But that setup is not there right now. It was an year back they used that.
I'll try to make those setup and send you the log. I think it may take some
2 days time, and I'm not very familiar with asterisk.

In the mean time, I request you to kindly go-through freeswitch debug log.
You may still get some clue.


On Tue, Oct 13, 2009 at 9:16 PM, Michael S Collins <msc at freeswitch.org>wrote:

>
> On Oct 13, 2009, at 6:45 AM, lakshmanan ganapathy <lakindia89 at gmail.com>
> wrote:
>
> We are using Reliance as the Carrier.
> I think, with this same Reliance carrier, in my office, they are able to
> make outgoing calls through asterisk+libpri.
>
>
> If that's the case I would be very interested in seeing a pri debug from a
> working call on an asterisk box. It might give us a clue as to what is not
> working. -MC
>
>
>
> On Tue, Oct 13, 2009 at 12:19 AM, Michael Collins < <msc at freeswitch.org>
> msc at freeswitch.org> wrote:
>
>> Lak,
>>
>> Okay I will need a little bit of time to dig into the IE's and what they
>> contain. In the meantime can you tell me who the carrier is? I'd like to
>> find out if they have some specific requirements. The fact that it doesn't
>> work with libpri surprises me because that would mean that Asterisk systems
>> would probably not work with this carrier as well.
>>
>> BTW, thanks for the very complete pastebin entries. :)
>> -MC
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lakindia89 at gmail.com  Tue Oct 13 22:34:49 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Wed, 14 Oct 2009 11:04:49 +0530
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <5E7A8451-6264-4822-8152-1B94240BDD48@freeswitch.org>
References: <7d79b3930910090407m5dc81d12pf2b0a47f6017804@mail.gmail.com>
	<20091009123615.BC50A3FA6E9@mail.cune.org>
	<7d79b3930910100647u7fd11413ucec0a535d68b3e3a@mail.gmail.com>
	<87f2f3b90910101009j1295d812jd98cae91e30b1eff@mail.gmail.com>
	<7d79b3930910120454m1ffc6dd4u42b91a4349597166@mail.gmail.com>
	<87f2f3b90910121149x6e2114f0q996a262ad272a1f6@mail.gmail.com>
	<7d79b3930910130645y658ae041v8219c4c881fa5698@mail.gmail.com>
	<5E7A8451-6264-4822-8152-1B94240BDD48@freeswitch.org>
Message-ID: <7d79b3930910132234k4b99c794q8854ce20d14ee74e@mail.gmail.com>

Hi. I made the setup quickly with one of my colleague, and here is the debug
log given by asterisk

Steps done by me :
	+ Call Made from 04443902743 to 04439114600
	+ When call lands to 04439114600, I just route it to 09176454982.

*CLI>
*CLI>
*CLI> pri debug span 1
Enabled debugging on span 1
*CLI>
*CLI>
*CLI>
*CLI> < Protocol Discriminator: Q.931 (8)  len=41
< Call Ref: len= 2 (reference 21/0x15) (Originator)
< Message type: SETUP (5)
< [a1]
< Sending Complete (len= 1)
< [04 03 80 90 a3]
< Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer
capability: Speech (0)
<                              Ext: 1  Trans mode/rate: 64kbps,
circuit-mode (16)
<                                User information layer 1: A-Law (35)
< [18 03 a1 83 9f]
< Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0
Preferred  Dchan: 0
<                        ChanSel: As indicated in following octets
<                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
<                       Ext: 1  Channel: 31 ]
< [6c 0c 01 81 34 34 34 33 39 30 32 37 34 33]
< Calling Number (len=14) [ Ext: 0  TON: Unknown Number Type (0)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
<                           Presentation: Presentation permitted, user
number passed network screening (1)  '4443902743' ]
< [70 09 c1 33 39 31 31 34 36 30 30]
< Called Number (len=11) [ Ext: 1  TON: Subscriber Number (4)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)  '39114600' ]
-- Making new call for cr 21
-- Processing Q.931 Call Setup
-- Processing IE 161 (cs0, Sending Complete)
-- Processing IE 4 (cs0, Bearer Capability)
-- Processing IE 24 (cs0, Channel Identification)
-- Processing IE 108 (cs0, Calling Party Number)
-- Processing IE 112 (cs0, Called Party Number)
q931.c:3504 q931_receive: call 21 on channel 31 enters state 6 (Call Present)
q931.c:2769 q931_call_proceeding: call 21 on channel 31 enters state 9
(Incoming Call Proceeding)

> Protocol Discriminator: Q.931 (8)  len=10
> Call Ref: len= 2 (reference 21/0x15) (Terminator)
> Message type: CALL PROCEEDING (2)
> [18 03 a9 83 9f]
> Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  Dchan: 0
>                        ChanSel: As indicated in following octets
>                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
>                       Ext: 1  Channel: 31 ]

    -- Accepting call from '4443902743' to '39114600' on channel 0/31, span 1
    -- Executing [39114600 at incoming-from-pri:1] NoOp("Zap/31-1", "FMS
Call Extn:39114600:| CID:: <cid::>| Channel:Zap/31-1:| ") in new stack
    -- Executing [39114600 at incoming-from-pri:2] Dial("Zap/31-1",
"Zap/g1/w09176454982||rj") in new stack
-- Making new call for cr 32770
    -- Requested transfer capability: 0x00 - SPEECH

> Protocol Discriminator: Q.931 (8)  len=43
> Call Ref: len= 2 (reference 2/0x2) (Originator)
> Message type: SETUP (5)
> [04 03 80 90 a3]
> Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer capability: Speech (0)
>                              Ext: 1  Trans mode/rate: 64kbps, circuit-mode (16)
>                                User information layer 1: A-Law (35)
> [18 03 a9 83 81]
> Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  Dchan: 0
>                        ChanSel: As indicated in following octets
>                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
>                       Ext: 1  Channel: 1 ]
> [6c 0c 00 81 34 34 34 33 39 30 32 37 34 33]
> Calling Number (len=14) [ Ext: 0  TON: Unknown Number Type (0)  NPI: Unknown Number Plan (0)
>                           Presentation: Presentation permitted, user number passed network screening (1)  '4443902743' ]
> [70 0c 80 30 39 31 37 36 34 35 34 39 38 32]
> Called Number (len=14) [ Ext: 1  TON: Unknown Number Type (0)  NPI: Unknown Number Plan (0)  '09176454982' ]

q931.c:3087 q931_setup: call 32770 on channel 1 enters state 1 (Call Initiated)
    -- Called g1/w09176454982
q931.c:2797 q931_alerting: call 21 on channel 31 enters state 7 (Call Received)

> Protocol Discriminator: Q.931 (8)  len=9
> Call Ref: len= 2 (reference 21/0x15) (Terminator)
> Message type: ALERTING (1)
> [1e 02 81 88]
> Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Private network serving the local user (1)
>                               Ext: 1  Progress Description: Inband information or appropriate pattern now available. (8) ]

< Protocol Discriminator: Q.931 (8)  len=10
< Call Ref: len= 2 (reference 2/0x2) (Terminator)
< Message type: CALL PROCEEDING (2)
< [18 03 a9 83 81]
< Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0
Exclusive  Dchan: 0
<                        ChanSel: As indicated in following octets
<                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
<                       Ext: 1  Channel: 1 ]
-- Processing IE 24 (cs0, Channel Identification)
q931.c:3636 q931_receive: call 32770 on channel 1 enters state 3
(Outgoing call  Proceeding)
    -- Zap/1-1 is proceeding passing it to Zap/31-1
< Protocol Discriminator: Q.931 (8)  len=13
< Call Ref: len= 2 (reference 2/0x2) (Terminator)
< Message type: ALERTING (1)
< [1e 02 84 88]
< Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard
(0)  0: 0  Location: Public network serving the remote user (4)
<                               Ext: 1  Progress Description: Inband
information or appropriate pattern now available. (8) ]
< [1e 02 84 82]
< Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard
(0)  0: 0  Location: Public network serving the remote user (4)
<                               Ext: 1  Progress Description: Called
equipment is non-ISDN. (2) ]
-- Processing IE 30 (cs0, Progress Indicator)
-- Processing IE 30 (cs0, Progress Indicator)
q931.c:3549 q931_receive: call 32770 on channel 1 enters state 4 (Call
Delivered)
    -- Zap/1-1 is ringing
< Protocol Discriminator: Q.931 (8)  len=9
< Call Ref: len= 2 (reference 2/0x2) (Terminator)
< Message type: PROGRESS (3)
< [1e 02 84 84]
< Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard
(0)  0: 0  Location: Public network serving the remote user (4)
<                               Ext: 1  Progress Description: Unknown (4) ]
-- Processing IE 30 (cs0, Progress Indicator)
    -- Zap/1-1 is making progress passing it to Zap/31-1
< Protocol Discriminator: Q.931 (8)  len=9
< Call Ref: len= 2 (reference 2/0x2) (Terminator)
< Message type: PROGRESS (3)
< [1e 02 84 88]
< Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard
(0)  0: 0  Location: Public network serving the remote user (4)
<                               Ext: 1  Progress Description: Inband
information or appropriate pattern now available. (8) ]
-- Processing IE 30 (cs0, Progress Indicator)
    -- Zap/1-1 is making progress passing it to Zap/31-1
< Protocol Discriminator: Q.931 (8)  len=19
< Call Ref: len= 2 (reference 2/0x2) (Terminator)
< Message type: CONNECT (7)
< [4c 0c 01 83 39 31 37 36 34 35 34 39 38 32]
< Connected Number (len=14) [ Ext: 0  TON: Unknown Number Type (0)
NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
<                             Ext: 1  Presentation: Presentation
allowed of network provided number (3)  '9176454982' ]
-- Processing IE 76 (cs0, Connected Number)
q931.c:3579 q931_receive: call 32770 on channel 1 enters state 10 (Active)

> Protocol Discriminator: Q.931 (8)  len=5
> Call Ref: len= 2 (reference 2/0x2) (Originator)
> Message type: CONNECT ACKNOWLEDGE (15)

    -- Zap/1-1 answered Zap/31-1
q931.c:2904 q931_connect: call 21 on channel 31 enters state 8 (Connect Request)

> Protocol Discriminator: Q.931 (8)  len=14
> Call Ref: len= 2 (reference 21/0x15) (Terminator)
> Message type: CONNECT (7)
> [18 03 a9 83 9f]
> Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  Dchan: 0
>                        ChanSel: As indicated in following octets
>                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
>                       Ext: 1  Channel: 31 ]
> [1e 02 81 82]
> Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Private network serving the local user (1)
>                               Ext: 1  Progress Description: Called equipment is non-ISDN. (2) ]

    -- Native bridging Zap/31-1 and Zap/1-1
< Protocol Discriminator: Q.931 (8)  len=5
< Call Ref: len= 2 (reference 21/0x15) (Originator)
< Message type: CONNECT ACKNOWLEDGE (15)
q931.c:3664 q931_receive: call 21 on channel 31 enters state 10 (Active)
< Protocol Discriminator: Q.931 (8)  len=9
< Call Ref: len= 2 (reference 2/0x2) (Terminator)
< Message type: DISCONNECT (69)
< [08 02 80 90]
< Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0
Location: User (0)
<                  Ext: 1  Cause: Normal Clearing (16), class = Normal
Event (1) ]
-- Processing IE 8 (cs0, Cause)
q931.c:3779 q931_receive: call 32770 on channel 1 enters state 12
(Disconnect Indication)
    -- Channel 0/1, span 1 got hangup request, cause 16
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication,
peerstate Disconnect Request
q931.c:2920 q931_release: call 32770 on channel 1 enters state 19
(Release Request)

> Protocol Discriminator: Q.931 (8)  len=9
> Call Ref: len= 2 (reference 2/0x2) (Originator)
> Message type: RELEASE (77)
> [08 02 81 90]
> Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  Location: Private network serving the local user (1)
>                  Ext: 1  Cause: Normal Clearing (16), class = Normal Event (1) ]

    -- Hungup 'Zap/1-1'
  == Spawn extension (incoming-from-pri, 39114600, 2) exited non-zero
on 'Zap/31-1'
    -- Executing [h at incoming-from-pri:1] Hangup("Zap/31-1", "") in new stack
  == Spawn extension (incoming-from-pri, h, 1) exited non-zero on 'Zap/31-1'
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Active
q931.c:2968 q931_disconnect: call 21 on channel 31 enters state 11
(Disconnect Request)

> Protocol Discriminator: Q.931 (8)  len=9
> Call Ref: len= 2 (reference 21/0x15) (Terminator)
> Message type: DISCONNECT (69)
> [08 02 81 90]
> Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  Location: Private network serving the local user (1)
>                  Ext: 1  Cause: Normal Clearing (16), class = Normal Event (1) ]

    -- Hungup 'Zap/31-1'
< Protocol Discriminator: Q.931 (8)  len=5
< Call Ref: len= 2 (reference 2/0x2) (Terminator)
< Message type: RELEASE COMPLETE (90)
q931.c:3719 q931_receive: call 32770 on channel 1 enters state 0 (Null)
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
< Protocol Discriminator: Q.931 (8)  len=5
< Call Ref: len= 2 (reference 21/0x15) (Originator)
< Message type: RELEASE (77)
q931.c:3754 q931_receive: call 21 on channel 31 enters state 0 (Null)
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request

> Protocol Discriminator: Q.931 (8)  len=9
> Call Ref: len= 2 (reference 21/0x15) (Terminator)
> Message type: RELEASE COMPLETE (90)
> [08 02 81 90]
> Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  Location: Private network serving the local user (1)
>                  Ext: 1  Cause: Normal Clearing (16), class = Normal Event (1) ]

NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null


On Tue, Oct 13, 2009 at 9:16 PM, Michael S Collins <msc at freeswitch.org>wrote:

>
> On Oct 13, 2009, at 6:45 AM, lakshmanan ganapathy <lakindia89 at gmail.com>
> wrote:
>
> We are using Reliance as the Carrier.
> I think, with this same Reliance carrier, in my office, they are able to
> make outgoing calls through asterisk+libpri.
>
>
> If that's the case I would be very interested in seeing a pri debug from a
> working call on an asterisk box. It might give us a clue as to what is not
> working. -MC
>
>
>
> On Tue, Oct 13, 2009 at 12:19 AM, Michael Collins < <msc at freeswitch.org>
> msc at freeswitch.org> wrote:
>
>> Lak,
>>
>> Okay I will need a little bit of time to dig into the IE's and what they
>> contain. In the meantime can you tell me who the carrier is? I'd like to
>> find out if they have some specific requirements. The fact that it doesn't
>> work with libpri surprises me because that would mean that Asterisk systems
>> would probably not work with this carrier as well.
>>
>> BTW, thanks for the very complete pastebin entries. :)
>> -MC
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From velu.technical at gmail.com  Tue Oct 13 22:50:39 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Wed, 14 Oct 2009 11:20:39 +0530
Subject: [Freeswitch-users] Perl event socket library problem
Message-ID: <1452e2980910132250w293596d6jce5515d787b4fa2a@mail.gmail.com>

Dear All,
       I am implementing an IVR framework using Perl event socket libraries.
At first I set socket mode async full. I have faced problem that the Perl
statements executing before dailplan execution. So, I couldn't control my
process.

       Next I have tried without async mode, it solved that race condition
problem. I had another requirement that while executing external application
I need to play some voice file some time limit. When I played that voice
file in playback, the playback application blocked. So, I can't break that
playback application after some time limit.

       I am hanging on this problem for past one week. Please any one help
me to solve above problems...

Thanks,
Velusamy.
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From msc at freeswitch.org  Tue Oct 13 23:03:06 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Oct 2009 23:03:06 -0700
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <7d79b3930910132234k4b99c794q8854ce20d14ee74e@mail.gmail.com>
References: <7d79b3930910090407m5dc81d12pf2b0a47f6017804@mail.gmail.com>
	<20091009123615.BC50A3FA6E9@mail.cune.org>
	<7d79b3930910100647u7fd11413ucec0a535d68b3e3a@mail.gmail.com>
	<87f2f3b90910101009j1295d812jd98cae91e30b1eff@mail.gmail.com>
	<7d79b3930910120454m1ffc6dd4u42b91a4349597166@mail.gmail.com>
	<87f2f3b90910121149x6e2114f0q996a262ad272a1f6@mail.gmail.com>
	<7d79b3930910130645y658ae041v8219c4c881fa5698@mail.gmail.com>
	<5E7A8451-6264-4822-8152-1B94240BDD48@freeswitch.org>
	<7d79b3930910132234k4b99c794q8854ce20d14ee74e@mail.gmail.com>
Message-ID: <87f2f3b90910132303i5966a477ue82667bef975cc93@mail.gmail.com>

Lak,

Okay, it stood out right away: FS is trying u-law but Asterisk is trying
A-law. I'm not sure where the codec for openzap gets selected but you can
try modifying this line in vars.xml:

<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>

Try putting PCMA first. Also, I've never tried this but perhaps in your
dialplan you can set absolute_codec_string to PCMA. I'm very interesting in
hearing about the results. Please try them as soon as you can and report
back.

Thanks,
MC

On Tue, Oct 13, 2009 at 10:34 PM, lakshmanan ganapathy <lakindia89 at gmail.com
> wrote:

> Hi. I made the setup quickly with one of my colleague, and here is the
> debug log given by asterisk
>
> Steps done by me :
> 	+ Call Made from 04443902743 to 04439114600
> 	+ When call lands to 04439114600, I just route it to 09176454982.
>
> *CLI>
> *CLI>
> *CLI> pri debug span 1
> Enabled debugging on span 1
> *CLI>
> *CLI>
> *CLI>
> *CLI> < Protocol Discriminator: Q.931 (8)  len=41
> < Call Ref: len= 2 (reference 21/0x15) (Originator)
>
> < Message type: SETUP (5)
> < [a1]
> < Sending Complete (len= 1)
> < [04 03 80 90 a3]
> < Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer capability: Speech (0)
> <                              Ext: 1  Trans mode/rate: 64kbps, circuit-mode (16)
>
> <                                User information layer 1: A-Law (35)
> < [18 03 a1 83 9f]
> < Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Preferred  Dchan: 0
> <                        ChanSel: As indicated in following octets
>
> <                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
> <                       Ext: 1  Channel: 31 ]
> < [6c 0c 01 81 34 34 34 33 39 30 32 37 34 33]
> < Calling Number (len=14) [ Ext: 0  TON: Unknown Number Type (0)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
>
> <                           Presentation: Presentation permitted, user number passed network screening (1)  '4443902743' ]
> < [70 09 c1 33 39 31 31 34 36 30 30]
> < Called Number (len=11) [ Ext: 1  TON: Subscriber Number (4)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)  '39114600' ]
>
> -- Making new call for cr 21
> -- Processing Q.931 Call Setup
> -- Processing IE 161 (cs0, Sending Complete)
> -- Processing IE 4 (cs0, Bearer Capability)
> -- Processing IE 24 (cs0, Channel Identification)
> -- Processing IE 108 (cs0, Calling Party Number)
>
> -- Processing IE 112 (cs0, Called Party Number)
> q931.c:3504 q931_receive: call 21 on channel 31 enters state 6 (Call Present)
> q931.c:2769 q931_call_proceeding: call 21 on channel 31 enters state 9 (Incoming Call Proceeding)
>
> > Protocol Discriminator: Q.931 (8)  len=10
> > Call Ref: len= 2 (reference 21/0x15) (Terminator)
> > Message type: CALL PROCEEDING (2)
> > [18 03 a9 83 9f]
> > Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  Dchan: 0
> >                        ChanSel: As indicated in following octets
> >                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
> >                       Ext: 1  Channel: 31 ]
>
>     -- Accepting call from '4443902743' to '39114600' on channel 0/31, span 1
>     -- Executing [39114600 at incoming-from-pri:1] NoOp("Zap/31-1", "FMS Call Extn:39114600:| CID::| Channel:Zap/31-1:| ") in new stack
>
>     -- Executing [39114600 at incoming-from-pri:2] Dial("Zap/31-1", "Zap/g1/w09176454982||rj") in new stack
> -- Making new call for cr 32770
>     -- Requested transfer capability: 0x00 - SPEECH
>
> > Protocol Discriminator: Q.931 (8)  len=43
> > Call Ref: len= 2 (reference 2/0x2) (Originator)
> > Message type: SETUP (5)
> > [04 03 80 90 a3]
> > Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer capability: Speech (0)
> >                              Ext: 1  Trans mode/rate: 64kbps, circuit-mode (16)
> >                                User information layer 1: A-Law (35)
> > [18 03 a9 83 81]
> > Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  Dchan: 0
> >                        ChanSel: As indicated in following octets
> >                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
> >                       Ext: 1  Channel: 1 ]
> > [6c 0c 00 81 34 34 34 33 39 30 32 37 34 33]
> > Calling Number (len=14) [ Ext: 0  TON: Unknown Number Type (0)  NPI: Unknown Number Plan (0)
> >                           Presentation: Presentation permitted, user number passed network screening (1)  '4443902743' ]
> > [70 0c 80 30 39 31 37 36 34 35 34 39 38 32]
> > Called Number (len=14) [ Ext: 1  TON: Unknown Number Type (0)  NPI: Unknown Number Plan (0)  '09176454982' ]
>
> q931.c:3087 q931_setup: call 32770 on channel 1 enters state 1 (Call Initiated)
>
>     -- Called g1/w09176454982
> q931.c:2797 q931_alerting: call 21 on channel 31 enters state 7 (Call Received)
>
> > Protocol Discriminator: Q.931 (8)  len=9
> > Call Ref: len= 2 (reference 21/0x15) (Terminator)
> > Message type: ALERTING (1)
> > [1e 02 81 88]
> > Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Private network serving the local user (1)
> >                               Ext: 1  Progress Description: Inband information or appropriate pattern now available. (8) ]
>
> < Protocol Discriminator: Q.931 (8)  len=10
> < Call Ref: len= 2 (reference 2/0x2) (Terminator)
> < Message type: CALL PROCEEDING (2)
> < [18 03 a9 83 81]
> < Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  Dchan: 0
>
> <                        ChanSel: As indicated in following octets
> <                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
> <                       Ext: 1  Channel: 1 ]
> -- Processing IE 24 (cs0, Channel Identification)
>
> q931.c:3636 q931_receive: call 32770 on channel 1 enters state 3 (Outgoing call  Proceeding)
>     -- Zap/1-1 is proceeding passing it to Zap/31-1
> < Protocol Discriminator: Q.931 (8)  len=13
> < Call Ref: len= 2 (reference 2/0x2) (Terminator)
>
> < Message type: ALERTING (1)
> < [1e 02 84 88]
> < Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Public network serving the remote user (4)
> <                               Ext: 1  Progress Description: Inband information or appropriate pattern now available. (8) ]
>
> < [1e 02 84 82]
> < Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Public network serving the remote user (4)
> <                               Ext: 1  Progress Description: Called equipment is non-ISDN. (2) ]
>
> -- Processing IE 30 (cs0, Progress Indicator)
> -- Processing IE 30 (cs0, Progress Indicator)
> q931.c:3549 q931_receive: call 32770 on channel 1 enters state 4 (Call Delivered)
>     -- Zap/1-1 is ringing
> < Protocol Discriminator: Q.931 (8)  len=9
>
> < Call Ref: len= 2 (reference 2/0x2) (Terminator)
> < Message type: PROGRESS (3)
> < [1e 02 84 84]
> < Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Public network serving the remote user (4)
>
> <                               Ext: 1  Progress Description: Unknown (4) ]
> -- Processing IE 30 (cs0, Progress Indicator)
>     -- Zap/1-1 is making progress passing it to Zap/31-1
> < Protocol Discriminator: Q.931 (8)  len=9
>
> < Call Ref: len= 2 (reference 2/0x2) (Terminator)
> < Message type: PROGRESS (3)
> < [1e 02 84 88]
> < Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Public network serving the remote user (4)
>
> <                               Ext: 1  Progress Description: Inband information or appropriate pattern now available. (8) ]
> -- Processing IE 30 (cs0, Progress Indicator)
>     -- Zap/1-1 is making progress passing it to Zap/31-1
>
> < Protocol Discriminator: Q.931 (8)  len=19
> < Call Ref: len= 2 (reference 2/0x2) (Terminator)
> < Message type: CONNECT (7)
> < [4c 0c 01 83 39 31 37 36 34 35 34 39 38 32]
> < Connected Number (len=14) [ Ext: 0  TON: Unknown Number Type (0)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
>
> <                             Ext: 1  Presentation: Presentation allowed of network provided number (3)  '9176454982' ]
> -- Processing IE 76 (cs0, Connected Number)
> q931.c:3579 q931_receive: call 32770 on channel 1 enters state 10 (Active)
>
> > Protocol Discriminator: Q.931 (8)  len=5
> > Call Ref: len= 2 (reference 2/0x2) (Originator)
> > Message type: CONNECT ACKNOWLEDGE (15)
>
>     -- Zap/1-1 answered Zap/31-1
> q931.c:2904 q931_connect: call 21 on channel 31 enters state 8 (Connect Request)
>
> > Protocol Discriminator: Q.931 (8)  len=14
> > Call Ref: len= 2 (reference 21/0x15) (Terminator)
> > Message type: CONNECT (7)
> > [18 03 a9 83 9f]
> > Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  Dchan: 0
> >                        ChanSel: As indicated in following octets
> >                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
> >                       Ext: 1  Channel: 31 ]
> > [1e 02 81 82]
> > Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Private network serving the local user (1)
> >                               Ext: 1  Progress Description: Called equipment is non-ISDN. (2) ]
>
>     -- Native bridging Zap/31-1 and Zap/1-1
> < Protocol Discriminator: Q.931 (8)  len=5
> < Call Ref: len= 2 (reference 21/0x15) (Originator)
> < Message type: CONNECT ACKNOWLEDGE (15)
>
> q931.c:3664 q931_receive: call 21 on channel 31 enters state 10 (Active)
> < Protocol Discriminator: Q.931 (8)  len=9
> < Call Ref: len= 2 (reference 2/0x2) (Terminator)
> < Message type: DISCONNECT (69)
> < [08 02 80 90]
>
> < Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  Location: User (0)
> <                  Ext: 1  Cause: Normal Clearing (16), class = Normal Event (1) ]
> -- Processing IE 8 (cs0, Cause)
> q931.c:3779 q931_receive: call 32770 on channel 1 enters state 12 (Disconnect Indication)
>
>     -- Channel 0/1, span 1 got hangup request, cause 16
> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request
> q931.c:2920 q931_release: call 32770 on channel 1 enters state 19 (Release Request)
>
> > Protocol Discriminator: Q.931 (8)  len=9
> > Call Ref: len= 2 (reference 2/0x2) (Originator)
> > Message type: RELEASE (77)
> > [08 02 81 90]
> > Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  Location: Private network serving the local user (1)
> >                  Ext: 1  Cause: Normal Clearing (16), class = Normal Event (1) ]
>
>     -- Hungup 'Zap/1-1'
>   == Spawn extension (incoming-from-pri, 39114600, 2) exited non-zero on 'Zap/31-1'
>
>     -- Executing [h at incoming-from-pri:1] Hangup("Zap/31-1", "") in new stack
>   == Spawn extension (incoming-from-pri, h, 1) exited non-zero on 'Zap/31-1'
> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Active
>
> q931.c:2968 q931_disconnect: call 21 on channel 31 enters state 11 (Disconnect Request)
>
> > Protocol Discriminator: Q.931 (8)  len=9
> > Call Ref: len= 2 (reference 21/0x15) (Terminator)
> > Message type: DISCONNECT (69)
> > [08 02 81 90]
> > Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  Location: Private network serving the local user (1)
> >                  Ext: 1  Cause: Normal Clearing (16), class = Normal Event (1) ]
>
>     -- Hungup 'Zap/31-1'
> < Protocol Discriminator: Q.931 (8)  len=5
>
> < Call Ref: len= 2 (reference 2/0x2) (Terminator)
> < Message type: RELEASE COMPLETE (90)
> q931.c:3719 q931_receive: call 32770 on channel 1 enters state 0 (Null)
> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
>
> NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
> < Protocol Discriminator: Q.931 (8)  len=5
> < Call Ref: len= 2 (reference 21/0x15) (Originator)
> < Message type: RELEASE (77)
> q931.c:3754 q931_receive: call 21 on channel 31 enters state 0 (Null)
>
> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request
>
> > Protocol Discriminator: Q.931 (8)  len=9
> > Call Ref: len= 2 (reference 21/0x15) (Terminator)
> > Message type: RELEASE COMPLETE (90)
> > [08 02 81 90]
> > Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  Location: Private network serving the local user (1)
> >                  Ext: 1  Cause: Normal Clearing (16), class = Normal Event (1) ]
>
> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
>
> NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
>
>
> On Tue, Oct 13, 2009 at 9:16 PM, Michael S Collins <msc at freeswitch.org>wrote:
>
>>
>> On Oct 13, 2009, at 6:45 AM, lakshmanan ganapathy <lakindia89 at gmail.com>
>> wrote:
>>
>> We are using Reliance as the Carrier.
>> I think, with this same Reliance carrier, in my office, they are able to
>> make outgoing calls through asterisk+libpri.
>>
>>
>> If that's the case I would be very interested in seeing a pri debug from a
>> working call on an asterisk box. It might give us a clue as to what is not
>> working. -MC
>>
>>
>>
>> On Tue, Oct 13, 2009 at 12:19 AM, Michael Collins < <msc at freeswitch.org>
>> msc at freeswitch.org> wrote:
>>
>>> Lak,
>>>
>>> Okay I will need a little bit of time to dig into the IE's and what they
>>> contain. In the meantime can you tell me who the carrier is? I'd like to
>>> find out if they have some specific requirements. The fact that it doesn't
>>> work with libpri surprises me because that would mean that Asterisk systems
>>> would probably not work with this carrier as well.
>>>
>>> BTW, thanks for the very complete pastebin entries. :)
>>> -MC
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From shiyanov at gmail.com  Tue Oct 13 23:31:05 2009
From: shiyanov at gmail.com (Artem Shiyanov)
Date: Wed, 14 Oct 2009 10:31:05 +0400
Subject: [Freeswitch-users] mod_socket: custom timeout for HEARTBEAT
	event
In-Reply-To: <87f2f3b90910131148m1853a219md8788f55e40ef4c3@mail.gmail.com>
References: <d211f54a0910110959y3d3deed7s43201a3e39756585@mail.gmail.com>
	<86a32abc0910111444j1ab2e446p3cb94b0ae04126e3@mail.gmail.com>
	<97B4BDC9-5B11-4A98-B1BF-CC00A82ECBD9@jerris.com>
	<d211f54a0910112238i5f69275ey7cad437e47e6b567@mail.gmail.com>
	<191c3a030910120726p46d6410ejd94b61612686fbbc@mail.gmail.com>
	<d211f54a0910130832m25aaadd3t4d6e938d1c71d701@mail.gmail.com>
	<87f2f3b90910131148m1853a219md8788f55e40ef4c3@mail.gmail.com>
Message-ID: <d211f54a0910132331j14ccf510xd87bcb092a43f0b3@mail.gmail.com>

Finally!!
Thank you Michael, I didn't know about "status" app. It satisfies all my
desires.

And again,
thanks for all the community for the strong support!

Artem




On Tue, Oct 13, 2009 at 10:48 PM, Michael Collins <msc at freeswitch.org>wrote:

>
>
> On Tue, Oct 13, 2009 at 8:32 AM, Artem Shiyanov <shiyanov at gmail.com>wrote:
>
>> Sorry for my foolishness but I stil can't grasp it.
>> I'm developing app based on inbound mod_event_socket and I don't know how
>> to run "enable_heartbeat" or " sched_heartbeat" without specifying any alive
>> session uuid. I tried to use "create_uuid" and send mentioned  commands to
>> the created uuid  but this approach doesn't work. I do need to monitor the
>> FS state itself rather then any particular call.
>>
>> Please, enlighten me!
>> Artem
>>
>
> So if I understand correctly, you want to see the system heartbeat more
> frequently than every 20 seconds? What, exactly, is in the heartbeat that
> you need? You can get the same information by sending the API "status" and
> reading the results.
>
> If I read the source code correctly it looks like the system heartbeat is
> hard-coded at 20 seconds. See switch_core.c, line 155. However, before you
> go messing with a core file I would make sure that you can't already use the
> tools that are available, like sending a "status" API every 5 seconds (or
> whatever interval you need).
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sjmudd at pobox.com  Tue Oct 13 23:33:43 2009
From: sjmudd at pobox.com (Simon J Mudd)
Date: 14 Oct 2009 08:33:43 +0200
Subject: [Freeswitch-users] Building Freeswitch with VPATH (src and obj
	directories are different)?
In-Reply-To: <D8EB1A6C-38A6-4BC2-AD8C-46CF11993366@freeswitch.org>
References: <20091013205907.GA7545@mad06.wl0.org>
	<D8EB1A6C-38A6-4BC2-AD8C-46CF11993366@freeswitch.org>
Message-ID: <m3eip65vso.fsf@mad06.wl0.org>

brian at freeswitch.org (Brian West) writes:

> You shouldn't have to make clean usually ... doing so might break your  
> tree...

Why?

In any case on my mac (Leopard) neither make clean or distclean fully
cleans up afterwards.  I would prefer to get a completely "untouched"
source tree after doing this.  If that's supposed to work and I should
post a bug or report the problem in more detail then I can do that.

> You can usually get by with "make current" that will ensure  
> the critical things are cleaned and built correctly... every now and  
> they you'll hit a snafu but we'll usually tell you about it.

ok. That's one of the reasons for liking to use VPATH builds. It's also 
why many BSD system builds use this so that you can mount the source
tree readonly (perhaps from another server) and build the binaries
in a separate directory independently.

If that's not supported then that's fine.

I was trying to build the latest svn version of freeswitch (Revision:
15138) and this didn't build. As I'm not too familiar with compiling
at least on the Mac I thought I'd first try to see if a VPATH build
would work better. I'll have to try and investigate further.

Simon



From tculjaga at gmail.com  Tue Oct 13 23:48:26 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 14 Oct 2009 08:48:26 +0200
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <48B637E0506F4A809E4C0CAF39BD08BB@chaos>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
	<191c3a030910130803k321f39e3n265e64ab3c84a5e6@mail.gmail.com>
	<5e414ed0910130826j11d45a66q86d785f900e4196e@mail.gmail.com>
	<65d96fc80910131012v1f72ac49o6bfe9f2267e541c5@mail.gmail.com>
	<191c3a030910131101p418e0961t373db68285248ff6@mail.gmail.com>
	<538F4CF43B294D0EACABA281E1913B4C@chaos>
	<65d96fc80910131224m73f35b87g355be818ad3c0caf@mail.gmail.com>
	<48B637E0506F4A809E4C0CAF39BD08BB@chaos>
Message-ID: <65d96fc80910132348t202905fbub57cc4c814eb4e21@mail.gmail.com>

I suppose he want to have a central dialplan and a dummy phone instead...
something as a MGCP phone behavior.

T.

On Tue, Oct 13, 2009 at 10:22 PM, Metik <freeswitch-users-list at metik.com>wrote:

>  As evidenced by various DTMF interop issues (with RFC2833, inband,
> etc) over the years, I would avoid it if at all possible.
>
> What does it particularly do that can not accomplished by using RFC 2833 or
> (less ideal) inband DTMF?  Or are you attempting to use it as a band-aid to
> address some sort of interop issue with the carrier involved that is
> wrecking havoc with your particular application?
>
> -metik
>
> ----- Original Message -----
> *From:* Tihomir Culjaga <tculjaga at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, October 13, 2009 3:24 PM
> *Subject:* Re: [Freeswitch-users] SIP Overlap support?
>
> i never found it working properly... i always had some interoperability
> issues and i finished having a "dialplan" on my phones being delivered
> through a config file via tftp or http .. depending of the phone capability.
>
> BTW: using overlap can lead to a greater system load... be careful when
> setting the minimum number of digits you will send in 1st message. I wish
> you luck...
>
> T.
>
> On Tue, Oct 13, 2009 at 9:03 PM, Metik <freeswitch-users-list at metik.com>wrote:
>
>>  Both support it.  In the Grandstream, I believe it is called Early Dial
>> (vs. SNOM's Overlap Dialing).  It can be problematic if you have a device
>> somewhere in the middle that doesn't support 484s.
>>
>> -metik
>>
>>   ----- Original Message -----
>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Sent:* Tuesday, October 13, 2009 2:01 PM
>> *Subject:* Re: [Freeswitch-users] SIP Overlap support?
>>
>> i do think some softphone can do it but i forgot which one it was either
>> snom or grandstream
>>
>>
>> On Tue, Oct 13, 2009 at 12:12 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>
>>> you need a softswitch.... i'm afraid a SIP phone is not designed for
>>> overlap...
>>>
>>> T.
>>>
>>>
>>> On Tue, Oct 13, 2009 at 5:26 PM, Dennis <odermann at googlemail.com> wrote:
>>>
>>>> how could we try? we played arround with a snom phone (snom seems to
>>>> support something in this direction, but are not shure, how we can
>>>> test it and how we can see if it is supported or not.
>>>>
>>>> any hint?
>>>>
>>>>
>>>> 2009/10/13 Anthony Minessale <anthony.minessale at gmail.com>:
>>>>  > have you tried it?
>>>> > I *think* either we did support it or we would with a small patch to
>>>> sofia
>>>> > lib that I cannot recall if we ever got committed.
>>>> >
>>>> >
>>>> > On Tue, Oct 13, 2009 at 8:51 AM, Dennis <odermann at googlemail.com>
>>>> wrote:
>>>> >>
>>>> >> hi there,
>>>> >>
>>>> >> i would like to ask, if fs has support for something like "SIP
>>>> Overlap"?
>>>> >>
>>>> >> instead of receiving the phonenumber from our carrier in a block, we
>>>> >> want to receive the phonenumber digit-by-digit and we want to tell fs
>>>> >> when the number is complete. our carrier could send us the
>>>> phonenumber
>>>> >> digit-by-digit, but what about the fs-side?
>>>> >>
>>>> >>
>>>> >> thanks and kind regards
>>>> >> dennis
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >
>>>> >
>>>> >
>>>> > --
>>>> > Anthony Minessale II
>>>> >
>>>> > FreeSWITCH http://www.freeswitch.org/
>>>> > ClueCon http://www.cluecon.com/
>>>> > Twitter: http://twitter.com/FreeSWITCH_wire
>>>> >
>>>> > AIM: anthm
>>>> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> > IRC: irc.freenode.net #freeswitch
>>>> >
>>>> > FreeSWITCH Developer Conference
>>>> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>> > iax:guest at conference.freeswitch.org/888
>>>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> > pstn:213-799-1400
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Tue Oct 13 23:59:00 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 14 Oct 2009 08:59:00 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910130014hc5f5fcet677f87b7fb57559e@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<65d96fc80910130014hc5f5fcet677f87b7fb57559e@mail.gmail.com>
Message-ID: <65d96fc80910132359i6d53d8ban10f943795481bf7c@mail.gmail.com>

the e-mail bounced back .. don't know whether you received this...






here are the results!


Outbound calls:

SIP_user(1001) => FS => H323_SoftSwitch:
http://pastebin.freeswitch.org/10672

i get coredumps but this might be because i'm missing the bridge format....
what i'm using is: <action application="bridge" data="h323/0$1@${NCX_IP}"/>
that can be wrong :) Please advice here.




Inbound Calls:

openPhone => FS => SIP_user(1001): http://pastebin.freeswitch.org/10671


the call went in, hit a registered sip user (1001) and the phone started
ringing. The phone answered the call but no audio can be heared (i see RTP
from the SIP uset towards FS). Also, the openPhone never receives H.225
connect nor any OLC before connect. After some time, the H323 leg constates
the user never answered and releases the call with cause 19.




this is the h323.config.xml

tculjaga at subZero:~$ cat
/usr/local/freeswitch/conf/autoload_configs/h323.conf.xml
<configuration name="h323.conf" description="H323 Endpoints">
  <settings>
    <param name="trace-level" value="10"/>
    <param name="context" value="default"/>
    <param name="dialplan" value="XML"/>
    <param name="codec-prefs" value="PCMA,GSM,G729,G726"/>
    <param name="gk-address" value=""/>    <!-- empty to disable, "*" to
search LAN -->
    <param name="gk-identifer" value=""/>  <!-- optional name of gk -->
    <param name="gk-interface" value=""/>  <!-- optional listener interface
name -->
  </settings>
  <listeners>
    <listener name="default">
      <param name="h323-ip" value="10.4.62.7"/>
      <param name="h323-port" value="1720"/>
    </listener>
  </listeners>
</configuration>



and this is a dialplan for outgoing that i'm using


<include>

  <extension name="EMERGENCY">
    <condition field="destination_number" expression="^0(112|9[23456])$">
      <action application="set"
data="effective_caller_id_number=1001282122"/>
      <action application="set" data="NCX_IP=10.4.4.254"/>
      <action application="set" data="call_timeout=30"/>
      <action application="set" data="hangup_after_bridge=true"/>

      <!--action application="bridge" data="opal/h323:0$1@${NCX_IP}"/-->
      <action application="bridge" data="h323/0$1@${NCX_IP}"/>
    </condition>
  </extension>

  <extension name="SPECIAL_SERVICES">
    <condition field="destination_number" expression="^0(9[01789]\d{3,4})$">
      <action application="set"
data="effective_caller_id_number=1001282122"/>
      <action application="set" data="NCX_IP=10.4.4.254"/>
      <action application="set" data="call_timeout=30"/>
      <action application="set" data="hangup_after_bridge=true"/>

      <!--action application="bridge" data="opal/h323:0$1@${NCX_IP}"/-->
      <action application="bridge" data="h323/0$1@${NCX_IP}"/>
    </condition>
  </extension>

  <extension name="ENYTHING_ELSE">
    <condition field="destination_number"
expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
      <action application="set"
data="effective_caller_id_number=1001282122"/>
      <action application="set" data="NCX_IP=10.4.4.254"/>
      <action application="set" data="call_timeout=30"/>
      <action application="set" data="hangup_after_bridge=true"/>

      <!--action application="bridge" data="opal/h323:0$1@${NCX_IP}"/-->
      <action application="bridge" data="h323/0$1@${NCX_IP}"/>
    </condition>
  </extension>

</include>



P.S.: sorry for the attached rar, but don't know how to upload it to
pastebin :P


T.




2009/10/12 Tihomir Culjaga <tculjaga at gmail.com>

hi,
>
> finally i compiled it right ... had a stupid issue with ekiga and wrong
> ptlib in place...
>
> anyhow, i loaded the module and will continue the tests tomorrow ...first
> thing i arrive in my office :P
>
>
>
> freeswitch at subZero>
> freeswitch at subZero>
> API CALL [console(loglevel 7)] output:
> +OK console log level set to DEBUG
>
> freeswitch at subZero> load mod_h323
> 2009-10-12 23:38:09.509279 [CONSOLE] mod_h323.cpp:93 Starting loading
> mod_h323
> 2009-10-12 23:38:09.511413 [DEBUG] mod_h323.cpp:461 Created Listener
> 'default'
> 2009-10-12 23:38:09.511413 [DEBUG] mod_h323.cpp:305 Config capabilliti
> PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.511413 [DEBUG] mod_h323.cpp:309 Find capabilliti PCMU
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.511413 [DEBUG] mod_h323.cpp:309 Find capabilliti PCMA
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.511413 [DEBUG] h323caps.cxx:3264 FindCapability:
> "G.711-uLaw-64k{sw}"
> 2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:3176 Added capability:
> G.711-uLaw-64k <1>
> 2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:3264 FindCapability:
> "G.711-uLaw-64k{sw}"
> 2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:3264 FindCapability:
> "G.711-uLaw-64k{sw}"
> 2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:3264 FindCapability:
> "G.711-uLaw-64k{sw}"
> 2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:3264 FindCapability:
> "G.711-uLaw-64k{sw}"
> 2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:1989 No Extended
> Capabilities found to load
> 2009-10-12 23:38:09.512480 [DEBUG] mod_h323.cpp:325 H.323 added 1
> capabilities 'G.711-uLaw-64k*{sw}'
> 2009-10-12 23:38:09.512480 [DEBUG] mod_h323.cpp:309 Find capabilliti gsm to
> PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti msgsm
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti speex
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti lpc10
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti ilbc20
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti ilbc30
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g723
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g726
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g728
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g729b
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g729
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti slin
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g729a
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g729ab
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti g723.1
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
> g723.1-5k3 to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
> g723.1a-5k3 to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
> g723.1a-6k3 to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
> g723.1a-6k3-cisco to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
> g726-16 to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
> g726-24 to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
> g726-32 to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
> g726-40 to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti ilbc
> to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
> speex-18k2 to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
> speex-15k to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
> speex-11k to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.515708 [DEBUG] mod_h323.cpp:309 Find capabilliti
> speex-8k to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.515708 [DEBUG] mod_h323.cpp:309 Find capabilliti
> speex-5k95 to PCMA,GSM,G729,G726
> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
> UserInput/hookflash <2>
> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
> UserInput/basicString <3>
> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
> UserInput/dtmf <4>
> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
> UserInput/RFC2833 <5>
> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
> UserInput/Navigation <6>
> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
> UserInput/Softkey <7>
> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
> UserInput/PointDevice <8>
> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
> UserInput/Modal <9>
> 2009-10-12 23:38:09.516760 [DEBUG] osutil.cxx:188 File handle high water
> mark set: 44 PTCPSocket
> 2009-10-12 23:38:09.516760 [DEBUG] h323ep.cxx:1995 Started listener
> Listener[ip$10.4.62.7:1720]
> 2009-10-12 23:38:09.516760 [DEBUG] tlibthrd.cxx:547 Thread high water mark
> set: 3
> 2009-10-12 23:38:09.516760 [CONSOLE] mod_h323.cpp:108 Opal manager
> initialized and running
> 2009-10-12 23:38:09.516760 [DEBUG] tlibthrd.cxx:454 Started thread
> 0xb75e0f18 H323 Listener:b5c98b90
> 2009-10-12 23:38:09.516760 [CONSOLE] switch_loadable_module.c:889
> Successfully Loaded [mod_h323]
> 2009-10-12 23:38:09.516760 [NOTICE] switch_loadable_module.c:142 Adding
> Endpoint 'h323'
> API CALL [load(mod_h323)] output:
> +OK
>
> 2009-10-12 23:38:09.516760 [DEBUG] transports.cxx:1521 Awaiting TCP
> connections on port 1720
> 2009-10-12 23:38:09.517832 [DEBUG] transports.cxx:1475 Waiting on socket
> accept on ip$10.4.62.7:1720
> 2009-10-12 23:38:09.517832 [DEBUG] tlibthrd.cxx:1023
> PThread::PXBlockOnIO(44,2)
> freeswitch at subZero>
>
>
> BTW: to compile everytihng,  I used FS trunk, ptlib 2.7.1 and h323plus
> 1.22.0
>
> Will let you know the results asap.
>
>
> T.
>
>
>
> 2009/10/12 Georgiewskiy Yuriy <bottleman at icf.org.ru>
>
>> On 2009-10-12 18:03 +0200, Tihomir Culjaga wrote
>> freeswitch-users at lists.fre...:
>>
>> i already write this, ptlib 2.6.5, tou can find link to it on
>> oplalvoip.org, h32plus latest CVS version,
>> you can find it on www.h323plus.org.
>>
>> TC>hi,
>> TC>
>> TC>can't make it...
>> TC>
>> TC>subZero:~/freeswitch-trunk$ make mod_h323
>> TC>
>> TC>making all mod_h323
>> TC>Compiling mod_h323.cpp...
>> TC>quiet_libtool: compile:  g++ -g -ggdb -I/usr/local/include/ptlib
>> TC>-I/usr/local/include/openh323 -I. -DPTRACING=1 -D_REENTRANT
>> -fno-exceptions
>> TC>-I/home/tculjaga/freeswitch-trunk/src/include
>> TC>-I/home/tculjaga/freeswitch-trunk/libs/libteletone/src -fPIC
>> TC>-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g
>> -O2
>> TC>-D_GNU_SOURCE -DHAVE_CONFIG_H -c mod_h323.cpp  -fPIC -DPIC -o
>> TC>.libs/mod_h323.o
>> TC>In file included from /usr/local/include/openh323/h323.h:493,
>> TC>                 from mod_h323.h:8,
>> TC>                 from mod_h323.cpp:3:
>> TC>/usr/local/include/openh323/h323ep.h: In member function ??????virtual
>> void
>> TC>NATFactoryStartup::OnShutdown()??????:
>> TC>/usr/local/include/openh323/h323ep.h:2731: error:
>> ??????NatFactory?????? has not
>> TC>been declared
>> TC>make[4]: *** [mod_h323.lo] Error 1
>> TC>make[3]: *** [all] Error 1
>> TC>make[2]: *** [mod_h323-all] Error 1
>> TC>make[1]: *** [mod_h323] Error 2
>> TC>make: *** [mod_h323] Error 2
>> TC>
>> TC>
>> TC>
>> TC>what exact ptlib and h323plus versions did you use? .. can you send us
>> a
>> TC>link so we can use the exact ones.
>> TC>
>> TC>
>> TC>T.
>> TC>
>> TC>2009/10/12 Georgiewskiy Yuriy <bottleman at icf.org.ru>
>> TC>
>> TC>> On 2009-10-12 09:43 -0500, Brian West wrote
>> TC>> freeswitch-users at lists.freeswit...:
>> TC>>
>> TC>> BW>We can host this in our SVN if you wish?
>> TC>>
>> TC>> If in fs svn i think yes. But i think may be little time later?
>> TC>> i don't known is it builds on trunk because i develop it on 1.0.4.
>> TC>>
>> TC>> BW>/b
>> TC>> BW>
>> TC>> BW>On Oct 12, 2009, at 8:31 AM, Georgiewskiy Yuriy wrote:
>> TC>> BW>
>> TC>> BW>> ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/ alfa code,
>> but
>> TC>> seems
>> TC>> BW>> it work, but should be buggy,
>> TC>> BW>> to build need libpt 2.6.5 and h323plus cvs version, i test it
>> now on
>> TC>> fs
>> TC>> BW>> 1.0.4.
>> TC>> BW>
>> TC>>
>> TC>> C ?????????                       With Best Regards
>> TC>> ???????????? ????.                Georgiewskiy Yuriy
>> TC>> +7 4872 711666                    +7 4872 711666
>> TC>> ???? +7 4872 711143               fax +7 4872 711143
>> TC>> ???????? ??? "?? ?? ??????"       IT Service Ltd
>> TC>> http://nkoort.ru                  http://nkoort.ru
>> TC>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
>> TC>> YG129-RIPE                        YG129-RIPE
>> TC>>
>> TC>> _______________________________________________
>> TC>> FreeSWITCH-users mailing list
>> TC>> FreeSWITCH-users at lists.freeswitch.org
>> TC>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> TC>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> TC>> http://www.freeswitch.org
>> TC>>
>> TC>>
>> TC>
>>
>> C ?????????                       With Best Regards
>> ???????????? ????.                Georgiewskiy Yuriy
>> +7 4872 711666                    +7 4872 711666
>> ???? +7 4872 711143               fax +7 4872 711143
>> ???????? ??? "?? ?? ??????"       IT Service Ltd
>> http://nkoort.ru                  http://nkoort.ru
>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
>> YG129-RIPE                        YG129-RIPE
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From bottleman at icf.org.ru  Wed Oct 14 00:36:13 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Wed, 14 Oct 2009 11:36:13 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910132359i6d53d8ban10f943795481bf7c@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<65d96fc80910130014hc5f5fcet677f87b7fb57559e@mail.gmail.com>
	<65d96fc80910132359i6d53d8ban10f943795481bf7c@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910141134270.11620@srv.icf.org.ru>

On 2009-10-14 08:59 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

try sow start on h323 channel, there is a bug in faststart, i will fix it later.

TC>the e-mail bounced back .. don't know whether you received this...
TC>
TC>
TC>
TC>
TC>
TC>
TC>here are the results!
TC>
TC>
TC>Outbound calls:
TC>
TC>SIP_user(1001) => FS => H323_SoftSwitch:
TC>http://pastebin.freeswitch.org/10672
TC>
TC>i get coredumps but this might be because i'm missing the bridge format....
TC>what i'm using is: <action application="bridge" data="h323/0$1@${NCX_IP}"/>
TC>that can be wrong :) Please advice here.
TC>
TC>
TC>
TC>
TC>Inbound Calls:
TC>
TC>openPhone => FS => SIP_user(1001): http://pastebin.freeswitch.org/10671
TC>
TC>
TC>the call went in, hit a registered sip user (1001) and the phone started
TC>ringing. The phone answered the call but no audio can be heared (i see RTP
TC>from the SIP uset towards FS). Also, the openPhone never receives H.225
TC>connect nor any OLC before connect. After some time, the H323 leg constates
TC>the user never answered and releases the call with cause 19.
TC>
TC>
TC>
TC>
TC>this is the h323.config.xml
TC>
TC>tculjaga at subZero:~$ cat
TC>/usr/local/freeswitch/conf/autoload_configs/h323.conf.xml
TC><configuration name="h323.conf" description="H323 Endpoints">
TC>  <settings>
TC>    <param name="trace-level" value="10"/>
TC>    <param name="context" value="default"/>
TC>    <param name="dialplan" value="XML"/>
TC>    <param name="codec-prefs" value="PCMA,GSM,G729,G726"/>
TC>    <param name="gk-address" value=""/>    <!-- empty to disable, "*" to
TC>search LAN -->
TC>    <param name="gk-identifer" value=""/>  <!-- optional name of gk -->
TC>    <param name="gk-interface" value=""/>  <!-- optional listener interface
TC>name -->
TC>  </settings>
TC>  <listeners>
TC>    <listener name="default">
TC>      <param name="h323-ip" value="10.4.62.7"/>
TC>      <param name="h323-port" value="1720"/>
TC>    </listener>
TC>  </listeners>
TC></configuration>
TC>
TC>
TC>
TC>and this is a dialplan for outgoing that i'm using
TC>
TC>
TC><include>
TC>
TC>  <extension name="EMERGENCY">
TC>    <condition field="destination_number" expression="^0(112|9[23456])$">
TC>      <action application="set"
TC>data="effective_caller_id_number=1001282122"/>
TC>      <action application="set" data="NCX_IP=10.4.4.254"/>
TC>      <action application="set" data="call_timeout=30"/>
TC>      <action application="set" data="hangup_after_bridge=true"/>
TC>
TC>      <!--action application="bridge" data="opal/h323:0$1@${NCX_IP}"/-->
TC>      <action application="bridge" data="h323/0$1@${NCX_IP}"/>
TC>    </condition>
TC>  </extension>
TC>
TC>  <extension name="SPECIAL_SERVICES">
TC>    <condition field="destination_number" expression="^0(9[01789]\d{3,4})$">
TC>      <action application="set"
TC>data="effective_caller_id_number=1001282122"/>
TC>      <action application="set" data="NCX_IP=10.4.4.254"/>
TC>      <action application="set" data="call_timeout=30"/>
TC>      <action application="set" data="hangup_after_bridge=true"/>
TC>
TC>      <!--action application="bridge" data="opal/h323:0$1@${NCX_IP}"/-->
TC>      <action application="bridge" data="h323/0$1@${NCX_IP}"/>
TC>    </condition>
TC>  </extension>
TC>
TC>  <extension name="ENYTHING_ELSE">
TC>    <condition field="destination_number"
TC>expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
TC>      <action application="set"
TC>data="effective_caller_id_number=1001282122"/>
TC>      <action application="set" data="NCX_IP=10.4.4.254"/>
TC>      <action application="set" data="call_timeout=30"/>
TC>      <action application="set" data="hangup_after_bridge=true"/>
TC>
TC>      <!--action application="bridge" data="opal/h323:0$1@${NCX_IP}"/-->
TC>      <action application="bridge" data="h323/0$1@${NCX_IP}"/>
TC>    </condition>
TC>  </extension>
TC>
TC></include>
TC>
TC>
TC>
TC>P.S.: sorry for the attached rar, but don't know how to upload it to
TC>pastebin :P
TC>
TC>
TC>T.
TC>
TC>
TC>
TC>
TC>2009/10/12 Tihomir Culjaga <tculjaga at gmail.com>
TC>
TC>hi,
TC>>
TC>> finally i compiled it right ... had a stupid issue with ekiga and wrong
TC>> ptlib in place...
TC>>
TC>> anyhow, i loaded the module and will continue the tests tomorrow ...first
TC>> thing i arrive in my office :P
TC>>
TC>>
TC>>
TC>> freeswitch at subZero>
TC>> freeswitch at subZero>
TC>> API CALL [console(loglevel 7)] output:
TC>> +OK console log level set to DEBUG
TC>>
TC>> freeswitch at subZero> load mod_h323
TC>> 2009-10-12 23:38:09.509279 [CONSOLE] mod_h323.cpp:93 Starting loading
TC>> mod_h323
TC>> 2009-10-12 23:38:09.511413 [DEBUG] mod_h323.cpp:461 Created Listener
TC>> 'default'
TC>> 2009-10-12 23:38:09.511413 [DEBUG] mod_h323.cpp:305 Config capabilliti
TC>> PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.511413 [DEBUG] mod_h323.cpp:309 Find capabilliti PCMU
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.511413 [DEBUG] mod_h323.cpp:309 Find capabilliti PCMA
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.511413 [DEBUG] h323caps.cxx:3264 FindCapability:
TC>> "G.711-uLaw-64k{sw}"
TC>> 2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:3176 Added capability:
TC>> G.711-uLaw-64k <1>
TC>> 2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:3264 FindCapability:
TC>> "G.711-uLaw-64k{sw}"
TC>> 2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:3264 FindCapability:
TC>> "G.711-uLaw-64k{sw}"
TC>> 2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:3264 FindCapability:
TC>> "G.711-uLaw-64k{sw}"
TC>> 2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:3264 FindCapability:
TC>> "G.711-uLaw-64k{sw}"
TC>> 2009-10-12 23:38:09.512480 [DEBUG] h323caps.cxx:1989 No Extended
TC>> Capabilities found to load
TC>> 2009-10-12 23:38:09.512480 [DEBUG] mod_h323.cpp:325 H.323 added 1
TC>> capabilities 'G.711-uLaw-64k*{sw}'
TC>> 2009-10-12 23:38:09.512480 [DEBUG] mod_h323.cpp:309 Find capabilliti gsm to
TC>> PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti msgsm
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti speex
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti lpc10
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti ilbc20
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti ilbc30
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g723
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g726
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g728
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g729b
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g729
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti slin
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g729a
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.513529 [DEBUG] mod_h323.cpp:309 Find capabilliti g729ab
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti g723.1
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
TC>> g723.1-5k3 to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
TC>> g723.1a-5k3 to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
TC>> g723.1a-6k3 to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
TC>> g723.1a-6k3-cisco to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
TC>> g726-16 to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
TC>> g726-24 to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
TC>> g726-32 to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
TC>> g726-40 to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti ilbc
TC>> to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
TC>> speex-18k2 to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
TC>> speex-15k to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.514615 [DEBUG] mod_h323.cpp:309 Find capabilliti
TC>> speex-11k to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.515708 [DEBUG] mod_h323.cpp:309 Find capabilliti
TC>> speex-8k to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.515708 [DEBUG] mod_h323.cpp:309 Find capabilliti
TC>> speex-5k95 to PCMA,GSM,G729,G726
TC>> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
TC>> UserInput/hookflash <2>
TC>> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
TC>> UserInput/basicString <3>
TC>> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
TC>> UserInput/dtmf <4>
TC>> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
TC>> UserInput/RFC2833 <5>
TC>> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
TC>> UserInput/Navigation <6>
TC>> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
TC>> UserInput/Softkey <7>
TC>> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
TC>> UserInput/PointDevice <8>
TC>> 2009-10-12 23:38:09.515708 [DEBUG] h323caps.cxx:3176 Added capability:
TC>> UserInput/Modal <9>
TC>> 2009-10-12 23:38:09.516760 [DEBUG] osutil.cxx:188 File handle high water
TC>> mark set: 44 PTCPSocket
TC>> 2009-10-12 23:38:09.516760 [DEBUG] h323ep.cxx:1995 Started listener
TC>> Listener[ip$10.4.62.7:1720]
TC>> 2009-10-12 23:38:09.516760 [DEBUG] tlibthrd.cxx:547 Thread high water mark
TC>> set: 3
TC>> 2009-10-12 23:38:09.516760 [CONSOLE] mod_h323.cpp:108 Opal manager
TC>> initialized and running
TC>> 2009-10-12 23:38:09.516760 [DEBUG] tlibthrd.cxx:454 Started thread
TC>> 0xb75e0f18 H323 Listener:b5c98b90
TC>> 2009-10-12 23:38:09.516760 [CONSOLE] switch_loadable_module.c:889
TC>> Successfully Loaded [mod_h323]
TC>> 2009-10-12 23:38:09.516760 [NOTICE] switch_loadable_module.c:142 Adding
TC>> Endpoint 'h323'
TC>> API CALL [load(mod_h323)] output:
TC>> +OK
TC>>
TC>> 2009-10-12 23:38:09.516760 [DEBUG] transports.cxx:1521 Awaiting TCP
TC>> connections on port 1720
TC>> 2009-10-12 23:38:09.517832 [DEBUG] transports.cxx:1475 Waiting on socket
TC>> accept on ip$10.4.62.7:1720
TC>> 2009-10-12 23:38:09.517832 [DEBUG] tlibthrd.cxx:1023
TC>> PThread::PXBlockOnIO(44,2)
TC>> freeswitch at subZero>
TC>>
TC>>
TC>> BTW: to compile everytihng,  I used FS trunk, ptlib 2.7.1 and h323plus
TC>> 1.22.0
TC>>
TC>> Will let you know the results asap.
TC>>
TC>>
TC>> T.
TC>>
TC>>
TC>>
TC>> 2009/10/12 Georgiewskiy Yuriy <bottleman at icf.org.ru>
TC>>
TC>>> On 2009-10-12 18:03 +0200, Tihomir Culjaga wrote
TC>>> freeswitch-users at lists.fre...:
TC>>>
TC>>> i already write this, ptlib 2.6.5, tou can find link to it on
TC>>> oplalvoip.org, h32plus latest CVS version,
TC>>> you can find it on www.h323plus.org.
TC>>>
TC>>> TC>hi,
TC>>> TC>
TC>>> TC>can't make it...
TC>>> TC>
TC>>> TC>subZero:~/freeswitch-trunk$ make mod_h323
TC>>> TC>
TC>>> TC>making all mod_h323
TC>>> TC>Compiling mod_h323.cpp...
TC>>> TC>quiet_libtool: compile:  g++ -g -ggdb -I/usr/local/include/ptlib
TC>>> TC>-I/usr/local/include/openh323 -I. -DPTRACING=1 -D_REENTRANT
TC>>> -fno-exceptions
TC>>> TC>-I/home/tculjaga/freeswitch-trunk/src/include
TC>>> TC>-I/home/tculjaga/freeswitch-trunk/libs/libteletone/src -fPIC
TC>>> TC>-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g
TC>>> -O2
TC>>> TC>-D_GNU_SOURCE -DHAVE_CONFIG_H -c mod_h323.cpp  -fPIC -DPIC -o
TC>>> TC>.libs/mod_h323.o
TC>>> TC>In file included from /usr/local/include/openh323/h323.h:493,
TC>>> TC>                 from mod_h323.h:8,
TC>>> TC>                 from mod_h323.cpp:3:
TC>>> TC>/usr/local/include/openh323/h323ep.h: In member function ??????virtual
TC>>> void
TC>>> TC>NATFactoryStartup::OnShutdown()??????:
TC>>> TC>/usr/local/include/openh323/h323ep.h:2731: error:
TC>>> ??????NatFactory?????? has not
TC>>> TC>been declared
TC>>> TC>make[4]: *** [mod_h323.lo] Error 1
TC>>> TC>make[3]: *** [all] Error 1
TC>>> TC>make[2]: *** [mod_h323-all] Error 1
TC>>> TC>make[1]: *** [mod_h323] Error 2
TC>>> TC>make: *** [mod_h323] Error 2
TC>>> TC>
TC>>> TC>
TC>>> TC>
TC>>> TC>what exact ptlib and h323plus versions did you use? .. can you send us
TC>>> a
TC>>> TC>link so we can use the exact ones.
TC>>> TC>
TC>>> TC>
TC>>> TC>T.
TC>>> TC>
TC>>> TC>2009/10/12 Georgiewskiy Yuriy <bottleman at icf.org.ru>
TC>>> TC>
TC>>> TC>> On 2009-10-12 09:43 -0500, Brian West wrote
TC>>> TC>> freeswitch-users at lists.freeswit...:
TC>>> TC>>
TC>>> TC>> BW>We can host this in our SVN if you wish?
TC>>> TC>>
TC>>> TC>> If in fs svn i think yes. But i think may be little time later?
TC>>> TC>> i don't known is it builds on trunk because i develop it on 1.0.4.
TC>>> TC>>
TC>>> TC>> BW>/b
TC>>> TC>> BW>
TC>>> TC>> BW>On Oct 12, 2009, at 8:31 AM, Georgiewskiy Yuriy wrote:
TC>>> TC>> BW>
TC>>> TC>> BW>> ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/ alfa code,
TC>>> but
TC>>> TC>> seems
TC>>> TC>> BW>> it work, but should be buggy,
TC>>> TC>> BW>> to build need libpt 2.6.5 and h323plus cvs version, i test it
TC>>> now on
TC>>> TC>> fs
TC>>> TC>> BW>> 1.0.4.
TC>>> TC>> BW>
TC>>> TC>>
TC>>> TC>> C ?????????                       With Best Regards
TC>>> TC>> ???????????? ????.                Georgiewskiy Yuriy
TC>>> TC>> +7 4872 711666                    +7 4872 711666
TC>>> TC>> ???? +7 4872 711143               fax +7 4872 711143
TC>>> TC>> ???????? ??? "?? ?? ??????"       IT Service Ltd
TC>>> TC>> http://nkoort.ru                  http://nkoort.ru
TC>>> TC>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
TC>>> TC>> YG129-RIPE                        YG129-RIPE
TC>>> TC>>
TC>>> TC>> _______________________________________________
TC>>> TC>> FreeSWITCH-users mailing list
TC>>> TC>> FreeSWITCH-users at lists.freeswitch.org
TC>>> TC>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>>> TC>> UNSUBSCRIBE:
TC>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>>> TC>> http://www.freeswitch.org
TC>>> TC>>
TC>>> TC>>
TC>>> TC>
TC>>>
TC>>> C ?????????                       With Best Regards
TC>>> ???????????? ????.                Georgiewskiy Yuriy
TC>>> +7 4872 711666                    +7 4872 711666
TC>>> ???? +7 4872 711143               fax +7 4872 711143
TC>>> ???????? ??? "?? ?? ??????"       IT Service Ltd
TC>>> http://nkoort.ru                  http://nkoort.ru
TC>>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
TC>>> YG129-RIPE                        YG129-RIPE
TC>>>
TC>>> _______________________________________________
TC>>> FreeSWITCH-users mailing list
TC>>> FreeSWITCH-users at lists.freeswitch.org
TC>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>>> http://www.freeswitch.org
TC>>>
TC>>>
TC>>
TC>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From shaheryarkh at googlemail.com  Wed Oct 14 00:55:22 2009
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Wed, 14 Oct 2009 13:55:22 +0600
Subject: [Freeswitch-users] Some help with my post-paid billing project
In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6703241D1E52@mse17be1.mse17.exchange.ms>
References: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
	<74730C11-75D0-4BC0-9F63-8B6C9F0F499F@me.com>
	<86a32abc0910131836w5f113a16s247caf5eeb0f8fe4@mail.gmail.com>
	<86a32abc0910131853v244ed61bof9c24f345a0944b2@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C6703241D1E52@mse17be1.mse17.exchange.ms>
Message-ID: <b16156850910140055r37581014iddf52d915aa7eaa4@mail.gmail.com>

I fully agree that direct matching is much faster then pattern matching in
SQL.

One of my clients had same problem, he had around 12 million number prefixes
in a table and during each call an AGI script use to query that table to
find longest prefix match, but this use to take like 3-5 seconds even with
indexed columns. So after a lots of R & D we come up with following logic,

1. We break the destination number length-wise, e.g. suppose number is
923344224088 then length chunks would be,

923344224088, 92334422408, 9233442240, 923344224, 92334422, 9233442, 923344,
92334, 9233, 923, 92, 9

2. Then use SQL LIKE function in WHERE clause (you can also use SQL OR
function if your DBMS doesn't support SQL LIKE function), and pass all these
chunks to it, e.g.

WHERE prefix LIKE (923344224088, 92334422408, 9233442240, 923344224,
92334422, 9233442, 923344, 92334, 9233, 923, 92, 9)

3. Lastly we ORDER the result by prefix length, e.g.

ORDER BY LENGTH(prefix) DESC LIMIT 1

4. The complete query will be,

SELECT * FROM prefixes
WHERE prefix LIKE (923344224088, 92334422408, 9233442240, 923344224,
92334422, 9233442, 923344, 92334, 9233, 923, 92, 9)
ORDER BY LENGTH(prefix) DESC LIMIT 1

Now the query takes less then 150 ms to execute. :-)

Here is an STL method that can generate this query, i am sure you can
convert it to any programming language of your choice easily.

=========================================================
*std::string GetQuery(std::string destination) {
        std::string query = "SELECT * FROM prefixes WHERE prefix LIKE ('" +
destination;

        for(int i=1; i<destination.length(); i++) {
                query += "','" + destination.substr(0, (i * -1));

        }

        query += "') ORDER BY LENGTH(prefix) DESC LIMIT 1";
        return query;
}
*=========================================================

I am pretty sure this query is 100% ANSI SQL compatible (
http://en.wikipedia.org/wiki/SQL).

Thank you.


On Wed, Oct 14, 2009 at 10:15 AM, Michael Giagnocavo <mgg at giagnocavo.net>wrote:

>  In our testing with SQL Server, we found that executing several queries
> for direct matches yielded far better performance than one query trying to
> check prefixes. (The column was also part of the clustered index, but AFAIK
> MySQL doesn?t support defining your own clustered indexes; you get the PK
> always.)
>
>
>
> -Michael
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Diego Viola
> *Sent:* Tuesday, October 13, 2009 7:54 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Some help with my post-paid billing
> project
>
>
>
> Wrong question.
>
> Is there a way to compare numbers with prefixes without using the prefix
> module?
>
> Diego
>
> On Wed, Oct 14, 2009 at 1:36 AM, Diego Viola <diego.viola at gmail.com>
> wrote:
>
> I'm using MySQL now but I will try PostgreSQL with the prefix module, is
> there a way to do that without the prefix module and with regular SQL?
>
> Any examples?
>
> Diego
>
>
>
> On Tue, Oct 13, 2009 at 10:45 PM, Even Andr? Fiskvik <grevenx at me.com>
> wrote:
>
> What database are you using?
> You could do this with regular SQL, but it would by a costly operation,
> for PostgreSQL we're using the prefix module:
> http://pgfoundry.org/projects/prefix/
>
> You can then match the closest prefix by using something like
> "WHERE myprefix_col @> caller_destination_number ORDER BY LENGTH
> (myprefix_col::text) LIMIT 1;"
>
>
> Best regards,
> Even Andr?
>
>
>
> On 13. okt. 2009, at 23.53, Diego Viola wrote:
>
> > Hello,
> >
> > I'm trying to write a post-paid billing script, I have the CDR on my
> > database and also a "rates" table, the CDR contains fields like
> > caller_destination_number, variable_duration, etc. and the rates
> > table contains: destination, prefix, rate (cost).
> >
> > The problem is that I can't just strip the destination number to
> > take the prefix from it because I have to deal with destination
> > numbers from different countries and they all have different prefix
> > lengths... so I need to find another way to take the prefix from the
> > destination number.
> >
> > Any ideas how to do this?
> >
> > Thanks,
> >
> > Diego
> >
>
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
________________________________________________________
|
                     |
| FATAL ERROR                                                           ---
O X |
|_______________________________________________________|
|                        You have moved the mouse.
 |
| Windows must be restarted for the changes to take effect.   |
|                                        <OK>
               |
####################################/


Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From tculjaga at gmail.com  Wed Oct 14 00:58:47 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 14 Oct 2009 09:58:47 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910141134270.11620@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<65d96fc80910130014hc5f5fcet677f87b7fb57559e@mail.gmail.com>
	<65d96fc80910132359i6d53d8ban10f943795481bf7c@mail.gmail.com>
	<Pine.LNX.4.64.0910141134270.11620@srv.icf.org.ru>
Message-ID: <65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>

2009/10/14 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-14 08:59 +0200, Tihomir Culjaga wrote
> freeswitch-users at lists.fre...:
>
> try sow start on h323 channel, there is a bug in faststart, i will fix it
> later.
>
>
there are few things,

1. capability PCMU/PCMA needs to be inverted
2. when you place outgoing calls SIP_user => FS => H323_endpoint FS
corediumps: what is the data format for bridge? Is that correct =>
data="h323/1001 at 10.1.1.1" ?
3. when you place incoming calls H323_endpoint => FS => SIPUser, the call
goes through but there is no audio. After H.225 connect and TCS/MSD, FS
stops sending RTP back to te originator.
4. when you place incoming calls H323_endpoint => FS => SIPUser and you
hangup from SIPUser side, the call is not released on H323 side...
switch_core_state_machine
complains about a wrong thread.


i will send you logs for:

1. slow start
2. slow start early h245
3. fast start tunneling true
4. faststart tunneling false
5. faststart with early h245 tunneling true
6. faststart with early h245 tunneling false


do you need a tcpdump for every scenario as well ?

T.
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From bottleman at icf.org.ru  Wed Oct 14 01:05:37 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Wed, 14 Oct 2009 12:05:37 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<65d96fc80910130014hc5f5fcet677f87b7fb57559e@mail.gmail.com>
	<65d96fc80910132359i6d53d8ban10f943795481bf7c@mail.gmail.com>
	<Pine.LNX.4.64.0910141134270.11620@srv.icf.org.ru>
	<65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910141202500.21535@srv.icf.org.ru>

On 2009-10-14 09:58 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

i need trace level 4 from mod_h323 and debug log of entire call, tcpdump may be needed later, i have no way 
to test it on this time, i do it later.


TC>2009/10/14 Georgiewskiy Yuriy <bottleman at icf.org.ru>
TC>
TC>> On 2009-10-14 08:59 +0200, Tihomir Culjaga wrote
TC>> freeswitch-users at lists.fre...:
TC>>
TC>> try sow start on h323 channel, there is a bug in faststart, i will fix it
TC>> later.
TC>>
TC>>
TC>there are few things,
TC>
TC>1. capability PCMU/PCMA needs to be inverted
TC>2. when you place outgoing calls SIP_user => FS => H323_endpoint FS
TC>corediumps: what is the data format for bridge? Is that correct =>
TC>data="h323/1001 at 10.1.1.1" ?
TC>3. when you place incoming calls H323_endpoint => FS => SIPUser, the call
TC>goes through but there is no audio. After H.225 connect and TCS/MSD, FS
TC>stops sending RTP back to te originator.
TC>4. when you place incoming calls H323_endpoint => FS => SIPUser and you
TC>hangup from SIPUser side, the call is not released on H323 side...
TC>switch_core_state_machine
TC>complains about a wrong thread.
TC>
TC>
TC>i will send you logs for:
TC>
TC>1. slow start
TC>2. slow start early h245
TC>3. fast start tunneling true
TC>4. faststart tunneling false
TC>5. faststart with early h245 tunneling true
TC>6. faststart with early h245 tunneling false
TC>
TC>
TC>do you need a tcpdump for every scenario as well ?
TC>
TC>T.
TC>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From tculjaga at gmail.com  Wed Oct 14 01:16:42 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 14 Oct 2009 10:16:42 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910141202500.21535@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<65d96fc80910130014hc5f5fcet677f87b7fb57559e@mail.gmail.com>
	<65d96fc80910132359i6d53d8ban10f943795481bf7c@mail.gmail.com>
	<Pine.LNX.4.64.0910141134270.11620@srv.icf.org.ru>
	<65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>
	<Pine.LNX.4.64.0910141202500.21535@srv.icf.org.ru>
Message-ID: <65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>

2009/10/14 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-14 09:58 +0200, Tihomir Culjaga wrote
> freeswitch-users at lists.fre...:
>
> i need trace level 4 from mod_h323 and debug log of entire call, tcpdump
> may be needed later, i have no way
> to test it on this time, i do it later.
>
>
> Ok, will generate this logs...

hope you will recover soon.

T.
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From diego.viola at gmail.com  Wed Oct 14 01:28:11 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Wed, 14 Oct 2009 08:28:11 +0000
Subject: [Freeswitch-users] Some help with my post-paid billing project
In-Reply-To: <4AD5324F.2030000@ttnc.co.uk>
References: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
	<74730C11-75D0-4BC0-9F63-8B6C9F0F499F@me.com>
	<86a32abc0910131836w5f113a16s247caf5eeb0f8fe4@mail.gmail.com>
	<4AD5324F.2030000@ttnc.co.uk>
Message-ID: <86a32abc0910140128y553786ean13d5699a799f116d@mail.gmail.com>

Nice, I just converted this to Ruby/Sequel.

DB[:rates].first{{prefix=>substring('number', 1, length(prefix))}}

Thanks for the help :).

On Wed, Oct 14, 2009 at 2:07 AM, TTNC - Adnan Barakat
<technical at ttnc.co.uk>wrote:

> Diego Viola wrote:
> > I'm using MySQL now but I will try PostgreSQL with the prefix module, is
> > there a way to do that without the prefix module and with regular SQL?
> >
> > Any examples?
> SELECT * FROM rates WHERE prefix = SUBSTRING('$NUMBER$', 1,
> LENGTH(prefix)) LIMIT 1
>
> Adnan
>
> > Diego
> >
> > On Tue, Oct 13, 2009 at 10:45 PM, Even Andr? Fiskvik <grevenx at me.com
> > <mailto:grevenx at me.com>> wrote:
> >
> >     What database are you using?
> >     You could do this with regular SQL, but it would by a costly
> operation,
> >     for PostgreSQL we're using the prefix module:
> >     http://pgfoundry.org/projects/prefix/
> >
> >     You can then match the closest prefix by using something like
> >     "WHERE myprefix_col @> caller_destination_number ORDER BY LENGTH
> >     (myprefix_col::text) LIMIT 1;"
> >
> >
> >     Best regards,
> >     Even Andr?
> >
> >
> >     On 13. okt. 2009, at 23.53, Diego Viola wrote:
> >
> >     > Hello,
> >     >
> >     > I'm trying to write a post-paid billing script, I have the CDR on
> my
> >     > database and also a "rates" table, the CDR contains fields like
> >     > caller_destination_number, variable_duration, etc. and the rates
> >     > table contains: destination, prefix, rate (cost).
> >     >
> >     > The problem is that I can't just strip the destination number to
> >     > take the prefix from it because I have to deal with destination
> >     > numbers from different countries and they all have different prefix
> >     > lengths... so I need to find another way to take the prefix from
> the
> >     > destination number.
> >     >
> >     > Any ideas how to do this?
> >     >
> >     > Thanks,
> >     >
> >     > Diego
> >     >
> >     > _______________________________________________
> >     > FreeSWITCH-users mailing list
> >     > FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-
> >     > users
> >     > http://www.freeswitch.org
> >
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From diego.viola at gmail.com  Wed Oct 14 01:42:13 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Wed, 14 Oct 2009 08:42:13 +0000
Subject: [Freeswitch-users] Some help with my post-paid billing project
In-Reply-To: <86a32abc0910140128y553786ean13d5699a799f116d@mail.gmail.com>
References: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
	<74730C11-75D0-4BC0-9F63-8B6C9F0F499F@me.com>
	<86a32abc0910131836w5f113a16s247caf5eeb0f8fe4@mail.gmail.com>
	<4AD5324F.2030000@ttnc.co.uk>
	<86a32abc0910140128y553786ean13d5699a799f116d@mail.gmail.com>
Message-ID: <86a32abc0910140142g3f3ee930p7afc6d39baf5b7ec@mail.gmail.com>

DB[:rates].where(:prefix => substring('number', 1, length(prefix)).first

Rather.

On Wed, Oct 14, 2009 at 8:28 AM, Diego Viola <diego.viola at gmail.com> wrote:

> Nice, I just converted this to Ruby/Sequel.
>
> DB[:rates].first{{prefix=>substring('number', 1, length(prefix))}}
>
> Thanks for the help :).
>
>
> On Wed, Oct 14, 2009 at 2:07 AM, TTNC - Adnan Barakat <
> technical at ttnc.co.uk> wrote:
>
>> Diego Viola wrote:
>> > I'm using MySQL now but I will try PostgreSQL with the prefix module, is
>> > there a way to do that without the prefix module and with regular SQL?
>> >
>> > Any examples?
>> SELECT * FROM rates WHERE prefix = SUBSTRING('$NUMBER$', 1,
>> LENGTH(prefix)) LIMIT 1
>>
>> Adnan
>>
>> > Diego
>> >
>> > On Tue, Oct 13, 2009 at 10:45 PM, Even Andr? Fiskvik <grevenx at me.com
>> > <mailto:grevenx at me.com>> wrote:
>> >
>> >     What database are you using?
>> >     You could do this with regular SQL, but it would by a costly
>> operation,
>> >     for PostgreSQL we're using the prefix module:
>> >     http://pgfoundry.org/projects/prefix/
>> >
>> >     You can then match the closest prefix by using something like
>> >     "WHERE myprefix_col @> caller_destination_number ORDER BY LENGTH
>> >     (myprefix_col::text) LIMIT 1;"
>> >
>> >
>> >     Best regards,
>> >     Even Andr?
>> >
>> >
>> >     On 13. okt. 2009, at 23.53, Diego Viola wrote:
>> >
>> >     > Hello,
>> >     >
>> >     > I'm trying to write a post-paid billing script, I have the CDR on
>> my
>> >     > database and also a "rates" table, the CDR contains fields like
>> >     > caller_destination_number, variable_duration, etc. and the rates
>> >     > table contains: destination, prefix, rate (cost).
>> >     >
>> >     > The problem is that I can't just strip the destination number to
>> >     > take the prefix from it because I have to deal with destination
>> >     > numbers from different countries and they all have different
>> prefix
>> >     > lengths... so I need to find another way to take the prefix from
>> the
>> >     > destination number.
>> >     >
>> >     > Any ideas how to do this?
>> >     >
>> >     > Thanks,
>> >     >
>> >     > Diego
>> >     >
>> >     > _______________________________________________
>> >     > FreeSWITCH-users mailing list
>> >     > FreeSWITCH-users at lists.freeswitch.org
>> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >     > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-
>> >     > users
>> >     > http://www.freeswitch.org
>> >
>> >
>> >     _______________________________________________
>> >     FreeSWITCH-users mailing list
>> >     FreeSWITCH-users at lists.freeswitch.org
>> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >     UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >     http://www.freeswitch.org
>> >
>> >
>> >
>> > ------------------------------------------------------------------------
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From god.nirvana at gmail.com  Wed Oct 14 01:45:55 2009
From: god.nirvana at gmail.com (qian ma)
Date: Wed, 14 Oct 2009 16:45:55 +0800
Subject: [Freeswitch-users] how can i get the caller-number when the caller
	in the conference
Message-ID: <49161ef40910140145r7b8d28dbu3d9e44b4c17547de@mail.gmail.com>

hi all:
      i can get the member_id in the conference,but how can can i get the
number??
      thx!
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From tculjaga at gmail.com  Wed Oct 14 02:15:30 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 14 Oct 2009 11:15:30 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<65d96fc80910130014hc5f5fcet677f87b7fb57559e@mail.gmail.com>
	<65d96fc80910132359i6d53d8ban10f943795481bf7c@mail.gmail.com>
	<Pine.LNX.4.64.0910141134270.11620@srv.icf.org.ru>
	<65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>
	<Pine.LNX.4.64.0910141202500.21535@srv.icf.org.ru>
	<65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>
Message-ID: <65d96fc80910140215y357fede1l37f6cba9725ddd89@mail.gmail.com>

On Wed, Oct 14, 2009 at 10:16 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:

>
>
> 2009/10/14 Georgiewskiy Yuriy <bottleman at icf.org.ru>
>
>> On 2009-10-14 09:58 +0200, Tihomir Culjaga wrote
>> freeswitch-users at lists.fre...:
>>
>> i need trace level 4 from mod_h323 and debug log of entire call, tcpdump
>> may be needed later, i have no way
>> to test it on this time, i do it later.
>>
>>

Here are the logs:

call flow is H323_endpoint => FS => SIP_user

slow start                => http://pastebin.freeswitch.org/10693
slow start w eH245            => http://pastebin.freeswitch.org/10694
fastStart w tunneling true         => http://pastebin.freeswitch.org/10701
fastStart w tunneling false        => http://pastebin.freeswitch.org/10702
fastStart w tunneling true eH245    => http://pastebin.freeswitch.org/10699
fastStart w tunneling false eH245    => http://pastebin.freeswitch.org/10700


In all cases, there is no audio on SIP User side... i see the SIP Phone
(terminator) sending RTP to FS but FS is not forwarding it back to the
originator. Also, i see originator (H323) sending RTP to FS but FS doesn't
forward it to the terminator (SIP).



call flow is SIP_user => FS => H323_endpoint is failing .. coredumped....
http://pastebin.freeswitch.org/10703



T.




>
>
> hope you will recover soon.
>
> T.
>
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From srinivas.ksvreddy at gmail.com  Wed Oct 14 02:49:40 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Wed, 14 Oct 2009 15:19:40 +0530
Subject: [Freeswitch-users] Fwd: Groups information in sqllite
In-Reply-To: <B3C83D64-BC34-4F67-B72E-9EAE6F3A362E@jerris.com>
References: <f8af5740910122301u338ca128macfc0b769ea0e961@mail.gmail.com>
	<f8af5740910122302i47ccde5bo9df2bae965408394@mail.gmail.com>
	<B3C83D64-BC34-4F67-B72E-9EAE6F3A362E@jerris.com>
Message-ID: <f8af5740910140249t67df3364k1bb81a9c544f185e@mail.gmail.com>

Hi michael,

Thank you very much for your valuable time,
you are right, i have seen freeswitch.xml.fsxml,
groups data is taking from these xml file. instead of taking from these xml
file can i take from sqllite?


Thanks
Srinivasula Reddy

On Tue, Oct 13, 2009 at 9:12 PM, Michael Jerris <mike at jerris.com> wrote:

> Group information is not stored in sqlite, it is pulled from the xml
> registry (switch_xml_locate_group function can find them) .  Also,
> please do not cross post between lists.
>
> http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups
> http://wiki.freeswitch.org/wiki/Mod_commands#in_group
> http://wiki.freeswitch.org/wiki/Mod_commands#group_call
>
> Mike
>
>
> On Oct 13, 2009, at 2:02 AM, srinivasula reddy wrote:
> > can any know where group information is exactly stored in sqllite
> > database, i have seen sip_registration here i can find the
> > registered users,
> > in the same way how i can i find the group information, and which
> > user belongs to which user?
> > any help would be great.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Srinivasula Reddy K
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From lakindia89 at gmail.com  Wed Oct 14 03:03:33 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Wed, 14 Oct 2009 15:33:33 +0530
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <87f2f3b90910132303i5966a477ue82667bef975cc93@mail.gmail.com>
References: <7d79b3930910090407m5dc81d12pf2b0a47f6017804@mail.gmail.com>
	<20091009123615.BC50A3FA6E9@mail.cune.org>
	<7d79b3930910100647u7fd11413ucec0a535d68b3e3a@mail.gmail.com>
	<87f2f3b90910101009j1295d812jd98cae91e30b1eff@mail.gmail.com>
	<7d79b3930910120454m1ffc6dd4u42b91a4349597166@mail.gmail.com>
	<87f2f3b90910121149x6e2114f0q996a262ad272a1f6@mail.gmail.com>
	<7d79b3930910130645y658ae041v8219c4c881fa5698@mail.gmail.com>
	<5E7A8451-6264-4822-8152-1B94240BDD48@freeswitch.org>
	<7d79b3930910132234k4b99c794q8854ce20d14ee74e@mail.gmail.com>
	<87f2f3b90910132303i5966a477ue82667bef975cc93@mail.gmail.com>
Message-ID: <7d79b3930910140303k4bff6d5fgbc55a173194e22c9@mail.gmail.com>

Hi,
Tried setting PCMA as first parameter in vars.xml, no use. Still I got the
same error.

Tried setting the absolute_codec_string to PCMA, but got the same error.

But I tried with a another parameter in openzap.conf.xml, which I got from
the following thread
http://www.nabble.com/Problem-dialing-out-via-E1-to22479047.html#a22626383

<param name="l1" value="alaw"/>

After that I was able to make outgoing call, with the dialplan that I've.

But I don't know why setting PCMA has not worked. Have any guess???

I planned to document this in the wiki.

I was struggling to originate a call from the CLI.
originate openzap/1/1/9952248266 openzap/1/1/9952248266 says "No route.
Aborting".

Have to find a way for this.

I sincerely thank you for your valuable time and effort on this matter.



On Wed, Oct 14, 2009 at 11:33 AM, Michael Collins <msc at freeswitch.org>wrote:

> Lak,
>
> Okay, it stood out right away: FS is trying u-law but Asterisk is trying
> A-law. I'm not sure where the codec for openzap gets selected but you can
> try modifying this line in vars.xml:
>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>
> Try putting PCMA first. Also, I've never tried this but perhaps in your
> dialplan you can set absolute_codec_string to PCMA. I'm very interesting in
> hearing about the results. Please try them as soon as you can and report
> back.
>
> Thanks,
> MC
>
>
> On Tue, Oct 13, 2009 at 10:34 PM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> Hi. I made the setup quickly with one of my colleague, and here is the
>> debug log given by asterisk
>>
>> Steps done by me :
>> 	+ Call Made from 04443902743 to 04439114600
>> 	+ When call lands to 04439114600, I just route it to 09176454982.
>>
>>
>> *CLI>
>> *CLI>
>> *CLI> pri debug span 1
>> Enabled debugging on span 1
>> *CLI>
>> *CLI>
>> *CLI>
>> *CLI> < Protocol Discriminator: Q.931 (8)  len=41
>> < Call Ref: len= 2 (reference 21/0x15) (Originator)
>>
>>
>> < Message type: SETUP (5)
>> < [a1]
>> < Sending Complete (len= 1)
>> < [04 03 80 90 a3]
>> < Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer capability: Speech (0)
>> <                              Ext: 1  Trans mode/rate: 64kbps, circuit-mode (16)
>>
>>
>> <                                User information layer 1: A-Law (35)
>> < [18 03 a1 83 9f]
>> < Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Preferred  Dchan: 0
>> <                        ChanSel: As indicated in following octets
>>
>>
>> <                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
>> <                       Ext: 1  Channel: 31 ]
>> < [6c 0c 01 81 34 34 34 33 39 30 32 37 34 33]
>> < Calling Number (len=14) [ Ext: 0  TON: Unknown Number Type (0)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
>>
>>
>> <                           Presentation: Presentation permitted, user number passed network screening (1)  '4443902743' ]
>> < [70 09 c1 33 39 31 31 34 36 30 30]
>> < Called Number (len=11) [ Ext: 1  TON: Subscriber Number (4)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)  '39114600' ]
>>
>>
>> -- Making new call for cr 21
>> -- Processing Q.931 Call Setup
>> -- Processing IE 161 (cs0, Sending Complete)
>> -- Processing IE 4 (cs0, Bearer Capability)
>> -- Processing IE 24 (cs0, Channel Identification)
>> -- Processing IE 108 (cs0, Calling Party Number)
>>
>>
>> -- Processing IE 112 (cs0, Called Party Number)
>> q931.c:3504 q931_receive: call 21 on channel 31 enters state 6 (Call Present)
>> q931.c:2769 q931_call_proceeding: call 21 on channel 31 enters state 9 (Incoming Call Proceeding)
>>
>> > Protocol Discriminator: Q.931 (8)  len=10
>> > Call Ref: len= 2 (reference 21/0x15) (Terminator)
>> > Message type: CALL PROCEEDING (2)
>>
>> > [18 03 a9 83 9f]
>> > Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  Dchan: 0
>> >                        ChanSel: As indicated in following octets
>>
>> >                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
>> >                       Ext: 1  Channel: 31 ]
>>
>>     -- Accepting call from '4443902743' to '39114600' on channel 0/31, span 1
>>     -- Executing [39114600 at incoming-from-pri:1] NoOp("Zap/31-1", "FMS Call Extn:39114600:| CID::| Channel:Zap/31-1:| ") in new stack
>>
>>
>>     -- Executing [39114600 at incoming-from-pri:2] Dial("Zap/31-1", "Zap/g1/w09176454982||rj") in new stack
>> -- Making new call for cr 32770
>>     -- Requested transfer capability: 0x00 - SPEECH
>>
>> > Protocol Discriminator: Q.931 (8)  len=43
>> > Call Ref: len= 2 (reference 2/0x2) (Originator)
>> > Message type: SETUP (5)
>> > [04 03 80 90 a3]
>> > Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer capability: Speech (0)
>> >                              Ext: 1  Trans mode/rate: 64kbps, circuit-mode (16)
>>
>> >                                User information layer 1: A-Law (35)
>> > [18 03 a9 83 81]
>> > Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  Dchan: 0
>>
>> >                        ChanSel: As indicated in following octets
>> >                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
>> >                       Ext: 1  Channel: 1 ]
>> > [6c 0c 00 81 34 34 34 33 39 30 32 37 34 33]
>> > Calling Number (len=14) [ Ext: 0  TON: Unknown Number Type (0)  NPI: Unknown Number Plan (0)
>>
>> >                           Presentation: Presentation permitted, user number passed network screening (1)  '4443902743' ]
>> > [70 0c 80 30 39 31 37 36 34 35 34 39 38 32]
>> > Called Number (len=14) [ Ext: 1  TON: Unknown Number Type (0)  NPI: Unknown Number Plan (0)  '09176454982' ]
>>
>> q931.c:3087 q931_setup: call 32770 on channel 1 enters state 1 (Call Initiated)
>>
>>
>>     -- Called g1/w09176454982
>> q931.c:2797 q931_alerting: call 21 on channel 31 enters state 7 (Call Received)
>>
>> > Protocol Discriminator: Q.931 (8)  len=9
>> > Call Ref: len= 2 (reference 21/0x15) (Terminator)
>> > Message type: ALERTING (1)
>> > [1e 02 81 88]
>> > Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Private network serving the local user (1)
>> >                               Ext: 1  Progress Description: Inband information or appropriate pattern now available. (8) ]
>>
>> < Protocol Discriminator: Q.931 (8)  len=10
>> < Call Ref: len= 2 (reference 2/0x2) (Terminator)
>> < Message type: CALL PROCEEDING (2)
>> < [18 03 a9 83 81]
>> < Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  Dchan: 0
>>
>>
>> <                        ChanSel: As indicated in following octets
>> <                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
>> <                       Ext: 1  Channel: 1 ]
>> -- Processing IE 24 (cs0, Channel Identification)
>>
>>
>> q931.c:3636 q931_receive: call 32770 on channel 1 enters state 3 (Outgoing call  Proceeding)
>>     -- Zap/1-1 is proceeding passing it to Zap/31-1
>> < Protocol Discriminator: Q.931 (8)  len=13
>> < Call Ref: len= 2 (reference 2/0x2) (Terminator)
>>
>>
>> < Message type: ALERTING (1)
>> < [1e 02 84 88]
>> < Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Public network serving the remote user (4)
>> <                               Ext: 1  Progress Description: Inband information or appropriate pattern now available. (8) ]
>>
>>
>> < [1e 02 84 82]
>> < Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Public network serving the remote user (4)
>> <                               Ext: 1  Progress Description: Called equipment is non-ISDN. (2) ]
>>
>>
>> -- Processing IE 30 (cs0, Progress Indicator)
>> -- Processing IE 30 (cs0, Progress Indicator)
>> q931.c:3549 q931_receive: call 32770 on channel 1 enters state 4 (Call Delivered)
>>     -- Zap/1-1 is ringing
>> < Protocol Discriminator: Q.931 (8)  len=9
>>
>>
>> < Call Ref: len= 2 (reference 2/0x2) (Terminator)
>> < Message type: PROGRESS (3)
>> < [1e 02 84 84]
>> < Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Public network serving the remote user (4)
>>
>>
>> <                               Ext: 1  Progress Description: Unknown (4) ]
>> -- Processing IE 30 (cs0, Progress Indicator)
>>     -- Zap/1-1 is making progress passing it to Zap/31-1
>> < Protocol Discriminator: Q.931 (8)  len=9
>>
>>
>> < Call Ref: len= 2 (reference 2/0x2) (Terminator)
>> < Message type: PROGRESS (3)
>> < [1e 02 84 88]
>> < Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Public network serving the remote user (4)
>>
>>
>> <                               Ext: 1  Progress Description: Inband information or appropriate pattern now available. (8) ]
>> -- Processing IE 30 (cs0, Progress Indicator)
>>     -- Zap/1-1 is making progress passing it to Zap/31-1
>>
>>
>> < Protocol Discriminator: Q.931 (8)  len=19
>> < Call Ref: len= 2 (reference 2/0x2) (Terminator)
>> < Message type: CONNECT (7)
>> < [4c 0c 01 83 39 31 37 36 34 35 34 39 38 32]
>> < Connected Number (len=14) [ Ext: 0  TON: Unknown Number Type (0)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
>>
>>
>> <                             Ext: 1  Presentation: Presentation allowed of network provided number (3)  '9176454982' ]
>> -- Processing IE 76 (cs0, Connected Number)
>> q931.c:3579 q931_receive: call 32770 on channel 1 enters state 10 (Active)
>>
>> > Protocol Discriminator: Q.931 (8)  len=5
>> > Call Ref: len= 2 (reference 2/0x2) (Originator)
>> > Message type: CONNECT ACKNOWLEDGE (15)
>>
>>     -- Zap/1-1 answered Zap/31-1
>> q931.c:2904 q931_connect: call 21 on channel 31 enters state 8 (Connect Request)
>>
>> > Protocol Discriminator: Q.931 (8)  len=14
>>
>> > Call Ref: len= 2 (reference 21/0x15) (Terminator)
>> > Message type: CONNECT (7)
>> > [18 03 a9 83 9f]
>> > Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  Dchan: 0
>> >                        ChanSel: As indicated in following octets
>> >                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
>> >                       Ext: 1  Channel: 31 ]
>> > [1e 02 81 82]
>> > Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Private network serving the local user (1)
>> >                               Ext: 1  Progress Description: Called equipment is non-ISDN. (2) ]
>>
>>     -- Native bridging Zap/31-1 and Zap/1-1
>> < Protocol Discriminator: Q.931 (8)  len=5
>> < Call Ref: len= 2 (reference 21/0x15) (Originator)
>> < Message type: CONNECT ACKNOWLEDGE (15)
>>
>>
>> q931.c:3664 q931_receive: call 21 on channel 31 enters state 10 (Active)
>> < Protocol Discriminator: Q.931 (8)  len=9
>> < Call Ref: len= 2 (reference 2/0x2) (Terminator)
>> < Message type: DISCONNECT (69)
>> < [08 02 80 90]
>>
>>
>> < Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  Location: User (0)
>> <                  Ext: 1  Cause: Normal Clearing (16), class = Normal Event (1) ]
>> -- Processing IE 8 (cs0, Cause)
>>
>> q931.c:3779 q931_receive: call 32770 on channel 1 enters state 12 (Disconnect Indication)
>>
>>     -- Channel 0/1, span 1 got hangup request, cause 16
>> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request
>> q931.c:2920 q931_release: call 32770 on channel 1 enters state 19 (Release Request)
>>
>> > Protocol Discriminator: Q.931 (8)  len=9
>> > Call Ref: len= 2 (reference 2/0x2) (Originator)
>> > Message type: RELEASE (77)
>> > [08 02 81 90]
>> > Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  Location: Private network serving the local user (1)
>> >                  Ext: 1  Cause: Normal Clearing (16), class = Normal Event (1) ]
>>
>>     -- Hungup 'Zap/1-1'
>>   == Spawn extension (incoming-from-pri, 39114600, 2) exited non-zero on 'Zap/31-1'
>>
>>
>>     -- Executing [h at incoming-from-pri:1] Hangup("Zap/31-1", "") in new stack
>>   == Spawn extension (incoming-from-pri, h, 1) exited non-zero on 'Zap/31-1'
>> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Active
>>
>>
>> q931.c:2968 q931_disconnect: call 21 on channel 31 enters state 11 (Disconnect Request)
>>
>> > Protocol Discriminator: Q.931 (8)  len=9
>> > Call Ref: len= 2 (reference 21/0x15) (Terminator)
>>
>> > Message type: DISCONNECT (69)
>> > [08 02 81 90]
>> > Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  Location: Private network serving the local user (1)
>>
>> >                  Ext: 1  Cause: Normal Clearing (16), class = Normal Event (1) ]
>>
>>     -- Hungup 'Zap/31-1'
>> < Protocol Discriminator: Q.931 (8)  len=5
>>
>> < Call Ref: len= 2 (reference 2/0x2) (Terminator)
>> < Message type: RELEASE COMPLETE (90)
>> q931.c:3719 q931_receive: call 32770 on channel 1 enters state 0 (Null)
>> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
>>
>>
>> NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
>> < Protocol Discriminator: Q.931 (8)  len=5
>> < Call Ref: len= 2 (reference 21/0x15) (Originator)
>> < Message type: RELEASE (77)
>> q931.c:3754 q931_receive: call 21 on channel 31 enters state 0 (Null)
>>
>>
>> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request
>>
>> > Protocol Discriminator: Q.931 (8)  len=9
>> > Call Ref: len= 2 (reference 21/0x15) (Terminator)
>>
>> > Message type: RELEASE COMPLETE (90)
>> > [08 02 81 90]
>> > Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  Location: Private network serving the local user (1)
>>
>> >                  Ext: 1  Cause: Normal Clearing (16), class = Normal Event (1) ]
>>
>> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
>>
>> NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
>>
>>
>> On Tue, Oct 13, 2009 at 9:16 PM, Michael S Collins <msc at freeswitch.org>wrote:
>>
>>>
>>> On Oct 13, 2009, at 6:45 AM, lakshmanan ganapathy <lakindia89 at gmail.com>
>>> wrote:
>>>
>>> We are using Reliance as the Carrier.
>>> I think, with this same Reliance carrier, in my office, they are able to
>>> make outgoing calls through asterisk+libpri.
>>>
>>>
>>> If that's the case I would be very interested in seeing a pri debug from
>>> a working call on an asterisk box. It might give us a clue as to what is not
>>> working. -MC
>>>
>>>
>>>
>>> On Tue, Oct 13, 2009 at 12:19 AM, Michael Collins < <msc at freeswitch.org>
>>> msc at freeswitch.org> wrote:
>>>
>>>> Lak,
>>>>
>>>> Okay I will need a little bit of time to dig into the IE's and what they
>>>> contain. In the meantime can you tell me who the carrier is? I'd like to
>>>> find out if they have some specific requirements. The fact that it doesn't
>>>> work with libpri surprises me because that would mean that Asterisk systems
>>>> would probably not work with this carrier as well.
>>>>
>>>> BTW, thanks for the very complete pastebin entries. :)
>>>> -MC
>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From nandy1925 at gmail.com  Wed Oct 14 03:41:59 2009
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Wed, 14 Oct 2009 18:41:59 +0800
Subject: [Freeswitch-users] Preannouncing a message before making a call
In-Reply-To: <20091013205114.GA4927@mad06.wl0.org>
References: <20091013205114.GA4927@mad06.wl0.org>
Message-ID: <7d0bfd8c0910140341l1ee4502fiaaff6c5087666048@mail.gmail.com>

hi simon, implementation is almost the same. here's my dialplan:

<extension name="say_destination_info">
 <condition field="destination_number" expression="^0044(\d)$">
   <action application="answer"/>
   <action application="playback" data="misc/dialing.wav"/>
   <action application="say" data="en name_spelled iterated $1"/>
   ... add more files to play here ...
   <action application="playback" data="misc/gateway.wav"/>
   <action application="bridge" data="sofia/gateway/$1/>
 </condition>
</extension>

warning: i have not tested this. i'm just showing for illustration purposes.

-nandy

On Wed, Oct 14, 2009 at 4:51 AM, Simon J Mudd <sjmudd at pobox.com> wrote:

> I'm looking to migrate to Freeswitch from Asterisk (SOHO usage).
> One thing I have configured at the moment which I'm unsure how to do in
> FreeSwitch is the following.
>
> When the call's dial pattern is recognised I've been getting Asterisk
> to say: "Dialing <destination> via <gateway>" where <destination> might
> be a country, or area and <gateway> may be the name of one of several
> SIP gateways I have configured.  Each phrase has been a prerecorded wav
> file played in sequence.
>
> [macro-XXXXX]
> exten => s,1,Playback(/etc/asterisk/wav/dialing)
> exten => s,2,Playback(/etc/asterisk/wav/${ARG2})    # Name of destination
> exten => s,3,Playback(/etc/asterisk/wav/via)
> exten => s,4,Playback(/etc/asterisk/wav/XXXXX)      # Gateway name
> exten => s,5,Dial(SIP/XXXXX/${ARG1},60,tT)          # ARG1 = Destination
> number
>
>
> # actual usage
> exten => _0044.,1,Macro(XXXXX,${EXTEN},touk)
>
> How would I approach doing something like this in FreeSwtich?
>
> Thanks for any pointers.
>
> Simon
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From Prometheus001 at gmx.net  Wed Oct 14 03:55:58 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Wed, 14 Oct 2009 12:55:58 +0200
Subject: [Freeswitch-users] T.38 via UPDATE request
Message-ID: <4AD5AE3E.9070206@gmx.net>

Hello,

we have the following problem.
2 Fax machines are communicating via Freeswitch. One is externally
attached via a Telco who is able to handle T.38. The other one is
attached locally.

When 2 Fax machines start syncing each other, the Telco sends a SIP
UPDATE message with T.38 SDP, as it detects fax during the fax negociations.
Freeswitch answers with an SIP OK message back to the telco, and I can
see the T.38 SDP on the debug console of freeswitch.
Then nothing happens any more until one of fax machines detects timeout.

We have set proxy-media to true. However is was done during call setup
when both machines communicated with G711 SDP.
The UPDATE message was commited by FS to the telco, but was not sent to
the other fax, so I think in this case Freeswitch is supposed to
transcode between T.38 and G711 which it cannot do, as we know.

How can I overcome this scenario? Is this a defect, should freeswitch
send the UPDATE message to the other fax?  Or is there a workaround?

Best regards
Peter

Here's the UPDATE message:
UPDATE sip:mod_sofia at 82.115.xx.xxx:5080 SIP/2.0.
Call-ID: 5283fe4e-334f-122d-d1b9-001517956764.
Contact: <sip:212.91.xxx.xxx:5060>.
Content-Type: application/sdp.
CSeq: 16340063 UPDATE.
From: <sip:06912345678 at sip.telco.de>;tag=00-08135-017e041a-4d21f6037.
Max-Forwards: 31.
Route: <sip:82.115.96.165;lr;ftag=pH663F4S02erm>.
To: "030987654321" <sip:030987654321 at 82.115.xx.xxx>;tag=pH663F4S02erm.
User-Agent: Cirpack/v4.41e (gw_sip).
Via: SIP/2.0/UDP 212.91.xxx.xxx:5060;branch=z9hG4bK-178C-1923B31.
Content-Length: 300.
.
v=0.
o=cp10 125551618103 125551618105 IN IP4 212.91.xxx.xx.
s=SIP Call.
c=IN IP4 212.91.xxx.xx.
t=0 0.
m=image 6860 udptl t38.
a=sendrecv.
a=T38FaxVersion:0.
a=T38MaxBitRate:9600.
a=T38FaxRateManagement:transferredTCF.
a=T38FaxMaxBuffer:1000.
a=T38FaxMaxDatagram:200.
a=T38FaxUdpEC:t38UDPRedundancy.



From srinivas.ksvreddy at gmail.com  Wed Oct 14 04:58:49 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Wed, 14 Oct 2009 17:28:49 +0530
Subject: [Freeswitch-users] adding new extension
Message-ID: <f8af5740910140458w4666ee2fwa93d1ecb70cc015a@mail.gmail.com>

Hi,

can any please let me know, how to add new extension(eg 1000.xml)
dynamically while running freeswitch?
while running freeswitch i have created new xml(eg: 1500.xml) file and i
have changed freeswitch.xml.fsxml still  not working,

Thanks
Srinivasula Reddy K
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From mgg at giagnocavo.net  Wed Oct 14 05:43:32 2009
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Wed, 14 Oct 2009 08:43:32 -0400
Subject: [Freeswitch-users] Some help with my post-paid billing project
In-Reply-To: <b16156850910140055r37581014iddf52d915aa7eaa4@mail.gmail.com>
References: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
	<74730C11-75D0-4BC0-9F63-8B6C9F0F499F@me.com>
	<86a32abc0910131836w5f113a16s247caf5eeb0f8fe4@mail.gmail.com>
	<86a32abc0910131853v244ed61bof9c24f345a0944b2@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C6703241D1E52@mse17be1.mse17.exchange.ms>
	<b16156850910140055r37581014iddf52d915aa7eaa4@mail.gmail.com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C6703241D1E8A@mse17be1.mse17.exchange.ms>

Wouldn?t that be SQL ?IN? instead of LIKE?

First off, see if you can do this in memory. If it?s just a ?rate sheet? list of prefixes + billing, it?s probably not that much data. I?d guess maybe 64-128 bytes per record (in Ruby it?s probably more, maybe 256 bytes?) inside a hashtable. Even so, we?re only talking about 1.5 to 3GB of RAM. Stick it in a nice performing dictionary of some sort and you?re set.

Just as a quick test, on one core of a Core2, it takes about 150ms to do 100K lookups against a 1M item hashtable (F#) ? that?s just a 1 line loop I tried out with no optimization, doing about 650K/sec. For comparison on the same machine, looping on a simple ?SELECT @@VERSION? command only achieved 5000 executions/second on a single thread ? that?s just the SQL overhead (to a local SQL instance!).

If the memory usage is too excessive, consider caching only subsets of the prefixes. Keep track of which leading prefix (1 or 2 digits) are the most used, and keep those in memory. Have a reasonably fast DB to fall back to.

If you?re sure the storage requirements or the change frequency truly eliminates keeping it in your own memory, here are some suggestions for working on DB performance:

A little perf gain is to pre-calculate possible lengths, so you?re performing less lookups. Just have a table based on the first 2 or 3 digits, giving you the valid prefix lengths. That can easily cut down the number of lookups from 10-12 to, say, 3-4. Compare the perf of temp tables, CTEs, and dynamic SQL ?IN?.

But I would be hesitant to do this from a higher level ORM or via dynamic SQL queries:

-          You?ll have a much harder time doing DB-specific optimizations/hints

-          Dynamic SQL needs to be parsed each time (possibly minor issue)

-          If the ad hoc queries are not properly parameterized, query plans might not get reused very well

-          You can easily send many times more data to the server (your entire SQL, versus just the sproc name + parameters)

Really look into doing programmability on the server. There?s all sorts of things you can do to minimize what you need to send to the database, and let the DB engine figure out the most efficient way. You might want to use a product that?s a bit more advanced than MySQL.

And finally, test, test, test. You should have a nice benchmark with several million rows and be able to execute, say, 10-50K lookups (on multiple threads) after each change. Modify your query, execute it, _inspect the execution plan_. Check your indexes ? make sure they are covering the query in the right way. (For example, if the table is (ID, Prefix, Price), you want an index of (Prefix, Price).) Review possible hints (for example, SQL Server usually needs a hint to choose a indexed view). Make sure your transaction level is set appropriately (can you do the query without locking?). Can you batch up several numbers in one command, saving on the per-command overhead?

Also consider it in light of the rest of your application. If everything is in the same database, then you can probably do the entire rate call/save cdr/update balances in a single command and transaction.

For us, our initial draft of routing inside SQL Server performed well under 100 calls/sec ? not usable. After a few days of playing with things, it went over 1000 calls/sec, which is comfortable enough for a $500 server. This is matching 100 gateways and dialplans with several thousand entries per dialplan, across 4 million+ routes, doing QBR/LCR in the process.

-Michael


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Muhammad Shahzad
Sent: Wednesday, October 14, 2009 1:55 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Some help with my post-paid billing project

I fully agree that direct matching is much faster then pattern matching in SQL.

One of my clients had same problem, he had around 12 million number prefixes in a table and during each call an AGI script use to query that table to find longest prefix match, but this use to take like 3-5 seconds even with indexed columns. So after a lots of R & D we come up with following logic,

1. We break the destination number length-wise, e.g. suppose number is 923344224088 then length chunks would be,

923344224088, 92334422408, 9233442240, 923344224, 92334422, 9233442, 923344, 92334, 9233, 923, 92, 9

2. Then use SQL LIKE function in WHERE clause (you can also use SQL OR function if your DBMS doesn't support SQL LIKE function), and pass all these chunks to it, e.g.

WHERE prefix LIKE (923344224088, 92334422408, 9233442240, 923344224, 92334422, 9233442, 923344, 92334, 9233, 923, 92, 9)

3. Lastly we ORDER the result by prefix length, e.g.

ORDER BY LENGTH(prefix) DESC LIMIT 1

4. The complete query will be,

SELECT * FROM prefixes
WHERE prefix LIKE (923344224088, 92334422408, 9233442240, 923344224, 92334422, 9233442, 923344, 92334, 9233, 923, 92, 9)
ORDER BY LENGTH(prefix) DESC LIMIT 1

Now the query takes less then 150 ms to execute. :-)

Here is an STL method that can generate this query, i am sure you can convert it to any programming language of your choice easily.

=========================================================
std::string GetQuery(std::string destination) {
        std::string query = "SELECT * FROM prefixes WHERE prefix LIKE ('" + destination;

        for(int i=1; i<destination.length(); i++) {
                query += "','" + destination.substr(0, (i * -1));

        }

        query += "') ORDER BY LENGTH(prefix) DESC LIMIT 1";
        return query;
}
=========================================================

I am pretty sure this query is 100% ANSI SQL compatible (http://en.wikipedia.org/wiki/SQL).

Thank you.
On Wed, Oct 14, 2009 at 10:15 AM, Michael Giagnocavo <mgg at giagnocavo.net<mailto:mgg at giagnocavo.net>> wrote:
In our testing with SQL Server, we found that executing several queries for direct matches yielded far better performance than one query trying to check prefixes. (The column was also part of the clustered index, but AFAIK MySQL doesn?t support defining your own clustered indexes; you get the PK always.)

-Michael

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Diego Viola
Sent: Tuesday, October 13, 2009 7:54 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Some help with my post-paid billing project

Wrong question.

Is there a way to compare numbers with prefixes without using the prefix module?

Diego
On Wed, Oct 14, 2009 at 1:36 AM, Diego Viola <diego.viola at gmail.com<mailto:diego.viola at gmail.com>> wrote:
I'm using MySQL now but I will try PostgreSQL with the prefix module, is there a way to do that without the prefix module and with regular SQL?

Any examples?

Diego

On Tue, Oct 13, 2009 at 10:45 PM, Even Andr? Fiskvik <grevenx at me.com<mailto:grevenx at me.com>> wrote:
What database are you using?
You could do this with regular SQL, but it would by a costly operation,
for PostgreSQL we're using the prefix module: http://pgfoundry.org/projects/prefix/

You can then match the closest prefix by using something like
"WHERE myprefix_col @> caller_destination_number ORDER BY LENGTH
(myprefix_col::text) LIMIT 1;"


Best regards,
Even Andr?


On 13. okt. 2009, at 23.53, Diego Viola wrote:

> Hello,
>
> I'm trying to write a post-paid billing script, I have the CDR on my
> database and also a "rates" table, the CDR contains fields like
> caller_destination_number, variable_duration, etc. and the rates
> table contains: destination, prefix, rate (cost).
>
> The problem is that I can't just strip the destination number to
> take the prefix from it because I have to deal with destination
> numbers from different countries and they all have different prefix
> lengths... so I need to find another way to take the prefix from the
> destination number.
>
> Any ideas how to do this?
>
> Thanks,
>
> Diego
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> users
> http://www.freeswitch.org


_______________________________________________
FreeSWITCH-users mailing list
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http://www.freeswitch.org



_______________________________________________
FreeSWITCH-users mailing list
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--
________________________________________________________
|                                                                                                |
| FATAL ERROR                                                           --- O X |
|_______________________________________________________|
|                        You have moved the mouse.                            |
| Windows must be restarted for the changes to take effect.   |
|                                        <OK>                                              |
####################################/


Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com<mailto:shari_786pk at hotmail.com>
Email: shaheryarkh at googlemail.com<mailto:shaheryarkh at googlemail.com>
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From Russell.Mosemann at cune.org  Wed Oct 14 06:21:18 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Wed, 14 Oct 2009 13:21:18 -0000
Subject: [Freeswitch-users] openzap Failure opening channel error
In-Reply-To: <7d79b3930910140303k4bff6d5fgbc55a173194e22c9@mail.gmail.com>
Message-ID: <20091014132118.3A34C403DBC@mail.cune.org>

lakshmanan ganapathy <lakindia89 at gmail.com> said:

> I was struggling to originate a call from the CLI.
> originate openzap/1/1/9952248266 openzap/1/1/9952248266 says "No route.
> Aborting".

The syntax is incorrect.

originate openzap/1/1/9952248266 <extension>

For example,

openzap/1/1/9952248266 1234

See the wiki for details about originate. I think I included a link in a
previous message.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!




From shaheryarkh at googlemail.com  Wed Oct 14 06:22:51 2009
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Wed, 14 Oct 2009 19:22:51 +0600
Subject: [Freeswitch-users] Some help with my post-paid billing project
In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6703241D1E8A@mse17be1.mse17.exchange.ms>
References: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
	<74730C11-75D0-4BC0-9F63-8B6C9F0F499F@me.com>
	<86a32abc0910131836w5f113a16s247caf5eeb0f8fe4@mail.gmail.com>
	<86a32abc0910131853v244ed61bof9c24f345a0944b2@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C6703241D1E52@mse17be1.mse17.exchange.ms>
	<b16156850910140055r37581014iddf52d915aa7eaa4@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C6703241D1E8A@mse17be1.mse17.exchange.ms>
Message-ID: <b16156850910140622w78b356b4hd7280bc504416685@mail.gmail.com>

Oops, you are right, this is what happens when you are doing more then one
thing at a time, i was writing a db function while replying in this email
thread and confused LIKE with IN. There is one more mistake in my email if
any one noticed, that is, string class in STL does not support negative
length in method substr.

Thanks for correction.


On Wed, Oct 14, 2009 at 6:43 PM, Michael Giagnocavo <mgg at giagnocavo.net>wrote:

>  Wouldn?t that be SQL ?IN? instead of LIKE?
>
>
>
> First off, see if you can do this in memory. If it?s just a ?rate sheet?
> list of prefixes + billing, it?s probably not that much data. I?d guess
> maybe 64-128 bytes per record (in Ruby it?s probably more, maybe 256 bytes?)
> inside a hashtable. Even so, we?re only talking about 1.5 to 3GB of RAM.
> Stick it in a nice performing dictionary of some sort and you?re set.
>
>
>
> Just as a quick test, on one core of a Core2, it takes about 150ms to do
> 100K lookups against a 1M item hashtable (F#) ? that?s just a 1 line loop I
> tried out with no optimization, doing about 650K/sec. For comparison on the
> same machine, looping on a simple ?SELECT @@VERSION? command only achieved
> 5000 executions/second on a single thread ? that?s just the SQL overhead (to
> a local SQL instance!).
>
>
>
> If the memory usage is too excessive, consider caching only subsets of the
> prefixes. Keep track of which leading prefix (1 or 2 digits) are the most
> used, and keep those in memory. Have a reasonably fast DB to fall back to.
>
>
>
> If you?re sure the storage requirements or the change frequency truly
> eliminates keeping it in your own memory, here are some suggestions for
> working on DB performance:
>
>
>
> A little perf gain is to pre-calculate possible lengths, so you?re
> performing less lookups. Just have a table based on the first 2 or 3 digits,
> giving you the valid prefix lengths. That can easily cut down the number of
> lookups from 10-12 to, say, 3-4. Compare the perf of temp tables, CTEs, and
> dynamic SQL ?IN?.
>
>
>
> But I would be hesitant to do this from a higher level ORM or via dynamic
> SQL queries:
>
> -          You?ll have a much harder time doing DB-specific
> optimizations/hints
>
> -          Dynamic SQL needs to be parsed each time (possibly minor issue)
>
> -          If the ad hoc queries are not properly parameterized, query
> plans might not get reused very well
>
> -          You can easily send many times more data to the server (your
> entire SQL, versus just the sproc name + parameters)
>
>
>
> Really look into doing programmability on the server. There?s all sorts of
> things you can do to minimize what you need to send to the database, and let
> the DB engine figure out the most efficient way. You might want to use a
> product that?s a bit more advanced than MySQL.
>
>
>
> And finally, test, test, test. You should have a nice benchmark with
> several million rows and be able to execute, say, 10-50K lookups (on
> multiple threads) after each change. Modify your query, execute it, _*inspect
> the execution plan*_. Check your indexes ? make sure they are covering the
> query in the right way. (For example, if the table is (ID, Prefix, Price),
> you want an index of (Prefix, Price).) Review possible hints (for example,
> SQL Server usually needs a hint to choose a indexed view). Make sure your
> transaction level is set appropriately (can you do the query without
> locking?). Can you batch up several numbers in one command, saving on the
> per-command overhead?
>
>
>
> Also consider it in light of the rest of your application. If everything is
> in the same database, then you can probably do the entire rate call/save
> cdr/update balances in a single command and transaction.
>
>
>
> For us, our initial draft of routing inside SQL Server performed well under
> 100 calls/sec ? not usable. After a few days of playing with things, it went
> over 1000 calls/sec, which is comfortable enough for a $500 server. This is
> matching 100 gateways and dialplans with several thousand entries per
> dialplan, across 4 million+ routes, doing QBR/LCR in the process.
>
>
>
> -Michael
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Muhammad
> Shahzad
> *Sent:* Wednesday, October 14, 2009 1:55 AM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Some help with my post-paid billing
> project
>
>
>
> I fully agree that direct matching is much faster then pattern matching in
> SQL.
>
> One of my clients had same problem, he had around 12 million number
> prefixes in a table and during each call an AGI script use to query that
> table to find longest prefix match, but this use to take like 3-5 seconds
> even with indexed columns. So after a lots of R & D we come up with
> following logic,
>
> 1. We break the destination number length-wise, e.g. suppose number is
> 923344224088 then length chunks would be,
>
> 923344224088, 92334422408, 9233442240, 923344224, 92334422, 9233442,
> 923344, 92334, 9233, 923, 92, 9
>
> 2. Then use SQL LIKE function in WHERE clause (you can also use SQL OR
> function if your DBMS doesn't support SQL LIKE function), and pass all these
> chunks to it, e.g.
>
> WHERE prefix LIKE (923344224088, 92334422408, 9233442240, 923344224,
> 92334422, 9233442, 923344, 92334, 9233, 923, 92, 9)
>
> 3. Lastly we ORDER the result by prefix length, e.g.
>
> ORDER BY LENGTH(prefix) DESC LIMIT 1
>
> 4. The complete query will be,
>
> SELECT * FROM prefixes
> WHERE prefix LIKE (923344224088, 92334422408, 9233442240, 923344224,
> 92334422, 9233442, 923344, 92334, 9233, 923, 92, 9)
> ORDER BY LENGTH(prefix) DESC LIMIT 1
>
> Now the query takes less then 150 ms to execute. :-)
>
> Here is an STL method that can generate this query, i am sure you can
> convert it to any programming language of your choice easily.
>
> =========================================================
> *std::string GetQuery(std::string destination) {
>         std::string query = "SELECT * FROM prefixes WHERE prefix LIKE ('" +
> destination;
>
>         for(int i=1; i<destination.length(); i++) {
>                 query += "','" + destination.substr(0, (i * -1));
>
>         }
>
>         query += "') ORDER BY LENGTH(prefix) DESC LIMIT 1";
>         return query;
> }
> *=========================================================
>
> I am pretty sure this query is 100% ANSI SQL compatible (
> http://en.wikipedia.org/wiki/SQL).
>
> Thank you.
>
> On Wed, Oct 14, 2009 at 10:15 AM, Michael Giagnocavo <mgg at giagnocavo.net>
> wrote:
>
> In our testing with SQL Server, we found that executing several queries for
> direct matches yielded far better performance than one query trying to check
> prefixes. (The column was also part of the clustered index, but AFAIK MySQL
> doesn?t support defining your own clustered indexes; you get the PK always.)
>
>
>
> -Michael
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Diego Viola
> *Sent:* Tuesday, October 13, 2009 7:54 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Some help with my post-paid billing
> project
>
>
>
> Wrong question.
>
> Is there a way to compare numbers with prefixes without using the prefix
> module?
>
> Diego
>
> On Wed, Oct 14, 2009 at 1:36 AM, Diego Viola <diego.viola at gmail.com>
> wrote:
>
> I'm using MySQL now but I will try PostgreSQL with the prefix module, is
> there a way to do that without the prefix module and with regular SQL?
>
> Any examples?
>
> Diego
>
>
>
> On Tue, Oct 13, 2009 at 10:45 PM, Even Andr? Fiskvik <grevenx at me.com>
> wrote:
>
> What database are you using?
> You could do this with regular SQL, but it would by a costly operation,
> for PostgreSQL we're using the prefix module:
> http://pgfoundry.org/projects/prefix/
>
> You can then match the closest prefix by using something like
> "WHERE myprefix_col @> caller_destination_number ORDER BY LENGTH
> (myprefix_col::text) LIMIT 1;"
>
>
> Best regards,
> Even Andr?
>
>
>
> On 13. okt. 2009, at 23.53, Diego Viola wrote:
>
> > Hello,
> >
> > I'm trying to write a post-paid billing script, I have the CDR on my
> > database and also a "rates" table, the CDR contains fields like
> > caller_destination_number, variable_duration, etc. and the rates
> > table contains: destination, prefix, rate (cost).
> >
> > The problem is that I can't just strip the destination number to
> > take the prefix from it because I have to deal with destination
> > numbers from different countries and they all have different prefix
> > lengths... so I need to find another way to take the prefix from the
> > destination number.
> >
> > Any ideas how to do this?
> >
> > Thanks,
> >
> > Diego
> >
>
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> ________________________________________________________
> |
>                      |
> | FATAL ERROR                                                           ---
> O X |
> |_______________________________________________________|
> |                        You have moved the mouse.
>    |
> | Windows must be restarted for the changes to take effect.   |
> |                                        <OK>
>                |
> ####################################/
>
>
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
________________________________________________________
|
                     |
| FATAL ERROR                                                           ---
O X |
|_______________________________________________________|
|                        You have moved the mouse.
 |
| Windows must be restarted for the changes to take effect.   |
|                                        <OK>
               |
####################################/


Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From rupa at rupa.com  Wed Oct 14 06:36:07 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 14 Oct 2009 08:36:07 -0500
Subject: [Freeswitch-users] Some help with my post-paid billing project
In-Reply-To: <b16156850910140622w78b356b4hd7280bc504416685@mail.gmail.com>
References: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
	<74730C11-75D0-4BC0-9F63-8B6C9F0F499F@me.com>
	<86a32abc0910131836w5f113a16s247caf5eeb0f8fe4@mail.gmail.com>
	<86a32abc0910131853v244ed61bof9c24f345a0944b2@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C6703241D1E52@mse17be1.mse17.exchange.ms>
	<b16156850910140055r37581014iddf52d915aa7eaa4@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C6703241D1E8A@mse17be1.mse17.exchange.ms>
	<b16156850910140622w78b356b4hd7280bc504416685@mail.gmail.com>
Message-ID: <b39273530910140636j72652456ga0ace41468b5c546@mail.gmail.com>

I would still suggest using mod_lcr for this...  If you have any real
volume, use postgresql with the prefix module.

It also supports IN lists, OR lists, optional quoting (since mysql is
retarded), and custom sql so you can interface with whatever stored
proc or deal with whatever database table you may need to support.

It also supports multiple profiles (so you can charge different
amounts based on some other criteria like account code) and...
interstate/intrastate/intralata rating.  probably other stuff I forgot
to mention...

Let me just recommend to reuse what already exists and improve on that.

On Wed, Oct 14, 2009 at 8:22 AM, Muhammad Shahzad
<shaheryarkh at googlemail.com> wrote:
> Oops, you are right, this is what happens when you are doing more then one
> thing at a time, i was writing a db function while replying in this email
> thread and confused LIKE with IN. There is one more mistake in my email if
> any one noticed, that is, string class in STL does not support negative
> length in method substr.
>
> Thanks for correction.
>
>
> On Wed, Oct 14, 2009 at 6:43 PM, Michael Giagnocavo <mgg at giagnocavo.net>
> wrote:
>>
>> Wouldn?t that be SQL ?IN? instead of LIKE?
>>
>>
>>
>> First off, see if you can do this in memory. If it?s just a ?rate sheet?
>> list of prefixes + billing, it?s probably not that much data. I?d guess
>> maybe 64-128 bytes per record (in Ruby it?s probably more, maybe 256 bytes?)
>> inside a hashtable. Even so, we?re only talking about 1.5 to 3GB of RAM.
>> Stick it in a nice performing dictionary of some sort and you?re set.
>>
>>
>>
>> Just as a quick test, on one core of a Core2, it takes about 150ms to do
>> 100K lookups against a 1M item hashtable (F#) ? that?s just a 1 line loop I
>> tried out with no optimization, doing about 650K/sec. For comparison on the
>> same machine, looping on a simple ?SELECT @@VERSION? command only achieved
>> 5000 executions/second on a single thread ? that?s just the SQL overhead (to
>> a local SQL instance!).
>>
>>
>>
>> If the memory usage is too excessive, consider caching only subsets of the
>> prefixes. Keep track of which leading prefix (1 or 2 digits) are the most
>> used, and keep those in memory. Have a reasonably fast DB to fall back to.
>>
>>
>>
>> If you?re sure the storage requirements or the change frequency truly
>> eliminates keeping it in your own memory, here are some suggestions for
>> working on DB performance:
>>
>>
>>
>> A little perf gain is to pre-calculate possible lengths, so you?re
>> performing less lookups. Just have a table based on the first 2 or 3 digits,
>> giving you the valid prefix lengths. That can easily cut down the number of
>> lookups from 10-12 to, say, 3-4. Compare the perf of temp tables, CTEs, and
>> dynamic SQL ?IN?.
>>
>>
>>
>> But I would be hesitant to do this from a higher level ORM or via dynamic
>> SQL queries:
>>
>> -????????? You?ll have a much harder time doing DB-specific
>> optimizations/hints
>>
>> -????????? Dynamic SQL needs to be parsed each time (possibly minor issue)
>>
>> -????????? If the ad hoc queries are not properly parameterized, query
>> plans might not get reused very well
>>
>> -????????? You can easily send many times more data to the server (your
>> entire SQL, versus just the sproc name + parameters)
>>
>>
>>
>> Really look into doing programmability on the server. There?s all sorts of
>> things you can do to minimize what you need to send to the database, and let
>> the DB engine figure out the most efficient way. You might want to use a
>> product that?s a bit more advanced than MySQL.
>>
>>
>>
>> And finally, test, test, test. You should have a nice benchmark with
>> several million rows and be able to execute, say, 10-50K lookups (on
>> multiple threads) after each change. Modify your query, execute it, _inspect
>> the execution plan_. Check your indexes ? make sure they are covering the
>> query in the right way. (For example, if the table is (ID, Prefix, Price),
>> you want an index of (Prefix, Price).) Review possible hints (for example,
>> SQL Server usually needs a hint to choose a indexed view). Make sure your
>> transaction level is set appropriately (can you do the query without
>> locking?). Can you batch up several numbers in one command, saving on the
>> per-command overhead?
>>
>>
>>
>> Also consider it in light of the rest of your application. If everything
>> is in the same database, then you can probably do the entire rate call/save
>> cdr/update balances in a single command and transaction.
>>
>>
>>
>> For us, our initial draft of routing inside SQL Server performed well
>> under 100 calls/sec ? not usable. After a few days of playing with things,
>> it went over 1000 calls/sec, which is comfortable enough for a $500 server.
>> This is matching 100 gateways and dialplans with several thousand entries
>> per dialplan, across 4 million+ routes, doing QBR/LCR in the process.
>>
>>
>>
>> -Michael
>>
>>
>>
>>
>>
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Muhammad
>> Shahzad
>> Sent: Wednesday, October 14, 2009 1:55 AM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Some help with my post-paid billing
>> project
>>
>>
>>
>> I fully agree that direct matching is much faster then pattern matching in
>> SQL.
>>
>> One of my clients had same problem, he had around 12 million number
>> prefixes in a table and during each call an AGI script use to query that
>> table to find longest prefix match, but this use to take like 3-5 seconds
>> even with indexed columns. So after a lots of R & D we come up with
>> following logic,
>>
>> 1. We break the destination number length-wise, e.g. suppose number is
>> 923344224088 then length chunks would be,
>>
>> 923344224088, 92334422408, 9233442240, 923344224, 92334422, 9233442,
>> 923344, 92334, 9233, 923, 92, 9
>>
>> 2. Then use SQL LIKE function in WHERE clause (you can also use SQL OR
>> function if your DBMS doesn't support SQL LIKE function), and pass all these
>> chunks to it, e.g.
>>
>> WHERE prefix LIKE (923344224088, 92334422408, 9233442240, 923344224,
>> 92334422, 9233442, 923344, 92334, 9233, 923, 92, 9)
>>
>> 3. Lastly we ORDER the result by prefix length, e.g.
>>
>> ORDER BY LENGTH(prefix) DESC LIMIT 1
>>
>> 4. The complete query will be,
>>
>> SELECT * FROM prefixes
>> WHERE prefix LIKE (923344224088, 92334422408, 9233442240, 923344224,
>> 92334422, 9233442, 923344, 92334, 9233, 923, 92, 9)
>> ORDER BY LENGTH(prefix) DESC LIMIT 1
>>
>> Now the query takes less then 150 ms to execute. :-)
>>
>> Here is an STL method that can generate this query, i am sure you can
>> convert it to any programming language of your choice easily.
>>
>> =========================================================
>> std::string GetQuery(std::string destination) {
>> ??????? std::string query = "SELECT * FROM prefixes WHERE prefix LIKE ('"
>> + destination;
>>
>> ??????? for(int i=1; i<destination.length(); i++) {
>> ??????????????? query += "','" + destination.substr(0, (i * -1));
>>
>> ??????? }
>>
>> ??????? query += "') ORDER BY LENGTH(prefix) DESC LIMIT 1";
>> ??????? return query;
>> }
>> =========================================================
>>
>> I am pretty sure this query is 100% ANSI SQL compatible
>> (http://en.wikipedia.org/wiki/SQL).
>>
>> Thank you.
>>
>> On Wed, Oct 14, 2009 at 10:15 AM, Michael Giagnocavo <mgg at giagnocavo.net>
>> wrote:
>>
>> In our testing with SQL Server, we found that executing several queries
>> for direct matches yielded far better performance than one query trying to
>> check prefixes. (The column was also part of the clustered index, but AFAIK
>> MySQL doesn?t support defining your own clustered indexes; you get the PK
>> always.)
>>
>>
>>
>> -Michael
>>
>>
>>
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Diego
>> Viola
>> Sent: Tuesday, October 13, 2009 7:54 PM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Some help with my post-paid billing
>> project
>>
>>
>>
>> Wrong question.
>>
>> Is there a way to compare numbers with prefixes without using the prefix
>> module?
>>
>> Diego
>>
>> On Wed, Oct 14, 2009 at 1:36 AM, Diego Viola <diego.viola at gmail.com>
>> wrote:
>>
>> I'm using MySQL now but I will try PostgreSQL with the prefix module, is
>> there a way to do that without the prefix module and with regular SQL?
>>
>> Any examples?
>>
>> Diego
>>
>>
>>
>> On Tue, Oct 13, 2009 at 10:45 PM, Even Andr? Fiskvik <grevenx at me.com>
>> wrote:
>>
>> What database are you using?
>> You could do this with regular SQL, but it would by a costly operation,
>> for PostgreSQL we're using the prefix module:
>> http://pgfoundry.org/projects/prefix/
>>
>> You can then match the closest prefix by using something like
>> "WHERE myprefix_col @> caller_destination_number ORDER BY LENGTH
>> (myprefix_col::text) LIMIT 1;"
>>
>>
>> Best regards,
>> Even Andr?
>>
>> On 13. okt. 2009, at 23.53, Diego Viola wrote:
>>
>> > Hello,
>> >
>> > I'm trying to write a post-paid billing script, I have the CDR on my
>> > database and also a "rates" table, the CDR contains fields like
>> > caller_destination_number, variable_duration, etc. and the rates
>> > table contains: destination, prefix, rate (cost).
>> >
>> > The problem is that I can't just strip the destination number to
>> > take the prefix from it because I have to deal with destination
>> > numbers from different countries and they all have different prefix
>> > lengths... so I need to find another way to take the prefix from the
>> > destination number.
>> >
>> > Any ideas how to do this?
>> >
>> > Thanks,
>> >
>> > Diego
>> >
>>
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> > users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> --
>> ________________________________________________________
>> |
>> ? ? ? ? ? ? ? ? ? ? ? ?|
>> | FATAL ERROR
>> --- O X |
>> |_______________________________________________________|
>> | ? ? ? ? ? ? ? ? ? ? ? ?You have moved the mouse.
>> ? ?|
>> | Windows must be restarted for the changes to take effect. ? |
>> | ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?<OK>
>> ? ? ? ? ? ? ? ? ?|
>> ####################################/
>>
>>
>> Muhammad Shahzad
>> -----------------------------------
>> CISCO Rich Media Communication Specialist (CRMCS)
>> CISCO Certified Network Associate (CCNA)
>> Cell: +92 334 422 40 88
>> MSN: shari_786pk at hotmail.com
>> Email: shaheryarkh at googlemail.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> ________________________________________________________
> |
> ? ? ? ? ? ? ? ? ? ? ?|
> | FATAL ERROR ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ---
> O X |
> |_______________________________________________________|
> | ? ? ? ? ? ? ? ? ? ? ? ?You have moved the mouse.
> ?|
> | Windows must be restarted for the changes to take effect. ? |
> | ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?<OK>
> ? ? ? ? ? ? ? ?|
> ####################################/
>
>
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa



From nagalenoj at gmail.com  Wed Oct 14 07:10:42 2009
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Wed, 14 Oct 2009 19:40:42 +0530
Subject: [Freeswitch-users] Bridge in event socket - async mode
Message-ID: <fe67a3340910140710g136e1aa2x7a23aaff77a7b6c0@mail.gmail.com>

Hi,

I'm using event outboud socket(perl) in async mode.

Scenario where I face problem:
When a call comes to an extension(1000), my program will play some message
to the user, and get some DTMF.
I'll get the DTMF event(as I'm in async mode) and store the digits in a
variable.

When he presses "#", I call $con->execute("bridge","user/$dtmf"), to bridge
him to the dialed number.

It is bridging sometimes, and sometimes it is not bridging.

Is it the problem with async mode??
I also set $con->setEventLock("true"), before executing the bridge. But
still the problem persists.

I've seen the logs too, I didn't find the bridge application getting
executed.
What makes the problem here..??

The freeswitch log, when bridge is happening:
    http://pastebin.freeswitch.org/10704

The freeswitch log, when bridging is not happening:
    http://pastebin.freeswitch.org/10705

Code:

 # some statements

if($name eq "CHANNEL_BRIDGE") { # checking for events
.
}
elsif ($name eq "DTMF") {
   my $digit = $e->getHeader("dtmf-digit");
	
   if($digit eq '#') {
	print "Going to bridge\n";
	$con->execute("bridge","user/$dtmf_digit");		
   }
   else {
	$dtmf_digit.=$digit;
   }
}


-- 
Regards,
Nagalenoj H.
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From anthony.minessale at gmail.com  Wed Oct 14 07:26:39 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 14 Oct 2009 09:26:39 -0500
Subject: [Freeswitch-users] Bridge in event socket - async mode
In-Reply-To: <fe67a3340910140710g136e1aa2x7a23aaff77a7b6c0@mail.gmail.com>
References: <fe67a3340910140710g136e1aa2x7a23aaff77a7b6c0@mail.gmail.com>
Message-ID: <191c3a030910140726y2198213fhc97f3c2d0c3aeb5d@mail.gmail.com>

try putting
ESL::eslSetLogLevel(7);
at the top so you can get a trace of the esl data on the stdout from your
script
also put some debug code to confirm you are getting the digit


On Wed, Oct 14, 2009 at 9:10 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> Hi,
>
> I'm using event outboud socket(perl) in async mode.
>
> Scenario where I face problem:
> When a call comes to an extension(1000), my program will play some message
> to the user, and get some DTMF.
> I'll get the DTMF event(as I'm in async mode) and store the digits in a
> variable.
>
> When he presses "#", I call $con->execute("bridge","user/$dtmf"), to bridge
> him to the dialed number.
>
> It is bridging sometimes, and sometimes it is not bridging.
>
> Is it the problem with async mode??
> I also set $con->setEventLock("true"), before executing the bridge. But
> still the problem persists.
>
> I've seen the logs too, I didn't find the bridge application getting
> executed.
> What makes the problem here..??
>
> The freeswitch log, when bridge is happening:
>     http://pastebin.freeswitch.org/10704
>
> The freeswitch log, when bridging is not happening:
>     http://pastebin.freeswitch.org/10705
>
> Code:
>
>  # some statements
>
> if($name eq "CHANNEL_BRIDGE") { # checking for events
> .
> }
> elsif ($name eq "DTMF") {
>    my $digit = $e->getHeader("dtmf-digit");
> 	
>    if($digit eq '#') {
>
> 	print "Going to bridge\n";
> 	$con->execute("bridge","user/$dtmf_digit");		
>    }
>    else {
> 	$dtmf_digit.=$digit;
>    }
> }
>
>
> --
> Regards,
> Nagalenoj H.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From odermann at googlemail.com  Wed Oct 14 07:31:31 2009
From: odermann at googlemail.com (Dennis)
Date: Wed, 14 Oct 2009 16:31:31 +0200
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <65d96fc80910132348t202905fbub57cc4c814eb4e21@mail.gmail.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
	<191c3a030910130803k321f39e3n265e64ab3c84a5e6@mail.gmail.com>
	<5e414ed0910130826j11d45a66q86d785f900e4196e@mail.gmail.com>
	<65d96fc80910131012v1f72ac49o6bfe9f2267e541c5@mail.gmail.com>
	<191c3a030910131101p418e0961t373db68285248ff6@mail.gmail.com>
	<538F4CF43B294D0EACABA281E1913B4C@chaos>
	<65d96fc80910131224m73f35b87g355be818ad3c0caf@mail.gmail.com>
	<48B637E0506F4A809E4C0CAF39BD08BB@chaos>
	<65d96fc80910132348t202905fbub57cc4c814eb4e21@mail.gmail.com>
Message-ID: <5e414ed0910140731w1c7ebedr150e69cda8073155@mail.gmail.com>

the thing we want to make working nicer is the following:

we want the main/basic phonenumber (123456) to be reachable, so that
the telephone rings. but we also want it to be expandable with
ddi-digits.

example: dial the 123456 to reach the company, dial the 123456 1 to
reach the support.

in the moment our carrier waits x (milli-)seconds to see, if the
123456 was dialed or if there were more digits attached. after this
time, the carrier sends us the whole number as a block (may it be
123456 or 1234561) and we answer.

we want to avoid the waiting time the carrier needs to wait for
(possible) more digits. we want to receive every single digit from our
carrier and we want to tell fs, when the number is complete and fs can
answer.



From anthony.minessale at gmail.com  Wed Oct 14 07:47:19 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 14 Oct 2009 09:47:19 -0500
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <5e414ed0910140731w1c7ebedr150e69cda8073155@mail.gmail.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
	<191c3a030910130803k321f39e3n265e64ab3c84a5e6@mail.gmail.com>
	<5e414ed0910130826j11d45a66q86d785f900e4196e@mail.gmail.com>
	<65d96fc80910131012v1f72ac49o6bfe9f2267e541c5@mail.gmail.com>
	<191c3a030910131101p418e0961t373db68285248ff6@mail.gmail.com>
	<538F4CF43B294D0EACABA281E1913B4C@chaos>
	<65d96fc80910131224m73f35b87g355be818ad3c0caf@mail.gmail.com>
	<48B637E0506F4A809E4C0CAF39BD08BB@chaos>
	<65d96fc80910132348t202905fbub57cc4c814eb4e21@mail.gmail.com>
	<5e414ed0910140731w1c7ebedr150e69cda8073155@mail.gmail.com>
Message-ID: <191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com>

So with overlap you will have to keep refusing the call until the right
amount of digits are dialed.
This mode would send 1 then 12 then 123 then 1234 then 12345 then 123456 as
they were being dialed.
once you have 123456 won't you still be unsure if he will type the next 1 or
not and be forced to refuse it and wait anyway?

This feature is mostly used to allow the server to accept the call the
instant it knows there is an exact match in the dialplan or a non-existent
number has been dialed.  For instance, if 1234 was a unique ext, it could
instantly return since it knows.


On Wed, Oct 14, 2009 at 9:31 AM, Dennis <odermann at googlemail.com> wrote:

> the thing we want to make working nicer is the following:
>
> we want the main/basic phonenumber (123456) to be reachable, so that
> the telephone rings. but we also want it to be expandable with
> ddi-digits.
>
> example: dial the 123456 to reach the company, dial the 123456 1 to
> reach the support.
>
> in the moment our carrier waits x (milli-)seconds to see, if the
> 123456 was dialed or if there were more digits attached. after this
> time, the carrier sends us the whole number as a block (may it be
> 123456 or 1234561) and we answer.
>
> we want to avoid the waiting time the carrier needs to wait for
> (possible) more digits. we want to receive every single digit from our
> carrier and we want to tell fs, when the number is complete and fs can
> answer.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Wed Oct 14 07:52:14 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 14 Oct 2009 09:52:14 -0500
Subject: [Freeswitch-users] mod_socket: custom timeout for HEARTBEAT
	event
In-Reply-To: <d211f54a0910132331j14ccf510xd87bcb092a43f0b3@mail.gmail.com>
References: <d211f54a0910110959y3d3deed7s43201a3e39756585@mail.gmail.com>
	<86a32abc0910111444j1ab2e446p3cb94b0ae04126e3@mail.gmail.com>
	<97B4BDC9-5B11-4A98-B1BF-CC00A82ECBD9@jerris.com>
	<d211f54a0910112238i5f69275ey7cad437e47e6b567@mail.gmail.com>
	<191c3a030910120726p46d6410ejd94b61612686fbbc@mail.gmail.com>
	<d211f54a0910130832m25aaadd3t4d6e938d1c71d701@mail.gmail.com>
	<87f2f3b90910131148m1853a219md8788f55e40ef4c3@mail.gmail.com>
	<d211f54a0910132331j14ccf510xd87bcb092a43f0b3@mail.gmail.com>
Message-ID: <191c3a030910140752v2360c4d1r536f7eefa97603d6@mail.gmail.com>

the heartbeat you were seeing that needs a uuid was the session heartbeat
which is a per-call heartbeat.
indeed, the system heartbeat is not customizable.


On Wed, Oct 14, 2009 at 1:31 AM, Artem Shiyanov <shiyanov at gmail.com> wrote:

> Finally!!
> Thank you Michael, I didn't know about "status" app. It satisfies all my
> desires.
>
> And again,
> thanks for all the community for the strong support!
>
> Artem
>
>
>
>
> On Tue, Oct 13, 2009 at 10:48 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Tue, Oct 13, 2009 at 8:32 AM, Artem Shiyanov <shiyanov at gmail.com>wrote:
>>
>>> Sorry for my foolishness but I stil can't grasp it.
>>> I'm developing app based on inbound mod_event_socket and I don't know how
>>> to run "enable_heartbeat" or " sched_heartbeat" without specifying any alive
>>> session uuid. I tried to use "create_uuid" and send mentioned  commands to
>>> the created uuid  but this approach doesn't work. I do need to monitor the
>>> FS state itself rather then any particular call.
>>>
>>> Please, enlighten me!
>>> Artem
>>>
>>
>> So if I understand correctly, you want to see the system heartbeat more
>> frequently than every 20 seconds? What, exactly, is in the heartbeat that
>> you need? You can get the same information by sending the API "status" and
>> reading the results.
>>
>> If I read the source code correctly it looks like the system heartbeat is
>> hard-coded at 20 seconds. See switch_core.c, line 155. However, before you
>> go messing with a core file I would make sure that you can't already use the
>> tools that are available, like sending a "status" API every 5 seconds (or
>> whatever interval you need).
>>
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From jmesquita at freeswitch.org  Wed Oct 14 08:08:46 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 14 Oct 2009 12:08:46 -0300
Subject: [Freeswitch-users] adding new extension
In-Reply-To: <f8af5740910140458w4666ee2fwa93d1ecb70cc015a@mail.gmail.com>
References: <f8af5740910140458w4666ee2fwa93d1ecb70cc015a@mail.gmail.com>
Message-ID: <ebbc2d550910140808h3bc09fb2s53224c76389efe08@mail.gmail.com>

Never _EVER_ change the freeswitch.xml.fsxml file directly. If you need
something dynamic you would have to implement directory using mod_xml_curl,
otherwise, you change the files on the conf/ tree and do a reloadxml on the
CLI.

JM

On Wed, Oct 14, 2009 at 8:58 AM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

> Hi,
>
> can any please let me know, how to add new extension(eg 1000.xml)
> dynamically while running freeswitch?
> while running freeswitch i have created new xml(eg: 1500.xml) file and i
> have changed freeswitch.xml.fsxml still  not working,
>
> Thanks
> Srinivasula Reddy K
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Wed Oct 14 08:11:59 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 14 Oct 2009 12:11:59 -0300
Subject: [Freeswitch-users] Some help with my post-paid billing project
In-Reply-To: <b39273530910140636j72652456ga0ace41468b5c546@mail.gmail.com>
References: <86a32abc0910131453vea22e4bn2f7ceafaeab9a885@mail.gmail.com>
	<74730C11-75D0-4BC0-9F63-8B6C9F0F499F@me.com>
	<86a32abc0910131836w5f113a16s247caf5eeb0f8fe4@mail.gmail.com>
	<86a32abc0910131853v244ed61bof9c24f345a0944b2@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C6703241D1E52@mse17be1.mse17.exchange.ms>
	<b16156850910140055r37581014iddf52d915aa7eaa4@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C6703241D1E8A@mse17be1.mse17.exchange.ms>
	<b16156850910140622w78b356b4hd7280bc504416685@mail.gmail.com>
	<b39273530910140636j72652456ga0ace41468b5c546@mail.gmail.com>
Message-ID: <ebbc2d550910140811q2e5d93cbx171d2cf358aa921f@mail.gmail.com>

Could not agree more with Rupa. mod_lcr is an awesome piece of software.
Thank you Rupa for that.

JM

On Wed, Oct 14, 2009 at 10:36 AM, Rupa Schomaker <rupa at rupa.com> wrote:

> I would still suggest using mod_lcr for this...  If you have any real
> volume, use postgresql with the prefix module.
>
> It also supports IN lists, OR lists, optional quoting (since mysql is
> retarded), and custom sql so you can interface with whatever stored
> proc or deal with whatever database table you may need to support.
>
> It also supports multiple profiles (so you can charge different
> amounts based on some other criteria like account code) and...
> interstate/intrastate/intralata rating.  probably other stuff I forgot
> to mention...
>
> Let me just recommend to reuse what already exists and improve on that.
>
> On Wed, Oct 14, 2009 at 8:22 AM, Muhammad Shahzad
> <shaheryarkh at googlemail.com> wrote:
> > Oops, you are right, this is what happens when you are doing more then
> one
> > thing at a time, i was writing a db function while replying in this email
> > thread and confused LIKE with IN. There is one more mistake in my email
> if
> > any one noticed, that is, string class in STL does not support negative
> > length in method substr.
> >
> > Thanks for correction.
> >
> >
> > On Wed, Oct 14, 2009 at 6:43 PM, Michael Giagnocavo <mgg at giagnocavo.net>
> > wrote:
> >>
> >> Wouldn?t that be SQL ?IN? instead of LIKE?
> >>
> >>
> >>
> >> First off, see if you can do this in memory. If it?s just a ?rate sheet?
> >> list of prefixes + billing, it?s probably not that much data. I?d guess
> >> maybe 64-128 bytes per record (in Ruby it?s probably more, maybe 256
> bytes?)
> >> inside a hashtable. Even so, we?re only talking about 1.5 to 3GB of RAM.
> >> Stick it in a nice performing dictionary of some sort and you?re set.
> >>
> >>
> >>
> >> Just as a quick test, on one core of a Core2, it takes about 150ms to do
> >> 100K lookups against a 1M item hashtable (F#) ? that?s just a 1 line
> loop I
> >> tried out with no optimization, doing about 650K/sec. For comparison on
> the
> >> same machine, looping on a simple ?SELECT @@VERSION? command only
> achieved
> >> 5000 executions/second on a single thread ? that?s just the SQL overhead
> (to
> >> a local SQL instance!).
> >>
> >>
> >>
> >> If the memory usage is too excessive, consider caching only subsets of
> the
> >> prefixes. Keep track of which leading prefix (1 or 2 digits) are the
> most
> >> used, and keep those in memory. Have a reasonably fast DB to fall back
> to.
> >>
> >>
> >>
> >> If you?re sure the storage requirements or the change frequency truly
> >> eliminates keeping it in your own memory, here are some suggestions for
> >> working on DB performance:
> >>
> >>
> >>
> >> A little perf gain is to pre-calculate possible lengths, so you?re
> >> performing less lookups. Just have a table based on the first 2 or 3
> digits,
> >> giving you the valid prefix lengths. That can easily cut down the number
> of
> >> lookups from 10-12 to, say, 3-4. Compare the perf of temp tables, CTEs,
> and
> >> dynamic SQL ?IN?.
> >>
> >>
> >>
> >> But I would be hesitant to do this from a higher level ORM or via
> dynamic
> >> SQL queries:
> >>
> >> -          You?ll have a much harder time doing DB-specific
> >> optimizations/hints
> >>
> >> -          Dynamic SQL needs to be parsed each time (possibly minor
> issue)
> >>
> >> -          If the ad hoc queries are not properly parameterized, query
> >> plans might not get reused very well
> >>
> >> -          You can easily send many times more data to the server (your
> >> entire SQL, versus just the sproc name + parameters)
> >>
> >>
> >>
> >> Really look into doing programmability on the server. There?s all sorts
> of
> >> things you can do to minimize what you need to send to the database, and
> let
> >> the DB engine figure out the most efficient way. You might want to use a
> >> product that?s a bit more advanced than MySQL.
> >>
> >>
> >>
> >> And finally, test, test, test. You should have a nice benchmark with
> >> several million rows and be able to execute, say, 10-50K lookups (on
> >> multiple threads) after each change. Modify your query, execute it,
> _inspect
> >> the execution plan_. Check your indexes ? make sure they are covering
> the
> >> query in the right way. (For example, if the table is (ID, Prefix,
> Price),
> >> you want an index of (Prefix, Price).) Review possible hints (for
> example,
> >> SQL Server usually needs a hint to choose a indexed view). Make sure
> your
> >> transaction level is set appropriately (can you do the query without
> >> locking?). Can you batch up several numbers in one command, saving on
> the
> >> per-command overhead?
> >>
> >>
> >>
> >> Also consider it in light of the rest of your application. If everything
> >> is in the same database, then you can probably do the entire rate
> call/save
> >> cdr/update balances in a single command and transaction.
> >>
> >>
> >>
> >> For us, our initial draft of routing inside SQL Server performed well
> >> under 100 calls/sec ? not usable. After a few days of playing with
> things,
> >> it went over 1000 calls/sec, which is comfortable enough for a $500
> server.
> >> This is matching 100 gateways and dialplans with several thousand
> entries
> >> per dialplan, across 4 million+ routes, doing QBR/LCR in the process.
> >>
> >>
> >>
> >> -Michael
> >>
> >>
> >>
> >>
> >>
> >> From: freeswitch-users-bounces at lists.freeswitch.org
> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Muhammad
> >> Shahzad
> >> Sent: Wednesday, October 14, 2009 1:55 AM
> >> To: freeswitch-users at lists.freeswitch.org
> >> Subject: Re: [Freeswitch-users] Some help with my post-paid billing
> >> project
> >>
> >>
> >>
> >> I fully agree that direct matching is much faster then pattern matching
> in
> >> SQL.
> >>
> >> One of my clients had same problem, he had around 12 million number
> >> prefixes in a table and during each call an AGI script use to query that
> >> table to find longest prefix match, but this use to take like 3-5
> seconds
> >> even with indexed columns. So after a lots of R & D we come up with
> >> following logic,
> >>
> >> 1. We break the destination number length-wise, e.g. suppose number is
> >> 923344224088 then length chunks would be,
> >>
> >> 923344224088, 92334422408, 9233442240, 923344224, 92334422, 9233442,
> >> 923344, 92334, 9233, 923, 92, 9
> >>
> >> 2. Then use SQL LIKE function in WHERE clause (you can also use SQL OR
> >> function if your DBMS doesn't support SQL LIKE function), and pass all
> these
> >> chunks to it, e.g.
> >>
> >> WHERE prefix LIKE (923344224088, 92334422408, 9233442240, 923344224,
> >> 92334422, 9233442, 923344, 92334, 9233, 923, 92, 9)
> >>
> >> 3. Lastly we ORDER the result by prefix length, e.g.
> >>
> >> ORDER BY LENGTH(prefix) DESC LIMIT 1
> >>
> >> 4. The complete query will be,
> >>
> >> SELECT * FROM prefixes
> >> WHERE prefix LIKE (923344224088, 92334422408, 9233442240, 923344224,
> >> 92334422, 9233442, 923344, 92334, 9233, 923, 92, 9)
> >> ORDER BY LENGTH(prefix) DESC LIMIT 1
> >>
> >> Now the query takes less then 150 ms to execute. :-)
> >>
> >> Here is an STL method that can generate this query, i am sure you can
> >> convert it to any programming language of your choice easily.
> >>
> >> =========================================================
> >> std::string GetQuery(std::string destination) {
> >>         std::string query = "SELECT * FROM prefixes WHERE prefix LIKE
> ('"
> >> + destination;
> >>
> >>         for(int i=1; i<destination.length(); i++) {
> >>                 query += "','" + destination.substr(0, (i * -1));
> >>
> >>         }
> >>
> >>         query += "') ORDER BY LENGTH(prefix) DESC LIMIT 1";
> >>         return query;
> >> }
> >> =========================================================
> >>
> >> I am pretty sure this query is 100% ANSI SQL compatible
> >> (http://en.wikipedia.org/wiki/SQL).
> >>
> >> Thank you.
> >>
> >> On Wed, Oct 14, 2009 at 10:15 AM, Michael Giagnocavo <
> mgg at giagnocavo.net>
> >> wrote:
> >>
> >> In our testing with SQL Server, we found that executing several queries
> >> for direct matches yielded far better performance than one query trying
> to
> >> check prefixes. (The column was also part of the clustered index, but
> AFAIK
> >> MySQL doesn?t support defining your own clustered indexes; you get the
> PK
> >> always.)
> >>
> >>
> >>
> >> -Michael
> >>
> >>
> >>
> >> From: freeswitch-users-bounces at lists.freeswitch.org
> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Diego
> >> Viola
> >> Sent: Tuesday, October 13, 2009 7:54 PM
> >> To: freeswitch-users at lists.freeswitch.org
> >> Subject: Re: [Freeswitch-users] Some help with my post-paid billing
> >> project
> >>
> >>
> >>
> >> Wrong question.
> >>
> >> Is there a way to compare numbers with prefixes without using the prefix
> >> module?
> >>
> >> Diego
> >>
> >> On Wed, Oct 14, 2009 at 1:36 AM, Diego Viola <diego.viola at gmail.com>
> >> wrote:
> >>
> >> I'm using MySQL now but I will try PostgreSQL with the prefix module, is
> >> there a way to do that without the prefix module and with regular SQL?
> >>
> >> Any examples?
> >>
> >> Diego
> >>
> >>
> >>
> >> On Tue, Oct 13, 2009 at 10:45 PM, Even Andr? Fiskvik <grevenx at me.com>
> >> wrote:
> >>
> >> What database are you using?
> >> You could do this with regular SQL, but it would by a costly operation,
> >> for PostgreSQL we're using the prefix module:
> >> http://pgfoundry.org/projects/prefix/
> >>
> >> You can then match the closest prefix by using something like
> >> "WHERE myprefix_col @> caller_destination_number ORDER BY LENGTH
> >> (myprefix_col::text) LIMIT 1;"
> >>
> >>
> >> Best regards,
> >> Even Andr?
> >>
> >> On 13. okt. 2009, at 23.53, Diego Viola wrote:
> >>
> >> > Hello,
> >> >
> >> > I'm trying to write a post-paid billing script, I have the CDR on my
> >> > database and also a "rates" table, the CDR contains fields like
> >> > caller_destination_number, variable_duration, etc. and the rates
> >> > table contains: destination, prefix, rate (cost).
> >> >
> >> > The problem is that I can't just strip the destination number to
> >> > take the prefix from it because I have to deal with destination
> >> > numbers from different countries and they all have different prefix
> >> > lengths... so I need to find another way to take the prefix from the
> >> > destination number.
> >> >
> >> > Any ideas how to do this?
> >> >
> >> > Thanks,
> >> >
> >> > Diego
> >> >
> >>
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> > users
> >> > http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >> --
> >> ________________________________________________________
> >> |
> >>                        |
> >> | FATAL ERROR
> >> --- O X |
> >> |_______________________________________________________|
> >> |                        You have moved the mouse.
> >>    |
> >> | Windows must be restarted for the changes to take effect.   |
> >> |                                        <OK>
> >>                  |
> >> ####################################/
> >>
> >>
> >> Muhammad Shahzad
> >> -----------------------------------
> >> CISCO Rich Media Communication Specialist (CRMCS)
> >> CISCO Certified Network Associate (CCNA)
> >> Cell: +92 334 422 40 88
> >> MSN: shari_786pk at hotmail.com
> >> Email: shaheryarkh at googlemail.com
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > --
> > ________________________________________________________
> > |
> >                      |
> > | FATAL ERROR
> ---
> > O X |
> > |_______________________________________________________|
> > |                        You have moved the mouse.
> >  |
> > | Windows must be restarted for the changes to take effect.   |
> > |                                        <OK>
> >                |
> > ####################################/
> >
> >
> > Muhammad Shahzad
> > -----------------------------------
> > CISCO Rich Media Communication Specialist (CRMCS)
> > CISCO Certified Network Associate (CCNA)
> > Cell: +92 334 422 40 88
> > MSN: shari_786pk at hotmail.com
> > Email: shaheryarkh at googlemail.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Wed Oct 14 08:11:33 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 14 Oct 2009 08:11:33 -0700
Subject: [Freeswitch-users] Fwd: Groups information in sqllite
In-Reply-To: <f8af5740910140249t67df3364k1bb81a9c544f185e@mail.gmail.com>
References: <f8af5740910122301u338ca128macfc0b769ea0e961@mail.gmail.com>
	<f8af5740910122302i47ccde5bo9df2bae965408394@mail.gmail.com>
	<B3C83D64-BC34-4F67-B72E-9EAE6F3A362E@jerris.com>
	<f8af5740910140249t67df3364k1bb81a9c544f185e@mail.gmail.com>
Message-ID: <42BA0F36-1BF7-45B3-A50A-321E2879C003@freeswitch.org>

I think we have said NO more than once in the thread... so to clear  
this up RIGHT now without coding anything else you can NOT load your  
groups from SQLITE.

/b

On Oct 14, 2009, at 2:49 AM, srinivasula reddy wrote:

> groups data is taking from these xml file. instead of taking from  
> these xml file can i take from sqllite?




From dipen at entvoice.com  Wed Oct 14 05:54:28 2009
From: dipen at entvoice.com (dipen at entvoice.com)
Date: Wed, 14 Oct 2009 08:54:28 -0400
Subject: [Freeswitch-users] Mod_java loading error
Message-ID: <34733.1255524868@entvoice.com>

Hi Baskar,

Can you tell us how did you solve your problem of Loading mod_java.
I mean which steps to follow to solve the problem.

Currently i am getting the error..

 [ERR] modjava.c:133 load_config() Error loading /opt/SDK/jdk/jre/lib/i386/client/libjvm.so
 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_java.so
**Module load routine returned an error**

Thanks & Regards
Dipen Velani.


On Wed 15/04/09  3:15 AM , Baskar <yudha2008 at gmail.com> wrote:

> Hi,
> 
>    Now i can able to load the mod_java in the freeswitch console.
> 
> After that i have followed these method to run the PhoneTest.java
> 1) VERIFIED MY CLASSPATH IN THE JAVA.CONF.XML:   
> 
> 2)MY PHONETEST.CLASS IS LOCATED IN /USR/LOCAL/FREESWITCH/SCRIPT -
> DIRECTORY, SAME AS WHERE FREESWITCH.JAR IS LOCATED 
> 
> 3) I CREATED PHONETEST.JAR FILE FROM PHONETEST.CLASS 
> 
> 4)CHANGED MY DIALPLAN TO:  
> Then i have complied and Run the PhoneTest.java  by this methods
> 
> Method1:
> [localhost scripts]# javac -cp freeswitch.jar PhoneTest.java 
> [localhost scripts]# java -cp freeswitch.jar PhoneTest
> Exception in thread "main" java.lang.NoClassDefFoundError: PhoneTest
> Caused by: java.lang.ClassNotFoundException: PhoneTest
>         at
> java.net.URLClassLoader$1.run(URLClassLoader.java:200)
>         at java.security.AccessController.doPrivileged(Native
> Method)
>         at
> java.net.URLClassLoader.findClass(URLClassLoader.java:188)
>         at
> java.lang.ClassLoader.loadClass(ClassLoader.java:306)
>         at
> sun.misc.Launcher$AppClassLoader.loadClass(Launcher.java:276)
>         at
> java.lang.ClassLoader.loadClass(ClassLoader.java:251)
>         at
> java.lang.ClassLoader.loadClassInternal(ClassLoader.java:319)
> 
> Method2:
> [localhost scripts]# javac -cp PhoneTest.jar:freeswitch.jar
> PhoneTest.java
> [localhost scripts]# java -cp PhoneTest.jar:freeswitch.jar PhoneTest
> Exception in thread "main" java.lang.NoSuchMethodError: main
> 
> But I get these above errors .So Please Correct me where i am wrong
> and assist me how to compile and run the PhoneTest.java.
> -- 
> Warm Regards,
> N.Baskar
> 
> 
> 
---- Msg sent via @Mail - http://atmail.com/



From nbelov at abisoft.spb.ru  Wed Oct 14 05:59:25 2009
From: nbelov at abisoft.spb.ru (Nikita Belov)
Date: Wed, 14 Oct 2009 16:59:25 +0400
Subject: [Freeswitch-users] conference call
Message-ID: <064A5201C10D4D20AC76EB93CA5EA4B5@abisoft.biz>

HI all,

 

I want to configure FS to make special conference call between three users
(A, B, C).  In this conference C will hear A and B, but A will hear only B.
Can I make it using FS API commands? Does anybody know which approach is
better to use?

 

___

 

Thanks, Nikita

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From szasz.szabolcs at gmail.com  Tue Oct 13 23:16:37 2009
From: szasz.szabolcs at gmail.com (Szasz Szabolcs)
Date: Wed, 14 Oct 2009 09:16:37 +0300
Subject: [Freeswitch-users] TLS Audiocodes
Message-ID: <4a4d27d30910132316g206580f0ya32287291af00c5b@mail.gmail.com>

Hi,

Did anybody set up TLS between Freeswitch and Audiodes MP11X ? I got to work
TLS between freeswitch  and a softphone (phonerlite), but I have problem
with Audiocodes during the TLS authentication. I've loaded the certification
but it still doesn't work. Can I debug the tls in freeswitch? Please help
me.

Thank you in advance.

Szasz Szabolcs
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From szasz.szabolcs at gmail.com  Wed Oct 14 03:02:20 2009
From: szasz.szabolcs at gmail.com (Szasz Szabolcs)
Date: Wed, 14 Oct 2009 13:02:20 +0300
Subject: [Freeswitch-users] TLS Audiocodes
In-Reply-To: <4a4d27d30910132316g206580f0ya32287291af00c5b@mail.gmail.com>
References: <4a4d27d30910132316g206580f0ya32287291af00c5b@mail.gmail.com>
Message-ID: <4a4d27d30910140302s7c4e109j3cee9a6fe0d755db@mail.gmail.com>

Hello,

Turning on debug I got the following messages:

tls_connect(0x942e8f0): events NEGOTIATING
tls_connect(0x942e8f0): events NEGOTIATING
tls_connect(0x942e8f0): TLS setup failed
(error:00000001:lib(0):func(0):reason(1))
tport_close(0x942e8f0): tls/82.77.201.227:64571/sips

Szasz Szabolcs

On Wed, Oct 14, 2009 at 9:16 AM, Szasz Szabolcs <szasz.szabolcs at gmail.com>wrote:

> Hi,
>
> Did anybody set up TLS between Freeswitch and Audiodes MP11X ? I got to
> work TLS between freeswitch  and a softphone (phonerlite), but I have
> problem with Audiocodes during the TLS authentication. I've loaded the
> certification but it still doesn't work. Can I debug the tls in freeswitch?
> Please help me.
>
> Thank you in advance.
>
> Szasz Szabolcs
>
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From larclap at yahoo.com  Wed Oct 14 08:28:05 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Wed, 14 Oct 2009 08:28:05 -0700
Subject: [Freeswitch-users] lua script causing FreeSwitch to crash?
In-Reply-To: <87f2f3b90910131312k7dac8980h70ce8be9c07aa7b2@mail.gmail.com>
References: <005a01ca4c3c$c183ceb0$448b6c10$@com>
	<87f2f3b90910131312k7dac8980h70ce8be9c07aa7b2@mail.gmail.com>
Message-ID: <000601ca4ce2$ed1de640$c759b2c0$@com>

Michael,

 

I upgraded to v15152. There is no apparent change in the behavior. When I
call into a number handled by the lua script, I may get a fast busy signal,
or a recording saying the connection cannot be made.

 

But FreeSwitch is removed from memory as a result. I cannot see from the log
that the FreeSwitch ended.

 

Is there anything I can try on my side?

 

http://pastebin.freeswitch.org/10707

 

Lars

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, October 13, 2009 1:12 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] lua script causing FreeSwitch to crash?

 

 

On Tue, Oct 13, 2009 at 12:38 PM, Lars Zeb <larclap at yahoo.com> wrote:

http://pastebin.freeswitch.org/10686

 

I am trying to use a lua script for inbound calls. The caller hears a busy
signal. After this call fails to go through, FreeSwitch is no longer in
memory - it does not appear in  'ps -ef' output.

 

I don't have a clue what I might be doing incorrectly in the script in order
to cause this to happen. There is no indication in the log that FreeSwitch
has terminated. But it is not resident in memory.

 

I am using v15134. I had been using this same script without incident as
late as v14929. Until I can resolve the issue, I have removed the call to
lua from the dialplan and done all I could in it instead.

We sent out a notice about a bug in event socket between revs 15119 and
15134. Update to latest trunk. (FYI, the fix was in r15135, so you just
barely missed it.)

After you update, let us know if the problem persists. 
-MC

 

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From jmesquita at freeswitch.org  Wed Oct 14 08:38:22 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 14 Oct 2009 12:38:22 -0300
Subject: [Freeswitch-users] conference call
In-Reply-To: <064A5201C10D4D20AC76EB93CA5EA4B5@abisoft.biz>
References: <064A5201C10D4D20AC76EB93CA5EA4B5@abisoft.biz>
Message-ID: <ebbc2d550910140838v3125579ev9fa16e3e83d25770@mail.gmail.com>

Look at eavesdrop on the wiki.

JM

2009/10/14 Nikita Belov <nbelov at abisoft.spb.ru>

>  HI all,
>
>
>
> I want to configure FS to make special conference call between three users
> (A, B, C).  In this conference C will hear A and B, but A will hear only B.
> Can I make it using FS API commands? Does anybody know which approach is
> better to use?
>
>
>
> ___
>
>
>
> Thanks, Nikita
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Wed Oct 14 09:02:09 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 14 Oct 2009 11:02:09 -0500
Subject: [Freeswitch-users] lua script causing FreeSwitch to crash?
In-Reply-To: <000601ca4ce2$ed1de640$c759b2c0$@com>
References: <005a01ca4c3c$c183ceb0$448b6c10$@com>
	<87f2f3b90910131312k7dac8980h70ce8be9c07aa7b2@mail.gmail.com>
	<000601ca4ce2$ed1de640$c759b2c0$@com>
Message-ID: <191c3a030910140902j401a9c6dg7704616d27bac051@mail.gmail.com>

By which process did you upgrade?
Did you use "make current"
if so, run freeswitch from the directory you built it from then reporduce
the problem and watch for it to crash and type

./support-d/fscore_pb

at your shell to upload a crash report to pastebin and tell us the url it
generated.


On Wed, Oct 14, 2009 at 10:28 AM, Lars Zeb <larclap at yahoo.com> wrote:

>  Michael,
>
>
>
> I upgraded to v15152. There is no apparent change in the behavior. When I
> call into a number handled by the lua script, I may get a fast busy signal,
> or a recording saying the connection cannot be made.
>
>
>
> But FreeSwitch is removed from memory as a result. I cannot see from the
> log that the FreeSwitch ended.
>
>
>
> Is there anything I can try on my side?
>
>
>
> http://pastebin.freeswitch.org/10707
>
>
>
> Lars
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Tuesday, October 13, 2009 1:12 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] lua script causing FreeSwitch to crash?
>
>
>
>
>
> On Tue, Oct 13, 2009 at 12:38 PM, Lars Zeb <larclap at yahoo.com> wrote:
>
> http://pastebin.freeswitch.org/10686
>
>
>
> I am trying to use a lua script for inbound calls. The caller hears a busy
> signal. After this call fails to go through, FreeSwitch is no longer in
> memory ? it does not appear in  ?ps ?ef? output.
>
>
>
> I don?t have a clue what I might be doing incorrectly in the script in
> order to cause this to happen. There is no indication in the log that
> FreeSwitch has terminated. But it is not resident in memory.
>
>
>
> I am using v15134. I had been using this same script without incident as
> late as v14929. Until I can resolve the issue, I have removed the call to
> lua from the dialplan and done all I could in it instead.
>
> We sent out a notice about a bug in event socket between revs 15119 and
> 15134. Update to latest trunk. (FYI, the fix was in r15135, so you just
> barely missed it.)
>
> After you update, let us know if the problem persists.
> -MC
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From bjbrashier at gmail.com  Wed Oct 14 09:11:51 2009
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Wed, 14 Oct 2009 09:11:51 -0700
Subject: [Freeswitch-users] how can i get the caller-number when the
	caller in the conference
In-Reply-To: <49161ef40910140145r7b8d28dbu3d9e44b4c17547de@mail.gmail.com>
References: <49161ef40910140145r7b8d28dbu3d9e44b4c17547de@mail.gmail.com>
Message-ID: <7bcfdd290910140911mfe30ccclea156e5dc397a6bb@mail.gmail.com>

What method are you trying to use?

If you're using the event socket, it's in several of the parameters.
For instance, with an add-member event, you could look at
Caller-Username, Caller-Caller-ID-Name, or Caller-Caller-ID-Number
depending on your specific needs. There's also a uuid available if
that's better.

BB

On Wed, Oct 14, 2009 at 1:45 AM, qian ma <god.nirvana at gmail.com> wrote:
> hi all:
> ??????i can get the member_id in the conference,but how can can i get the
> number??
> ????? thx!
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From larclap at yahoo.com  Wed Oct 14 10:09:17 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Wed, 14 Oct 2009 10:09:17 -0700
Subject: [Freeswitch-users] lua script causing FreeSwitch to crash?
In-Reply-To: <191c3a030910140902j401a9c6dg7704616d27bac051@mail.gmail.com>
References: <005a01ca4c3c$c183ceb0$448b6c10$@com>	<87f2f3b90910131312k7dac8980h70ce8be9c07aa7b2@mail.gmail.com>	<000601ca4ce2$ed1de640$c759b2c0$@com>
	<191c3a030910140902j401a9c6dg7704616d27bac051@mail.gmail.com>
Message-ID: <005b01ca4cf1$10a30970$31e91c50$@com>

I did svn up, ./configure, make and make install.

 

Do you want me to do make current before proceeding, or just try to make it
stop with the current build from the above commands?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Wednesday, October 14, 2009 9:02 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] lua script causing FreeSwitch to crash?

 

By which process did you upgrade?
Did you use "make current"
if so, run freeswitch from the directory you built it from then reporduce
the problem and watch for it to crash and type

./support-d/fscore_pb

at your shell to upload a crash report to pastebin and tell us the url it
generated.



On Wed, Oct 14, 2009 at 10:28 AM, Lars Zeb <larclap at yahoo.com> wrote:

Michael,

 

I upgraded to v15152. There is no apparent change in the behavior. When I
call into a number handled by the lua script, I may get a fast busy signal,
or a recording saying the connection cannot be made.

 

But FreeSwitch is removed from memory as a result. I cannot see from the log
that the FreeSwitch ended.

 

Is there anything I can try on my side?

 

http://pastebin.freeswitch.org/10707

 

Lars

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, October 13, 2009 1:12 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] lua script causing FreeSwitch to crash?

 

 

On Tue, Oct 13, 2009 at 12:38 PM, Lars Zeb <larclap at yahoo.com> wrote:

http://pastebin.freeswitch.org/10686

 

I am trying to use a lua script for inbound calls. The caller hears a busy
signal. After this call fails to go through, FreeSwitch is no longer in
memory - it does not appear in  'ps -ef' output.

 

I don't have a clue what I might be doing incorrectly in the script in order
to cause this to happen. There is no indication in the log that FreeSwitch
has terminated. But it is not resident in memory.

 

I am using v15134. I had been using this same script without incident as
late as v14929. Until I can resolve the issue, I have removed the call to
lua from the dialplan and done all I could in it instead.

We sent out a notice about a bug in event socket between revs 15119 and
15134. Update to latest trunk. (FYI, the fix was in r15135, so you just
barely missed it.)

After you update, let us know if the problem persists. 
-MC

 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:213-799-1400

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From ryannyl at gmail.com  Wed Oct 14 10:40:55 2009
From: ryannyl at gmail.com (Ryanny Lin)
Date: Thu, 15 Oct 2009 01:40:55 +0800
Subject: [Freeswitch-users] how to make more profiles listen on port 5060 ?
Message-ID: <4bfcac7e0910141040i6edb596ej5877176035084262@mail.gmail.com>

Hi, all,

I read the article "Performance testing and configurations" on the wiki.
Here is a Recommended from wiki:
"libsofia only handles 1 thread per profile, so if that is your bottle
neck use more profiles"
How can I use more profiles and listen on port 5060? Is it possible?
I used sipp to tested SIP REGISTER performance which is about 165 cps.
It is about 260 cps when turned lots of modules.
Does anyone test higher than this?

-- 
Sincerely regards,
Ryanny
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From msc at freeswitch.org  Wed Oct 14 11:05:49 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 14 Oct 2009 11:05:49 -0700
Subject: [Freeswitch-users] how to make more profiles listen on port
	5060 ?
In-Reply-To: <4bfcac7e0910141040i6edb596ej5877176035084262@mail.gmail.com>
References: <4bfcac7e0910141040i6edb596ej5877176035084262@mail.gmail.com>
Message-ID: <87f2f3b90910141105m2a31ca4btef4b06f9253a275a@mail.gmail.com>

On Wed, Oct 14, 2009 at 10:40 AM, Ryanny Lin <ryannyl at gmail.com> wrote:

> Hi, all,
>
> I read the article "Performance testing and configurations" on the wiki.
> Here is a Recommended from wiki:
> "libsofia only handles 1 thread per profile, so if that is your bottle neck use more profiles"
>
> How can I use more profiles and listen on port 5060? Is it possible?
> I used sipp to tested SIP REGISTER performance which is about 165 cps. It is about 260 cps when turned lots of modules.
> Does anyone test higher than this?
>
> You can't have more than one profile listening on a particular port/IP
since by definition a profile is a UA that services exactly one IP/port
combo. As for SIPp, unless you are an expert in load testing then tread very
carefully. Your best bet for testing is to put the machine into production
and see how it perform with real traffic.

-MC
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From msc at freeswitch.org  Wed Oct 14 11:36:35 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 14 Oct 2009 11:36:35 -0700
Subject: [Freeswitch-users] adding new extension
In-Reply-To: <f8af5740910140458w4666ee2fwa93d1ecb70cc015a@mail.gmail.com>
References: <f8af5740910140458w4666ee2fwa93d1ecb70cc015a@mail.gmail.com>
Message-ID: <87f2f3b90910141136s2a233cabg166bb91f8b4e795b@mail.gmail.com>

On Wed, Oct 14, 2009 at 4:58 AM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

> Hi,
>
> can any please let me know, how to add new extension(eg 1000.xml)
> dynamically while running freeswitch?
> while running freeswitch i have created new xml(eg: 1500.xml) file and i
> have changed freeswitch.xml.fsxml still  not working,
>
> Boy, it would be great if someone wrote an article that covers this and
published it in a magazine... oh wait, we have that!
http://bit.ly/EpVrv

I cover this exact scenario. In fact, my example actually uses "1500" as a
new extension to add. Definitely read this article - it will help ease your
transition into the world of FreeSWITCH.
-MC
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From msc at freeswitch.org  Wed Oct 14 11:45:10 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 14 Oct 2009 11:45:10 -0700
Subject: [Freeswitch-users] FreeSWITCH training poll: please vote soon
Message-ID: <87f2f3b90910141145u56a2a14eh5e1a5c61918d3af5@mail.gmail.com>

Everyone,

Please go vote here:
http://www.freeswitch.org/node/208

Be honest - we need real feedback. Don't click the first choice unless you
really have the money. :)

Thanks everyone! We appreciate having such a great community.
-MC
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From msc at freeswitch.org  Wed Oct 14 11:48:11 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 14 Oct 2009 11:48:11 -0700
Subject: [Freeswitch-users] lua script causing FreeSwitch to crash?
In-Reply-To: <005b01ca4cf1$10a30970$31e91c50$@com>
References: <005a01ca4c3c$c183ceb0$448b6c10$@com>
	<87f2f3b90910131312k7dac8980h70ce8be9c07aa7b2@mail.gmail.com>
	<000601ca4ce2$ed1de640$c759b2c0$@com>
	<191c3a030910140902j401a9c6dg7704616d27bac051@mail.gmail.com>
	<005b01ca4cf1$10a30970$31e91c50$@com>
Message-ID: <87f2f3b90910141148p7d2d33a3od7098142853b0964@mail.gmail.com>

On Wed, Oct 14, 2009 at 10:09 AM, Lars Zeb <larclap at yahoo.com> wrote:

>  I did svn up, ./configure, make and make install.
>
>
>
> Do you want me to do make current before proceeding, or just try to make it
> stop with the current build from the above commands?
>
> Lars,

The "make current" method was designed exactly for the scenario you are in:
to upgrade smoothly and cleanly. It help prevents build skew among other
things. Please go do "make current" and then run your test again and report
back.

Thanks!
-MC
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From sjmudd at pobox.com  Wed Oct 14 12:46:47 2009
From: sjmudd at pobox.com (Simon J Mudd)
Date: 14 Oct 2009 21:46:47 +0200
Subject: [Freeswitch-users] FreeSWITCH training poll: please vote soon
In-Reply-To: <87f2f3b90910141145u56a2a14eh5e1a5c61918d3af5@mail.gmail.com>
References: <87f2f3b90910141145u56a2a14eh5e1a5c61918d3af5@mail.gmail.com>
Message-ID: <m3zl7tzrko.fsf@mad06.wl0.org>

msc at freeswitch.org (Michael Collins) writes:

> Please go vote here:
> http://www.freeswitch.org/node/208
> 
> Be honest - we need real feedback. Don't click the first choice unless you
> really have the money. :)

There's a missing option IMO at least for SOHO people like
myself. That's better documentation, such as books on FreeSWITCH,
perhaps showing how to build a small SOHO system, perhaps using
"standard" example components like a few hard SIP phones and a SIP
gateway, and perhaps a local SIP -> PSTN gateway.  Then show how to
configure things like recording conversations, enable "out of hours"
voicemail for "non-special" outside callers.  Something like that.

I'd buy a book that covered stuff like that today. I bought several
books on VoIP, a couple of Asterisk books and have read some related
RFCs but I'm not a VoIP professioal.  So any (good) books on FreeSWITCH
would help me adapt my knowledge on Asterisk to FreeSWITCH, and help
me setup FreeSWITCH the way I want.

Simon



From msc at freeswitch.org  Wed Oct 14 12:56:31 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 14 Oct 2009 12:56:31 -0700
Subject: [Freeswitch-users] FreeSWITCH training poll: please vote soon
In-Reply-To: <m3zl7tzrko.fsf@mad06.wl0.org>
References: <87f2f3b90910141145u56a2a14eh5e1a5c61918d3af5@mail.gmail.com>
	<m3zl7tzrko.fsf@mad06.wl0.org>
Message-ID: <87f2f3b90910141256i565d9062te9235f49ccf8b668@mail.gmail.com>

On Wed, Oct 14, 2009 at 12:46 PM, Simon J Mudd <sjmudd at pobox.com> wrote:

> msc at freeswitch.org (Michael Collins) writes:
>
> > Please go vote here:
> > http://www.freeswitch.org/node/208
> >
> > Be honest - we need real feedback. Don't click the first choice unless
> you
> > really have the money. :)
>
> There's a missing option IMO at least for SOHO people like
> myself. That's better documentation, such as books on FreeSWITCH,
> perhaps showing how to build a small SOHO system, perhaps using
> "standard" example components like a few hard SIP phones and a SIP
> gateway, and perhaps a local SIP -> PSTN gateway.  Then show how to
> configure things like recording conversations, enable "out of hours"
> voicemail for "non-special" outside callers.  Something like that.
>
> I'd buy a book that covered stuff like that today. I bought several
> books on VoIP, a couple of Asterisk books and have read some related
> RFCs but I'm not a VoIP professioal.  So any (good) books on FreeSWITCH
> would help me adapt my knowledge on Asterisk to FreeSWITCH, and help
> me setup FreeSWITCH the way I want.
>

We are aware of this issue and are addressing it separately. Pretty much
everyone is on board with better written documentation. This poll is
specifically asking about on-site training or a boot-camp style training
program, both of which would require a serious financial commitment from the
participants. If you are not in a position for serious FS training then
option #3 would be for you.
-MC
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From anthony.minessale at gmail.com  Wed Oct 14 13:01:30 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 14 Oct 2009 15:01:30 -0500
Subject: [Freeswitch-users] Perl event socket library problem
In-Reply-To: <1452e2980910132250w293596d6jce5515d787b4fa2a@mail.gmail.com>
References: <1452e2980910132250w293596d6jce5515d787b4fa2a@mail.gmail.com>
Message-ID: <191c3a030910141301q21301c8v5f3ff7cd9ac79a0f@mail.gmail.com>

That's what asynchronous means "do not block"

try $con->setEventLock("true"); with the socket in async mode, then your
requests will stack in order.



On Wed, Oct 14, 2009 at 12:50 AM, velusamy velu <velu.technical at gmail.com>wrote:

> Dear All,
>        I am implementing an IVR framework using Perl event socket
> libraries. At first I set socket mode async full. I have faced problem that
> the Perl statements executing before dailplan execution. So, I couldn't
> control my process.
>
>        Next I have tried without async mode, it solved that race condition
> problem. I had another requirement that while executing external application
> I need to play some voice file some time limit. When I played that voice
> file in playback, the playback application blocked. So, I can't break that
> playback application after some time limit.
>
>        I am hanging on this problem for past one week. Please any one help
> me to solve above problems...
>
> Thanks,
> Velusamy.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From sjmudd at pobox.com  Wed Oct 14 15:01:35 2009
From: sjmudd at pobox.com (Simon J Mudd)
Date: Thu, 15 Oct 2009 00:01:35 +0200
Subject: [Freeswitch-users] macros for performing repeated dialplan actions?
Message-ID: <20091014220135.GA12614@mad06.wl0.org>

Hi again,

I'm trying to build up a soho dialplan to replicate an existing Asterisk
setup I want to replace.

I'm editing
/usr/local/freeswitch/conf/dialplan/default/00_mydomain.com.xml to add
some dialplan rules for various destination numbers I'm using. These
dialplans may use one of several gateways I have configured.

I currently have:

<include>
  <extension name="spanish_mobiles.mydomain.com">
    <condition field="destination_number" expression="^(6\d{8})$">
      <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
      <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>
      <action application="bridge" data="sofia/gateway/asterisk/$1"/>
    </condition>
  </extension>

  <extension name="spanish_fijos.mydomain.com">
    <condition field="destination_number" expression="^(9\d{8})$">
      <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
      <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>
      <action application="bridge" data="sofia/gateway/asterisk/$1"/>
    </condition>
  </extension>

  ... more stuff not shown ...

</include>

Which routes out to my existing asterisk installation.

I now want to record all calls. Looking at
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session
it seems I need to add a section like into each entry prior to the
application="bridge".

    <action application="set" data="gateway_name=XXXXX"/>

    <action application="set" data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/>
    <action application="set" data="RECORD_COPYRIGHT=(C) 2009 my data here."/>
    <action application="set" data="RECORD_SOFTWARE=FreeSwitch"/>
    <action application="set" data="RECORD_ARTIST=My name"/>
    <action application="set" data="RECORD_COMMENT=My comment"/>
    <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/>
    <action application="set" data="RECORD_STEREO=true"/>
    <action application="record_session" data="$${base_dir}/recordings/${gateway_name}_${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/>

My question is how do I avoid repeatedly using these same 8-9 lines and
"merge" this into some macro which given the right parameters allows me
to define various extensions using different gateways but all configured
"the same way"?

Thanks for any pointers to how this should be done.

Simon



From msc at freeswitch.org  Wed Oct 14 15:10:51 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 14 Oct 2009 15:10:51 -0700
Subject: [Freeswitch-users] macros for performing repeated dialplan
	actions?
In-Reply-To: <20091014220135.GA12614@mad06.wl0.org>
References: <20091014220135.GA12614@mad06.wl0.org>
Message-ID: <87f2f3b90910141510n45401995y11e3e380653e9790@mail.gmail.com>

On Wed, Oct 14, 2009 at 3:01 PM, Simon J Mudd <sjmudd at pobox.com> wrote:

> Hi again,
>
> I'm trying to build up a soho dialplan to replicate an existing Asterisk
> setup I want to replace.
>
> I'm editing
> /usr/local/freeswitch/conf/dialplan/default/00_mydomain.com.xml to add
> some dialplan rules for various destination numbers I'm using. These
> dialplans may use one of several gateways I have configured.
>
> I currently have:
>
> <include>
>  <extension name="spanish_mobiles.mydomain.com">
>    <condition field="destination_number" expression="^(6\d{8})$">
>      <action application="set"
> data="effective_caller_id_number=${outbound_caller_id_number}"/>
>      <action application="set"
> data="effective_caller_id_name=${outbound_caller_id_name}"/>
>      <action application="bridge" data="sofia/gateway/asterisk/$1"/>
>    </condition>
>  </extension>
>
>  <extension name="spanish_fijos.mydomain.com">
>    <condition field="destination_number" expression="^(9\d{8})$">
>      <action application="set"
> data="effective_caller_id_number=${outbound_caller_id_number}"/>
>      <action application="set"
> data="effective_caller_id_name=${outbound_caller_id_name}"/>
>      <action application="bridge" data="sofia/gateway/asterisk/$1"/>
>    </condition>
>  </extension>
>
>  ... more stuff not shown ...
>
> </include>
>
> Which routes out to my existing asterisk installation.
>
> I now want to record all calls. Looking at
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session
> it seems I need to add a section like into each entry prior to the
> application="bridge".
>
>    <action application="set" data="gateway_name=XXXXX"/>
>
>    <action application="set" data="RECORD_TITLE=Recording
> ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/>
>    <action application="set" data="RECORD_COPYRIGHT=(C) 2009 my data
> here."/>
>    <action application="set" data="RECORD_SOFTWARE=FreeSwitch"/>
>    <action application="set" data="RECORD_ARTIST=My name"/>
>    <action application="set" data="RECORD_COMMENT=My comment"/>
>    <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d
> %H:%M)}"/>
>    <action application="set" data="RECORD_STEREO=true"/>
>    <action application="record_session"
> data="$${base_dir}/recordings/${gateway_name}_${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/>
>
> My question is how do I avoid repeatedly using these same 8-9 lines and
> "merge" this into some macro which given the right parameters allows me
> to define various extensions using different gateways but all configured
> "the same way"?
>
> Thanks for any pointers to how this should be done.
>

Try using the execute_extension app. It's kinda like a gosub in BASIC: it'll
go and execute an extension and then come back and continue executing. Put
all those lines into a single extension and then call them from other
extensions via execute_extension.
-MC
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From larclap at yahoo.com  Wed Oct 14 16:26:11 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Wed, 14 Oct 2009 16:26:11 -0700
Subject: [Freeswitch-users] Outbound sip address call fails
Message-ID: <00d701ca4d25$b7d615f0$278241d0$@com>

I am trying to make a test call to the FreeSwitch conference address,
sip:888 at conference.freeswitch.org on a Bria softphone. However, it fails
with a 404 Not found message.

 

I have struggled with this for a while. Is there something special I must do
on the Bria to make this happen correctly?

 

Thanks, Lars

 

http://pastebin.freeswitch.org/10715

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From orien at tx.rr.com  Wed Oct 14 16:31:43 2009
From: orien at tx.rr.com (Orien Love)
Date: Wed, 14 Oct 2009 18:31:43 -0500
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 129
In-Reply-To: <mailman.1259.1255550525.2495.freeswitch-users@lists.freeswitch.org>
References: <mailman.1259.1255550525.2495.freeswitch-users@lists.freeswitch.org>
Message-ID: <4AD65F5F.9090306@tx.rr.com>

Hello, a few weeks ago I asked about a setup, and about using the 
freeswitch package on pfSense. Since then I have received most of my 
hardware and started setting up freeswitch.

First my setup
WWW --> 1st pfsense box <-- private (192.168.2.xxx) --> pfsense w 
freeswtich <-- Test (192.168.3.xxx) --> phones

pfsense w freeswtich Port Addresses
WAN 192.168.2.210
LAN 192.168.3.1

this is setup to simulate a pfsense box hooked up to an external network.


here are the things I have done.

1st attempt
#1 set my WAN IP port to fixed in pfsense (192.168.2.210)
#2 set my soft phone to register to 192.168.2.210 (The phones registered 
and showed up in the status section)
#3 Tried to make a call to extension 1004, the soft phone tells me that 
it is "Temporally Unavailable 480).

2nd attempt
#1 set my WAN IP port to fixed in pfsense (192.168.2.210)
#2 deleted the lan.xml profile
#3 rebooted freeswitch.
#4 set my soft phone to register to 192.168.2.210 (The phones registered 
and showed up in the status section)
#5 Tried to make a call to extension 1004, the soft phone tells me that 
it is "Temporally Unavailable 480).

3rd attempt
#1 set my WAN IP port to fixed in pfsense (192.168.2.210)
#2 edited the internal.xml file as described in the multi-home tutorial 
(192.168.3.1 is my phone lan address)
    <param name="rtp-ip" value="192.168.3.1"/>
    <param name="sip-ip" value="192.168.3.1"/>
    <param name="ext-rtp-ip" value="192.168.3.1"/>
    <param name="ext-sip-ip" value="192.168.3.1"/>
#3 rebooted freeswitch.
#4 set my soft phone to register to 192.168.2.210 (The phone would not 
register)
#5 set my soft phone to register to 192.168.3.1 (The phone registered 
and showed up in the status section)
#6 Tried to make a call to extension 1004, the soft phone tells me that 
it is "Temporally Unavailable 480).
#7 edited the file /usr/local/freeswitch/conf/directory/default.xml
  <domain name="192.168.3.1">
#8 rebooted freeswitch.
#9 set my soft phone to register to 192.168.2.210 (The phone would not 
register)
#10 set my soft phone to register to 192.168.3.1 (The phone would not 
register got a forbidden error)
#11 did some reading and found a place in the vars tab to change the domain
#12 edited the file in the vars tab
  <X-PRE-PROCESS cmd="set" data="domain=192.168.3.1"/>
#13 rebooted freeswitch.
#14 set my soft phone to register to 192.168.2.210 (The phone would not 
register)
#15 set my soft phone to register to 192.168.3.1 (The phone registered 
and showed up in the status section)
#16 Tried to make a call to extension 1004, the soft phone tells me that 
it is "Temporally Unavailable 480).

here is part of the log file:
2009-10-14 22:18:43.560147 [DEBUG] sofia.c:910 Write lock lan
2009-10-14 22:18:43.563938 [NOTICE] sofia.c:917 Waiting for worker thread
2009-10-14 22:18:43.563938 [DEBUG] sofia.c:975 Write unlock lan
2009-10-14 22:26:00.23234 [DEBUG] sofia.c:4549 IP 192.168.3.199 Rejected 
by acl "domains". Falling back to Digest auth.
2009-10-14 22:26:00.55225 [DEBUG] sofia.c:4549 IP 192.168.3.199 Rejected 
by acl "domains". Falling back to Digest auth.
2009-10-14 22:26:00.55225 [NOTICE] switch_channel.c:602 New Channel 
sofia/internal/1001 at 192.168.2.210 [8aa7cd8c-10b9-de11-a0e9-00a0cc5a11e3]
2009-10-14 22:26:00.55225 [DEBUG] sofia.c:5196 Setting NAT mode based on 
rfc1918.auto
2009-10-14 22:26:00.55225 [DEBUG] sofia.c:3210 Channel 
sofia/internal/1001 at 192.168.2.210 entering state [received][100]
2009-10-14 22:26:00.55225 [DEBUG] sofia.c:3217 Remote SDP:
v=0
o=- 78902911 78902911 IN IP4 192.168.189.1
s=http://www.portsip.com
c=IN IP4 192.168.189.1
t=0 0
m=audio 21328 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
m=video 40216 RTP/AVP 34 125
a=rtpmap:34 H263/90000
a=fmtp:34 CIF=2
a=rtpmap:125 H264/90000
a=fmtp:125 profile-level-id=42e015

2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3081 Audio Codec Compare 
[PCMU:0:8000:20]/[G7221:115:32000:20]
2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3081 Audio Codec Compare 
[PCMU:0:8000:20]/[G7221:107:16000:20]
2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3081 Audio Codec Compare 
[PCMU:0:8000:20]/[G722:9:8000:20]
2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3081 Audio Codec Compare 
[PCMU:0:8000:20]/[PCMU:0:8000:20]
2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:2039 Set Codec 
sofia/internal/1001 at 192.168.2.210 PCMU/8000 20 ms 160 samples
2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3041 Set 2833 dtmf 
payload to 101
2009-10-14 22:26:00.55225 [DEBUG] sofia.c:3376 
(sofia/internal/1001 at 192.168.2.210) State Change CS_NEW -> CS_INIT
2009-10-14 22:26:00.55225 [DEBUG] switch_core_session.c:933 Send signal 
sofia/internal/1001 at 192.168.2.210 [BREAK]
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397 
(sofia/internal/1001 at 192.168.2.210) Running State Change CS_INIT
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:480 
(sofia/internal/1001 at 192.168.2.210) State INIT
2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:83 
sofia/internal/1001 at 192.168.2.210 SOFIA INIT
2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:111 
(sofia/internal/1001 at 192.168.2.210) State Change CS_INIT -> CS_ROUTING
2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:933 Send signal 
sofia/internal/1001 at 192.168.2.210 [BREAK]
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:480 
(sofia/internal/1001 at 192.168.2.210) State INIT going to sleep
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397 
(sofia/internal/1001 at 192.168.2.210) Running State Change CS_ROUTING
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:483 
(sofia/internal/1001 at 192.168.2.210) State ROUTING
2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:130 
sofia/internal/1001 at 192.168.2.210 SOFIA ROUTING
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:78 
sofia/internal/1001 at 192.168.2.210 Standard ROUTING
2009-10-14 22:26:00.63551 [INFO] mod_dialplan_xml.c:252 Processing 
1001->1004 in context public
Dialplan: sofia/internal/1001 at 192.168.2.210 parsing [public->unloop] 
continue=false
Dialplan: sofia/internal/1001 at 192.168.2.210 Regex (PASS) [unloop] 
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1001 at 192.168.2.210 Regex (FAIL) [unloop] 
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/1001 at 192.168.2.210 parsing 
[public->outside_call] continue=true
Dialplan: sofia/internal/1001 at 192.168.2.210 Absolute Condition 
[outside_call]
Dialplan: sofia/internal/1001 at 192.168.2.210 Action set(outside_call=true)
Dialplan: sofia/internal/1001 at 192.168.2.210 parsing [public->call_debug] 
continue=true
Dialplan: sofia/internal/1001 at 192.168.2.210 Regex (FAIL) [call_debug] 
${call_debug}(false) =~ /^true$/ break=never
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:114 
(sofia/internal/1001 at 192.168.2.210) State Change CS_ROUTING -> CS_EXECUTE
2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:933 Send signal 
sofia/internal/1001 at 192.168.2.210 [BREAK]
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:483 
(sofia/internal/1001 at 192.168.2.210) State ROUTING going to sleep
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397 
(sofia/internal/1001 at 192.168.2.210) Running State Change CS_EXECUTE
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:490 
(sofia/internal/1001 at 192.168.2.210) State EXECUTE
2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:173 
sofia/internal/1001 at 192.168.2.210 SOFIA EXECUTE
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:151 
sofia/internal/1001 at 192.168.2.210 Standard EXECUTE
EXECUTE sofia/internal/1001 at 192.168.2.210 set(outside_call=true)
2009-10-14 22:26:00.63551 [DEBUG] mod_dptools.c:748 
sofia/internal/1001 at 192.168.2.210 SET [outside_call]=[true]
2009-10-14 22:26:00.63551 [NOTICE] switch_core_state_machine.c:179 
Hangup sofia/internal/1001 at 192.168.2.210 [CS_EXECUTE] [NORMAL_CLEARING]
2009-10-14 22:26:00.63551 [DEBUG] switch_channel.c:1683 Send signal 
sofia/internal/1001 at 192.168.2.210 [KILL]
2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:933 Send signal 
sofia/internal/1001 at 192.168.2.210 [BREAK]
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:490 
(sofia/internal/1001 at 192.168.2.210) State EXECUTE going to sleep
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397 
(sofia/internal/1001 at 192.168.2.210) Running State Change CS_HANGUP
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:433 
(sofia/internal/1001 at 192.168.2.210) State HANGUP
2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:338 Channel 
sofia/internal/1001 at 192.168.2.210 hanging up, cause: NORMAL_CLEARING
2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:414 Responding to INVITE 
with: 480
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:46 
sofia/internal/1001 at 192.168.2.210 Standard HANGUP, cause: NORMAL_CLEARING
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:433 
(sofia/internal/1001 at 192.168.2.210) State HANGUP going to sleep
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:475 
(sofia/internal/1001 at 192.168.2.210) State Change CS_HANGUP -> CS_REPORTING
2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:933 Send signal 
sofia/internal/1001 at 192.168.2.210 [BREAK]
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397 
(sofia/internal/1001 at 192.168.2.210) Running State Change CS_REPORTING
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:607 
(sofia/internal/1001 at 192.168.2.210) State REPORTING
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:53 
sofia/internal/1001 at 192.168.2.210 Standard REPORTING, cause: NORMAL_CLEARING
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:607 
(sofia/internal/1001 at 192.168.2.210) State REPORTING going to sleep
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:410 
(sofia/internal/1001 at 192.168.2.210) State Change CS_REPORTING -> CS_DESTROY
2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:1067 Session 2 
(sofia/internal/1001 at 192.168.2.210) Locked, Waiting on external entities
2009-10-14 22:26:00.63551 [NOTICE] switch_core_session.c:1085 Session 2 
(sofia/internal/1001 at 192.168.2.210) Ended
2009-10-14 22:26:00.63551 [NOTICE] switch_core_session.c:1087 Close 
Channel sofia/internal/1001 at 192.168.2.210 [CS_DESTROY]
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:559 
(sofia/internal/1001 at 192.168.2.210) State DESTROY
2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:255 
sofia/internal/1001 at 192.168.2.210 SOFIA DESTROY
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:60 
sofia/internal/1001 at 192.168.2.210 Standard DESTROY
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:559 
(sofia/internal/1001 at 192.168.2.210) State DESTROY going to sleep
2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:559 
(sofia/internal/1001 at 192.168.2.210) State DESTROY going to sleep

I am at a loss, any help would be welcome.

Orien Love





From Mailings at kh-dev.de  Wed Oct 14 16:48:09 2009
From: Mailings at kh-dev.de (Klaus Hochlehnert)
Date: Thu, 15 Oct 2009 01:48:09 +0200
Subject: [Freeswitch-users] Strange problems after upgrade...
Message-ID: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330252@srv02.khdev.corp>

Hi,

I just upgraded FS from 15094 to 15161 and after the upgrade I have the following problem:
I use t38modem together with a SIP/ISDN gateway.

With the old release everything fax related works fine. I call the fax -> I hear a fax tone.
With the new release I call the fax -> I don't hear a fax tone. The modem answers the call but there's no media.

I use proxy_media=true, bypass_media=false and inbound-late-negotiation=true.

Is there anything else I need to do with the new release?

Thanks, Klaus
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From Mailings at kh-dev.de  Wed Oct 14 16:56:32 2009
From: Mailings at kh-dev.de (Klaus Hochlehnert)
Date: Thu, 15 Oct 2009 01:56:32 +0200
Subject: [Freeswitch-users] Media got stuck after attended transfer...
Message-ID: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330253@srv02.khdev.corp>

Hi,

sometimes I have the problem that after doing an attended transfer the media got stuck in FS.
Meaning the call goes through, but I don't hear anything and the caller still hears music.

Now I found out that setting the sofia parameter  media-option=resume-media-on-hold  helps here.
But after setting this parameter I always get the error "Codec PROXY PASS-THROUGH encoder error!" when using t38modem with proxy_media=true.

So I stuck here. I can either use fax or do attended transfers.

Does anyone have a solution for this?

Thanks, Klaus
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From msc at freeswitch.org  Wed Oct 14 17:03:18 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 14 Oct 2009 17:03:18 -0700
Subject: [Freeswitch-users] Strange problems after upgrade...
In-Reply-To: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330252@srv02.khdev.corp>
References: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330252@srv02.khdev.corp>
Message-ID: <87f2f3b90910141703hc8311bfg8c1d1280708d70f0@mail.gmail.com>

On Wed, Oct 14, 2009 at 4:48 PM, Klaus Hochlehnert <Mailings at kh-dev.de>wrote:

>  Hi,
>
>
>
> I just upgraded FS from 15094 to 15161 and after the upgrade I have the
> following problem:
>
> I use t38modem together with a SIP/ISDN gateway.
>
>
>
> With the old release everything fax related works fine. I call the fax -> I
> hear a fax tone.
>
> With the new release I call the fax -> I don?t hear a fax tone. The modem
> answers the call but there?s no media.
>
>
>
> I use proxy_media=true, bypass_media=false and
> inbound-late-negotiation=true.
>
>
>
> Is there anything else I need to do with the new release?
>
> How exactly did you update?
-MC
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From Mailings at kh-dev.de  Wed Oct 14 17:14:10 2009
From: Mailings at kh-dev.de (Klaus Hochlehnert)
Date: Thu, 15 Oct 2009 02:14:10 +0200
Subject: [Freeswitch-users] Strange problems after upgrade...
In-Reply-To: <87f2f3b90910141703hc8311bfg8c1d1280708d70f0@mail.gmail.com>
References: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330252@srv02.khdev.corp>
	<87f2f3b90910141703hc8311bfg8c1d1280708d70f0@mail.gmail.com>
Message-ID: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330254@srv02.khdev.corp>

Basically I did an normal make + make install

The conf and the scripts dir were untouched after that (which I wanted because I didn't want to reconfigure everything).

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Thursday, October 15, 2009 2:03 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Strange problems after upgrade...


On Wed, Oct 14, 2009 at 4:48 PM, Klaus Hochlehnert <Mailings at kh-dev.de<mailto:Mailings at kh-dev.de>> wrote:
Hi,

I just upgraded FS from 15094 to 15161 and after the upgrade I have the following problem:
I use t38modem together with a SIP/ISDN gateway.

With the old release everything fax related works fine. I call the fax -> I hear a fax tone.
With the new release I call the fax -> I don't hear a fax tone. The modem answers the call but there's no media.

I use proxy_media=true, bypass_media=false and inbound-late-negotiation=true.

Is there anything else I need to do with the new release?
How exactly did you update?
-MC

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From mgende at gendesign.com  Wed Oct 14 19:57:57 2009
From: mgende at gendesign.com (Michael Gende)
Date: Wed, 14 Oct 2009 21:57:57 -0500
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 129
In-Reply-To: <4AD65F5F.9090306@tx.rr.com>
References: <mailman.1259.1255550525.2495.freeswitch-users@lists.freeswitch.org>
	<4AD65F5F.9090306@tx.rr.com>
Message-ID: <a87d43fd0910141957t13523f73p580847d7e5875288@mail.gmail.com>

Hey Orien,

All of these guys are more experienced than I, but I'll try to throw you a
line.

I'll have to check a thing or two out at work, but here's an observation or
two (I'll post again tomorrow morning after I get in).

The mult-homed example is set to work in a very specifc way: A dual-homed
box where the WAN is used for the WWW connection, and then using the LAN
side for phones.

To work like the example, you need to have your phones on the LAN network.
Then, configure your internal.xml file on FS to use the LAN port, not the
WAN. Next, set your phones to register to that LAN address.

One thing I prefer to do on pfsense, is at the FS box console (or ssh to the
box), kill the "pfsense FS" running the background, and run FS in the
foreground. Then you can "reloadxml" instead of rebooting when XML changes
are made (if I understand your post).

As I said, let me check a thing or two out, and I'll ask to see some
configuration files. I'm sure we can get you going.

BTW: What kind of phones are you using?

Regards,

Mike G.


On Wed, Oct 14, 2009 at 6:31 PM, Orien Love <orien at tx.rr.com> wrote:

> Hello, a few weeks ago I asked about a setup, and about using the
> freeswitch package on pfSense. Since then I have received most of my
> hardware and started setting up freeswitch.
>
> First my setup
> WWW --> 1st pfsense box <-- private (192.168.2.xxx) --> pfsense w
> freeswtich <-- Test (192.168.3.xxx) --> phones
>
> pfsense w freeswtich Port Addresses
> WAN 192.168.2.210
> LAN 192.168.3.1
>
> this is setup to simulate a pfsense box hooked up to an external network.
>
>
> here are the things I have done.
>
> 1st attempt
> #1 set my WAN IP port to fixed in pfsense (192.168.2.210)
> #2 set my soft phone to register to 192.168.2.210 (The phones registered
> and showed up in the status section)
> #3 Tried to make a call to extension 1004, the soft phone tells me that
> it is "Temporally Unavailable 480).
>
> 2nd attempt
> #1 set my WAN IP port to fixed in pfsense (192.168.2.210)
> #2 deleted the lan.xml profile
> #3 rebooted freeswitch.
> #4 set my soft phone to register to 192.168.2.210 (The phones registered
> and showed up in the status section)
> #5 Tried to make a call to extension 1004, the soft phone tells me that
> it is "Temporally Unavailable 480).
>
> 3rd attempt
> #1 set my WAN IP port to fixed in pfsense (192.168.2.210)
> #2 edited the internal.xml file as described in the multi-home tutorial
> (192.168.3.1 is my phone lan address)
>    <param name="rtp-ip" value="192.168.3.1"/>
>    <param name="sip-ip" value="192.168.3.1"/>
>    <param name="ext-rtp-ip" value="192.168.3.1"/>
>    <param name="ext-sip-ip" value="192.168.3.1"/>
> #3 rebooted freeswitch.
> #4 set my soft phone to register to 192.168.2.210 (The phone would not
> register)
> #5 set my soft phone to register to 192.168.3.1 (The phone registered
> and showed up in the status section)
> #6 Tried to make a call to extension 1004, the soft phone tells me that
> it is "Temporally Unavailable 480).
> #7 edited the file /usr/local/freeswitch/conf/directory/default.xml
>  <domain name="192.168.3.1">
> #8 rebooted freeswitch.
> #9 set my soft phone to register to 192.168.2.210 (The phone would not
> register)
> #10 set my soft phone to register to 192.168.3.1 (The phone would not
> register got a forbidden error)
> #11 did some reading and found a place in the vars tab to change the domain
> #12 edited the file in the vars tab
>  <X-PRE-PROCESS cmd="set" data="domain=192.168.3.1"/>
> #13 rebooted freeswitch.
> #14 set my soft phone to register to 192.168.2.210 (The phone would not
> register)
> #15 set my soft phone to register to 192.168.3.1 (The phone registered
> and showed up in the status section)
> #16 Tried to make a call to extension 1004, the soft phone tells me that
> it is "Temporally Unavailable 480).
>
> here is part of the log file:
> 2009-10-14 22:18:43.560147 [DEBUG] sofia.c:910 Write lock lan
> 2009-10-14 22:18:43.563938 [NOTICE] sofia.c:917 Waiting for worker thread
> 2009-10-14 22:18:43.563938 [DEBUG] sofia.c:975 Write unlock lan
> 2009-10-14 22:26:00.23234 [DEBUG] sofia.c:4549 IP 192.168.3.199 Rejected
> by acl "domains". Falling back to Digest auth.
> 2009-10-14 22:26:00.55225 [DEBUG] sofia.c:4549 IP 192.168.3.199 Rejected
> by acl "domains". Falling back to Digest auth.
> 2009-10-14 22:26:00.55225 [NOTICE] switch_channel.c:602 New Channel
> sofia/internal/1001 at 192.168.2.210 [8aa7cd8c-10b9-de11-a0e9-00a0cc5a11e3]
> 2009-10-14 22:26:00.55225 [DEBUG] sofia.c:5196 Setting NAT mode based on
> rfc1918.auto
> 2009-10-14 22:26:00.55225 [DEBUG] sofia.c:3210 Channel
> sofia/internal/1001 at 192.168.2.210 entering state [received][100]
> 2009-10-14 22:26:00.55225 [DEBUG] sofia.c:3217 Remote SDP:
> v=0
> o=- 78902911 78902911 IN IP4 192.168.189.1
> s=http://www.portsip.com
> c=IN IP4 192.168.189.1
> t=0 0
> m=audio 21328 RTP/AVP 0 8 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
> m=video 40216 RTP/AVP 34 125
> a=rtpmap:34 H263/90000
> a=fmtp:34 CIF=2
> a=rtpmap:125 H264/90000
> a=fmtp:125 profile-level-id=42e015
>
> 2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3081 Audio Codec Compare
> [PCMU:0:8000:20]/[G7221:115:32000:20]
> 2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3081 Audio Codec Compare
> [PCMU:0:8000:20]/[G7221:107:16000:20]
> 2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3081 Audio Codec Compare
> [PCMU:0:8000:20]/[G722:9:8000:20]
> 2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3081 Audio Codec Compare
> [PCMU:0:8000:20]/[PCMU:0:8000:20]
> 2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:2039 Set Codec
> sofia/internal/1001 at 192.168.2.210 PCMU/8000 20 ms 160 samples
> 2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3041 Set 2833 dtmf
> payload to 101
> 2009-10-14 22:26:00.55225 [DEBUG] sofia.c:3376
> (sofia/internal/1001 at 192.168.2.210) State Change CS_NEW -> CS_INIT
> 2009-10-14 22:26:00.55225 [DEBUG] switch_core_session.c:933 Send signal
> sofia/internal/1001 at 192.168.2.210 [BREAK]
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397
> (sofia/internal/1001 at 192.168.2.210) Running State Change CS_INIT
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:480
> (sofia/internal/1001 at 192.168.2.210) State INIT
> 2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:83
> sofia/internal/1001 at 192.168.2.210 SOFIA INIT
> 2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:111
> (sofia/internal/1001 at 192.168.2.210) State Change CS_INIT -> CS_ROUTING
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:933 Send signal
> sofia/internal/1001 at 192.168.2.210 [BREAK]
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:480
> (sofia/internal/1001 at 192.168.2.210) State INIT going to sleep
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397
> (sofia/internal/1001 at 192.168.2.210) Running State Change CS_ROUTING
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:483
> (sofia/internal/1001 at 192.168.2.210) State ROUTING
> 2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:130
> sofia/internal/1001 at 192.168.2.210 SOFIA ROUTING
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:78
> sofia/internal/1001 at 192.168.2.210 Standard ROUTING
> 2009-10-14 22:26:00.63551 [INFO] mod_dialplan_xml.c:252 Processing
> 1001->1004 in context public
> Dialplan: sofia/internal/1001 at 192.168.2.210 parsing [public->unloop]
> continue=false
> Dialplan: sofia/internal/1001 at 192.168.2.210 Regex (PASS) [unloop]
> ${unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/internal/1001 at 192.168.2.210 Regex (FAIL) [unloop]
> ${sip_looped_call}() =~ /^true$/ break=on-false
> Dialplan: sofia/internal/1001 at 192.168.2.210 parsing
> [public->outside_call] continue=true
> Dialplan: sofia/internal/1001 at 192.168.2.210 Absolute Condition
> [outside_call]
> Dialplan: sofia/internal/1001 at 192.168.2.210 Action set(outside_call=true)
> Dialplan: sofia/internal/1001 at 192.168.2.210 parsing [public->call_debug]
> continue=true
> Dialplan: sofia/internal/1001 at 192.168.2.210 Regex (FAIL) [call_debug]
> ${call_debug}(false) =~ /^true$/ break=never
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:114
> (sofia/internal/1001 at 192.168.2.210) State Change CS_ROUTING -> CS_EXECUTE
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:933 Send signal
> sofia/internal/1001 at 192.168.2.210 [BREAK]
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:483
> (sofia/internal/1001 at 192.168.2.210) State ROUTING going to sleep
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397
> (sofia/internal/1001 at 192.168.2.210) Running State Change CS_EXECUTE
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:490
> (sofia/internal/1001 at 192.168.2.210) State EXECUTE
> 2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:173
> sofia/internal/1001 at 192.168.2.210 SOFIA EXECUTE
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:151
> sofia/internal/1001 at 192.168.2.210 Standard EXECUTE
> EXECUTE sofia/internal/1001 at 192.168.2.210 set(outside_call=true)
> 2009-10-14 22:26:00.63551 [DEBUG] mod_dptools.c:748
> sofia/internal/1001 at 192.168.2.210 SET [outside_call]=[true]
> 2009-10-14 22:26:00.63551 [NOTICE] switch_core_state_machine.c:179
> Hangup sofia/internal/1001 at 192.168.2.210 [CS_EXECUTE] [NORMAL_CLEARING]
> 2009-10-14 22:26:00.63551 [DEBUG] switch_channel.c:1683 Send signal
> sofia/internal/1001 at 192.168.2.210 [KILL]
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:933 Send signal
> sofia/internal/1001 at 192.168.2.210 [BREAK]
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:490
> (sofia/internal/1001 at 192.168.2.210) State EXECUTE going to sleep
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397
> (sofia/internal/1001 at 192.168.2.210) Running State Change CS_HANGUP
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:433
> (sofia/internal/1001 at 192.168.2.210) State HANGUP
> 2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:338 Channel
> sofia/internal/1001 at 192.168.2.210 hanging up, cause: NORMAL_CLEARING
> 2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:414 Responding to INVITE
> with: 480
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/1001 at 192.168.2.210 Standard HANGUP, cause: NORMAL_CLEARING
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:433
> (sofia/internal/1001 at 192.168.2.210) State HANGUP going to sleep
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:475
> (sofia/internal/1001 at 192.168.2.210) State Change CS_HANGUP -> CS_REPORTING
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:933 Send signal
> sofia/internal/1001 at 192.168.2.210 [BREAK]
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397
> (sofia/internal/1001 at 192.168.2.210) Running State Change CS_REPORTING
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:607
> (sofia/internal/1001 at 192.168.2.210) State REPORTING
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/1001 at 192.168.2.210 Standard REPORTING, cause:
> NORMAL_CLEARING
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:607
> (sofia/internal/1001 at 192.168.2.210) State REPORTING going to sleep
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:410
> (sofia/internal/1001 at 192.168.2.210) State Change CS_REPORTING ->
> CS_DESTROY
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:1067 Session 2
> (sofia/internal/1001 at 192.168.2.210) Locked, Waiting on external entities
> 2009-10-14 22:26:00.63551 [NOTICE] switch_core_session.c:1085 Session 2
> (sofia/internal/1001 at 192.168.2.210) Ended
> 2009-10-14 22:26:00.63551 [NOTICE] switch_core_session.c:1087 Close
> Channel sofia/internal/1001 at 192.168.2.210 [CS_DESTROY]
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:559
> (sofia/internal/1001 at 192.168.2.210) State DESTROY
> 2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:255
> sofia/internal/1001 at 192.168.2.210 SOFIA DESTROY
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:60
> sofia/internal/1001 at 192.168.2.210 Standard DESTROY
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:559
> (sofia/internal/1001 at 192.168.2.210) State DESTROY going to sleep
> 2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:559
> (sofia/internal/1001 at 192.168.2.210) State DESTROY going to sleep
>
> I am at a loss, any help would be welcome.
>
> Orien Love
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From orien at tx.rr.com  Wed Oct 14 20:30:03 2009
From: orien at tx.rr.com (Orien Love)
Date: Wed, 14 Oct 2009 22:30:03 -0500
Subject: [Freeswitch-users] FreeSWITCH-users Digest,
	Mike G. answer to Orien L
In-Reply-To: <mailman.1269.1255575509.2495.freeswitch-users@lists.freeswitch.org>
References: <mailman.1269.1255575509.2495.freeswitch-users@lists.freeswitch.org>
Message-ID: <4AD6973B.70106@tx.rr.com>

Mike,
     Thank you for your quick response, this has been driving me nuts.

in my third attempt I tried to follow the mulit home example. I did set 
the phones to register to the local lan IP. actually had them 
registering, but it did help it in any way still the same problem.
After my post I reinstalled the package and set the firewall to allow 
phones on the wan to connect, as per 
http://doc.pfsense.org/index.php/FreeSWITCH#Firewall
put my computer with the soft phone on the wan network (192,168.2.xxx) 
had it registering and it had the same problem. so far all the setups I 
have tried are not working, the phones register, but cannot 
dial/connect  another  phone, also does not go to voice mail, maybe it 
is in my dial plan. which is default.

Shoot, I am not picky, I just want to get it to work. so any suggestions 
on how to accomplish this are greatly appreciated.

My phones are currently a couple of polycom 601's, and a Pangolin Softphone.

Thanks
   Orien L.

>
> ------------------------------------------------------------------------
>
> Subject:
> Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 129
> From:
> Michael Gende <mgende at gendesign.com>
> Date:
> Wed, 14 Oct 2009 21:57:57 -0500
> To:
> freeswitch-users at lists.freeswitch.org
>
> To:
> freeswitch-users at lists.freeswitch.org
>
>
> Hey Orien,
>
> All of these guys are more experienced than I, but I'll try to throw 
> you a line.
>
> I'll have to check a thing or two out at work, but here's an 
> observation or two (I'll post again tomorrow morning after I get in).
>
> The mult-homed example is set to work in a very specifc way: A 
> dual-homed box where the WAN is used for the WWW connection, and then 
> using the LAN side for phones.
>
> To work like the example, you need to have your phones on the LAN 
> network. Then, configure your internal.xml file on FS to use the LAN 
> port, not the WAN. Next, set your phones to register to that LAN 
> address. 
>
> One thing I prefer to do on pfsense, is at the FS box console (or ssh 
> to the box), kill the "pfsense FS" running the background, and run FS 
> in the foreground. Then you can "reloadxml" instead of rebooting when 
> XML changes are made (if I understand your post).
>
> As I said, let me check a thing or two out, and I'll ask to see some 
> configuration files. I'm sure we can get you going.
>
> BTW: What kind of phones are you using?
>
> Regards,
>
> Mike G.
>
>
> On Wed, Oct 14, 2009 at 6:31 PM, Orien Love <orien at tx.rr.com 
> <mailto:orien at tx.rr.com>> wrote:
>
>     Hello, a few weeks ago I asked about a setup, and about using the
>     freeswitch package on pfSense. Since then I have received most of my
>     hardware and started setting up freeswitch.
>
>     First my setup
>     WWW --> 1st pfsense box <-- private (192.168.2.xxx) --> pfsense w
>     freeswtich <-- Test (192.168.3.xxx) --> phones
>
>     pfsense w freeswtich Port Addresses
>     WAN 192.168.2.210
>     LAN 192.168.3.1
>
>     this is setup to simulate a pfsense box hooked up to an external
>     network.
>
>
>     here are the things I have done.
>
>     1st attempt
>     #1 set my WAN IP port to fixed in pfsense (192.168.2.210)
>     #2 set my soft phone to register to 192.168.2.210 (The phones
>     registered
>     and showed up in the status section)
>     #3 Tried to make a call to extension 1004, the soft phone tells me
>     that
>     it is "Temporally Unavailable 480).
>
>     2nd attempt
>     #1 set my WAN IP port to fixed in pfsense (192.168.2.210)
>     #2 deleted the lan.xml profile
>     #3 rebooted freeswitch.
>     #4 set my soft phone to register to 192.168.2.210 (The phones
>     registered
>     and showed up in the status section)
>     #5 Tried to make a call to extension 1004, the soft phone tells me
>     that
>     it is "Temporally Unavailable 480).
>
>     3rd attempt
>     #1 set my WAN IP port to fixed in pfsense (192.168.2.210)
>     #2 edited the internal.xml file as described in the multi-home
>     tutorial
>     (192.168.3.1 is my phone lan address)
>        <param name="rtp-ip" value="192.168.3.1"/>
>        <param name="sip-ip" value="192.168.3.1"/>
>        <param name="ext-rtp-ip" value="192.168.3.1"/>
>        <param name="ext-sip-ip" value="192.168.3.1"/>
>     #3 rebooted freeswitch.
>     #4 set my soft phone to register to 192.168.2.210 (The phone would not
>     register)
>     #5 set my soft phone to register to 192.168.3.1 (The phone registered
>     and showed up in the status section)
>     #6 Tried to make a call to extension 1004, the soft phone tells me
>     that
>     it is "Temporally Unavailable 480).
>     #7 edited the file /usr/local/freeswitch/conf/directory/default.xml
>      <domain name="192.168.3.1">
>     #8 rebooted freeswitch.
>     #9 set my soft phone to register to 192.168.2.210 (The phone would not
>     register)
>     #10 set my soft phone to register to 192.168.3.1 (The phone would not
>     register got a forbidden error)
>     #11 did some reading and found a place in the vars tab to change
>     the domain
>     #12 edited the file in the vars tab
>      <X-PRE-PROCESS cmd="set" data="domain=192.168.3.1"/>
>     #13 rebooted freeswitch.
>     #14 set my soft phone to register to 192.168.2.210 (The phone
>     would not
>     register)
>     #15 set my soft phone to register to 192.168.3.1 (The phone registered
>     and showed up in the status section)
>     #16 Tried to make a call to extension 1004, the soft phone tells
>     me that
>     it is "Temporally Unavailable 480).
>
>     here is part of the log file:
>     2009-10-14 22:18:43.560147 [DEBUG] sofia.c:910 Write lock lan
>     2009-10-14 22:18:43.563938 [NOTICE] sofia.c:917 Waiting for worker
>     thread
>     2009-10-14 22:18:43.563938 [DEBUG] sofia.c:975 Write unlock lan
>     2009-10-14 22:26:00.23234 [DEBUG] sofia.c:4549 IP 192.168.3.199
>     Rejected
>     by acl "domains". Falling back to Digest auth.
>     2009-10-14 22:26:00.55225 [DEBUG] sofia.c:4549 IP 192.168.3.199
>     Rejected
>     by acl "domains". Falling back to Digest auth.
>     2009-10-14 22:26:00.55225 [NOTICE] switch_channel.c:602 New Channel
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>
>     [8aa7cd8c-10b9-de11-a0e9-00a0cc5a11e3]
>     2009-10-14 22:26:00.55225 [DEBUG] sofia.c:5196 Setting NAT mode
>     based on
>     rfc1918.auto
>     2009-10-14 22:26:00.55225 [DEBUG] sofia.c:3210 Channel
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>
>     entering state [received][100]
>     2009-10-14 22:26:00.55225 [DEBUG] sofia.c:3217 Remote SDP:
>     v=0
>     o=- 78902911 78902911 IN IP4 192.168.189.1
>     s=http://www.portsip.com
>     c=IN IP4 192.168.189.1
>     t=0 0
>     m=audio 21328 RTP/AVP 0 8 18 101
>     a=rtpmap:0 PCMU/8000
>     a=rtpmap:8 PCMA/8000
>     a=rtpmap:18 G729/8000
>     a=fmtp:18 annexb=no
>     a=rtpmap:101 telephone-event/8000
>     a=fmtp:101 0-15
>     a=ptime:20
>     m=video 40216 RTP/AVP 34 125
>     a=rtpmap:34 H263/90000
>     a=fmtp:34 CIF=2
>     a=rtpmap:125 H264/90000
>     a=fmtp:125 profile-level-id=42e015
>
>     2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3081 Audio Codec
>     Compare
>     [PCMU:0:8000:20]/[G7221:115:32000:20]
>     2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3081 Audio Codec
>     Compare
>     [PCMU:0:8000:20]/[G7221:107:16000:20]
>     2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3081 Audio Codec
>     Compare
>     [PCMU:0:8000:20]/[G722:9:8000:20]
>     2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3081 Audio Codec
>     Compare
>     [PCMU:0:8000:20]/[PCMU:0:8000:20]
>     2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:2039 Set Codec
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>
>     PCMU/8000 20 ms 160 samples
>     2009-10-14 22:26:00.55225 [DEBUG] sofia_glue.c:3041 Set 2833 dtmf
>     payload to 101
>     2009-10-14 22:26:00.55225 [DEBUG] sofia.c:3376
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State Change CS_NEW -> CS_INIT
>     2009-10-14 22:26:00.55225 [DEBUG] switch_core_session.c:933 Send
>     signal
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210> [BREAK]
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     Running State Change CS_INIT
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:480
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State INIT
>     2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:83
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>
>     SOFIA INIT
>     2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:111
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State Change CS_INIT -> CS_ROUTING
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:933 Send
>     signal
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210> [BREAK]
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:480
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State INIT going to sleep
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     Running State Change CS_ROUTING
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:483
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State ROUTING
>     2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:130
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>
>     SOFIA ROUTING
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:78
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>
>     Standard ROUTING
>     2009-10-14 22:26:00.63551 [INFO] mod_dialplan_xml.c:252 Processing
>     1001->1004 in context public
>     Dialplan: sofia/internal/1001 at 192.168.2.210
>     <mailto:1001 at 192.168.2.210> parsing [public->unloop]
>     continue=false
>     Dialplan: sofia/internal/1001 at 192.168.2.210
>     <mailto:1001 at 192.168.2.210> Regex (PASS) [unloop]
>     ${unroll_loops}(true) =~ /^true$/ break=on-false
>     Dialplan: sofia/internal/1001 at 192.168.2.210
>     <mailto:1001 at 192.168.2.210> Regex (FAIL) [unloop]
>     ${sip_looped_call}() =~ /^true$/ break=on-false
>     Dialplan: sofia/internal/1001 at 192.168.2.210
>     <mailto:1001 at 192.168.2.210> parsing
>     [public->outside_call] continue=true
>     Dialplan: sofia/internal/1001 at 192.168.2.210
>     <mailto:1001 at 192.168.2.210> Absolute Condition
>     [outside_call]
>     Dialplan: sofia/internal/1001 at 192.168.2.210
>     <mailto:1001 at 192.168.2.210> Action set(outside_call=true)
>     Dialplan: sofia/internal/1001 at 192.168.2.210
>     <mailto:1001 at 192.168.2.210> parsing [public->call_debug]
>     continue=true
>     Dialplan: sofia/internal/1001 at 192.168.2.210
>     <mailto:1001 at 192.168.2.210> Regex (FAIL) [call_debug]
>     ${call_debug}(false) =~ /^true$/ break=never
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:114
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State Change CS_ROUTING -> CS_EXECUTE
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:933 Send
>     signal
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210> [BREAK]
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:483
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State ROUTING going to sleep
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     Running State Change CS_EXECUTE
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:490
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State EXECUTE
>     2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:173
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>
>     SOFIA EXECUTE
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:151
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>
>     Standard EXECUTE
>     EXECUTE sofia/internal/1001 at 192.168.2.210
>     <mailto:1001 at 192.168.2.210> set(outside_call=true)
>     2009-10-14 22:26:00.63551 [DEBUG] mod_dptools.c:748
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210> SET
>     [outside_call]=[true]
>     2009-10-14 22:26:00.63551 [NOTICE] switch_core_state_machine.c:179
>     Hangup sofia/internal/1001 at 192.168.2.210
>     <mailto:1001 at 192.168.2.210> [CS_EXECUTE] [NORMAL_CLEARING]
>     2009-10-14 22:26:00.63551 [DEBUG] switch_channel.c:1683 Send signal
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210> [KILL]
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:933 Send
>     signal
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210> [BREAK]
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:490
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State EXECUTE going to sleep
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     Running State Change CS_HANGUP
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:433
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State HANGUP
>     2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:338 Channel
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>
>     hanging up, cause: NORMAL_CLEARING
>     2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:414 Responding to INVITE
>     with: 480
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:46
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>
>     Standard HANGUP, cause: NORMAL_CLEARING
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:433
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State HANGUP going to sleep
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:475
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State Change CS_HANGUP -> CS_REPORTING
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:933 Send
>     signal
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210> [BREAK]
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:397
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     Running State Change CS_REPORTING
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:607
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State REPORTING
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:53
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>
>     Standard REPORTING, cause: NORMAL_CLEARING
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:607
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State REPORTING going to sleep
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:410
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State Change CS_REPORTING -> CS_DESTROY
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_session.c:1067 Session 2
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     Locked, Waiting on external entities
>     2009-10-14 22:26:00.63551 [NOTICE] switch_core_session.c:1085
>     Session 2
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>) Ended
>     2009-10-14 22:26:00.63551 [NOTICE] switch_core_session.c:1087 Close
>     Channel sofia/internal/1001 at 192.168.2.210
>     <mailto:1001 at 192.168.2.210> [CS_DESTROY]
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:559
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State DESTROY
>     2009-10-14 22:26:00.63551 [DEBUG] mod_sofia.c:255
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>
>     SOFIA DESTROY
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:60
>     sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>
>     Standard DESTROY
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:559
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State DESTROY going to sleep
>     2009-10-14 22:26:00.63551 [DEBUG] switch_core_state_machine.c:559
>     (sofia/internal/1001 at 192.168.2.210 <mailto:1001 at 192.168.2.210>)
>     State DESTROY going to sleep
>
>     I am at a loss, any help would be welcome.
>
>     Orien Love
>
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   




From brian at freeswitch.org  Wed Oct 14 20:55:14 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 14 Oct 2009 20:55:14 -0700
Subject: [Freeswitch-users] Outbound sip address call fails
In-Reply-To: <00d701ca4d25$b7d615f0$278241d0$@com>
References: <00d701ca4d25$b7d615f0$278241d0$@com>
Message-ID: <8A5CCD06-A37C-423B-807B-01F24D537E49@freeswitch.org>

Sounds like bria is sending the invite via the proxy which is your  
local FS box and NOT direct.

/b

On Oct 14, 2009, at 4:26 PM, Lars Zeb wrote:

> I am trying to make a test call to the FreeSwitch conference  
> address, sip:888 at conference.freeswitch.org on a Bria softphone.  
> However, it fails with a 404 Not found message.
>
> I have struggled with this for a while. Is there something special I  
> must do on the Bria to make this happen correctly?
>
> Thanks, Lars
>
> http://pastebin.freeswitch.org/10715

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From brian at freeswitch.org  Wed Oct 14 20:55:48 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 14 Oct 2009 20:55:48 -0700
Subject: [Freeswitch-users] Media got stuck after attended transfer...
In-Reply-To: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330253@srv02.khdev.corp>
References: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330253@srv02.khdev.corp>
Message-ID: <EEB6A298-BCA7-45C5-9550-E0DB43162093@freeswitch.org>

Why exactly are you doing proxy media?

/b

On Oct 14, 2009, at 4:56 PM, Klaus Hochlehnert wrote:

> Does anyone have a solution for this?

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From brian at freeswitch.org  Wed Oct 14 20:56:15 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 14 Oct 2009 20:56:15 -0700
Subject: [Freeswitch-users] Strange problems after upgrade...
In-Reply-To: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330254@srv02.khdev.corp>
References: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330252@srv02.khdev.corp>
	<87f2f3b90910141703hc8311bfg8c1d1280708d70f0@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F04027F02330254@srv02.khdev.corp>
Message-ID: <1ADD58A4-E64B-4E16-9366-B205FBB33980@freeswitch.org>

Make current will NOT touch your configs and is designed to prevent  
build skew.

/b

On Oct 14, 2009, at 5:14 PM, Klaus Hochlehnert wrote:

> The conf and the scripts dir were untouched after that (which I  
> wanted because I didn?t want to reconfigure everything).

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From mgende at gendesign.com  Wed Oct 14 21:34:33 2009
From: mgende at gendesign.com (Michael Gende)
Date: Wed, 14 Oct 2009 23:34:33 -0500
Subject: [Freeswitch-users] FreeSWITCH-users Digest,
	Mike G. answer to 	Orien L
In-Reply-To: <4AD6973B.70106@tx.rr.com>
References: <mailman.1269.1255575509.2495.freeswitch-users@lists.freeswitch.org>
	<4AD6973B.70106@tx.rr.com>
Message-ID: <a87d43fd0910142134v366551a1p52f74e436f7f5709@mail.gmail.com>

I'll get to you in the morning (where do you live?) and we'll see what's
what.


On Wed, Oct 14, 2009 at 10:30 PM, Orien Love <orien at tx.rr.com> wrote:

> Mike,
>     Thank you for your quick response, this has been driving me nuts.
>
> in my third attempt I tried to follow the mulit home example. I did set
> the phones to register to the local lan IP. actually had them
> registering, but it did help it in any way still the same problem.
> After my post I reinstalled the package and set the firewall to allow
> phones on the wan to connect, as per
> http://doc.pfsense.org/index.php/FreeSWITCH#Firewall
> put my computer with the soft phone on the wan network (192,168.2.xxx)
> had it registering and it had the same problem. so far all the setups I
> have tried are not working, the phones register, but cannot
> dial/connect  another  phone, also does not go to voice mail, maybe it
> is in my dial plan. which is default.
>
> Shoot, I am not picky, I just want to get it to work. so any suggestions
> on how to accomplish this are greatly appreciated.
>
> My phones are currently a couple of polycom 601's, and a Pangolin
> Softphone.
>
> Thanks
>   Orien L.
>
>
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From msc at freeswitch.org  Wed Oct 14 21:38:24 2009
From: msc at freeswitch.org (Michael S Collins)
Date: Wed, 14 Oct 2009 21:38:24 -0700
Subject: [Freeswitch-users] Strange problems after upgrade...
In-Reply-To: <1ADD58A4-E64B-4E16-9366-B205FBB33980@freeswitch.org>
References: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330252@srv02.khdev.corp>
	<87f2f3b90910141703hc8311bfg8c1d1280708d70f0@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F04027F02330254@srv02.khdev.corp>
	<1ADD58A4-E64B-4E16-9366-B205FBB33980@freeswitch.org>
Message-ID: <3EFB01FB-6990-4D50-9A2A-0FE42CCE4C4E@freeswitch.org>

Everyone repeat after me:
make current is my friend
make current is my friend
make current is my friend

ALWAYS use make current unless you know more about FreeSWITCH than the  
project's authors do.

-MC

On Oct 14, 2009, at 8:56 PM, Brian West <brian at freeswitch.org> wrote:

> Make current will NOT touch your configs and is designed to prevent  
> build skew.
>
> /b
>
> On Oct 14, 2009, at 5:14 PM, Klaus Hochlehnert wrote:
>
>> The conf and the scripts dir were untouched after that (which I  
>> wanted because I didn?t want to reconfigure everything).
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From larclap at yahoo.com  Thu Oct 15 00:00:45 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Thu, 15 Oct 2009 00:00:45 -0700
Subject: [Freeswitch-users] Outbound sip address call fails
In-Reply-To: <8A5CCD06-A37C-423B-807B-01F24D537E49@freeswitch.org>
References: <00d701ca4d25$b7d615f0$278241d0$@com>
	<8A5CCD06-A37C-423B-807B-01F24D537E49@freeswitch.org>
Message-ID: <001e01ca4d65$3814bfc0$a83e3f40$@com>

Brian,

 

Does this mean that Bria should be sending the invite directly to
sip:888 at conference.freeswitch.org and the From address should be the address
of the Bria itself rather than FS?

 

Lars

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Wednesday, October 14, 2009 8:55 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Outbound sip address call fails

 

Sounds like bria is sending the invite via the proxy which is your local FS
box and NOT direct.

 

/b

 

On Oct 14, 2009, at 4:26 PM, Lars Zeb wrote:





I am trying to make a test call to the FreeSwitch conference address,
sip:888 at conference.freeswitch.org on a Bria softphone. However, it fails
with a 404 Not found message.

 

I have struggled with this for a while. Is there something special I must do
on the Bria to make this happen correctly?

 

Thanks, Lars

 

http://pastebin.freeswitch.org/10715

 

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From odermann at googlemail.com  Thu Oct 15 00:47:28 2009
From: odermann at googlemail.com (Dennis)
Date: Thu, 15 Oct 2009 09:47:28 +0200
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
	<5e414ed0910130826j11d45a66q86d785f900e4196e@mail.gmail.com>
	<65d96fc80910131012v1f72ac49o6bfe9f2267e541c5@mail.gmail.com>
	<191c3a030910131101p418e0961t373db68285248ff6@mail.gmail.com>
	<538F4CF43B294D0EACABA281E1913B4C@chaos>
	<65d96fc80910131224m73f35b87g355be818ad3c0caf@mail.gmail.com>
	<48B637E0506F4A809E4C0CAF39BD08BB@chaos>
	<65d96fc80910132348t202905fbub57cc4c814eb4e21@mail.gmail.com>
	<5e414ed0910140731w1c7ebedr150e69cda8073155@mail.gmail.com>
	<191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com>
Message-ID: <5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com>

> once you have 123456 won't you still be unsure if he will type the next 1 or
> not and be forced to refuse it and wait anyway?

basically you are right. BUT, we know, that a basic phone number has 6
digits - so, we do not have to check anything before. as soon as we
have 6 digits, we look in our database for this number. a flag in the
database tells us, if this number is allowed to use ddi (further
digits) or not. if the number is not allowed, we can answer the call
directly without any further waiting.
other numbers are maked to have 2 ddi's - here we know, that we have
to wait for 2 more digits, before we answer.
if a number can have 6 digits or 8 digits, you are right, we have to
wait anyway.

but without having overlap, there is ALWAYS a waiting time.

we would like to make the answer-times as short as possible. if no
waiting is needed, we do not want to wait.

we would like to tell fs to send a 484 response incomplete to the
cirpack of our carrier if we need/want more digits.
perhaps this works with fs, but not with "socket outbound", which we are using?

we know, that this is more work for our servers, but they are powerful
enough and have enough resources, that they should be able to handle
that.
of course we do not want to become to experimental, because fs works
so extremly smooth and reliable, that it is nearly unbelievable.



From nbelov at abisoft.spb.ru  Thu Oct 15 01:16:21 2009
From: nbelov at abisoft.spb.ru (Nikita Belov)
Date: Thu, 15 Oct 2009 12:16:21 +0400
Subject: [Freeswitch-users] conference call
In-Reply-To: <ebbc2d550910140838v3125579ev9fa16e3e83d25770@mail.gmail.com>
References: <064A5201C10D4D20AC76EB93CA5EA4B5@abisoft.biz>
	<ebbc2d550910140838v3125579ev9fa16e3e83d25770@mail.gmail.com>
Message-ID: <BE4EDDA84CB64AA98F18D412B51912FC@abisoft.biz>

Thanks for answer. 

 

As I've understood eavesdrop allows C to hear A and B, but A and B will not
hear C. But in my case B SHOULD hear C. In my case A - client, B - agent
worked with client and C - coach/supervisor for B. Can I make such
conference in FS using eavesdrop or other API command?

 

___

 

Thanks, Nikita

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Joao
Mesquita
Sent: Wednesday, October 14, 2009 7:38 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] conference call

 

Look at eavesdrop on the wiki.

JM

2009/10/14 Nikita Belov <nbelov at abisoft.spb.ru>

HI all,

 

I want to configure FS to make special conference call between three users
(A, B, C).  In this conference C will hear A and B, but A will hear only B.
Can I make it using FS API commands? Does anybody know which approach is
better to use?

 

___

 

Thanks, Nikita


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

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From god.nirvana at gmail.com  Thu Oct 15 04:44:49 2009
From: god.nirvana at gmail.com (god.nirvana)
Date: Thu, 15 Oct 2009 19:44:49 +0800
Subject: [Freeswitch-users] conference digits and conference control
Message-ID: <200910151944464068246@gmail.com>

hi all
      how can i get the digits when users in the conference??
      and,in conference.conf.xml
      <control action="mute" digits="0"/> the "action" will set another value?e.g:transfer?
      thanks

2009-10-15 



god.nirvana 
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From rupa at rupa.com  Thu Oct 15 05:04:17 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 15 Oct 2009 07:04:17 -0500
Subject: [Freeswitch-users] conference call
In-Reply-To: <BE4EDDA84CB64AA98F18D412B51912FC@abisoft.biz>
References: <064A5201C10D4D20AC76EB93CA5EA4B5@abisoft.biz>
	<ebbc2d550910140838v3125579ev9fa16e3e83d25770@mail.gmail.com>
	<BE4EDDA84CB64AA98F18D412B51912FC@abisoft.biz>
Message-ID: <b39273530910150504l5c7411b9g39b9ed33c592e85f@mail.gmail.com>

The wiki has a pretty good explanation of how eavesdrop works.
Enabling a talk path to A or B or both A and B requires dtmf.

So, if C hits the 1 button on the phone they can talk to the UUID you
bound the eavesdrop to.   2 talks to the other party.  3 talks to both
parties.

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop

Adding more api calls to control eavesdrop via event socket should be
possible, though I haven't looked at the code to see what it would
take...

On Thu, Oct 15, 2009 at 3:16 AM, Nikita Belov <nbelov at abisoft.spb.ru> wrote:
> Thanks for answer.
>
>
>
> As I?ve understood eavesdrop allows C to hear A and B, but A and B will not
> hear C. But in my case B SHOULD hear C. In my case A ? client, B ? agent
> worked with client and C ? coach/supervisor for B. Can I make such
> conference in FS using eavesdrop or other API command?
>
>
>
> ___
>
>
>
> Thanks, Nikita
>
>
>
> ________________________________
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Joao
> Mesquita
> Sent: Wednesday, October 14, 2009 7:38 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] conference call
>
>
>
> Look at eavesdrop on the wiki.
>
> JM
>
> 2009/10/14 Nikita Belov <nbelov at abisoft.spb.ru>
>
> HI all,
>
>
>
> I want to configure FS to make special conference call between three users
> (A, B, C).? In this conference C will hear A and B, but A will hear only B.
> Can I make it using FS API commands? Does anybody know which approach is
> better to use?
>
>
>
> ___
>
>
>
> Thanks, Nikita
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa



From anthony.minessale at gmail.com  Thu Oct 15 06:53:37 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 15 Oct 2009 08:53:37 -0500
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
	<65d96fc80910131012v1f72ac49o6bfe9f2267e541c5@mail.gmail.com>
	<191c3a030910131101p418e0961t373db68285248ff6@mail.gmail.com>
	<538F4CF43B294D0EACABA281E1913B4C@chaos>
	<65d96fc80910131224m73f35b87g355be818ad3c0caf@mail.gmail.com>
	<48B637E0506F4A809E4C0CAF39BD08BB@chaos>
	<65d96fc80910132348t202905fbub57cc4c814eb4e21@mail.gmail.com>
	<5e414ed0910140731w1c7ebedr150e69cda8073155@mail.gmail.com>
	<191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com>
	<5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com>
Message-ID: <191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com>

right you can reply 484 in your dp at any time
<action application="respond" data="484 Address Incomplete"/>

then it should try again.

The bit i can't remember is if we committed a certain 1 line patch that
makes sofia parse the next invite to the same call properly, the patch was
to the sofia lib itself so test it and see.  I may need to dig up the answer
again from the sofia dev.



On Thu, Oct 15, 2009 at 2:47 AM, Dennis <odermann at googlemail.com> wrote:

> > once you have 123456 won't you still be unsure if he will type the next 1
> or
> > not and be forced to refuse it and wait anyway?
>
> basically you are right. BUT, we know, that a basic phone number has 6
> digits - so, we do not have to check anything before. as soon as we
> have 6 digits, we look in our database for this number. a flag in the
> database tells us, if this number is allowed to use ddi (further
> digits) or not. if the number is not allowed, we can answer the call
> directly without any further waiting.
> other numbers are maked to have 2 ddi's - here we know, that we have
> to wait for 2 more digits, before we answer.
> if a number can have 6 digits or 8 digits, you are right, we have to
> wait anyway.
>
> but without having overlap, there is ALWAYS a waiting time.
>
> we would like to make the answer-times as short as possible. if no
> waiting is needed, we do not want to wait.
>
> we would like to tell fs to send a 484 response incomplete to the
> cirpack of our carrier if we need/want more digits.
> perhaps this works with fs, but not with "socket outbound", which we are
> using?
>
> we know, that this is more work for our servers, but they are powerful
> enough and have enough resources, that they should be able to handle
> that.
> of course we do not want to become to experimental, because fs works
> so extremly smooth and reliable, that it is nearly unbelievable.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From brian at freeswitch.org  Thu Oct 15 06:54:58 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 15 Oct 2009 06:54:58 -0700
Subject: [Freeswitch-users] Outbound sip address call fails
In-Reply-To: <001e01ca4d65$3814bfc0$a83e3f40$@com>
References: <00d701ca4d25$b7d615f0$278241d0$@com>
	<8A5CCD06-A37C-423B-807B-01F24D537E49@freeswitch.org>
	<001e01ca4d65$3814bfc0$a83e3f40$@com>
Message-ID: <D9915F47-D22D-4D6F-8B27-F76356A7CBBB@freeswitch.org>

no clue it depends on how you have it configured.  If you have an  
outbound proxy set it'll send all calls to the proxy.  If not then  
most likely it will go direct to the destination.

/b

On Oct 15, 2009, at 12:00 AM, Lars Zeb wrote:

> Brian,
>
> Does this mean that Bria should be sending the invite directly to sip:888 at conference.freeswitch.org 
>  and the From address should be the address of the Bria itself  
> rather than FS?
>
> Lars
>

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From anthony.minessale at gmail.com  Thu Oct 15 06:57:48 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 15 Oct 2009 08:57:48 -0500
Subject: [Freeswitch-users] Media got stuck after attended transfer...
In-Reply-To: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330253@srv02.khdev.corp>
References: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330253@srv02.khdev.corp>
Message-ID: <191c3a030910150657r668eb5a3q24c641e312d2b113@mail.gmail.com>

This is a known limitation until we add actual t38 support to the project.


On Wed, Oct 14, 2009 at 6:56 PM, Klaus Hochlehnert <Mailings at kh-dev.de>wrote:

>  Hi,
>
>
>
> sometimes I have the problem that after doing an attended transfer the
> media got stuck in FS.
>
> Meaning the call goes through, but I don?t hear anything and the caller
> still hears music.
>
>
>
> Now I found out that setting the sofia parameter
>  media-option=resume-media-on-hold  helps here.
>
> But after setting this parameter I always get the error ?Codec PROXY
> PASS-THROUGH encoder error!? when using t38modem with proxy_media=true.
>
>
>
> So I stuck here. I can either use fax or do attended transfers.
>
>
>
> Does anyone have a solution for this?
>
>
>
> Thanks, Klaus
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From jbarou at sqli.com  Thu Oct 15 07:10:53 2009
From: jbarou at sqli.com (Jonathan Barou)
Date: Thu, 15 Oct 2009 16:10:53 +0200
Subject: [Freeswitch-users] Alias user mapping
In-Reply-To: <4AC27BC1.2050201@ParcelPool.com>
References: <8048ff7f0909290633t24f15224v40fcfe5aa0c8fdf0@mail.gmail.com>
	<4AC27BC1.2050201@ParcelPool.com>
Message-ID: <8048ff7f0910150710g810c7e2kbb0f8b0f0a28ef6b@mail.gmail.com>

Thanks, now it's working, but in my user directory I have something like
this :

  <user id="john" number-alias="1368">

and in my dialplan.xml I have to put the two entry for working :

<condition field="destination_number" expression="^(toto|john|8400|1368)$">

If I only put john, when I try to call the "1368" it's not working. Is it a
normal way ?

Thanks



2009/9/29 Peter Hinman <Peter.Hinman at parcelpool.com>

>  I've noticed that when using an alphanumeric ID, case sensitivity causes
> issues.  It doesn't seem to matter what case the userID is in the XML
> directory.  What does seem to matter is how the user registered their
> phone.  If the user registered in all upper case (JOHN_SMITH), then dialing
> John_Smith will not work.
>
> My work around for this is to tell all users to register their phones in
> lower case, and then my dial plan uses LUA to force all dialed extensions to
> lowercase.
>
> Peter
>
>
> Jonathan Barou wrote:
>
> Hi everybody,
>  I have a problem with "Alphanumeric to numeric user mapping"
>
>  I have done like it's written here :
>
>
>  Alphanumeric to numeric user mapping
>
> Say you want a user's id to be alphanumeric (like an email username), such
> as johnsmith at pbx.example.com. These users have alphanumeric usernames in
> their sip phone config, but you want to map them from their sip username to
> a numeric extension, and vice versa.
>
> As of version 1.0.4, Freeswitch makes this trivial to accomplish. A user's
> ID can be any alphanumeric string, and this can be simply tied to an
> extension number using the 'number-alias' property. This property creates an
> aliased directory entry that points to the alphanumeric user entry.
>
> *NOTE: When using this attribute, you must be careful not to create a
> directory collision by having another user whose ID is the same as another
> user's alias*
>
> Here is an example from the user directory:
>
>   <user id="johnsmith" number-alias="1001">
>     <!-- Insert the usual user configuration variables and params here,
>          including user password, voicemail password, caller ID info, etc -->
>   </user>
>
>
> So when a user dials extension number 1001, your dialplan can use the
> 'user_data' function to look up the ID attribute associated with that number
> alias. In the default dialplan, the 'Local Extension' section can be made to
> work with a small change to the 'bridge' line:
>
>   <action application="bridge" data="user/${user_data(${dialed_extension}@${domain_name} attr id)}@${domain_name}"/>
>
>
> *NOTE: Using this user_data function in combination with mod_xml_curl will
> generate an additional request each time the user_data function is called.
> Note that it is already called once in the Local Extension section to
> determine the callgroup. Beware of performance implications of this with
> high-volume systems.*
>
>
>  But when I want to call my alias-number, FS says "No Route, Abording"
>
>  My version of FreeSWITCH is the 1.0.4pre9.
>
>  Do you have any ideas ? Did I forget something to do?
>
>  Thanks
>
>
>
> --
> Jonathan BAROU
> SQLI LYON - CRCI
> jbarou at sqli.com
> lyon.crci at sqli.com
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Jonathan BAROU
SQLI LYON - CRCI
jbarou at sqli.com
lyon.crci at sqli.com
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From odermann at googlemail.com  Thu Oct 15 07:18:56 2009
From: odermann at googlemail.com (Dennis)
Date: Thu, 15 Oct 2009 16:18:56 +0200
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
	<191c3a030910131101p418e0961t373db68285248ff6@mail.gmail.com>
	<538F4CF43B294D0EACABA281E1913B4C@chaos>
	<65d96fc80910131224m73f35b87g355be818ad3c0caf@mail.gmail.com>
	<48B637E0506F4A809E4C0CAF39BD08BB@chaos>
	<65d96fc80910132348t202905fbub57cc4c814eb4e21@mail.gmail.com>
	<5e414ed0910140731w1c7ebedr150e69cda8073155@mail.gmail.com>
	<191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com>
	<5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com>
	<191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com>
Message-ID: <5e414ed0910150718k41d747a6t6ca98daf1c2a01f4@mail.gmail.com>

ok, we will try this with the cirpack of our carrier. this will take
some days, till everything is set up.

after the tests i will come back to report.


2009/10/15 Anthony Minessale <anthony.minessale at gmail.com>:
> right you can reply 484 in your dp at any time
> <action application="respond" data="484 Address Incomplete"/>
>
> then it should try again.
>
> The bit i can't remember is if we committed a certain 1 line patch that
> makes sofia parse the next invite to the same call properly, the patch was
> to the sofia lib itself so test it and see.? I may need to dig up the answer
> again from the sofia dev.



From mgende at gendesign.com  Thu Oct 15 07:36:55 2009
From: mgende at gendesign.com (Michael Gende)
Date: Thu, 15 Oct 2009 09:36:55 -0500
Subject: [Freeswitch-users] FreeSWITCH-users Digest,
	Mike G. answer to 	Orien L
In-Reply-To: <a87d43fd0910142134v366551a1p52f74e436f7f5709@mail.gmail.com>
References: <mailman.1269.1255575509.2495.freeswitch-users@lists.freeswitch.org>
	<4AD6973B.70106@tx.rr.com>
	<a87d43fd0910142134v366551a1p52f74e436f7f5709@mail.gmail.com>
Message-ID: <a87d43fd0910150736k2690d130gcebf829f74d6950@mail.gmail.com>

Hey Orien,

OK, let's have a look here. How about going to the console of your FS and
posting the output of the following commands:

sofia status profile internal
sofia status profile external

We'll take it from there. Your problems are just like what I ran into
initially, so I'm hopeful we get round them.

Mike G.

On Wed, Oct 14, 2009 at 11:34 PM, Michael Gende <mgende at gendesign.com>wrote:

> I'll get to you in the morning (where do you live?) and we'll see what's
> what.
>
>
>
> On Wed, Oct 14, 2009 at 10:30 PM, Orien Love <orien at tx.rr.com> wrote:
>
>> Mike,
>>     Thank you for your quick response, this has been driving me nuts.
>>
>> in my third attempt I tried to follow the mulit home example. I did set
>> the phones to register to the local lan IP. actually had them
>> registering, but it did help it in any way still the same problem.
>> After my post I reinstalled the package and set the firewall to allow
>> phones on the wan to connect, as per
>> http://doc.pfsense.org/index.php/FreeSWITCH#Firewall
>> put my computer with the soft phone on the wan network (192,168.2.xxx)
>> had it registering and it had the same problem. so far all the setups I
>> have tried are not working, the phones register, but cannot
>> dial/connect  another  phone, also does not go to voice mail, maybe it
>> is in my dial plan. which is default.
>>
>> Shoot, I am not picky, I just want to get it to work. so any suggestions
>> on how to accomplish this are greatly appreciated.
>>
>> My phones are currently a couple of polycom 601's, and a Pangolin
>> Softphone.
>>
>> Thanks
>>   Orien L.
>>
>>
>
>
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From larclap at yahoo.com  Thu Oct 15 07:56:52 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Thu, 15 Oct 2009 07:56:52 -0700
Subject: [Freeswitch-users] lua script causing FreeSwitch to crash?
In-Reply-To: <87f2f3b90910131312k7dac8980h70ce8be9c07aa7b2@mail.gmail.com>
References: <005a01ca4c3c$c183ceb0$448b6c10$@com>
	<87f2f3b90910131312k7dac8980h70ce8be9c07aa7b2@mail.gmail.com>
Message-ID: <007901ca4da7$bafed320$30fc7960$@com>

Thanks, Michael, using "make current" resolved the issue. I will use only
that command in the future.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, October 13, 2009 1:12 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] lua script causing FreeSwitch to crash?

 

 

On Tue, Oct 13, 2009 at 12:38 PM, Lars Zeb <larclap at yahoo.com> wrote:

http://pastebin.freeswitch.org/10686

 

I am trying to use a lua script for inbound calls. The caller hears a busy
signal. After this call fails to go through, FreeSwitch is no longer in
memory - it does not appear in  'ps -ef' output.

 

I don't have a clue what I might be doing incorrectly in the script in order
to cause this to happen. There is no indication in the log that FreeSwitch
has terminated. But it is not resident in memory.

 

I am using v15134. I had been using this same script without incident as
late as v14929. Until I can resolve the issue, I have removed the call to
lua from the dialplan and done all I could in it instead.

We sent out a notice about a bug in event socket between revs 15119 and
15134. Update to latest trunk. (FYI, the fix was in r15135, so you just
barely missed it.)

After you update, let us know if the problem persists. 
-MC

 

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From sjmudd at pobox.com  Thu Oct 15 08:03:56 2009
From: sjmudd at pobox.com (Simon J Mudd)
Date: 15 Oct 2009 17:03:56 +0200
Subject: [Freeswitch-users] macros for performing repeated dialplan
	actions?
In-Reply-To: <87f2f3b90910141510n45401995y11e3e380653e9790@mail.gmail.com>
References: <20091014220135.GA12614@mad06.wl0.org>
	<87f2f3b90910141510n45401995y11e3e380653e9790@mail.gmail.com>
Message-ID: <m3eip41yxv.fsf@mad06.wl0.org>

Hi Michael,

msc at freeswitch.org (Michael Collins) writes:

> On Wed, Oct 14, 2009 at 3:01 PM, Simon J Mudd <sjmudd at pobox.com> wrote:
> 
> > I'm trying to build up a soho dialplan to replicate an existing Asterisk
> > setup I want to replace.

...

> > My question is how do I avoid repeatedly using these same 8-9 lines and
> > "merge" this into some macro which given the right parameters allows me
> > to define various extensions using different gateways but all configured
> > "the same way"?
> >
> > Thanks for any pointers to how this should be done.
> 
> Try using the execute_extension app. It's kinda like a gosub in BASIC: it'll
> go and execute an extension and then come back and continue executing. Put
> all those lines into a single extension and then call them from other
> extensions via execute_extension.

OK. That looks clear. However it doesn't look as if it's possible to
pass parameters this way which means that data can only be passed to
the "extension" via variables. That works but isn't so clean. I'll
check this out further shortly. I'm planning later to pass info like
"which gateway to talk to" and perhaps some sort of flag which
indicates "record: yes/no", and even include the bridge application
into this extension/macro.  That reduces the original dialplan pattern
matching to a much simpler and easier to configure setup for me.

excecute_extension does avoid a lot of redundancy, so thanks for the
pointer.  I'll get that working first.

Simon



From larclap at yahoo.com  Thu Oct 15 08:14:28 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Thu, 15 Oct 2009 08:14:28 -0700
Subject: [Freeswitch-users] Outbound sip address call fails
In-Reply-To: <D9915F47-D22D-4D6F-8B27-F76356A7CBBB@freeswitch.org>
References: <00d701ca4d25$b7d615f0$278241d0$@com>	<8A5CCD06-A37C-423B-807B-01F24D537E49@freeswitch.org>	<001e01ca4d65$3814bfc0$a83e3f40$@com>
	<D9915F47-D22D-4D6F-8B27-F76356A7CBBB@freeswitch.org>
Message-ID: <007e01ca4daa$30a74c90$91f5e5b0$@com>

I don't think I have a proxy. It's pretty standard installation.

 

                     Name          Type                               Data
State

============================================================================
=====================

                 internal       profile   sip:mod_sofia at 192.168.10.29:5060
RUNNING (0)

            internal-ipv6       profile           sip:mod_sofia@[::1]:5060
RUNNING (0)

                 external       profile   sip:mod_sofia at 192.168.10.29:5090
RUNNING (0)

                flowroute       gateway        sip:55555 at sip.flowroute.com
REGED

            192.168.10.29         alias                           internal
ALIASED

============================================================================
=====================

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, October 15, 2009 6:55 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Outbound sip address call fails

 

no clue it depends on how you have it configured.  If you have an outbound
proxy set it'll send all calls to the proxy.  If not then most likely it
will go direct to the destination.

 

/b

 

On Oct 15, 2009, at 12:00 AM, Lars Zeb wrote:





Brian,

 

Does this mean that Bria should be sending the invite directly to
sip:888 at conference.freeswitch.org and the From address should be the address
of the Bria itself rather than FS?

 

Lars

 

 

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From chad at apartmentlines.com  Thu Oct 15 09:33:55 2009
From: chad at apartmentlines.com (Chad Phillips -- Apartment Lines)
Date: Thu, 15 Oct 2009 12:33:55 -0400
Subject: [Freeswitch-users] macros for performing repeated dialplan
	actions?
In-Reply-To: <m3eip41yxv.fsf@mad06.wl0.org>
References: <20091014220135.GA12614@mad06.wl0.org>
	<87f2f3b90910141510n45401995y11e3e380653e9790@mail.gmail.com>
	<m3eip41yxv.fsf@mad06.wl0.org>
Message-ID: <1C80FD78-71B7-4D62-8C45-2F5BC8C1564C@apartmentlines.com>


On Oct 15, 2009, at 11:03 AM, Simon J Mudd wrote:

> OK. That looks clear. However it doesn't look as if it's possible to
> pass parameters this way which means that data can only be passed to
> the "extension" via variables. That works but isn't so clean. I'll
> check this out further shortly. I'm planning later to pass info like
> "which gateway to talk to" and perhaps some sort of flag which
> indicates "record: yes/no", and even include the bridge application
> into this extension/macro.  That reduces the original dialplan pattern
> matching to a much simpler and easier to configure setup for me.

here's a tip from a recovering asterisk user: don't feel like you have  
to do everything via the dialplan.  this was a constant mantra in  
asterisk, because calling AGI scripts was expensive.  this is not the  
case with FreeSWITCH.  the general rule of thumb is: as soon as it  
gets hard to do in the dialplan, farm it out to a script.  it's the  
way FreeSWITCH is designed, so don't fight it  ;)

that said, you still might be able to use execute_extension along with  
setting some channel variables before you call it.  those channel  
variables will be available in the executed extension, and can act  
pretty much like the arguments to a macro.



From Russell.Mosemann at cune.org  Thu Oct 15 09:48:10 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Thu, 15 Oct 2009 16:48:10 -0000
Subject: [Freeswitch-users] Outbound sip address call fails
In-Reply-To: <007e01ca4daa$30a74c90$91f5e5b0$@com>
Message-ID: <20091015164810.B6A253F3FA9@mail.cune.org>

Lars Zeb <larclap at yahoo.com> said:

> I don't think I have a proxy. It's pretty standard installation.

Is the phone itself configured to use a proxy?

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!




From Stefan.Weigel at allianz-warranty.com  Thu Oct 15 08:16:22 2009
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Thu, 15 Oct 2009 17:16:22 +0200
Subject: [Freeswitch-users] Changing display text on phones (sip_from_user)
Message-ID: <5003D7D3E06F514E8C682F18D223265C0465862EAA@AZWSMS03.azwarranty.int>

Hi list,

I'm playing around with this wonderful, mighty solution.
Setup so far:

I created a gateway pointing to my ITSP (sip.qsc.de). Outbound calls getting in an depending on the channel variable

sip_to_user I want to rewrite the sip_from_user (=text which get's displayed in the phones):

      <condition field="${sip_to_user}" expression="^0.*974$" break="never">
        <action application="set" data="sip_from_user='IT_Hotline'"/>
       ...

 My telephone rings but it's always showing the original dialer number. I also tried the variables

username
caller_id_name
sip_contact_user

in the application info log it's showing me:

Caller-Username: [01732898376]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [01732898376]
Caller-Caller-ID-Number: [01732898376]

[..]

variable_sip_from_user: ['IT_Hotline']
variable_sip_from_user_stripped: ['IT_Hotline']
variable_sip_contact_user: ['IT_Hotline']


Is there any chance to rewrite only the part that's getting displayed on the phone  (e.g. From: IT_Hotline sip:123456 at sip.qsc.de) ?


Thanks and best regards

Stefan

Stefan Weigel
Advanced IT-Professional
Tel.: +49 89 2000 48 975
Fax: +49 89 2000 48 566
eMail: Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com>
Allianz Automotive Services GmbH
Einsteinring 28
85609 Aschheim
Germany

http://www.allianz-warranty.com
Gesch?ftsf?hrung: Andreas R?sing, Horst Ziegler
Amtsgericht M?nchen, HRB 175682
F?r Umsatzsteuerzwecke: Ust-ID-Nr.: DE 262 617 720

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From Stefan.Weigel at allianz-warranty.com  Thu Oct 15 08:23:19 2009
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Thu, 15 Oct 2009 17:23:19 +0200
Subject: [Freeswitch-users] WG: Changing display text on phones
	(sip_from_user)
Message-ID: <5003D7D3E06F514E8C682F18D223265C0465862EAC@AZWSMS03.azwarranty.int>

Hi,

just found the answer:

just use

"effective_caller_id_name"

and  it work's :)



Cheers

Stefan

Stefan Weigel
Advanced IT-Professional
Tel.: +49 89 2000 48 975
Fax: +49 89 2000 48 566
eMail: Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com>
Allianz Automotive Services GmbH
Einsteinring 28
85609 Aschheim
Germany

http://www.allianz-warranty.com
Gesch?ftsf?hrung: Andreas R?sing, Horst Ziegler
Amtsgericht M?nchen, HRB 175682
F?r Umsatzsteuerzwecke: Ust-ID-Nr.: DE 262 617 720

Von: Weigel, Stefan
Gesendet: Donnerstag, 15. Oktober 2009 17:16
An: 'freeswitch-users at lists.freeswitch.org'
Betreff: Changing display text on phones (sip_from_user)

Hi list,

I'm playing around with this wonderful, mighty solution.
Setup so far:

I created a gateway pointing to my ITSP (sip.qsc.de). Outbound calls getting in an depending on the channel variable

sip_to_user I want to rewrite the sip_from_user (=text which get's displayed in the phones):

      <condition field="${sip_to_user}" expression="^0.*974$" break="never">
        <action application="set" data="sip_from_user='IT_Hotline'"/>
       ...

 My telephone rings but it's always showing the original dialer number. I also tried the variables

username
caller_id_name
sip_contact_user

in the application info log it's showing me:

Caller-Username: [01732898376]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [01732898376]
Caller-Caller-ID-Number: [01732898376]

[..]

variable_sip_from_user: ['IT_Hotline']
variable_sip_from_user_stripped: ['IT_Hotline']
variable_sip_contact_user: ['IT_Hotline']


Is there any chance to rewrite only the part that's getting displayed on the phone  (e.g. From: IT_Hotline sip:123456 at sip.qsc.de) ?


Thanks and best regards

Stefan

Stefan Weigel
Advanced IT-Professional
Tel.: +49 89 2000 48 975
Fax: +49 89 2000 48 566
eMail: Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com>
Allianz Automotive Services GmbH
Einsteinring 28
85609 Aschheim
Germany

http://www.allianz-warranty.com
Gesch?ftsf?hrung: Andreas R?sing, Horst Ziegler
Amtsgericht M?nchen, HRB 175682
F?r Umsatzsteuerzwecke: Ust-ID-Nr.: DE 262 617 720

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From larclap at yahoo.com  Thu Oct 15 10:17:25 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Thu, 15 Oct 2009 10:17:25 -0700
Subject: [Freeswitch-users] Outbound sip address call fails
In-Reply-To: <20091015164810.B6A253F3FA9@mail.cune.org>
References: <007e01ca4daa$30a74c90$91f5e5b0$@com>
	<20091015164810.B6A253F3FA9@mail.cune.org>
Message-ID: <00be01ca4dbb$5d913160$18b39420$@com>

I don't think Bria is set to use a proxy.

In the SIP Account/Account tab, it is set to "Register with domain and
receive calls" and "Send outbound via" is set to use Domain rather than
"Proxy Address".

The Account ID is set to 1009 at 192.168.10.29, where the IP is the address of
FreeSwitch.

> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-
> bounces at lists.freeswitch.org] On Behalf Of Russell.Mosemann at cune.org
> Sent: Thursday, October 15, 2009 9:48 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Outbound sip address call fails
> 
> Lars Zeb <larclap at yahoo.com> said:
> 
> > I don't think I have a proxy. It's pretty standard installation.
> 
> Is the phone itself configured to use a proxy?
> 
> --
> Russell Mosemann
> 
> 
> 
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From freeswitch at cartissolutions.com  Thu Oct 15 10:30:35 2009
From: freeswitch at cartissolutions.com (Yossi Neiman)
Date: Thu, 15 Oct 2009 12:30:35 -0500
Subject: [Freeswitch-users] apr_queue
In-Reply-To: <f8af5740910090458w6540db9fg972ad1ba8a691c30@mail.gmail.com>
References: <f8af5740910090458w6540db9fg972ad1ba8a691c30@mail.gmail.com>
Message-ID: <4AD75C3B.3030007@cartissolutions.com>

Not sure at all what you are asking.  apr_queue is simply a data 
management container that is threadsafe.

-- 
Yossi Neiman
Cartis Solutions, Inc. - http://www.cartissolutions.com



srinivasula reddy wrote:
> Hi all,
>
> does any know about How apr_queue is maintaing and retriving all 
> registered and all stuff....
>
>
> -- 
> Srinivasula Reddy K
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From Russell.Mosemann at cune.org  Thu Oct 15 10:33:13 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Thu, 15 Oct 2009 17:33:13 -0000
Subject: [Freeswitch-users] Outbound sip address call fails
In-Reply-To: <00be01ca4dbb$5d913160$18b39420$@com>
Message-ID: <20091015173313.3834B199FE2@mail.cune.org>

Lars Zeb <larclap at yahoo.com> said:

> I don't think Bria is set to use a proxy.
> 
> In the SIP Account/Account tab, it is set to "Register with domain and
> receive calls" and "Send outbound via" is set to use Domain rather than
> "Proxy Address".
> 
> The Account ID is set to 1009 at 192.168.10.29, where the IP is the
> address of FreeSwitch.

That's exactly how I have X-Lite configured, and I am able to call the
conference directly by entering the conference URL. I'm sitting behind
NAT, too. Under the Topology tab, it is set to "Discover global address"
and "Discover server" (for STUN). "Enable ICE" is checked. The next thing
to check might be if the phone traffic is going directly to the Internet
and getting past your firewall.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!




From orien at tx.rr.com  Thu Oct 15 10:43:21 2009
From: orien at tx.rr.com (Orien Love)
Date: Thu, 15 Oct 2009 12:43:21 -0500
Subject: [Freeswitch-users] Subject:, Re:  FreeSWITCH-users Digest,
 Orien L. answer to Mike G.
In-Reply-To: <mailman.1290.1255619700.2495.freeswitch-users@lists.freeswitch.org>
References: <mailman.1290.1255619700.2495.freeswitch-users@lists.freeswitch.org>
Message-ID: <4AD75F39.6010209@tx.rr.com>

Mike,
    I copied the status screen from the freeswitch package in the web 
setup. I you need a console view, I will need instructions on how to 
console into pfsense, Since I have never done that.

Orien L.

                     Name       Type                                  
Data    State
=================================================================================================
                 internal    profile       
sip:mod_sofia at 192.168.10.1:5060    RUNNING (0)
                 external    profile      
sip:mod_sofia at 192.168.2.210:5080    RUNNING (0)
             192.168.10.1      alias                              
internal    ALIASED
            internal-ipv6    profile              
sip:mod_sofia@[::1]:5060    RUNNING (0)
=================================================================================================
3 profiles 1 alias



sofia status profile external    

=================================================================================================
Name                 external
Domain Name          N/A
DBName               sofia_reg_external
Pres Hosts          
Dialplan             XML
Context              public
Challenge Realm      auto_to
RTP-IP               192.168.2.210
Ext-RTP-IP           192.168.2.210
SIP-IP               192.168.2.210
Ext-SIP-IP           192.168.2.210
URL                  sip:mod_sofia at 192.168.2.210:5080
BIND-URL             sip:mod_sofia at 192.168.2.210:5080
HOLD-MUSIC           local_stream://moh
OUTBOUND-PROXY       N/A
CODECS               PCMU,PCMA,GSM
TEL-EVENT            101
DTMF-MODE            rfc2833
CNG                  13
SESSION-TO           0
MAX-DIALOG           0
NOMEDIA              false
LATE-NEG             false
PROXY-MEDIA          false
AGGRESSIVENAT        false
STUN-ENABLED         true
STUN-AUTO-DISABLE    false
CALLS-IN             0
FAILED-CALLS-IN      0
CALLS-OUT            0
FAILED-CALLS-OUT     0

Registrations:
=================================================================================================
=================================================================================================



sofia status profile internal    

=================================================================================================
Name                 internal
Domain Name          N/A
DBName               sofia_reg_internal
Pres Hosts          
Dialplan             XML
Context              public
Challenge Realm      auto_from
RTP-IP               192.168.10.1
Ext-RTP-IP           192.168.10.1
SIP-IP               192.168.10.1
Ext-SIP-IP           192.168.10.1
URL                  sip:mod_sofia at 192.168.10.1:5060
BIND-URL             sip:mod_sofia at 192.168.10.1:5060
HOLD-MUSIC           local_stream://moh
OUTBOUND-PROXY       N/A
CODECS               G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
TEL-EVENT            101
DTMF-MODE            rfc2833
CNG                  13
SESSION-TO           0
MAX-DIALOG           0
NOMEDIA              false
LATE-NEG             false
PROXY-MEDIA          false
AGGRESSIVENAT        false
STUN-ENABLED         true
STUN-AUTO-DISABLE    false
CALLS-IN             6
FAILED-CALLS-IN      3
CALLS-OUT            0
FAILED-CALLS-OUT     0

Registrations:
=================================================================================================
Call-ID:        NjczMzQzNTVlYTJhYjIzYmFiZmFjZGYzNzNlMTVkMTc.
User:           1001 at 192.168.10.1
Contact:        "user"
Agent:          Pangolin v5.2, Build 09182009
Status:         Registered(UDP)(unknown) EXP(2009-10-15 05:03:17)
Host:           orienhome2.local
IP:             192.168.10.243
Port:           3789
Auth-User:      1001
Auth-Realm:     192.168.10.1

=================================================================================================



sofia status profile internal-ipv6    

=================================================================================================
Name                 internal-ipv6
Domain Name          N/A
DBName               sofia_reg_internal-ipv6
Pres Hosts          
Dialplan             XML
Context              public
Challenge Realm      auto_to
RTP-IP               ::1
SIP-IP               ::1
URL                  sip:mod_sofia@[::1]:5060
BIND-URL             sip:mod_sofia@[::1]:5060
HOLD-MUSIC           local_stream://moh
OUTBOUND-PROXY       N/A
CODECS               G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
TEL-EVENT            101
DTMF-MODE            rfc2833
CNG                  13
SESSION-TO           0
MAX-DIALOG           0
NOMEDIA              false
LATE-NEG             false
PROXY-MEDIA          false
AGGRESSIVENAT        false
STUN-ENABLED         true
STUN-AUTO-DISABLE    false
CALLS-IN             0
FAILED-CALLS-IN      0
CALLS-OUT            0
FAILED-CALLS-OUT     0

Registrations:
=================================================================================================
=================================================================================================



status

UP 0 years, 0 days, 6 hours, 55 minutes, 32 seconds, 640 milliseconds, 
638 microseconds
43 session(s) since startup
0 session(s) 0/30
1000 session(s) max


> Subject:
> Re: [Freeswitch-users] FreeSWITCH-users Digest, Mike G. answer to Orien L
> From:
> Michael Gende <mgende at gendesign.com>
> Date:
> Thu, 15 Oct 2009 09:36:55 -0500
> To:
> freeswitch-users at lists.freeswitch.org
>
> To:
> freeswitch-users at lists.freeswitch.org
>
>
> Hey Orien,
>
> OK, let's have a look here. How about going to the console of your FS 
> and posting the output of the following commands:
>
> sofia status profile internal
> sofia status profile external
>
> We'll take it from there. Your problems are just like what I ran into 
> initially, so I'm hopeful we get round them.
>
> Mike G.
>
> On Wed, Oct 14, 2009 at 11:34 PM, Michael Gende <mgende at gendesign.com 
> <mailto:mgende at gendesign.com>> wrote:
>
>     I'll get to you in the morning (where do you live?) and we'll see
>     what's what.
>
>
>
>     On Wed, Oct 14, 2009 at 10:30 PM, Orien Love <orien at tx.rr.com
>     <mailto:orien at tx.rr.com>> wrote:
>
>         Mike,
>             Thank you for your quick response, this has been driving
>         me nuts.
>
>         in my third attempt I tried to follow the mulit home example.
>         I did set
>         the phones to register to the local lan IP. actually had them
>         registering, but it did help it in any way still the same problem.
>         After my post I reinstalled the package and set the firewall
>         to allow
>         phones on the wan to connect, as per
>         http://doc.pfsense.org/index.php/FreeSWITCH#Firewall
>         put my computer with the soft phone on the wan network
>         (192,168.2.xxx)
>         had it registering and it had the same problem. so far all the
>         setups I
>         have tried are not working, the phones register, but cannot
>         dial/connect  another  phone, also does not go to voice mail,
>         maybe it
>         is in my dial plan. which is default.
>
>         Shoot, I am not picky, I just want to get it to work. so any
>         suggestions
>         on how to accomplish this are greatly appreciated.
>
>         My phones are currently a couple of polycom 601's, and a
>         Pangolin Softphone.
>
>         Thanks
>           Orien L.
>          
>
>
>




From msc at freeswitch.org  Thu Oct 15 11:09:22 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 15 Oct 2009 11:09:22 -0700
Subject: [Freeswitch-users] lua script causing FreeSwitch to crash?
In-Reply-To: <007901ca4da7$bafed320$30fc7960$@com>
References: <005a01ca4c3c$c183ceb0$448b6c10$@com>
	<87f2f3b90910131312k7dac8980h70ce8be9c07aa7b2@mail.gmail.com>
	<007901ca4da7$bafed320$30fc7960$@com>
Message-ID: <87f2f3b90910151109g1c317c97n43bed3d90b408548@mail.gmail.com>

On Thu, Oct 15, 2009 at 7:56 AM, Lars Zeb <larclap at yahoo.com> wrote:

>  Thanks, Michael, using ?make current? resolved the issue. I will use only
> that command in the future.
>
Excellent! Thanks for the feedback.
-MC
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From msc at freeswitch.org  Thu Oct 15 11:13:33 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 15 Oct 2009 11:13:33 -0700
Subject: [Freeswitch-users] Alias user mapping
In-Reply-To: <8048ff7f0910150710g810c7e2kbb0f8b0f0a28ef6b@mail.gmail.com>
References: <8048ff7f0909290633t24f15224v40fcfe5aa0c8fdf0@mail.gmail.com>
	<4AC27BC1.2050201@ParcelPool.com>
	<8048ff7f0910150710g810c7e2kbb0f8b0f0a28ef6b@mail.gmail.com>
Message-ID: <87f2f3b90910151113x43871d2fw41e99d638787563a@mail.gmail.com>

On Thu, Oct 15, 2009 at 7:10 AM, Jonathan Barou <jbarou at sqli.com> wrote:

> Thanks, now it's working, but in my user directory I have something like
> this :
>
>   <user id="john" number-alias="1368">
>
> and in my dialplan.xml I have to put the two entry for working :
>
> <condition field="destination_number" expression="^(toto|john|8400|1368)$">
>
> If I only put john, when I try to call the "1368" it's not working. Is it a
> normal way ?
>

Correct. The number-alias in the directory has no relation to the
destination_number matching in the dialplan. The destination_number match
will match whatever the person physically dialed on the telephone, so leave
the 1368 in there.

-MC
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From davis.erwin at gmail.com  Thu Oct 15 11:30:19 2009
From: davis.erwin at gmail.com (Erwin Davis)
Date: Thu, 15 Oct 2009 14:30:19 -0400
Subject: [Freeswitch-users] failed in bridging with SIPp
Message-ID: <a38a91d50910151130s1413cbc8yc39c58f8ca6473c7@mail.gmail.com>

Hi,

I tried to originate a call from x-lite and hope to terminate to SIPp
running in another machine (192.168.1.36:5069)
through FS. My x-lite received service no available and the FS report
NORMAL_TEMPOARY_FAILURE
(http://pastebin.freeswitch.org/10720). My purpose for FS is to simply
forward the call to the destination IP addr,
which is 192.168.1.36. Did I miss anything in the configuration? Thanks,

Below is my configuration,

in dialplan/default.xml,
    <extension name="to-sipp">
      <condition field="destination_number" expression="^6666$">
    <action application="bridge" data="sofia/internal/profile1/$
0 at 192.168.1.36"/>
      </condition>
    </extension>

in sip-profile/profile1.xml

<profile name="profile1">
  <param name="debug" value="1"/>
  <param name="rfc2833-pt" value="101"/>
  <param name="sip-ip" value="192.168.1.36"/>
  <param name="sip-port" value="5069"/>
  <param name="dialplan" value="XML"/>
  <param name="dtmf-duration" value="100"/>
  <param name="codec-ms" value="20"/>
  <param name="use-rtp-timer" value="true"/>
</profile>
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From tculjaga at gmail.com  Thu Oct 15 11:54:34 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 15 Oct 2009 20:54:34 +0200
Subject: [Freeswitch-users] Media got stuck after attended transfer...
In-Reply-To: <191c3a030910150657r668eb5a3q24c641e312d2b113@mail.gmail.com>
References: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330253@srv02.khdev.corp>
	<191c3a030910150657r668eb5a3q24c641e312d2b113@mail.gmail.com>
Message-ID: <65d96fc80910151154w2468ebeie06211d0966b4548@mail.gmail.com>

hi, any clue when can t38 be added?

T.

On Thu, Oct 15, 2009 at 3:57 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> This is a known limitation until we add actual t38 support to the project.
>
>
> On Wed, Oct 14, 2009 at 6:56 PM, Klaus Hochlehnert <Mailings at kh-dev.de>wrote:
>
>>  Hi,
>>
>>
>>
>> sometimes I have the problem that after doing an attended transfer the
>> media got stuck in FS.
>>
>> Meaning the call goes through, but I don?t hear anything and the caller
>> still hears music.
>>
>>
>>
>> Now I found out that setting the sofia parameter
>>  media-option=resume-media-on-hold  helps here.
>>
>> But after setting this parameter I always get the error ?Codec PROXY
>> PASS-THROUGH encoder error!? when using t38modem with proxy_media=true.
>>
>>
>>
>> So I stuck here. I can either use fax or do attended transfers.
>>
>>
>>
>> Does anyone have a solution for this?
>>
>>
>>
>> Thanks, Klaus
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From larclap at yahoo.com  Thu Oct 15 13:09:30 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Thu, 15 Oct 2009 13:09:30 -0700
Subject: [Freeswitch-users] Outbound sip address call fails
In-Reply-To: <20091015173313.3834B199FE2@mail.cune.org>
References: <00be01ca4dbb$5d913160$18b39420$@com>
	<20091015173313.3834B199FE2@mail.cune.org>
Message-ID: <011301ca4dd3$67d486a0$377d93e0$@com>

Russell,

My settings in Bria are just like yours.

I captured the SIP trace from the Bria and it says it is using the
FreeSwitch as the Outbound Proxy. I am very confused.

I tried dial sip:echo at iptel.org. In the pastebin you can see that this
address came over to the cli as "echo". Is it Bria which is messing up, or
my FS configuration?

Where did Bria get the Contact info at lines 79 & 82? 192.168.10.11 is the
address of the softphone, so I don't think sip:1009 at 192.168.10.11 makes
sense.

http://pastebin.freeswitch.org/10723

Lars


> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-
> bounces at lists.freeswitch.org] On Behalf Of Russell.Mosemann at cune.org
> Sent: Thursday, October 15, 2009 10:33 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Outbound sip address call fails
> 
> Lars Zeb <larclap at yahoo.com> said:
> 
> > I don't think Bria is set to use a proxy.
> >
> > In the SIP Account/Account tab, it is set to "Register with domain and
> > receive calls" and "Send outbound via" is set to use Domain rather than
> > "Proxy Address".
> >
> > The Account ID is set to 1009 at 192.168.10.29, where the IP is the
> > address of FreeSwitch.
> 
> That's exactly how I have X-Lite configured, and I am able to call the
> conference directly by entering the conference URL. I'm sitting behind
> NAT, too. Under the Topology tab, it is set to "Discover global address"
> and "Discover server" (for STUN). "Enable ICE" is checked. The next thing
> to check might be if the phone traffic is going directly to the Internet
> and getting past your firewall.
> 
> --
> Russell Mosemann
> 
> 
> 
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From alexagone at gmail.com  Thu Oct 15 12:20:30 2009
From: alexagone at gmail.com (Alexandre Savard)
Date: Thu, 15 Oct 2009 15:20:30 -0400
Subject: [Freeswitch-users] TLS client authentification
Message-ID: <87747c930910151220p474e079bt609019e562698dfb@mail.gmail.com>

Hi,

Does Freeswitch support TLS Client-Authenticated handshake. Openssl does,
but it has to be enabled in order to send the certificate request to the
client.

I tested simple TLS hanshaking and it wotks well.


Thanks.
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From mcampbellsmith at gmail.com  Thu Oct 15 15:19:21 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Fri, 16 Oct 2009 09:19:21 +1100
Subject: [Freeswitch-users] Dingaling / Jingle DTMF support?
Message-ID: <33c87fa30910151519x637618d3p109f1b8edbf3432e@mail.gmail.com>

Hi!

I was wondering if the Dingaling implementation in FS supports DTMF?
This is now supported in the jingle specs (
http://www.jabberforum.org/showthread.php?t=2709 ), even though Google
Talk client does not currently support DTMF.

If not, are there plans to implement this?

Thanks!



From jason at jasonjgw.net  Thu Oct 15 16:22:29 2009
From: jason at jasonjgw.net (Jason White)
Date: Fri, 16 Oct 2009 10:22:29 +1100
Subject: [Freeswitch-users] failed in bridging with SIPp
In-Reply-To: <a38a91d50910151130s1413cbc8yc39c58f8ca6473c7@mail.gmail.com>
References: <a38a91d50910151130s1413cbc8yc39c58f8ca6473c7@mail.gmail.com>
Message-ID: <20091015232229.GA10415@jdc.jasonjgw.net>

Erwin Davis <davis.erwin at gmail.com> wrote:
>     <action application="bridge" data="sofia/internal/profile1/$
> 0 at 192.168.1.36"/>

The above syntax won't work.

data="user/$1 at 192.168.1.36" (assuming that your FreeSWITCH domain is
192.168.1.36) would be better.

If the other end doesn't require authentication (i.e., it isn't registering
with FS) you can use sofia/internal/$1 at domain




From msc at freeswitch.org  Thu Oct 15 16:56:11 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 15 Oct 2009 16:56:11 -0700
Subject: [Freeswitch-users] failed in bridging with SIPp
In-Reply-To: <20091015232229.GA10415@jdc.jasonjgw.net>
References: <a38a91d50910151130s1413cbc8yc39c58f8ca6473c7@mail.gmail.com>
	<20091015232229.GA10415@jdc.jasonjgw.net>
Message-ID: <87f2f3b90910151656tc908855qf1011e84a8075a7b@mail.gmail.com>

On Thu, Oct 15, 2009 at 4:22 PM, Jason White <jason at jasonjgw.net> wrote:

> Erwin Davis <davis.erwin at gmail.com> wrote:
> >     <action application="bridge" data="sofia/internal/profile1/$
> > 0 at 192.168.1.36"/>
>
> The above syntax won't work.
>
> data="user/$1 at 192.168.1.36" (assuming that your FreeSWITCH domain is
> 192.168.1.36) would be better.
>
> If the other end doesn't require authentication (i.e., it isn't registering
> with FS) you can use sofia/internal/$1 at domain
>

Erwin,

Good catch on the sofia syntax. FYI, for those who don't already know, be
sure to check out this wiki page:

http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings

It discusses the various syntaxes for sofia dial strings. Hint: in the
#freeswitch IRC channel you can type "~whatis anatomy" and get this URL.
It's "anatomy" of a SIP dial string.
-MC
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From davis.erwin at gmail.com  Thu Oct 15 17:01:17 2009
From: davis.erwin at gmail.com (Erwin Davis)
Date: Thu, 15 Oct 2009 20:01:17 -0400
Subject: [Freeswitch-users] failed in bridging with SIPp
In-Reply-To: <20091015232229.GA10415@jdc.jasonjgw.net>
References: <a38a91d50910151130s1413cbc8yc39c58f8ca6473c7@mail.gmail.com>
	<20091015232229.GA10415@jdc.jasonjgw.net>
Message-ID: <a38a91d50910151701s4fc8dfd3r513ee41f39e99648@mail.gmail.com>

I prefer the second case. How to define the domain?  Thanks,

e

On Thu, Oct 15, 2009 at 7:22 PM, Jason White <jason at jasonjgw.net> wrote:

> Erwin Davis <davis.erwin at gmail.com> wrote:
> >     <action application="bridge" data="sofia/internal/profile1/$
> > 0 at 192.168.1.36"/>
>
> The above syntax won't work.
>
> data="user/$1 at 192.168.1.36" (assuming that your FreeSWITCH domain is
> 192.168.1.36) would be better.
>
> If the other end doesn't require authentication (i.e., it isn't registering
> with FS) you can use sofia/internal/$1 at domain
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Thu Oct 15 17:10:22 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 15 Oct 2009 17:10:22 -0700
Subject: [Freeswitch-users] Media got stuck after attended transfer...
In-Reply-To: <65d96fc80910151154w2468ebeie06211d0966b4548@mail.gmail.com>
References: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330253@srv02.khdev.corp>
	<191c3a030910150657r668eb5a3q24c641e312d2b113@mail.gmail.com>
	<65d96fc80910151154w2468ebeie06211d0966b4548@mail.gmail.com>
Message-ID: <87f2f3b90910151710k34e4092eg26108dd819d9c041@mail.gmail.com>

On Thu, Oct 15, 2009 at 11:54 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:

> hi, any clue when can t38 be added?
>
>
"Eventually." :)  Of course, if we could get more to add to the bounty it
might grease the wheels of innovation.

http://wiki.freeswitch.org/wiki/Bounty#spanDSP_.2B_t.38_.28origination.2C_termination.2C_.26_gateway.29_in_Freeswitch

Of course, I was listening to my A.M radio the other day and they said that
there was this new invention called the Internet that would let people send
documents to each other electronically. Maybe you should look into that.
Next thing you know they'll come up with telephones that people don't have
to plug into the wall and can take with them in the car. ;)

-MC
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From jason at jasonjgw.net  Thu Oct 15 17:31:51 2009
From: jason at jasonjgw.net (Jason White)
Date: Fri, 16 Oct 2009 11:31:51 +1100
Subject: [Freeswitch-users] failed in bridging with SIPp
In-Reply-To: <a38a91d50910151701s4fc8dfd3r513ee41f39e99648@mail.gmail.com>
References: <a38a91d50910151130s1413cbc8yc39c58f8ca6473c7@mail.gmail.com>
	<20091015232229.GA10415@jdc.jasonjgw.net>
	<a38a91d50910151701s4fc8dfd3r513ee41f39e99648@mail.gmail.com>
Message-ID: <20091016003151.GA26958@jdc.jasonjgw.net>

Erwin Davis <davis.erwin at gmail.com> wrote:
> I prefer the second case. How to define the domain?  Thanks,

It's the domain of FreeSWITCH itself. You could just specify $${domain} if you
aren't using multiple domains.
user/6666@$${domain}

By default, the domain is the IPv4 address of the machine running FreeSWITCH,
which you can change by setting it in vars.xml (assuming that you're using the
default configuration; it will be different if you totally rewrite the
configuration, of course.)




From mike at jerris.com  Thu Oct 15 18:50:40 2009
From: mike at jerris.com (Michael Jerris)
Date: Thu, 15 Oct 2009 21:50:40 -0400
Subject: [Freeswitch-users] Sending an Event to a Session for onInput
In-Reply-To: <4256bf830910112311t409c45a3q2f26cef3464e49e7@mail.gmail.com>
References: <4256bf830910091234t11f4f4d0w28283f341bd1fb06@mail.gmail.com>
	<ACA9A602-0925-4D6F-8B11-0B214A462500@jerris.com>
	<4256bf830910112311t409c45a3q2f26cef3464e49e7@mail.gmail.com>
Message-ID: <0BDCD23D-D551-4A2F-9CD7-CC5A2F1382CB@jerris.com>

It updates the display on a phone if the phone supports this.  This  
works on some sip phones right now including polycom and snom.

Mike

On Oct 12, 2009, at 2:11 AM, Matthew Fong wrote:

> Hi Mike,
>
> I'm just trying to send it an event with some custom event headers,  
> just so an external program can communicate with a session without  
> having to transfer the session to a different program.  I'm curious  
> what uuid_display does...the wiki only gives a brief description and  
> my Google'ing could not find any examples. Thanks for the help.
>
> --matt
> http://www.hellohunter.com
>
> On Mon, Oct 12, 2009 at 5:04 AM, Michael Jerris <mike at jerris.com>  
> wrote:
> We don't have session messages directly exposed, except for things
> like display, respond, and deflect.  What specifically are you trying
> to send ?
>
> Mike
>
> On Oct 9, 2009, at 3:34 PM, Matthew Fong wrote:
>
> > I'm used to using the onInput callbacks inside lua and javascript to
> > listen for dtmf and other events and perform a task accordingly. I'm
> > wondering if there is a way to send an event to a session or channel
> > that can be caught using the setInputCallback inside lua from
> > outside the session program. Maybe an API command that can generate
> > an event for a specific UUID. Does a mechanism exist to do this that
> > I'm over looking? Thanks.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From mcampbellsmith at gmail.com  Thu Oct 15 20:35:20 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Fri, 16 Oct 2009 14:35:20 +1100
Subject: [Freeswitch-users] Dingaling: using a hostname instead of stun
	for rtp
In-Reply-To: <C874E25D-6D73-49AA-8DD3-0A0022617FD2@freeswitch.org>
References: <33c87fa30910131502t2af24801i1c3319518b5db50b@mail.gmail.com>
	<C874E25D-6D73-49AA-8DD3-0A0022617FD2@freeswitch.org>
Message-ID: <33c87fa30910152035w43e260faoc031a7a4ac949d9e@mail.gmail.com>

Thanks Brian.  Is this something that is planned to be implemented?
The workaround is to set the stun server also in the dingaling
configuration, but as I said, for some reason the stun times for me
out occasionally with dingaling.

Thanks!

On Wed, Oct 14, 2009 at 11:33 AM, Brian West <brian at freeswitch.org> wrote:
> I don't think mod_dingaling will do a lookup for host: like sofia will
> as it doesn't have the code for that last I checked... I could be
> wrong but I don't recall it doing that.
>
> /b
>
> On Oct 13, 2009, at 3:02 PM, Mark Campbell-Smith wrote:
>
>> I have a hostname set in vars.conf.xml for the parameters
>> external_rtp_ip and the external_sip_ip instead of the usual stun. ?I
>> found that stun was timing out and was causing some problems. ?And as
>> I have a hostname, it makes sense to use that instead of relying on
>> stun.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From mgende at gendesign.com  Thu Oct 15 22:02:16 2009
From: mgende at gendesign.com (Michael Gende)
Date: Fri, 16 Oct 2009 00:02:16 -0500
Subject: [Freeswitch-users] Subject:, Re: FreeSWITCH-users Digest,
	Orien 	L. answer to Mike G.
In-Reply-To: <4AD75F39.6010209@tx.rr.com>
References: <mailman.1290.1255619700.2495.freeswitch-users@lists.freeswitch.org>
	<4AD75F39.6010209@tx.rr.com>
Message-ID: <a87d43fd0910152202k2c7b3300ya139a81385f60bf2@mail.gmail.com>

Hey Orien,

pfsense has a nice interface, but I only use it for adding the FS package,
maybe messing with LAN and WAN addresses. Using the "command line" helped me
to better understand what FS was doing and how. Pfsense just made it easier
to have an OS and really good firewall married to FS without me needing to
do a lot of work. In my pfsense installs, I didn't need to add any firewall
rules to allow SIP ports "right of way", the pfsense folks had taken care of
that detail.


One way to talk "direct" to FS on pfsense is to use the console of the
FreeBSD box you have pfsense installed on. Login as root. You'll get a
character screen with options 1-12 (I believe).

SSHing in gets you the same interface options.

Hit option 8 and you'll go to a shell. I'm assuming you can get around in
Linux or Unix (If not, tell me). "ps aux | grep free" and kill the PID for
FS (pfsense runs FS in the background) Then, invoke FS with
/usr/local/freeswitch/bin/freeswitch.

Now, you're talking direct to FS in the foreground, and it is talking to
you. After starting FS, hit "enter" if there's stuff on the screen - which
there will be - and you'll see the FS prompt. Try the sofia commands.


If you are still configured with:

WAN: 192.168.2.210
LAN: 192.168.3.1
Phones on network 192.168.3.0/24

...then make sure you follow the multi-home wiki instructions in section 4.
I'd include that here, but its better formatted there. I've updated it last
week so have another look if you've not seen it lately.

http://wiki.freeswitch.org/wiki/Multi_home_tutorial

Also, make sure you understand section 3, the FS COMMANDS FOR STARTERS and
WHERE STUFF IS FOR STARTERS. It'll help.

Following those instructions, unless there's some other issue, will allow
for phones on the LAN to talk to one another. Its a start.

If it does not, we'll take it from there.

Regards,

Mike G.






On Thu, Oct 15, 2009 at 12:43 PM, Orien Love <orien at tx.rr.com> wrote:

> Mike,
>    I copied the status screen from the freeswitch package in the web
> setup. I you need a console view, I will need instructions on how to
> console into pfsense, Since I have never done that.
>
> Orien L.
>
>                     Name       Type
> Data    State
>
> =================================================================================================
>                 internal    profile
> sip:mod_sofia at 192.168.10.1:5060    RUNNING (0)
>                 external    profile
> sip:mod_sofia at 192.168.2.210:5080    RUNNING (0)
>             192.168.10.1      alias
> internal    ALIASED
>            internal-ipv6    profile
> sip:mod_sofia@[::1]:5060    RUNNING (0)
>
> =================================================================================================
> 3 profiles 1 alias
>
>
>
> sofia status profile external
>
>
> =================================================================================================
> Name                 external
> Domain Name          N/A
> DBName               sofia_reg_external
> Pres Hosts
> Dialplan             XML
> Context              public
> Challenge Realm      auto_to
> RTP-IP               192.168.2.210
> Ext-RTP-IP           192.168.2.210
> SIP-IP               192.168.2.210
> Ext-SIP-IP           192.168.2.210
> URL                  sip:mod_sofia at 192.168.2.210:5080
> BIND-URL             sip:mod_sofia at 192.168.2.210:5080
> HOLD-MUSIC           local_stream://moh
> OUTBOUND-PROXY       N/A
> CODECS               PCMU,PCMA,GSM
> TEL-EVENT            101
> DTMF-MODE            rfc2833
> CNG                  13
> SESSION-TO           0
> MAX-DIALOG           0
> NOMEDIA              false
> LATE-NEG             false
> PROXY-MEDIA          false
> AGGRESSIVENAT        false
> STUN-ENABLED         true
> STUN-AUTO-DISABLE    false
> CALLS-IN             0
> FAILED-CALLS-IN      0
> CALLS-OUT            0
> FAILED-CALLS-OUT     0
>
> Registrations:
>
> =================================================================================================
>
> =================================================================================================
>
>
>
> sofia status profile internal
>
>
> =================================================================================================
> Name                 internal
> Domain Name          N/A
> DBName               sofia_reg_internal
> Pres Hosts
> Dialplan             XML
> Context              public
> Challenge Realm      auto_from
> RTP-IP               192.168.10.1
> Ext-RTP-IP           192.168.10.1
> SIP-IP               192.168.10.1
> Ext-SIP-IP           192.168.10.1
> URL                  sip:mod_sofia at 192.168.10.1:5060
> BIND-URL             sip:mod_sofia at 192.168.10.1:5060
> HOLD-MUSIC           local_stream://moh
> OUTBOUND-PROXY       N/A
> CODECS               G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
> TEL-EVENT            101
> DTMF-MODE            rfc2833
> CNG                  13
> SESSION-TO           0
> MAX-DIALOG           0
> NOMEDIA              false
> LATE-NEG             false
> PROXY-MEDIA          false
> AGGRESSIVENAT        false
> STUN-ENABLED         true
> STUN-AUTO-DISABLE    false
> CALLS-IN             6
> FAILED-CALLS-IN      3
> CALLS-OUT            0
> FAILED-CALLS-OUT     0
>
> Registrations:
>
> =================================================================================================
> Call-ID:        NjczMzQzNTVlYTJhYjIzYmFiZmFjZGYzNzNlMTVkMTc.
> User:           1001 at 192.168.10.1
> Contact:        "user"
> Agent:          Pangolin v5.2, Build 09182009
> Status:         Registered(UDP)(unknown) EXP(2009-10-15 05:03:17)
> Host:           orienhome2.local
> IP:             192.168.10.243
> Port:           3789
> Auth-User:      1001
> Auth-Realm:     192.168.10.1
>
>
> =================================================================================================
>
>
>
> sofia status profile internal-ipv6
>
>
> =================================================================================================
> Name                 internal-ipv6
> Domain Name          N/A
> DBName               sofia_reg_internal-ipv6
> Pres Hosts
> Dialplan             XML
> Context              public
> Challenge Realm      auto_to
> RTP-IP               ::1
> SIP-IP               ::1
> URL                  sip:mod_sofia@[::1]:5060
> BIND-URL             sip:mod_sofia@[::1]:5060
> HOLD-MUSIC           local_stream://moh
> OUTBOUND-PROXY       N/A
> CODECS               G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
> TEL-EVENT            101
> DTMF-MODE            rfc2833
> CNG                  13
> SESSION-TO           0
> MAX-DIALOG           0
> NOMEDIA              false
> LATE-NEG             false
> PROXY-MEDIA          false
> AGGRESSIVENAT        false
> STUN-ENABLED         true
> STUN-AUTO-DISABLE    false
> CALLS-IN             0
> FAILED-CALLS-IN      0
> CALLS-OUT            0
> FAILED-CALLS-OUT     0
>
> Registrations:
>
> =================================================================================================
>
> =================================================================================================
>
>
>
> status
>
> UP 0 years, 0 days, 6 hours, 55 minutes, 32 seconds, 640 milliseconds,
> 638 microseconds
> 43 session(s) since startup
> 0 session(s) 0/30
> 1000 session(s) max
>
>
> > Subject:
> > Re: [Freeswitch-users] FreeSWITCH-users Digest, Mike G. answer to Orien L
> > From:
> > Michael Gende <mgende at gendesign.com>
> > Date:
> > Thu, 15 Oct 2009 09:36:55 -0500
> > To:
> > freeswitch-users at lists.freeswitch.org
> >
> > To:
> > freeswitch-users at lists.freeswitch.org
> >
> >
> > Hey Orien,
> >
> > OK, let's have a look here. How about going to the console of your FS
> > and posting the output of the following commands:
> >
> > sofia status profile internal
> > sofia status profile external
> >
> > We'll take it from there. Your problems are just like what I ran into
> > initially, so I'm hopeful we get round them.
> >
> > Mike G.
> >
> > On Wed, Oct 14, 2009 at 11:34 PM, Michael Gende <mgende at gendesign.com
> > <mailto:mgende at gendesign.com>> wrote:
> >
> >     I'll get to you in the morning (where do you live?) and we'll see
> >     what's what.
> >
> >
> >
> >     On Wed, Oct 14, 2009 at 10:30 PM, Orien Love <orien at tx.rr.com
> >     <mailto:orien at tx.rr.com>> wrote:
> >
> >         Mike,
> >             Thank you for your quick response, this has been driving
> >         me nuts.
> >
> >         in my third attempt I tried to follow the mulit home example.
> >         I did set
> >         the phones to register to the local lan IP. actually had them
> >         registering, but it did help it in any way still the same
> problem.
> >         After my post I reinstalled the package and set the firewall
> >         to allow
> >         phones on the wan to connect, as per
> >         http://doc.pfsense.org/index.php/FreeSWITCH#Firewall
> >         put my computer with the soft phone on the wan network
> >         (192,168.2.xxx)
> >         had it registering and it had the same problem. so far all the
> >         setups I
> >         have tried are not working, the phones register, but cannot
> >         dial/connect  another  phone, also does not go to voice mail,
> >         maybe it
> >         is in my dial plan. which is default.
> >
> >         Shoot, I am not picky, I just want to get it to work. so any
> >         suggestions
> >         on how to accomplish this are greatly appreciated.
> >
> >         My phones are currently a couple of polycom 601's, and a
> >         Pangolin Softphone.
> >
> >         Thanks
> >           Orien L.
> >
> >
> >
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From rogelio.perez at gmail.com  Thu Oct 15 23:44:23 2009
From: rogelio.perez at gmail.com (Rogelio Perez)
Date: Fri, 16 Oct 2009 03:44:23 -0300
Subject: [Freeswitch-users] SIP CANCEL error
Message-ID: <E1FE097E-599C-479E-9D1C-2308F213109C@gmail.com>

Hi guys,

I'm having a signaling issue on FS when calling from an endpoint to a  
GW.
To reproduce the error I just call and hungup the call right away.
FS gets the CANCEL and replies back with 481 Call/Transaction Does Not  
Exist, so the call continues and eventually connects, even when the A  
side of the call is closed.
I have verified that the Call-ID of the CANCEL matches that from the  
INVITE, so I see no reasons for FS to reject it.

SIP+FS logs in pastebin 10738.

Any ideas?

Thanks,
Rogelio






From codecomplete at free.fr  Fri Oct 16 00:11:16 2009
From: codecomplete at free.fr (Fred-145)
Date: Fri, 16 Oct 2009 00:11:16 -0700 (PDT)
Subject: [Freeswitch-users]  PBX vs. SIP proxy?
Message-ID: <25920788.post@talk.nabble.com>


Hello

I was reading an article the other day about Freeswitch in which the author
used the terms "SIP proxy" and PBX. Is an SIP proxy just a way for SIP
clients to register and connect to each other, while the PBX part adds
services like voice mail, etc.?

Thank you.
-- 
View this message in context: http://www.nabble.com/PBX-vs.-SIP-proxy--tp25920788p25920788.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From Russell.Mosemann at cune.org  Fri Oct 16 00:47:31 2009
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Fri, 16 Oct 2009 02:47:31 -0500
Subject: [Freeswitch-users] Subject:, Re: FreeSWITCH-users Digest,
	Orien 	L. answer to Mike G.
In-Reply-To: <a87d43fd0910152202k2c7b3300ya139a81385f60bf2@mail.gmail.com>
References: <mailman.1290.1255619700.2495.freeswitch-users@lists.freeswitch.org><4AD75F39.6010209@tx.rr.com>
	<a87d43fd0910152202k2c7b3300ya139a81385f60bf2@mail.gmail.com>
Message-ID: <C8C9B2D0FA5E40C4AA2658B34D0092DD@cune.pri>

Michael Gende wrote:

> Hit option 8 and you'll go to a shell. I'm assuming you can get around in
> Linux or Unix (If not, tell me). "ps aux | grep free" and kill the PID for
> FS (pfsense runs FS in the background) Then, invoke FS with
> /usr/local/freeswitch/bin/freeswitch.

You've mentioned this twice, now, but a better way to do it is to _not_ stop FreeSwitch. Instead, run /usr/local/freeswitch/bin/fs_cli, which connects to FS, and then you can do everything that you can do directly in FS. When you are done, enter "\quit" or "\bye" or "\exit". See "\help" and "help" for more information.

--
Russell Mosemann




From Russell.Mosemann at cune.org  Fri Oct 16 00:54:18 2009
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Fri, 16 Oct 2009 02:54:18 -0500
Subject: [Freeswitch-users] Subject:, Re: FreeSWITCH-users Digest,
	Orien 	L. answer to Mike G.
In-Reply-To: <C8C9B2D0FA5E40C4AA2658B34D0092DD@cune.pri>
References: <mailman.1290.1255619700.2495.freeswitch-users@lists.freeswitch.org><4AD75F39.6010209@tx.rr.com><a87d43fd0910152202k2c7b3300ya139a81385f60bf2@mail.gmail.com>
	<C8C9B2D0FA5E40C4AA2658B34D0092DD@cune.pri>
Message-ID: <D328986AA10A4B0796265D08C355EFFC@cune.pri>

> When you are done, enter "\quit" or "\bye" or "\exit". See "\help" and
> "help" for more information.

Yup, I got the slashes wrong, and I was staring right at the commands when I wrote the message. I'll blame it on how late it is here. :-) It should be "/quit", "/bye", "/exit" and "/help".

--
Russell Mosemann




From jason at jasonjgw.net  Fri Oct 16 01:06:08 2009
From: jason at jasonjgw.net (Jason White)
Date: Fri, 16 Oct 2009 19:06:08 +1100
Subject: [Freeswitch-users] Subject:, Re: FreeSWITCH-users Digest,
 Orien 	L. answer to Mike G.
In-Reply-To: <D328986AA10A4B0796265D08C355EFFC@cune.pri>
References: <mailman.1290.1255619700.2495.freeswitch-users@lists.freeswitch.org>
	<4AD75F39.6010209@tx.rr.com>
	<a87d43fd0910152202k2c7b3300ya139a81385f60bf2@mail.gmail.com>
	<C8C9B2D0FA5E40C4AA2658B34D0092DD@cune.pri>
	<D328986AA10A4B0796265D08C355EFFC@cune.pri>
Message-ID: <20091016080608.GA21211@jdc.jasonjgw.net>

Russell Mosemann <Russell.Mosemann at cune.org> wrote:
> 
> Yup, I got the slashes wrong, and I was staring right at the commands when I
> wrote the message. I'll blame it on how late it is here. :-) It should be
> "/quit", "/bye", "/exit" and "/help".

ctrl-d is also recognized and it's easier to type.




From helmut.kuper at ewetel.de  Fri Oct 16 04:31:48 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Fri, 16 Oct 2009 13:31:48 +0200
Subject: [Freeswitch-users] Qustion about INFO messages after Connect/Answer
Message-ID: <4AD859A4.5000101@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

after updating FS to trunk a few days ago I found that callee's display
is updated serveral times to caller's name after callee picked up. The
first two equal INFO messages looks like this:

INFO sip:2850 at 85.16.245.213:1040;line=367hfn9i SIP/2.0
Via: SIP/2.0/UDP 85.16.246.6;rport;branch=z9hG4bKU4803UHXaKD2r
Max-Forwards: 70
From: "Snom1 an PBX" <sip:4918 at 85.16.246.6>;tag=ccc4B2F9SD2rm
To: <sip:2850 at 85.16.245.213:1040;line=367hfn9i>;tag=zbrirvg9ow
Call-ID: 97a0e683-34e6-122d-8db2-00144fe6e330
CSeq: 121727215 INFO
Contact: <sip:mod_sofia at 85.16.246.6:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15137M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Type: message/sipfrag
Content-Length: 27

From:
To: "unknown" 2850


The third one is this:

INFO sip:2850 at 85.16.245.213:1040;line=367hfn9i SIP/2.0
Via: SIP/2.0/UDP 85.16.246.6;rport;branch=z9hG4bK0H7vc35evZvZj
Max-Forwards: 70
From: "Snom1 an PBX" <sip:4918 at 85.16.246.6>;tag=ccc4B2F9SD2rm
To: <sip:2850 at 85.16.245.213:1040;line=367hfn9i>;tag=zbrirvg9ow
Call-ID: 97a0e683-34e6-122d-8db2-00144fe6e330
CSeq: 121727216 INFO
Contact: <sip:mod_sofia at 85.16.246.6:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15137M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Type: message/sipfrag
Content-Length: 32

From:
To: "Snom1 an PBX" 4918


What calle sees is this:
1. Phone rings: caller's displayname
2. Callee picks up: switching from dislayname to unknown
3. Switching from unknown to displayname

After callee has picked up caller's display is also updated with
callee's name (this is a copy of callee's number rather than callee's
name) and number.

First Question:
Can this behaviour be disabled? Or can it be modified by dialplan, so
that there is no "unknown"?




Unfortunately when you do an outgoing call via openzap and callee picks
up the phone caller's display is updated with two equal INFO message
like this:

INFO sip:2850 at 85.16.245.213:1040;line=367hfn9i SIP/2.0
Via: SIP/2.0/UDP 85.16.246.6;rport;branch=z9hG4bK7jFa9Ztg39UUr
Max-Forwards: 70
From: <sip:890327 at 85.16.246.6;user=phone>;tag=9QZamXtmUyXvm
To: "Helmut Kuper" <sip:2850 at 85.16.246.6>;tag=5lf2e0q1on
Call-ID: 3c2a2539a966-0m3i9s85my46
CSeq: 121727707 INFO
Contact: <sip:890327 at 85.16.246.6:5060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15137M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Type: message/sipfrag
Content-Length: 36

From:
To: "1/a/890327" 1/a/890327


Second Question:
Can I change the name and number via dialplan, so that the correkt name
and number is viewed to caller?


regards
helmut
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From mike at jerris.com  Fri Oct 16 06:27:21 2009
From: mike at jerris.com (Michael Jerris)
Date: Fri, 16 Oct 2009 09:27:21 -0400
Subject: [Freeswitch-users] sched_api doesn't get launched
In-Reply-To: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
References: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
Message-ID: <3F1747F9-C543-4C6F-BB01-4DD5FA5B7BC5@jerris.com>

sched_api is a fsapi command not a dialplan application, I believe  
sched_hangup is both.

Mike


On Oct 13, 2009, at 6:14 AM, Henry Huang wrote:

> Hi:
>
> I am using mod_java. And in my script I was able to achieve using:
>
> execute("sched_hangup", "+300 alloted_timeout");
>
> However, when I try to run sched_api in the same way, system log  
> returns that it's an invalid application. I have also tried to  
> trigger it with many conditional channel variable api calls , but  
> non of them seemed to execute the api command (because I turned on  
> the highest level of debugging and see no where the sched_api is  
> being called.
>
> The closest thing I got was by using "api_after_bridge" like the  
> following, but it only launches when the bridge is teared down(which  
> is not what I want). I originally thought after bridge means right  
> after the 2 party is connected. All I want is to be able to play  
> some message to leg A at certain time.
>
> setVariable("api_after_bridge", "sched_api +10 none uuid_displace $ 
> {uuid} start /path/to/some.wav 20 mux");
>
> I have been struggling with different combination for a week now..  
> Please shed some light if you know something.
>
> Thanks,
>
> -- 
> Henry Huang
> UniC Solution - Communication Unified
> VoIP & Open Source software Consultant
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From mike at jerris.com  Fri Oct 16 06:29:39 2009
From: mike at jerris.com (Michael Jerris)
Date: Fri, 16 Oct 2009 09:29:39 -0400
Subject: [Freeswitch-users] Building Freeswitch with VPATH (src and obj
	directories are different)?
In-Reply-To: <m3eip65vso.fsf@mad06.wl0.org>
References: <20091013205907.GA7545@mad06.wl0.org>
	<D8EB1A6C-38A6-4BC2-AD8C-46CF11993366@freeswitch.org>
	<m3eip65vso.fsf@mad06.wl0.org>
Message-ID: <8C5898A4-E257-4FBE-A8EC-439402965FDB@jerris.com>

I would love to see this work in tree, but i am pretty sure it has  
never worked.  I would gladly accept patches that implement this.


Mike

On Oct 14, 2009, at 2:33 AM, Simon J Mudd wrote:

> brian at freeswitch.org (Brian West) writes:
>
>> You shouldn't have to make clean usually ... doing so might break  
>> your
>> tree...
>
> Why?
>
> In any case on my mac (Leopard) neither make clean or distclean fully
> cleans up afterwards.  I would prefer to get a completely "untouched"
> source tree after doing this.  If that's supposed to work and I should
> post a bug or report the problem in more detail then I can do that.
>
>> You can usually get by with "make current" that will ensure
>> the critical things are cleaned and built correctly... every now and
>> they you'll hit a snafu but we'll usually tell you about it.
>
> ok. That's one of the reasons for liking to use VPATH builds. It's  
> also
> why many BSD system builds use this so that you can mount the source
> tree readonly (perhaps from another server) and build the binaries
> in a separate directory independently.
>
> If that's not supported then that's fine.
>
> I was trying to build the latest svn version of freeswitch (Revision:
> 15138) and this didn't build. As I'm not too familiar with compiling
> at least on the Mac I thought I'd first try to see if a VPATH build
> would work better. I'll have to try and investigate further.




From mike at jerris.com  Fri Oct 16 06:33:59 2009
From: mike at jerris.com (Michael Jerris)
Date: Fri, 16 Oct 2009 09:33:59 -0400
Subject: [Freeswitch-users] T.38 via UPDATE request
In-Reply-To: <4AD5AE3E.9070206@gmx.net>
References: <4AD5AE3E.9070206@gmx.net>
Message-ID: <0C137031-4C23-4A70-A05F-DE038D005839@jerris.com>

There was just a bunch of work on UPDATE, can you confirm this is the  
same behavior with trunk?

On Oct 14, 2009, at 6:55 AM, Peter P GMX wrote:

> Hello,
>
> we have the following problem.
> 2 Fax machines are communicating via Freeswitch. One is externally
> attached via a Telco who is able to handle T.38. The other one is
> attached locally.
>
> When 2 Fax machines start syncing each other, the Telco sends a SIP
> UPDATE message with T.38 SDP, as it detects fax during the fax  
> negociations.
> Freeswitch answers with an SIP OK message back to the telco, and I can
> see the T.38 SDP on the debug console of freeswitch.
> Then nothing happens any more until one of fax machines detects  
> timeout.
>
> We have set proxy-media to true. However is was done during call setup
> when both machines communicated with G711 SDP.
> The UPDATE message was commited by FS to the telco, but was not sent  
> to
> the other fax, so I think in this case Freeswitch is supposed to
> transcode between T.38 and G711 which it cannot do, as we know.
>
> How can I overcome this scenario? Is this a defect, should freeswitch
> send the UPDATE message to the other fax?  Or is there a workaround?
>
> Best regards
> Peter




From mike at jerris.com  Fri Oct 16 06:36:29 2009
From: mike at jerris.com (Michael Jerris)
Date: Fri, 16 Oct 2009 09:36:29 -0400
Subject: [Freeswitch-users] TLS Audiocodes
In-Reply-To: <4a4d27d30910132316g206580f0ya32287291af00c5b@mail.gmail.com>
References: <4a4d27d30910132316g206580f0ya32287291af00c5b@mail.gmail.com>
Message-ID: <E82402B2-857E-45EC-AC6A-0FF18983F9CE@jerris.com>

Try turning up all the sofia debug to 9.

Mike

On Oct 14, 2009, at 2:16 AM, Szasz Szabolcs wrote:

> Hi,
>
> Did anybody set up TLS between Freeswitch and Audiodes MP11X ? I got  
> to work TLS between freeswitch  and a softphone (phonerlite), but I  
> have problem with Audiocodes during the TLS authentication. I've  
> loaded the certification but it still doesn't work. Can I debug the  
> tls in freeswitch? Please help me.
>



From pedroprado at msn.com  Fri Oct 16 06:48:48 2009
From: pedroprado at msn.com (Pedro Prado)
Date: Fri, 16 Oct 2009 10:48:48 -0300
Subject: [Freeswitch-users] Brazilians (Off-Topic)
Message-ID: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>


Hi,

Do you have a group of Brazilians here?

Thanks,
Pedro Prado
 		 	   		  
_________________________________________________________________
Voc? sabia que com o Hotmail voc? tem espa?o ilimitado para guardar seus e-mails? Come?e a usar j?!
http://www.microsoft.com/brasil/windows/windowslive/products/hotmail.aspx
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From mgende at gendesign.com  Fri Oct 16 07:02:28 2009
From: mgende at gendesign.com (Michael Gende)
Date: Fri, 16 Oct 2009 09:02:28 -0500
Subject: [Freeswitch-users] Subject:, Re: FreeSWITCH-users Digest,
	Orien 	L. answer to Mike G.
In-Reply-To: <C8C9B2D0FA5E40C4AA2658B34D0092DD@cune.pri>
References: <mailman.1290.1255619700.2495.freeswitch-users@lists.freeswitch.org>
	<4AD75F39.6010209@tx.rr.com>
	<a87d43fd0910152202k2c7b3300ya139a81385f60bf2@mail.gmail.com>
	<C8C9B2D0FA5E40C4AA2658B34D0092DD@cune.pri>
Message-ID: <a87d43fd0910160702y5ab70c29w3bb22a903ddc59b7@mail.gmail.com>

I didn't know that. Thanks Russell.


>
>
>
> You've mentioned this twice, now, but a better way to do it is to _not_
> stop FreeSwitch. Instead, run /usr/local/freeswitch/bin/fs_cli, which
> connects to FS, and then you can do everything that you can do directly in
> FS. When you are done, enter "\quit" or "\bye" or "\exit". See "\help" and
> "help" for more information.
>
> --
>
> Russell Mosemann
>
>
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From itamar at ispbrasil.com.br  Fri Oct 16 07:02:28 2009
From: itamar at ispbrasil.com.br (Itamar Reis Peixoto)
Date: Fri, 16 Oct 2009 11:02:28 -0300
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
Message-ID: <d3ceea380910160702x8acbc4ci3bf3cfc1fdf9a9b7@mail.gmail.com>

I think no.


On Fri, Oct 16, 2009 at 10:48 AM, Pedro Prado <pedroprado at msn.com> wrote:
> Hi,
>
> Do you have a group of Brazilians here?
>
> Thanks,
> Pedro Prado



-- 
------------

Itamar Reis Peixoto

e-mail/msn: itamar at ispbrasil.com.br
sip: itamar at ispbrasil.com.br
skype: itamarjp
icq: 81053601
+55 11 4063 5033
+55 34 3221 8599



From jmesquita at freeswitch.org  Fri Oct 16 07:25:25 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 16 Oct 2009 11:25:25 -0300
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <d3ceea380910160702x8acbc4ci3bf3cfc1fdf9a9b7@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<d3ceea380910160702x8acbc4ci3bf3cfc1fdf9a9b7@mail.gmail.com>
Message-ID: <ebbc2d550910160725x3141ce14re1821926d4027d42@mail.gmail.com>

We do. :-)

Jo?o Mesquita

On Fri, Oct 16, 2009 at 11:02 AM, Itamar Reis Peixoto <
itamar at ispbrasil.com.br> wrote:

> I think no.
>
>
> On Fri, Oct 16, 2009 at 10:48 AM, Pedro Prado <pedroprado at msn.com> wrote:
> > Hi,
> >
> > Do you have a group of Brazilians here?
> >
> > Thanks,
> > Pedro Prado
>
>
>
> --
> ------------
>
> Itamar Reis Peixoto
>
> e-mail/msn: itamar at ispbrasil.com.br
> sip: itamar at ispbrasil.com.br
> skype: itamarjp
> icq: 81053601
> +55 11 4063 5033
> +55 34 3221 8599
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Prometheus001 at gmx.net  Fri Oct 16 07:35:22 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 16 Oct 2009 16:35:22 +0200
Subject: [Freeswitch-users] T.38 via UPDATE request
In-Reply-To: <0C137031-4C23-4A70-A05F-DE038D005839@jerris.com>
References: <4AD5AE3E.9070206@gmx.net>
	<0C137031-4C23-4A70-A05F-DE038D005839@jerris.com>
Message-ID: <4AD884AA.1050805@gmx.net>

Hello Michael

this is a productive system, so I can currently not update to the
current trunk.
But the installed SVN is 14741. Were those changes after 14741? Then
I'll try to find a timeslot at night in order to update freeswitch.

But Freeswitch should forward the UPDATE request in proxy-media mode, right?

Best regards
Peter

Michael Jerris schrieb:
> There was just a bunch of work on UPDATE, can you confirm this is the  
> same behavior with trunk?
>
> On Oct 14, 2009, at 6:55 AM, Peter P GMX wrote:
>
>   
>> Hello,
>>
>> we have the following problem.
>> 2 Fax machines are communicating via Freeswitch. One is externally
>> attached via a Telco who is able to handle T.38. The other one is
>> attached locally.
>>
>> When 2 Fax machines start syncing each other, the Telco sends a SIP
>> UPDATE message with T.38 SDP, as it detects fax during the fax  
>> negociations.
>> Freeswitch answers with an SIP OK message back to the telco, and I can
>> see the T.38 SDP on the debug console of freeswitch.
>> Then nothing happens any more until one of fax machines detects  
>> timeout.
>>
>> We have set proxy-media to true. However is was done during call setup
>> when both machines communicated with G711 SDP.
>> The UPDATE message was commited by FS to the telco, but was not sent  
>> to
>> the other fax, so I think in this case Freeswitch is supposed to
>> transcode between T.38 and G711 which it cannot do, as we know.
>>
>> How can I overcome this scenario? Is this a defect, should freeswitch
>> send the UPDATE message to the other fax?  Or is there a workaround?
>>
>> Best regards
>> Peter
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   



From brian at freeswitch.org  Fri Oct 16 07:48:37 2009
From: brian at freeswitch.org (Brian West)
Date: Fri, 16 Oct 2009 09:48:37 -0500
Subject: [Freeswitch-users] Dingaling: using a hostname instead of stun
	for rtp
In-Reply-To: <33c87fa30910152035w43e260faoc031a7a4ac949d9e@mail.gmail.com>
References: <33c87fa30910131502t2af24801i1c3319518b5db50b@mail.gmail.com>
	<C874E25D-6D73-49AA-8DD3-0A0022617FD2@freeswitch.org>
	<33c87fa30910152035w43e260faoc031a7a4ac949d9e@mail.gmail.com>
Message-ID: <65BF398D-E3A6-4A98-83D9-D4E271070CCA@freeswitch.org>

If you setup your own stun server it wouldn't do that....  But the  
hostlookup only solves half the problem .. getting the external IP vs  
poking holes for RTP which is what stun will do.

/b

On Oct 15, 2009, at 10:35 PM, Mark Campbell-Smith wrote:

> Thanks Brian.  Is this something that is planned to be implemented?
> The workaround is to set the stun server also in the dingaling
> configuration, but as I said, for some reason the stun times for me
> out occasionally with dingaling.
>
> Thanks!




From nbelov at abisoft.spb.ru  Fri Oct 16 08:54:47 2009
From: nbelov at abisoft.spb.ru (Nikita Belov)
Date: Fri, 16 Oct 2009 19:54:47 +0400
Subject: [Freeswitch-users] uuid_send_dtmf fails (was: conference call)
In-Reply-To: <b39273530910150504l5c7411b9g39b9ed33c592e85f@mail.gmail.com>
References: <064A5201C10D4D20AC76EB93CA5EA4B5@abisoft.biz><ebbc2d550910140838v3125579ev9fa16e3e83d25770@mail.gmail.com><BE4EDDA84CB64AA98F18D412B51912FC@abisoft.biz>
	<b39273530910150504l5c7411b9g39b9ed33c592e85f@mail.gmail.com>
Message-ID: <02C2EFD4CF8D4ACF8EEC0F77C4D75028@abisoft.biz>

Yes, it is what I need. But now I have problem with sending dtmf. Here what
I've done:
[root at centos4-4-vm ~]# telnet localhost 8021
Trying 127.0.0.1...
Connected to localhost.localdomain (127.0.0.1).
Escape character is '^]'.
Content-Type: auth/request

auth ClueCon

Content-Type: command/reply
Reply-Text: +OK accepted

api originate user/1007 at master.agent.starpoundtech.net &park()

Content-Type: api/response
Content-Length: 41

+OK bba3b45a-4cc1-48af-a15d-1052d5f11371

SendMsg bba3b45a-4cc1-48af-a15d-1052d5f11371
call-command: execute
execute-app-name: eavesdrop
execute-app-arg: cd99f999-9b47-457e-8439-1d366e015b8c

Content-Type: command/reply
Reply-Text: +OK

Here I had started to hear A and B. Here what I saw in FS log:
2009-10-18 03:22:47 [DEBUG] switch_core_session.c:706
switch_core_session_queue_private_event() Send signal
sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes [BREAK]
2009-10-18 03:22:47 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event()
sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes Command Execute
eavesdrop(cd99f999-9b47-457e-8439-1d366e015b8c)
2009-10-18 03:22:47 [DEBUG] switch_core_media_bug.c:297
switch_core_media_bug_add() Attaching BUG to
sofia/internal/1006 at master.agent.starpoundtech.net
2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234
switch_core_session_read_frame()
sofia/internal/1006 at master.agent.starpoundtech.net receive message
[TRANSCODING_NECESSARY]
2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234
switch_core_session_read_frame()
sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes receive message
[TRANSCODING_NECESSARY]

Then I run command:
api uuid_send_dtmf bba3b45a-4cc1-48af-a15d-1052d5f11371 1

Content-Type: api/response
Content-Length: 14

-ERR no reply

Log:
2009-10-18 03:24:01 [DEBUG] switch_core_io.c:1190
switch_core_session_send_dtmf_string()
sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes send dtmf
digit=1 ms=250 samples=2000
2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1282 do_2833() Send start packet
for [1] ts=2241760 dur=160/160/2000 seq=21346
2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=320/320/2000 seq=21347
2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=480/480/2000 seq=21348
2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=640/640/2000 seq=21349
2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=800/800/2000 seq=21350
2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=960/960/2000 seq=21351
2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=1120/1120/2000 seq=21352
2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=1280/1280/2000 seq=21353
2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=1440/1440/2000 seq=21354
2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=1600/1600/2000 seq=21355
2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=1760/1760/2000 seq=21356
2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=1920/1920/2000 seq=21357
2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end packet for
[1] ts=2241760 dur=2080/2080/2000 seq=21358
2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end packet for
[1] ts=2241760 dur=2080/2080/2000 seq=21359
2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end packet for
[1] ts=2241760 dur=2080/2080/2000 seq=21360

But both A and B couldn't hear me. 
Btw, after I had send dtmf "1" manually from my phone. B started to hear me.
There was this record in log:
2009-10-18 03:47:55 [DEBUG] switch_rtp.c:1767 switch_rtp_dequeue_dtmf() RTP
RECV DTMF 1:2240

Does anybody know, what had I done wrong?

___

Thanks, Nikita
    
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker
> Sent: Thursday, October 15, 2009 4:04 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] conference call
> 
> The wiki has a pretty good explanation of how eavesdrop works.
> Enabling a talk path to A or B or both A and B requires dtmf.
> 
> So, if C hits the 1 button on the phone they can talk to the UUID you
> bound the eavesdrop to.   2 talks to the other party.  3 talks to both
> parties.
> 
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop
> 
> Adding more api calls to control eavesdrop via event socket should be
> possible, though I haven't looked at the code to see what it would
> take...
> 
> On Thu, Oct 15, 2009 at 3:16 AM, Nikita Belov <nbelov at abisoft.spb.ru>
> wrote:
> > Thanks for answer.
> >
> >
> >
> > As I've understood eavesdrop allows C to hear A and B, but A and B will
> not
> > hear C. But in my case B SHOULD hear C. In my case A - client, B - agent
> > worked with client and C - coach/supervisor for B. Can I make such
> > conference in FS using eavesdrop or other API command?
> >
> >
> >
> > ___
> >
> >
> >
> > Thanks, Nikita
> >
> >
> >
> > ________________________________
> >
> > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Joao
> > Mesquita
> > Sent: Wednesday, October 14, 2009 7:38 PM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] conference call
> >
> >
> >
> > Look at eavesdrop on the wiki.
> >
> > JM
> >
> > 2009/10/14 Nikita Belov <nbelov at abisoft.spb.ru>
> >
> > HI all,
> >
> >
> >
> > I want to configure FS to make special conference call between three
> users
> > (A, B, C).? In this conference C will hear A and B, but A will hear only
> B.
> > Can I make it using FS API commands? Does anybody know which approach is
> > better to use?
> >
> >
> >
> > ___
> >
> >
> >
> > Thanks, Nikita
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> 
> 
> --
> -Rupa
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From msc at freeswitch.org  Fri Oct 16 08:57:04 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 16 Oct 2009 08:57:04 -0700
Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Starting
	Shortly, Please Start 	Calling In
Message-ID: <87f2f3b90910160857v36acb772s324389b0debb489c@mail.gmail.com>

The weekly conference call is about to start. Please call in. The agenda and
calling instructions are here:

http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_16

Looking forward to hearing from everyone!
-Michael
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From anthony.minessale at gmail.com  Fri Oct 16 09:18:12 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 16 Oct 2009 11:18:12 -0500
Subject: [Freeswitch-users] Qustion about INFO messages after
	Connect/Answer
In-Reply-To: <4AD859A4.5000101@ewetel.de>
References: <4AD859A4.5000101@ewetel.de>
Message-ID: <191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>

1) you should update again there were a few issues.
2) you can set the variable sip_callee_id_name and sip_callee_id number on
the inbound leg before you answer to control what it says.


On Fri, Oct 16, 2009 at 6:31 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> after updating FS to trunk a few days ago I found that callee's display
> is updated serveral times to caller's name after callee picked up. The
> first two equal INFO messages looks like this:
>
> INFO sip:2850 at 85.16.245.213:1040;line=367hfn9i SIP/2.0
> Via: SIP/2.0/UDP 85.16.246.6;rport;branch=z9hG4bKU4803UHXaKD2r
> Max-Forwards: 70
> From: "Snom1 an PBX" <sip:4918 at 85.16.246.6 <sip%3A4918 at 85.16.246.6>
> >;tag=ccc4B2F9SD2rm
> To: <sip:2850 at 85.16.245.213:1040;line=367hfn9i>;tag=zbrirvg9ow
> Call-ID: 97a0e683-34e6-122d-8db2-00144fe6e330
> CSeq: 121727215 INFO
> Contact: <sip:mod_sofia at 85.16.246.6:5060>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15137M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
> REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Content-Type: message/sipfrag
> Content-Length: 27
>
> From:
> To: "unknown" 2850
>
>
> The third one is this:
>
> INFO sip:2850 at 85.16.245.213:1040;line=367hfn9i SIP/2.0
> Via: SIP/2.0/UDP 85.16.246.6;rport;branch=z9hG4bK0H7vc35evZvZj
> Max-Forwards: 70
> From: "Snom1 an PBX" <sip:4918 at 85.16.246.6 <sip%3A4918 at 85.16.246.6>
> >;tag=ccc4B2F9SD2rm
> To: <sip:2850 at 85.16.245.213:1040;line=367hfn9i>;tag=zbrirvg9ow
> Call-ID: 97a0e683-34e6-122d-8db2-00144fe6e330
> CSeq: 121727216 INFO
> Contact: <sip:mod_sofia at 85.16.246.6:5060>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15137M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
> REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Content-Type: message/sipfrag
> Content-Length: 32
>
> From:
> To: "Snom1 an PBX" 4918
>
>
> What calle sees is this:
> 1. Phone rings: caller's displayname
> 2. Callee picks up: switching from dislayname to unknown
> 3. Switching from unknown to displayname
>
> After callee has picked up caller's display is also updated with
> callee's name (this is a copy of callee's number rather than callee's
> name) and number.
>
> First Question:
> Can this behaviour be disabled? Or can it be modified by dialplan, so
> that there is no "unknown"?
>
>
>
>
> Unfortunately when you do an outgoing call via openzap and callee picks
> up the phone caller's display is updated with two equal INFO message
> like this:
>
> INFO sip:2850 at 85.16.245.213:1040;line=367hfn9i SIP/2.0
> Via: SIP/2.0/UDP 85.16.246.6;rport;branch=z9hG4bK7jFa9Ztg39UUr
> Max-Forwards: 70
> From: <sip:890327 at 85.16.246.6 <sip%3A890327 at 85.16.246.6>
> ;user=phone>;tag=9QZamXtmUyXvm
> To: "Helmut Kuper" <sip:2850 at 85.16.246.6 <sip%3A2850 at 85.16.246.6>
> >;tag=5lf2e0q1on
> Call-ID: 3c2a2539a966-0m3i9s85my46
> CSeq: 121727707 INFO
> Contact: <sip:890327 at 85.16.246.6:5060;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15137M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
> REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Content-Type: message/sipfrag
> Content-Length: 36
>
> From:
> To: "1/a/890327" 1/a/890327
>
>
> Second Question:
> Can I change the name and number via dialplan, so that the correkt name
> and number is viewed to caller?
>
>
> regards
> helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
>
> iD8DBQFK2Fmk4tZeNddg3dwRApPUAKCQLBEi/l6mwf1c55r5mlnEqvLLowCgoP9t
> VSQFKUCWpsxVm3evyfWQT/Y=
> =FGsp
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From jerry.richards at teotech.com  Fri Oct 16 09:47:17 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 16 Oct 2009 09:47:17 -0700
Subject: [Freeswitch-users] scripts/contrib/trixter/makemodconf.pl: No such
	file or directory
Message-ID: <43360B9674B446119482FC37680F6F5F@greyhawk.tonecommander.com>

I am building Freeswitch on a Centos 5.3 machine and the last step below
gets an error because there is no scripts/contrib folder.  Anyone know why?

./configure
make
make all install sounds-install uhd-moh-install moh-install
scripts/contrib/trixter/makemodconf.pl modules.conf >
/usr/local/freeswitch/conf/autoload_configs/modules.conf.xml

bash: scripts/contrib/trixter/makemodconf.pl: No such file or directory

Best Regards,
Jerry




From jerry.richards at teotech.com  Fri Oct 16 10:16:04 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 16 Oct 2009 10:16:04 -0700
Subject: [Freeswitch-users] scripts/contrib/trixter/makemodconf.pl: No
	such file or directory
Message-ID: <1E8E134F44F146558C9CC9613FF48C2C@greyhawk.tonecommander.com>


Okay, I think the contrib folder moved up one level.  So the Wiki
installation documentation should probably be updated to reflect that.

Best Regards,
Jerry


-----Original Message-----
From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Friday, October 16, 2009 9:47 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: scripts/contrib/trixter/makemodconf.pl: No such file or directory

I am building Freeswitch on a Centos 5.3 machine and the last step below
gets an error because there is no scripts/contrib folder.  Anyone know why?

./configure
make
make all install sounds-install uhd-moh-install moh-install
scripts/contrib/trixter/makemodconf.pl modules.conf >
/usr/local/freeswitch/conf/autoload_configs/modules.conf.xml

bash: scripts/contrib/trixter/makemodconf.pl: No such file or directory

Best Regards,
Jerry




From miles.chet at gmail.com  Fri Oct 16 10:20:37 2009
From: miles.chet at gmail.com (Roberto Martins)
Date: Fri, 16 Oct 2009 14:20:37 -0300
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <ebbc2d550910160725x3141ce14re1821926d4027d42@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<d3ceea380910160702x8acbc4ci3bf3cfc1fdf9a9b7@mail.gmail.com>
	<ebbc2d550910160725x3141ce14re1821926d4027d42@mail.gmail.com>
Message-ID: <2E845C1E-C084-4025-806D-178A175FF90A@gmail.com>

we do!.

roberto
On Oct 16, 2009, at 11:25 AM, Jo?o Mesquita wrote:

> We do. :-)
>
> Jo?o Mesquita
>
> On Fri, Oct 16, 2009 at 11:02 AM, Itamar Reis Peixoto <itamar at ispbrasil.com.br 
> > wrote:
> I think no.
>
>
> On Fri, Oct 16, 2009 at 10:48 AM, Pedro Prado <pedroprado at msn.com>  
> wrote:
> > Hi,
> >
> > Do you have a group of Brazilians here?
> >
> > Thanks,
> > Pedro Prado
>
>
>
> --
> ------------
>
> Itamar Reis Peixoto
>
> e-mail/msn: itamar at ispbrasil.com.br
> sip: itamar at ispbrasil.com.br
> skype: itamarjp
> icq: 81053601
> +55 11 4063 5033
> +55 34 3221 8599
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From Russell.Mosemann at cune.org  Fri Oct 16 10:29:36 2009
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Fri, 16 Oct 2009 17:29:36 -0000
Subject: [Freeswitch-users] Outbound sip address call fails
In-Reply-To: <011301ca4dd3$67d486a0$377d93e0$@com>
Message-ID: <20091016172936.CCE7C41E8F6@mail.cune.org>

Lars Zeb <larclap at yahoo.com> said:

> I tried dial sip:echo at iptel.org. In the pastebin you can see that this
> address came over to the cli as "echo". Is it Bria which is messing up, or
> my FS configuration?

It looks like a Bria problem. If you open the account information and go
to the Advanced tab, is the box checked to "Send outgoing request
directly to target"?


-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!




From brian at freeswitch.org  Fri Oct 16 10:38:51 2009
From: brian at freeswitch.org (Brian West)
Date: Fri, 16 Oct 2009 12:38:51 -0500
Subject: [Freeswitch-users] scripts/contrib/trixter/makemodconf.pl: No
	such file or directory
In-Reply-To: <1E8E134F44F146558C9CC9613FF48C2C@greyhawk.tonecommander.com>
References: <1E8E134F44F146558C9CC9613FF48C2C@greyhawk.tonecommander.com>
Message-ID: <803E6637-5D7C-4B88-84B1-9947A88E4B95@freeswitch.org>

You can signup for a wiki account and correct the documentation so  
others don't repeat the same issue.

Thanks,
Brian

On Oct 16, 2009, at 12:16 PM, Jerry Richards wrote:

>
> Okay, I think the contrib folder moved up one level.  So the Wiki
> installation documentation should probably be updated to reflect that.
>
> Best Regards,
> Jerry




From red.rain.seven at gmail.com  Fri Oct 16 11:53:43 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Sat, 17 Oct 2009 02:53:43 +0800
Subject: [Freeswitch-users] sched_api doesn't get launched
In-Reply-To: <3F1747F9-C543-4C6F-BB01-4DD5FA5B7BC5@jerris.com>
References: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
	<3F1747F9-C543-4C6F-BB01-4DD5FA5B7BC5@jerris.com>
Message-ID: <59ad9ca10910161153h5d531533vd844e49b9ca6d896@mail.gmail.com>

So how would you trigger it from a script dialplan? The only time it seemed
to work is when I did setVariable("api_after_bridge", "sched_api blah blah
blah");
but then it gets executed after the channel's been teared down. I thought
api_after_bridge means right after the call gets connected.

I need something to execute an api command right before or right after the
call gets bridged.

On Fri, Oct 16, 2009 at 9:27 PM, Michael Jerris <mike at jerris.com> wrote:

> sched_api is a fsapi command not a dialplan application, I believe
> sched_hangup is both.
> Mike
>
>
> On Oct 13, 2009, at 6:14 AM, Henry Huang wrote:
>
> Hi:
> I am using mod_java. And in my script I was able to achieve using:
>
> execute("sched_hangup", "+300 alloted_timeout");
>
> However, when I try to run sched_api in the same way, system log returns
> that it's an invalid application. I have also tried to trigger it with many
> conditional channel variable api calls , but non of them seemed to execute
> the api command (because I turned on the highest level of debugging and see
> no where the sched_api is being called.
>
> The closest thing I got was by using "api_after_bridge" like the following,
> but it only launches when the bridge is teared down(which is not what I
> want). I originally thought after bridge means right after the 2 party is
> connected. All I want is to be able to play some message to leg A at certain
> time.
>
> setVariable("api_after_bridge", "sched_api +10 none uuid_displace ${uuid}
> start /path/to/some.wav 20 mux");
>
> I have been struggling with different combination for a week now.. Please
> shed some light if you know something.
>
> Thanks,
>
> --
> Henry Huang
> UniC Solution - Communication Unified
> VoIP & Open Source software Consultant
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From msc at freeswitch.org  Fri Oct 16 12:37:20 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 16 Oct 2009 12:37:20 -0700
Subject: [Freeswitch-users] sched_api doesn't get launched
In-Reply-To: <59ad9ca10910161153h5d531533vd844e49b9ca6d896@mail.gmail.com>
References: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
	<3F1747F9-C543-4C6F-BB01-4DD5FA5B7BC5@jerris.com>
	<59ad9ca10910161153h5d531533vd844e49b9ca6d896@mail.gmail.com>
Message-ID: <87f2f3b90910161237h356b7aa2q6ec12c6ecf78dc34@mail.gmail.com>

On Fri, Oct 16, 2009 at 11:53 AM, Henry Huang <red.rain.seven at gmail.com>wrote:

> So how would you trigger it from a script dialplan? The only time it seemed
> to work is when I did setVariable("api_after_bridge", "sched_api blah blah
> blah");
> but then it gets executed after the channel's been teared down. I thought
> api_after_bridge means right after the call gets connected.
>
> I need something to execute an api command right before or right after the
> call gets bridged.
>
> api_after_bridge is a channel variable, so using setVariable works just
fine. If you need to sched_api is an API only. Check these out:
http://wiki.freeswitch.org/wiki/Mod_commands#Misc._Commands

So you need an API object in order to use it. I don't know the syntax for
creating an api obj in Java but in Lua it goes like this:
api = freeswitch.API();
res = api:execute("sched_api","+300 none my_api my_api_args")

Remember, if the method you are using isn't found in the dial plan tools
then it isn't a dial plan application. Make sure it's on the list:
http://wiki.freeswitch.org/wiki/Mod_dptools

On the other hand, API commands are listed here:
http://wiki.freeswitch.org/wiki/Mod_commands

dptools require a session object, api commands require an api object...

-MC
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From fabioferrari at gmail.com  Fri Oct 16 13:02:20 2009
From: fabioferrari at gmail.com (Fabio Ferrari)
Date: Fri, 16 Oct 2009 17:02:20 -0300
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <2E845C1E-C084-4025-806D-178A175FF90A@gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<d3ceea380910160702x8acbc4ci3bf3cfc1fdf9a9b7@mail.gmail.com> 
	<ebbc2d550910160725x3141ce14re1821926d4027d42@mail.gmail.com> 
	<2E845C1E-C084-4025-806D-178A175FF90A@gmail.com>
Message-ID: <f3cdc69d0910161302v6a97ead2xc2af5d73b5a048d7@mail.gmail.com>

Just to let you know, Jeremias Silva from Khomp registered the
freeswitchbrasil.org domain. I think there is space to build an wiki
with portuguese content there.

Fabio Ferrari

On Fri, Oct 16, 2009 at 14:20, Roberto Martins <miles.chet at gmail.com> wrote:
> we do!.
>
> roberto
> On Oct 16, 2009, at 11:25 AM, Jo?o Mesquita wrote:
>
>> We do. :-)
>>
>> Jo?o Mesquita
>>
>> On Fri, Oct 16, 2009 at 11:02 AM, Itamar Reis Peixoto <itamar at ispbrasil.com.br
>> > wrote:
>> I think no.
>>
>>
>> On Fri, Oct 16, 2009 at 10:48 AM, Pedro Prado <pedroprado at msn.com>
>> wrote:
>> > Hi,
>> >
>> > Do you have a group of Brazilians here?
>> >
>> > Thanks,
>> > Pedro Prado
>>
>>
>>
>> --
>> ------------
>>
>> Itamar Reis Peixoto
>>
>> e-mail/msn: itamar at ispbrasil.com.br
>> sip: itamar at ispbrasil.com.br
>> skype: itamarjp
>> icq: 81053601
>> +55 11 4063 5033
>> +55 34 3221 8599
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From tculjaga at gmail.com  Fri Oct 16 13:13:07 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 16 Oct 2009 22:13:07 +0200
Subject: [Freeswitch-users] Media got stuck after attended transfer...
In-Reply-To: <87f2f3b90910151710k34e4092eg26108dd819d9c041@mail.gmail.com>
References: <A2F1C0E212BBBA4E8C0E7BE568361F04027F02330253@srv02.khdev.corp>
	<191c3a030910150657r668eb5a3q24c641e312d2b113@mail.gmail.com>
	<65d96fc80910151154w2468ebeie06211d0966b4548@mail.gmail.com>
	<87f2f3b90910151710k34e4092eg26108dd819d9c041@mail.gmail.com>
Message-ID: <65d96fc80910161313p5d0aacecq1f1bb71ad514921e@mail.gmail.com>

> Of course, I was listening to my A.M radio the other day and they said that
> there was this new invention called the Internet that would let people send
> documents to each other electronically. Maybe you should look into that.
> Next thing you know they'll come up with telephones that people don't have
> to plug into the wall and can take with them in the car. ;)
>
>
yes, in that galaxy far far away :P



> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From itamar at ispbrasil.com.br  Fri Oct 16 13:28:30 2009
From: itamar at ispbrasil.com.br (Itamar Reis Peixoto)
Date: Fri, 16 Oct 2009 17:28:30 -0300
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <f3cdc69d0910161302v6a97ead2xc2af5d73b5a048d7@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<d3ceea380910160702x8acbc4ci3bf3cfc1fdf9a9b7@mail.gmail.com>
	<ebbc2d550910160725x3141ce14re1821926d4027d42@mail.gmail.com>
	<2E845C1E-C084-4025-806D-178A175FF90A@gmail.com>
	<f3cdc69d0910161302v6a97ead2xc2af5d73b5a048d7@mail.gmail.com>
Message-ID: <d3ceea380910161328w4c843e9as49b2341dc6f8eb7d@mail.gmail.com>

instead of creating thousand of small sites, why don't put the content
in only place.

freeswitch.org already exist, no need for a new domain.



On Fri, Oct 16, 2009 at 5:02 PM, Fabio Ferrari <fabioferrari at gmail.com> wrote:
> Just to let you know, Jeremias Silva from Khomp registered the
> freeswitchbrasil.org domain. I think there is space to build an wiki
> with portuguese content there.
>
> Fabio Ferrari


------------

Itamar Reis Peixoto

e-mail/msn: itamar at ispbrasil.com.br
sip: itamar at ispbrasil.com.br
skype: itamarjp
icq: 81053601
+55 11 4063 5033
+55 34 3221 8599



From anthony.minessale at gmail.com  Fri Oct 16 14:56:44 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 16 Oct 2009 16:56:44 -0500
Subject: [Freeswitch-users] Dingaling / Jingle DTMF support?
In-Reply-To: <33c87fa30910151519x637618d3p109f1b8edbf3432e@mail.gmail.com>
References: <33c87fa30910151519x637618d3p109f1b8edbf3432e@mail.gmail.com>
Message-ID: <191c3a030910161456y65d49120p8479c7c5df675125@mail.gmail.com>

It already should support at least recv of rfc2833 any time.
as a workaround in googletalk any string that starts with + typed in the
chat box is treated as dtmf by FS
e.g. +1

Once the jingle spec stops being a moving target we will re-investigate
making sure its supported properly.


On Thu, Oct 15, 2009 at 5:19 PM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> Hi!
>
> I was wondering if the Dingaling implementation in FS supports DTMF?
> This is now supported in the jingle specs (
> http://www.jabberforum.org/showthread.php?t=2709 ), even though Google
> Talk client does not currently support DTMF.
>
> If not, are there plans to implement this?
>
> Thanks!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Fri Oct 16 14:59:48 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 16 Oct 2009 16:59:48 -0500
Subject: [Freeswitch-users] T.38 via UPDATE request
In-Reply-To: <4AD884AA.1050805@gmx.net>
References: <4AD5AE3E.9070206@gmx.net>
	<0C137031-4C23-4A70-A05F-DE038D005839@jerris.com>
	<4AD884AA.1050805@gmx.net>
Message-ID: <191c3a030910161459t77162434k3b6a8977b594f393@mail.gmail.com>

I'm sure it won't work because we only implemented forwarding t38 with
invite
it would require adding code to say the least.


On Fri, Oct 16, 2009 at 9:35 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Hello Michael
>
> this is a productive system, so I can currently not update to the
> current trunk.
> But the installed SVN is 14741. Were those changes after 14741? Then
> I'll try to find a timeslot at night in order to update freeswitch.
>
> But Freeswitch should forward the UPDATE request in proxy-media mode,
> right?
>
> Best regards
> Peter
>
> Michael Jerris schrieb:
> > There was just a bunch of work on UPDATE, can you confirm this is the
> > same behavior with trunk?
> >
> > On Oct 14, 2009, at 6:55 AM, Peter P GMX wrote:
> >
> >
> >> Hello,
> >>
> >> we have the following problem.
> >> 2 Fax machines are communicating via Freeswitch. One is externally
> >> attached via a Telco who is able to handle T.38. The other one is
> >> attached locally.
> >>
> >> When 2 Fax machines start syncing each other, the Telco sends a SIP
> >> UPDATE message with T.38 SDP, as it detects fax during the fax
> >> negociations.
> >> Freeswitch answers with an SIP OK message back to the telco, and I can
> >> see the T.38 SDP on the debug console of freeswitch.
> >> Then nothing happens any more until one of fax machines detects
> >> timeout.
> >>
> >> We have set proxy-media to true. However is was done during call setup
> >> when both machines communicated with G711 SDP.
> >> The UPDATE message was commited by FS to the telco, but was not sent
> >> to
> >> the other fax, so I think in this case Freeswitch is supposed to
> >> transcode between T.38 and G711 which it cannot do, as we know.
> >>
> >> How can I overcome this scenario? Is this a defect, should freeswitch
> >> send the UPDATE message to the other fax?  Or is there a workaround?
> >>
> >> Best regards
> >> Peter
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From diego.viola at gmail.com  Fri Oct 16 15:10:35 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 16 Oct 2009 22:10:35 +0000
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <6b65470d0910131009y1cf136c3q8c0932375c19f272@mail.gmail.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
	<891080.87141.qm@web56406.mail.re3.yahoo.com>
	<87f2f3b90910130952r5657c8e2g6e11c22140da919c@mail.gmail.com>
	<6b65470d0910131009y1cf136c3q8c0932375c19f272@mail.gmail.com>
Message-ID: <86a32abc0910161510ma323718g410bcf1029156077@mail.gmail.com>

Can someone please convert this IRC log to documentation form:

http://wiki.freeswitch.org/wiki/Cacti

Try to explain what the IRC log explains in documentation form, and get rid
of the IRC log.

I'm working on fixing typos and stuff like that on the wiki now, then I will
focus on fixing the format, so all pages look the same (titles, choice of
case in text, etc.)

Thanks,

Diego

On Tue, Oct 13, 2009 at 5:09 PM, William Suffill
<william.suffill at gmail.com>wrote:

> Under special pages there is ways to get a list of all the wiki pages. (
> http://wiki.freeswitch.org/wiki/Special:SpecialPages)
>
> http://wiki.freeswitch.org/wiki/Special:AllPages
>
> Due to the number of pages it's broken into sub pages based in alphabetical
> order of the page names.
>
> -- W
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tayeb.meftah at gmail.com  Fri Oct 16 17:51:10 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sat, 17 Oct 2009 00:51:10 +0000
Subject: [Freeswitch-users] PBX vs. SIP proxy?
In-Reply-To: <25920788.post@talk.nabble.com>
References: <25920788.post@talk.nabble.com>
Message-ID: <4AD914FE.4020709@gmail.com>

hello
for sip proxy,
Click here <http://www.opensips.org/>
(http://www.opensips.org)
for a Ip PBX, sted with freeswitch (
http://www.Freeswitch.org)
sip provide a registrare to let phones register and proxy there request 
to other route or something
while PBX let local dialing, conferencing, voicemail, call  parcking and 
other stufs...
thank you and welcome to freeswitch
Fred-145 a ?crit :
> Hello
>
> I was reading an article the other day about Freeswitch in which the author
> used the terms "SIP proxy" and PBX. Is an SIP proxy just a way for SIP
> clients to register and connect to each other, while the PBX part adds
> services like voice mail, etc.?
>
> Thank you.
>   




__________ Information from ESET NOD32 Antivirus, version of virus signature database 4516 (20091016) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From mgende at gendesign.com  Fri Oct 16 17:29:46 2009
From: mgende at gendesign.com (Michael Gende)
Date: Fri, 16 Oct 2009 19:29:46 -0500
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <86a32abc0910161510ma323718g410bcf1029156077@mail.gmail.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
	<891080.87141.qm@web56406.mail.re3.yahoo.com>
	<87f2f3b90910130952r5657c8e2g6e11c22140da919c@mail.gmail.com>
	<6b65470d0910131009y1cf136c3q8c0932375c19f272@mail.gmail.com>
	<86a32abc0910161510ma323718g410bcf1029156077@mail.gmail.com>
Message-ID: <a87d43fd0910161729u42db5ea7qafb5604c0680954@mail.gmail.com>

Hey Diego,

You seem to know your way around the documentation and clearly have many
contributions. Thus, a question:

Can you point me to doc on dial plan applications (such as transfer, bridge,
etc)? It would help me greatly. Sorry if this is in an obvious place and
I've just not seen it despite the nose on my face.

Regards,

Mike G.

On Fri, Oct 16, 2009 at 5:10 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Can someone please convert this IRC log to documentation form:
>
> http://wiki.freeswitch.org/wiki/Cacti
>
> Try to explain what the IRC log explains in documentation form, and get rid
> of the IRC log.
>
> I'm working on fixing typos and stuff like that on the wiki now, then I
> will focus on fixing the format, so all pages look the same (titles, choice
> of case in text, etc.)
>
> Thanks,
>
> Diego
>
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From diego.viola at gmail.com  Fri Oct 16 18:11:42 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 17 Oct 2009 01:11:42 +0000
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <a87d43fd0910161729u42db5ea7qafb5604c0680954@mail.gmail.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
	<891080.87141.qm@web56406.mail.re3.yahoo.com>
	<87f2f3b90910130952r5657c8e2g6e11c22140da919c@mail.gmail.com>
	<6b65470d0910131009y1cf136c3q8c0932375c19f272@mail.gmail.com>
	<86a32abc0910161510ma323718g410bcf1029156077@mail.gmail.com>
	<a87d43fd0910161729u42db5ea7qafb5604c0680954@mail.gmail.com>
Message-ID: <86a32abc0910161811h15eac05dwc3fe0c2533b7397a@mail.gmail.com>

Hi Michael,

You might want to look at the mod_dptools wiki.

http://wiki.freeswitch.org/wiki/Mod_dptools

mod_dptools is the module that contains the dialplan apps.

Regards,

Diego

On Sat, Oct 17, 2009 at 12:29 AM, Michael Gende <mgende at gendesign.com>wrote:

> Hey Diego,
>
> You seem to know your way around the documentation and clearly have many
> contributions. Thus, a question:
>
> Can you point me to doc on dial plan applications (such as transfer,
> bridge, etc)? It would help me greatly. Sorry if this is in an obvious place
> and I've just not seen it despite the nose on my face.
>
> Regards,
>
> Mike G.
>
>
> On Fri, Oct 16, 2009 at 5:10 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> Can someone please convert this IRC log to documentation form:
>>
>> http://wiki.freeswitch.org/wiki/Cacti
>>
>> Try to explain what the IRC log explains in documentation form, and get
>> rid of the IRC log.
>>
>> I'm working on fixing typos and stuff like that on the wiki now, then I
>> will focus on fixing the format, so all pages look the same (titles, choice
>> of case in text, etc.)
>>
>> Thanks,
>>
>> Diego
>>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ruda at ruda.com.br  Fri Oct 16 18:22:19 2009
From: ruda at ruda.com.br (=?ISO-8859-1?Q?Rud=E1_Cunha?=)
Date: Fri, 16 Oct 2009 22:22:19 -0300
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <d3ceea380910161328w4c843e9as49b2341dc6f8eb7d@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<d3ceea380910160702x8acbc4ci3bf3cfc1fdf9a9b7@mail.gmail.com> 
	<ebbc2d550910160725x3141ce14re1821926d4027d42@mail.gmail.com> 
	<2E845C1E-C084-4025-806D-178A175FF90A@gmail.com>
	<f3cdc69d0910161302v6a97ead2xc2af5d73b5a048d7@mail.gmail.com> 
	<d3ceea380910161328w4c843e9as49b2341dc6f8eb7d@mail.gmail.com>
Message-ID: <7600794b0910161822j37848bbfk8b0eaf35eeeae189@mail.gmail.com>

Hello,

I wanted to create a forum so we can discuss FreeSWITCH.

That person who registered the domain, I can create and host the forum, so
we can talk there and ask questions


2009/10/16 Itamar Reis Peixoto <itamar at ispbrasil.com.br>

> instead of creating thousand of small sites, why don't put the content
> in only place.
>
> freeswitch.org already exist, no need for a new domain.
>
>
>
> On Fri, Oct 16, 2009 at 5:02 PM, Fabio Ferrari <fabioferrari at gmail.com>
> wrote:
> > Just to let you know, Jeremias Silva from Khomp registered the
> > freeswitchbrasil.org domain. I think there is space to build an wiki
> > with portuguese content there.
> >
> > Fabio Ferrari
>
>
> ------------
>
> Itamar Reis Peixoto
>
> e-mail/msn: itamar at ispbrasil.com.br
> sip: itamar at ispbrasil.com.br
> skype: itamarjp
> icq: 81053601
> +55 11 4063 5033
> +55 34 3221 8599
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From diego.viola at gmail.com  Fri Oct 16 18:47:56 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 17 Oct 2009 01:47:56 +0000
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <86a32abc0910161811h15eac05dwc3fe0c2533b7397a@mail.gmail.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
	<891080.87141.qm@web56406.mail.re3.yahoo.com>
	<87f2f3b90910130952r5657c8e2g6e11c22140da919c@mail.gmail.com>
	<6b65470d0910131009y1cf136c3q8c0932375c19f272@mail.gmail.com>
	<86a32abc0910161510ma323718g410bcf1029156077@mail.gmail.com>
	<a87d43fd0910161729u42db5ea7qafb5604c0680954@mail.gmail.com>
	<86a32abc0910161811h15eac05dwc3fe0c2533b7397a@mail.gmail.com>
Message-ID: <86a32abc0910161847k415b9722ubc5439ea1b9f6e78@mail.gmail.com>

My plan to improve the FreeSWITCH wiki


   1. Correct all typos from all pages.
   2. Delete unused/irrelevant pages.
   3. Make sure we don't repeat stuff (multi-tenant/multi-companies,
   features/spec-sheet, etc)
   4. Define a size for text, titles/subtitles (so all pages look the same).
   5. Define a case for text, titles/subtitles (so all pages look the same).

More to come...



On Sat, Oct 17, 2009 at 1:11 AM, Diego Viola <diego.viola at gmail.com> wrote:

> Hi Michael,
>
> You might want to look at the mod_dptools wiki.
>
> http://wiki.freeswitch.org/wiki/Mod_dptools
>
> mod_dptools is the module that contains the dialplan apps.
>
> Regards,
>
> Diego
>
> On Sat, Oct 17, 2009 at 12:29 AM, Michael Gende <mgende at gendesign.com>wrote:
>
>> Hey Diego,
>>
>> You seem to know your way around the documentation and clearly have many
>> contributions. Thus, a question:
>>
>> Can you point me to doc on dial plan applications (such as transfer,
>> bridge, etc)? It would help me greatly. Sorry if this is in an obvious place
>> and I've just not seen it despite the nose on my face.
>>
>> Regards,
>>
>> Mike G.
>>
>>
>> On Fri, Oct 16, 2009 at 5:10 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>
>>> Can someone please convert this IRC log to documentation form:
>>>
>>> http://wiki.freeswitch.org/wiki/Cacti
>>>
>>> Try to explain what the IRC log explains in documentation form, and get
>>> rid of the IRC log.
>>>
>>> I'm working on fixing typos and stuff like that on the wiki now, then I
>>> will focus on fixing the format, so all pages look the same (titles, choice
>>> of case in text, etc.)
>>>
>>> Thanks,
>>>
>>> Diego
>>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From jason at jasonjgw.net  Fri Oct 16 18:54:27 2009
From: jason at jasonjgw.net (Jason White)
Date: Sat, 17 Oct 2009 12:54:27 +1100
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <7600794b0910161822j37848bbfk8b0eaf35eeeae189@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<d3ceea380910160702x8acbc4ci3bf3cfc1fdf9a9b7@mail.gmail.com>
	<ebbc2d550910160725x3141ce14re1821926d4027d42@mail.gmail.com>
	<2E845C1E-C084-4025-806D-178A175FF90A@gmail.com>
	<f3cdc69d0910161302v6a97ead2xc2af5d73b5a048d7@mail.gmail.com>
	<d3ceea380910161328w4c843e9as49b2341dc6f8eb7d@mail.gmail.com>
	<7600794b0910161822j37848bbfk8b0eaf35eeeae189@mail.gmail.com>
Message-ID: <20091017015427.GA7044@jdc.jasonjgw.net>

Rud? Cunha <ruda at ruda.com.br> wrote:
> Hello,
> 
> I wanted to create a forum so we can discuss FreeSWITCH.

You're writing to one now. It can be accessed as a mailing list, on the Web at
gmane.org, and via NNTP at gmane.org. I think there's a Web archive somewhere
else, too.




From diego.viola at gmail.com  Fri Oct 16 18:54:16 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 17 Oct 2009 01:54:16 +0000
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <86a32abc0910161847k415b9722ubc5439ea1b9f6e78@mail.gmail.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
	<891080.87141.qm@web56406.mail.re3.yahoo.com>
	<87f2f3b90910130952r5657c8e2g6e11c22140da919c@mail.gmail.com>
	<6b65470d0910131009y1cf136c3q8c0932375c19f272@mail.gmail.com>
	<86a32abc0910161510ma323718g410bcf1029156077@mail.gmail.com>
	<a87d43fd0910161729u42db5ea7qafb5604c0680954@mail.gmail.com>
	<86a32abc0910161811h15eac05dwc3fe0c2533b7397a@mail.gmail.com>
	<86a32abc0910161847k415b9722ubc5439ea1b9f6e78@mail.gmail.com>
Message-ID: <86a32abc0910161854k7fd3fca6pc0686e928c772d2c@mail.gmail.com>

s/so all pages look the same/so all the pages have the same look and feel/

On Sat, Oct 17, 2009 at 1:47 AM, Diego Viola <diego.viola at gmail.com> wrote:

> My plan to improve the FreeSWITCH wiki
>
>
>    1. Correct all typos from all pages.
>    2. Delete unused/irrelevant pages.
>    3. Make sure we don't repeat stuff (multi-tenant/multi-companies,
>    features/spec-sheet, etc)
>    4. Define a size for text, titles/subtitles (so all pages look the
>    same).
>    5. Define a case for text, titles/subtitles (so all pages look the
>    same).
>
> More to come...
>
>
>
>
> On Sat, Oct 17, 2009 at 1:11 AM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> Hi Michael,
>>
>> You might want to look at the mod_dptools wiki.
>>
>> http://wiki.freeswitch.org/wiki/Mod_dptools
>>
>> mod_dptools is the module that contains the dialplan apps.
>>
>> Regards,
>>
>> Diego
>>
>> On Sat, Oct 17, 2009 at 12:29 AM, Michael Gende <mgende at gendesign.com>wrote:
>>
>>> Hey Diego,
>>>
>>> You seem to know your way around the documentation and clearly have many
>>> contributions. Thus, a question:
>>>
>>> Can you point me to doc on dial plan applications (such as transfer,
>>> bridge, etc)? It would help me greatly. Sorry if this is in an obvious place
>>> and I've just not seen it despite the nose on my face.
>>>
>>> Regards,
>>>
>>> Mike G.
>>>
>>>
>>> On Fri, Oct 16, 2009 at 5:10 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>
>>>> Can someone please convert this IRC log to documentation form:
>>>>
>>>> http://wiki.freeswitch.org/wiki/Cacti
>>>>
>>>> Try to explain what the IRC log explains in documentation form, and get
>>>> rid of the IRC log.
>>>>
>>>> I'm working on fixing typos and stuff like that on the wiki now, then I
>>>> will focus on fixing the format, so all pages look the same (titles, choice
>>>> of case in text, etc.)
>>>>
>>>> Thanks,
>>>>
>>>> Diego
>>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
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From jason at jasonjgw.net  Fri Oct 16 19:53:23 2009
From: jason at jasonjgw.net (Jason White)
Date: Sat, 17 Oct 2009 13:53:23 +1100
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <d3ceea380910161328w4c843e9as49b2341dc6f8eb7d@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<d3ceea380910160702x8acbc4ci3bf3cfc1fdf9a9b7@mail.gmail.com>
	<ebbc2d550910160725x3141ce14re1821926d4027d42@mail.gmail.com>
	<2E845C1E-C084-4025-806D-178A175FF90A@gmail.com>
	<f3cdc69d0910161302v6a97ead2xc2af5d73b5a048d7@mail.gmail.com>
	<d3ceea380910161328w4c843e9as49b2341dc6f8eb7d@mail.gmail.com>
Message-ID: <20091017025323.GA15649@jdc.jasonjgw.net>

Itamar Reis Peixoto <itamar at ispbrasil.com.br> wrote:
> instead of creating thousand of small sites, why don't put the content
> in only place.
> 
> freeswitch.org already exist, no need for a new domain.

This mailing list, the development mailing list, the IRC channel and the
FreeSWITCH conference all exist - no need for another "forum".




From moises.silva at gmail.com  Fri Oct 16 20:05:12 2009
From: moises.silva at gmail.com (Moises Silva)
Date: Fri, 16 Oct 2009 23:05:12 -0400
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <20091017025323.GA15649@jdc.jasonjgw.net>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<d3ceea380910160702x8acbc4ci3bf3cfc1fdf9a9b7@mail.gmail.com>
	<ebbc2d550910160725x3141ce14re1821926d4027d42@mail.gmail.com>
	<2E845C1E-C084-4025-806D-178A175FF90A@gmail.com>
	<f3cdc69d0910161302v6a97ead2xc2af5d73b5a048d7@mail.gmail.com>
	<d3ceea380910161328w4c843e9as49b2341dc6f8eb7d@mail.gmail.com>
	<20091017025323.GA15649@jdc.jasonjgw.net>
Message-ID: <c4d05cbe0910162005r3f094fc7ne448b905446282f4@mail.gmail.com>

>
> This mailing list, the development mailing list, the IRC channel and the
> FreeSWITCH conference all exist - no need for another "forum".
>
>
I think he meant a forum in Portuguese. May be it's hard to believe for some
people, but not everybody speaks English. Programmers tend to do it by
default, but there are a lot of Asterisk users in  Brazil or
spanish-speaking countries that post daily to spanish and portuguese forums.
I don't see a problem with creating a forum in portuguese if it helps to
bring adoption of FreeSWTICH.

I understand we want to avoid duplication of efforts, but the language
barrier is there and is not going to change.

-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From chris at fowler.cc  Fri Oct 16 20:06:52 2009
From: chris at fowler.cc (Chris Fowler)
Date: Fri, 16 Oct 2009 20:06:52 -0700
Subject: [Freeswitch-users] NAT problems migrating from Version 1.0.trunk
 (13168M) to Version 1.0.trunk (15166)
Message-ID: <1255748812.24395.1340540355@webmail.messagingengine.com>

Hi,

We've been using 13168M in production for some time now (works great). 
I want to get us onto the latest build but am having problems getting
NAT to work.

Phones can register; can dial 9999 test #, but after 100 seconds the
call is disconnected with error:
2009-10-16 19:52:26.936618 [NOTICE] sofia.c:4038 Hangup
sofia/internal/1001 at myhost.mydomain.com [CS_EXECUTE]
[RECOVERY_ON_TIMER_EXPIRE]

I took the standard internal.xml and vars.xml files from the new build
and made the following modifications - which worked previously:

modify conf/vars.xml and update
  <X-PRE-PROCESS cmd="set" data="domain=myhost.mydomain.com"/>

  <X-PRE-PROCESS cmd="set" data="bind_server_ip=1.2.3.4"/>
  <X-PRE-PROCESS cmd="set" data="external_rtp_ip=1.2.3.4"/>
  <X-PRE-PROCESS cmd="set" data="external_sip_ip=1.2.3.4"/>

Modify conf/sip_profiles/internal.xml
 <param name="aggressive-nat-detection" value="true"/>  <param
 name="ext-rtp-ip" value="$${external_rtp_ip}"/>  <param
 name="ext-sip-ip" value="$${external_sip_ip}"/>  <param
 name="NDLB-received-in-nat-reg-contact" value="true"/>  <param
 name="NDLB-force-rport" value="true"/>  <param
 name="NDLB-broken-auth-hash" value="true"/>

The big difference I note is that on PRODUCTION (which works) "sofia
status profile internal" yields:
URL                     sip:mod_sofia at 1.2.3.4:5060
BIND-URL                sip:mod_sofia at 1.2.3.4:5060;maddr=10.250.35.224

But on Test I see:
URL                     sip:mod_sofia at 10.250.66.210:5060
BIND-URL                sip:mod_sofia at 10.250.66.210:5060

Any ideas?

Thanks, Chris.




From diego.viola at gmail.com  Fri Oct 16 20:19:37 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 17 Oct 2009 03:19:37 +0000
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <c4d05cbe0910162005r3f094fc7ne448b905446282f4@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<d3ceea380910160702x8acbc4ci3bf3cfc1fdf9a9b7@mail.gmail.com>
	<ebbc2d550910160725x3141ce14re1821926d4027d42@mail.gmail.com>
	<2E845C1E-C084-4025-806D-178A175FF90A@gmail.com>
	<f3cdc69d0910161302v6a97ead2xc2af5d73b5a048d7@mail.gmail.com>
	<d3ceea380910161328w4c843e9as49b2341dc6f8eb7d@mail.gmail.com>
	<20091017025323.GA15649@jdc.jasonjgw.net>
	<c4d05cbe0910162005r3f094fc7ne448b905446282f4@mail.gmail.com>
Message-ID: <86a32abc0910162019m6315692bw566e282aaf8e2900@mail.gmail.com>

I'm with Moises and with the other people supporting this initiative.

I'm not Brazilian, but they should be able to do whatever they want, after
all, that's how open source works, if you can do it go ahead and do it.

Diego

On Sat, Oct 17, 2009 at 3:05 AM, Moises Silva <moises.silva at gmail.com>wrote:

> This mailing list, the development mailing list, the IRC channel and the
>> FreeSWITCH conference all exist - no need for another "forum".
>>
>>
> I think he meant a forum in Portuguese. May be it's hard to believe for
> some people, but not everybody speaks English. Programmers tend to do it by
> default, but there are a lot of Asterisk users in  Brazil or
> spanish-speaking countries that post daily to spanish and portuguese forums.
> I don't see a problem with creating a forum in portuguese if it helps to
> bring adoption of FreeSWTICH.
>
> I understand we want to avoid duplication of efforts, but the language
> barrier is there and is not going to change.
>
> --
> Moises Silva
> Software Developer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From diego.viola at gmail.com  Fri Oct 16 20:50:12 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 17 Oct 2009 03:50:12 +0000
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <86a32abc0910161854k7fd3fca6pc0686e928c772d2c@mail.gmail.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
	<891080.87141.qm@web56406.mail.re3.yahoo.com>
	<87f2f3b90910130952r5657c8e2g6e11c22140da919c@mail.gmail.com>
	<6b65470d0910131009y1cf136c3q8c0932375c19f272@mail.gmail.com>
	<86a32abc0910161510ma323718g410bcf1029156077@mail.gmail.com>
	<a87d43fd0910161729u42db5ea7qafb5604c0680954@mail.gmail.com>
	<86a32abc0910161811h15eac05dwc3fe0c2533b7397a@mail.gmail.com>
	<86a32abc0910161847k415b9722ubc5439ea1b9f6e78@mail.gmail.com>
	<86a32abc0910161854k7fd3fca6pc0686e928c772d2c@mail.gmail.com>
Message-ID: <86a32abc0910162050m6bad9a9do171b268e083b3d0d@mail.gmail.com>

This needs to be converted to documentation form too.

http://wiki.freeswitch.org/wiki/Mod_vmd#Related_conversations

Diego

On Sat, Oct 17, 2009 at 1:54 AM, Diego Viola <diego.viola at gmail.com> wrote:

> s/so all pages look the same/so all the pages have the same look and feel/
>
>
> On Sat, Oct 17, 2009 at 1:47 AM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> My plan to improve the FreeSWITCH wiki
>>
>>
>>    1. Correct all typos from all pages.
>>    2. Delete unused/irrelevant pages.
>>    3. Make sure we don't repeat stuff (multi-tenant/multi-companies,
>>    features/spec-sheet, etc)
>>    4. Define a size for text, titles/subtitles (so all pages look the
>>    same).
>>    5. Define a case for text, titles/subtitles (so all pages look the
>>    same).
>>
>> More to come...
>>
>>
>>
>>
>> On Sat, Oct 17, 2009 at 1:11 AM, Diego Viola <diego.viola at gmail.com>wrote:
>>
>>> Hi Michael,
>>>
>>> You might want to look at the mod_dptools wiki.
>>>
>>> http://wiki.freeswitch.org/wiki/Mod_dptools
>>>
>>> mod_dptools is the module that contains the dialplan apps.
>>>
>>> Regards,
>>>
>>> Diego
>>>
>>> On Sat, Oct 17, 2009 at 12:29 AM, Michael Gende <mgende at gendesign.com>wrote:
>>>
>>>> Hey Diego,
>>>>
>>>> You seem to know your way around the documentation and clearly have many
>>>> contributions. Thus, a question:
>>>>
>>>> Can you point me to doc on dial plan applications (such as transfer,
>>>> bridge, etc)? It would help me greatly. Sorry if this is in an obvious place
>>>> and I've just not seen it despite the nose on my face.
>>>>
>>>> Regards,
>>>>
>>>> Mike G.
>>>>
>>>>
>>>> On Fri, Oct 16, 2009 at 5:10 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>>
>>>>> Can someone please convert this IRC log to documentation form:
>>>>>
>>>>> http://wiki.freeswitch.org/wiki/Cacti
>>>>>
>>>>> Try to explain what the IRC log explains in documentation form, and get
>>>>> rid of the IRC log.
>>>>>
>>>>> I'm working on fixing typos and stuff like that on the wiki now, then I
>>>>> will focus on fixing the format, so all pages look the same (titles, choice
>>>>> of case in text, etc.)
>>>>>
>>>>> Thanks,
>>>>>
>>>>> Diego
>>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>
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From jason at jasonjgw.net  Fri Oct 16 20:52:15 2009
From: jason at jasonjgw.net (Jason White)
Date: Sat, 17 Oct 2009 14:52:15 +1100
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <86a32abc0910162019m6315692bw566e282aaf8e2900@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<d3ceea380910160702x8acbc4ci3bf3cfc1fdf9a9b7@mail.gmail.com>
	<ebbc2d550910160725x3141ce14re1821926d4027d42@mail.gmail.com>
	<2E845C1E-C084-4025-806D-178A175FF90A@gmail.com>
	<f3cdc69d0910161302v6a97ead2xc2af5d73b5a048d7@mail.gmail.com>
	<d3ceea380910161328w4c843e9as49b2341dc6f8eb7d@mail.gmail.com>
	<20091017025323.GA15649@jdc.jasonjgw.net>
	<c4d05cbe0910162005r3f094fc7ne448b905446282f4@mail.gmail.com>
	<86a32abc0910162019m6315692bw566e282aaf8e2900@mail.gmail.com>
Message-ID: <20091017035215.GA24308@jdc.jasonjgw.net>

Diego Viola <diego.viola at gmail.com> wrote:
> I'm with Moises and with the other people supporting this initiative.
> 
> I'm not Brazilian, but they should be able to do whatever they want, after
> all, that's how open source works, if you can do it go ahead and do it.

Correct. We have enough of them as far as English-language fora are concerned;
other languages are a different question altogether, though.




From jmesquita at freeswitch.org  Fri Oct 16 21:23:40 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 17 Oct 2009 01:23:40 -0300
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <20091017035215.GA24308@jdc.jasonjgw.net>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<d3ceea380910160702x8acbc4ci3bf3cfc1fdf9a9b7@mail.gmail.com>
	<ebbc2d550910160725x3141ce14re1821926d4027d42@mail.gmail.com>
	<2E845C1E-C084-4025-806D-178A175FF90A@gmail.com>
	<f3cdc69d0910161302v6a97ead2xc2af5d73b5a048d7@mail.gmail.com>
	<d3ceea380910161328w4c843e9as49b2341dc6f8eb7d@mail.gmail.com>
	<20091017025323.GA15649@jdc.jasonjgw.net>
	<c4d05cbe0910162005r3f094fc7ne448b905446282f4@mail.gmail.com>
	<86a32abc0910162019m6315692bw566e282aaf8e2900@mail.gmail.com>
	<20091017035215.GA24308@jdc.jasonjgw.net>
Message-ID: <ebbc2d550910162123p33a80f0fnd8ebac5fc569588d@mail.gmail.com>

Just a heads up, I have talked to Jeremias from Khomp today and he is
setting up the wiki. I will personally be adding contents to the that wiki
if it ever picks up.

Regards,

jm

On Sat, Oct 17, 2009 at 12:52 AM, Jason White <jason at jasonjgw.net> wrote:

> Diego Viola <diego.viola at gmail.com> wrote:
> > I'm with Moises and with the other people supporting this initiative.
> >
> > I'm not Brazilian, but they should be able to do whatever they want,
> after
> > all, that's how open source works, if you can do it go ahead and do it.
>
> Correct. We have enough of them as far as English-language fora are
> concerned;
> other languages are a different question altogether, though.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mgende at gendesign.com  Fri Oct 16 22:23:22 2009
From: mgende at gendesign.com (Michael Gende)
Date: Sat, 17 Oct 2009 00:23:22 -0500
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <86a32abc0910161811h15eac05dwc3fe0c2533b7397a@mail.gmail.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
	<891080.87141.qm@web56406.mail.re3.yahoo.com>
	<87f2f3b90910130952r5657c8e2g6e11c22140da919c@mail.gmail.com>
	<6b65470d0910131009y1cf136c3q8c0932375c19f272@mail.gmail.com>
	<86a32abc0910161510ma323718g410bcf1029156077@mail.gmail.com>
	<a87d43fd0910161729u42db5ea7qafb5604c0680954@mail.gmail.com>
	<86a32abc0910161811h15eac05dwc3fe0c2533b7397a@mail.gmail.com>
Message-ID: <a87d43fd0910162223l45e2adb5xf529f8d56107c786@mail.gmail.com>

That's exactly what I needed. Many thanks.

On Fri, Oct 16, 2009 at 8:11 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Hi Michael,
>
> You might want to look at the mod_dptools wiki.
>
> http://wiki.freeswitch.org/wiki/Mod_dptools
>
> mod_dptools is the module that contains the dialplan apps.
>
> Regards,
>
> Diego
>
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From msc at freeswitch.org  Fri Oct 16 22:54:15 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 16 Oct 2009 22:54:15 -0700
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <86a32abc0910162050m6bad9a9do171b268e083b3d0d@mail.gmail.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
	<891080.87141.qm@web56406.mail.re3.yahoo.com>
	<87f2f3b90910130952r5657c8e2g6e11c22140da919c@mail.gmail.com>
	<6b65470d0910131009y1cf136c3q8c0932375c19f272@mail.gmail.com>
	<86a32abc0910161510ma323718g410bcf1029156077@mail.gmail.com>
	<a87d43fd0910161729u42db5ea7qafb5604c0680954@mail.gmail.com>
	<86a32abc0910161811h15eac05dwc3fe0c2533b7397a@mail.gmail.com>
	<86a32abc0910161847k415b9722ubc5439ea1b9f6e78@mail.gmail.com>
	<86a32abc0910161854k7fd3fca6pc0686e928c772d2c@mail.gmail.com>
	<86a32abc0910162050m6bad9a9do171b268e083b3d0d@mail.gmail.com>
Message-ID: <87f2f3b90910162254q23893e4cx160fe4a5ae64cc91@mail.gmail.com>

Diego, when I get some time I will help convert these from IRC to wiki.
Thanks,
MC

On Fri, Oct 16, 2009 at 8:50 PM, Diego Viola <diego.viola at gmail.com> wrote:

> This needs to be converted to documentation form too.
>
> http://wiki.freeswitch.org/wiki/Mod_vmd#Related_conversations
>
> Diego
>
>
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From msc at freeswitch.org  Fri Oct 16 23:00:03 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 16 Oct 2009 23:00:03 -0700
Subject: [Freeswitch-users] Sending an Event to a Session for onInput
In-Reply-To: <0BDCD23D-D551-4A2F-9CD7-CC5A2F1382CB@jerris.com>
References: <4256bf830910091234t11f4f4d0w28283f341bd1fb06@mail.gmail.com>
	<ACA9A602-0925-4D6F-8B11-0B214A462500@jerris.com>
	<4256bf830910112311t409c45a3q2f26cef3464e49e7@mail.gmail.com>
	<0BDCD23D-D551-4A2F-9CD7-CC5A2F1382CB@jerris.com>
Message-ID: <87f2f3b90910162300x30701935k9f35162fb88838d9@mail.gmail.com>

Thanks for the explanation. I added this to the wiki.
-MC

On Thu, Oct 15, 2009 at 6:50 PM, Michael Jerris <mike at jerris.com> wrote:

> It updates the display on a phone if the phone supports this.  This works
> on some sip phones right now including polycom and snom.
> Mike
>
> On Oct 12, 2009, at 2:11 AM, Matthew Fong wrote:
>
>
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From msc at freeswitch.org  Fri Oct 16 23:06:51 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 16 Oct 2009 23:06:51 -0700
Subject: [Freeswitch-users] Qustion about INFO messages after
	Connect/Answer
In-Reply-To: <191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>
References: <4AD859A4.5000101@ewetel.de>
	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>
Message-ID: <87f2f3b90910162306s14356f1bmbc484761b99bcc42@mail.gmail.com>

On Fri, Oct 16, 2009 at 9:18 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> 1) you should update again there were a few issues.
> 2) you can set the variable sip_callee_id_name and sip_callee_id number on
> the inbound leg before you answer to control what it says.
>
>
> Thanks for the heads up. I added these two vars to the wiki on the chan
vars page, SIP-related vars section.
-MC
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From diego.viola at gmail.com  Fri Oct 16 23:12:38 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 17 Oct 2009 06:12:38 +0000
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <87f2f3b90910162254q23893e4cx160fe4a5ae64cc91@mail.gmail.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
	<87f2f3b90910130952r5657c8e2g6e11c22140da919c@mail.gmail.com>
	<6b65470d0910131009y1cf136c3q8c0932375c19f272@mail.gmail.com>
	<86a32abc0910161510ma323718g410bcf1029156077@mail.gmail.com>
	<a87d43fd0910161729u42db5ea7qafb5604c0680954@mail.gmail.com>
	<86a32abc0910161811h15eac05dwc3fe0c2533b7397a@mail.gmail.com>
	<86a32abc0910161847k415b9722ubc5439ea1b9f6e78@mail.gmail.com>
	<86a32abc0910161854k7fd3fca6pc0686e928c772d2c@mail.gmail.com>
	<86a32abc0910162050m6bad9a9do171b268e083b3d0d@mail.gmail.com>
	<87f2f3b90910162254q23893e4cx160fe4a5ae64cc91@mail.gmail.com>
Message-ID: <86a32abc0910162312le6c08b8jb53ae83bcafadab1@mail.gmail.com>

Thanks, our wiki is getting a lot better :).

Diego

On Sat, Oct 17, 2009 at 5:54 AM, Michael Collins <msc at freeswitch.org> wrote:

> Diego, when I get some time I will help convert these from IRC to wiki.
> Thanks,
> MC
>
>
> On Fri, Oct 16, 2009 at 8:50 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> This needs to be converted to documentation form too.
>>
>> http://wiki.freeswitch.org/wiki/Mod_vmd#Related_conversations
>>
>> Diego
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sjmudd at pobox.com  Fri Oct 16 23:44:14 2009
From: sjmudd at pobox.com (Simon J Mudd)
Date: 17 Oct 2009 08:44:14 +0200
Subject: [Freeswitch-users] macros for performing repeated dialplan
	actions?
In-Reply-To: <1C80FD78-71B7-4D62-8C45-2F5BC8C1564C@apartmentlines.com>
References: <20091014220135.GA12614@mad06.wl0.org>
	<87f2f3b90910141510n45401995y11e3e380653e9790@mail.gmail.com>
	<m3eip41yxv.fsf@mad06.wl0.org>
	<1C80FD78-71B7-4D62-8C45-2F5BC8C1564C@apartmentlines.com>
Message-ID: <m3aazq1pvl.fsf@mad06.wl0.org>

chad at apartmentlines.com (Chad Phillips -- Apartment Lines) writes:

> On Oct 15, 2009, at 11:03 AM, Simon J Mudd wrote:
> 
> > OK. That looks clear. However it doesn't look as if it's possible to
> > pass parameters this way which means that data can only be passed to
> > the "extension" via variables. That works but isn't so clean. I'll
> > check this out further shortly. I'm planning later to pass info like
> > "which gateway to talk to" and perhaps some sort of flag which
> > indicates "record: yes/no", and even include the bridge application
> > into this extension/macro.  That reduces the original dialplan pattern
> > matching to a much simpler and easier to configure setup for me.
> 
> here's a tip from a recovering asterisk user: don't feel like you have  
> to do everything via the dialplan.  this was a constant mantra in  
> asterisk, because calling AGI scripts was expensive.  this is not the  
> case with FreeSWITCH.  the general rule of thumb is: as soon as it  
> gets hard to do in the dialplan, farm it out to a script.  it's the  
> way FreeSWITCH is designed, so don't fight it  ;)

OK. That makes a lot of sense. Perhaps I'm just missing examples to copy
which do some of the things I want to do.

Since there are indications on this list that FreeSWITCH can do many
things the changeover from Asterisk to FreeSWITCH is quite difficult
at least once you start to try and do more complex things.

I've other plans like:

1. choose the cheapest gateway based on some scripts which download
rates from the various gateways web pages and let me compare rates.
Hard partly as the "charges pages" aren't designed to allow me to
match a number to a charge. Shame the VoIP community can't come up
with a common format "charge/rate file" to make this easier.

2. Try one gateway and if that fails try another one. Many of the
gateways I've been using are low cost and don't always work, so to
avoid me having to adjust things each time this might be useful.

3. Remove a lot of the "intelligence" about what to do to a backend
script/program or whatever which decides how to route calls and tells
FreeSWITCH what it decides. This may not scale for a large setup but
for SOHO is never going to be an issue. It also avoids me writing so
many xml configuration files which seems to get spread all over
various places of the conf/ tree.

However I've still got a lot of experimenting to do yet.

Simon



From rupa at rupa.com  Sat Oct 17 00:18:17 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 17 Oct 2009 02:18:17 -0500
Subject: [Freeswitch-users] macros for performing repeated dialplan
	actions?
In-Reply-To: <m3aazq1pvl.fsf@mad06.wl0.org>
References: <20091014220135.GA12614@mad06.wl0.org>
	<87f2f3b90910141510n45401995y11e3e380653e9790@mail.gmail.com>
	<m3eip41yxv.fsf@mad06.wl0.org>
	<1C80FD78-71B7-4D62-8C45-2F5BC8C1564C@apartmentlines.com>
	<m3aazq1pvl.fsf@mad06.wl0.org>
Message-ID: <b39273530910170018h5063c309xa96ab59b15256055@mail.gmail.com>

On Sat, Oct 17, 2009 at 1:44 AM, Simon J Mudd <sjmudd at pobox.com> wrote:
> I've other plans like:
>
> 1. choose the cheapest gateway based on some scripts which download
> rates from the various gateways web pages and let me compare rates.
> Hard partly as the "charges pages" aren't designed to allow me to
> match a number to a charge. Shame the VoIP community can't come up
> with a common format "charge/rate file" to make this easier.

Look at mod_lcr.  The only part you have to write is the part that
loads the database from your providers.

> 2. Try one gateway and if that fails try another one. Many of the
> gateways I've been using are low cost and don't always work, so to
> avoid me having to adjust things each time this might be useful.

bridge to ${lcr_auto_route} after calling the lcr application.

> 3. Remove a lot of the "intelligence" about what to do to a backend
> script/program or whatever which decides how to route calls and tells
> FreeSWITCH what it decides. This may not scale for a large setup but
> for SOHO is never going to be an issue. It also avoids me writing so
> many xml configuration files which seems to get spread all over
> various places of the conf/ tree.

This is for incoming calls to your DIDs?  Or something else?  I'm not
sure what you mean here.

> However I've still got a lot of experimenting to do yet.
>
> Simon



-- 
-Rupa



From ruda at ruda.com.br  Sat Oct 17 04:06:26 2009
From: ruda at ruda.com.br (=?ISO-8859-1?Q?Rud=E1_Cunha?=)
Date: Sat, 17 Oct 2009 08:06:26 -0300
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <ebbc2d550910162123p33a80f0fnd8ebac5fc569588d@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<ebbc2d550910160725x3141ce14re1821926d4027d42@mail.gmail.com> 
	<2E845C1E-C084-4025-806D-178A175FF90A@gmail.com>
	<f3cdc69d0910161302v6a97ead2xc2af5d73b5a048d7@mail.gmail.com> 
	<d3ceea380910161328w4c843e9as49b2341dc6f8eb7d@mail.gmail.com> 
	<20091017025323.GA15649@jdc.jasonjgw.net>
	<c4d05cbe0910162005r3f094fc7ne448b905446282f4@mail.gmail.com> 
	<86a32abc0910162019m6315692bw566e282aaf8e2900@mail.gmail.com> 
	<20091017035215.GA24308@jdc.jasonjgw.net>
	<ebbc2d550910162123p33a80f0fnd8ebac5fc569588d@mail.gmail.com>
Message-ID: <7600794b0910170406m164e3e2od198586f53de5efc@mail.gmail.com>

Jo?o Mesquita,

Could check with Jeremiah or contact me through, so I talk to him about the
forum?

He could redirect the domain to my hosting server and we can put there the
wiki and forums and we can communicate!

Rud? Cunha

2009/10/17 Jo?o Mesquita <jmesquita at freeswitch.org>

> Just a heads up, I have talked to Jeremias from Khomp today and he is
> setting up the wiki. I will personally be adding contents to the that wiki
> if it ever picks up.
>
> Regards,
>
> jm
>
>
> On Sat, Oct 17, 2009 at 12:52 AM, Jason White <jason at jasonjgw.net> wrote:
>
>> Diego Viola <diego.viola at gmail.com> wrote:
>> > I'm with Moises and with the other people supporting this initiative.
>> >
>> > I'm not Brazilian, but they should be able to do whatever they want,
>> after
>> > all, that's how open source works, if you can do it go ahead and do it.
>>
>> Correct. We have enough of them as far as English-language fora are
>> concerned;
>> other languages are a different question altogether, though.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ruda at ruda.com.br  Sat Oct 17 07:38:58 2009
From: ruda at ruda.com.br (=?ISO-8859-1?Q?Rud=E1_Cunha?=)
Date: Sat, 17 Oct 2009 11:38:58 -0300
Subject: [Freeswitch-users] SPA 3000 - Disconnect Tone
Message-ID: <7600794b0910170738l612e3163i567c5522313109ad@mail.gmail.com>

I am using SPA 3000 (FXO) except that the Disconnect Tone is not working for
an operator (GVT) now works in another carrier (OI). I'm from Brazil.

Anybody know how to find the Disconnect Tone Operator?

Att,
Rud? Cunha
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From anthony.minessale at gmail.com  Sat Oct 17 09:25:08 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 17 Oct 2009 11:25:08 -0500
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <7600794b0910170406m164e3e2od198586f53de5efc@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<2E845C1E-C084-4025-806D-178A175FF90A@gmail.com>
	<f3cdc69d0910161302v6a97ead2xc2af5d73b5a048d7@mail.gmail.com>
	<d3ceea380910161328w4c843e9as49b2341dc6f8eb7d@mail.gmail.com>
	<20091017025323.GA15649@jdc.jasonjgw.net>
	<c4d05cbe0910162005r3f094fc7ne448b905446282f4@mail.gmail.com>
	<86a32abc0910162019m6315692bw566e282aaf8e2900@mail.gmail.com>
	<20091017035215.GA24308@jdc.jasonjgw.net>
	<ebbc2d550910162123p33a80f0fnd8ebac5fc569588d@mail.gmail.com>
	<7600794b0910170406m164e3e2od198586f53de5efc@mail.gmail.com>
Message-ID: <191c3a030910170925i7aa7d4f3p9d382e0d855b7c8b@mail.gmail.com>

or someone could just ask us for pg.freeswitch.org or whatever other
sub-domain to avoid trademark infringement of registering 40 domains with
freeswitch in the name for no reason.


On Sat, Oct 17, 2009 at 6:06 AM, Rud? Cunha <ruda at ruda.com.br> wrote:

> Jo?o Mesquita,
>
> Could check with Jeremiah or contact me through, so I talk to him about the
> forum?
>
> He could redirect the domain to my hosting server and we can put there the
> wiki and forums and we can communicate!
>
> Rud? Cunha
>
> 2009/10/17 Jo?o Mesquita <jmesquita at freeswitch.org>
>
> Just a heads up, I have talked to Jeremias from Khomp today and he is
>> setting up the wiki. I will personally be adding contents to the that wiki
>> if it ever picks up.
>>
>> Regards,
>>
>> jm
>>
>>
>> On Sat, Oct 17, 2009 at 12:52 AM, Jason White <jason at jasonjgw.net> wrote:
>>
>>> Diego Viola <diego.viola at gmail.com> wrote:
>>> > I'm with Moises and with the other people supporting this initiative.
>>> >
>>> > I'm not Brazilian, but they should be able to do whatever they want,
>>> after
>>> > all, that's how open source works, if you can do it go ahead and do it.
>>>
>>> Correct. We have enough of them as far as English-language fora are
>>> concerned;
>>> other languages are a different question altogether, though.
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Sat Oct 17 09:30:45 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 17 Oct 2009 11:30:45 -0500
Subject: [Freeswitch-users] Sending an Event to a Session for onInput
In-Reply-To: <ACA9A602-0925-4D6F-8B11-0B214A462500@jerris.com>
References: <4256bf830910091234t11f4f4d0w28283f341bd1fb06@mail.gmail.com>
	<ACA9A602-0925-4D6F-8B11-0B214A462500@jerris.com>
Message-ID: <191c3a030910170930i30c0b428ya415162418db851b@mail.gmail.com>

actually,

If you use sendevent command and add a unique-id header to the event you
send, it will be sent to the channel rather than globally.


On Sun, Oct 11, 2009 at 5:04 PM, Michael Jerris <mike at jerris.com> wrote:

> We don't have session messages directly exposed, except for things
> like display, respond, and deflect.  What specifically are you trying
> to send ?
>
> Mike
>
> On Oct 9, 2009, at 3:34 PM, Matthew Fong wrote:
>
> > I'm used to using the onInput callbacks inside lua and javascript to
> > listen for dtmf and other events and perform a task accordingly. I'm
> > wondering if there is a way to send an event to a session or channel
> > that can be caught using the setInputCallback inside lua from
> > outside the session program. Maybe an API command that can generate
> > an event for a specific UUID. Does a mechanism exist to do this that
> > I'm over looking? Thanks.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Sat Oct 17 09:49:09 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 17 Oct 2009 11:49:09 -0500
Subject: [Freeswitch-users] SIP CANCEL error
In-Reply-To: <E1FE097E-599C-479E-9D1C-2308F213109C@gmail.com>
References: <E1FE097E-599C-479E-9D1C-2308F213109C@gmail.com>
Message-ID: <191c3a030910170949l1b4553dcub5c5d4a6ae529ae6@mail.gmail.com>

I think the phone is making a mistake with the CANCEL packet.

Here is the Via header from the INVITE the phone sends:
Via: SIP/2.0/UDP
190.137.190.131;rport;branch=z9hG4bKac109601000000fb4ad814425106977400000792

Here is the Via header from the CANCEL
Via: SIP/2.0/UDP
10.0.0.8;rport;branch=z9hG4bKac109601000000fb4ad814425106977400000792

it's identical aside from the private address contained in the CANCEL and a
public one in the INVITE

So the anti-nat settings on your phone are not working properly and they
forgot to substitute the public IP for the CANCEL.

That makes Sofia unable to match it to an existing call.



On Fri, Oct 16, 2009 at 1:44 AM, Rogelio Perez <rogelio.perez at gmail.com>wrote:

> Hi guys,
>
> I'm having a signaling issue on FS when calling from an endpoint to a
> GW.
> To reproduce the error I just call and hungup the call right away.
> FS gets the CANCEL and replies back with 481 Call/Transaction Does Not
> Exist, so the call continues and eventually connects, even when the A
> side of the call is closed.
> I have verified that the Call-ID of the CANCEL matches that from the
> INVITE, so I see no reasons for FS to reject it.
>
> SIP+FS logs in pastebin 10738.
>
> Any ideas?
>
> Thanks,
> Rogelio
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From miles.chet at gmail.com  Sat Oct 17 13:01:57 2009
From: miles.chet at gmail.com (Roberto Martins)
Date: Sat, 17 Oct 2009 17:01:57 -0300
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <191c3a030910170925i7aa7d4f3p9d382e0d855b7c8b@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<2E845C1E-C084-4025-806D-178A175FF90A@gmail.com>
	<f3cdc69d0910161302v6a97ead2xc2af5d73b5a048d7@mail.gmail.com>
	<d3ceea380910161328w4c843e9as49b2341dc6f8eb7d@mail.gmail.com>
	<20091017025323.GA15649@jdc.jasonjgw.net>
	<c4d05cbe0910162005r3f094fc7ne448b905446282f4@mail.gmail.com>
	<86a32abc0910162019m6315692bw566e282aaf8e2900@mail.gmail.com>
	<20091017035215.GA24308@jdc.jasonjgw.net>
	<ebbc2d550910162123p33a80f0fnd8ebac5fc569588d@mail.gmail.com>
	<7600794b0910170406m164e3e2od198586f53de5efc@mail.gmail.com>
	<191c3a030910170925i7aa7d4f3p9d382e0d855b7c8b@mail.gmail.com>
Message-ID: <6701786D-A8AF-444E-A5A7-29E69F6E055C@gmail.com>

sub-domain seem to be a very wise choice...
ptbr



On Oct 17, 2009, at 1:25 PM, Anthony Minessale wrote:

> or someone could just ask us for pg.freeswitch.org or whatever other  
> sub-domain to avoid trademark infringement of registering 40 domains  
> with freeswitch in the name for no reason.
>
>
> On Sat, Oct 17, 2009 at 6:06 AM, Rud? Cunha <ruda at ruda.com.br> wrote:
> Jo?o Mesquita,
>
> Could check with Jeremiah or contact me through, so I talk to him  
> about the forum?
>
> He could redirect the domain to my hosting server and we can put  
> there the wiki and forums and we can communicate!
>
> Rud? Cunha
>
> 2009/10/17 Jo?o Mesquita <jmesquita at freeswitch.org>
>
> Just a heads up, I have talked to Jeremias from Khomp today and he  
> is setting up the wiki. I will personally be adding contents to the  
> that wiki if it ever picks up.
>
> Regards,
>
> jm
>
>
> On Sat, Oct 17, 2009 at 12:52 AM, Jason White <jason at jasonjgw.net>  
> wrote:
> Diego Viola <diego.viola at gmail.com> wrote:
> > I'm with Moises and with the other people supporting this  
> initiative.
> >
> > I'm not Brazilian, but they should be able to do whatever they  
> want, after
> > all, that's how open source works, if you can do it go ahead and  
> do it.
>
> Correct. We have enough of them as far as English-language fora are  
> concerned;
> other languages are a different question altogether, though.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From diego.viola at gmail.com  Sat Oct 17 13:24:54 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 17 Oct 2009 20:24:54 +0000
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <191c3a030910170925i7aa7d4f3p9d382e0d855b7c8b@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<f3cdc69d0910161302v6a97ead2xc2af5d73b5a048d7@mail.gmail.com>
	<d3ceea380910161328w4c843e9as49b2341dc6f8eb7d@mail.gmail.com>
	<20091017025323.GA15649@jdc.jasonjgw.net>
	<c4d05cbe0910162005r3f094fc7ne448b905446282f4@mail.gmail.com>
	<86a32abc0910162019m6315692bw566e282aaf8e2900@mail.gmail.com>
	<20091017035215.GA24308@jdc.jasonjgw.net>
	<ebbc2d550910162123p33a80f0fnd8ebac5fc569588d@mail.gmail.com>
	<7600794b0910170406m164e3e2od198586f53de5efc@mail.gmail.com>
	<191c3a030910170925i7aa7d4f3p9d382e0d855b7c8b@mail.gmail.com>
Message-ID: <86a32abc0910171324l454b6719xba3c920d80d35b61@mail.gmail.com>

I'm with Anthony, the sub-domain idea is nice, I wish we could find a way to
do internalization in our wiki in a easy/elegantly way.

Diego

On Sat, Oct 17, 2009 at 4:25 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> or someone could just ask us for pg.freeswitch.org or whatever other
> sub-domain to avoid trademark infringement of registering 40 domains with
> freeswitch in the name for no reason.
>
>
>
> On Sat, Oct 17, 2009 at 6:06 AM, Rud? Cunha <ruda at ruda.com.br> wrote:
>
>> Jo?o Mesquita,
>>
>> Could check with Jeremiah or contact me through, so I talk to him about
>> the forum?
>>
>> He could redirect the domain to my hosting server and we can put there the
>> wiki and forums and we can communicate!
>>
>> Rud? Cunha
>>
>> 2009/10/17 Jo?o Mesquita <jmesquita at freeswitch.org>
>>
>> Just a heads up, I have talked to Jeremias from Khomp today and he is
>>> setting up the wiki. I will personally be adding contents to the that wiki
>>> if it ever picks up.
>>>
>>> Regards,
>>>
>>> jm
>>>
>>>
>>> On Sat, Oct 17, 2009 at 12:52 AM, Jason White <jason at jasonjgw.net>wrote:
>>>
>>>> Diego Viola <diego.viola at gmail.com> wrote:
>>>> > I'm with Moises and with the other people supporting this initiative.
>>>> >
>>>> > I'm not Brazilian, but they should be able to do whatever they want,
>>>> after
>>>> > all, that's how open source works, if you can do it go ahead and do
>>>> it.
>>>>
>>>> Correct. We have enough of them as far as English-language fora are
>>>> concerned;
>>>> other languages are a different question altogether, though.
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Sat Oct 17 13:29:55 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 17 Oct 2009 16:29:55 -0400
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <6701786D-A8AF-444E-A5A7-29E69F6E055C@gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<2E845C1E-C084-4025-806D-178A175FF90A@gmail.com>
	<f3cdc69d0910161302v6a97ead2xc2af5d73b5a048d7@mail.gmail.com>
	<d3ceea380910161328w4c843e9as49b2341dc6f8eb7d@mail.gmail.com>
	<20091017025323.GA15649@jdc.jasonjgw.net>
	<c4d05cbe0910162005r3f094fc7ne448b905446282f4@mail.gmail.com>
	<86a32abc0910162019m6315692bw566e282aaf8e2900@mail.gmail.com>
	<20091017035215.GA24308@jdc.jasonjgw.net>
	<ebbc2d550910162123p33a80f0fnd8ebac5fc569588d@mail.gmail.com>
	<7600794b0910170406m164e3e2od198586f53de5efc@mail.gmail.com>
	<191c3a030910170925i7aa7d4f3p9d382e0d855b7c8b@mail.gmail.com>
	<6701786D-A8AF-444E-A5A7-29E69F6E055C@gmail.com>
Message-ID: <B69597BF-DA4A-440B-8CB0-08F52F33DD17@jerris.com>

I think we strongly lean towards using a sub-domain (if necessary) and  
maintaining other language content in the same wiki in alternate  
pages.  If the wiki software we are using is not effective to create  
multiple languages we should find a way to do it all in one.  We can  
set up an additional mailing list if necessary but also we have no  
issue with going across multiple languages in both the mailing list  
and in irc.

Mike

On Oct 17, 2009, at 4:01 PM, Roberto Martins wrote:

> sub-domain seem to be a very wise choice...
> ptbr
>
>
>
> On Oct 17, 2009, at 1:25 PM, Anthony Minessale wrote:
>
>> or someone could just ask us for pg.freeswitch.org or whatever other
>> sub-domain to avoid trademark infringement of registering 40 domains
>> with freeswitch in the name for no reason.
>>
>>
>> On Sat, Oct 17, 2009 at 6:06 AM, Rud? Cunha <ruda at ruda.com.br> wrote:
>> Jo?o Mesquita,
>>
>> Could check with Jeremiah or contact me through, so I talk to him
>> about the forum?
>>
>> He could redirect the domain to my hosting server and we can put
>> there the wiki and forums and we can communicate!
>>
>> Rud? Cunha
>>
>> 2009/10/17 Jo?o Mesquita <jmesquita at freeswitch.org>
>>
>> Just a heads up, I have talked to Jeremias from Khomp today and he
>> is setting up the wiki. I will personally be adding contents to the
>> that wiki if it ever picks up.
>>
>> Regards,
>>
>> jm
>>
>>
>> On Sat, Oct 17, 2009 at 12:52 AM, Jason White <jason at jasonjgw.net>
>> wrote:
>> Diego Viola <diego.viola at gmail.com> wrote:
>>> I'm with Moises and with the other people supporting this
>> initiative.
>>>
>>> I'm not Brazilian, but they should be able to do whatever they
>> want, after
>>> all, that's how open source works, if you can do it go ahead and
>> do it.
>>
>> Correct. We have enough of them as far as English-language fora are
>> concerned;
>> other languages are a different question altogether, though.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> -- 
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:213-799-1400
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From diego.viola at gmail.com  Sat Oct 17 13:39:53 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 17 Oct 2009 20:39:53 +0000
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <B69597BF-DA4A-440B-8CB0-08F52F33DD17@jerris.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<20091017025323.GA15649@jdc.jasonjgw.net>
	<c4d05cbe0910162005r3f094fc7ne448b905446282f4@mail.gmail.com>
	<86a32abc0910162019m6315692bw566e282aaf8e2900@mail.gmail.com>
	<20091017035215.GA24308@jdc.jasonjgw.net>
	<ebbc2d550910162123p33a80f0fnd8ebac5fc569588d@mail.gmail.com>
	<7600794b0910170406m164e3e2od198586f53de5efc@mail.gmail.com>
	<191c3a030910170925i7aa7d4f3p9d382e0d855b7c8b@mail.gmail.com>
	<6701786D-A8AF-444E-A5A7-29E69F6E055C@gmail.com>
	<B69597BF-DA4A-440B-8CB0-08F52F33DD17@jerris.com>
Message-ID: <86a32abc0910171339o6a300932v66b7f00419a0cb69@mail.gmail.com>

If we find a better wiki software that does internalization in a better way,
I could help porting the documentation over to the new software.

The sub-domain idea is also very nice.

Diego


On Sat, Oct 17, 2009 at 8:29 PM, Michael Jerris <mike at jerris.com> wrote:

> I think we strongly lean towards using a sub-domain (if necessary) and
> maintaining other language content in the same wiki in alternate
> pages.  If the wiki software we are using is not effective to create
> multiple languages we should find a way to do it all in one.  We can
> set up an additional mailing list if necessary but also we have no
> issue with going across multiple languages in both the mailing list
> and in irc.
>
> Mike
>
> On Oct 17, 2009, at 4:01 PM, Roberto Martins wrote:
>
> > sub-domain seem to be a very wise choice...
> > ptbr
> >
> >
> >
> > On Oct 17, 2009, at 1:25 PM, Anthony Minessale wrote:
> >
> >> or someone could just ask us for pg.freeswitch.org or whatever other
> >> sub-domain to avoid trademark infringement of registering 40 domains
> >> with freeswitch in the name for no reason.
> >>
> >>
> >> On Sat, Oct 17, 2009 at 6:06 AM, Rud? Cunha <ruda at ruda.com.br> wrote:
> >> Jo?o Mesquita,
> >>
> >> Could check with Jeremiah or contact me through, so I talk to him
> >> about the forum?
> >>
> >> He could redirect the domain to my hosting server and we can put
> >> there the wiki and forums and we can communicate!
> >>
> >> Rud? Cunha
> >>
> >> 2009/10/17 Jo?o Mesquita <jmesquita at freeswitch.org>
> >>
> >> Just a heads up, I have talked to Jeremias from Khomp today and he
> >> is setting up the wiki. I will personally be adding contents to the
> >> that wiki if it ever picks up.
> >>
> >> Regards,
> >>
> >> jm
> >>
> >>
> >> On Sat, Oct 17, 2009 at 12:52 AM, Jason White <jason at jasonjgw.net>
> >> wrote:
> >> Diego Viola <diego.viola at gmail.com> wrote:
> >>> I'm with Moises and with the other people supporting this
> >> initiative.
> >>>
> >>> I'm not Brazilian, but they should be able to do whatever they
> >> want, after
> >>> all, that's how open source works, if you can do it go ahead and
> >> do it.
> >>
> >> Correct. We have enough of them as far as English-language fora are
> >> concerned;
> >> other languages are a different question altogether, though.
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> iax:guest at conference.freeswitch.org/888
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> pstn:213-799-1400
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091017/e5b233da/attachment-0002.html 

From mike at jerris.com  Sat Oct 17 14:04:07 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 17 Oct 2009 17:04:07 -0400
Subject: [Freeswitch-users] Dingaling: using a hostname instead of stun
	for rtp
In-Reply-To: <65BF398D-E3A6-4A98-83D9-D4E271070CCA@freeswitch.org>
References: <33c87fa30910131502t2af24801i1c3319518b5db50b@mail.gmail.com>
	<C874E25D-6D73-49AA-8DD3-0A0022617FD2@freeswitch.org>
	<33c87fa30910152035w43e260faoc031a7a4ac949d9e@mail.gmail.com>
	<65BF398D-E3A6-4A98-83D9-D4E271070CCA@freeswitch.org>
Message-ID: <7710644E-CEA1-4D48-8A03-EEE1C18F1B44@jerris.com>

If you don't have working stun, jingle is not going to work very  
well.  It is a required part of the protocol.  You need to be able to  
determine your external ports for media on each call, using a host  
name will not do this for you.

Mike


On Oct 16, 2009, at 10:48 AM, Brian West wrote:

> If you setup your own stun server it wouldn't do that....  But the
> hostlookup only solves half the problem .. getting the external IP vs
> poking holes for RTP which is what stun will do.
>
> /b
>
> On Oct 15, 2009, at 10:35 PM, Mark Campbell-Smith wrote:
>
>> Thanks Brian.  Is this something that is planned to be implemented?
>> The workaround is to set the stun server also in the dingaling
>> configuration, but as I said, for some reason the stun times for me
>> out occasionally with dingaling.
>>
>> Thanks!
>




From diego.viola at gmail.com  Sat Oct 17 15:13:53 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 17 Oct 2009 22:13:53 +0000
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <86a32abc0910171339o6a300932v66b7f00419a0cb69@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<c4d05cbe0910162005r3f094fc7ne448b905446282f4@mail.gmail.com>
	<86a32abc0910162019m6315692bw566e282aaf8e2900@mail.gmail.com>
	<20091017035215.GA24308@jdc.jasonjgw.net>
	<ebbc2d550910162123p33a80f0fnd8ebac5fc569588d@mail.gmail.com>
	<7600794b0910170406m164e3e2od198586f53de5efc@mail.gmail.com>
	<191c3a030910170925i7aa7d4f3p9d382e0d855b7c8b@mail.gmail.com>
	<6701786D-A8AF-444E-A5A7-29E69F6E055C@gmail.com>
	<B69597BF-DA4A-440B-8CB0-08F52F33DD17@jerris.com>
	<86a32abc0910171339o6a300932v66b7f00419a0cb69@mail.gmail.com>
Message-ID: <86a32abc0910171513n35d7d0echff940a16685ae1a4@mail.gmail.com>

Would you be open to test another wiki software, that supports
internalization/translation easier, if i am open to port the entire
documentation to the new wiki software?




On Sat, Oct 17, 2009 at 8:39 PM, Diego Viola <diego.viola at gmail.com> wrote:

> If we find a better wiki software that does internalization in a better
> way, I could help porting the documentation over to the new software.
>
> The sub-domain idea is also very nice.
>
> Diego
>
>
>
> On Sat, Oct 17, 2009 at 8:29 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> I think we strongly lean towards using a sub-domain (if necessary) and
>> maintaining other language content in the same wiki in alternate
>> pages.  If the wiki software we are using is not effective to create
>> multiple languages we should find a way to do it all in one.  We can
>> set up an additional mailing list if necessary but also we have no
>> issue with going across multiple languages in both the mailing list
>> and in irc.
>>
>> Mike
>>
>> On Oct 17, 2009, at 4:01 PM, Roberto Martins wrote:
>>
>> > sub-domain seem to be a very wise choice...
>> > ptbr
>> >
>> >
>> >
>> > On Oct 17, 2009, at 1:25 PM, Anthony Minessale wrote:
>> >
>> >> or someone could just ask us for pg.freeswitch.org or whatever other
>> >> sub-domain to avoid trademark infringement of registering 40 domains
>> >> with freeswitch in the name for no reason.
>> >>
>> >>
>> >> On Sat, Oct 17, 2009 at 6:06 AM, Rud? Cunha <ruda at ruda.com.br> wrote:
>> >> Jo?o Mesquita,
>> >>
>> >> Could check with Jeremiah or contact me through, so I talk to him
>> >> about the forum?
>> >>
>> >> He could redirect the domain to my hosting server and we can put
>> >> there the wiki and forums and we can communicate!
>> >>
>> >> Rud? Cunha
>> >>
>> >> 2009/10/17 Jo?o Mesquita <jmesquita at freeswitch.org>
>> >>
>> >> Just a heads up, I have talked to Jeremias from Khomp today and he
>> >> is setting up the wiki. I will personally be adding contents to the
>> >> that wiki if it ever picks up.
>> >>
>> >> Regards,
>> >>
>> >> jm
>> >>
>> >>
>> >> On Sat, Oct 17, 2009 at 12:52 AM, Jason White <jason at jasonjgw.net>
>> >> wrote:
>> >> Diego Viola <diego.viola at gmail.com> wrote:
>> >>> I'm with Moises and with the other people supporting this
>> >> initiative.
>> >>>
>> >>> I'm not Brazilian, but they should be able to do whatever they
>> >> want, after
>> >>> all, that's how open source works, if you can do it go ahead and
>> >> do it.
>> >>
>> >> Correct. We have enough of them as far as English-language fora are
>> >> concerned;
>> >> other languages are a different question altogether, though.
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> >> users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> >> users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> >> users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH http://www.freeswitch.org/
>> >> ClueCon http://www.cluecon.com/
>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>
>> >> AIM: anthm
>> >> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> >> IRC: irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>> >> iax:guest at conference.freeswitch.org/888
>> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> >> pstn:213-799-1400
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> >> users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> > users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From diego.viola at gmail.com  Sat Oct 17 15:21:32 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 17 Oct 2009 22:21:32 +0000
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <86a32abc0910171513n35d7d0echff940a16685ae1a4@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<86a32abc0910162019m6315692bw566e282aaf8e2900@mail.gmail.com>
	<20091017035215.GA24308@jdc.jasonjgw.net>
	<ebbc2d550910162123p33a80f0fnd8ebac5fc569588d@mail.gmail.com>
	<7600794b0910170406m164e3e2od198586f53de5efc@mail.gmail.com>
	<191c3a030910170925i7aa7d4f3p9d382e0d855b7c8b@mail.gmail.com>
	<6701786D-A8AF-444E-A5A7-29E69F6E055C@gmail.com>
	<B69597BF-DA4A-440B-8CB0-08F52F33DD17@jerris.com>
	<86a32abc0910171339o6a300932v66b7f00419a0cb69@mail.gmail.com>
	<86a32abc0910171513n35d7d0echff940a16685ae1a4@mail.gmail.com>
Message-ID: <86a32abc0910171521n403094bfvf27aa7c2a1c1f798@mail.gmail.com>

Some interesting pages on the subject.

http://en.wikipedia.org/wiki/Comparison_of_wiki_software
http://www.osnews.com/story/21865/Recommend_an_Open_Source_Wiki_Platform



On Sat, Oct 17, 2009 at 10:13 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Would you be open to test another wiki software, that supports
> internalization/translation easier, if i am open to port the entire
> documentation to the new wiki software?
>
>
>
>
>
> On Sat, Oct 17, 2009 at 8:39 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> If we find a better wiki software that does internalization in a better
>> way, I could help porting the documentation over to the new software.
>>
>> The sub-domain idea is also very nice.
>>
>> Diego
>>
>>
>>
>> On Sat, Oct 17, 2009 at 8:29 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> I think we strongly lean towards using a sub-domain (if necessary) and
>>> maintaining other language content in the same wiki in alternate
>>> pages.  If the wiki software we are using is not effective to create
>>> multiple languages we should find a way to do it all in one.  We can
>>> set up an additional mailing list if necessary but also we have no
>>> issue with going across multiple languages in both the mailing list
>>> and in irc.
>>>
>>> Mike
>>>
>>> On Oct 17, 2009, at 4:01 PM, Roberto Martins wrote:
>>>
>>> > sub-domain seem to be a very wise choice...
>>> > ptbr
>>> >
>>> >
>>> >
>>> > On Oct 17, 2009, at 1:25 PM, Anthony Minessale wrote:
>>> >
>>> >> or someone could just ask us for pg.freeswitch.org or whatever other
>>> >> sub-domain to avoid trademark infringement of registering 40 domains
>>> >> with freeswitch in the name for no reason.
>>> >>
>>> >>
>>> >> On Sat, Oct 17, 2009 at 6:06 AM, Rud? Cunha <ruda at ruda.com.br> wrote:
>>> >> Jo?o Mesquita,
>>> >>
>>> >> Could check with Jeremiah or contact me through, so I talk to him
>>> >> about the forum?
>>> >>
>>> >> He could redirect the domain to my hosting server and we can put
>>> >> there the wiki and forums and we can communicate!
>>> >>
>>> >> Rud? Cunha
>>> >>
>>> >> 2009/10/17 Jo?o Mesquita <jmesquita at freeswitch.org>
>>> >>
>>> >> Just a heads up, I have talked to Jeremias from Khomp today and he
>>> >> is setting up the wiki. I will personally be adding contents to the
>>> >> that wiki if it ever picks up.
>>> >>
>>> >> Regards,
>>> >>
>>> >> jm
>>> >>
>>> >>
>>> >> On Sat, Oct 17, 2009 at 12:52 AM, Jason White <jason at jasonjgw.net>
>>> >> wrote:
>>> >> Diego Viola <diego.viola at gmail.com> wrote:
>>> >>> I'm with Moises and with the other people supporting this
>>> >> initiative.
>>> >>>
>>> >>> I'm not Brazilian, but they should be able to do whatever they
>>> >> want, after
>>> >>> all, that's how open source works, if you can do it go ahead and
>>> >> do it.
>>> >>
>>> >> Correct. We have enough of them as far as English-language fora are
>>> >> concerned;
>>> >> other languages are a different question altogether, though.
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> >> users
>>> >> http://www.freeswitch.org
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> >> users
>>> >> http://www.freeswitch.org
>>> >>
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> >> users
>>> >> http://www.freeswitch.org
>>> >>
>>> >>
>>> >>
>>> >>
>>> >> --
>>> >> Anthony Minessale II
>>> >>
>>> >> FreeSWITCH http://www.freeswitch.org/
>>> >> ClueCon http://www.cluecon.com/
>>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>>> >>
>>> >> AIM: anthm
>>> >> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> >> IRC: irc.freenode.net #freeswitch
>>> >>
>>> >> FreeSWITCH Developer Conference
>>> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>> >> iax:guest at conference.freeswitch.org/888
>>> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> >> pstn:213-799-1400
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> >> users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> > users
>>> > http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
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From diego.viola at gmail.com  Sat Oct 17 15:35:43 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 17 Oct 2009 22:35:43 +0000
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <86a32abc0910171521n403094bfvf27aa7c2a1c1f798@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<20091017035215.GA24308@jdc.jasonjgw.net>
	<ebbc2d550910162123p33a80f0fnd8ebac5fc569588d@mail.gmail.com>
	<7600794b0910170406m164e3e2od198586f53de5efc@mail.gmail.com>
	<191c3a030910170925i7aa7d4f3p9d382e0d855b7c8b@mail.gmail.com>
	<6701786D-A8AF-444E-A5A7-29E69F6E055C@gmail.com>
	<B69597BF-DA4A-440B-8CB0-08F52F33DD17@jerris.com>
	<86a32abc0910171339o6a300932v66b7f00419a0cb69@mail.gmail.com>
	<86a32abc0910171513n35d7d0echff940a16685ae1a4@mail.gmail.com>
	<86a32abc0910171521n403094bfvf27aa7c2a1c1f798@mail.gmail.com>
Message-ID: <86a32abc0910171535l606eb9c6id54f7b3b5ae1b670@mail.gmail.com>

We could also do this for a better documentation.


   1. Pick a new wiki engine that does internalization/translation easier.
   2. Set some documentation guidelines from the start.
   3. Move stuff over to the new wiki.
   4. Start with the translation/internalization on the new wiki.


Diego

On Sat, Oct 17, 2009 at 10:21 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Some interesting pages on the subject.
>
> http://en.wikipedia.org/wiki/Comparison_of_wiki_software
> http://www.osnews.com/story/21865/Recommend_an_Open_Source_Wiki_Platform
>
>
>
>
> On Sat, Oct 17, 2009 at 10:13 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> Would you be open to test another wiki software, that supports
>> internalization/translation easier, if i am open to port the entire
>> documentation to the new wiki software?
>>
>>
>>
>>
>>
>> On Sat, Oct 17, 2009 at 8:39 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>
>>> If we find a better wiki software that does internalization in a better
>>> way, I could help porting the documentation over to the new software.
>>>
>>> The sub-domain idea is also very nice.
>>>
>>> Diego
>>>
>>>
>>>
>>> On Sat, Oct 17, 2009 at 8:29 PM, Michael Jerris <mike at jerris.com> wrote:
>>>
>>>> I think we strongly lean towards using a sub-domain (if necessary) and
>>>> maintaining other language content in the same wiki in alternate
>>>> pages.  If the wiki software we are using is not effective to create
>>>> multiple languages we should find a way to do it all in one.  We can
>>>> set up an additional mailing list if necessary but also we have no
>>>> issue with going across multiple languages in both the mailing list
>>>> and in irc.
>>>>
>>>> Mike
>>>>
>>>> On Oct 17, 2009, at 4:01 PM, Roberto Martins wrote:
>>>>
>>>> > sub-domain seem to be a very wise choice...
>>>> > ptbr
>>>> >
>>>> >
>>>> >
>>>> > On Oct 17, 2009, at 1:25 PM, Anthony Minessale wrote:
>>>> >
>>>> >> or someone could just ask us for pg.freeswitch.org or whatever other
>>>> >> sub-domain to avoid trademark infringement of registering 40 domains
>>>> >> with freeswitch in the name for no reason.
>>>> >>
>>>> >>
>>>> >> On Sat, Oct 17, 2009 at 6:06 AM, Rud? Cunha <ruda at ruda.com.br>
>>>> wrote:
>>>> >> Jo?o Mesquita,
>>>> >>
>>>> >> Could check with Jeremiah or contact me through, so I talk to him
>>>> >> about the forum?
>>>> >>
>>>> >> He could redirect the domain to my hosting server and we can put
>>>> >> there the wiki and forums and we can communicate!
>>>> >>
>>>> >> Rud? Cunha
>>>> >>
>>>> >> 2009/10/17 Jo?o Mesquita <jmesquita at freeswitch.org>
>>>> >>
>>>> >> Just a heads up, I have talked to Jeremias from Khomp today and he
>>>> >> is setting up the wiki. I will personally be adding contents to the
>>>> >> that wiki if it ever picks up.
>>>> >>
>>>> >> Regards,
>>>> >>
>>>> >> jm
>>>> >>
>>>> >>
>>>> >> On Sat, Oct 17, 2009 at 12:52 AM, Jason White <jason at jasonjgw.net>
>>>> >> wrote:
>>>> >> Diego Viola <diego.viola at gmail.com> wrote:
>>>> >>> I'm with Moises and with the other people supporting this
>>>> >> initiative.
>>>> >>>
>>>> >>> I'm not Brazilian, but they should be able to do whatever they
>>>> >> want, after
>>>> >>> all, that's how open source works, if you can do it go ahead and
>>>> >> do it.
>>>> >>
>>>> >> Correct. We have enough of them as far as English-language fora are
>>>> >> concerned;
>>>> >> other languages are a different question altogether, though.
>>>> >>
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>> >> users
>>>> >> http://www.freeswitch.org
>>>> >>
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>> >> users
>>>> >> http://www.freeswitch.org
>>>> >>
>>>> >>
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>> >> users
>>>> >> http://www.freeswitch.org
>>>> >>
>>>> >>
>>>> >>
>>>> >>
>>>> >> --
>>>> >> Anthony Minessale II
>>>> >>
>>>> >> FreeSWITCH http://www.freeswitch.org/
>>>> >> ClueCon http://www.cluecon.com/
>>>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>>>> >>
>>>> >> AIM: anthm
>>>> >> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> >> IRC: irc.freenode.net #freeswitch
>>>> >>
>>>> >> FreeSWITCH Developer Conference
>>>> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>> >> iax:guest at conference.freeswitch.org/888
>>>> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> >> pstn:213-799-1400
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>> >> users
>>>> >> http://www.freeswitch.org
>>>> >
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>> > users
>>>> > http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>
>
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From diego.viola at gmail.com  Sat Oct 17 15:45:40 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 17 Oct 2009 22:45:40 +0000
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <86a32abc0910171535l606eb9c6id54f7b3b5ae1b670@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<ebbc2d550910162123p33a80f0fnd8ebac5fc569588d@mail.gmail.com>
	<7600794b0910170406m164e3e2od198586f53de5efc@mail.gmail.com>
	<191c3a030910170925i7aa7d4f3p9d382e0d855b7c8b@mail.gmail.com>
	<6701786D-A8AF-444E-A5A7-29E69F6E055C@gmail.com>
	<B69597BF-DA4A-440B-8CB0-08F52F33DD17@jerris.com>
	<86a32abc0910171339o6a300932v66b7f00419a0cb69@mail.gmail.com>
	<86a32abc0910171513n35d7d0echff940a16685ae1a4@mail.gmail.com>
	<86a32abc0910171521n403094bfvf27aa7c2a1c1f798@mail.gmail.com>
	<86a32abc0910171535l606eb9c6id54f7b3b5ae1b670@mail.gmail.com>
Message-ID: <86a32abc0910171545l17f3f539ldba432db550c3eaa@mail.gmail.com>

Instead of adding million of different domains for the languages, it would
be nice if we only keep wiki.freeswitch.org, and the wiki software provides
a list of languages to choose from.

Then when we click in one of the languages we have the wiki translated to
the desired language.

I believe if we go this way we will be more organized, and there is more
chances that things will get translated, and we wont be repeating too much
information here and there too.

Just a thought... I'm open to do this work to make it happen.

The different domains per language is a good idea too.

Diego

On Sat, Oct 17, 2009 at 10:35 PM, Diego Viola <diego.viola at gmail.com> wrote:

> We could also do this for a better documentation.
>
>
>    1. Pick a new wiki engine that does internalization/translation easier.
>    2. Set some documentation guidelines from the start.
>    3. Move stuff over to the new wiki.
>    4. Start with the translation/internalization on the new wiki.
>
>
> Diego
>
>
> On Sat, Oct 17, 2009 at 10:21 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> Some interesting pages on the subject.
>>
>> http://en.wikipedia.org/wiki/Comparison_of_wiki_software
>> http://www.osnews.com/story/21865/Recommend_an_Open_Source_Wiki_Platform
>>
>>
>>
>>
>> On Sat, Oct 17, 2009 at 10:13 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>
>>> Would you be open to test another wiki software, that supports
>>> internalization/translation easier, if i am open to port the entire
>>> documentation to the new wiki software?
>>>
>>>
>>>
>>>
>>>
>>> On Sat, Oct 17, 2009 at 8:39 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>
>>>> If we find a better wiki software that does internalization in a better
>>>> way, I could help porting the documentation over to the new software.
>>>>
>>>> The sub-domain idea is also very nice.
>>>>
>>>> Diego
>>>>
>>>>
>>>>
>>>> On Sat, Oct 17, 2009 at 8:29 PM, Michael Jerris <mike at jerris.com>wrote:
>>>>
>>>>> I think we strongly lean towards using a sub-domain (if necessary) and
>>>>> maintaining other language content in the same wiki in alternate
>>>>> pages.  If the wiki software we are using is not effective to create
>>>>> multiple languages we should find a way to do it all in one.  We can
>>>>> set up an additional mailing list if necessary but also we have no
>>>>> issue with going across multiple languages in both the mailing list
>>>>> and in irc.
>>>>>
>>>>> Mike
>>>>>
>>>>> On Oct 17, 2009, at 4:01 PM, Roberto Martins wrote:
>>>>>
>>>>> > sub-domain seem to be a very wise choice...
>>>>> > ptbr
>>>>> >
>>>>> >
>>>>> >
>>>>> > On Oct 17, 2009, at 1:25 PM, Anthony Minessale wrote:
>>>>> >
>>>>> >> or someone could just ask us for pg.freeswitch.org or whatever
>>>>> other
>>>>> >> sub-domain to avoid trademark infringement of registering 40 domains
>>>>> >> with freeswitch in the name for no reason.
>>>>> >>
>>>>> >>
>>>>> >> On Sat, Oct 17, 2009 at 6:06 AM, Rud? Cunha <ruda at ruda.com.br>
>>>>> wrote:
>>>>> >> Jo?o Mesquita,
>>>>> >>
>>>>> >> Could check with Jeremiah or contact me through, so I talk to him
>>>>> >> about the forum?
>>>>> >>
>>>>> >> He could redirect the domain to my hosting server and we can put
>>>>> >> there the wiki and forums and we can communicate!
>>>>> >>
>>>>> >> Rud? Cunha
>>>>> >>
>>>>> >> 2009/10/17 Jo?o Mesquita <jmesquita at freeswitch.org>
>>>>> >>
>>>>> >> Just a heads up, I have talked to Jeremias from Khomp today and he
>>>>> >> is setting up the wiki. I will personally be adding contents to the
>>>>> >> that wiki if it ever picks up.
>>>>> >>
>>>>> >> Regards,
>>>>> >>
>>>>> >> jm
>>>>> >>
>>>>> >>
>>>>> >> On Sat, Oct 17, 2009 at 12:52 AM, Jason White <jason at jasonjgw.net>
>>>>> >> wrote:
>>>>> >> Diego Viola <diego.viola at gmail.com> wrote:
>>>>> >>> I'm with Moises and with the other people supporting this
>>>>> >> initiative.
>>>>> >>>
>>>>> >>> I'm not Brazilian, but they should be able to do whatever they
>>>>> >> want, after
>>>>> >>> all, that's how open source works, if you can do it go ahead and
>>>>> >> do it.
>>>>> >>
>>>>> >> Correct. We have enough of them as far as English-language fora are
>>>>> >> concerned;
>>>>> >> other languages are a different question altogether, though.
>>>>> >>
>>>>> >>
>>>>> >> _______________________________________________
>>>>> >> FreeSWITCH-users mailing list
>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>> >> users
>>>>> >> http://www.freeswitch.org
>>>>> >>
>>>>> >>
>>>>> >> _______________________________________________
>>>>> >> FreeSWITCH-users mailing list
>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>> >> users
>>>>> >> http://www.freeswitch.org
>>>>> >>
>>>>> >>
>>>>> >>
>>>>> >> _______________________________________________
>>>>> >> FreeSWITCH-users mailing list
>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>> >> users
>>>>> >> http://www.freeswitch.org
>>>>> >>
>>>>> >>
>>>>> >>
>>>>> >>
>>>>> >> --
>>>>> >> Anthony Minessale II
>>>>> >>
>>>>> >> FreeSWITCH http://www.freeswitch.org/
>>>>> >> ClueCon http://www.cluecon.com/
>>>>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>> >>
>>>>> >> AIM: anthm
>>>>> >> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>> >> IRC: irc.freenode.net #freeswitch
>>>>> >>
>>>>> >> FreeSWITCH Developer Conference
>>>>> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>> >> iax:guest at conference.freeswitch.org/888
>>>>> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>> >> pstn:213-799-1400
>>>>> >> _______________________________________________
>>>>> >> FreeSWITCH-users mailing list
>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>> >> users
>>>>> >> http://www.freeswitch.org
>>>>> >
>>>>> >
>>>>> > _______________________________________________
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>> > users
>>>>> > http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>
>>
>
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From diego.viola at gmail.com  Sat Oct 17 16:01:13 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 17 Oct 2009 23:01:13 +0000
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <86a32abc0910171545l17f3f539ldba432db550c3eaa@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<7600794b0910170406m164e3e2od198586f53de5efc@mail.gmail.com>
	<191c3a030910170925i7aa7d4f3p9d382e0d855b7c8b@mail.gmail.com>
	<6701786D-A8AF-444E-A5A7-29E69F6E055C@gmail.com>
	<B69597BF-DA4A-440B-8CB0-08F52F33DD17@jerris.com>
	<86a32abc0910171339o6a300932v66b7f00419a0cb69@mail.gmail.com>
	<86a32abc0910171513n35d7d0echff940a16685ae1a4@mail.gmail.com>
	<86a32abc0910171521n403094bfvf27aa7c2a1c1f798@mail.gmail.com>
	<86a32abc0910171535l606eb9c6id54f7b3b5ae1b670@mail.gmail.com>
	<86a32abc0910171545l17f3f539ldba432db550c3eaa@mail.gmail.com>
Message-ID: <86a32abc0910171601u6ab06e42xcf303301c85c57b1@mail.gmail.com>

Take a look at this for instance:

http://wiki.rubyonrails.org/start

This uses DokuWiki, check the langauges option at the top (en, de, el, es,
fr, etc) and when you click one of them it translates the actual page to the
desired language... I'd love to have something like that.

Diego

On Sat, Oct 17, 2009 at 10:45 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Instead of adding million of different domains for the languages, it would
> be nice if we only keep wiki.freeswitch.org, and the wiki software
> provides a list of languages to choose from.
>
> Then when we click in one of the languages we have the wiki translated to
> the desired language.
>
> I believe if we go this way we will be more organized, and there is more
> chances that things will get translated, and we wont be repeating too much
> information here and there too.
>
> Just a thought... I'm open to do this work to make it happen.
>
> The different domains per language is a good idea too.
>
> Diego
>
>
> On Sat, Oct 17, 2009 at 10:35 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> We could also do this for a better documentation.
>>
>>
>>    1. Pick a new wiki engine that does internalization/translation
>>    easier.
>>    2. Set some documentation guidelines from the start.
>>    3. Move stuff over to the new wiki.
>>    4. Start with the translation/internalization on the new wiki.
>>
>>
>> Diego
>>
>>
>> On Sat, Oct 17, 2009 at 10:21 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>
>>> Some interesting pages on the subject.
>>>
>>> http://en.wikipedia.org/wiki/Comparison_of_wiki_software
>>> http://www.osnews.com/story/21865/Recommend_an_Open_Source_Wiki_Platform
>>>
>>>
>>>
>>>
>>> On Sat, Oct 17, 2009 at 10:13 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>
>>>> Would you be open to test another wiki software, that supports
>>>> internalization/translation easier, if i am open to port the entire
>>>> documentation to the new wiki software?
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Sat, Oct 17, 2009 at 8:39 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>>
>>>>> If we find a better wiki software that does internalization in a better
>>>>> way, I could help porting the documentation over to the new software.
>>>>>
>>>>> The sub-domain idea is also very nice.
>>>>>
>>>>> Diego
>>>>>
>>>>>
>>>>>
>>>>> On Sat, Oct 17, 2009 at 8:29 PM, Michael Jerris <mike at jerris.com>wrote:
>>>>>
>>>>>> I think we strongly lean towards using a sub-domain (if necessary) and
>>>>>> maintaining other language content in the same wiki in alternate
>>>>>> pages.  If the wiki software we are using is not effective to create
>>>>>> multiple languages we should find a way to do it all in one.  We can
>>>>>> set up an additional mailing list if necessary but also we have no
>>>>>> issue with going across multiple languages in both the mailing list
>>>>>> and in irc.
>>>>>>
>>>>>> Mike
>>>>>>
>>>>>> On Oct 17, 2009, at 4:01 PM, Roberto Martins wrote:
>>>>>>
>>>>>> > sub-domain seem to be a very wise choice...
>>>>>> > ptbr
>>>>>> >
>>>>>> >
>>>>>> >
>>>>>> > On Oct 17, 2009, at 1:25 PM, Anthony Minessale wrote:
>>>>>> >
>>>>>> >> or someone could just ask us for pg.freeswitch.org or whatever
>>>>>> other
>>>>>> >> sub-domain to avoid trademark infringement of registering 40
>>>>>> domains
>>>>>> >> with freeswitch in the name for no reason.
>>>>>> >>
>>>>>> >>
>>>>>> >> On Sat, Oct 17, 2009 at 6:06 AM, Rud? Cunha <ruda at ruda.com.br>
>>>>>> wrote:
>>>>>> >> Jo?o Mesquita,
>>>>>> >>
>>>>>> >> Could check with Jeremiah or contact me through, so I talk to him
>>>>>> >> about the forum?
>>>>>> >>
>>>>>> >> He could redirect the domain to my hosting server and we can put
>>>>>> >> there the wiki and forums and we can communicate!
>>>>>> >>
>>>>>> >> Rud? Cunha
>>>>>> >>
>>>>>> >> 2009/10/17 Jo?o Mesquita <jmesquita at freeswitch.org>
>>>>>> >>
>>>>>> >> Just a heads up, I have talked to Jeremias from Khomp today and he
>>>>>> >> is setting up the wiki. I will personally be adding contents to the
>>>>>> >> that wiki if it ever picks up.
>>>>>> >>
>>>>>> >> Regards,
>>>>>> >>
>>>>>> >> jm
>>>>>> >>
>>>>>> >>
>>>>>> >> On Sat, Oct 17, 2009 at 12:52 AM, Jason White <jason at jasonjgw.net>
>>>>>> >> wrote:
>>>>>> >> Diego Viola <diego.viola at gmail.com> wrote:
>>>>>> >>> I'm with Moises and with the other people supporting this
>>>>>> >> initiative.
>>>>>> >>>
>>>>>> >>> I'm not Brazilian, but they should be able to do whatever they
>>>>>> >> want, after
>>>>>> >>> all, that's how open source works, if you can do it go ahead and
>>>>>> >> do it.
>>>>>> >>
>>>>>> >> Correct. We have enough of them as far as English-language fora are
>>>>>> >> concerned;
>>>>>> >> other languages are a different question altogether, though.
>>>>>> >>
>>>>>> >>
>>>>>> >> _______________________________________________
>>>>>> >> FreeSWITCH-users mailing list
>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> >> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>> >> users
>>>>>> >> http://www.freeswitch.org
>>>>>> >>
>>>>>> >>
>>>>>> >> _______________________________________________
>>>>>> >> FreeSWITCH-users mailing list
>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> >> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>> >> users
>>>>>> >> http://www.freeswitch.org
>>>>>> >>
>>>>>> >>
>>>>>> >>
>>>>>> >> _______________________________________________
>>>>>> >> FreeSWITCH-users mailing list
>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> >> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>> >> users
>>>>>> >> http://www.freeswitch.org
>>>>>> >>
>>>>>> >>
>>>>>> >>
>>>>>> >>
>>>>>> >> --
>>>>>> >> Anthony Minessale II
>>>>>> >>
>>>>>> >> FreeSWITCH http://www.freeswitch.org/
>>>>>> >> ClueCon http://www.cluecon.com/
>>>>>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>> >>
>>>>>> >> AIM: anthm
>>>>>> >> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>> >> IRC: irc.freenode.net #freeswitch
>>>>>> >>
>>>>>> >> FreeSWITCH Developer Conference
>>>>>> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>> >> iax:guest at conference.freeswitch.org/888
>>>>>> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>> >> pstn:213-799-1400
>>>>>> >> _______________________________________________
>>>>>> >> FreeSWITCH-users mailing list
>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> >> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>> >> users
>>>>>> >> http://www.freeswitch.org
>>>>>> >
>>>>>> >
>>>>>> > _______________________________________________
>>>>>> > FreeSWITCH-users mailing list
>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>> > users
>>>>>> > http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>
>>>
>>
>
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From diego.viola at gmail.com  Sat Oct 17 16:05:33 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 17 Oct 2009 23:05:33 +0000
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <86a32abc0910171601u6ab06e42xcf303301c85c57b1@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<191c3a030910170925i7aa7d4f3p9d382e0d855b7c8b@mail.gmail.com>
	<6701786D-A8AF-444E-A5A7-29E69F6E055C@gmail.com>
	<B69597BF-DA4A-440B-8CB0-08F52F33DD17@jerris.com>
	<86a32abc0910171339o6a300932v66b7f00419a0cb69@mail.gmail.com>
	<86a32abc0910171513n35d7d0echff940a16685ae1a4@mail.gmail.com>
	<86a32abc0910171521n403094bfvf27aa7c2a1c1f798@mail.gmail.com>
	<86a32abc0910171535l606eb9c6id54f7b3b5ae1b670@mail.gmail.com>
	<86a32abc0910171545l17f3f539ldba432db550c3eaa@mail.gmail.com>
	<86a32abc0910171601u6ab06e42xcf303301c85c57b1@mail.gmail.com>
Message-ID: <86a32abc0910171605y73ecb75cqfb13923ad19adc56@mail.gmail.com>

With sub-domains you'll need to start all over and it will be a pain to keep
the documentation in-sync.

With a wiki that supports real translation it's easier to add a new line
with the translation tag or whatever.

Diego

On Sat, Oct 17, 2009 at 11:01 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Take a look at this for instance:
>
> http://wiki.rubyonrails.org/start
>
> This uses DokuWiki, check the langauges option at the top (en, de, el, es,
> fr, etc) and when you click one of them it translates the actual page to the
> desired language... I'd love to have something like that.
>
> Diego
>
>
> On Sat, Oct 17, 2009 at 10:45 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> Instead of adding million of different domains for the languages, it would
>> be nice if we only keep wiki.freeswitch.org, and the wiki software
>> provides a list of languages to choose from.
>>
>> Then when we click in one of the languages we have the wiki translated to
>> the desired language.
>>
>> I believe if we go this way we will be more organized, and there is more
>> chances that things will get translated, and we wont be repeating too much
>> information here and there too.
>>
>> Just a thought... I'm open to do this work to make it happen.
>>
>> The different domains per language is a good idea too.
>>
>> Diego
>>
>>
>> On Sat, Oct 17, 2009 at 10:35 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>
>>> We could also do this for a better documentation.
>>>
>>>
>>>    1. Pick a new wiki engine that does internalization/translation
>>>    easier.
>>>    2. Set some documentation guidelines from the start.
>>>    3. Move stuff over to the new wiki.
>>>    4. Start with the translation/internalization on the new wiki.
>>>
>>>
>>> Diego
>>>
>>>
>>> On Sat, Oct 17, 2009 at 10:21 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>
>>>> Some interesting pages on the subject.
>>>>
>>>> http://en.wikipedia.org/wiki/Comparison_of_wiki_software
>>>> http://www.osnews.com/story/21865/Recommend_an_Open_Source_Wiki_Platform
>>>>
>>>>
>>>>
>>>>
>>>> On Sat, Oct 17, 2009 at 10:13 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>>
>>>>> Would you be open to test another wiki software, that supports
>>>>> internalization/translation easier, if i am open to port the entire
>>>>> documentation to the new wiki software?
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Sat, Oct 17, 2009 at 8:39 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>>>
>>>>>> If we find a better wiki software that does internalization in a
>>>>>> better way, I could help porting the documentation over to the new software.
>>>>>>
>>>>>> The sub-domain idea is also very nice.
>>>>>>
>>>>>> Diego
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Sat, Oct 17, 2009 at 8:29 PM, Michael Jerris <mike at jerris.com>wrote:
>>>>>>
>>>>>>> I think we strongly lean towards using a sub-domain (if necessary)
>>>>>>> and
>>>>>>> maintaining other language content in the same wiki in alternate
>>>>>>> pages.  If the wiki software we are using is not effective to create
>>>>>>> multiple languages we should find a way to do it all in one.  We can
>>>>>>> set up an additional mailing list if necessary but also we have no
>>>>>>> issue with going across multiple languages in both the mailing list
>>>>>>> and in irc.
>>>>>>>
>>>>>>> Mike
>>>>>>>
>>>>>>> On Oct 17, 2009, at 4:01 PM, Roberto Martins wrote:
>>>>>>>
>>>>>>> > sub-domain seem to be a very wise choice...
>>>>>>> > ptbr
>>>>>>> >
>>>>>>> >
>>>>>>> >
>>>>>>> > On Oct 17, 2009, at 1:25 PM, Anthony Minessale wrote:
>>>>>>> >
>>>>>>> >> or someone could just ask us for pg.freeswitch.org or whatever
>>>>>>> other
>>>>>>> >> sub-domain to avoid trademark infringement of registering 40
>>>>>>> domains
>>>>>>> >> with freeswitch in the name for no reason.
>>>>>>> >>
>>>>>>> >>
>>>>>>> >> On Sat, Oct 17, 2009 at 6:06 AM, Rud? Cunha <ruda at ruda.com.br>
>>>>>>> wrote:
>>>>>>> >> Jo?o Mesquita,
>>>>>>> >>
>>>>>>> >> Could check with Jeremiah or contact me through, so I talk to him
>>>>>>> >> about the forum?
>>>>>>> >>
>>>>>>> >> He could redirect the domain to my hosting server and we can put
>>>>>>> >> there the wiki and forums and we can communicate!
>>>>>>> >>
>>>>>>> >> Rud? Cunha
>>>>>>> >>
>>>>>>> >> 2009/10/17 Jo?o Mesquita <jmesquita at freeswitch.org>
>>>>>>> >>
>>>>>>> >> Just a heads up, I have talked to Jeremias from Khomp today and he
>>>>>>> >> is setting up the wiki. I will personally be adding contents to
>>>>>>> the
>>>>>>> >> that wiki if it ever picks up.
>>>>>>> >>
>>>>>>> >> Regards,
>>>>>>> >>
>>>>>>> >> jm
>>>>>>> >>
>>>>>>> >>
>>>>>>> >> On Sat, Oct 17, 2009 at 12:52 AM, Jason White <jason at jasonjgw.net
>>>>>>> >
>>>>>>> >> wrote:
>>>>>>> >> Diego Viola <diego.viola at gmail.com> wrote:
>>>>>>> >>> I'm with Moises and with the other people supporting this
>>>>>>> >> initiative.
>>>>>>> >>>
>>>>>>> >>> I'm not Brazilian, but they should be able to do whatever they
>>>>>>> >> want, after
>>>>>>> >>> all, that's how open source works, if you can do it go ahead and
>>>>>>> >> do it.
>>>>>>> >>
>>>>>>> >> Correct. We have enough of them as far as English-language fora
>>>>>>> are
>>>>>>> >> concerned;
>>>>>>> >> other languages are a different question altogether, though.
>>>>>>> >>
>>>>>>> >>
>>>>>>> >> _______________________________________________
>>>>>>> >> FreeSWITCH-users mailing list
>>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>>> >> users
>>>>>>> >> http://www.freeswitch.org
>>>>>>> >>
>>>>>>> >>
>>>>>>> >> _______________________________________________
>>>>>>> >> FreeSWITCH-users mailing list
>>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>>> >> users
>>>>>>> >> http://www.freeswitch.org
>>>>>>> >>
>>>>>>> >>
>>>>>>> >>
>>>>>>> >> _______________________________________________
>>>>>>> >> FreeSWITCH-users mailing list
>>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>>> >> users
>>>>>>> >> http://www.freeswitch.org
>>>>>>> >>
>>>>>>> >>
>>>>>>> >>
>>>>>>> >>
>>>>>>> >> --
>>>>>>> >> Anthony Minessale II
>>>>>>> >>
>>>>>>> >> FreeSWITCH http://www.freeswitch.org/
>>>>>>> >> ClueCon http://www.cluecon.com/
>>>>>>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>> >>
>>>>>>> >> AIM: anthm
>>>>>>> >> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>>> >> IRC: irc.freenode.net #freeswitch
>>>>>>> >>
>>>>>>> >> FreeSWITCH Developer Conference
>>>>>>> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>>> >> iax:guest at conference.freeswitch.org/888
>>>>>>> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>>> >> pstn:213-799-1400
>>>>>>> >> _______________________________________________
>>>>>>> >> FreeSWITCH-users mailing list
>>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>>> >> users
>>>>>>> >> http://www.freeswitch.org
>>>>>>> >
>>>>>>> >
>>>>>>> > _______________________________________________
>>>>>>> > FreeSWITCH-users mailing list
>>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> > UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>>> > users
>>>>>>> > http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>
>>>
>>
>
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From mcampbellsmith at gmail.com  Sat Oct 17 17:10:01 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sun, 18 Oct 2009 11:10:01 +1100
Subject: [Freeswitch-users] Dingaling: using a hostname instead of stun
	for rtp
In-Reply-To: <7710644E-CEA1-4D48-8A03-EEE1C18F1B44@jerris.com>
References: <33c87fa30910131502t2af24801i1c3319518b5db50b@mail.gmail.com>
	<C874E25D-6D73-49AA-8DD3-0A0022617FD2@freeswitch.org>
	<33c87fa30910152035w43e260faoc031a7a4ac949d9e@mail.gmail.com>
	<65BF398D-E3A6-4A98-83D9-D4E271070CCA@freeswitch.org>
	<7710644E-CEA1-4D48-8A03-EEE1C18F1B44@jerris.com>
Message-ID: <33c87fa30910171710t7b9038cap5dcf13259486ec13@mail.gmail.com>

Thanks for the response Mike and Brian.

Using stun is not a problem and I have it working okay now.  Is there
something different in the implementation of the stun lookup between
dingaling and sofia?

I normally use stun.freeswitch.org and sofia never had a problem.

It doesn't matter which stun server I use with dingaling (I have tried
with numerous servers) and the same thing seems to happen.  The first
stun lookup fails with a timeout (on the first inbound call), and then
if another dingaling call is tried directly, the second stun lookup
works.

I noticed a similar behaviour by issuing the stun command from fs_cli

Thanks

On Sun, Oct 18, 2009 at 8:04 AM, Michael Jerris <mike at jerris.com> wrote:
> If you don't have working stun, jingle is not going to work very
> well. ?It is a required part of the protocol. ?You need to be able to
> determine your external ports for media on each call, using a host
> name will not do this for you.
>
> Mike
>
>
> On Oct 16, 2009, at 10:48 AM, Brian West wrote:
>
>> If you setup your own stun server it wouldn't do that.... ?But the
>> hostlookup only solves half the problem .. getting the external IP vs
>> poking holes for RTP which is what stun will do.
>>
>> /b
>>
>> On Oct 15, 2009, at 10:35 PM, Mark Campbell-Smith wrote:
>>
>>> Thanks Brian. ?Is this something that is planned to be implemented?
>>> The workaround is to set the stun server also in the dingaling
>>> configuration, but as I said, for some reason the stun times for me
>>> out occasionally with dingaling.
>>>
>>> Thanks!
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From diego.viola at gmail.com  Sat Oct 17 17:11:40 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 18 Oct 2009 00:11:40 +0000
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <86a32abc0910171605y73ecb75cqfb13923ad19adc56@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<6701786D-A8AF-444E-A5A7-29E69F6E055C@gmail.com>
	<B69597BF-DA4A-440B-8CB0-08F52F33DD17@jerris.com>
	<86a32abc0910171339o6a300932v66b7f00419a0cb69@mail.gmail.com>
	<86a32abc0910171513n35d7d0echff940a16685ae1a4@mail.gmail.com>
	<86a32abc0910171521n403094bfvf27aa7c2a1c1f798@mail.gmail.com>
	<86a32abc0910171535l606eb9c6id54f7b3b5ae1b670@mail.gmail.com>
	<86a32abc0910171545l17f3f539ldba432db550c3eaa@mail.gmail.com>
	<86a32abc0910171601u6ab06e42xcf303301c85c57b1@mail.gmail.com>
	<86a32abc0910171605y73ecb75cqfb13923ad19adc56@mail.gmail.com>
Message-ID: <86a32abc0910171711je1e57bduf2bb672472365ce1@mail.gmail.com>

http://moinmo.in/ is also pretty cool it seems and it's used for the apache
wiki and others.

http://wiki.apache.org/

It also has i18n support, full text search, etc.

Diego

On Sat, Oct 17, 2009 at 11:05 PM, Diego Viola <diego.viola at gmail.com> wrote:

> With sub-domains you'll need to start all over and it will be a pain to
> keep the documentation in-sync.
>
> With a wiki that supports real translation it's easier to add a new line
> with the translation tag or whatever.
>
> Diego
>
>
> On Sat, Oct 17, 2009 at 11:01 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> Take a look at this for instance:
>>
>> http://wiki.rubyonrails.org/start
>>
>> This uses DokuWiki, check the langauges option at the top (en, de, el, es,
>> fr, etc) and when you click one of them it translates the actual page to the
>> desired language... I'd love to have something like that.
>>
>> Diego
>>
>>
>> On Sat, Oct 17, 2009 at 10:45 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>
>>> Instead of adding million of different domains for the languages, it
>>> would be nice if we only keep wiki.freeswitch.org, and the wiki software
>>> provides a list of languages to choose from.
>>>
>>> Then when we click in one of the languages we have the wiki translated to
>>> the desired language.
>>>
>>> I believe if we go this way we will be more organized, and there is more
>>> chances that things will get translated, and we wont be repeating too much
>>> information here and there too.
>>>
>>> Just a thought... I'm open to do this work to make it happen.
>>>
>>> The different domains per language is a good idea too.
>>>
>>> Diego
>>>
>>>
>>> On Sat, Oct 17, 2009 at 10:35 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>
>>>> We could also do this for a better documentation.
>>>>
>>>>
>>>>    1. Pick a new wiki engine that does internalization/translation
>>>>    easier.
>>>>    2. Set some documentation guidelines from the start.
>>>>    3. Move stuff over to the new wiki.
>>>>    4. Start with the translation/internalization on the new wiki.
>>>>
>>>>
>>>> Diego
>>>>
>>>>
>>>> On Sat, Oct 17, 2009 at 10:21 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>>
>>>>> Some interesting pages on the subject.
>>>>>
>>>>> http://en.wikipedia.org/wiki/Comparison_of_wiki_software
>>>>>
>>>>> http://www.osnews.com/story/21865/Recommend_an_Open_Source_Wiki_Platform
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Sat, Oct 17, 2009 at 10:13 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>>>
>>>>>> Would you be open to test another wiki software, that supports
>>>>>> internalization/translation easier, if i am open to port the entire
>>>>>> documentation to the new wiki software?
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Sat, Oct 17, 2009 at 8:39 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>>>>
>>>>>>> If we find a better wiki software that does internalization in a
>>>>>>> better way, I could help porting the documentation over to the new software.
>>>>>>>
>>>>>>> The sub-domain idea is also very nice.
>>>>>>>
>>>>>>> Diego
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Sat, Oct 17, 2009 at 8:29 PM, Michael Jerris <mike at jerris.com>wrote:
>>>>>>>
>>>>>>>> I think we strongly lean towards using a sub-domain (if necessary)
>>>>>>>> and
>>>>>>>> maintaining other language content in the same wiki in alternate
>>>>>>>> pages.  If the wiki software we are using is not effective to create
>>>>>>>> multiple languages we should find a way to do it all in one.  We can
>>>>>>>> set up an additional mailing list if necessary but also we have no
>>>>>>>> issue with going across multiple languages in both the mailing list
>>>>>>>> and in irc.
>>>>>>>>
>>>>>>>> Mike
>>>>>>>>
>>>>>>>> On Oct 17, 2009, at 4:01 PM, Roberto Martins wrote:
>>>>>>>>
>>>>>>>> > sub-domain seem to be a very wise choice...
>>>>>>>> > ptbr
>>>>>>>> >
>>>>>>>> >
>>>>>>>> >
>>>>>>>> > On Oct 17, 2009, at 1:25 PM, Anthony Minessale wrote:
>>>>>>>> >
>>>>>>>> >> or someone could just ask us for pg.freeswitch.org or whatever
>>>>>>>> other
>>>>>>>> >> sub-domain to avoid trademark infringement of registering 40
>>>>>>>> domains
>>>>>>>> >> with freeswitch in the name for no reason.
>>>>>>>> >>
>>>>>>>> >>
>>>>>>>> >> On Sat, Oct 17, 2009 at 6:06 AM, Rud? Cunha <ruda at ruda.com.br>
>>>>>>>> wrote:
>>>>>>>> >> Jo?o Mesquita,
>>>>>>>> >>
>>>>>>>> >> Could check with Jeremiah or contact me through, so I talk to him
>>>>>>>> >> about the forum?
>>>>>>>> >>
>>>>>>>> >> He could redirect the domain to my hosting server and we can put
>>>>>>>> >> there the wiki and forums and we can communicate!
>>>>>>>> >>
>>>>>>>> >> Rud? Cunha
>>>>>>>> >>
>>>>>>>> >> 2009/10/17 Jo?o Mesquita <jmesquita at freeswitch.org>
>>>>>>>> >>
>>>>>>>> >> Just a heads up, I have talked to Jeremias from Khomp today and
>>>>>>>> he
>>>>>>>> >> is setting up the wiki. I will personally be adding contents to
>>>>>>>> the
>>>>>>>> >> that wiki if it ever picks up.
>>>>>>>> >>
>>>>>>>> >> Regards,
>>>>>>>> >>
>>>>>>>> >> jm
>>>>>>>> >>
>>>>>>>> >>
>>>>>>>> >> On Sat, Oct 17, 2009 at 12:52 AM, Jason White <
>>>>>>>> jason at jasonjgw.net>
>>>>>>>> >> wrote:
>>>>>>>> >> Diego Viola <diego.viola at gmail.com> wrote:
>>>>>>>> >>> I'm with Moises and with the other people supporting this
>>>>>>>> >> initiative.
>>>>>>>> >>>
>>>>>>>> >>> I'm not Brazilian, but they should be able to do whatever they
>>>>>>>> >> want, after
>>>>>>>> >>> all, that's how open source works, if you can do it go ahead and
>>>>>>>> >> do it.
>>>>>>>> >>
>>>>>>>> >> Correct. We have enough of them as far as English-language fora
>>>>>>>> are
>>>>>>>> >> concerned;
>>>>>>>> >> other languages are a different question altogether, though.
>>>>>>>> >>
>>>>>>>> >>
>>>>>>>> >> _______________________________________________
>>>>>>>> >> FreeSWITCH-users mailing list
>>>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>>>> >> users
>>>>>>>> >> http://www.freeswitch.org
>>>>>>>> >>
>>>>>>>> >>
>>>>>>>> >> _______________________________________________
>>>>>>>> >> FreeSWITCH-users mailing list
>>>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>>>> >> users
>>>>>>>> >> http://www.freeswitch.org
>>>>>>>> >>
>>>>>>>> >>
>>>>>>>> >>
>>>>>>>> >> _______________________________________________
>>>>>>>> >> FreeSWITCH-users mailing list
>>>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>>>> >> users
>>>>>>>> >> http://www.freeswitch.org
>>>>>>>> >>
>>>>>>>> >>
>>>>>>>> >>
>>>>>>>> >>
>>>>>>>> >> --
>>>>>>>> >> Anthony Minessale II
>>>>>>>> >>
>>>>>>>> >> FreeSWITCH http://www.freeswitch.org/
>>>>>>>> >> ClueCon http://www.cluecon.com/
>>>>>>>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>>> >>
>>>>>>>> >> AIM: anthm
>>>>>>>> >> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>>>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>>>> >> IRC: irc.freenode.net #freeswitch
>>>>>>>> >>
>>>>>>>> >> FreeSWITCH Developer Conference
>>>>>>>> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>>>> >> iax:guest at conference.freeswitch.org/888
>>>>>>>> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>>>> >> pstn:213-799-1400
>>>>>>>> >> _______________________________________________
>>>>>>>> >> FreeSWITCH-users mailing list
>>>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>>>> >> users
>>>>>>>> >> http://www.freeswitch.org
>>>>>>>> >
>>>>>>>> >
>>>>>>>> > _______________________________________________
>>>>>>>> > FreeSWITCH-users mailing list
>>>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> > UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>>>> > users
>>>>>>>> > http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>
>>>
>>
>
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From wiltingtree at gmail.com  Sat Oct 17 17:13:59 2009
From: wiltingtree at gmail.com (Adam Wilt)
Date: Sat, 17 Oct 2009 20:13:59 -0400
Subject: [Freeswitch-users] Problem connecting to Spidermonkey ODBC
Message-ID: <eb7797d70910171713wc3a064fi6231745381bb2d1b@mail.gmail.com>

Hi, I know there are a lot of answered questions about this already, but I
think I read all of the answers and I'm still stuck.
* I'm using FreeSWITCH Version 1.0.trunk (14961).
* I installed UnixODBC
* My odbc.ini and odbcinst.ini files are in /usr/local/etc/
* I have symlinks to /usr/local/etc from /usr/etc and from
/usr/local/freeswitch/etc/
* Im running freeswitch as root, and I copied odbc.ini to /root/.odbc.ini
* I have mod_spidermonkey_odbc uncommented in spidermonkey.conf.xml
* When I run "isql dbname" from the command prompt it works fine.

But when I try to run my script, I get:

2009-10-17 23:56:21.848930 [ERR] switch_odbc.c:188 STATE: IM002 CODE 0
ERROR: [unixODBC][Driver Manager]Data source name not found, and no default
driver specified

2009-10-17 23:56:21.848930 [ERR] mod_spidermonkey_odbc.c:245 Database is not
connected!


Please help!
Thanks.
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From brian at freeswitch.org  Sat Oct 17 17:56:36 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 17 Oct 2009 19:56:36 -0500
Subject: [Freeswitch-users] Problem connecting to Spidermonkey ODBC
In-Reply-To: <eb7797d70910171713wc3a064fi6231745381bb2d1b@mail.gmail.com>
References: <eb7797d70910171713wc3a064fi6231745381bb2d1b@mail.gmail.com>
Message-ID: <3BEC6702-4754-41BB-B458-835250ECD897@freeswitch.org>

How are you using this from your javascript code?

/b

On Oct 17, 2009, at 7:13 PM, Adam Wilt wrote:

>
> 2009-10-17 23:56:21.848930 [ERR] switch_odbc.c:188 STATE: IM002 CODE  
> 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and  
> no default driver specified
>
> 2009-10-17 23:56:21.848930 [ERR] mod_spidermonkey_odbc.c:245  
> Database is not connected!
>




From brian at freeswitch.org  Sat Oct 17 18:01:09 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 17 Oct 2009 20:01:09 -0500
Subject: [Freeswitch-users] Dingaling: using a hostname instead of stun
	for rtp
In-Reply-To: <33c87fa30910171710t7b9038cap5dcf13259486ec13@mail.gmail.com>
References: <33c87fa30910131502t2af24801i1c3319518b5db50b@mail.gmail.com>
	<C874E25D-6D73-49AA-8DD3-0A0022617FD2@freeswitch.org>
	<33c87fa30910152035w43e260faoc031a7a4ac949d9e@mail.gmail.com>
	<65BF398D-E3A6-4A98-83D9-D4E271070CCA@freeswitch.org>
	<7710644E-CEA1-4D48-8A03-EEE1C18F1B44@jerris.com>
	<33c87fa30910171710t7b9038cap5dcf13259486ec13@mail.gmail.com>
Message-ID: <9B8E1ADF-B0EB-495F-8D27-A9D1211A4D2C@freeswitch.org>

They all use the same core API.  I suspect since our stun server is  
just a cname to many public services out there you're just thinking  
its dingaling related but its really not...

/b

On Oct 17, 2009, at 7:10 PM, Mark Campbell-Smith wrote:

> I noticed a similar behaviour by issuing the stun command from fs_cli
>
> Thanks




From wiltingtree at gmail.com  Sat Oct 17 18:11:27 2009
From: wiltingtree at gmail.com (Adam Wilt)
Date: Sat, 17 Oct 2009 21:11:27 -0400
Subject: [Freeswitch-users] Problem connecting to Spidermonkey ODBC
In-Reply-To: <3BEC6702-4754-41BB-B458-835250ECD897@freeswitch.org>
References: <eb7797d70910171713wc3a064fi6231745381bb2d1b@mail.gmail.com>
	<3BEC6702-4754-41BB-B458-835250ECD897@freeswitch.org>
Message-ID: <eb7797d70910171811y57f2d29q2360de42c0ad3c81@mail.gmail.com>

Brian,

Here's how I'm using the ODBC in my .js file:

use("ODBC");

var DSN="mydsn";
var DB_USER="myuser";
var DB_PASS="mypass";
.
.
.
var db = new ODBC(DSN, DB_USER, DB_PASS);
db.connect();
.
.
.

Thanks.


On Sat, Oct 17, 2009 at 8:56 PM, Brian West <brian at freeswitch.org> wrote:

> How are you using this from your javascript code?
>
> /b
>
> On Oct 17, 2009, at 7:13 PM, Adam Wilt wrote:
>
> >
> > 2009-10-17 23:56:21.848930 [ERR] switch_odbc.c:188 STATE: IM002 CODE
> > 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and
> > no default driver specified
> >
> > 2009-10-17 23:56:21.848930 [ERR] mod_spidermonkey_odbc.c:245
> > Database is not connected!
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mcampbellsmith at gmail.com  Sat Oct 17 20:07:50 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sun, 18 Oct 2009 14:07:50 +1100
Subject: [Freeswitch-users] Dingaling / Jingle DTMF support?
In-Reply-To: <191c3a030910161456y65d49120p8479c7c5df675125@mail.gmail.com>
References: <33c87fa30910151519x637618d3p109f1b8edbf3432e@mail.gmail.com>
	<191c3a030910161456y65d49120p8479c7c5df675125@mail.gmail.com>
Message-ID: <33c87fa30910172007p57eec112qb03a9f31c763b139@mail.gmail.com>

Perfect... thanks for the + tip with googletalk...   I had been using
gtalk2voip but didn't want to rely on another service

Wki for dingaling updated.


On Sat, Oct 17, 2009 at 8:56 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> It already should support at least recv of rfc2833 any time.
> as a workaround in googletalk any string that starts with + typed in the
> chat box is treated as dtmf by FS
> e.g. +1
>
> Once the jingle spec stops being a moving target we will re-investigate
> making sure its supported properly.
>
>
> On Thu, Oct 15, 2009 at 5:19 PM, Mark Campbell-Smith
> <mcampbellsmith at gmail.com> wrote:
>>
>> Hi!
>>
>> I was wondering if the Dingaling implementation in FS supports DTMF?
>> This is now supported in the jingle specs (
>> http://www.jabberforum.org/showthread.php?t=2709 ), even though Google
>> Talk client does not currently support DTMF.
>>
>> If not, are there plans to implement this?
>>
>> Thanks!
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From wiltingtree at gmail.com  Sat Oct 17 20:16:05 2009
From: wiltingtree at gmail.com (Adam Wilt)
Date: Sat, 17 Oct 2009 23:16:05 -0400
Subject: [Freeswitch-users] Problem connecting to Spidermonkey ODBC
In-Reply-To: <eb7797d70910171811y57f2d29q2360de42c0ad3c81@mail.gmail.com>
References: <eb7797d70910171713wc3a064fi6231745381bb2d1b@mail.gmail.com>
	<3BEC6702-4754-41BB-B458-835250ECD897@freeswitch.org>
	<eb7797d70910171811y57f2d29q2360de42c0ad3c81@mail.gmail.com>
Message-ID: <eb7797d70910172016i37c6c1c0tfe2464c60e06d62d@mail.gmail.com>

So, where is FreeSWITCH looking for the odbc.ini file?
I keep reading conflicting information, including
/usr/local/freeswitch/etc/,  /usr/local/etc,  /usr/etc,  /home/username/



On Sat, Oct 17, 2009 at 9:11 PM, Adam Wilt <wiltingtree at gmail.com> wrote:

> Brian,
>
> Here's how I'm using the ODBC in my .js file:
>
> use("ODBC");
>
> var DSN="mydsn";
> var DB_USER="myuser";
> var DB_PASS="mypass";
> .
> .
> .
> var db = new ODBC(DSN, DB_USER, DB_PASS);
> db.connect();
> .
> .
> .
>
> Thanks.
>
>
> On Sat, Oct 17, 2009 at 8:56 PM, Brian West <brian at freeswitch.org> wrote:
>
>> How are you using this from your javascript code?
>>
>> /b
>>
>> On Oct 17, 2009, at 7:13 PM, Adam Wilt wrote:
>>
>> >
>> > 2009-10-17 23:56:21.848930 [ERR] switch_odbc.c:188 STATE: IM002 CODE
>> > 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and
>> > no default driver specified
>> >
>> > 2009-10-17 23:56:21.848930 [ERR] mod_spidermonkey_odbc.c:245
>> > Database is not connected!
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From brian at freeswitch.org  Sat Oct 17 20:26:58 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 17 Oct 2009 22:26:58 -0500
Subject: [Freeswitch-users] Problem connecting to Spidermonkey ODBC
In-Reply-To: <eb7797d70910172016i37c6c1c0tfe2464c60e06d62d@mail.gmail.com>
References: <eb7797d70910171713wc3a064fi6231745381bb2d1b@mail.gmail.com>
	<3BEC6702-4754-41BB-B458-835250ECD897@freeswitch.org>
	<eb7797d70910171811y57f2d29q2360de42c0ad3c81@mail.gmail.com>
	<eb7797d70910172016i37c6c1c0tfe2464c60e06d62d@mail.gmail.com>
Message-ID: <4CAB8EB3-6D37-448D-8B33-59C84AB685C3@freeswitch.org>

Its in /etc/ or where ever your default is.. is FS running as root?

/b

On Oct 17, 2009, at 10:16 PM, Adam Wilt wrote:

> So, where is FreeSWITCH looking for the odbc.ini file?
> I keep reading conflicting information, including /usr/local/ 
> freeswitch/etc/,  /usr/local/etc,  /usr/etc,  /home/username/
>
>




From wiltingtree at gmail.com  Sat Oct 17 20:38:45 2009
From: wiltingtree at gmail.com (Adam Wilt)
Date: Sat, 17 Oct 2009 23:38:45 -0400
Subject: [Freeswitch-users] Problem connecting to Spidermonkey ODBC
In-Reply-To: <4CAB8EB3-6D37-448D-8B33-59C84AB685C3@freeswitch.org>
References: <eb7797d70910171713wc3a064fi6231745381bb2d1b@mail.gmail.com>
	<3BEC6702-4754-41BB-B458-835250ECD897@freeswitch.org>
	<eb7797d70910171811y57f2d29q2360de42c0ad3c81@mail.gmail.com>
	<eb7797d70910172016i37c6c1c0tfe2464c60e06d62d@mail.gmail.com>
	<4CAB8EB3-6D37-448D-8B33-59C84AB685C3@freeswitch.org>
Message-ID: <eb7797d70910172038v4e1f7da2ge36679974f010cac@mail.gmail.com>

Yes, I have FreeSWITCH running as root.  So the odbc.ini file should be in
/etc/ ?
How do you set the default?


On Sat, Oct 17, 2009 at 11:26 PM, Brian West <brian at freeswitch.org> wrote:

> Its in /etc/ or where ever your default is.. is FS running as root?
>
> /b
>
> On Oct 17, 2009, at 10:16 PM, Adam Wilt wrote:
>
> > So, where is FreeSWITCH looking for the odbc.ini file?
> > I keep reading conflicting information, including /usr/local/
> > freeswitch/etc/,  /usr/local/etc,  /usr/etc,  /home/username/
> >
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From wiltingtree at gmail.com  Sat Oct 17 20:46:04 2009
From: wiltingtree at gmail.com (Adam Wilt)
Date: Sat, 17 Oct 2009 23:46:04 -0400
Subject: [Freeswitch-users] Problem connecting to Spidermonkey ODBC
In-Reply-To: <4CAB8EB3-6D37-448D-8B33-59C84AB685C3@freeswitch.org>
References: <eb7797d70910171713wc3a064fi6231745381bb2d1b@mail.gmail.com>
	<3BEC6702-4754-41BB-B458-835250ECD897@freeswitch.org>
	<eb7797d70910171811y57f2d29q2360de42c0ad3c81@mail.gmail.com>
	<eb7797d70910172016i37c6c1c0tfe2464c60e06d62d@mail.gmail.com>
	<4CAB8EB3-6D37-448D-8B33-59C84AB685C3@freeswitch.org>
Message-ID: <eb7797d70910172046u6b2435dcqd2fc833bdc8a9b29@mail.gmail.com>

Brian,
Putting the odbc.ini and odbcinst.ini in /etc/ fixed the problem.  Thanks.

Thanks,
Marc


On Sat, Oct 17, 2009 at 11:26 PM, Brian West <brian at freeswitch.org> wrote:

> Its in /etc/ or where ever your default is.. is FS running as root?
>
> /b
>
> On Oct 17, 2009, at 10:16 PM, Adam Wilt wrote:
>
> > So, where is FreeSWITCH looking for the odbc.ini file?
> > I keep reading conflicting information, including /usr/local/
> > freeswitch/etc/,  /usr/local/etc,  /usr/etc,  /home/username/
> >
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From max.bridgewater at gmail.com  Sat Oct 17 20:47:55 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sat, 17 Oct 2009 23:47:55 -0400
Subject: [Freeswitch-users] Detecting Forwarding Device Caller ID (Off Topic)
Message-ID: <c55d46950910172047y6de63448h656a1d889077b8f5@mail.gmail.com>

Hey Guys,

Pleas help me answer the following question. Sorry for being a bit
off-topic.  I have a GSM-VoiP modem with just 4 SIM Cards. Is there a known
mechanism that allows me to use this modem for more than 4 (albeit not at
the same time) GSM numbers?  Let's say the phone numbers of the SIM cards
inserted in the modem are call real phone numbers. My goal is to have some
10 additional virtual phone numbers served by this very same modem.

One way i thought this could be done is by forwarding the virtual phone
numbers the 4 real phone numbers. Then, when somebody calls one of the
virtual numbers, the Telco will forward it to the real phone number and it
will end on Freeswitch box. My only problem is now that on the Freeswitch
box I have no idea what virtual phone number was actually dialed. Any idea
how to recover this phone number? I looked at the GSM specification and
couldn't find a way.

PS. Purchasing DIDs won't solve my underlying problem.

Thanks,
Max.
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From dmitry.bely at gmail.com  Sun Oct 18 02:49:33 2009
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Sun, 18 Oct 2009 13:49:33 +0400
Subject: [Freeswitch-users] Status of ubuntu/debian packages.
In-Reply-To: <4ABA47FB.2050100@gmail.com>
References: <4ABA47FB.2050100@gmail.com>
Message-ID: <90823c940910180249u69f18aaamc383f9a5a0c65848@mail.gmail.com>

William,

When I upgrade freeswitch-config package, all my customized config
files are silently overwritten with the default ones from .deb
package. This bug is quite annoying.

- Dmitry Bely



From mattdfong at gmail.com  Sun Oct 18 07:25:10 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Sun, 18 Oct 2009 21:25:10 +0700
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
In-Reply-To: <3D0B855A-5B2C-4C6F-9306-64B0ECE97091@jerris.com>
References: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
	<177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>
	<4256bf830908041151l55980e3fifb85c75b87535426@mail.gmail.com>
	<3D0B855A-5B2C-4C6F-9306-64B0ECE97091@jerris.com>
Message-ID: <4256bf830910180725y72539d98gb138934afc18d58b@mail.gmail.com>

I'm still having this issue with random EXCHANGE_ROUTING_ERROR disconnects
for users utilizing my inbound DID to connect to my FreeSWITCH server. It's
a predictive dialing application, with one agent session being bridged with
multiple calls and transfered back and forth between extensions in my dial
plan. After a random number of bridging and transferring, FreeSWITCH
suddenly sends a BYE to my DID provider citing an EXCHANGE_ROUTING_ERROR. It
does not happen at any one-point in my dial plan, or applications--it just
randomly disconnects when a call that the Agent is bridged to hangs-up or is
disconnected. It seems to only happen when two external sip profiles are
being bridged together, and not when an internal and external profile is
being bridged.
I turned

sip trace on and
sofia loglevel all 9

below is the the snippet. I've posted the entire Agent session at the
following pastebin http://pastebin.freeswitch.org/10756

tport_deliver(0x18413c0): msg 0x7faeb818ea30 (304 bytes) from udp/
208.76.18.254:5080/sip next=(nil)
nta: received 200 OK for BYE (121818983)
nta: 200 OK is going to a transaction
nta_outgoing: RTT is 84.409 ms
tport_release(0x18413c0): 0x1a15cc0 by 0x1a16a00 with 0x7faeb818ea30
nua(0x1ad6fb0): event r_bye 200 OK
nua(0x1ad6fb0): call state changed: terminating -> terminated
nua(0x1ad6fb0): event i_state 200 to BYE
nua: nua_application_event: entering
nua(0x1ad6fb0): event i_terminated 200 to BYE
nua: nua_handle_magic: entering
nua(0x1ad6fb0): removing session usage
soa_destroy(static::0x1b5ae90) called
nua: nua_application_event: entering
nta_leg_destroy(0x1b594a0)
nua: nua_handle_magic: entering
nua: nua_handle_bind: entering
nua: nua_application_event: entering
nua: nua_handle_magic: entering
nua: terminated session 0x1ad6fb0
nua: nua_handle_destroy: entering
nua(0x1ad6fb0): recv signal r_destroy
nta_leg_destroy((nil))
nua(0x1ad6fb0): sent signal r_destroy
nta: timer set next to 28 ms
nta: timer E fired, retransmit BYE (121818989)
tport_release(0x18413c0): 0x1b5c9b0 by 0x7faeb817d830 with (nil)
tport_tsend(0x18413c0) tpn = */209.216.2.211:5060
tport_resolve addrinfo = 209.216.2.211:5060
tport_by_addrinfo(0x18413c0): not found by name */209.216.2.211:5060
tport_vsend(0x18413c0): 862 bytes of 862 to udp/209.216.2.211:5060
tport_vsend returned 862
send 862 bytes to udp/[209.216.2.211]:5060 at 14:04:11.753690:
   ------------------------------------------------------------------------
   BYE sip:199.173.100.16:5060;transport=UDP SIP/2.0
   Via: SIP/2.0/UDP 67.220.216.146:5080;rport;branch=z9hG4bK02jNX8a4HrNyN
   Route: <sip:209.216.2.211;ftag=4adc7-13c4-1ab03-71ce3705-1ab03;lr=on>
   Route: <sip:65.211.120.237:5060;lr>
   Route: <sip:63.110.102.239;lr>
   Max-Forwards: 70
   From: <sip:+12133304391 at 63.110.102.239:5060;user=phone>;tag=cgBe054jZrt3a
   To: <sip:+14158867717 at 199.173.100.16:5060
;user=phone>;tag=4adc7-13c4-1ab03-71ce3705-1ab03
   Call-ID: a0f656a01064adc713c41ab036840746ee20ca11c06b2d8-0440-5124
   CSeq: 121818989 BYE
   Contact: <sip:+12133304391 at 67.220.216.146:5080;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15135
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER,
UPDATE, NOTIFY
   Supported: timer, precondition, path, replaces
   Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
   Content-Length: 0

   ------------------------------------------------------------------------

Thanks.
--matt

On Sat, Aug 8, 2009 at 12:42 PM, Michael Jerris <mike at jerris.com> wrote:

> http://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP
> turn the logging all the way up and see what it says.
>
> Mike
>
> On Aug 4, 2009, at 2:51 PM, Matthew Fong wrote:
>
> Hi Mathieu, thanks for the reply. I enabled sip trace logging and got the
> logs below, but I am still at a loss at being able to identify the error or
> reproduce it consistently. The below log indicates to me that my FS server
> is initiating sending 2 BYE message to my DID provider (didforsale.com).
> Is there a way I can look further inside FreeSWITCH to see why it is sending
> this BYE packet?
>
> sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:44.679208:
> BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0
> Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD
> Route: <sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on>
> Route: <sip:65.217.40.210:5060;lr>
> Route: <sip:63.110.102.238;lr>
> Max-Forwards: 70
> From: <sip:+1212381XXXX at 63.110.102.238:5060;user=phone>;tag=Ztr5ycrv3QZ1g
> To: <sip:+1909635XXXX at 199.173.100.144:5060
> ;user=phone>;tag=dc7-13c4-2401b7-46dea593-2401b7
> Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441
> CSeq: 118584736 BYE
> Contact: <sip:+1212381XXXX at 66.197.142.69:5080;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO
> Supported: timer, precondition, path, replaces
> Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
> Content-Length: 0
>
>
> sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:45.182589:
> BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0
> Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD
> Route: <sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on>
> Route: <sip:65.217.40.210:5060;lr>
> Route: <sip:63.110.102.238;lr>
> Max-Forwards: 70
> From: <sip:+1212381XXXX at 63.110.102.238:5060;user=phone>;tag=Ztr5ycrv3QZ1g
> To: <sip:+1909635XXXX at 199.173.100.144:5060
> ;user=phone>;tag=dc7-13c4-2401b7-46dea593-2401b7
> Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441
> CSeq: 118584736 BYE
> Contact: <sip:+1212381XXXX at 66.197.142.69:5080;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO
> Supported: timer, precondition, path, replaces
> Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
> Content-Length: 0
>
> On Sun, Aug 2, 2009 at 11:20 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
>
>> Hi,
>>
>> Digging a bit in mod_sofia releaved that it can be caused by a SIP
>> code 482 (loop detected), 483 (too many hops) or 484 (address
>> incomplete).
>>
>> Do a SIP trace to sched more light on what's happening.
>>
>> Mathieu Rene
>> Avant-Garde Solutions Inc
>> Office: + 1 (514) 664-1044 x100
>> Cell: +1 (514) 664-1044 x200
>> mrene at avgs.ca
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From juanbackson at gmail.com  Sun Oct 18 08:29:45 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Sun, 18 Oct 2009 23:29:45 +0800
Subject: [Freeswitch-users] anyway to not to retry
Message-ID: <27c25bc40910180829y6ced15d7hdb75e7dda8daf04c@mail.gmail.com>

Hi,

Is there anyway to configure freeswitch so that it won't retry the SIP
message when 200 OK is not received?

thanks,
jb
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From dave at 3c.co.uk  Sun Oct 18 09:04:32 2009
From: dave at 3c.co.uk (David Knell)
Date: Sun, 18 Oct 2009 09:04:32 -0700
Subject: [Freeswitch-users] Detecting Forwarding Device Caller ID
	(Off	Topic)
In-Reply-To: <c55d46950910172047y6de63448h656a1d889077b8f5@mail.gmail.com>
References: <c55d46950910172047y6de63448h656a1d889077b8f5@mail.gmail.com>
Message-ID: <1255881872.5374.8.camel@localhost.localdomain>

Hi Max,

Some operators (e.g. Orange in the UK) will allow you to have two
numbers attached to one SIM, which might get you 4/10ths of the way to
your goal.

I'm struggling to see why purchasing DIDs won't help, unless your FS box
has no internet connectivity.  A bit more background on the problem
might help with finding a solution?

--Dave

> Hey Guys,
> 
> Pleas help me answer the following question. Sorry for being a bit
> off-topic.  I have a GSM-VoiP modem with just 4 SIM Cards. Is there a
> known mechanism that allows me to use this modem for more than 4
> (albeit not at the same time) GSM numbers?  Let's say the phone
> numbers of the SIM cards inserted in the modem are call real phone
> numbers. My goal is to have some 10 additional virtual phone numbers
> served by this very same modem.
> 
> One way i thought this could be done is by forwarding the virtual
> phone numbers the 4 real phone numbers. Then, when somebody calls one
> of the virtual numbers, the Telco will forward it to the real phone
> number and it will end on Freeswitch box. My only problem is now that
> on the Freeswitch box I have no idea what virtual phone number was
> actually dialed. Any idea how to recover this phone number? I looked
> at the GSM specification and couldn't find a way.
> 
> PS. Purchasing DIDs won't solve my underlying problem.
> 
> Thanks,
> Max.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From frank at carmickle.com  Sun Oct 18 09:40:00 2009
From: frank at carmickle.com (Frank Carmickle)
Date: Sun, 18 Oct 2009 12:40:00 -0400
Subject: [Freeswitch-users] Status of ubuntu/debian packages.
In-Reply-To: <90823c940910180249u69f18aaamc383f9a5a0c65848@mail.gmail.com>
References: <4ABA47FB.2050100@gmail.com>
	<90823c940910180249u69f18aaamc383f9a5a0c65848@mail.gmail.com>
Message-ID: <20091018164000.GD13895@base.carmickle.com>

On Sun, Oct 18, Dmitry Bely wrote:
> William,
> 
> When I upgrade freeswitch-config package, all my customized config
> files are silently overwritten with the default ones from .deb
> package. This bug is quite annoying.

Very sorry.  I can try to include some code that detects if your moving from the old versions that don't have a config package to the new stuff that does.  My way around this is not to install the freeswitch-config package at all.  I extract it some where else and diff my way through it.

--FC



From max.bridgewater at gmail.com  Sun Oct 18 09:51:04 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sun, 18 Oct 2009 12:51:04 -0400
Subject: [Freeswitch-users] Detecting Forwarding Device Caller ID (Off
	Topic)
In-Reply-To: <1255881872.5374.8.camel@localhost.localdomain>
References: <c55d46950910172047y6de63448h656a1d889077b8f5@mail.gmail.com>
	<1255881872.5374.8.camel@localhost.localdomain>
Message-ID: <c55d46950910180951t25ad668et385f6f3d78ce69e9@mail.gmail.com>

Thanks Dave. I'm essentially trying to build a service in one country where
there is no DID provider. If DIDs were available for such countries, that
would have done the job.

Using the two-numbers-per-SIM approach  would help significantly, if I could
only figure out, how to detect the Called Number using AT commands or
Unsolicited Result codes. I was hoping the CLIP code could give me this
information, but I'm still struggling to if it does or not. Any idea?

Max.

On Sun, Oct 18, 2009 at 12:04 PM, David Knell <dave at 3c.co.uk> wrote:

> Hi Max,
>
> Some operators (e.g. Orange in the UK) will allow you to have two
> numbers attached to one SIM, which might get you 4/10ths of the way to
> your goal.
>
> I'm struggling to see why purchasing DIDs won't help, unless your FS box
> has no internet connectivity.  A bit more background on the problem
> might help with finding a solution?
>
> --Dave
>
> > Hey Guys,
> >
> > Pleas help me answer the following question. Sorry for being a bit
> > off-topic.  I have a GSM-VoiP modem with just 4 SIM Cards. Is there a
> > known mechanism that allows me to use this modem for more than 4
> > (albeit not at the same time) GSM numbers?  Let's say the phone
> > numbers of the SIM cards inserted in the modem are call real phone
> > numbers. My goal is to have some 10 additional virtual phone numbers
> > served by this very same modem.
> >
> > One way i thought this could be done is by forwarding the virtual
> > phone numbers the 4 real phone numbers. Then, when somebody calls one
> > of the virtual numbers, the Telco will forward it to the real phone
> > number and it will end on Freeswitch box. My only problem is now that
> > on the Freeswitch box I have no idea what virtual phone number was
> > actually dialed. Any idea how to recover this phone number? I looked
> > at the GSM specification and couldn't find a way.
> >
> > PS. Purchasing DIDs won't solve my underlying problem.
> >
> > Thanks,
> > Max.
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sun Oct 18 11:43:25 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 18 Oct 2009 13:43:25 -0500
Subject: [Freeswitch-users] anyway to not to retry
In-Reply-To: <27c25bc40910180829y6ced15d7hdb75e7dda8daf04c@mail.gmail.com>
References: <27c25bc40910180829y6ced15d7hdb75e7dda8daf04c@mail.gmail.com>
Message-ID: <705F9C29-44F8-4E5E-93E1-0B234DAEECF4@freeswitch.org>

Nope its part of SIP it'll retry if a response isn't received in a  
certain time.

/b

On Oct 18, 2009, at 10:29 AM, Juan Backson wrote:

> Hi,
>
> Is there anyway to configure freeswitch so that it won't retry the  
> SIP message when 200 OK is not received?
>
> thanks,
> jb




From dmitry.bely at gmail.com  Sun Oct 18 11:54:02 2009
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Sun, 18 Oct 2009 22:54:02 +0400
Subject: [Freeswitch-users] Status of ubuntu/debian packages.
In-Reply-To: <20091018164000.GD13895@base.carmickle.com>
References: <4ABA47FB.2050100@gmail.com>
	<90823c940910180249u69f18aaamc383f9a5a0c65848@mail.gmail.com>
	<20091018164000.GD13895@base.carmickle.com>
Message-ID: <90823c940910181154w615b9ac5v89a30a3a3288e619@mail.gmail.com>

On Sun, Oct 18, 2009 at 8:40 PM, Frank Carmickle <frank at carmickle.com> wrote:
> On Sun, Oct 18, Dmitry Bely wrote:
>> William,
>>
>> When I upgrade freeswitch-config package, all my customized config
>> files are silently overwritten with the default ones from .deb
>> package. This bug is quite annoying.
>
> Very sorry. ?I can try to include some code that detects if your moving from the old versions that don't have a config package to the new stuff that does.

I did have the previous version of freeswitch-config installed but
nevertheless the files were lost.

> ?My way around this is not to install the freeswitch-config package at all. ?I extract it
> some where else and diff my way through it.

Yes, but the bug should be fixed.

- Dmitry Bely



From brian at freeswitch.org  Sun Oct 18 12:06:13 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 18 Oct 2009 14:06:13 -0500
Subject: [Freeswitch-users] anyway to not to retry
In-Reply-To: <27c25bc40910180829y6ced15d7hdb75e7dda8daf04c@mail.gmail.com>
References: <27c25bc40910180829y6ced15d7hdb75e7dda8daf04c@mail.gmail.com>
Message-ID: <EDA7684B-73B8-486D-A7AA-B689A268AFA5@freeswitch.org>

You can also try this http://wiki.freeswitch.org/wiki/Channel_Variables#progress_timeout

/b

On Oct 18, 2009, at 10:29 AM, Juan Backson wrote:

> Hi,
>
> Is there anyway to configure freeswitch so that it won't retry the  
> SIP message when 200 OK is not received?
>
> thanks,
> jb
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From tayeb.meftah at gmail.com  Sun Oct 18 13:08:53 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sun, 18 Oct 2009 20:08:53 +0000
Subject: [Freeswitch-users] freeswitch french community
Message-ID: <4ADB75D5.9000705@gmail.com>

hello folk
we are starting a new freeswitch community for french users
the newly created Irc channel is #freeswitch-fr
freeswitch staf, please can we register this channel officialy to the 
freeswitch freenode group?
also can we build a french documentation and host it somehere? in the wiki?
my long time friend, diegoviola i going to help about the freeswitch 
wiki markup
we are waiting for french users to join, and welcome
thank toAnthony Minessale for accepting the community
welcome!


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4520 (20091018) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com





From sjmudd at pobox.com  Sun Oct 18 13:09:07 2009
From: sjmudd at pobox.com (Simon J Mudd)
Date: 18 Oct 2009 22:09:07 +0200
Subject: [Freeswitch-users] macros for performing repeated dialplan
	actions?
In-Reply-To: <b39273530910170018h5063c309xa96ab59b15256055@mail.gmail.com>
References: <20091014220135.GA12614@mad06.wl0.org>
	<87f2f3b90910141510n45401995y11e3e380653e9790@mail.gmail.com>
	<m3eip41yxv.fsf@mad06.wl0.org>
	<1C80FD78-71B7-4D62-8C45-2F5BC8C1564C@apartmentlines.com>
	<m3aazq1pvl.fsf@mad06.wl0.org>
	<b39273530910170018h5063c309xa96ab59b15256055@mail.gmail.com>
Message-ID: <m363ac1n30.fsf@mad06.wl0.org>

rupa at rupa.com (Rupa Schomaker) writes:

> On Sat, Oct 17, 2009 at 1:44 AM, Simon J Mudd <sjmudd at pobox.com> wrote:
> > I've other plans like:
> >
> > 1. choose the cheapest gateway based on some scripts which download
> > rates from the various gateways web pages and let me compare rates.
> > Hard partly as the "charges pages" aren't designed to allow me to
> > match a number to a charge. Shame the VoIP community can't come up
> > with a common format "charge/rate file" to make this easier.
> 
> Look at mod_lcr.  The only part you have to write is the part that
> loads the database from your providers.

Sounds good. Thanks for the pointer.

> > 2. Try one gateway and if that fails try another one. Many of the
> > gateways I've been using are low cost and don't always work, so to
> > avoid me having to adjust things each time this might be useful.
> 
> bridge to ${lcr_auto_route} after calling the lcr application.

You make it sound easy in FreeSWITCH. :-)

> > 3. Remove a lot of the "intelligence" about what to do to a backend
> > script/program or whatever which decides how to route calls and tells
> > FreeSWITCH what it decides. This may not scale for a large setup but
> > for SOHO is never going to be an issue. It also avoids me writing so
> > many xml configuration files which seems to get spread all over
> > various places of the conf/ tree.
> 
> This is for incoming calls to your DIDs?  Or something else?  I'm not
> sure what you mean here.

Well it seems/ed that the dialplan actions and xml files get complex
if you want to do anything complicated. however it seems that I'm
doing things the wrong way and perhaps that's part of the problem.

I'll look at the mod_lcr and see how it works.

Thanks for the pointers.

Simon



From diego.viola at gmail.com  Sun Oct 18 16:13:34 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 18 Oct 2009 23:13:34 +0000
Subject: [Freeswitch-users] freeswitch french community
In-Reply-To: <4ADB75D5.9000705@gmail.com>
References: <4ADB75D5.9000705@gmail.com>
Message-ID: <86a32abc0910181613g321b8315t6204f6e45be7a212@mail.gmail.com>

Hi Meftah,

I'm more than happy to help with the documentation and anything else related
to FreeSWITCH.

As you mention the documentation and translation, there are some things that
I would really like to improve in the documentation.

I propose that we start a new documentation from the ground up, we really
need a better wiki engine that supports inline translation and a better
search engine. I'm open to do all the hard work like porting the entire
documentation to it.

We could put the new wiki on a sub-domain like: newdocs.freeswitch.org and I
do all the work there, without affecting others.

That way we could have inline documentation like this:

<text:en>status - show status information</text>
<text:es>status - mostrar informacion de estado</text>

It's up to others to decide if this is really worth or not.

I'm open to help and to give my time to improve things like documentation,
etc :).

Diego

On Sun, Oct 18, 2009 at 8:08 PM, Meftah Tayeb <tayeb.meftah at gmail.com>wrote:

> hello folk
> we are starting a new freeswitch community for french users
> the newly created Irc channel is #freeswitch-fr
> freeswitch staf, please can we register this channel officialy to the
> freeswitch freenode group?
> also can we build a french documentation and host it somehere? in the wiki?
> my long time friend, diegoviola i going to help about the freeswitch
> wiki markup
> we are waiting for french users to join, and welcome
> thank toAnthony Minessale for accepting the community
> welcome!
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus
> signature database 4520 (20091018) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From diego.viola at gmail.com  Sun Oct 18 16:17:27 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 18 Oct 2009 23:17:27 +0000
Subject: [Freeswitch-users] freeswitch french community
In-Reply-To: <86a32abc0910181613g321b8315t6204f6e45be7a212@mail.gmail.com>
References: <4ADB75D5.9000705@gmail.com>
	<86a32abc0910181613g321b8315t6204f6e45be7a212@mail.gmail.com>
Message-ID: <86a32abc0910181617o212e25e4i78f3b595860b8a32@mail.gmail.com>

The other way we could do is set up a fr.freeswitch.org with mediawiki on it
and you start to translate from the English wiki.

But I think inline translation would be better for this.

Just my two cents.

Diego


On Sun, Oct 18, 2009 at 11:13 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Hi Meftah,
>
> I'm more than happy to help with the documentation and anything else
> related to FreeSWITCH.
>
> As you mention the documentation and translation, there are some things
> that I would really like to improve in the documentation.
>
> I propose that we start a new documentation from the ground up, we really
> need a better wiki engine that supports inline translation and a better
> search engine. I'm open to do all the hard work like porting the entire
> documentation to it.
>
> We could put the new wiki on a sub-domain like: newdocs.freeswitch.org and
> I do all the work there, without affecting others.
>
> That way we could have inline documentation like this:
>
> <text:en>status - show status information</text>
> <text:es>status - mostrar informacion de estado</text>
>
> It's up to others to decide if this is really worth or not.
>
> I'm open to help and to give my time to improve things like documentation,
> etc :).
>
> Diego
>
>
> On Sun, Oct 18, 2009 at 8:08 PM, Meftah Tayeb <tayeb.meftah at gmail.com>wrote:
>
>> hello folk
>> we are starting a new freeswitch community for french users
>> the newly created Irc channel is #freeswitch-fr
>> freeswitch staf, please can we register this channel officialy to the
>> freeswitch freenode group?
>> also can we build a french documentation and host it somehere? in the
>> wiki?
>> my long time friend, diegoviola i going to help about the freeswitch
>> wiki markup
>> we are waiting for french users to join, and welcome
>> thank toAnthony Minessale for accepting the community
>> welcome!
>>
>>
>> __________ Information from ESET NOD32 Antivirus, version of virus
>> signature database 4520 (20091018) __________
>>
>> The message was checked by ESET NOD32 Antivirus.
>>
>> http://www.eset.com
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From diego.viola at gmail.com  Sun Oct 18 16:19:41 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 18 Oct 2009 23:19:41 +0000
Subject: [Freeswitch-users] freeswitch french community
In-Reply-To: <86a32abc0910181617o212e25e4i78f3b595860b8a32@mail.gmail.com>
References: <4ADB75D5.9000705@gmail.com>
	<86a32abc0910181613g321b8315t6204f6e45be7a212@mail.gmail.com>
	<86a32abc0910181617o212e25e4i78f3b595860b8a32@mail.gmail.com>
Message-ID: <86a32abc0910181619oa26ed79l4c55b967fa2a5e95@mail.gmail.com>

Take a look at this for instance, for an example of inline translation:
http://wiki.rubyonrails.org/

Check the: "Translations of this page" -- all in the same page, I'd love to
have something like that for FreeSWITCH.

Along with a good search engine on the wiki (not Google).

Diego

On Sun, Oct 18, 2009 at 11:17 PM, Diego Viola <diego.viola at gmail.com> wrote:

> The other way we could do is set up a fr.freeswitch.org with mediawiki on
> it and you start to translate from the English wiki.
>
> But I think inline translation would be better for this.
>
> Just my two cents.
>
> Diego
>
>
>
> On Sun, Oct 18, 2009 at 11:13 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> Hi Meftah,
>>
>> I'm more than happy to help with the documentation and anything else
>> related to FreeSWITCH.
>>
>> As you mention the documentation and translation, there are some things
>> that I would really like to improve in the documentation.
>>
>> I propose that we start a new documentation from the ground up, we really
>> need a better wiki engine that supports inline translation and a better
>> search engine. I'm open to do all the hard work like porting the entire
>> documentation to it.
>>
>> We could put the new wiki on a sub-domain like: newdocs.freeswitch.organd I do all the work there, without affecting others.
>>
>> That way we could have inline documentation like this:
>>
>> <text:en>status - show status information</text>
>> <text:es>status - mostrar informacion de estado</text>
>>
>> It's up to others to decide if this is really worth or not.
>>
>> I'm open to help and to give my time to improve things like documentation,
>> etc :).
>>
>> Diego
>>
>>
>> On Sun, Oct 18, 2009 at 8:08 PM, Meftah Tayeb <tayeb.meftah at gmail.com>wrote:
>>
>>> hello folk
>>> we are starting a new freeswitch community for french users
>>> the newly created Irc channel is #freeswitch-fr
>>> freeswitch staf, please can we register this channel officialy to the
>>> freeswitch freenode group?
>>> also can we build a french documentation and host it somehere? in the
>>> wiki?
>>> my long time friend, diegoviola i going to help about the freeswitch
>>> wiki markup
>>> we are waiting for french users to join, and welcome
>>> thank toAnthony Minessale for accepting the community
>>> welcome!
>>>
>>>
>>> __________ Information from ESET NOD32 Antivirus, version of virus
>>> signature database 4520 (20091018) __________
>>>
>>> The message was checked by ESET NOD32 Antivirus.
>>>
>>> http://www.eset.com
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
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From jason at jasonjgw.net  Sun Oct 18 16:58:58 2009
From: jason at jasonjgw.net (Jason White)
Date: Mon, 19 Oct 2009 10:58:58 +1100
Subject: [Freeswitch-users] freeswitch french community
In-Reply-To: <86a32abc0910181613g321b8315t6204f6e45be7a212@mail.gmail.com>
References: <4ADB75D5.9000705@gmail.com>
	<86a32abc0910181613g321b8315t6204f6e45be7a212@mail.gmail.com>
Message-ID: <20091018235858.GA19968@jdc.jasonjgw.net>

Diego Viola <diego.viola at gmail.com> wrote:
 
> I propose that we start a new documentation from the ground up, we really
> need a better wiki engine that supports inline translation and a better
> search engine. I'm open to do all the hard work like porting the entire
> documentation to it.

Is there a good wiki engine which allows the documentation to be edited as
files in a file system as well as via a Web interface - something like
Ikiwiki which uses Git or another version control system to store the files,
perhaps?

On occasion I have wished I could add to or update the wiki easily, but there
are two problems.

1. The current wiki requires a captcha to be completed in order to register an
account. I can't see the captcha - this is an accessibility issue.

2. Editing anything through a Web interface is painful. If I could edit the
content as files in Emacs or Vim it would be much easier, and I would be more
likely to do it instead of deciding that the difficulty and inconvenience are
simply too great.

Obviously this is only one person's perspective: other people will have very
different needs from mine.




From larclap at yahoo.com  Sun Oct 18 17:02:57 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Sun, 18 Oct 2009 17:02:57 -0700
Subject: [Freeswitch-users] Outbound sip address call fails
In-Reply-To: <20091016172936.CCE7C41E8F6@mail.cune.org>
References: <011301ca4dd3$67d486a0$377d93e0$@com>
	<20091016172936.CCE7C41E8F6@mail.cune.org>
Message-ID: <01e201ca504f$83bc61c0$8b352540$@com>

No, the checkbox is unchecked.

> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-
> bounces at lists.freeswitch.org] On Behalf Of Russell.Mosemann at cune.org
> Sent: Friday, October 16, 2009 10:30 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Outbound sip address call fails
> 
> Lars Zeb <larclap at yahoo.com> said:
> 
> > I tried dial sip:echo at iptel.org. In the pastebin you can see that this
> > address came over to the cli as "echo". Is it Bria which is messing up,
or
> > my FS configuration?
> 
> It looks like a Bria problem. If you open the account information and go
> to the Advanced tab, is the box checked to "Send outgoing request
> directly to target"?
> 
> 
> --
> Russell Mosemann
> 
> 
> 
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From Russell.Mosemann at cune.org  Sun Oct 18 17:40:09 2009
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Sun, 18 Oct 2009 19:40:09 -0500
Subject: [Freeswitch-users] Outbound sip address call fails
In-Reply-To: <01e201ca504f$83bc61c0$8b352540$@com>
References: <011301ca4dd3$67d486a0$377d93e0$@com><20091016172936.CCE7C41E8F6@mail.cune.org>
	<01e201ca504f$83bc61c0$8b352540$@com>
Message-ID: <0542FF31EAB34BD8977E8E87263AA88A@cune.pri>

> No, the checkbox is unchecked.

Well, check it.

--
Russell Mosemann




From freeswitch-list at puzzled.xs4all.nl  Sun Oct 18 04:31:17 2009
From: freeswitch-list at puzzled.xs4all.nl (Patrick List)
Date: Sun, 18 Oct 2009 13:31:17 +0200
Subject: [Freeswitch-users] Status of ubuntu/debian packages.
In-Reply-To: <90823c940910180249u69f18aaamc383f9a5a0c65848@mail.gmail.com>
References: <4ABA47FB.2050100@gmail.com>
	<90823c940910180249u69f18aaamc383f9a5a0c65848@mail.gmail.com>
Message-ID: <4ADAFC85.6000301@puzzled.xs4all.nl>

On 10/18/2009 11:49 AM, Dmitry Bely wrote:
> William,
>
> When I upgrade freeswitch-config package, all my customized config
> files are silently overwritten with the default ones from .deb
> package. This bug is quite annoying.

Hope you have a recent backup. Years ago when I created rpm packages for 
Asterisk I chose to put the default config files in it's own directory 
so there was no chance of your customized configs getting overwritten. 
Example:
/opt/freeswitch/etc/freeswitch-$svn_version-samples/

Maybe the maintainer of the debs can do something similar.

Regards,
Patrick



From mike at jerris.com  Sun Oct 18 18:56:05 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 18 Oct 2009 21:56:05 -0400
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
In-Reply-To: <4256bf830910180725y72539d98gb138934afc18d58b@mail.gmail.com>
References: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
	<177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>
	<4256bf830908041151l55980e3fifb85c75b87535426@mail.gmail.com>
	<3D0B855A-5B2C-4C6F-9306-64B0ECE97091@jerris.com>
	<4256bf830910180725y72539d98gb138934afc18d58b@mail.gmail.com>
Message-ID: <A5ECDCD9-1A12-4A31-89F0-E6F4C79FE9E9@jerris.com>

FreeSWITCH debug level logs should help tell you exactly what is  
killing the call.


On Oct 18, 2009, at 10:25 AM, Matthew Fong wrote:

> I'm still having this issue with random EXCHANGE_ROUTING_ERROR  
> disconnects for users utilizing my inbound DID to connect to my  
> FreeSWITCH server. It's a predictive dialing application, with one  
> agent session being bridged with multiple calls and transfered back  
> and forth between extensions in my dial plan. After a random number  
> of bridging and transferring, FreeSWITCH suddenly sends a BYE to my  
> DID provider citing an EXCHANGE_ROUTING_ERROR. It does not happen at  
> any one-point in my dial plan, or applications--it just randomly  
> disconnects when a call that the Agent is bridged to hangs-up or is  
> disconnected. It seems to only happen when two external sip profiles  
> are being bridged together, and not when an internal and external  
> profile is being bridged.
>
> I turned
>
> sip trace on and
> sofia loglevel all 9
>
> below is the the snippet. I've posted the entire Agent session at  
> the following pastebin http://pastebin.freeswitch.org/10756
>
> tport_deliver(0x18413c0): msg 0x7faeb818ea30 (304 bytes) from udp/ 
> 208.76.18.254:5080/sip next=(nil)
> nta: received 200 OK for BYE (121818983)
> nta: 200 OK is going to a transaction
> nta_outgoing: RTT is 84.409 ms
> tport_release(0x18413c0): 0x1a15cc0 by 0x1a16a00 with 0x7faeb818ea30
> nua(0x1ad6fb0): event r_bye 200 OK
> nua(0x1ad6fb0): call state changed: terminating -> terminated
> nua(0x1ad6fb0): event i_state 200 to BYE
> nua: nua_application_event: entering
> nua(0x1ad6fb0): event i_terminated 200 to BYE
> nua: nua_handle_magic: entering
> nua(0x1ad6fb0): removing session usage
> soa_destroy(static::0x1b5ae90) called
> nua: nua_application_event: entering
> nta_leg_destroy(0x1b594a0)
> nua: nua_handle_magic: entering
> nua: nua_handle_bind: entering
> nua: nua_application_event: entering
> nua: nua_handle_magic: entering
> nua: terminated session 0x1ad6fb0
> nua: nua_handle_destroy: entering
> nua(0x1ad6fb0): recv signal r_destroy
> nta_leg_destroy((nil))
> nua(0x1ad6fb0): sent signal r_destroy
> nta: timer set next to 28 ms
> nta: timer E fired, retransmit BYE (121818989)
> tport_release(0x18413c0): 0x1b5c9b0 by 0x7faeb817d830 with (nil)
> tport_tsend(0x18413c0) tpn = */209.216.2.211:5060
> tport_resolve addrinfo = 209.216.2.211:5060
> tport_by_addrinfo(0x18413c0): not found by name */209.216.2.211:5060
> tport_vsend(0x18413c0): 862 bytes of 862 to udp/209.216.2.211:5060
> tport_vsend returned 862
> send 862 bytes to udp/[209.216.2.211]:5060 at 14:04:11.753690:
>     
> ------------------------------------------------------------------------
>    BYE sip:199.173.100.16:5060;transport=UDP SIP/2.0
>    Via: SIP/2.0/UDP  
> 67.220.216.146:5080;rport;branch=z9hG4bK02jNX8a4HrNyN
>    Route: <sip:209.216.2.211;ftag=4adc7-13c4-1ab03-71ce3705-1ab03;lr=on 
> >
>    Route: <sip:65.211.120.237:5060;lr>
>    Route: <sip:63.110.102.239;lr>
>    Max-Forwards: 70
>    From: <sip: 
> +12133304391 at 63.110.102.239:5060;user=phone>;tag=cgBe054jZrt3a
>    To: <sip: 
> + 
> 14158867717 
> @199.173.100.16:5060;user=phone>;tag=4adc7-13c4-1ab03-71ce3705-1ab03
>    Call-ID: a0f656a01064adc713c41ab036840746ee20ca11c06b2d8-0440-5124
>    CSeq: 121818989 BYE
>    Contact: <sip:+12133304391 at 67.220.216.146:5080;transport=udp>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15135
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO,  
> REGISTER, REFER, UPDATE, NOTIFY
>    Supported: timer, precondition, path, replaces
>    Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
>    Content-Length: 0
>
>     
> ------------------------------------------------------------------------
>
> Thanks.
> --matt
>
> On Sat, Aug 8, 2009 at 12:42 PM, Michael Jerris <mike at jerris.com>  
> wrote:
> http://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP
>
> turn the logging all the way up and see what it says.
>
> Mike
>
> On Aug 4, 2009, at 2:51 PM, Matthew Fong wrote:
>
>> Hi Mathieu, thanks for the reply. I enabled sip trace logging and  
>> got the logs below, but I am still at a loss at being able to  
>> identify the error or reproduce it consistently. The below log  
>> indicates to me that my FS server is initiating sending 2 BYE  
>> message to my DID provider (didforsale.com). Is there a way I can  
>> look further inside FreeSWITCH to see why it is sending this BYE  
>> packet?
>>
>>
>> sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:44.679208:
>> BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0
>> Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD
>> Route: <sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on>
>> Route: <sip:65.217.40.210:5060;lr>
>> Route: <sip:63.110.102.238;lr>
>> Max-Forwards: 70
>> From: <sip: 
>> +1212381XXXX at 63.110.102.238:5060;user=phone>;tag=Ztr5ycrv3QZ1g
>> To: <sip: 
>> + 
>> 1909635XXXX 
>> @199.173.100.144:5060;user=phone>;tag=dc7-13c4-2401b7-46dea593-2401b7
>> Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441
>> CSeq: 118584736 BYE
>> Contact: <sip:+1212381XXXX at 66.197.142.69:5080;transport=udp>
>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,  
>> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
>> Supported: timer, precondition, path, replaces
>> Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
>> Content-Length: 0
>>
>>
>> sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:45.182589:
>> BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0
>> Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD
>> Route: <sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on>
>> Route: <sip:65.217.40.210:5060;lr>
>> Route: <sip:63.110.102.238;lr>
>> Max-Forwards: 70
>> From: <sip: 
>> +1212381XXXX at 63.110.102.238:5060;user=phone>;tag=Ztr5ycrv3QZ1g
>> To: <sip: 
>> + 
>> 1909635XXXX 
>> @199.173.100.144:5060;user=phone>;tag=dc7-13c4-2401b7-46dea593-2401b7
>> Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441
>> CSeq: 118584736 BYE
>> Contact: <sip:+1212381XXXX at 66.197.142.69:5080;transport=udp>
>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,  
>> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
>> Supported: timer, precondition, path, replaces
>> Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
>> Content-Length: 0
>>
>> On Sun, Aug 2, 2009 at 11:20 PM, Mathieu Rene <mrene_lists at avgs.ca>  
>> wrote:
>> Hi,
>>
>> Digging a bit in mod_sofia releaved that it can be caused by a SIP
>> code 482 (loop detected), 483 (too many hops) or 484 (address
>> incomplete).
>>
>> Do a SIP trace to sched more light on what's happening.
>>
>> Mathieu Rene
>> Avant-Garde Solutions Inc
>> Office: + 1 (514) 664-1044 x100
>> Cell: +1 (514) 664-1044 x200
>> m
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From larclap at yahoo.com  Sun Oct 18 19:22:45 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Sun, 18 Oct 2009 19:22:45 -0700
Subject: [Freeswitch-users] Outbound sip address call fails
In-Reply-To: <0542FF31EAB34BD8977E8E87263AA88A@cune.pri>
References: <011301ca4dd3$67d486a0$377d93e0$@com><20091016172936.CCE7C41E8F6@mail.cune.org>	<01e201ca504f$83bc61c0$8b352540$@com>
	<0542FF31EAB34BD8977E8E87263AA88A@cune.pri>
Message-ID: <01e301ca5063$0c498690$24dc93b0$@com>

Thanks, Russell, that did it. Works perfectly!

> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-
> bounces at lists.freeswitch.org] On Behalf Of Russell Mosemann
> Sent: Sunday, October 18, 2009 5:40 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Outbound sip address call fails
> 
> > No, the checkbox is unchecked.
> 
> Well, check it.
> 
> --
> Russell Mosemann
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From dujinfang at gmail.com  Sun Oct 18 20:04:00 2009
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 19 Oct 2009 11:04:00 +0800
Subject: [Freeswitch-users] freeswitch french community
In-Reply-To: <86a32abc0910181617o212e25e4i78f3b595860b8a32@mail.gmail.com>
References: <4ADB75D5.9000705@gmail.com>
	<86a32abc0910181613g321b8315t6204f6e45be7a212@mail.gmail.com>
	<86a32abc0910181617o212e25e4i78f3b595860b8a32@mail.gmail.com>
Message-ID: <23f91030910182004g6bfdafb6g6198d107220f8320@mail.gmail.com>

There are some plugins for mediawiki to support multilanguage, either inline
or on a separate page. However, both have Pros and Cons. e.g.

multi-language support this with a separate page

some_page #default to en
some_page/fr Franch
some_page/zh Chinese

Pros: clear, available languages can be listed in "Other languages for this
page"
Cons: hard to keep all languages in sync

other plugins can do inline translation:

<@en|hello>
<@fr|Ca Va>
<@zh|??>

Pros: untranslated items remains in default lang, no sync headache
Cons: It breaks a page into sentences or items, also if we have more than 10
languages the source code maybe really a mess.


2009/10/19 Diego Viola <diego.viola at gmail.com>

> The other way we could do is set up a fr.freeswitch.org with mediawiki on
> it and you start to translate from the English wiki.
>
> But I think inline translation would be better for this.
>
> Just my two cents.
>
> Diego
>
>
>
> On Sun, Oct 18, 2009 at 11:13 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> Hi Meftah,
>>
>> I'm more than happy to help with the documentation and anything else
>> related to FreeSWITCH.
>>
>> As you mention the documentation and translation, there are some things
>> that I would really like to improve in the documentation.
>>
>> I propose that we start a new documentation from the ground up, we really
>> need a better wiki engine that supports inline translation and a better
>> search engine. I'm open to do all the hard work like porting the entire
>> documentation to it.
>>
>> We could put the new wiki on a sub-domain like: newdocs.freeswitch.organd I do all the work there, without affecting others.
>>
>> That way we could have inline documentation like this:
>>
>> <text:en>status - show status information</text>
>> <text:es>status - mostrar informacion de estado</text>
>>
>> It's up to others to decide if this is really worth or not.
>>
>> I'm open to help and to give my time to improve things like documentation,
>> etc :).
>>
>> Diego
>>
>>
>> On Sun, Oct 18, 2009 at 8:08 PM, Meftah Tayeb <tayeb.meftah at gmail.com>wrote:
>>
>>> hello folk
>>> we are starting a new freeswitch community for french users
>>> the newly created Irc channel is #freeswitch-fr
>>> freeswitch staf, please can we register this channel officialy to the
>>> freeswitch freenode group?
>>> also can we build a french documentation and host it somehere? in the
>>> wiki?
>>> my long time friend, diegoviola i going to help about the freeswitch
>>> wiki markup
>>> we are waiting for french users to join, and welcome
>>> thank toAnthony Minessale for accepting the community
>>> welcome!
>>>
>>>
>>> __________ Information from ESET NOD32 Antivirus, version of virus
>>> signature database 4520 (20091018) __________
>>>
>>> The message was checked by ESET NOD32 Antivirus.
>>>
>>> http://www.eset.com
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From chris at fowler.cc  Sun Oct 18 20:14:49 2009
From: chris at fowler.cc (Chris Fowler)
Date: Sun, 18 Oct 2009 20:14:49 -0700
Subject: [Freeswitch-users] NAT problems migrating from Version
 1.0.trunk (13168M) to Version 1.0.trunk (15166)
Message-ID: <1255922089.4653.1340747457@webmail.messagingengine.com>

I've tried all sorts of debug and parameter changes over the weekend,
but still can't figure out the correct solution.

If I disable timers on the sip profile then all works fine.
 <param name="enable-timer" value="false"/>

But that seems like a hack; not a correct solution.  With the build
13168M (which is pre the new NAT functionality) everything worked fine.

The SIP trace shows the phones and FreeSWITCH happily exchanging NOTIFY
and "200 OK" messages.  Audio's working - just calls timeout after 100
seconds with RECOVERY_ON_TIMER_EXPIRE.  Is enforcement of this timer new
functionality - and really just exposing a problem I've always had
before?

The config is (50 Polycom Phones -> NAT -> Internet -> Amazon EC2)

I would really appreciate some pointers on what to look for; additional
trace that might reveal something.

Thanks, Chris.


On Fri, 16 Oct 2009 20:06:52 -0700, "Chris Fowler" <chris at fowler.cc>
said:
> Hi,
> 
> We've been using 13168M in production for some time now (works great). 
> I want to get us onto the latest build but am having problems getting
> NAT to work.
> 
> Phones can register; can dial 9999 test #, but after 100 seconds the
> call is disconnected with error:
> 2009-10-16 19:52:26.936618 [NOTICE] sofia.c:4038 Hangup
> sofia/internal/1001 at myhost.mydomain.com [CS_EXECUTE]
> [RECOVERY_ON_TIMER_EXPIRE]
> 
> I took the standard internal.xml and vars.xml files from the new build
> and made the following modifications - which worked previously:
> 
> modify conf/vars.xml and update
>   <X-PRE-PROCESS cmd="set" data="domain=myhost.mydomain.com"/>
> 
>   <X-PRE-PROCESS cmd="set" data="bind_server_ip=1.2.3.4"/>
>   <X-PRE-PROCESS cmd="set" data="external_rtp_ip=1.2.3.4"/>
>   <X-PRE-PROCESS cmd="set" data="external_sip_ip=1.2.3.4"/>
> 
> Modify conf/sip_profiles/internal.xml
>  <param name="aggressive-nat-detection" value="true"/>  <param
>  name="ext-rtp-ip" value="$${external_rtp_ip}"/>  <param
>  name="ext-sip-ip" value="$${external_sip_ip}"/>  <param
>  name="NDLB-received-in-nat-reg-contact" value="true"/>  <param
>  name="NDLB-force-rport" value="true"/>  <param
>  name="NDLB-broken-auth-hash" value="true"/>
> 
> The big difference I note is that on PRODUCTION (which works) "sofia
> status profile internal" yields:
> URL                     sip:mod_sofia at 1.2.3.4:5060
> BIND-URL                sip:mod_sofia at 1.2.3.4:5060;maddr=10.250.35.224
> 
> But on Test I see:
> URL                     sip:mod_sofia at 10.250.66.210:5060
> BIND-URL                sip:mod_sofia at 10.250.66.210:5060
> 
> Any ideas?
> 
> Thanks, Chris.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From msc at freeswitch.org  Sun Oct 18 20:37:16 2009
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 18 Oct 2009 20:37:16 -0700
Subject: [Freeswitch-users] Excellent FreeSWITCH Article For You To Read
Message-ID: <87f2f3b90910182037x5c1d1279ue50f724f0a02f94c@mail.gmail.com>

Hello all,

Just a heads up that there is a great article out there for you to read.
Please visit the main freeswitch.org page here:
http://www.freeswitch.org/node/211

Click the link and digg the article after you read it. Please spread the
word! The twittersphere is starting to warm up already. Let's keep the
momentum going!

Thanks,
Michael
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From mattdfong at gmail.com  Sun Oct 18 21:48:25 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Mon, 19 Oct 2009 11:48:25 +0700
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
In-Reply-To: <A5ECDCD9-1A12-4A31-89F0-E6F4C79FE9E9@jerris.com>
References: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
	<177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>
	<4256bf830908041151l55980e3fifb85c75b87535426@mail.gmail.com>
	<3D0B855A-5B2C-4C6F-9306-64B0ECE97091@jerris.com>
	<4256bf830910180725y72539d98gb138934afc18d58b@mail.gmail.com>
	<A5ECDCD9-1A12-4A31-89F0-E6F4C79FE9E9@jerris.com>
Message-ID: <4256bf830910182148mf7f2bf9vdcccd687e0caa1e7@mail.gmail.com>

The debug level logs to the console, right? The pastebin, had log level
debug, sofia trace on for external and default, and sofia loglevel all 9. Is
there another log enable command I'm missing? It seems loglevel all 9
outputs enter and exit functions, but at least to my novice eye, it's not
too obvious why freeswitch is sending a BYE to my DID provider.

I did do some additional testing, and my prior comment about it working in
the internal profile is incorrect. Even if I put my DID provider in my
internal profile, I still receive the Exchange_Routing_Error after being
bridged with a few channels. However, I purchased a DID from icall, and that
DID worked. So icall YES, didforsale.com NO...It's too bad tho that
didforsale.com doesn't work too well with FreeSWITCH because their 20
unlimited inbound channels can't be beat.

--matt
http://www.hellohunter.com


On Mon, Oct 19, 2009 at 8:56 AM, Michael Jerris <mike at jerris.com> wrote:

> FreeSWITCH debug level logs should help tell you exactly what is killing
> the call.
>
>
> On Oct 18, 2009, at 10:25 AM, Matthew Fong wrote:
>
> I'm still having this issue with random EXCHANGE_ROUTING_ERROR disconnects
> for users utilizing my inbound DID to connect to my FreeSWITCH server. It's
> a predictive dialing application, with one agent session being bridged with
> multiple calls and transfered back and forth between extensions in my dial
> plan. After a random number of bridging and transferring, FreeSWITCH
> suddenly sends a BYE to my DID provider citing an EXCHANGE_ROUTING_ERROR. It
> does not happen at any one-point in my dial plan, or applications--it just
> randomly disconnects when a call that the Agent is bridged to hangs-up or is
> disconnected. It seems to only happen when two external sip profiles are
> being bridged together, and not when an internal and external profile is
> being bridged.
> I turned
>
> sip trace on and
> sofia loglevel all 9
>
> below is the the snippet. I've posted the entire Agent session at the
> following pastebin http://pastebin.freeswitch.org/10756
>
> tport_deliver(0x18413c0): msg 0x7faeb818ea30 (304 bytes) from udp/
> 208.76.18.254:5080/sip next=(nil)
> nta: received 200 OK for BYE (121818983)
> nta: 200 OK is going to a transaction
> nta_outgoing: RTT is 84.409 ms
> tport_release(0x18413c0): 0x1a15cc0 by 0x1a16a00 with 0x7faeb818ea30
> nua(0x1ad6fb0): event r_bye 200 OK
> nua(0x1ad6fb0): call state changed: terminating -> terminated
> nua(0x1ad6fb0): event i_state 200 to BYE
> nua: nua_application_event: entering
> nua(0x1ad6fb0): event i_terminated 200 to BYE
> nua: nua_handle_magic: entering
> nua(0x1ad6fb0): removing session usage
> soa_destroy(static::0x1b5ae90) called
> nua: nua_application_event: entering
> nta_leg_destroy(0x1b594a0)
> nua: nua_handle_magic: entering
> nua: nua_handle_bind: entering
> nua: nua_application_event: entering
> nua: nua_handle_magic: entering
> nua: terminated session 0x1ad6fb0
> nua: nua_handle_destroy: entering
> nua(0x1ad6fb0): recv signal r_destroy
> nta_leg_destroy((nil))
> nua(0x1ad6fb0): sent signal r_destroy
> nta: timer set next to 28 ms
> nta: timer E fired, retransmit BYE (121818989)
> tport_release(0x18413c0): 0x1b5c9b0 by 0x7faeb817d830 with (nil)
> tport_tsend(0x18413c0) tpn = */209.216.2.211:5060
> tport_resolve addrinfo = 209.216.2.211:5060
> tport_by_addrinfo(0x18413c0): not found by name */209.216.2.211:5060
> tport_vsend(0x18413c0): 862 bytes of 862 to udp/209.216.2.211:5060
> tport_vsend returned 862
> send 862 bytes to udp/[209.216.2.211]:5060 at 14:04:11.753690:
>    ------------------------------------------------------------------------
>    BYE sip:199.173.100.16:5060;transport=UDP SIP/2.0
>    Via: SIP/2.0/UDP 67.220.216.146:5080;rport;branch=z9hG4bK02jNX8a4HrNyN
>    Route: <sip:209.216.2.211;ftag=4adc7-13c4-1ab03-71ce3705-1ab03;lr=on>
>    Route: <sip:65.211.120.237:5060;lr>
>    Route: <sip:63.110.102.239;lr>
>    Max-Forwards: 70
>    From: <sip:+12133304391 at 63.110.102.239:5060;user=phone
> >;tag=cgBe054jZrt3a
>    To: <sip:+14158867717 at 199.173.100.16:5060;user=phone
> >;tag=4adc7-13c4-1ab03-71ce3705-1ab03
>    Call-ID: a0f656a01064adc713c41ab036840746ee20ca11c06b2d8-0440-5124
>    CSeq: 121818989 BYE
>    Contact: <sip:+12133304391 at 67.220.216.146:5080;transport=udp>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15135
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
> REFER, UPDATE, NOTIFY
>    Supported: timer, precondition, path, replaces
>    Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
>
> Thanks.
> --matt
>
> On Sat, Aug 8, 2009 at 12:42 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> http://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP
>> turn the logging all the way up and see what it says.
>>
>> Mike
>>
>> On Aug 4, 2009, at 2:51 PM, Matthew Fong wrote:
>>
>> Hi Mathieu, thanks for the reply. I enabled sip trace logging and got the
>> logs below, but I am still at a loss at being able to identify the error or
>> reproduce it consistently. The below log indicates to me that my FS server
>> is initiating sending 2 BYE message to my DID provider (didforsale.com).
>> Is there a way I can look further inside FreeSWITCH to see why it is sending
>> this BYE packet?
>>
>> sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:44.679208:
>> BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0
>> Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD
>> Route: <sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on>
>> Route: <sip:65.217.40.210:5060;lr>
>> Route: <sip:63.110.102.238;lr>
>> Max-Forwards: 70
>> From: <sip:+1212381XXXX at 63.110.102.238:5060;user=phone>;tag=Ztr5ycrv3QZ1g
>> To: <sip:+1909635XXXX at 199.173.100.144:5060;user=phone
>> >;tag=dc7-13c4-2401b7-46dea593-2401b7
>> Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441
>> CSeq: 118584736 BYE
>> Contact: <sip:+1212381XXXX at 66.197.142.69:5080;transport=udp>
>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>> NOTIFY, REFER, UPDATE, REGISTER, INFO
>> Supported: timer, precondition, path, replaces
>> Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
>> Content-Length: 0
>>
>>
>> sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:45.182589:
>> BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0
>> Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD
>> Route: <sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on>
>> Route: <sip:65.217.40.210:5060;lr>
>> Route: <sip:63.110.102.238;lr>
>> Max-Forwards: 70
>> From: <sip:+1212381XXXX at 63.110.102.238:5060;user=phone>;tag=Ztr5ycrv3QZ1g
>> To: <sip:+1909635XXXX at 199.173.100.144:5060;user=phone
>> >;tag=dc7-13c4-2401b7-46dea593-2401b7
>> Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441
>> CSeq: 118584736 BYE
>> Contact: <sip:+1212381XXXX at 66.197.142.69:5080;transport=udp>
>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>> NOTIFY, REFER, UPDATE, REGISTER, INFO
>> Supported: timer, precondition, path, replaces
>> Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
>> Content-Length: 0
>>
>> On Sun, Aug 2, 2009 at 11:20 PM, Mathieu Rene <mrene_lists at avgs.ca>wrote:
>>
>>> Hi,
>>>
>>> Digging a bit in mod_sofia releaved that it can be caused by a SIP
>>> code 482 (loop detected), 483 (too many hops) or 484 (address
>>> incomplete).
>>>
>>> Do a SIP trace to sched more light on what's happening.
>>>
>>> Mathieu Rene
>>> Avant-Garde Solutions Inc
>>> Office: + 1 (514) 664-1044 x100
>>> Cell: +1 (514) 664-1044 x200
>>> m <mrene at avgs.ca>
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mcampbellsmith at gmail.com  Sun Oct 18 21:53:02 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Mon, 19 Oct 2009 15:53:02 +1100
Subject: [Freeswitch-users] validating dtmf digits received
Message-ID: <33c87fa30910182153t23e52fa0p2b967b3843eb1175@mail.gmail.com>

Hi!

I simply want to validate the dtmf digits I read from a user.    From
the wiki, it appears I need to use inline="true" when setting the
variable so it can be used directly within the same extension.

What have I done wrong below?   I have tried many different
alternatives, but the second condition field, which is meant to match
the dtmf digits received (in this case 0000) is never matched, and the
anti-action is called instead.

:
      <some code here>
      <action application="read" data="1 10
ivr/ivr-please_enter_pin_followed_by_pound.wav res 10000 9"/>
       <action application="phrase" data="spell,${res}"/>
       <action inline="true" application="set" data="code=${digits_dialed}"/>
       <!-- <action inline="true" application="set" data="code=${res}"/> -->
  </condition>
  <condition field="digits_dialed" expression="^0000$">
  <!-- <condition field="${code}" expression="^0000$"> -->
  <!-- <condition field="${res}" expression="^0000$"> -->
        <some code here>
       <anti-action application="hangup"/>

Thanks!



From tzury.by at reguluslabs.com  Mon Oct 19 00:23:57 2009
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Mon, 19 Oct 2009 09:23:57 +0200
Subject: [Freeswitch-users] How to enforce freeswitch replying to the source
	port instead of to the one specified in v/m header parametes
	(Symmetric NAT)
Message-ID: <10128ef10910190023t17574fafic892bd4105cc1703@mail.gmail.com>

Hi,

I am struggling with a cellular operator which removes the rport from
the SIP messages sent by  the client.
During the troubleshooting process I have been playing with the NDLB
parameters I found mentioned at the fs-wiki

<param name="NDLB-force-rport" value="true"/>
<param name="NDLB-received-in-nat-reg-contact" value="true"/>

as well as with sip-force-contact at the extension xml config file.

<variables>
    <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/>
</variables>

I Want to make freeswitch replying to the source port instead of the
port supplied by the client, that is
to port 12543 instead of to the 7608 as you can see below some pcap
dumps of the case.

I am not sure if that would solve thep roblem and am open to other
suggestions as well.

Thanks in advance,
Tzury

pcap dumps below:

### REGISTER ###
User Datagram Protocol, Src Port: 12543 (12543), Dst Port: sip (5060)
REGISTER sip:example.net SIP/2.0
v: SIP/2.0/UDP 212.154.128.222:7608;branch=z9hG4bKPj41fa0abd044c35c75598d561b2f93167
Max-Forwards: 70
f: <sip:1013 at example.net>;tag=9194adecbf8e8179530eb589e83631db
t: <sip:1013 at example.net>
i: ebf2e3c254517938f90f33d8fd89326d
CSeq: 17355 REGISTER
m: <sip:1013 at 212.154.128.222:7608>
Expires: 300
l:  0

### 401 UNAUTHORIZED ###
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
212.154.128.222:7608;branch=z9hG4bKPj41fa0abd044c35c75598d561b2f93167
From: <sip:1013 at example.net>;tag=9194adecbf8e8179530eb589e83631db
To: <sip:1013 at example.net>;tag=166HraN0B149g
Call-ID: ebf2e3c254517938f90f33d8fd89326d
CSeq: 17355 REGISTER
User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
WWW-Authenticate: Digest realm="example.net",
nonce="f2f1370c-bc7c-11de-b000-fbe81e221ab4", algorithm=MD5,
qop="auth"
Content-Length: 0



From irmatov at gmail.com  Mon Oct 19 01:13:18 2009
From: irmatov at gmail.com (Timur Irmatov)
Date: Mon, 19 Oct 2009 13:13:18 +0500
Subject: [Freeswitch-users] set up sound on windows
Message-ID: <241d382f0910190113v5d2cc8b2kf28eabd2feec3062@mail.gmail.com>

Hi!


We are currently developing small call-center for our company using
FreeSWITCH. Our demands are pretty low at the moment, and we have
implemented all necessary features, now moving to testing.. We manage
FreeSWITCH through mod_erlang_event. FreeSWITCH and mod_erlang_event
are great, I want to thank all developers for those two great pieces
of technology!

We met strange problem with windows clients though. It is not related
to FreeSWITCH, but I don't know where to ask. Tried searching google,
but found nothing. As soon as we plug headset with microphone into
windows machine, we immediately start hearing environment though
headset. It seems as sound from microphone is transferred to earbuds,
which is pretty annoying.

Does anyone have any hints regarding configuration of audio/mics on Windows?


-- 
Timur Irmatov, xmpp:irmatov at jabber.ru



From durk.debeer at isp.solcon.nl  Mon Oct 19 00:11:24 2009
From: durk.debeer at isp.solcon.nl (Durk de Beer)
Date: Mon, 19 Oct 2009 09:11:24 +0200
Subject: [Freeswitch-users] Freeswitch 1.0.4 problems with music on hold
Message-ID: <mailman.13.1269220909.3324.freeswitch-users@lists.freeswitch.org>

Hello,

This is something I came across on Freeswitch 1.0.4

First let me explain what I'm trying to do.

I want Free-Switch to behave as a proxy so in the settings section of Sofia.conf.xml I use



and



As fare as I am able to follow the RTP stream it is passing by Freeswitch making it a proxy.

So far so good. Now I want to serv music on hold if a call is put on hold.

The only way I am able to find for Freeswitch to serv music on hold (MOH) is for it to be in media.

Ok we've got a command for this situation namely



Ok if I put the last parram command into the Sofia.conf.xml Freeswitch is starting to behave funny.

Let me try to explain this by an example. Let suppose Alice is calling Bob without the last command, the ?resume media on hold?.

If Bob puts Alice on hold al is silent on Bob's and Alice's phone.

If Bob takes Alice of hold there's two way audio.

Ok no I put the command in sofia.onf.xml, the ?resume media on hold?.

Now Alice is calling Bob again and Bob is putting her on hold again.

You are expecting MOH on this moment but it isn't, all there is are the sounds of silence.

Ok mistake made checking local streams, and they are there?. Yes it is.

Ok other mistake, didn't put the ?hold-music?command in sofia.conf.xml. Noop it's in there so there must be MOH. Codecs then?.

Noop there in vars.xml. Ok for me it is time to throw in the towel, I'll find it out later.
Ok Bob is now taking Alice of hold and surprise surprise there is only one way audio, between Alice and Bob.

I didn't have the time to look with Wireshark where al the steams go so sorry for that.
Did anyone came across the same problem with Freeswitch 1.0.4. I've a 1.0.2 running and that is doing everything a ok.



Kind regards,

 Durk

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From ineyaa at gmail.com  Mon Oct 19 05:16:02 2009
From: ineyaa at gmail.com (ineya ineya)
Date: Mon, 19 Oct 2009 14:16:02 +0200
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
Message-ID: <7b8592a0910190516rafa4398q62c7a99ef703540f@mail.gmail.com>

Hi,

I installed the phonenix release, and changed IPv4, IPv6 addresses and SIP
port. Directory and dialplan should be default ones.
My phones are able to register when using just IPv4, and I can call from one
to another.

When I try the same thing with IPv6, the call to other phone fails with [
USER_NOT_REGISTERED], however I'm able to call 9999 and hear MOH.
Here you can find the log: http://pastebin.freeswitch.org/10757
Freeswitch is running on 10.80.62.40/2000:2::1
phones are 2000:2::1001 and 2000:2::1002

Am I doing something wrong? Maybe I forgot to configure something, but I
can't see what it is.

Any hints are appreciated.
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From frank at carmickle.com  Mon Oct 19 06:07:17 2009
From: frank at carmickle.com (Frank Carmickle)
Date: Mon, 19 Oct 2009 09:07:17 -0400
Subject: [Freeswitch-users] Status of ubuntu/debian packages.
In-Reply-To: <4ADAFC85.6000301@puzzled.xs4all.nl>
References: <4ABA47FB.2050100@gmail.com>
	<90823c940910180249u69f18aaamc383f9a5a0c65848@mail.gmail.com>
	<4ADAFC85.6000301@puzzled.xs4all.nl>
Message-ID: <20091019130717.GG13895@base.carmickle.com>

On Sun, Oct 18, Patrick List wrote:
> so there was no chance of your customized configs getting overwritten. 
> Example:
> /opt/freeswitch/etc/freeswitch-$svn_version-samples/
> 
> Maybe the maintainer of the debs can do something similar.

We are planning on putting all of the different configs in /usr/share/doc/freeswitch/examples.  Now if I could find the time to get this done.

--FC



From frank at carmickle.com  Mon Oct 19 06:09:47 2009
From: frank at carmickle.com (Frank Carmickle)
Date: Mon, 19 Oct 2009 09:09:47 -0400
Subject: [Freeswitch-users] Status of ubuntu/debian packages.
In-Reply-To: <90823c940910181154w615b9ac5v89a30a3a3288e619@mail.gmail.com>
References: <4ABA47FB.2050100@gmail.com>
	<90823c940910180249u69f18aaamc383f9a5a0c65848@mail.gmail.com>
	<20091018164000.GD13895@base.carmickle.com>
	<90823c940910181154w615b9ac5v89a30a3a3288e619@mail.gmail.com>
Message-ID: <20091019130946.GH13895@base.carmickle.com>

On Sun, Oct 18, Dmitry Bely wrote:
> On Sun, Oct 18, 2009 at 8:40 PM, Frank Carmickle <frank at carmickle.com> wrote:
> > On Sun, Oct 18, Dmitry Bely wrote:
> >> William,
> >>
> >> When I upgrade freeswitch-config package, all my customized config
> >> files are silently overwritten with the default ones from .deb
> >> package. This bug is quite annoying.
> >
> > Very sorry. ?I can try to include some code that detects if your moving from the old versions that don't have a config package to the new stuff that does.
> 
> I did have the previous version of freeswitch-config installed but
> nevertheless the files were lost.

Well I am very sorry.  I wish I new why this happened for you.  I have not experienced this ever.  I will do some testing.

--FC



From mike at jerris.com  Mon Oct 19 06:10:47 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Oct 2009 09:10:47 -0400
Subject: [Freeswitch-users] NAT problems migrating from Version
	1.0.trunk (13168M) to Version 1.0.trunk (15166)
In-Reply-To: <1255922089.4653.1340747457@webmail.messagingengine.com>
References: <1255922089.4653.1340747457@webmail.messagingengine.com>
Message-ID: <9713E1DA-B66B-4A01-8864-49F71F8FF4EE@jerris.com>

we added some params for new better automatic nat handling, grep the  
new defailt configs for localnet and you will find what you are missing.

Mike

On Oct 18, 2009, at 11:14 PM, Chris Fowler wrote:

> I've tried all sorts of debug and parameter changes over the weekend,
> but still can't figure out the correct solution.
>
> If I disable timers on the sip profile then all works fine.
> <param name="enable-timer" value="false"/>
>
> But that seems like a hack; not a correct solution.  With the build
> 13168M (which is pre the new NAT functionality) everything worked  
> fine.
>
> The SIP trace shows the phones and FreeSWITCH happily exchanging  
> NOTIFY
> and "200 OK" messages.  Audio's working - just calls timeout after 100
> seconds with RECOVERY_ON_TIMER_EXPIRE.  Is enforcement of this timer  
> new
> functionality - and really just exposing a problem I've always had
> before?
>
> The config is (50 Polycom Phones -> NAT -> Internet -> Amazon EC2)
>
> I would really appreciate some pointers on what to look for;  
> additional
> trace that might reveal something.
>
> Thanks, Chris.
>
>
> On Fri, 16 Oct 2009 20:06:52 -0700, "Chris Fowler" <chris at fowler.cc>
> said:
>> Hi,
>>
>> We've been using 13168M in production for some time now (works  
>> great).
>> I want to get us onto the latest build but am having problems getting
>> NAT to work.
>>
>> Phones can register; can dial 9999 test #, but after 100 seconds the
>> call is disconnected with error:
>> 2009-10-16 19:52:26.936618 [NOTICE] sofia.c:4038 Hangup
>> sofia/internal/1001 at myhost.mydomain.com [CS_EXECUTE]
>> [RECOVERY_ON_TIMER_EXPIRE]
>>
>> I took the standard internal.xml and vars.xml files from the new  
>> build
>> and made the following modifications - which worked previously:
>>
>> modify conf/vars.xml and update
>>  <X-PRE-PROCESS cmd="set" data="domain=myhost.mydomain.com"/>
>>
>>  <X-PRE-PROCESS cmd="set" data="bind_server_ip=1.2.3.4"/>
>>  <X-PRE-PROCESS cmd="set" data="external_rtp_ip=1.2.3.4"/>
>>  <X-PRE-PROCESS cmd="set" data="external_sip_ip=1.2.3.4"/>
>>
>> Modify conf/sip_profiles/internal.xml
>> <param name="aggressive-nat-detection" value="true"/>  <param
>> name="ext-rtp-ip" value="$${external_rtp_ip}"/>  <param
>> name="ext-sip-ip" value="$${external_sip_ip}"/>  <param
>> name="NDLB-received-in-nat-reg-contact" value="true"/>  <param
>> name="NDLB-force-rport" value="true"/>  <param
>> name="NDLB-broken-auth-hash" value="true"/>
>>
>> The big difference I note is that on PRODUCTION (which works) "sofia
>> status profile internal" yields:
>> URL                     sip:mod_sofia at 1.2.3.4:5060
>> BIND-URL                 
>> sip:mod_sofia at 1.2.3.4:5060;maddr=10.250.35.224
>>
>> But on Test I see:
>> URL                     sip:mod_sofia at 10.250.66.210:5060
>> BIND-URL                sip:mod_sofia at 10.250.66.210:5060
>>
>> Any ideas?
>>
>> Thanks, Chris.




From srinivas.ksvreddy at gmail.com  Mon Oct 19 06:14:06 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Mon, 19 Oct 2009 18:44:06 +0530
Subject: [Freeswitch-users] Freeswitch.managed
Message-ID: <f8af5740910190614u75a98403icd5908c4a0663469@mail.gmail.com>

Hi,

How can i use freeswitch.managed project.  what are the parameters for
calling Execute method? and how can i call?
any help

-- 
Srinivasula Reddy K
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From mike at jerris.com  Mon Oct 19 06:19:25 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Oct 2009 09:19:25 -0400
Subject: [Freeswitch-users] uuid_send_dtmf fails (was: conference call)
In-Reply-To: <02C2EFD4CF8D4ACF8EEC0F77C4D75028@abisoft.biz>
References: <064A5201C10D4D20AC76EB93CA5EA4B5@abisoft.biz><ebbc2d550910140838v3125579ev9fa16e3e83d25770@mail.gmail.com><BE4EDDA84CB64AA98F18D412B51912FC@abisoft.biz>
	<b39273530910150504l5c7411b9g39b9ed33c592e85f@mail.gmail.com>
	<02C2EFD4CF8D4ACF8EEC0F77C4D75028@abisoft.biz>
Message-ID: <414DC781-A0F6-418E-9AAB-FA4307634932@jerris.com>

There is an event you can send as well to switch them, it your trying  
to switch it via event socket, that should be better, its not on the  
wiki, but

a session message with
eavesdrop-command header with data as the same as dtmf
should do the trick

Mike


On Oct 16, 2009, at 11:54 AM, Nikita Belov wrote:

> Yes, it is what I need. But now I have problem with sending dtmf.  
> Here what
> I've done:
> [root at centos4-4-vm ~]# telnet localhost 8021
> Trying 127.0.0.1...
> Connected to localhost.localdomain (127.0.0.1).
> Escape character is '^]'.
> Content-Type: auth/request
>
> auth ClueCon
>
> Content-Type: command/reply
> Reply-Text: +OK accepted
>
> api originate user/1007 at master.agent.starpoundtech.net &park()
>
> Content-Type: api/response
> Content-Length: 41
>
> +OK bba3b45a-4cc1-48af-a15d-1052d5f11371
>
> SendMsg bba3b45a-4cc1-48af-a15d-1052d5f11371
> call-command: execute
> execute-app-name: eavesdrop
> execute-app-arg: cd99f999-9b47-457e-8439-1d366e015b8c
>
> Content-Type: command/reply
> Reply-Text: +OK
>
> Here I had started to hear A and B. Here what I saw in FS log:
> 2009-10-18 03:22:47 [DEBUG] switch_core_session.c:706
> switch_core_session_queue_private_event() Send signal
> sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes [BREAK]
> 2009-10-18 03:22:47 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event()
> sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes Command Execute
> eavesdrop(cd99f999-9b47-457e-8439-1d366e015b8c)
> 2009-10-18 03:22:47 [DEBUG] switch_core_media_bug.c:297
> switch_core_media_bug_add() Attaching BUG to
> sofia/internal/1006 at master.agent.starpoundtech.net
> 2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234
> switch_core_session_read_frame()
> sofia/internal/1006 at master.agent.starpoundtech.net receive message
> [TRANSCODING_NECESSARY]
> 2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234
> switch_core_session_read_frame()
> sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes receive message
> [TRANSCODING_NECESSARY]
>
> Then I run command:
> api uuid_send_dtmf bba3b45a-4cc1-48af-a15d-1052d5f11371 1
>
> Content-Type: api/response
> Content-Length: 14
>
> -ERR no reply
>
> Log:
> 2009-10-18 03:24:01 [DEBUG] switch_core_io.c:1190
> switch_core_session_send_dtmf_string()
> sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes send dtmf
> digit=1 ms=250 samples=2000
> 2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1282 do_2833() Send start  
> packet
> for [1] ts=2241760 dur=160/160/2000 seq=21346
> 2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle  
> packet
> for [1] ts=2241760 dur=320/320/2000 seq=21347
> 2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle  
> packet
> for [1] ts=2241760 dur=480/480/2000 seq=21348
> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle  
> packet
> for [1] ts=2241760 dur=640/640/2000 seq=21349
> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle  
> packet
> for [1] ts=2241760 dur=800/800/2000 seq=21350
> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle  
> packet
> for [1] ts=2241760 dur=960/960/2000 seq=21351
> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle  
> packet
> for [1] ts=2241760 dur=1120/1120/2000 seq=21352
> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle  
> packet
> for [1] ts=2241760 dur=1280/1280/2000 seq=21353
> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle  
> packet
> for [1] ts=2241760 dur=1440/1440/2000 seq=21354
> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle  
> packet
> for [1] ts=2241760 dur=1600/1600/2000 seq=21355
> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle  
> packet
> for [1] ts=2241760 dur=1760/1760/2000 seq=21356
> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle  
> packet
> for [1] ts=2241760 dur=1920/1920/2000 seq=21357
> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end  
> packet for
> [1] ts=2241760 dur=2080/2080/2000 seq=21358
> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end  
> packet for
> [1] ts=2241760 dur=2080/2080/2000 seq=21359
> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end  
> packet for
> [1] ts=2241760 dur=2080/2080/2000 seq=21360
>
> But both A and B couldn't hear me.
> Btw, after I had send dtmf "1" manually from my phone. B started to  
> hear me.
> There was this record in log:
> 2009-10-18 03:47:55 [DEBUG] switch_rtp.c:1767 switch_rtp_dequeue_dtmf 
> () RTP
> RECV DTMF 1:2240
>
> Does anybody know, what had I done wrong?
>
> ___
>
> Thanks, Nikita
>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org  
>> [mailto:freeswitch-
>> users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker
>> Sent: Thursday, October 15, 2009 4:04 PM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] conference call
>>
>> The wiki has a pretty good explanation of how eavesdrop works.
>> Enabling a talk path to A or B or both A and B requires dtmf.
>>
>> So, if C hits the 1 button on the phone they can talk to the UUID you
>> bound the eavesdrop to.   2 talks to the other party.  3 talks to  
>> both
>> parties.
>>
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop
>>
>> Adding more api calls to control eavesdrop via event socket should be
>> possible, though I haven't looked at the code to see what it would
>> take...
>>
>> On Thu, Oct 15, 2009 at 3:16 AM, Nikita Belov <nbelov at abisoft.spb.ru>
>> wrote:
>>> Thanks for answer.
>>>
>>>
>>>
>>> As I've understood eavesdrop allows C to hear A and B, but A and B  
>>> will
>> not
>>> hear C. But in my case B SHOULD hear C. In my case A - client, B -  
>>> agent
>>> worked with client and C - coach/supervisor for B. Can I make such
>>> conference in FS using eavesdrop or other API command?
>>>
>>>
>>>
>>> ___
>>>
>>>
>>>
>>> Thanks, Nikita
>>>
>>>
>>>
>>> ________________________________
>>>
>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf  
>>> Of Joao
>>> Mesquita
>>> Sent: Wednesday, October 14, 2009 7:38 PM
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] conference call
>>>
>>>
>>>
>>> Look at eavesdrop on the wiki.
>>>
>>> JM
>>>
>>> 2009/10/14 Nikita Belov <nbelov at abisoft.spb.ru>
>>>
>>> HI all,
>>>
>>>
>>>
>>> I want to configure FS to make special conference call between three
>> users
>>> (A, B, C).  In this conference C will hear A and B, but A will  
>>> hear only
>> B.
>>> Can I make it using FS API commands? Does anybody know which  
>>> approach is
>>> better to use?
>>>
>>>
>>>
>>> ___
>>>
>>>
>>>
>>> Thanks, Nikita
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>>> users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>>> users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From srinivas.ksvreddy at gmail.com  Mon Oct 19 06:19:45 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Mon, 19 Oct 2009 18:49:45 +0530
Subject: [Freeswitch-users] Freeswitch.managed
In-Reply-To: <f8af5740910190614u75a98403icd5908c4a0663469@mail.gmail.com>
References: <f8af5740910190614u75a98403icd5908c4a0663469@mail.gmail.com>
Message-ID: <f8af5740910190619i32b83541s1ab238eebb87e757@mail.gmail.com>

Hi,

How can i use freeswitch.managed project.  what are the parameters for
calling Execute method? and how can i call?
any help

-- 
Srinivasula Reddy K
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From mike at jerris.com  Mon Oct 19 06:22:27 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Oct 2009 09:22:27 -0400
Subject: [Freeswitch-users] Freeswitch.managed
In-Reply-To: <f8af5740910190614u75a98403icd5908c4a0663469@mail.gmail.com>
References: <f8af5740910190614u75a98403icd5908c4a0663469@mail.gmail.com>
Message-ID: <7C661CE3-8D8F-4850-B751-D815F17DE49F@jerris.com>

Try starting out reading this.

http://wiki.freeswitch.org/wiki/Mod_managed

Mike

On Oct 19, 2009, at 9:14 AM, srinivasula reddy wrote:

> Hi,
>
> How can i use freeswitch.managed project.  what are the parameters  
> for calling Execute method? and how can i call?
> any help




From mike at jerris.com  Mon Oct 19 06:28:39 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Oct 2009 09:28:39 -0400
Subject: [Freeswitch-users] sofia gateways and linux multipath routing
In-Reply-To: <1255428585.11786.90.camel@francois.tc.commsmundi.com>
References: <1255428585.11786.90.camel@francois.tc.commsmundi.com>
Message-ID: <C0403714-EBBB-4758-8CF9-44E9D7E7F23C@jerris.com>

You need a sofia profile for each identity, if your using multiple  
external ip addresses, you will need a profile for each.  If you are  
using bgp or something of the sort and only using one external ip, you  
can use a single profile and route using standard routing.

Mike

On Oct 13, 2009, at 6:09 AM, Fran?ois Delawarde wrote:

> Hello all,
>
> I'm interested in using mod_sofia with multiple Internet connections
> (configured as a unique load-balancing route using multipath).
>
> One solution would be to define a different profile for each  
> connection,
> but it would be more practical having a unique external profile that
> would automatically handle everything (detecting multiple public IPs  
> and
> selecting the right one for a call, being able to select the router  
> for
> gateway registration...).
>
> Routing table:
> 192.168.10.0/24 dev eth0  proto kernel  scope link  src 192.168.10.1
> 192.168.1.0/24 dev eth1  proto kernel  scope link  src 192.168.1.2
> 192.168.2.0/24 dev eth2  proto kernel  scope link  src 192.168.2.2
> default  proto static
> 	nexthop via 192.168.1.1  dev eth1 weight 1
> 	nexthop via 192.168.2.1  dev eth2 weight 1
>
> Both default routers (192.168.1.1 and 192.168.2.1) would have a  
> distinct
> public IP.
>
> Several questions cross my mind:
> - Can a unique sofia profile be bound to multiple IPs (not 0.0.0.0)?
>
> - How would FS behave with a unique external profile in that  
> situation?
>   * Would FS reply to an incoming call using the same router it came
> from forcing packet source address?
>   * Would FS stick to a unique router for all flows of an outgoing  
> call
> (SIP, RTP, UDPTL)?
>   * Can I force a gateway to use a given router (for calls,
> registration, ...)?
>   * Would the NAT system (using stun or auto-nat) work in that
> situation, or does it assume only one default router (and a unique
> public IP) exists per profile?
>
> - Knowing the above, would it be necessary to use a different profile
> for each router/interface, and define the same gateway in each of  
> these?
>
> - Tricky question: What if multiple routers are on the same
> network/interface (192.168.1.1, 192.168.1.2, ...)?
>
>
> Thanks in advance,
> Fran?ois.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From mike at jerris.com  Mon Oct 19 06:30:02 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Oct 2009 09:30:02 -0400
Subject: [Freeswitch-users] TLS client authentification
In-Reply-To: <87747c930910151220p474e079bt609019e562698dfb@mail.gmail.com>
References: <87747c930910151220p474e079bt609019e562698dfb@mail.gmail.com>
Message-ID: <0FA87111-29F1-44E1-BE3A-2B8B2F841704@jerris.com>

I can't recall if we ever exposed an option for this, take a look at  
sofia-sip and see if they have a tag to enable this, if so it would  
probably be a fairly simple patch to add.

Mike

On Oct 15, 2009, at 3:20 PM, Alexandre Savard wrote:

> Hi,
>
> Does Freeswitch support TLS Client-Authenticated handshake. Openssl  
> does, but it has to be enabled in order to send the certificate  
> request to the client.
>
> I tested simple TLS hanshaking and it wotks well.




From mike at jerris.com  Mon Oct 19 06:32:28 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Oct 2009 09:32:28 -0400
Subject: [Freeswitch-users] validating dtmf digits received
In-Reply-To: <33c87fa30910182153t23e52fa0p2b967b3843eb1175@mail.gmail.com>
References: <33c87fa30910182153t23e52fa0p2b967b3843eb1175@mail.gmail.com>
Message-ID: <81433357-3203-4208-98C1-FF3FE2AC2D1E@jerris.com>

inline is new, it won't work unless your using recent trunk.  That  
being said, read is not being run inline, so the set is actually being  
run before digits_dialed is set.  You will most likely need to use  
transfer in this situation.

Mike

On Oct 19, 2009, at 12:53 AM, Mark Campbell-Smith wrote:

> Hi!
>
> I simply want to validate the dtmf digits I read from a user.    From
> the wiki, it appears I need to use inline="true" when setting the
> variable so it can be used directly within the same extension.
>
> What have I done wrong below?   I have tried many different
> alternatives, but the second condition field, which is meant to match
> the dtmf digits received (in this case 0000) is never matched, and the
> anti-action is called instead.
>
> :
>      <some code here>
>      <action application="read" data="1 10
> ivr/ivr-please_enter_pin_followed_by_pound.wav res 10000 9"/>
>       <action application="phrase" data="spell,${res}"/>
>       <action inline="true" application="set" data="code=$ 
> {digits_dialed}"/>
>       <!-- <action inline="true" application="set" data="code=$ 
> {res}"/> -->
>  </condition>
>  <condition field="digits_dialed" expression="^0000$">
>  <!-- <condition field="${code}" expression="^0000$"> -->
>  <!-- <condition field="${res}" expression="^0000$"> -->
>        <some code here>
>       <anti-action application="hangup"/>
>
> Thanks!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From mike at jerris.com  Mon Oct 19 06:34:47 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Oct 2009 09:34:47 -0400
Subject: [Freeswitch-users] How to enforce freeswitch replying to the
	source port instead of to the one specified in v/m header
	parametes (Symmetric NAT)
In-Reply-To: <10128ef10910190023t17574fafic892bd4105cc1703@mail.gmail.com>
References: <10128ef10910190023t17574fafic892bd4105cc1703@mail.gmail.com>
Message-ID: <06D9255C-2D72-4BD1-A17E-922C55A76C56@jerris.com>

This is what force-rport is supposed to do.  That being said, I can't  
tell from your trace where it is actually going to, just what it says  
in the packet, which can be different.

Mike

On Oct 19, 2009, at 3:23 AM, Tzury Bar Yochay wrote:

> Hi,
>
> I am struggling with a cellular operator which removes the rport from
> the SIP messages sent by  the client.
> During the troubleshooting process I have been playing with the NDLB
> parameters I found mentioned at the fs-wiki
>
> <param name="NDLB-force-rport" value="true"/>
> <param name="NDLB-received-in-nat-reg-contact" value="true"/>
>
> as well as with sip-force-contact at the extension xml config file.
>
> <variables>
>    <variable name="sip-force-contact" value="NDLB-connectile- 
> dysfunction"/>
> </variables>
>
> I Want to make freeswitch replying to the source port instead of the
> port supplied by the client, that is
> to port 12543 instead of to the 7608 as you can see below some pcap
> dumps of the case.
>
> I am not sure if that would solve thep roblem and am open to other
> suggestions as well.
>
> Thanks in advance,
> Tzury
>
> pcap dumps below:
>
> ### REGISTER ###
> User Datagram Protocol, Src Port: 12543 (12543), Dst Port: sip (5060)
> REGISTER sip:example.net SIP/2.0
> v: SIP/2.0/UDP  
> 212.154.128.222:7608;branch=z9hG4bKPj41fa0abd044c35c75598d561b2f93167
> Max-Forwards: 70
> f: <sip:1013 at example.net>;tag=9194adecbf8e8179530eb589e83631db
> t: <sip:1013 at example.net>
> i: ebf2e3c254517938f90f33d8fd89326d
> CSeq: 17355 REGISTER
> m: <sip:1013 at 212.154.128.222:7608>
> Expires: 300
> l:  0
>
> ### 401 UNAUTHORIZED ###
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP
> 212.154.128.222:7608;branch=z9hG4bKPj41fa0abd044c35c75598d561b2f93167
> From: <sip:1013 at example.net>;tag=9194adecbf8e8179530eb589e83631db
> To: <sip:1013 at example.net>;tag=166HraN0B149g
> Call-ID: ebf2e3c254517938f90f33d8fd89326d
> CSeq: 17355 REGISTER
> User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
> Supported: timer, precondition, path, replaces
> WWW-Authenticate: Digest realm="example.net",
> nonce="f2f1370c-bc7c-11de-b000-fbe81e221ab4", algorithm=MD5,
> qop="auth"
> Content-Length: 0




From mike at jerris.com  Mon Oct 19 06:47:41 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Oct 2009 09:47:41 -0400
Subject: [Freeswitch-users] Freeswitch 1.0.4 problems with music on hold
In-Reply-To: <4adc679c.a013f30a.539b.ffffb45dSMTPIN_ADDED@mx.google.com>
References: <4adc679c.a013f30a.539b.ffffb45dSMTPIN_ADDED@mx.google.com>
Message-ID: <359F2E83-6A5A-4CE9-B12D-118D8E9BA325@jerris.com>

Try out trunk and see if this issue is resolved please.

mike

On Oct 19, 2009, at 3:11 AM, Durk de Beer wrote:

> Hello,
> This is something I came across on Freeswitch 1.0.4
> First let me explain what I'm trying to do.
> I want Free-Switch to behave as a proxy so in the settings section  
> of Sofia.conf.xml I use
> <param name="media-option" value="bypass-media-after-att-xfer"/>
> and
> <param name="inbound-bypass-media" value="true"/>
> As fare as I am able to follow the RTP stream it is passing by  
> Freeswitch making it a proxy.
> So far so good. Now I want to serv music on hold if a call is put on  
> hold.
> The only way I am able to find for Freeswitch to serv music on hold  
> (MOH) is for it to be in media.
> Ok we've got a command for this situation namely
> <param name="media-option" value="resume-media-on-hold"/>
> Ok if I put the last parram command into the Sofia.conf.xml  
> Freeswitch is starting to behave funny.
> Let me try to explain this by an example. Let suppose Alice is  
> calling Bob without the last command, the ?resume media on hold?.
> If Bob puts Alice on hold al is silent on Bob's and Alice's phone.
> If Bob takes Alice of hold there's two way audio.
> Ok no I put the command in sofia.onf.xml, the ?resume media on hold?.
> Now Alice is calling Bob again and Bob is putting her on hold again.
> You are expecting MOH on this moment but it isn't, all there is are  
> the sounds of silence.
> Ok mistake made checking local streams, and they are there?. Yes it  
> is.
> Ok other mistake, didn't put the ?hold-music?command in  
> sofia.conf.xml. Noop it's in there so there must be MOH. Codecs then?.
> Noop there in vars.xml. Ok for me it is time to throw in the towel,  
> I'll find it out later.
> Ok Bob is now taking Alice of hold and surprise surprise there is  
> only one way audio, between Alice and Bob.
> I didn't have the time to look with Wireshark where al the steams go  
> so sorry for that.
> Did anyone came across the same problem with Freeswitch 1.0.4. I've  
> a 1.0.2 running and that is doing everything a ok.

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From helmut.kuper at ewetel.de  Mon Oct 19 07:47:37 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Mon, 19 Oct 2009 16:47:37 +0200
Subject: [Freeswitch-users] Qustion about INFO messages
	after	Connect/Answer
In-Reply-To: <191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>
References: <4AD859A4.5000101@ewetel.de>
	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>
Message-ID: <4ADC7C09.6070503@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello Anthony,

I updated and restarted my test FS to "FreeSWITCH Version 1.0.trunk
(15174M)". Callee's experience didn't change:

> 1. Phone rings: caller's displayname
> 2. Callee picks up: switching from dislayname to unknown
> 3. Switching from unknown to displayname

I used the two chvars you mentioned, set them via "set" and as well via
"export" but no change (neither on caller's nor in callee's display nor
in SIP INFO messages)


My dialplan portion for this is:
    <extension name="Local_Extension">
      <condition field="${ET_is_local}" expression="^true$">
        <action application="set"
data="dialed_extension=${destination_number}"/>
        <action application="set" data="sip_callee_id_number=1111"/>
        <action application="set" data="sip_callee_id_name=hubu"/>
        <action application="export" data="sip_callee_id_number=1111"/>
        <action application="export" data="sip_callee_id_name=hubu"/>
        <action application="info"/>
[...]


Here is the output of the info app after setting those chvars:

[INFO] mod_dptools.c:961 CHANNEL_DATA:
Channel-State: [CS_EXECUTE]
Channel-State-Number: [4]
Channel-Name: [sofia/internal/1001 at 85.16.246.12:5061]
Unique-ID: [4b143750-bcbd-11de-9f91-c9cd82739033]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Answer-State: [ringing]
Caller-Username: [1001]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [1001 an PBX1]
Caller-Caller-ID-Number: [1001]
Caller-Network-Addr: [85.16.245.206]
Caller-Destination-Number: [1000]
Caller-Unique-ID: [4b143750-bcbd-11de-9f91-c9cd82739033]
Caller-Source: [mod_sofia]
Caller-Context: [default]
Caller-RDNIS: [1000]
Caller-Channel-Name: [sofia/internal/1001 at 85.16.246.12:5061]
Caller-Profile-Index: [2]
Caller-Profile-Created-Time: [1255963206242587]
Caller-Channel-Created-Time: [1255963206214959]
Caller-Channel-Answered-Time: [0]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
variable_sip_received_ip: [85.16.245.206]
variable_sip_received_port: [1024]
variable_sip_via_protocol: [udp]
variable_sip_authorized: [true]
variable_sip_from_user: [1001]
variable_sip_from_port: [5061]
variable_sip_from_uri: [1001 at 85.16.246.12:5061]
variable_sip_from_host: [85.16.246.12]
variable_sip_from_user_stripped: [1001]
variable_sip_from_tag: [snfuiue6ga]
variable_sofia_profile_name: [internal]
variable_sip_req_params: [user=phone]
variable_sip_req_user: [1000]
variable_sip_req_port: [5061]
variable_sip_req_uri: [1000 at 85.16.246.12:5061]
variable_sip_req_host: [85.16.246.12]
variable_sip_to_params: [user=phone]
variable_sip_to_user: [1000]
variable_sip_to_port: [5061]
variable_sip_to_uri: [1000 at 85.16.246.12:5061]
variable_sip_to_host: [85.16.246.12]
variable_sip_contact_params: [line=eg3wp69a]
variable_sip_contact_user: [1001]
variable_sip_contact_port: [1024]
variable_sip_contact_uri: [1001 at 85.16.245.206:1024]
variable_sip_contact_host: [85.16.245.206]
variable_channel_name: [sofia/internal/1001 at 85.16.246.12:5061]
variable_sip_call_id: [3c2d2d8f9a49-edzr2i2iezjp]
variable_sip_user_agent: [snom820/8.2.16]
variable_sip_via_host: [85.16.245.206]
variable_sip_via_port: [1024]
variable_sip_via_rport: [1024]
variable_presence_id: [1001 at 85.16.246.12]
variable_sip_h_X-Serialnumber: [0004134002CB]
variable_sip_h_P-Key-Flags: [resolution="31x13", keys="4"]
variable_switch_r_sdp: [v=0
o=root 1331667919 1331667919 IN IP4 85.16.245.206
s=call
c=IN IP4 85.16.245.206
t=0 0
m=audio 62882 RTP/SAVP 0 8 9 99 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:m6fas/KsLF57r9RnU7X0WEWeJw9Y6+a66YUIf9Dc
a=ptime:20
m=audio 62882 RTP/AVP 0 8 9 99 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
]
variable_ep_codec_string: [G722 at 8000h@20i,PCMA at 8000h@20i]
variable_endpoint_disposition: [DELAYED NEGOTIATION]
variable_ET_is_local: [true]
variable_max_forwards: [69]
variable_domain_name: [85.16.246.12]
variable_dialed_extension: [1000]
variable_sip_callee_id_number: [1111]
variable_sip_callee_id_name: [hubu]
variable_export_vars: [sip_callee_id_number,sip_callee_id_name]
variable_current_application: [info]


On 16.10.2009 18:18, Anthony Minessale wrote:
> 1) you should update again there were a few issues.
> 2) you can set the variable sip_callee_id_name and sip_callee_id number
> on the inbound leg before you answer to control what it says.

regards
Helmut
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From t.mahe at telemaque.fr  Mon Oct 19 07:55:58 2009
From: t.mahe at telemaque.fr (=?ISO-8859-1?Q?Tristan_Mah=E9?=)
Date: Mon, 19 Oct 2009 16:55:58 +0200
Subject: [Freeswitch-users] Qustion about INFO
	messages	after	Connect/Answer
In-Reply-To: <4ADC7C09.6070503@ewetel.de>
References: <4AD859A4.5000101@ewetel.de>	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>
	<4ADC7C09.6070503@ewetel.de>
Message-ID: <4ADC7DFE.5000305@telemaque.fr>

Hi Helmut,

Just to add my 2 cents to the discussion, I have the same behaviour there...

Regards,

Gled.

Helmut Kuper a ?crit :
> Hello Anthony,
>
> I updated and restarted my test FS to "FreeSWITCH Version 1.0.trunk
> (15174M)". Callee's experience didn't change:
>
> > 1. Phone rings: caller's displayname
> > 2. Callee picks up: switching from dislayname to unknown
> > 3. Switching from unknown to displayname
>
> I used the two chvars you mentioned, set them via "set" and as well via
> "export" but no change (neither on caller's nor in callee's display nor
> in SIP INFO messages)
>
>
> My dialplan portion for this is:
>     <extension name="Local_Extension">
>       <condition field="${ET_is_local}" expression="^true$">
>         <action application="set"
> data="dialed_extension=${destination_number}"/>
>         <action application="set" data="sip_callee_id_number=1111"/>
>         <action application="set" data="sip_callee_id_name=hubu"/>
>         <action application="export" data="sip_callee_id_number=1111"/>
>         <action application="export" data="sip_callee_id_name=hubu"/>
>         <action application="info"/>
> [...]
>
>
> Here is the output of the info app after setting those chvars:
>
> [INFO] mod_dptools.c:961 CHANNEL_DATA:
> Channel-State: [CS_EXECUTE]
> Channel-State-Number: [4]
> Channel-Name: [sofia/internal/1001 at 85.16.246.12:5061]
> Unique-ID: [4b143750-bcbd-11de-9f91-c9cd82739033]
> Call-Direction: [inbound]
> Presence-Call-Direction: [inbound]
> Answer-State: [ringing]
> Caller-Username: [1001]
> Caller-Dialplan: [XML]
> Caller-Caller-ID-Name: [1001 an PBX1]
> Caller-Caller-ID-Number: [1001]
> Caller-Network-Addr: [85.16.245.206]
> Caller-Destination-Number: [1000]
> Caller-Unique-ID: [4b143750-bcbd-11de-9f91-c9cd82739033]
> Caller-Source: [mod_sofia]
> Caller-Context: [default]
> Caller-RDNIS: [1000]
> Caller-Channel-Name: [sofia/internal/1001 at 85.16.246.12:5061]
> Caller-Profile-Index: [2]
> Caller-Profile-Created-Time: [1255963206242587]
> Caller-Channel-Created-Time: [1255963206214959]
> Caller-Channel-Answered-Time: [0]
> Caller-Channel-Progress-Time: [0]
> Caller-Channel-Progress-Media-Time: [0]
> Caller-Channel-Hangup-Time: [0]
> Caller-Channel-Transfer-Time: [0]
> Caller-Screen-Bit: [true]
> Caller-Privacy-Hide-Name: [false]
> Caller-Privacy-Hide-Number: [false]
> variable_sip_received_ip: [85.16.245.206]
> variable_sip_received_port: [1024]
> variable_sip_via_protocol: [udp]
> variable_sip_authorized: [true]
> variable_sip_from_user: [1001]
> variable_sip_from_port: [5061]
> variable_sip_from_uri: [1001 at 85.16.246.12:5061]
> variable_sip_from_host: [85.16.246.12]
> variable_sip_from_user_stripped: [1001]
> variable_sip_from_tag: [snfuiue6ga]
> variable_sofia_profile_name: [internal]
> variable_sip_req_params: [user=phone]
> variable_sip_req_user: [1000]
> variable_sip_req_port: [5061]
> variable_sip_req_uri: [1000 at 85.16.246.12:5061]
> variable_sip_req_host: [85.16.246.12]
> variable_sip_to_params: [user=phone]
> variable_sip_to_user: [1000]
> variable_sip_to_port: [5061]
> variable_sip_to_uri: [1000 at 85.16.246.12:5061]
> variable_sip_to_host: [85.16.246.12]
> variable_sip_contact_params: [line=eg3wp69a]
> variable_sip_contact_user: [1001]
> variable_sip_contact_port: [1024]
> variable_sip_contact_uri: [1001 at 85.16.245.206:1024]
> variable_sip_contact_host: [85.16.245.206]
> variable_channel_name: [sofia/internal/1001 at 85.16.246.12:5061]
> variable_sip_call_id: [3c2d2d8f9a49-edzr2i2iezjp]
> variable_sip_user_agent: [snom820/8.2.16]
> variable_sip_via_host: [85.16.245.206]
> variable_sip_via_port: [1024]
> variable_sip_via_rport: [1024]
> variable_presence_id: [1001 at 85.16.246.12]
> variable_sip_h_X-Serialnumber: [0004134002CB]
> variable_sip_h_P-Key-Flags: [resolution="31x13", keys="4"]
> variable_switch_r_sdp: [v=0
> o=root 1331667919 1331667919 IN IP4 85.16.245.206
> s=call
> c=IN IP4 85.16.245.206
> t=0 0
> m=audio 62882 RTP/SAVP 0 8 9 99 3 18 4 101
> a=rtpmap:0 pcmu/8000
> a=rtpmap:8 pcma/8000
> a=rtpmap:9 g722/8000
> a=rtpmap:99 g726-32/8000
> a=rtpmap:3 gsm/8000
> a=rtpmap:18 g729/8000
> a=rtpmap:4 g723/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=crypto:1 AES_CM_128_HMAC_SHA1_32
> inline:m6fas/KsLF57r9RnU7X0WEWeJw9Y6+a66YUIf9Dc
> a=ptime:20
> m=audio 62882 RTP/AVP 0 8 9 99 3 18 4 101
> a=rtpmap:0 pcmu/8000
> a=rtpmap:8 pcma/8000
> a=rtpmap:9 g722/8000
> a=rtpmap:99 g726-32/8000
> a=rtpmap:3 gsm/8000
> a=rtpmap:18 g729/8000
> a=rtpmap:4 g723/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> ]
> variable_ep_codec_string: [G722 at 8000h@20i,PCMA at 8000h@20i]
> variable_endpoint_disposition: [DELAYED NEGOTIATION]
> variable_ET_is_local: [true]
> variable_max_forwards: [69]
> variable_domain_name: [85.16.246.12]
> variable_dialed_extension: [1000]
> variable_sip_callee_id_number: [1111]
> variable_sip_callee_id_name: [hubu]
> variable_export_vars: [sip_callee_id_number,sip_callee_id_name]
> variable_current_application: [info]
>
>
> On 16.10.2009 18:18, Anthony Minessale wrote:
> > 1) you should update again there were a few issues.
> > 2) you can set the variable sip_callee_id_name and sip_callee_id number
> > on the inbound leg before you answer to control what it says.
>
> regards
> Helmut

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From anthony.minessale at gmail.com  Mon Oct 19 08:00:49 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 19 Oct 2009 10:00:49 -0500
Subject: [Freeswitch-users] Qustion about INFO messages after
	Connect/Answer
In-Reply-To: <4ADC7C09.6070503@ewetel.de>
References: <4AD859A4.5000101@ewetel.de>
	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>
	<4ADC7C09.6070503@ewetel.de>
Message-ID: <191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>

you only need to "set" it on the inbound leg and you must answer and bridge
it somewhere.


On Mon, Oct 19, 2009 at 9:47 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello Anthony,
>
> I updated and restarted my test FS to "FreeSWITCH Version 1.0.trunk
> (15174M)". Callee's experience didn't change:
>
> > 1. Phone rings: caller's displayname
> > 2. Callee picks up: switching from dislayname to unknown
> > 3. Switching from unknown to displayname
>
> I used the two chvars you mentioned, set them via "set" and as well via
> "export" but no change (neither on caller's nor in callee's display nor
> in SIP INFO messages)
>
>
> My dialplan portion for this is:
>    <extension name="Local_Extension">
>      <condition field="${ET_is_local}" expression="^true$">
>        <action application="set"
> data="dialed_extension=${destination_number}"/>
>        <action application="set" data="sip_callee_id_number=1111"/>
>        <action application="set" data="sip_callee_id_name=hubu"/>
>        <action application="export" data="sip_callee_id_number=1111"/>
>        <action application="export" data="sip_callee_id_name=hubu"/>
>        <action application="info"/>
> [...]
>
>
> Here is the output of the info app after setting those chvars:
>
> [INFO] mod_dptools.c:961 CHANNEL_DATA:
> Channel-State: [CS_EXECUTE]
> Channel-State-Number: [4]
> Channel-Name: [sofia/internal/1001 at 85.16.246.12:5061]
> Unique-ID: [4b143750-bcbd-11de-9f91-c9cd82739033]
> Call-Direction: [inbound]
> Presence-Call-Direction: [inbound]
> Answer-State: [ringing]
> Caller-Username: [1001]
> Caller-Dialplan: [XML]
> Caller-Caller-ID-Name: [1001 an PBX1]
> Caller-Caller-ID-Number: [1001]
> Caller-Network-Addr: [85.16.245.206]
> Caller-Destination-Number: [1000]
> Caller-Unique-ID: [4b143750-bcbd-11de-9f91-c9cd82739033]
> Caller-Source: [mod_sofia]
> Caller-Context: [default]
> Caller-RDNIS: [1000]
> Caller-Channel-Name: [sofia/internal/1001 at 85.16.246.12:5061]
> Caller-Profile-Index: [2]
> Caller-Profile-Created-Time: [1255963206242587]
> Caller-Channel-Created-Time: [1255963206214959]
> Caller-Channel-Answered-Time: [0]
> Caller-Channel-Progress-Time: [0]
> Caller-Channel-Progress-Media-Time: [0]
> Caller-Channel-Hangup-Time: [0]
> Caller-Channel-Transfer-Time: [0]
> Caller-Screen-Bit: [true]
> Caller-Privacy-Hide-Name: [false]
> Caller-Privacy-Hide-Number: [false]
> variable_sip_received_ip: [85.16.245.206]
> variable_sip_received_port: [1024]
> variable_sip_via_protocol: [udp]
> variable_sip_authorized: [true]
> variable_sip_from_user: [1001]
> variable_sip_from_port: [5061]
> variable_sip_from_uri: [1001 at 85.16.246.12:5061]
> variable_sip_from_host: [85.16.246.12]
> variable_sip_from_user_stripped: [1001]
> variable_sip_from_tag: [snfuiue6ga]
> variable_sofia_profile_name: [internal]
> variable_sip_req_params: [user=phone]
> variable_sip_req_user: [1000]
> variable_sip_req_port: [5061]
> variable_sip_req_uri: [1000 at 85.16.246.12:5061]
> variable_sip_req_host: [85.16.246.12]
> variable_sip_to_params: [user=phone]
> variable_sip_to_user: [1000]
> variable_sip_to_port: [5061]
> variable_sip_to_uri: [1000 at 85.16.246.12:5061]
> variable_sip_to_host: [85.16.246.12]
> variable_sip_contact_params: [line=eg3wp69a]
> variable_sip_contact_user: [1001]
> variable_sip_contact_port: [1024]
> variable_sip_contact_uri: [1001 at 85.16.245.206:1024]
> variable_sip_contact_host: [85.16.245.206]
> variable_channel_name: [sofia/internal/1001 at 85.16.246.12:5061]
> variable_sip_call_id: [3c2d2d8f9a49-edzr2i2iezjp]
> variable_sip_user_agent: [snom820/8.2.16]
> variable_sip_via_host: [85.16.245.206]
> variable_sip_via_port: [1024]
> variable_sip_via_rport: [1024]
> variable_presence_id: [1001 at 85.16.246.12]
> variable_sip_h_X-Serialnumber: [0004134002CB]
> variable_sip_h_P-Key-Flags: [resolution="31x13", keys="4"]
> variable_switch_r_sdp: [v=0
> o=root 1331667919 1331667919 IN IP4 85.16.245.206
> s=call
> c=IN IP4 85.16.245.206
> t=0 0
> m=audio 62882 RTP/SAVP 0 8 9 99 3 18 4 101
> a=rtpmap:0 pcmu/8000
> a=rtpmap:8 pcma/8000
> a=rtpmap:9 g722/8000
> a=rtpmap:99 g726-32/8000
> a=rtpmap:3 gsm/8000
> a=rtpmap:18 g729/8000
> a=rtpmap:4 g723/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=crypto:1 AES_CM_128_HMAC_SHA1_32
> inline:m6fas/KsLF57r9RnU7X0WEWeJw9Y6+a66YUIf9Dc
> a=ptime:20
> m=audio 62882 RTP/AVP 0 8 9 99 3 18 4 101
> a=rtpmap:0 pcmu/8000
> a=rtpmap:8 pcma/8000
> a=rtpmap:9 g722/8000
> a=rtpmap:99 g726-32/8000
> a=rtpmap:3 gsm/8000
> a=rtpmap:18 g729/8000
> a=rtpmap:4 g723/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> ]
> variable_ep_codec_string: [G722 at 8000h@20i,PCMA at 8000h@20i]
> variable_endpoint_disposition: [DELAYED NEGOTIATION]
> variable_ET_is_local: [true]
> variable_max_forwards: [69]
> variable_domain_name: [85.16.246.12]
> variable_dialed_extension: [1000]
> variable_sip_callee_id_number: [1111]
> variable_sip_callee_id_name: [hubu]
> variable_export_vars: [sip_callee_id_number,sip_callee_id_name]
> variable_current_application: [info]
>
>
> On 16.10.2009 18:18, Anthony Minessale wrote:
> > 1) you should update again there were a few issues.
> > 2) you can set the variable sip_callee_id_name and sip_callee_id number
> > on the inbound leg before you answer to control what it says.
>
> regards
> Helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
>
> iD8DBQFK3HwJ4tZeNddg3dwRAieqAKDCn47dzctaYpweRGzovbMPcKD5KQCgjNx+
> nNTz4r7+N7mI3Wj4GayFdTk=
> =MYOb
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From nbelov at abisoft.spb.ru  Mon Oct 19 08:11:03 2009
From: nbelov at abisoft.spb.ru (Nikita Belov)
Date: Mon, 19 Oct 2009 19:11:03 +0400
Subject: [Freeswitch-users] uuid_send_dtmf fails (was: conference call)
In-Reply-To: <414DC781-A0F6-418E-9AAB-FA4307634932@jerris.com>
References: <064A5201C10D4D20AC76EB93CA5EA4B5@abisoft.biz><ebbc2d550910140838v3125579ev9fa16e3e83d25770@mail.gmail.com><BE4EDDA84CB64AA98F18D412B51912FC@abisoft.biz><b39273530910150504l5c7411b9g39b9ed33c592e85f@mail.gmail.com><02C2EFD4CF8D4ACF8EEC0F77C4D75028@abisoft.biz>
	<414DC781-A0F6-418E-9AAB-FA4307634932@jerris.com>
Message-ID: <14E8C50A8F4D41559AFCCD8838350F4F@abisoft.biz>

Thanks, Mike, for idea. But what is the syntax for this session message?
I tried this:

sendmsg e8e4f0ed-a0cc-4dff-b7e1-09eeade5df05
eavesdrop-command: 1

but it doesn't work.


> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
> Sent: Monday, October 19, 2009 5:19 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] uuid_send_dtmf fails (was: conference
> call)
> 
> There is an event you can send as well to switch them, it your trying
> to switch it via event socket, that should be better, its not on the
> wiki, but
> 
> a session message with
> eavesdrop-command header with data as the same as dtmf
> should do the trick
> 
> Mike
> 
> 
> On Oct 16, 2009, at 11:54 AM, Nikita Belov wrote:
> 
> > Yes, it is what I need. But now I have problem with sending dtmf.
> > Here what
> > I've done:
> > [root at centos4-4-vm ~]# telnet localhost 8021
> > Trying 127.0.0.1...
> > Connected to localhost.localdomain (127.0.0.1).
> > Escape character is '^]'.
> > Content-Type: auth/request
> >
> > auth ClueCon
> >
> > Content-Type: command/reply
> > Reply-Text: +OK accepted
> >
> > api originate user/1007 at master.agent.starpoundtech.net &park()
> >
> > Content-Type: api/response
> > Content-Length: 41
> >
> > +OK bba3b45a-4cc1-48af-a15d-1052d5f11371
> >
> > SendMsg bba3b45a-4cc1-48af-a15d-1052d5f11371
> > call-command: execute
> > execute-app-name: eavesdrop
> > execute-app-arg: cd99f999-9b47-457e-8439-1d366e015b8c
> >
> > Content-Type: command/reply
> > Reply-Text: +OK
> >
> > Here I had started to hear A and B. Here what I saw in FS log:
> > 2009-10-18 03:22:47 [DEBUG] switch_core_session.c:706
> > switch_core_session_queue_private_event() Send signal
> > sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes [BREAK]
> > 2009-10-18 03:22:47 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event()
> > sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes Command Execute
> > eavesdrop(cd99f999-9b47-457e-8439-1d366e015b8c)
> > 2009-10-18 03:22:47 [DEBUG] switch_core_media_bug.c:297
> > switch_core_media_bug_add() Attaching BUG to
> > sofia/internal/1006 at master.agent.starpoundtech.net
> > 2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234
> > switch_core_session_read_frame()
> > sofia/internal/1006 at master.agent.starpoundtech.net receive message
> > [TRANSCODING_NECESSARY]
> > 2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234
> > switch_core_session_read_frame()
> > sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes receive message
> > [TRANSCODING_NECESSARY]
> >
> > Then I run command:
> > api uuid_send_dtmf bba3b45a-4cc1-48af-a15d-1052d5f11371 1
> >
> > Content-Type: api/response
> > Content-Length: 14
> >
> > -ERR no reply
> >
> > Log:
> > 2009-10-18 03:24:01 [DEBUG] switch_core_io.c:1190
> > switch_core_session_send_dtmf_string()
> > sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes send dtmf
> > digit=1 ms=250 samples=2000
> > 2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1282 do_2833() Send start
> > packet
> > for [1] ts=2241760 dur=160/160/2000 seq=21346
> > 2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
> > packet
> > for [1] ts=2241760 dur=320/320/2000 seq=21347
> > 2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
> > packet
> > for [1] ts=2241760 dur=480/480/2000 seq=21348
> > 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
> > packet
> > for [1] ts=2241760 dur=640/640/2000 seq=21349
> > 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
> > packet
> > for [1] ts=2241760 dur=800/800/2000 seq=21350
> > 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
> > packet
> > for [1] ts=2241760 dur=960/960/2000 seq=21351
> > 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
> > packet
> > for [1] ts=2241760 dur=1120/1120/2000 seq=21352
> > 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
> > packet
> > for [1] ts=2241760 dur=1280/1280/2000 seq=21353
> > 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
> > packet
> > for [1] ts=2241760 dur=1440/1440/2000 seq=21354
> > 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
> > packet
> > for [1] ts=2241760 dur=1600/1600/2000 seq=21355
> > 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
> > packet
> > for [1] ts=2241760 dur=1760/1760/2000 seq=21356
> > 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
> > packet
> > for [1] ts=2241760 dur=1920/1920/2000 seq=21357
> > 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end
> > packet for
> > [1] ts=2241760 dur=2080/2080/2000 seq=21358
> > 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end
> > packet for
> > [1] ts=2241760 dur=2080/2080/2000 seq=21359
> > 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end
> > packet for
> > [1] ts=2241760 dur=2080/2080/2000 seq=21360
> >
> > But both A and B couldn't hear me.
> > Btw, after I had send dtmf "1" manually from my phone. B started to
> > hear me.
> > There was this record in log:
> > 2009-10-18 03:47:55 [DEBUG] switch_rtp.c:1767 switch_rtp_dequeue_dtmf
> > () RTP
> > RECV DTMF 1:2240
> >
> > Does anybody know, what had I done wrong?
> >
> > ___
> >
> > Thanks, Nikita
> >
> >> -----Original Message-----
> >> From: freeswitch-users-bounces at lists.freeswitch.org
> >> [mailto:freeswitch-
> >> users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker
> >> Sent: Thursday, October 15, 2009 4:04 PM
> >> To: freeswitch-users at lists.freeswitch.org
> >> Subject: Re: [Freeswitch-users] conference call
> >>
> >> The wiki has a pretty good explanation of how eavesdrop works.
> >> Enabling a talk path to A or B or both A and B requires dtmf.
> >>
> >> So, if C hits the 1 button on the phone they can talk to the UUID you
> >> bound the eavesdrop to.   2 talks to the other party.  3 talks to
> >> both
> >> parties.
> >>
> >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop
> >>
> >> Adding more api calls to control eavesdrop via event socket should be
> >> possible, though I haven't looked at the code to see what it would
> >> take...
> >>
> >> On Thu, Oct 15, 2009 at 3:16 AM, Nikita Belov <nbelov at abisoft.spb.ru>
> >> wrote:
> >>> Thanks for answer.
> >>>
> >>>
> >>>
> >>> As I've understood eavesdrop allows C to hear A and B, but A and B
> >>> will
> >> not
> >>> hear C. But in my case B SHOULD hear C. In my case A - client, B -
> >>> agent
> >>> worked with client and C - coach/supervisor for B. Can I make such
> >>> conference in FS using eavesdrop or other API command?
> >>>
> >>>
> >>>
> >>> ___
> >>>
> >>>
> >>>
> >>> Thanks, Nikita
> >>>
> >>>
> >>>
> >>> ________________________________
> >>>
> >>> From: freeswitch-users-bounces at lists.freeswitch.org
> >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf
> >>> Of Joao
> >>> Mesquita
> >>> Sent: Wednesday, October 14, 2009 7:38 PM
> >>> To: freeswitch-users at lists.freeswitch.org
> >>> Subject: Re: [Freeswitch-users] conference call
> >>>
> >>>
> >>>
> >>> Look at eavesdrop on the wiki.
> >>>
> >>> JM
> >>>
> >>> 2009/10/14 Nikita Belov <nbelov at abisoft.spb.ru>
> >>>
> >>> HI all,
> >>>
> >>>
> >>>
> >>> I want to configure FS to make special conference call between three
> >> users
> >>> (A, B, C).  In this conference C will hear A and B, but A will
> >>> hear only
> >> B.
> >>> Can I make it using FS API commands? Does anybody know which
> >>> approach is
> >>> better to use?
> >>>
> >>>
> >>>
> >>> ___
> >>>
> >>>
> >>>
> >>> Thanks, Nikita
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >>> users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >>> users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>
> >>
> >>
> >> --
> >> -Rupa
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From brian at freeswitch.org  Mon Oct 19 08:13:04 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 19 Oct 2009 10:13:04 -0500
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <7b8592a0910190516rafa4398q62c7a99ef703540f@mail.gmail.com>
References: <7b8592a0910190516rafa4398q62c7a99ef703540f@mail.gmail.com>
Message-ID: <5B924C57-B658-44C0-A1CE-00A3ECB5AA07@freeswitch.org>

I think you just have it misconfigured because if ipv6 was broken I  
would have William and Jason jumping on me.

Can you post your xml configs for the profile please?

/b

On Oct 19, 2009, at 7:16 AM, ineya ineya wrote:

> Hi,
>
> I installed the phonenix release, and changed IPv4, IPv6 addresses  
> and SIP port. Directory and dialplan should be default ones.
> My phones are able to register when using just IPv4, and I can call  
> from one to another.
>
> When I try the same thing with IPv6, the call to other phone fails  
> with [USER_NOT_REGISTERED], however I'm able to call 9999 and hear  
> MOH.
> Here you can find the log: http://pastebin.freeswitch.org/10757
> Freeswitch is running on 10.80.62.40/2000:2::1
> phones are 2000:2::1001 and 2000:2::1002
>
> Am I doing something wrong? Maybe I forgot to configure something,  
> but I can't see what it is.
>
> Any hints are appreciated.

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From helmut.kuper at ewetel.de  Mon Oct 19 08:29:04 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Mon, 19 Oct 2009 17:29:04 +0200
Subject: [Freeswitch-users] Qustion about INFO messages
	after	Connect/Answer
In-Reply-To: <191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>
References: <4AD859A4.5000101@ewetel.de>	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>	<4ADC7C09.6070503@ewetel.de>
	<191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>
Message-ID: <4ADC85C0.7060908@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

well, when I add "answer" before I bridge, there is a small change:
There is no INFO message with "unknown" send to callee.

The caller's display isn't affected by "sip_callee_*" chvars.



On 19.10.2009 17:00, Anthony Minessale wrote:
> you only need to "set" it on the inbound leg and you must answer and
> bridge it somewhere.

My current dialplan is this:

    <extension name="Local_Extension">
      <condition field="${ET_is_local}" expression="^true$">
        <action application="set"
data="dialed_extension=${destination_number}"/>
        <action application="answer"/>
        <action application="set" data="sip_callee_id_number=1111"/>
        <action application="set" data="sip_callee_id_name=hubu"/>
        <action application="export" data="sip_callee_id_number=1111"/>
        <action application="export" data="sip_callee_id_name=hubu"/>
        <action application="info"/>
        <action application="export"
data="dialed_extension=${destination_number}"/>
        <action application="set" data="transfer_ringback=$${hold_music}"/>

        <action application="set" data="hangup_after_bridge=true"/>
        <action application="export"
data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name}
var sip_secure_media)}"/>
        <action application="db"
data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
        <action application="set" data="continue_on_fail=false"/>
        <action application="bridge"
data="user/${dialed_extension}@$${domain}"/>
      </condition>
    </extension>



Here is the complete debug log:



2009-10-19 17:27:20.559060 [DEBUG] sofia.c:4906 IP 85.16.245.206
Approved by acl "clients[]". Access Granted.
2009-10-19 17:27:20.560087 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/1001 at 85.16.246.12:5061 [e47af3a6-bcc3-11de-9f91-c9cd82739033]
2009-10-19 17:27:20.560087 [DEBUG] sofia.c:3492 Channel
sofia/internal/1001 at 85.16.246.12:5061 entering state [received][100]
2009-10-19 17:27:20.560087 [DEBUG] sofia.c:3499 Remote SDP:
v=0
o=root 903333206 903333206 IN IP4 85.16.245.206
s=call
c=IN IP4 85.16.245.206
t=0 0
m=audio 54598 RTP/SAVP 0 8 9 99 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:zS0VqvwW7z/dnmeWvtZS0AwhJ1ru1J1tuHJO2JRD
a=ptime:20
m=audio 54598 RTP/AVP 0 8 9 99 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

2009-10-19 17:27:20.560087 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_NEW
2009-10-19 17:27:20.560087 [DEBUG] switch_core_state_machine.c:312
(sofia/internal/1001 at 85.16.246.12:5061) State NEW
2009-10-19 17:27:20.560087 [DEBUG] sofia.c:3620
(sofia/internal/1001 at 85.16.246.12:5061) State Change CS_NEW -> CS_INIT
2009-10-19 17:27:20.560087 [DEBUG] switch_core_session.c:969 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-19 17:27:20.560087 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_INIT
2009-10-19 17:27:20.560087 [DEBUG] switch_core_state_machine.c:330
(sofia/internal/1001 at 85.16.246.12:5061) State INIT
2009-10-19 17:27:20.560087 [DEBUG] mod_sofia.c:83
sofia/internal/1001 at 85.16.246.12:5061 SOFIA INIT
2009-10-19 17:27:20.560087 [DEBUG] mod_sofia.c:111
(sofia/internal/1001 at 85.16.246.12:5061) State Change CS_INIT -> CS_ROUTING
2009-10-19 17:27:20.560087 [DEBUG] switch_core_session.c:969 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-19 17:27:20.560087 [DEBUG] switch_core_state_machine.c:330
(sofia/internal/1001 at 85.16.246.12:5061) State INIT going to sleep
2009-10-19 17:27:20.560087 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_ROUTING
2009-10-19 17:27:20.560087 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/1001 at 85.16.246.12:5061) State ROUTING
2009-10-19 17:27:20.560087 [DEBUG] mod_sofia.c:130
sofia/internal/1001 at 85.16.246.12:5061 SOFIA ROUTING
2009-10-19 17:27:20.560087 [DEBUG] switch_core_state_machine.c:78
sofia/internal/1001 at 85.16.246.12:5061 Standard ROUTING
2009-10-19 17:27:20.560087 [INFO] mod_dialplan_xml.c:391 Processing 1001
an PBX1->1000 in context default
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->anonymous] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [anonymous]
destination_number(1000) =~ /^\*31([0-9]+)$/ break=never
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 ANTI-Action
set(dialed_extension=${destination_number})
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->is_local] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [is_local]
${ET_is_local}() =~ /(true|false)/ break=on-true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 ANTI-Action
lua(ET_is_local.lua)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 ANTI-Action
transfer(${destination_number} XML default)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->set_domain] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (PASS)
[set_domain] destination_number(1000) =~ /^.*$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
set(domain_name=85.16.246.12)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->302_zero_problem] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[302_zero_problem] ${sip_looped_call}() =~ /true/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->fakeTrisko-Servicenumber as callerid] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[fakeTrisko-Servicenumber as callerid] caller_id_number(1001) =~
/^(49[1-2][0-9])$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->outbound-ssw-ddi] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[outbound-ssw-ddi] ${dialed_extension}() =~ /^0([0-9]+)$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing [default->mail]
continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [mail]
destination_number(1000) =~ /^9997$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing [default->pizza]
continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [pizza]
destination_number(1000) =~ /^2006$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing [default->pbook]
continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [pbook]
destination_number(1000) =~ /^2007$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->shout_1] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [shout_1]
${dialed_extension}() =~ /^9123$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->shout_2] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [shout_2]
${dialed_extension}() =~ /^9124$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->speex_32000] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[speex_32000] ${dialed_extension}() =~ /^9125$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->speex_32000_2] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[speex_32000_2] ${dialed_extension}() =~ /^9126$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->jukebox] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [jukebox]
destination_number(1000) =~ /^9111$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->pickup-extension] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[pickup-extension] ${dialed_extension}() =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->Voicemail-Verwaltung] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[Voicemail-Verwaltung] ${dialed_extension}() =~ /^(1001|9999)$/
break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->Local_Extension] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[Local_Extension] ${ET_is_local}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->parallel call] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [parallel
call] ${dialed_extension}() =~ /^(6000)$/ break=on-false
2009-10-19 17:27:20.562064 [DEBUG] switch_core_state_machine.c:114
(sofia/internal/1001 at 85.16.246.12:5061) State Change CS_ROUTING ->
CS_EXECUTE
2009-10-19 17:27:20.562064 [DEBUG] switch_core_session.c:969 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-19 17:27:20.562064 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/1001 at 85.16.246.12:5061) State ROUTING going to sleep
2009-10-19 17:27:20.563061 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_EXECUTE
2009-10-19 17:27:20.563061 [DEBUG] switch_core_state_machine.c:340
(sofia/internal/1001 at 85.16.246.12:5061) State EXECUTE
2009-10-19 17:27:20.563061 [DEBUG] mod_sofia.c:173
sofia/internal/1001 at 85.16.246.12:5061 SOFIA EXECUTE
2009-10-19 17:27:20.563061 [DEBUG] switch_core_state_machine.c:151
sofia/internal/1001 at 85.16.246.12:5061 Standard EXECUTE
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 set(dialed_extension=1000)
2009-10-19 17:27:20.563061 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET [dialed_extension]=[1000]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 lua(ET_is_local.lua)
2009-10-19 17:27:20.588079 [DEBUG] switch_cpp.cpp:925 destroy/unlink
session from object
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 transfer(1000 XML default)
2009-10-19 17:27:20.588079 [DEBUG] switch_ivr.c:1381
(sofia/internal/1001 at 85.16.246.12:5061) State Change CS_EXECUTE ->
CS_ROUTING
2009-10-19 17:27:20.588079 [DEBUG] switch_core_session.c:969 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-19 17:27:20.588079 [DEBUG] switch_ivr.c:1385
sofia/internal/1001 at 85.16.246.12:5061 receive message [TRANSFER]
2009-10-19 17:27:20.588079 [DEBUG] switch_core_session.c:639 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-19 17:27:20.588079 [NOTICE] switch_ivr.c:1387 Transfer
sofia/internal/1001 at 85.16.246.12:5061 to XML[1000 at default]
2009-10-19 17:27:20.588079 [DEBUG] switch_core_state_machine.c:340
(sofia/internal/1001 at 85.16.246.12:5061) State EXECUTE going to sleep
2009-10-19 17:27:20.588079 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_ROUTING
2009-10-19 17:27:20.588079 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/1001 at 85.16.246.12:5061) State ROUTING
2009-10-19 17:27:20.588079 [DEBUG] mod_sofia.c:130
sofia/internal/1001 at 85.16.246.12:5061 SOFIA ROUTING
2009-10-19 17:27:20.588079 [DEBUG] switch_core_state_machine.c:78
sofia/internal/1001 at 85.16.246.12:5061 Standard ROUTING
2009-10-19 17:27:20.588079 [INFO] mod_dialplan_xml.c:391 Processing 1001
an PBX1->1000 in context default
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->anonymous] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [anonymous]
destination_number(1000) =~ /^\*31([0-9]+)$/ break=never
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 ANTI-Action
set(dialed_extension=${destination_number})
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->is_local] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (PASS) [is_local]
${ET_is_local}(true) =~ /(true|false)/ break=on-true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->set_domain] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (PASS)
[set_domain] destination_number(1000) =~ /^.*$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
set(domain_name=85.16.246.12)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->302_zero_problem] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[302_zero_problem] ${sip_looped_call}() =~ /true/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->fakeTrisko-Servicenumber as callerid] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[fakeTrisko-Servicenumber as callerid] caller_id_number(1001) =~
/^(49[1-2][0-9])$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->outbound-ssw-ddi] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[outbound-ssw-ddi] ${dialed_extension}(1000) =~ /^0([0-9]+)$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing [default->mail]
continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [mail]
destination_number(1000) =~ /^9997$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing [default->pizza]
continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [pizza]
destination_number(1000) =~ /^2006$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing [default->pbook]
continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [pbook]
destination_number(1000) =~ /^2007$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->shout_1] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [shout_1]
${dialed_extension}(1000) =~ /^9123$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->shout_2] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [shout_2]
${dialed_extension}(1000) =~ /^9124$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->speex_32000] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[speex_32000] ${dialed_extension}(1000) =~ /^9125$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->speex_32000_2] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[speex_32000_2] ${dialed_extension}(1000) =~ /^9126$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->jukebox] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [jukebox]
destination_number(1000) =~ /^9111$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->pickup-extension] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[pickup-extension] ${dialed_extension}(1000) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->Voicemail-Verwaltung] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[Voicemail-Verwaltung] ${dialed_extension}(1000) =~ /^(1001|9999)$/
break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->Local_Extension] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (PASS)
[Local_Extension] ${ET_is_local}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
set(dialed_extension=${destination_number})
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action answer()
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
set(sip_callee_id_number=1111)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
set(sip_callee_id_name=hubu)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
export(sip_callee_id_number=1111)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
export(sip_callee_id_name=hubu)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action info()
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
export(dialed_extension=${destination_number})
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
set(transfer_ringback=local_stream://moh)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
set(hangup_after_bridge=true)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
export(nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name}
var sip_secure_media)})
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
db(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid})
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
set(continue_on_fail=false)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
bridge(user/${dialed_extension}@85.16.246.12)
2009-10-19 17:27:20.590066 [DEBUG] switch_core_state_machine.c:114
(sofia/internal/1001 at 85.16.246.12:5061) State Change CS_ROUTING ->
CS_EXECUTE
2009-10-19 17:27:20.590066 [DEBUG] switch_core_session.c:969 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-19 17:27:20.590066 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/1001 at 85.16.246.12:5061) State ROUTING going to sleep
2009-10-19 17:27:20.590066 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_EXECUTE
2009-10-19 17:27:20.590066 [DEBUG] switch_core_state_machine.c:340
(sofia/internal/1001 at 85.16.246.12:5061) State EXECUTE
2009-10-19 17:27:20.590066 [DEBUG] mod_sofia.c:173
sofia/internal/1001 at 85.16.246.12:5061 SOFIA EXECUTE
2009-10-19 17:27:20.591072 [DEBUG] switch_core_state_machine.c:151
sofia/internal/1001 at 85.16.246.12:5061 Standard EXECUTE
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 set(dialed_extension=1000)
2009-10-19 17:27:20.591072 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET [dialed_extension]=[1000]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 set(domain_name=85.16.246.12)
2009-10-19 17:27:20.591072 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET [domain_name]=[85.16.246.12]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 set(dialed_extension=1000)
2009-10-19 17:27:20.591072 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET [dialed_extension]=[1000]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 answer()
2009-10-19 17:27:20.591072 [DEBUG] mod_dptools.c:653
sofia/internal/1001 at 85.16.246.12:5061 receive message [ANSWER]
2009-10-19 17:27:20.591072 [DEBUG] sofia_glue.c:3028 Set Remote Key [1
AES_CM_128_HMAC_SHA1_32 inline:zS0VqvwW7z/dnmeWvtZS0AwhJ1ru1J1tuHJO2JRD]
2009-10-19 17:27:20.591072 [DEBUG] sofia_glue.c:2160 Set Local Key [1
AES_CM_128_HMAC_SHA1_32 inline:GteACgi6L6LMWBxdEnW4/XSOUhsmp/iKk9dZjmBJ]
2009-10-19 17:27:20.591072 [DEBUG] sofia_glue.c:3142 Audio Codec Compare
[pcmu:0:8000:20]/[CELT:95:48000:10]
2009-10-19 17:27:20.591072 [DEBUG] sofia_glue.c:3142 Audio Codec Compare
[pcmu:0:8000:20]/[SPEEX:103:32000:20]
2009-10-19 17:27:20.591072 [DEBUG] sofia_glue.c:3142 Audio Codec Compare
[pcmu:0:8000:20]/[G722:9:8000:20]
2009-10-19 17:27:20.591072 [DEBUG] sofia_glue.c:3142 Audio Codec Compare
[pcmu:0:8000:20]/[PCMA:8:8000:20]
2009-10-19 17:27:20.591072 [DEBUG] sofia_glue.c:3142 Audio Codec Compare
[pcma:8:8000:20]/[CELT:95:48000:10]
2009-10-19 17:27:20.591072 [DEBUG] sofia_glue.c:3142 Audio Codec Compare
[pcma:8:8000:20]/[SPEEX:103:32000:20]
2009-10-19 17:27:20.591072 [DEBUG] sofia_glue.c:3142 Audio Codec Compare
[pcma:8:8000:20]/[G722:9:8000:20]
2009-10-19 17:27:20.591072 [DEBUG] sofia_glue.c:3142 Audio Codec Compare
[pcma:8:8000:20]/[PCMA:8:8000:20]
2009-10-19 17:27:20.591072 [DEBUG] sofia_glue.c:2100 Set Codec
sofia/internal/1001 at 85.16.246.12:5061 PCMA/8000 20 ms 160 samples
2009-10-19 17:27:20.591072 [DEBUG] sofia_glue.c:3102 Set 2833 dtmf
payload to 101
2009-10-19 17:27:20.591072 [DEBUG] sofia_glue.c:3076 Our existing sdp is
still good [pcma 85.16.245.206:54598], let's keep it.
2009-10-19 17:27:20.591072 [DEBUG] sofia_glue.c:2334 AUDIO RTP
[sofia/internal/1001 at 85.16.246.12:5061] 85.16.246.12 port 14580 ->
85.16.245.206 port 54598 codec: 8 ms: 20
2009-10-19 17:27:20.591072 [DEBUG] switch_rtp.c:1155 Starting timer
[soft] 160 bytes per 20ms
2009-10-19 17:27:20.594059 [INFO] switch_rtp.c:1019 Activating Secure
RTP SEND
2009-10-19 17:27:20.594059 [INFO] switch_rtp.c:999 Activating Secure RTP
RECV
2009-10-19 17:27:20.594059 [DEBUG] switch_core_sqldb.c:396 Secure Type:
srtp:AES_CM_128_HMAC_SHA1_32
2009-10-19 17:27:20.594059 [DEBUG] switch_core_sqldb.c:396 Secure Type:
srtp:AES_CM_128_HMAC_SHA1_32
2009-10-19 17:27:20.594059 [NOTICE] sofia_glue.c:2769 Pre-Answer
sofia/internal/1001 at 85.16.246.12:5061!
2009-10-19 17:27:20.594059 [DEBUG] mod_sofia.c:543 Local SDP
sofia/internal/1001 at 85.16.246.12:5061:
v=0
o=FreeSWITCH 1255951460 1255951461 IN IP4 85.16.246.12
s=FreeSWITCH
c=IN IP4 85.16.246.12
t=0 0
m=audio 14580 RTP/SAVP 8 101
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:GteACgi6L6LMWBxdEnW4/XSOUhsmp/iKk9dZjmBJ

2009-10-19 17:27:20.594059 [DEBUG] switch_core_session.c:639 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-19 17:27:20.594059 [NOTICE] mod_dptools.c:653 Channel
[sofia/internal/1001 at 85.16.246.12:5061] has been answered
2009-10-19 17:27:20.594059 [DEBUG] switch_channel.c:182
sofia/internal/1001 at 85.16.246.12:5061 receive message [AUDIO_SYNC]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 set(sip_callee_id_number=1111)
2009-10-19 17:27:20.594059 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET [sip_callee_id_number]=[1111]
2009-10-19 17:27:20.595060 [DEBUG] sofia.c:3492 Channel
sofia/internal/1001 at 85.16.246.12:5061 entering state [completed][200]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 set(sip_callee_id_name=hubu)
2009-10-19 17:27:20.595060 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET [sip_callee_id_name]=[hubu]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061
export(sip_callee_id_number=1111)
2009-10-19 17:27:20.595060 [DEBUG] mod_dptools.c:896 EXPORT
[sip_callee_id_number]=[1111]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061
export(sip_callee_id_name=hubu)
2009-10-19 17:27:20.595060 [DEBUG] mod_dptools.c:896 EXPORT
[sip_callee_id_name]=[hubu]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 info()
2009-10-19 17:27:20.596059 [INFO] mod_dptools.c:961 CHANNEL_DATA:
Channel-State: [CS_EXECUTE]
Channel-State-Number: [4]
Channel-Name: [sofia/internal/1001 at 85.16.246.12:5061]
Unique-ID: [e47af3a6-bcc3-11de-9f91-c9cd82739033]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Answer-State: [answered]
Channel-Read-Codec-Name: [PCMA]
Channel-Read-Codec-Rate: [8000]
Channel-Write-Codec-Name: [PCMA]
Channel-Write-Codec-Rate: [8000]
Caller-Username: [1001]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [1001 an PBX1]
Caller-Caller-ID-Number: [1001]
Caller-Network-Addr: [85.16.245.206]
Caller-Destination-Number: [1000]
Caller-Unique-ID: [e47af3a6-bcc3-11de-9f91-c9cd82739033]
Caller-Source: [mod_sofia]
Caller-Context: [default]
Caller-RDNIS: [1000]
Caller-Channel-Name: [sofia/internal/1001 at 85.16.246.12:5061]
Caller-Profile-Index: [2]
Caller-Profile-Created-Time: [1255966040588079]
Caller-Channel-Created-Time: [1255966040560087]
Caller-Channel-Answered-Time: [1255966040594059]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [1255966040594059]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
variable_sip_received_ip: [85.16.245.206]
variable_sip_received_port: [1024]
variable_sip_via_protocol: [udp]
variable_sip_authorized: [true]
variable_sip_from_user: [1001]
variable_sip_from_port: [5061]
variable_sip_from_uri: [1001 at 85.16.246.12:5061]
variable_sip_from_host: [85.16.246.12]
variable_sip_from_user_stripped: [1001]
variable_sip_from_tag: [02p9o5jbg8]
variable_sofia_profile_name: [internal]
variable_sip_req_params: [user=phone]
variable_sip_req_user: [1000]
variable_sip_req_port: [5061]
variable_sip_req_uri: [1000 at 85.16.246.12:5061]
variable_sip_req_host: [85.16.246.12]
variable_sip_to_params: [user=phone]
variable_sip_to_user: [1000]
variable_sip_to_port: [5061]
variable_sip_to_uri: [1000 at 85.16.246.12:5061]
variable_sip_to_host: [85.16.246.12]
variable_sip_contact_params: [line=eg3wp69a]
variable_sip_contact_user: [1001]
variable_sip_contact_port: [1024]
variable_sip_contact_uri: [1001 at 85.16.245.206:1024]
variable_sip_contact_host: [85.16.245.206]
variable_channel_name: [sofia/internal/1001 at 85.16.246.12:5061]
variable_sip_call_id: [3c2d38a1cd82-iphmvbhl46hj]
variable_sip_user_agent: [snom820/8.2.16]
variable_sip_via_host: [85.16.245.206]
variable_sip_via_port: [1024]
variable_sip_via_rport: [1024]
variable_presence_id: [1001 at 85.16.246.12]
variable_sip_h_X-Serialnumber: [0004134002CB]
variable_sip_h_P-Key-Flags: [resolution="31x13", keys="4"]
variable_switch_r_sdp: [v=0
o=root 903333206 903333206 IN IP4 85.16.245.206
s=call
c=IN IP4 85.16.245.206
t=0 0
m=audio 54598 RTP/SAVP 0 8 9 99 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:zS0VqvwW7z/dnmeWvtZS0AwhJ1ru1J1tuHJO2JRD
a=ptime:20
m=audio 54598 RTP/AVP 0 8 9 99 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
]
variable_ep_codec_string: [G722 at 8000h@20i,PCMA at 8000h@20i]
variable_ET_is_local: [true]
variable_max_forwards: [69]
variable_domain_name: [85.16.246.12]
variable_dialed_extension: [1000]
variable_sip_has_crypto: [AES_CM_128_HMAC_SHA1_32]
variable_sip_crypto_mandatory: [true]
variable_sip_secure_media: [true]
variable_remote_media_ip: [85.16.245.206]
variable_remote_media_port: [54598]
variable_read_codec: [PCMA]
variable_read_rate: [8000]
variable_write_codec: [PCMA]
variable_write_rate: [8000]
variable_local_media_ip: [85.16.246.12]
variable_local_media_port: [14580]
variable_sip_secure_media_confirmed: [true]
variable_endpoint_disposition: [ANSWER]
variable_sip_callee_id_number: [1111]
variable_sip_callee_id_name: [hubu]
variable_export_vars: [sip_callee_id_number,sip_callee_id_name]
variable_current_application: [info]


EXECUTE sofia/internal/1001 at 85.16.246.12:5061 export(dialed_extension=1000)
2009-10-19 17:27:20.596059 [DEBUG] mod_dptools.c:896 EXPORT
[dialed_extension]=[1000]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061
set(transfer_ringback=local_stream://moh)
2009-10-19 17:27:20.596059 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET
[transfer_ringback]=[local_stream://moh]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 set(hangup_after_bridge=true)
2009-10-19 17:27:20.597073 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061
export(nolocal:sip_secure_media=)
2009-10-19 17:27:20.621072 [DEBUG] mod_dptools.c:896 EXPORT (REMOTE
ONLY) [sip_secure_media]=[UNDEF]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061
db(insert/85.16.246.12-last_dial//e47af3a6-bcc3-11de-9f91-c9cd82739033)
2009-10-19 17:27:20.832283 [DEBUG] sofia.c:3492 Channel
sofia/internal/1001 at 85.16.246.12:5061 entering state [ready][200]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 set(continue_on_fail=false)
2009-10-19 17:27:20.841267 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET [continue_on_fail]=[false]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 bridge(user/1000 at 85.16.246.12)
2009-10-19 17:27:20.865064 [DEBUG] switch_ivr_originate.c:1162 variable
string 0 = [presence_id=1000 at 85.16.246.12]
2009-10-19 17:27:20.865064 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/sip:1000 at 85.16.245.228:1024
[e4a9a7a0-bcc3-11de-9f91-c9cd82739033]
2009-10-19 17:27:20.866297 [DEBUG] mod_sofia.c:2981
(sofia/internal/sip:1000 at 85.16.245.228:1024) State Change CS_NEW -> CS_INIT
2009-10-19 17:27:20.866297 [DEBUG] switch_core_session.c:969 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-19 17:27:20.866297 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/sip:1000 at 85.16.245.228:1024) Running State Change CS_INIT
2009-10-19 17:27:20.866297 [DEBUG] switch_core_state_machine.c:330
(sofia/internal/sip:1000 at 85.16.245.228:1024) State INIT
2009-10-19 17:27:20.866297 [DEBUG] mod_sofia.c:83
sofia/internal/sip:1000 at 85.16.245.228:1024 SOFIA INIT
2009-10-19 17:27:20.866297 [DEBUG] mod_sofia.c:111
(sofia/internal/sip:1000 at 85.16.245.228:1024) State Change CS_INIT ->
CS_ROUTING
2009-10-19 17:27:20.866297 [DEBUG] switch_core_session.c:969 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-19 17:27:20.866297 [DEBUG] switch_core_state_machine.c:330
(sofia/internal/sip:1000 at 85.16.245.228:1024) State INIT going to sleep
2009-10-19 17:27:20.866297 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/sip:1000 at 85.16.245.228:1024) Running State Change CS_ROUTING
2009-10-19 17:27:20.866297 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/sip:1000 at 85.16.245.228:1024) State ROUTING
2009-10-19 17:27:20.866297 [DEBUG] mod_sofia.c:130
sofia/internal/sip:1000 at 85.16.245.228:1024 SOFIA ROUTING
2009-10-19 17:27:20.866297 [DEBUG] switch_ivr_originate.c:66
(sofia/internal/sip:1000 at 85.16.245.228:1024) State Change CS_ROUTING ->
CS_CONSUME_MEDIA
2009-10-19 17:27:20.866297 [DEBUG] switch_core_session.c:969 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-19 17:27:20.866297 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/sip:1000 at 85.16.245.228:1024) State ROUTING going to sleep
2009-10-19 17:27:20.866297 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/sip:1000 at 85.16.245.228:1024) Running State Change
CS_CONSUME_MEDIA
2009-10-19 17:27:20.866297 [DEBUG] switch_ivr_originate.c:1880 Raw Codec
Activation Success L16 at 8000hz 1 channel 20ms
2009-10-19 17:27:20.866297 [DEBUG] switch_ivr_originate.c:1912 Play
Ringback File [local_stream://moh]
2009-10-19 17:27:20.866297 [DEBUG] mod_local_stream.c:408 Opening Stream
[moh/8000] 8000hz
2009-10-19 17:27:20.866297 [DEBUG] switch_core_state_machine.c:352
(sofia/internal/sip:1000 at 85.16.245.228:1024) State CONSUME_MEDIA
2009-10-19 17:27:20.866297 [DEBUG] switch_core_state_machine.c:352
(sofia/internal/sip:1000 at 85.16.245.228:1024) State CONSUME_MEDIA going
to sleep
2009-10-19 17:27:20.868064 [DEBUG] sofia.c:3492 Channel
sofia/internal/sip:1000 at 85.16.245.228:1024 entering state [calling][0]
2009-10-19 17:27:20.877064 [DEBUG] switch_core_io.c:652
sofia/internal/1001 at 85.16.246.12:5061 receive message
[TRANSCODING_NECESSARY]
2009-10-19 17:27:20.887288 [DEBUG] sofia.c:3492 Channel
sofia/internal/sip:1000 at 85.16.245.228:1024 entering state [proceeding][180]
2009-10-19 17:27:20.887288 [NOTICE] sofia.c:3556 Ring-Ready
sofia/internal/sip:1000 at 85.16.245.228:1024!
2009-10-19 17:27:21.392073 [DEBUG] sofia.c:3492 Channel
sofia/internal/sip:1000 at 85.16.245.228:1024 entering state [proceeding][180]
2009-10-19 17:27:22.403086 [DEBUG] sofia.c:3492 Channel
sofia/internal/sip:1000 at 85.16.245.228:1024 entering state [proceeding][180]
2009-10-19 17:27:22.849092 [DEBUG] sofia.c:3492 Channel
sofia/internal/sip:1000 at 85.16.245.228:1024 entering state [ready][200]
2009-10-19 17:27:22.849092 [DEBUG] sofia.c:3499 Remote SDP:
v=0
o=root 1600007648 1600007649 IN IP4 85.16.245.228
s=call
c=IN IP4 85.16.245.228
t=0 0
m=audio 50468 RTP/AVP 8 9 101
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

2009-10-19 17:27:22.849092 [DEBUG] sofia_glue.c:3142 Audio Codec Compare
[pcma:8:8000:20]/[PCMA:8:8000:20]
2009-10-19 17:27:22.849092 [DEBUG] sofia_glue.c:2100 Set Codec
sofia/internal/sip:1000 at 85.16.245.228:1024 PCMA/8000 20 ms 160 samples
2009-10-19 17:27:22.849092 [DEBUG] sofia_glue.c:3102 Set 2833 dtmf
payload to 101
2009-10-19 17:27:22.849092 [DEBUG] sofia_glue.c:2334 AUDIO RTP
[sofia/internal/sip:1000 at 85.16.245.228:1024] 85.16.246.12 port 16276 ->
85.16.245.228 port 50468 codec: 8 ms: 20
2009-10-19 17:27:22.850096 [DEBUG] switch_rtp.c:1155 Starting timer
[soft] 160 bytes per 20ms
2009-10-19 17:27:22.851093 [DEBUG] switch_channel.c:2046 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-19 17:27:22.851093 [NOTICE] sofia.c:3982 Channel
[sofia/internal/sip:1000 at 85.16.245.228:1024] has been answered
2009-10-19 17:27:22.851093 [DEBUG] switch_channel.c:182
sofia/internal/sip:1000 at 85.16.245.228:1024 receive message [AUDIO_SYNC]
2009-10-19 17:27:22.857294 [DEBUG] switch_core_codec.c:122 Restore
original codec.
2009-10-19 17:27:22.857294 [DEBUG] switch_ivr_originate.c:2363 Originate
Resulted in Success: [sofia/internal/sip:1000 at 85.16.245.228:1024]
2009-10-19 17:27:22.857294 [DEBUG] switch_channel.c:182
sofia/internal/sip:1000 at 85.16.245.228:1024 receive message [AUDIO_SYNC]
2009-10-19 17:27:22.857294 [DEBUG] switch_channel.c:182
sofia/internal/1001 at 85.16.246.12:5061 receive message [AUDIO_SYNC]
2009-10-19 17:27:22.857294 [DEBUG] switch_ivr_originate.c:1880 Raw Codec
Activation Success L16 at 8000hz 1 channel 20ms
2009-10-19 17:27:22.858097 [DEBUG] switch_ivr_originate.c:1912 Play
Ringback File [local_stream://moh]
2009-10-19 17:27:22.858097 [DEBUG] mod_local_stream.c:408 Opening Stream
[moh/8000] 8000hz
2009-10-19 17:27:22.858097 [DEBUG] switch_core_codec.c:122 Restore
original codec.
2009-10-19 17:27:22.858097 [DEBUG] switch_ivr_originate.c:2363 Originate
Resulted in Success: [sofia/internal/sip:1000 at 85.16.245.228:1024]
2009-10-19 17:27:22.858097 [DEBUG] switch_channel.c:182
sofia/internal/sip:1000 at 85.16.245.228:1024 receive message [AUDIO_SYNC]
2009-10-19 17:27:22.858097 [DEBUG] switch_channel.c:182
sofia/internal/1001 at 85.16.246.12:5061 receive message [AUDIO_SYNC]
2009-10-19 17:27:22.858097 [DEBUG] switch_ivr_bridge.c:966
sofia/internal/sip:1000 at 85.16.245.228:1024 receive message [BRIDGE]
2009-10-19 17:27:22.858097 [DEBUG] switch_core_session.c:639 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-19 17:27:22.858097 [DEBUG] switch_ivr_bridge.c:973
sofia/internal/1001 at 85.16.246.12:5061 receive message [BRIDGE]
2009-10-19 17:27:22.858097 [DEBUG] switch_core_session.c:639 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-19 17:27:22.858097 [DEBUG] switch_ivr_bridge.c:1017
(sofia/internal/sip:1000 at 85.16.245.228:1024) State Change
CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2009-10-19 17:27:22.858097 [DEBUG] switch_core_session.c:969 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-19 17:27:22.858097 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/sip:1000 at 85.16.245.228:1024) Running State Change
CS_EXCHANGE_MEDIA
2009-10-19 17:27:22.858097 [DEBUG] switch_core_state_machine.c:343
(sofia/internal/sip:1000 at 85.16.245.228:1024) State EXCHANGE_MEDIA
2009-10-19 17:27:22.858097 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK
2009-10-19 17:27:22.877303 [DEBUG] switch_ivr_bridge.c:115
sofia/internal/sip:1000 at 85.16.245.228:1024 receive message [DISPLAY]
2009-10-19 17:27:22.877303 [DEBUG] switch_ivr_bridge.c:115
sofia/internal/1001 at 85.16.246.12:5061 receive message [DISPLAY]
2009-10-19 17:27:24.568124 [NOTICE] sofia.c:328 Hangup
sofia/internal/sip:1000 at 85.16.245.228:1024 [CS_EXCHANGE_MEDIA]
[NORMAL_CLEARING]
2009-10-19 17:27:24.568124 [DEBUG] switch_channel.c:1837 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [KILL]
2009-10-19 17:27:24.568124 [DEBUG] switch_core_session.c:969 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-19 17:27:24.568124 [DEBUG] switch_core_state_machine.c:449
thread mismatch skipping state handler.
2009-10-19 17:27:24.577124 [DEBUG] switch_ivr_bridge.c:446
sofia/internal/sip:1000 at 85.16.245.228:1024 ending bridge by request from
read function
2009-10-19 17:27:24.577124 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
DONE [sofia/internal/sip:1000 at 85.16.245.228:1024]
2009-10-19 17:27:24.577124 [DEBUG] switch_ivr_bridge.c:523 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-19 17:27:24.577124 [DEBUG] switch_core_state_machine.c:343
(sofia/internal/sip:1000 at 85.16.245.228:1024) State EXCHANGE_MEDIA going
to sleep
2009-10-19 17:27:24.577124 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/sip:1000 at 85.16.245.228:1024) Running State Change CS_HANGUP
2009-10-19 17:27:24.577124 [DEBUG] switch_ivr_bridge.c:440
sofia/internal/sip:1000 at 85.16.245.228:1024 ending bridge by request from
write function
2009-10-19 17:27:24.577124 [DEBUG] switch_ivr_bridge.c:495
sofia/internal/1001 at 85.16.246.12:5061 receive message [UNBRIDGE]
2009-10-19 17:27:24.577124 [DEBUG] switch_core_session.c:639 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-19 17:27:24.577124 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
DONE [sofia/internal/1001 at 85.16.246.12:5061]
2009-10-19 17:27:24.577124 [DEBUG] switch_ivr_bridge.c:523 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-19 17:27:24.577124 [NOTICE] switch_ivr_bridge.c:1093 Hangup
sofia/internal/1001 at 85.16.246.12:5061 [CS_EXECUTE] [NORMAL_CLEARING]
2009-10-19 17:27:24.577124 [DEBUG] switch_channel.c:1837 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [KILL]
2009-10-19 17:27:24.577124 [DEBUG] switch_core_session.c:969 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-19 17:27:24.577124 [DEBUG] switch_core_state_machine.c:476
(sofia/internal/sip:1000 at 85.16.245.228:1024) State HANGUP
2009-10-19 17:27:24.577124 [DEBUG] mod_sofia.c:344 Channel
sofia/internal/sip:1000 at 85.16.245.228:1024 hanging up, cause:
NORMAL_CLEARING
2009-10-19 17:27:24.577124 [DEBUG] switch_core_state_machine.c:46
sofia/internal/sip:1000 at 85.16.245.228:1024 Standard HANGUP, cause:
NORMAL_CLEARING
2009-10-19 17:27:24.577124 [DEBUG] switch_core_state_machine.c:476
(sofia/internal/sip:1000 at 85.16.245.228:1024) State HANGUP going to sleep
2009-10-19 17:27:24.578354 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/sip:1000 at 85.16.245.228:1024) State Change CS_HANGUP ->
CS_REPORTING
2009-10-19 17:27:24.578354 [DEBUG] switch_core_session.c:969 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-19 17:27:24.578354 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/sip:1000 at 85.16.245.228:1024) Running State Change
CS_REPORTING
2009-10-19 17:27:24.578354 [DEBUG] switch_core_state_machine.c:567
(sofia/internal/sip:1000 at 85.16.245.228:1024) State REPORTING
2009-10-19 17:27:24.578354 [DEBUG] switch_core_state_machine.c:53
sofia/internal/sip:1000 at 85.16.245.228:1024 Standard REPORTING, cause:
NORMAL_CLEARING
2009-10-19 17:27:24.578354 [DEBUG] switch_core_state_machine.c:567
(sofia/internal/sip:1000 at 85.16.245.228:1024) State REPORTING going to sleep
2009-10-19 17:27:24.578354 [DEBUG] switch_core_state_machine.c:319
(sofia/internal/sip:1000 at 85.16.245.228:1024) State Change CS_REPORTING
- -> CS_DESTROY
2009-10-19 17:27:24.578354 [DEBUG] switch_core_session.c:969 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-19 17:27:24.578354 [DEBUG] switch_core_state_machine.c:476
(sofia/internal/1001 at 85.16.246.12:5061) State HANGUP
2009-10-19 17:27:24.578354 [DEBUG] mod_sofia.c:344 Channel
sofia/internal/1001 at 85.16.246.12:5061 hanging up, cause: NORMAL_CLEARING
2009-10-19 17:27:24.578354 [DEBUG] switch_core_session.c:1106 Session 46
(sofia/internal/sip:1000 at 85.16.245.228:1024) Locked, Waiting on external
entities
2009-10-19 17:27:24.578354 [DEBUG] mod_sofia.c:382 Sending BYE to
sofia/internal/1001 at 85.16.246.12:5061
2009-10-19 17:27:24.579127 [DEBUG] switch_core_state_machine.c:46
sofia/internal/1001 at 85.16.246.12:5061 Standard HANGUP, cause:
NORMAL_CLEARING
2009-10-19 17:27:24.579127 [DEBUG] switch_core_state_machine.c:476
(sofia/internal/1001 at 85.16.246.12:5061) State HANGUP going to sleep
2009-10-19 17:27:24.579127 [NOTICE] switch_core_session.c:1124 Session
46 (sofia/internal/sip:1000 at 85.16.245.228:1024) Ended
2009-10-19 17:27:24.579127 [NOTICE] switch_core_session.c:1126 Close
Channel sofia/internal/sip:1000 at 85.16.245.228:1024 [CS_DESTROY]
2009-10-19 17:27:24.579127 [DEBUG] switch_core_state_machine.c:413
(sofia/internal/sip:1000 at 85.16.245.228:1024) Running State Change CS_DESTROY
2009-10-19 17:27:24.579127 [DEBUG] switch_core_state_machine.c:340
(sofia/internal/1001 at 85.16.246.12:5061) State EXECUTE going to sleep
2009-10-19 17:27:24.579127 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_HANGUP
2009-10-19 17:27:24.579127 [DEBUG] switch_core_state_machine.c:424
(sofia/internal/sip:1000 at 85.16.245.228:1024) State DESTROY
2009-10-19 17:27:24.579127 [DEBUG] mod_sofia.c:261
sofia/internal/sip:1000 at 85.16.245.228:1024 SOFIA DESTROY
2009-10-19 17:27:24.579127 [DEBUG] switch_core_state_machine.c:60
sofia/internal/sip:1000 at 85.16.245.228:1024 Standard DESTROY
2009-10-19 17:27:24.579127 [DEBUG] switch_core_state_machine.c:454
handler already called, skipping state handler.
2009-10-19 17:27:24.579127 [DEBUG] switch_core_state_machine.c:424
(sofia/internal/sip:1000 at 85.16.245.228:1024) State DESTROY going to sleep
2009-10-19 17:27:24.579127 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/1001 at 85.16.246.12:5061) State Change CS_HANGUP ->
CS_REPORTING
2009-10-19 17:27:24.579127 [DEBUG] switch_core_session.c:969 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-19 17:27:24.579127 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_REPORTING
2009-10-19 17:27:24.579127 [DEBUG] switch_core_state_machine.c:567
(sofia/internal/1001 at 85.16.246.12:5061) State REPORTING
2009-10-19 17:27:24.581127 [DEBUG] switch_core_state_machine.c:53
sofia/internal/1001 at 85.16.246.12:5061 Standard REPORTING, cause:
NORMAL_CLEARING
2009-10-19 17:27:24.581127 [DEBUG] switch_core_state_machine.c:567
(sofia/internal/1001 at 85.16.246.12:5061) State REPORTING going to sleep
2009-10-19 17:27:24.581127 [DEBUG] switch_core_state_machine.c:319
(sofia/internal/1001 at 85.16.246.12:5061) State Change CS_REPORTING ->
CS_DESTROY
2009-10-19 17:27:24.581127 [DEBUG] switch_core_session.c:969 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-19 17:27:24.581127 [DEBUG] switch_core_session.c:1106 Session 45
(sofia/internal/1001 at 85.16.246.12:5061) Locked, Waiting on external entities
2009-10-19 17:27:24.581127 [NOTICE] switch_core_session.c:1124 Session
45 (sofia/internal/1001 at 85.16.246.12:5061) Ended
2009-10-19 17:27:24.581127 [NOTICE] switch_core_session.c:1126 Close
Channel sofia/internal/1001 at 85.16.246.12:5061 [CS_DESTROY]
2009-10-19 17:27:24.581127 [DEBUG] switch_core_state_machine.c:413
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_DESTROY
2009-10-19 17:27:24.581127 [DEBUG] switch_core_state_machine.c:424
(sofia/internal/1001 at 85.16.246.12:5061) State DESTROY
2009-10-19 17:27:24.581127 [DEBUG] mod_sofia.c:261
sofia/internal/1001 at 85.16.246.12:5061 SOFIA DESTROY
2009-10-19 17:27:24.581127 [DEBUG] switch_core_state_machine.c:60
sofia/internal/1001 at 85.16.246.12:5061 Standard DESTROY
2009-10-19 17:27:24.581127 [DEBUG] switch_core_state_machine.c:424
(sofia/internal/1001 at 85.16.246.12:5061) State DESTROY going to sleep





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From mike at jerris.com  Mon Oct 19 08:33:35 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Oct 2009 11:33:35 -0400
Subject: [Freeswitch-users] uuid_send_dtmf fails (was: conference call)
In-Reply-To: <14E8C50A8F4D41559AFCCD8838350F4F@abisoft.biz>
References: <064A5201C10D4D20AC76EB93CA5EA4B5@abisoft.biz><ebbc2d550910140838v3125579ev9fa16e3e83d25770@mail.gmail.com><BE4EDDA84CB64AA98F18D412B51912FC@abisoft.biz><b39273530910150504l5c7411b9g39b9ed33c592e85f@mail.gmail.com><02C2EFD4CF8D4ACF8EEC0F77C4D75028@abisoft.biz>
	<414DC781-A0F6-418E-9AAB-FA4307634932@jerris.com>
	<14E8C50A8F4D41559AFCCD8838350F4F@abisoft.biz>
Message-ID: <5F579783-33FE-4520-9DC9-7C880DFB34EF@jerris.com>

You will want to use sendevent with a unique-id header and a eavesdrop- 
command header.  Also please note you will want to use svn revision  
15175 or later, I just fixed a segfault in that code.

Mike

On Oct 19, 2009, at 11:11 AM, Nikita Belov wrote:

> Thanks, Mike, for idea. But what is the syntax for this session  
> message?
> I tried this:
>
> sendmsg e8e4f0ed-a0cc-4dff-b7e1-09eeade5df05
> eavesdrop-command: 1
>
> but it doesn't work.
>
>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org  
>> [mailto:freeswitch-
>> users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
>> Sent: Monday, October 19, 2009 5:19 PM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] uuid_send_dtmf fails (was: conference
>> call)
>>
>> There is an event you can send as well to switch them, it your trying
>> to switch it via event socket, that should be better, its not on the
>> wiki, but
>>
>> a session message with
>> eavesdrop-command header with data as the same as dtmf
>> should do the trick
>>
>> Mike
>>
>>
>> On Oct 16, 2009, at 11:54 AM, Nikita Belov wrote:
>>
>>> Yes, it is what I need. But now I have problem with sending dtmf.
>>> Here what
>>> I've done:
>>> [root at centos4-4-vm ~]# telnet localhost 8021
>>> Trying 127.0.0.1...
>>> Connected to localhost.localdomain (127.0.0.1).
>>> Escape character is '^]'.
>>> Content-Type: auth/request
>>>
>>> auth ClueCon
>>>
>>> Content-Type: command/reply
>>> Reply-Text: +OK accepted
>>>
>>> api originate user/1007 at master.agent.starpoundtech.net &park()
>>>
>>> Content-Type: api/response
>>> Content-Length: 41
>>>
>>> +OK bba3b45a-4cc1-48af-a15d-1052d5f11371
>>>
>>> SendMsg bba3b45a-4cc1-48af-a15d-1052d5f11371
>>> call-command: execute
>>> execute-app-name: eavesdrop
>>> execute-app-arg: cd99f999-9b47-457e-8439-1d366e015b8c
>>>
>>> Content-Type: command/reply
>>> Reply-Text: +OK
>>>
>>> Here I had started to hear A and B. Here what I saw in FS log:
>>> 2009-10-18 03:22:47 [DEBUG] switch_core_session.c:706
>>> switch_core_session_queue_private_event() Send signal
>>> sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes [BREAK]
>>> 2009-10-18 03:22:47 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event 
>>> ()
>>> sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes Command Execute
>>> eavesdrop(cd99f999-9b47-457e-8439-1d366e015b8c)
>>> 2009-10-18 03:22:47 [DEBUG] switch_core_media_bug.c:297
>>> switch_core_media_bug_add() Attaching BUG to
>>> sofia/internal/1006 at master.agent.starpoundtech.net
>>> 2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234
>>> switch_core_session_read_frame()
>>> sofia/internal/1006 at master.agent.starpoundtech.net receive message
>>> [TRANSCODING_NECESSARY]
>>> 2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234
>>> switch_core_session_read_frame()
>>> sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes receive message
>>> [TRANSCODING_NECESSARY]
>>>
>>> Then I run command:
>>> api uuid_send_dtmf bba3b45a-4cc1-48af-a15d-1052d5f11371 1
>>>
>>> Content-Type: api/response
>>> Content-Length: 14
>>>
>>> -ERR no reply
>>>
>>> Log:
>>> 2009-10-18 03:24:01 [DEBUG] switch_core_io.c:1190
>>> switch_core_session_send_dtmf_string()
>>> sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes send dtmf
>>> digit=1 ms=250 samples=2000
>>> 2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1282 do_2833() Send start
>>> packet
>>> for [1] ts=2241760 dur=160/160/2000 seq=21346
>>> 2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
>>> packet
>>> for [1] ts=2241760 dur=320/320/2000 seq=21347
>>> 2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
>>> packet
>>> for [1] ts=2241760 dur=480/480/2000 seq=21348
>>> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
>>> packet
>>> for [1] ts=2241760 dur=640/640/2000 seq=21349
>>> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
>>> packet
>>> for [1] ts=2241760 dur=800/800/2000 seq=21350
>>> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
>>> packet
>>> for [1] ts=2241760 dur=960/960/2000 seq=21351
>>> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
>>> packet
>>> for [1] ts=2241760 dur=1120/1120/2000 seq=21352
>>> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
>>> packet
>>> for [1] ts=2241760 dur=1280/1280/2000 seq=21353
>>> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
>>> packet
>>> for [1] ts=2241760 dur=1440/1440/2000 seq=21354
>>> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
>>> packet
>>> for [1] ts=2241760 dur=1600/1600/2000 seq=21355
>>> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
>>> packet
>>> for [1] ts=2241760 dur=1760/1760/2000 seq=21356
>>> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle
>>> packet
>>> for [1] ts=2241760 dur=1920/1920/2000 seq=21357
>>> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end
>>> packet for
>>> [1] ts=2241760 dur=2080/2080/2000 seq=21358
>>> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end
>>> packet for
>>> [1] ts=2241760 dur=2080/2080/2000 seq=21359
>>> 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end
>>> packet for
>>> [1] ts=2241760 dur=2080/2080/2000 seq=21360
>>>
>>> But both A and B couldn't hear me.
>>> Btw, after I had send dtmf "1" manually from my phone. B started to
>>> hear me.
>>> There was this record in log:
>>> 2009-10-18 03:47:55 [DEBUG] switch_rtp.c:1767  
>>> switch_rtp_dequeue_dtmf
>>> () RTP
>>> RECV DTMF 1:2240
>>>
>>> Does anybody know, what had I done wrong?
>>>
>>> ___
>>>
>>> Thanks, Nikita
>>>
>>>> -----Original Message-----
>>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>>> [mailto:freeswitch-
>>>> users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker
>>>> Sent: Thursday, October 15, 2009 4:04 PM
>>>> To: freeswitch-users at lists.freeswitch.org
>>>> Subject: Re: [Freeswitch-users] conference call
>>>>
>>>> The wiki has a pretty good explanation of how eavesdrop works.
>>>> Enabling a talk path to A or B or both A and B requires dtmf.
>>>>
>>>> So, if C hits the 1 button on the phone they can talk to the UUID  
>>>> you
>>>> bound the eavesdrop to.   2 talks to the other party.  3 talks to
>>>> both
>>>> parties.
>>>>
>>>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop
>>>>
>>>> Adding more api calls to control eavesdrop via event socket  
>>>> should be
>>>> possible, though I haven't looked at the code to see what it would
>>>> take...
>>>>
>>>> On Thu, Oct 15, 2009 at 3:16 AM, Nikita Belov <nbelov at abisoft.spb.ru 
>>>> >
>>>> wrote:
>>>>> Thanks for answer.
>>>>>
>>>>>
>>>>>
>>>>> As I've understood eavesdrop allows C to hear A and B, but A and B
>>>>> will
>>>> not
>>>>> hear C. But in my case B SHOULD hear C. In my case A - client, B -
>>>>> agent
>>>>> worked with client and C - coach/supervisor for B. Can I make such
>>>>> conference in FS using eavesdrop or other API command?
>>>>>
>>>>>
>>>>>
>>>>> ___
>>>>>
>>>>>
>>>>>
>>>>> Thanks, Nikita
>>>>>
>>>>>
>>>>>
>>>>> ________________________________
>>>>>
>>>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf
>>>>> Of Joao
>>>>> Mesquita
>>>>> Sent: Wednesday, October 14, 2009 7:38 PM
>>>>> To: freeswitch-users at lists.freeswitch.org
>>>>> Subject: Re: [Freeswitch-users] conference call
>>>>>
>>>>>
>>>>>
>>>>> Look at eavesdrop on the wiki.
>>>>>
>>>>> JM
>>>>>
>>>>> 2009/10/14 Nikita Belov <nbelov at abisoft.spb.ru>
>>>>>
>>>>> HI all,
>>>>>
>>>>>
>>>>>
>>>>> I want to configure FS to make special conference call between  
>>>>> three
>>>> users
>>>>> (A, B, C).  In this conference C will hear A and B, but A will
>>>>> hear only
>>>> B.
>>>>> Can I make it using FS API commands? Does anybody know which
>>>>> approach is
>>>>> better to use?
>>>>>
>>>>>
>>>>>
>>>>> ___
>>>>>
>>>>>
>>>>>
>>>>> Thanks, Nikita
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
>>>>> freeswitch-
>>>>> users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
>>>>> freeswitch-
>>>>> users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> -Rupa
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>> users
>>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From helmut.kuper at ewetel.de  Mon Oct 19 08:53:18 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Mon, 19 Oct 2009 17:53:18 +0200
Subject: [Freeswitch-users] Qustion about INFO
	messages	after	Connect/Answer
In-Reply-To: <4ADC85C0.7060908@ewetel.de>
References: <4AD859A4.5000101@ewetel.de>	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>	<4ADC7C09.6070503@ewetel.de>	<191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>
	<4ADC85C0.7060908@ewetel.de>
Message-ID: <4ADC8B6E.7070404@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Anthony,

just to make it clear: my goal is to avoid to see something (newly
introduced since 1 or 2 weeks) like "1/a/890327" for outgoing in the
caller's display after answering the call (for openzap calls). I want
simply e.g. "89327". I don't want to put the call into "answer" state
before the called side demands it. So I'm not sure if sip_callee_* is
the right way to follow as long as it needs the answer state.

regards
helmut


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From christian.loeschenkohl at xpirio.com  Mon Oct 19 09:07:10 2009
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Mon, 19 Oct 2009 18:07:10 +0200
Subject: [Freeswitch-users] check to see if freeswitch is alive
Message-ID: <4ADC8EAE.8020603@xpirio.com>

hello

we have the problem here that our freeswitch server "freezes" from time
to time (no sip traffic is possible any more).

has somebody here expirience or any idea how to monitor freeswitch to answer
the questions

- is the server alive
- does he process sip messages (invites, registrations)
and so on

the monitoring should work via cron local and remote (with esl for example)

a simple "fs_cli -x 'version'" wouldn't work - it's too simple, am i right?

any ideas would be helpful

br


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com



From jerry.richards at teotech.com  Mon Oct 19 09:13:48 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 19 Oct 2009 09:13:48 -0700
Subject: [Freeswitch-users] Adding Leading Digits To CALLING Number of
	Outgoing Call
Message-ID: <49D347B4795647BF9C35D28EDD207099@greyhawk.tonecommander.com>


How do I use the dial plan to add leading digits to an outgoing call through
a gateway?

My internal phone number is 5380, but when FS sends the call to the gateway
I want the CALLING party to be 4253495380.

Best Regards,
Jerry




From rob4manhere at gmail.com  Mon Oct 19 09:23:44 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Mon, 19 Oct 2009 11:23:44 -0500
Subject: [Freeswitch-users] check to see if freeswitch is alive
In-Reply-To: <4ADC8EAE.8020603@xpirio.com>
References: <4ADC8EAE.8020603@xpirio.com>
Message-ID: <8A38EE37-E03F-48F3-8B54-220BC2DC2363@gmail.com>

Hi Christian,

You can subscribe and monitor the heartbeat event either locally or  
remotely:
http://wiki.freeswitch.org/wiki/Perl_freeswitch_client_example

You could also send an application level SIP packet, like a ping, to  
Freeswitch externally.  I have a small script if you can't find one.

More importantly, while monitoring is good, its probably worth  
investigating the root cause of the freezes since thats not normal.

Rob


On Oct 19, 2009, at 11:07 AM, Christian L?schenkohl wrote:

> hello
>
> we have the problem here that our freeswitch server "freezes" from  
> time
> to time (no sip traffic is possible any more).
>
> has somebody here expirience or any idea how to monitor freeswitch  
> to answer
> the questions
>
> - is the server alive
> - does he process sip messages (invites, registrations)
> and so on
>
> the monitoring should work via cron local and remote (with esl for  
> example)
>
> a simple "fs_cli -x 'version'" wouldn't work - it's too simple, am i  
> right?
>
> any ideas would be helpful
>
> br
>
>
> -- 
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 (0) 5 77 11 - 1000
> F  +43 (0) 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From woof at iwoof.org  Mon Oct 19 09:28:47 2009
From: woof at iwoof.org (Andy Spitzer)
Date: Mon, 19 Oct 2009 12:28:47 -0400
Subject: [Freeswitch-users] check to see if freeswitch is alive
In-Reply-To: <4ADC8EAE.8020603@xpirio.com>
References: <4ADC8EAE.8020603@xpirio.com>
Message-ID: <op.u112p9fjg0dksq@woof3.fuzzy>

Woof!

On Mon, 19 Oct 2009 12:07:10 -0400, Christian L?schenkohl  
<christian.loeschenkohl at xpirio.com> wrote:

> any ideas would be helpful

We run a perl script that checks if the servers are responding to  
requests.  It can send OPTIONS, and PING requests to various servers  
periodically.  If the response it gets back isn't correct, it can send  
e-mail.  It'll keep testing and send an e-mail again when the system is  
working again.  It's defaults setup for testing sipXecs servers, but it's  
easy enough to set the various command line arguments to point it at  
FreeSWITCH directly.

Code here.
    http://code.sipfoundry.org/browse/~raw,r=14874/sipXecs/main/sipXtools/src/sipx-servtest.in

--Woof!



From woof at iwoof.org  Mon Oct 19 09:28:47 2009
From: woof at iwoof.org (Andy Spitzer)
Date: Mon, 19 Oct 2009 12:28:47 -0400
Subject: [Freeswitch-users] check to see if freeswitch is alive
In-Reply-To: <4ADC8EAE.8020603@xpirio.com>
References: <4ADC8EAE.8020603@xpirio.com>
Message-ID: <op.u112p9fjg0dksq@woof3.fuzzy>

Woof!

On Mon, 19 Oct 2009 12:07:10 -0400, Christian L?schenkohl  
<christian.loeschenkohl at xpirio.com> wrote:

> any ideas would be helpful

We run a perl script that checks if the servers are responding to  
requests.  It can send OPTIONS, and PING requests to various servers  
periodically.  If the response it gets back isn't correct, it can send  
e-mail.  It'll keep testing and send an e-mail again when the system is  
working again.  It's defaults setup for testing sipXecs servers, but it's  
easy enough to set the various command line arguments to point it at  
FreeSWITCH directly.

Code here.
    http://code.sipfoundry.org/browse/~raw,r=14874/sipXecs/main/sipXtools/src/sipx-servtest.in

--Woof!



From msc at freeswitch.org  Mon Oct 19 09:39:53 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 19 Oct 2009 09:39:53 -0700
Subject: [Freeswitch-users] check to see if freeswitch is alive
In-Reply-To: <4ADC8EAE.8020603@xpirio.com>
References: <4ADC8EAE.8020603@xpirio.com>
Message-ID: <87f2f3b90910190939h29ced665gb9681a8946d00212@mail.gmail.com>

2009/10/19 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>

> hello
>
> we have the problem here that our freeswitch server "freezes" from time
> to time (no sip traffic is possible any more).
>

Monitoring is definitely important, but I'm sure the FS devs would like to
know more about what happens when FS "freezes" - is it FS or some other
process that is messing things up? If there's an issue with FS then the
developers would like to know more.
-MC
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From nbelov at abisoft.spb.ru  Mon Oct 19 09:42:26 2009
From: nbelov at abisoft.spb.ru (Nikita Belov)
Date: Mon, 19 Oct 2009 20:42:26 +0400
Subject: [Freeswitch-users] uuid_send_dtmf fails (was: conference call)
In-Reply-To: <5F579783-33FE-4520-9DC9-7C880DFB34EF@jerris.com>
References: <064A5201C10D4D20AC76EB93CA5EA4B5@abisoft.biz><ebbc2d550910140838v3125579ev9fa16e3e83d25770@mail.gmail.com><BE4EDDA84CB64AA98F18D412B51912FC@abisoft.biz><b39273530910150504l5c7411b9g39b9ed33c592e85f@mail.gmail.com><02C2EFD4CF8D4ACF8EEC0F77C4D75028@abisoft.biz><414DC781-A0F6-418E-9AAB-FA4307634932@jerris.com><14E8C50A8F4D41559AFCCD8838350F4F@abisoft.biz>
	<5F579783-33FE-4520-9DC9-7C880DFB34EF@jerris.com>
Message-ID: <FFF06EC411464736B44C44B58646A8C6@abisoft.biz>

And what event name to use for sendevent command? Sorry for importunity.

sendevent ???

Unique-ID: b9d6a35c-ee0c-4203-8d65-ed816e0a9c19

eavesdrop-command: 1

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Monday, October 19, 2009 7:34 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] uuid_send_dtmf fails (was: conference call)

 

You will want to use sendevent with a unique-id header and a
eavesdrop-command header.  Also please note you will want to use svn
revision 15175 or later, I just fixed a segfault in that code.

 

Mike

 

On Oct 19, 2009, at 11:11 AM, Nikita Belov wrote:





Thanks, Mike, for idea. But what is the syntax for this session message?
I tried this:

sendmsg e8e4f0ed-a0cc-4dff-b7e1-09eeade5df05
eavesdrop-command: 1

but it doesn't work.





-----Original Message-----

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-

users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris

Sent: Monday, October 19, 2009 5:19 PM

To: freeswitch-users at lists.freeswitch.org

Subject: Re: [Freeswitch-users] uuid_send_dtmf fails (was: conference

call)

 

There is an event you can send as well to switch them, it your trying

to switch it via event socket, that should be better, its not on the

wiki, but

 

a session message with

eavesdrop-command header with data as the same as dtmf

should do the trick

 

Mike

 

 

On Oct 16, 2009, at 11:54 AM, Nikita Belov wrote:

 

Yes, it is what I need. But now I have problem with sending dtmf.

Here what

I've done:

[root at centos4-4-vm ~]# telnet localhost 8021

Trying 127.0.0.1...

Connected to localhost.localdomain (127.0.0.1).

Escape character is '^]'.

Content-Type: auth/request

 

auth ClueCon

 

Content-Type: command/reply

Reply-Text: +OK accepted

 

api originate user/1007 at master.agent.starpoundtech.net &park()

 

Content-Type: api/response

Content-Length: 41

 

+OK bba3b45a-4cc1-48af-a15d-1052d5f11371

 

SendMsg bba3b45a-4cc1-48af-a15d-1052d5f11371

call-command: execute

execute-app-name: eavesdrop

execute-app-arg: cd99f999-9b47-457e-8439-1d366e015b8c

 

Content-Type: command/reply

Reply-Text: +OK

 

Here I had started to hear A and B. Here what I saw in FS log:

2009-10-18 03:22:47 [DEBUG] switch_core_session.c:706

switch_core_session_queue_private_event() Send signal

sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes [BREAK]

2009-10-18 03:22:47 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event()

sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes Command Execute

eavesdrop(cd99f999-9b47-457e-8439-1d366e015b8c)

2009-10-18 03:22:47 [DEBUG] switch_core_media_bug.c:297

switch_core_media_bug_add() Attaching BUG to

sofia/internal/1006 at master.agent.starpoundtech.net

2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234

switch_core_session_read_frame()

sofia/internal/1006 at master.agent.starpoundtech.net receive message

[TRANSCODING_NECESSARY]

2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234

switch_core_session_read_frame()

sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes receive message

[TRANSCODING_NECESSARY]

 

Then I run command:

api uuid_send_dtmf bba3b45a-4cc1-48af-a15d-1052d5f11371 1

 

Content-Type: api/response

Content-Length: 14

 

-ERR no reply

 

Log:

2009-10-18 03:24:01 [DEBUG] switch_core_io.c:1190

switch_core_session_send_dtmf_string()

sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes send dtmf

digit=1 ms=250 samples=2000

2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1282 do_2833() Send start

packet

for [1] ts=2241760 dur=160/160/2000 seq=21346

2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle

packet

for [1] ts=2241760 dur=320/320/2000 seq=21347

2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle

packet

for [1] ts=2241760 dur=480/480/2000 seq=21348

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle

packet

for [1] ts=2241760 dur=640/640/2000 seq=21349

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle

packet

for [1] ts=2241760 dur=800/800/2000 seq=21350

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle

packet

for [1] ts=2241760 dur=960/960/2000 seq=21351

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle

packet

for [1] ts=2241760 dur=1120/1120/2000 seq=21352

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle

packet

for [1] ts=2241760 dur=1280/1280/2000 seq=21353

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle

packet

for [1] ts=2241760 dur=1440/1440/2000 seq=21354

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle

packet

for [1] ts=2241760 dur=1600/1600/2000 seq=21355

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle

packet

for [1] ts=2241760 dur=1760/1760/2000 seq=21356

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle

packet

for [1] ts=2241760 dur=1920/1920/2000 seq=21357

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end

packet for

[1] ts=2241760 dur=2080/2080/2000 seq=21358

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end

packet for

[1] ts=2241760 dur=2080/2080/2000 seq=21359

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end

packet for

[1] ts=2241760 dur=2080/2080/2000 seq=21360

 

But both A and B couldn't hear me.

Btw, after I had send dtmf "1" manually from my phone. B started to

hear me.

There was this record in log:

2009-10-18 03:47:55 [DEBUG] switch_rtp.c:1767 switch_rtp_dequeue_dtmf

() RTP

RECV DTMF 1:2240

 

Does anybody know, what had I done wrong?

 

___

 

Thanks, Nikita

 

-----Original Message-----

From: freeswitch-users-bounces at lists.freeswitch.org

[mailto:freeswitch-

users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker

Sent: Thursday, October 15, 2009 4:04 PM

To: freeswitch-users at lists.freeswitch.org

Subject: Re: [Freeswitch-users] conference call

 

The wiki has a pretty good explanation of how eavesdrop works.

Enabling a talk path to A or B or both A and B requires dtmf.

 

So, if C hits the 1 button on the phone they can talk to the UUID you

bound the eavesdrop to.   2 talks to the other party.  3 talks to

both

parties.

 

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop

 

Adding more api calls to control eavesdrop via event socket should be

possible, though I haven't looked at the code to see what it would

take...

 

On Thu, Oct 15, 2009 at 3:16 AM, Nikita Belov <nbelov at abisoft.spb.ru>

wrote:

Thanks for answer.

 

 

 

As I've understood eavesdrop allows C to hear A and B, but A and B

will

not

hear C. But in my case B SHOULD hear C. In my case A - client, B -

agent

worked with client and C - coach/supervisor for B. Can I make such

conference in FS using eavesdrop or other API command?

 

 

 

___

 

 

 

Thanks, Nikita

 

 

 

________________________________

 

From: freeswitch-users-bounces at lists.freeswitch.org

[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf

Of Joao

Mesquita

Sent: Wednesday, October 14, 2009 7:38 PM

To: freeswitch-users at lists.freeswitch.org

Subject: Re: [Freeswitch-users] conference call

 

 

 

Look at eavesdrop on the wiki.

 

JM

 

2009/10/14 Nikita Belov <nbelov at abisoft.spb.ru>

 

HI all,

 

 

 

I want to configure FS to make special conference call between three

users

(A, B, C).  In this conference C will hear A and B, but A will

hear only

B.

Can I make it using FS API commands? Does anybody know which

approach is

better to use?

 

 

 

___

 

 

 

Thanks, Nikita

 

_______________________________________________

FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-

users

http://www.freeswitch.org

 

 

 

_______________________________________________

FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-

users

http://www.freeswitch.org

 

 

 

 

 

--

-Rupa

 

_______________________________________________

FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-

users

http://www.freeswitch.org

 

 

 

_______________________________________________

FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-

users

http://www.freeswitch.org

 

 

_______________________________________________

FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org




_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

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From brian at freeswitch.org  Mon Oct 19 09:58:49 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 19 Oct 2009 11:58:49 -0500
Subject: [Freeswitch-users] uuid_send_dtmf fails (was: conference call)
In-Reply-To: <FFF06EC411464736B44C44B58646A8C6@abisoft.biz>
References: <064A5201C10D4D20AC76EB93CA5EA4B5@abisoft.biz><ebbc2d550910140838v3125579ev9fa16e3e83d25770@mail.gmail.com><BE4EDDA84CB64AA98F18D412B51912FC@abisoft.biz><b39273530910150504l5c7411b9g39b9ed33c592e85f@mail.gmail.com><02C2EFD4CF8D4ACF8EEC0F77C4D75028@abisoft.biz><414DC781-A0F6-418E-9AAB-FA4307634932@jerris.com><14E8C50A8F4D41559AFCCD8838350F4F@abisoft.biz>
	<5F579783-33FE-4520-9DC9-7C880DFB34EF@jerris.com>
	<FFF06EC411464736B44C44B58646A8C6@abisoft.biz>
Message-ID: <01AAA3C0-54DE-4BE0-B1D4-35B5B9C493DF@freeswitch.org>

Please review this:

http://wiki.freeswitch.org/wiki/Mod_event_socket#sendmsg

SendMsg <uuid>
call-command: execute
execute-app-name: <one of the applications>
execute-app-arg: <application data>



/b

On Oct 19, 2009, at 11:42 AM, Nikita Belov wrote:

> And what event name to use for sendevent command? Sorry for  
> importunity.
> sendevent ???
> Unique-ID: b9d6a35c-ee0c-4203-8d65-ed816e0a9c19
> eavesdrop-command: 1
>




From jerry.richards at teotech.com  Mon Oct 19 10:15:21 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 19 Oct 2009 10:15:21 -0700
Subject: [Freeswitch-users] Adding Leading Digits To CALLING Number of
	Outgoing Call
Message-ID: <3121A49E9BE549989158DDC0008498E7@greyhawk.tonecommander.com>


Okay, I figured it out.  I added the following line to the default.xml file,
just prior to the "bridge" action:

<action application="set"
data="effective_caller_id_number=425349${caller_id_number}"/>

Now, 425349 is prepended to the outgoing call's caller ID.

Best Regards,
Jerry


-----Original Message-----
From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Monday, October 19, 2009 9:14 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: Adding Leading Digits To CALLING Number of Outgoing Call


How do I use the dial plan to add leading digits to an outgoing call through
a gateway?

My internal phone number is 5380, but when FS sends the call to the gateway
I want the CALLING party to be 4253495380.

Best Regards,
Jerry




From m.sobkow at marketelsystems.com  Mon Oct 19 10:35:10 2009
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Mon, 19 Oct 2009 11:35:10 -0600
Subject: [Freeswitch-users] Is anyone running Ubuntu 8.04/Hardy?
Message-ID: <4ADCA34E.5000700@marketelsystems.com>

Everyone I've emailed with on the dev list is running the current 
release of Ubuntu, not 8.04/Hardy.  We're having some problems with 
loading mod_sofia on Hardy, so unless I get some progress/help by end of 
day Wednesday we're going to have to rebuild our development server with 
the most recent release of Ubuntu on Thursday and Friday.

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 




From nbelov at abisoft.spb.ru  Mon Oct 19 10:36:32 2009
From: nbelov at abisoft.spb.ru (Nikita Belov)
Date: Mon, 19 Oct 2009 21:36:32 +0400
Subject: [Freeswitch-users] uuid_send_dtmf fails (was: conference call)
In-Reply-To: <01AAA3C0-54DE-4BE0-B1D4-35B5B9C493DF@freeswitch.org>
References: <064A5201C10D4D20AC76EB93CA5EA4B5@abisoft.biz><ebbc2d550910140838v3125579ev9fa16e3e83d25770@mail.gmail.com><BE4EDDA84CB64AA98F18D412B51912FC@abisoft.biz><b39273530910150504l5c7411b9g39b9ed33c592e85f@mail.gmail.com><02C2EFD4CF8D4ACF8EEC0F77C4D75028@abisoft.biz><414DC781-A0F6-418E-9AAB-FA4307634932@jerris.com><14E8C50A8F4D41559AFCCD8838350F4F@abisoft.biz><5F579783-33FE-4520-9DC9-7C880DFB34EF@jerris.com><FFF06EC411464736B44C44B58646A8C6@abisoft.biz>
	<01AAA3C0-54DE-4BE0-B1D4-35B5B9C493DF@freeswitch.org>
Message-ID: <3F5FD88C28BF428CAA935EDE7357B686@abisoft.biz>

Yeah. I'm using it for starting eavesdrop:
SendMsg cced4b9a-b6de-4be1-8c12-3d18cc6e8454
call-command: execute
execute-app-name: eavesdrop
execute-app-arg: 723f3dbb-b87b-4cd4-98fc-698eed7f2bdb

But we are discussing how to switch eavesdrop to allow "spy" to speak with
one of the sides. Do you know any application which can I use for this? Mike
said I should use sendevent command to control eavesdrop, but I still
confused how can I do it.

> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> users-bounces at lists.freeswitch.org] On Behalf Of Brian West
> Sent: Monday, October 19, 2009 8:59 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] uuid_send_dtmf fails (was: conference
> call)
> 
> Please review this:
> 
> http://wiki.freeswitch.org/wiki/Mod_event_socket#sendmsg
> 
> SendMsg <uuid>
> call-command: execute
> execute-app-name: <one of the applications>
> execute-app-arg: <application data>
> 
> 
> 
> /b
> 
> On Oct 19, 2009, at 11:42 AM, Nikita Belov wrote:
> 
> > And what event name to use for sendevent command? Sorry for
> > importunity.
> > sendevent ???
> > Unique-ID: b9d6a35c-ee0c-4203-8d65-ed816e0a9c19
> > eavesdrop-command: 1
> >
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From chris at cloudtel.com  Mon Oct 19 10:38:28 2009
From: chris at cloudtel.com (Chris Burns)
Date: Mon, 19 Oct 2009 13:38:28 -0400
Subject: [Freeswitch-users] check to see if freeswitch is alive
In-Reply-To: <87f2f3b90910190939h29ced665gb9681a8946d00212@mail.gmail.com>
References: <4ADC8EAE.8020603@xpirio.com>
	<87f2f3b90910190939h29ced665gb9681a8946d00212@mail.gmail.com>
Message-ID: <200910191338.28439.chris@cloudtel.com>

There's a check_sip plugin for nagios if you're into that sorta thing

On October 19, 2009 12:39:53 pm Michael Collins wrote:
> 2009/10/19 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>
>
> > hello
> >
> > we have the problem here that our freeswitch server "freezes" from time
> > to time (no sip traffic is possible any more).
>
> Monitoring is definitely important, but I'm sure the FS devs would like to
> know more about what happens when FS "freezes" - is it FS or some other
> process that is messing things up? If there's an issue with FS then the
> developers would like to know more.
> -MC





From gmaruzz at celliax.org  Mon Oct 19 10:46:28 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 19 Oct 2009 19:46:28 +0200
Subject: [Freeswitch-users] Is anyone running Ubuntu 8.04/Hardy?
In-Reply-To: <4ADCA34E.5000700@marketelsystems.com>
References: <4ADCA34E.5000700@marketelsystems.com>
Message-ID: <7b197bef0910191046y6aa41986nf1a4f77061385bd7@mail.gmail.com>

8.04 was known to be very stable and reliable with FreeSWITCH.

First time I heard of problems with it...

What kind of problems? Problems you have not found with current version?

-giovanni

On Mon, Oct 19, 2009 at 7:35 PM, Mark Sobkow
<m.sobkow at marketelsystems.com> wrote:
> Everyone I've emailed with on the dev list is running the current
> release of Ubuntu, not 8.04/Hardy. ?We're having some problems with
> loading mod_sofia on Hardy, so unless I get some progress/help by end of
> day Wednesday we're going to have to rebuild our development server with
> the most recent release of Ubuntu on Thursday and Friday.
>
> --
> Mark Sobkow
> Senior Developer
> MarkeTel Multi-Line Dialing Systems LTD.
> 428 Victoria Ave
> Regina, SK S4N-0P6
> Toll-Free: 800-289-8616-X533
> Local: 306-359-6893-X533
> Fax: 306-359-6879
> Email: m.sobkow at marketelsystems.com
> Web: http://www.marketelsystems.com
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From christian.loeschenkohl at xpirio.com  Mon Oct 19 11:07:19 2009
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Mon, 19 Oct 2009 20:07:19 +0200
Subject: [Freeswitch-users] check to see if freeswitch is alive
In-Reply-To: <87f2f3b90910190939h29ced665gb9681a8946d00212@mail.gmail.com>
References: <4ADC8EAE.8020603@xpirio.com>
	<87f2f3b90910190939h29ced665gb9681a8946d00212@mail.gmail.com>
Message-ID: <4ADCAAD7.8010401@xpirio.com>

i know, we tried to to get closer to this with anthony (see thread "stability problems",
started on 03.09.2009).
the old issue described a server that used more profiles (8-10) and some event socket scripts.
we came to the conclusion that it could be a race condition when using multiple profiles.

now on the server we need monitor we use only internal functions and conditions
- no external scripts
- no database
- only on profile
- only freeswitch stock functions

the symptoms are the same
- no more progress displays on the fs_cli console (debug level info)
- invites are answered with trying, but no more communication after this
- customers said that option requests are answered (they could not switch to fallback
   because this worked)

we did use the trunk version and have now updated to 15162, but i don't think our problem is
solved with any update so far.

br

On 2009-10-19 18:39, Michael Collins wrote:
>
>
> 2009/10/19 Christian L?schenkohl <christian.loeschenkohl at xpirio.com
> <mailto:christian.loeschenkohl at xpirio.com>>
>
>     hello
>
>     we have the problem here that our freeswitch server "freezes" from time
>     to time (no sip traffic is possible any more).
>
>
> Monitoring is definitely important, but I'm sure the FS devs would like
> to know more about what happens when FS "freezes" - is it FS or some
> other process that is messing things up? If there's an issue with FS
> then the developers would like to know more.
> -MC
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com



From christian.loeschenkohl at xpirio.com  Mon Oct 19 11:18:07 2009
From: christian.loeschenkohl at xpirio.com (=?UTF-8?B?Q2hyaXN0aWFuIEzDtnNjaGVua29obA==?=)
Date: Mon, 19 Oct 2009 20:18:07 +0200
Subject: [Freeswitch-users] check to see if freeswitch is alive
In-Reply-To: <op.u112p9fjg0dksq@woof3.fuzzy>
References: <4ADC8EAE.8020603@xpirio.com> <op.u112p9fjg0dksq@woof3.fuzzy>
Message-ID: <4ADCAD5F.7020702@xpirio.com>

thank you for the script, we will try this asap

br

On 2009-10-19 18:28, Andy Spitzer wrote:
> Woof!
>
> On Mon, 19 Oct 2009 12:07:10 -0400, Christian L?schenkohl
> <christian.loeschenkohl at xpirio.com>  wrote:
>
>> any ideas would be helpful
>
> We run a perl script that checks if the servers are responding to
> requests.  It can send OPTIONS, and PING requests to various servers
> periodically.  If the response it gets back isn't correct, it can send
> e-mail.  It'll keep testing and send an e-mail again when the system is
> working again.  It's defaults setup for testing sipXecs servers, but it's
> easy enough to set the various command line arguments to point it at
> FreeSWITCH directly.
>
> Code here.
>      http://code.sipfoundry.org/browse/~raw,r=14874/sipXecs/main/sipXtools/src/sipx-servtest.in
>
> --Woof!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com



From anthony.minessale at gmail.com  Mon Oct 19 11:19:25 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 19 Oct 2009 13:19:25 -0500
Subject: [Freeswitch-users] check to see if freeswitch is alive
In-Reply-To: <4ADCAAD7.8010401@xpirio.com>
References: <4ADC8EAE.8020603@xpirio.com>
	<87f2f3b90910190939h29ced665gb9681a8946d00212@mail.gmail.com>
	<4ADCAAD7.8010401@xpirio.com>
Message-ID: <191c3a030910191119n3dae38d8y17c5361f74724d50@mail.gmail.com>

do you have a jira ticket open for this?
The mailing list is not a good place to ask for bugs to be fixed because I
can't possibly scour them for responses.

If you can update to latest trunk and still reproduce this problem:
make sure gdb and gcore is installed on your box

from the FS build root enter

./support-d/fscore_pb gcore

and submit the url it generates to me


2009/10/19 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>

> i know, we tried to to get closer to this with anthony (see thread
> "stability problems",
> started on 03.09.2009).
> the old issue described a server that used more profiles (8-10) and some
> event socket scripts.
> we came to the conclusion that it could be a race condition when using
> multiple profiles.
>
> now on the server we need monitor we use only internal functions and
> conditions
> - no external scripts
> - no database
> - only on profile
> - only freeswitch stock functions
>
> the symptoms are the same
> - no more progress displays on the fs_cli console (debug level info)
> - invites are answered with trying, but no more communication after this
> - customers said that option requests are answered (they could not switch
> to fallback
>   because this worked)
>
> we did use the trunk version and have now updated to 15162, but i don't
> think our problem is
> solved with any update so far.
>
> br
>
> On 2009-10-19 18:39, Michael Collins wrote:
> >
> >
> > 2009/10/19 Christian L?schenkohl <christian.loeschenkohl at xpirio.com
> > <mailto:christian.loeschenkohl at xpirio.com>>
> >
> >     hello
> >
> >     we have the problem here that our freeswitch server "freezes" from
> time
> >     to time (no sip traffic is possible any more).
> >
> >
> > Monitoring is definitely important, but I'm sure the FS devs would like
> > to know more about what happens when FS "freezes" - is it FS or some
> > other process that is messing things up? If there's an issue with FS
> > then the developers would like to know more.
> > -MC
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 (0) 5 77 11 - 1000
> F  +43 (0) 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Mon Oct 19 11:44:50 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 19 Oct 2009 11:44:50 -0700
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
In-Reply-To: <4256bf830910182148mf7f2bf9vdcccd687e0caa1e7@mail.gmail.com>
References: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
	<177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>
	<4256bf830908041151l55980e3fifb85c75b87535426@mail.gmail.com>
	<3D0B855A-5B2C-4C6F-9306-64B0ECE97091@jerris.com>
	<4256bf830910180725y72539d98gb138934afc18d58b@mail.gmail.com>
	<A5ECDCD9-1A12-4A31-89F0-E6F4C79FE9E9@jerris.com>
	<4256bf830910182148mf7f2bf9vdcccd687e0caa1e7@mail.gmail.com>
Message-ID: <87f2f3b90910191144q25750bayfd7a7448e82e78ff@mail.gmail.com>

On Sun, Oct 18, 2009 at 9:48 PM, Matthew Fong <mattdfong at gmail.com> wrote:

> The debug level logs to the console, right? The pastebin, had log level
> debug, sofia trace on for external and default, and sofia loglevel all 9. Is
> there another log enable command I'm missing? It seems loglevel all 9
> outputs enter and exit functions, but at least to my novice eye, it's not
> too obvious why freeswitch is sending a BYE to my DID provider.
>
> Evidently the console loglevel debug did not take because there isn't a
single DEBUG level line of output. Please try again. Just remember that
debug level log messages will have A LOT of information so be patient while
sifting through it. Of course, feel free to pastebin the output and add a
link to it in this email thread.

-MC
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From freeswitch-list at puzzled.xs4all.nl  Mon Oct 19 11:45:25 2009
From: freeswitch-list at puzzled.xs4all.nl (Patrick List)
Date: Mon, 19 Oct 2009 20:45:25 +0200
Subject: [Freeswitch-users] Status of ubuntu/debian packages.
In-Reply-To: <20091019130717.GG13895@base.carmickle.com>
References: <4ABA47FB.2050100@gmail.com>	<90823c940910180249u69f18aaamc383f9a5a0c65848@mail.gmail.com>	<4ADAFC85.6000301@puzzled.xs4all.nl>
	<20091019130717.GG13895@base.carmickle.com>
Message-ID: <4ADCB3C5.1070409@puzzled.xs4all.nl>

On 10/19/2009 03:07 PM, Frank Carmickle wrote:
> On Sun, Oct 18, Patrick List wrote:
>> so there was no chance of your customized configs getting overwritten.
>> Example:
>> /opt/freeswitch/etc/freeswitch-$svn_version-samples/
>>
>> Maybe the maintainer of the debs can do something similar.
>
> We are planning on putting all of the different configs in /usr/share/doc/freeswitch/examples.  Now if I could find the time to get this done.

That would obviously work. The advantage of sticking example configs in 
their own folder designating the release (svn rev) they belong to is 
that you can do a diff between two folders to see what has changed. 
Maybe that's useful to some. Just a thought though.

Regards,
Patrick



From msc at freeswitch.org  Mon Oct 19 11:46:54 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 19 Oct 2009 11:46:54 -0700
Subject: [Freeswitch-users] Adding Leading Digits To CALLING Number of
	Outgoing Call
In-Reply-To: <3121A49E9BE549989158DDC0008498E7@greyhawk.tonecommander.com>
References: <3121A49E9BE549989158DDC0008498E7@greyhawk.tonecommander.com>
Message-ID: <87f2f3b90910191146m7a0215abjc2eddda40621eef8@mail.gmail.com>

On Mon, Oct 19, 2009 at 10:15 AM, Jerry Richards <jerry.richards at teotech.com
> wrote:

>
> Okay, I figured it out.  I added the following line to the default.xml
> file,
> just prior to the "bridge" action:
>
Way to go all Chuck Norris on this issue! :)
-MC
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From anthony.minessale at gmail.com  Mon Oct 19 13:07:04 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 19 Oct 2009 15:07:04 -0500
Subject: [Freeswitch-users] Qustion about INFO messages after
	Connect/Answer
In-Reply-To: <4ADC8B6E.7070404@ewetel.de>
References: <4AD859A4.5000101@ewetel.de>
	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>
	<4ADC7C09.6070503@ewetel.de>
	<191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>
	<4ADC85C0.7060908@ewetel.de> <4ADC8B6E.7070404@ewetel.de>
Message-ID: <191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>

please update and test trunk

1) I changed the core to remove the excess data by default in your scenario
2) I added variables you can use to control it origination_callee_id_name
origination_callee_id_number which belong in {} in the dial string eg
{origination_callee_id_number=1234}openzap/1/a/1234


On Mon, Oct 19, 2009 at 10:53 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi Anthony,
>
> just to make it clear: my goal is to avoid to see something (newly
> introduced since 1 or 2 weeks) like "1/a/890327" for outgoing in the
> caller's display after answering the call (for openzap calls). I want
> simply e.g. "89327". I don't want to put the call into "answer" state
> before the called side demands it. So I'm not sure if sip_callee_* is
> the right way to follow as long as it needs the answer state.
>
> regards
> helmut
>
>
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
>
> iD8DBQFK3Itu4tZeNddg3dwRAvREAKCrZcRM+TnypdVOczpe2+q26b3BYQCfadrY
> tdA2XIc/hpuIv/916eu4GFo=
> =y6Md
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From dmitry.bely at gmail.com  Mon Oct 19 13:23:19 2009
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Tue, 20 Oct 2009 00:23:19 +0400
Subject: [Freeswitch-users] Status of ubuntu/debian packages.
In-Reply-To: <4ADCB3C5.1070409@puzzled.xs4all.nl>
References: <4ABA47FB.2050100@gmail.com>
	<90823c940910180249u69f18aaamc383f9a5a0c65848@mail.gmail.com>
	<4ADAFC85.6000301@puzzled.xs4all.nl>
	<20091019130717.GG13895@base.carmickle.com>
	<4ADCB3C5.1070409@puzzled.xs4all.nl>
Message-ID: <90823c940910191323m9497ca6t7b78c9af6ec43294@mail.gmail.com>

On Mon, Oct 19, 2009 at 10:45 PM, Patrick List
<freeswitch-list at puzzled.xs4all.nl> wrote:
> On 10/19/2009 03:07 PM, Frank Carmickle wrote:
>> On Sun, Oct 18, Patrick List wrote:
>>> so there was no chance of your customized configs getting overwritten.
>>> Example:
>>> /opt/freeswitch/etc/freeswitch-$svn_version-samples/
>>>
>>> Maybe the maintainer of the debs can do something similar.
>>
>> We are planning on putting all of the different configs in /usr/share/doc/freeswitch/examples. ?Now if I could find the time to get this done.
>
> That would obviously work. The advantage of sticking example configs in
> their own folder designating the release (svn rev) they belong to is
> that you can do a diff between two folders to see what has changed.
> Maybe that's useful to some. Just a thought though.

For most Debian packages dpkg just asks the user what he/she would
like to do with the newer config file from .deb: keep the old file,
replace it with the new file, diff old with new files etc. Not an
option here?

- Dmitry Bely



From miles.chet at gmail.com  Mon Oct 19 13:24:19 2009
From: miles.chet at gmail.com (Roberto Martins)
Date: Mon, 19 Oct 2009 18:24:19 -0200
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <86a32abc0910171711je1e57bduf2bb672472365ce1@mail.gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<6701786D-A8AF-444E-A5A7-29E69F6E055C@gmail.com>
	<B69597BF-DA4A-440B-8CB0-08F52F33DD17@jerris.com>
	<86a32abc0910171339o6a300932v66b7f00419a0cb69@mail.gmail.com>
	<86a32abc0910171513n35d7d0echff940a16685ae1a4@mail.gmail.com>
	<86a32abc0910171521n403094bfvf27aa7c2a1c1f798@mail.gmail.com>
	<86a32abc0910171535l606eb9c6id54f7b3b5ae1b670@mail.gmail.com>
	<86a32abc0910171545l17f3f539ldba432db550c3eaa@mail.gmail.com>
	<86a32abc0910171601u6ab06e42xcf303301c85c57b1@mail.gmail.com>
	<86a32abc0910171605y73ecb75cqfb13923ad19adc56@mail.gmail.com>
	<86a32abc0910171711je1e57bduf2bb672472365ce1@mail.gmail.com>
Message-ID: <00E26046-328F-440A-81F4-671F197F114C@gmail.com>

what about http://www.atlassian.com/software/confluence/ they give  
free licenses to open source project, and FS is using JIRA.

roberto

On Oct 17, 2009, at 9:11 PM, Diego Viola wrote:

> http://moinmo.in/ is also pretty cool it seems and it's used for the  
> apache wiki and others.
>
> http://wiki.apache.org/
>
> It also has i18n support, full text search, etc.
>
> Diego
>
> On Sat, Oct 17, 2009 at 11:05 PM, Diego Viola  
> <diego.viola at gmail.com> wrote:
> With sub-domains you'll need to start all over and it will be a pain  
> to keep the documentation in-sync.
>
> With a wiki that supports real translation it's easier to add a new  
> line with the translation tag or whatever.
>
> Diego
>
>
> On Sat, Oct 17, 2009 at 11:01 PM, Diego Viola  
> <diego.viola at gmail.com> wrote:
> Take a look at this for instance:
>
> http://wiki.rubyonrails.org/start
>
> This uses DokuWiki, check the langauges option at the top (en, de,  
> el, es, fr, etc) and when you click one of them it translates the  
> actual page to the desired language... I'd love to have something  
> like that.
>
> Diego
>
>
> On Sat, Oct 17, 2009 at 10:45 PM, Diego Viola  
> <diego.viola at gmail.com> wrote:
> Instead of adding million of different domains for the languages, it  
> would be nice if we only keep wiki.freeswitch.org, and the wiki  
> software provides a list of languages to choose from.
>
> Then when we click in one of the languages we have the wiki  
> translated to the desired language.
>
> I believe if we go this way we will be more organized, and there is  
> more chances that things will get translated, and we wont be  
> repeating too much information here and there too.
>
> Just a thought... I'm open to do this work to make it happen.
>
> The different domains per language is a good idea too.
>
> Diego
>
>
> On Sat, Oct 17, 2009 at 10:35 PM, Diego Viola  
> <diego.viola at gmail.com> wrote:
> We could also do this for a better documentation.
>
> 	? Pick a new wiki engine that does internalization/translation  
> easier.
> 	? Set some documentation guidelines from the start.
> 	? Move stuff over to the new wiki.
> 	? Start with the translation/internalization on the new wiki.
>
> Diego
>
>
> On Sat, Oct 17, 2009 at 10:21 PM, Diego Viola  
> <diego.viola at gmail.com> wrote:
> Some interesting pages on the subject.
>
> http://en.wikipedia.org/wiki/Comparison_of_wiki_software
> http://www.osnews.com/story/21865/Recommend_an_Open_Source_Wiki_Platform
>
>
>
>
> On Sat, Oct 17, 2009 at 10:13 PM, Diego Viola  
> <diego.viola at gmail.com> wrote:
> Would you be open to test another wiki software, that supports  
> internalization/translation easier, if i am open to port the entire  
> documentation to the new wiki software?
>
>
>
>
>
> On Sat, Oct 17, 2009 at 8:39 PM, Diego Viola <diego.viola at gmail.com>  
> wrote:
> If we find a better wiki software that does internalization in a  
> better way, I could help porting the documentation over to the new  
> software.
>
> The sub-domain idea is also very nice.
>
> Diego
>
>
>
> On Sat, Oct 17, 2009 at 8:29 PM, Michael Jerris <mike at jerris.com>  
> wrote:
> I think we strongly lean towards using a sub-domain (if necessary) and
> maintaining other language content in the same wiki in alternate
> pages.  If the wiki software we are using is not effective to create
> multiple languages we should find a way to do it all in one.  We can
> set up an additional mailing list if necessary but also we have no
> issue with going across multiple languages in both the mailing list
> and in irc.
>
> Mike
>
> On Oct 17, 2009, at 4:01 PM, Roberto Martins wrote:
>
> > sub-domain seem to be a very wise choice...
> > ptbr
> >
> >
> >
> > On Oct 17, 2009, at 1:25 PM, Anthony Minessale wrote:
> >
> >> or someone could just ask us for pg.freeswitch.org or whatever  
> other
> >> sub-domain to avoid trademark infringement of registering 40  
> domains
> >> with freeswitch in the name for no reason.
> >>
> >>
> >> On Sat, Oct 17, 2009 at 6:06 AM, Rud? Cunha <ruda at ruda.com.br>  
> wrote:
> >> Jo?o Mesquita,
> >>
> >> Could check with Jeremiah or contact me through, so I talk to him
> >> about the forum?
> >>
> >> He could redirect the domain to my hosting server and we can put
> >> there the wiki and forums and we can communicate!
> >>
> >> Rud? Cunha
> >>
> >> 2009/10/17 Jo?o Mesquita <jmesquita at freeswitch.org>
> >>
> >> Just a heads up, I have talked to Jeremias from Khomp today and he
> >> is setting up the wiki. I will personally be adding contents to the
> >> that wiki if it ever picks up.
> >>
> >> Regards,
> >>
> >> jm
> >>
> >>
> >> On Sat, Oct 17, 2009 at 12:52 AM, Jason White <jason at jasonjgw.net>
> >> wrote:
> >> Diego Viola <diego.viola at gmail.com> wrote:
> >>> I'm with Moises and with the other people supporting this
> >> initiative.
> >>>
> >>> I'm not Brazilian, but they should be able to do whatever they
> >> want, after
> >>> all, that's how open source works, if you can do it go ahead and
> >> do it.
> >>
> >> Correct. We have enough of them as far as English-language fora are
> >> concerned;
> >> other languages are a different question altogether, though.
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org
> >> iax:guest at conference.freeswitch.org/888
> >> googletalk:conf+888 at conference.freeswitch.org
> >> pstn:213-799-1400
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From dmitry.bely at gmail.com  Mon Oct 19 13:32:04 2009
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Tue, 20 Oct 2009 00:32:04 +0400
Subject: [Freeswitch-users] Status of ubuntu/debian packages.
In-Reply-To: <4ABA47FB.2050100@gmail.com>
References: <4ABA47FB.2050100@gmail.com>
Message-ID: <90823c940910191332r719970f0m2adce06ad40738d1@mail.gmail.com>

Can you build FreeSWITCH .deb package with tls transport support?
Anyway it already depends on openssl package.

Another problem:

2009-10-20 00:28:50.728557 [CRIT] switch_loadable_module.c:871 Error
Loading module /opt/freeswitch/mod/mod_cluechoo.so
**/opt/freeswitch/mod/mod_cluechoo.so: cannot open shared object file:
No such file or directory**
2009-10-20 00:28:50.728604 [CRIT] switch_loadable_module.c:871 Error
Loading module /opt/freeswitch/mod/mod_valet_parking.so
**/opt/freeswitch/mod/mod_valet_parking.so: cannot open shared object
file: No such file or directory**

Modules mod_cluechoo & mod_valet_parking are referenced in the config
but not present in the package.

- Dmitry Bely



From abovebluesky at gmail.com  Mon Oct 19 13:41:37 2009
From: abovebluesky at gmail.com (Uncle Johny)
Date: Mon, 19 Oct 2009 13:41:37 -0700 (PDT)
Subject: [Freeswitch-users]  bind_meta_app problem with custome extension
Message-ID: <25963971.post@talk.nabble.com>


Hi guys,

I hope you can help me with this one since it looks like some FS bug or my
user error :) .I have a simple dialplan in FS which allows to transfer a
person to some 3rd party phone but it works only if you call it from default
context.
If it is not called from default context I get this error all the time:
2009-09-12 11:58:17.484680 [ERR] mod_native_file.c:68 Error opening
/usr/local/freeswitch/conf/dialplan/execute_extension:roar XML features.PCMU

With the following configuration 2 parties will be connected but whenever
you press *7 it just does not work and FS splits the message. Very weird
that this works if I put the extension into default.xml.

Any ideas what is wrong here?

My config:

I tried to create my own context and call it pstn.
So when FS gets a call from proxy it goes to pstn context, since I have this
line in sip_proxies/internal.xml


in dialplan directory I have pstn.xml
<include>
  <context name="pstn">
   <extension name="pstn_enum">
    <condition field="destination_number" expression="^(.*)$">
                <action application="answer"/>
                <action application="bind_meta_app" data="7 ab s
/usr/local/freeswitch/conf/dialplan/execute_extension:roar XML features"/>
                <action application="set" data="hangup_after_bridge=true"/>
                <action application="bridge"
data="sofia/external/${destination_number}@207.88.122.16"/>
    </condition>
  </extension>
</context>
</include>

In features.xml (It can be anything there since FS never reaches this code
and dies before executing it):
<extension name="roar">
     <condition field="destination_number" expression="^roar$">
        <action application="set" data="continue_on_fail=true"/>
        <action application="log" data="INFO transfer context" />
        <action application="read" data="1 1 /root/1.gsm attxfer_callthis
1000 #"/>
        <action application="log" data="INFO blah ${attxfer_callthis}"/>
        <action application="set" data="origination_cancel_key=#"/>
        <action application="log" data="INFO after read"/>
     </condition>
</extension>

 
-- 
View this message in context: http://www.nabble.com/bind_meta_app-problem-with-custome-extension-tp25963971p25963971.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From ineyaa at gmail.com  Mon Oct 19 13:51:07 2009
From: ineyaa at gmail.com (ineya ineya)
Date: Mon, 19 Oct 2009 22:51:07 +0200
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <5B924C57-B658-44C0-A1CE-00A3ECB5AA07@freeswitch.org>
References: <7b8592a0910190516rafa4398q62c7a99ef703540f@mail.gmail.com>
	<5B924C57-B658-44C0-A1CE-00A3ECB5AA07@freeswitch.org>
Message-ID: <7b8592a0910191351g14cbb47dj3f19362ff6197157@mail.gmail.com>

Here is internal.xml
http://pastebin.freeswitch.org/10766

and internal-ipv6.xml
http://pastebin.freeswitch.org/10767



On Mon, Oct 19, 2009 at 5:13 PM, Brian West <brian at freeswitch.org> wrote:

> I think you just have it misconfigured because if ipv6 was broken I would
> have William and Jason jumping on me.
> Can you post your xml configs for the profile please?
>
> /b
>
> On Oct 19, 2009, at 7:16 AM, ineya ineya wrote:
>
> Hi,
>
> I installed the phonenix release, and changed IPv4, IPv6 addresses and SIP
> port. Directory and dialplan should be default ones.
> My phones are able to register when using just IPv4, and I can call from
> one to another.
>
> When I try the same thing with IPv6, the call to other phone fails with [
> USER_NOT_REGISTERED], however I'm able to call 9999 and hear MOH.
> Here you can find the log: http://pastebin.freeswitch.org/10757
> Freeswitch is running on 10.80.62.40/2000:2::1
> phones are 2000:2::1001 and 2000:2::1002
>
> Am I doing something wrong? Maybe I forgot to configure something, but I
> can't see what it is.
>
> Any hints are appreciated.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Mon Oct 19 14:24:55 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 19 Oct 2009 16:24:55 -0500
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <7b8592a0910191351g14cbb47dj3f19362ff6197157@mail.gmail.com>
References: <7b8592a0910190516rafa4398q62c7a99ef703540f@mail.gmail.com>
	<5B924C57-B658-44C0-A1CE-00A3ECB5AA07@freeswitch.org>
	<7b8592a0910191351g14cbb47dj3f19362ff6197157@mail.gmail.com>
Message-ID: <18202C58-EF07-43F6-9649-A40134B30240@freeswitch.org>

And you're sure you have that IP on your interface?

William or Jason care to let me know if latest SVN IPv6 works for you?

/b

On Oct 19, 2009, at 3:51 PM, ineya ineya wrote:

> Here is internal.xml
> http://pastebin.freeswitch.org/10766
>
> and internal-ipv6.xml
> http://pastebin.freeswitch.org/10767
>
>
>
> On Mon, Oct 19, 2009 at 5:13 PM, Brian West <brian at freeswitch.org>  
> wrote:
> I think you just have it misconfigured because if ipv6 was broken I  
> would have William and Jason jumping on me.
>
> Can you post your xml configs for the profile please?
>
> /b

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From quentusrex at gmail.com  Mon Oct 19 14:28:39 2009
From: quentusrex at gmail.com (William King)
Date: Mon, 19 Oct 2009 14:28:39 -0700
Subject: [Freeswitch-users] Status of ubuntu/debian packages.
In-Reply-To: <90823c940910191332r719970f0m2adce06ad40738d1@mail.gmail.com>
References: <4ABA47FB.2050100@gmail.com>
	<90823c940910191332r719970f0m2adce06ad40738d1@mail.gmail.com>
Message-ID: <4ADCDA07.2000001@gmail.com>

Thanks for the catch. I have fixed the valet_parking as of last night.
And cluechoo will be fixed after this upcoming nightly.

Does anyone know how the other packages deal with locally modified
config files? I like the idea of adding the diffs, but prompting first
per file to see if it should be ignored, diffed, or overwritten.

-William King

Dmitry Bely wrote:
> Can you build FreeSWITCH .deb package with tls transport support?
> Anyway it already depends on openssl package.
>
> Another problem:
>
> 2009-10-20 00:28:50.728557 [CRIT] switch_loadable_module.c:871 Error
> Loading module /opt/freeswitch/mod/mod_cluechoo.so
> **/opt/freeswitch/mod/mod_cluechoo.so: cannot open shared object file:
> No such file or directory**
> 2009-10-20 00:28:50.728604 [CRIT] switch_loadable_module.c:871 Error
> Loading module /opt/freeswitch/mod/mod_valet_parking.so
> **/opt/freeswitch/mod/mod_valet_parking.so: cannot open shared object
> file: No such file or directory**
>
> Modules mod_cluechoo & mod_valet_parking are referenced in the config
> but not present in the package.
>
> - Dmitry Bely
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From msc at freeswitch.org  Mon Oct 19 14:35:10 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 19 Oct 2009 14:35:10 -0700
Subject: [Freeswitch-users] Qustion about INFO messages after
	Connect/Answer
In-Reply-To: <191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>
References: <4AD859A4.5000101@ewetel.de>
	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>
	<4ADC7C09.6070503@ewetel.de>
	<191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>
	<4ADC85C0.7060908@ewetel.de> <4ADC8B6E.7070404@ewetel.de>
	<191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>
Message-ID: <87f2f3b90910191435s7528e256r85df5f12fc634f24@mail.gmail.com>

On Mon, Oct 19, 2009 at 1:07 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> please update and test trunk
>
> 1) I changed the core to remove the excess data by default in your scenario
> 2) I added variables you can use to control it origination_callee_id_name
> origination_callee_id_number which belong in {} in the dial string eg
> {origination_callee_id_number=1234}openzap/1/a/1234
>
>
> After you test, please confirm the behavior and then we'll update the wiki
on these two new chan vars.
-MC
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From mcampbellsmith at gmail.com  Mon Oct 19 14:40:26 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 20 Oct 2009 08:40:26 +1100
Subject: [Freeswitch-users] validating dtmf digits received
In-Reply-To: <81433357-3203-4208-98C1-FF3FE2AC2D1E@jerris.com>
References: <33c87fa30910182153t23e52fa0p2b967b3843eb1175@mail.gmail.com>
	<81433357-3203-4208-98C1-FF3FE2AC2D1E@jerris.com>
Message-ID: <33c87fa30910191440ib033cd6l64bded22394915cc@mail.gmail.com>

Thanks Mike,

I have a lateish trunk and inline seems to work okay.

Does the inline statement below set variable ${code} to be used
directly or does it require transfer also?  ie is digits_dialed
available for use right after a read statement (<action
application="read" data="1 10
ivr/ivr-please_enter_pin_followed_by_pound.wav res 10000 9"/> in my
case) or is it not 'set' until after the transfer?

        <action inline="true" application="set" data="code=${digits_dialed}

Thanks!

On Tue, Oct 20, 2009 at 12:32 AM, Michael Jerris <mike at jerris.com> wrote:
> inline is new, it won't work unless your using recent trunk. ?That
> being said, read is not being run inline, so the set is actually being
> run before digits_dialed is set. ?You will most likely need to use
> transfer in this situation.
>
> Mike
>
> On Oct 19, 2009, at 12:53 AM, Mark Campbell-Smith wrote:
>
>> Hi!
>>
>> I simply want to validate the dtmf digits I read from a user. ? ?From
>> the wiki, it appears I need to use inline="true" when setting the
>> variable so it can be used directly within the same extension.
>>
>> What have I done wrong below? ? I have tried many different
>> alternatives, but the second condition field, which is meant to match
>> the dtmf digits received (in this case 0000) is never matched, and the
>> anti-action is called instead.
>>
>> :
>> ? ? ?<some code here>
>> ? ? ?<action application="read" data="1 10
>> ivr/ivr-please_enter_pin_followed_by_pound.wav res 10000 9"/>
>> ? ? ? <action application="phrase" data="spell,${res}"/>
>> ? ? ? <action inline="true" application="set" data="code=$
>> {digits_dialed}"/>
>> ? ? ? <!-- <action inline="true" application="set" data="code=$
>> {res}"/> -->
>> ?</condition>
>> ?<condition field="digits_dialed" expression="^0000$">
>> ?<!-- <condition field="${code}" expression="^0000$"> -->
>> ?<!-- <condition field="${res}" expression="^0000$"> -->
>> ? ? ? ?<some code here>
>> ? ? ? <anti-action application="hangup"/>
>>
>> Thanks!
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From rupa at rupa.com  Mon Oct 19 14:40:30 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 19 Oct 2009 16:40:30 -0500
Subject: [Freeswitch-users] Status of ubuntu/debian packages.
In-Reply-To: <4ADCDA07.2000001@gmail.com>
References: <4ABA47FB.2050100@gmail.com>
	<90823c940910191332r719970f0m2adce06ad40738d1@mail.gmail.com>
	<4ADCDA07.2000001@gmail.com>
Message-ID: <b39273530910191440s6efd081bie3d8b3138dbae55e@mail.gmail.com>

I was under the impression that if you mark the files as conffiles you
get the default behavior which prompts for diff/overwrite/edit
options.  If you want to get real fancy, look into using ucf to do
three-way diffs between installed default, user modifications and new
default.

Finally, I have checked the current package state, but please put all
the confs in a separate package and make it a suggests not a depends.
I absolutely hate how debian packages keep creating files that I've
removed (ex: in our case would be eg: 1000.xml).  Since conf is so
small,  maybe two.  one that puts them in /etc for people that want
quick out of the box setup and one that puts 'em in
/usr/share/doc/freeswitch/examples for people that want to use the
current defaults as a sample rather than as the basis for their
config.

On Mon, Oct 19, 2009 at 4:28 PM, William King <quentusrex at gmail.com> wrote:
> Thanks for the catch. I have fixed the valet_parking as of last night.
> And cluechoo will be fixed after this upcoming nightly.
>
> Does anyone know how the other packages deal with locally modified
> config files? I like the idea of adding the diffs, but prompting first
> per file to see if it should be ignored, diffed, or overwritten.
>
> -William King
>
> Dmitry Bely wrote:
>> Can you build FreeSWITCH .deb package with tls transport support?
>> Anyway it already depends on openssl package.
>>
>> Another problem:
>>
>> 2009-10-20 00:28:50.728557 [CRIT] switch_loadable_module.c:871 Error
>> Loading module /opt/freeswitch/mod/mod_cluechoo.so
>> **/opt/freeswitch/mod/mod_cluechoo.so: cannot open shared object file:
>> No such file or directory**
>> 2009-10-20 00:28:50.728604 [CRIT] switch_loadable_module.c:871 Error
>> Loading module /opt/freeswitch/mod/mod_valet_parking.so
>> **/opt/freeswitch/mod/mod_valet_parking.so: cannot open shared object
>> file: No such file or directory**
>>
>> Modules mod_cluechoo & mod_valet_parking are referenced in the config
>> but not present in the package.
>>
>> - Dmitry Bely
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa



From hads at nice.net.nz  Mon Oct 19 14:43:59 2009
From: hads at nice.net.nz (Hadley Rich)
Date: Tue, 20 Oct 2009 10:43:59 +1300
Subject: [Freeswitch-users] Status of ubuntu/debian packages.
In-Reply-To: <4ADCDA07.2000001@gmail.com>
References: <4ABA47FB.2050100@gmail.com>
	<90823c940910191332r719970f0m2adce06ad40738d1@mail.gmail.com>
	<4ADCDA07.2000001@gmail.com>
Message-ID: <1255988639.3830.1.camel@sodium>

On Mon, 2009-10-19 at 14:28 -0700, William King wrote:
> Does anyone know how the other packages deal with locally modified
> config files? I like the idea of adding the diffs, but prompting first
> per file to see if it should be ignored, diffed, or overwritten.

dpkg handles config files;

http://www.debian.org/doc/debian-policy/ap-pkg-conffiles.html

hads

-- 
http://nicegear.co.nz
New Zealand's Open Source Hardware Supplier




From mike at jerris.com  Mon Oct 19 15:55:54 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Oct 2009 18:55:54 -0400
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <00E26046-328F-440A-81F4-671F197F114C@gmail.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<6701786D-A8AF-444E-A5A7-29E69F6E055C@gmail.com>
	<B69597BF-DA4A-440B-8CB0-08F52F33DD17@jerris.com>
	<86a32abc0910171339o6a300932v66b7f00419a0cb69@mail.gmail.com>
	<86a32abc0910171513n35d7d0echff940a16685ae1a4@mail.gmail.com>
	<86a32abc0910171521n403094bfvf27aa7c2a1c1f798@mail.gmail.com>
	<86a32abc0910171535l606eb9c6id54f7b3b5ae1b670@mail.gmail.com>
	<86a32abc0910171545l17f3f539ldba432db550c3eaa@mail.gmail.com>
	<86a32abc0910171601u6ab06e42xcf303301c85c57b1@mail.gmail.com>
	<86a32abc0910171605y73ecb75cqfb13923ad19adc56@mail.gmail.com>
	<86a32abc0910171711je1e57bduf2bb672472365ce1@mail.gmail.com>
	<00E26046-328F-440A-81F4-671F197F114C@gmail.com>
Message-ID: <7A73AEA0-9138-4E02-897A-85159405A4EE@jerris.com>

we do have a license for this, people didn't seem to like it last time  
we looked at it, I can't recall why.

On Oct 19, 2009, at 4:24 PM, Roberto Martins wrote:

> what about http://www.atlassian.com/software/confluence/ they give
> free licenses to open source project, and FS is using JIRA.
>
> roberto
>



From mike at jerris.com  Mon Oct 19 16:00:13 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Oct 2009 19:00:13 -0400
Subject: [Freeswitch-users] validating dtmf digits received
In-Reply-To: <33c87fa30910191440ib033cd6l64bded22394915cc@mail.gmail.com>
References: <33c87fa30910182153t23e52fa0p2b967b3843eb1175@mail.gmail.com>
	<81433357-3203-4208-98C1-FF3FE2AC2D1E@jerris.com>
	<33c87fa30910191440ib033cd6l64bded22394915cc@mail.gmail.com>
Message-ID: <490A9EB1-0CAD-421A-A66C-FBA4C52E9F22@jerris.com>

inline is run when the dialplan in parsed, everything else is run  
later.  So read sets digits dialed after it is finished parsing the  
dialplan, if you transfer to another extensions after the read you can  
then condition on that value.

Mike

On Oct 19, 2009, at 5:40 PM, Mark Campbell-Smith wrote:

> Thanks Mike,
>
> I have a lateish trunk and inline seems to work okay.
>
> Does the inline statement below set variable ${code} to be used
> directly or does it require transfer also?  ie is digits_dialed
> available for use right after a read statement (<action
> application="read" data="1 10
> ivr/ivr-please_enter_pin_followed_by_pound.wav res 10000 9"/> in my
> case) or is it not 'set' until after the transfer?
>
>        <action inline="true" application="set" data="code=$ 
> {digits_dialed}
>
> Thanks!
>
> On Tue, Oct 20, 2009 at 12:32 AM, Michael Jerris <mike at jerris.com>  
> wrote:
>> inline is new, it won't work unless your using recent trunk.  That
>> being said, read is not being run inline, so the set is actually  
>> being
>> run before digits_dialed is set.  You will most likely need to use
>> transfer in this situation.
>>
>> Mike
>>
>> On Oct 19, 2009, at 12:53 AM, Mark Campbell-Smith wrote:
>>
>>> Hi!
>>>
>>> I simply want to validate the dtmf digits I read from a user.     
>>> From
>>> the wiki, it appears I need to use inline="true" when setting the
>>> variable so it can be used directly within the same extension.
>>>
>>> What have I done wrong below?   I have tried many different
>>> alternatives, but the second condition field, which is meant to  
>>> match
>>> the dtmf digits received (in this case 0000) is never matched, and  
>>> the
>>> anti-action is called instead.
>>>
>>> :
>>>      <some code here>
>>>      <action application="read" data="1 10
>>> ivr/ivr-please_enter_pin_followed_by_pound.wav res 10000 9"/>
>>>       <action application="phrase" data="spell,${res}"/>
>>>       <action inline="true" application="set" data="code=$
>>> {digits_dialed}"/>
>>>       <!-- <action inline="true" application="set" data="code=$
>>> {res}"/> -->
>>>  </condition>
>>>  <condition field="digits_dialed" expression="^0000$">
>>>  <!-- <condition field="${code}" expression="^0000$"> -->
>>>  <!-- <condition field="${res}" expression="^0000$"> -->
>>>        <some code here>
>>>       <anti-action application="hangup"/>
>>>
>>> Thanks!
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From diego.viola at gmail.com  Mon Oct 19 17:35:38 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 20 Oct 2009 00:35:38 +0000
Subject: [Freeswitch-users] Excellent FreeSWITCH Article For You To Read
In-Reply-To: <87f2f3b90910182037x5c1d1279ue50f724f0a02f94c@mail.gmail.com>
References: <87f2f3b90910182037x5c1d1279ue50f724f0a02f94c@mail.gmail.com>
Message-ID: <86a32abc0910191735l53a5f146o8c46844d6902ffb8@mail.gmail.com>

Nice story.

I see you guys submitted to Slashdot as well, but they have not accepted it
yet.

So please vote the story on Slashdot as well so they accept it :D.

http://slashdot.org/submission/1095795/Open-Source-Carrier-Grade-Telephone-Switch

Diego

On Mon, Oct 19, 2009 at 3:37 AM, Michael Collins <msc at freeswitch.org> wrote:

> Hello all,
>
> Just a heads up that there is a great article out there for you to read.
> Please visit the main freeswitch.org page here:
> http://www.freeswitch.org/node/211
>
> Click the link and digg the article after you read it. Please spread the
> word! The twittersphere is starting to warm up already. Let's keep the
> momentum going!
>
> Thanks,
> Michael
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Mon Oct 19 17:41:05 2009
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 20 Oct 2009 08:41:05 +0800
Subject: [Freeswitch-users] uuid_send_dtmf fails (was: conference call)
In-Reply-To: <3F5FD88C28BF428CAA935EDE7357B686@abisoft.biz>
References: <064A5201C10D4D20AC76EB93CA5EA4B5@abisoft.biz>
	<BE4EDDA84CB64AA98F18D412B51912FC@abisoft.biz>
	<b39273530910150504l5c7411b9g39b9ed33c592e85f@mail.gmail.com>
	<02C2EFD4CF8D4ACF8EEC0F77C4D75028@abisoft.biz>
	<414DC781-A0F6-418E-9AAB-FA4307634932@jerris.com>
	<14E8C50A8F4D41559AFCCD8838350F4F@abisoft.biz>
	<5F579783-33FE-4520-9DC9-7C880DFB34EF@jerris.com>
	<FFF06EC411464736B44C44B58646A8C6@abisoft.biz>
	<01AAA3C0-54DE-4BE0-B1D4-35B5B9C493DF@freeswitch.org>
	<3F5FD88C28BF428CAA935EDE7357B686@abisoft.biz>
Message-ID: <23f91030910191741u5c6093e4w73e3351cd8769aa6@mail.gmail.com>

not sure about this, but did you try send dtmf to uuid
723f3dbb-b87b-4cd4-98fc-698eed7f2bdb other than cced4b9a-b6de-4be1-8c12-
>
> 3d18cc6e8454 ?
>


2009/10/20 Nikita Belov <nbelov at abisoft.spb.ru>

> Yeah. I'm using it for starting eavesdrop:
> SendMsg cced4b9a-b6de-4be1-8c12-3d18cc6e8454
> call-command: execute
> execute-app-name: eavesdrop
> execute-app-arg: 723f3dbb-b87b-4cd4-98fc-698eed7f2bdb
>
> But we are discussing how to switch eavesdrop to allow "spy" to speak with
> one of the sides. Do you know any application which can I use for this?
> Mike
> said I should use sendevent command to control eavesdrop, but I still
> confused how can I do it.
>
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> > users-bounces at lists.freeswitch.org] On Behalf Of Brian West
> > Sent: Monday, October 19, 2009 8:59 PM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] uuid_send_dtmf fails (was: conference
> > call)
> >
> > Please review this:
> >
> > http://wiki.freeswitch.org/wiki/Mod_event_socket#sendmsg
> >
> > SendMsg <uuid>
> > call-command: execute
> > execute-app-name: <one of the applications>
> > execute-app-arg: <application data>
> >
> >
> >
> > /b
> >
> > On Oct 19, 2009, at 11:42 AM, Nikita Belov wrote:
> >
> > > And what event name to use for sendevent command? Sorry for
> > > importunity.
> > > sendevent ???
> > > Unique-ID: b9d6a35c-ee0c-4203-8d65-ed816e0a9c19
> > > eavesdrop-command: 1
> > >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Mon Oct 19 17:53:56 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 19 Oct 2009 19:53:56 -0500
Subject: [Freeswitch-users] uuid_send_dtmf fails (was: conference call)
In-Reply-To: <23f91030910191741u5c6093e4w73e3351cd8769aa6@mail.gmail.com>
References: <064A5201C10D4D20AC76EB93CA5EA4B5@abisoft.biz>
	<BE4EDDA84CB64AA98F18D412B51912FC@abisoft.biz>
	<b39273530910150504l5c7411b9g39b9ed33c592e85f@mail.gmail.com>
	<02C2EFD4CF8D4ACF8EEC0F77C4D75028@abisoft.biz>
	<414DC781-A0F6-418E-9AAB-FA4307634932@jerris.com>
	<14E8C50A8F4D41559AFCCD8838350F4F@abisoft.biz>
	<5F579783-33FE-4520-9DC9-7C880DFB34EF@jerris.com>
	<FFF06EC411464736B44C44B58646A8C6@abisoft.biz>
	<01AAA3C0-54DE-4BE0-B1D4-35B5B9C493DF@freeswitch.org>
	<3F5FD88C28BF428CAA935EDE7357B686@abisoft.biz>
	<23f91030910191741u5c6093e4w73e3351cd8769aa6@mail.gmail.com>
Message-ID: <D2161B4F-CFEC-4717-AE64-B447C751C3B6@freeswitch.org>

OK what makes you think it failed?  The fact you don't hear it?

/b

On Oct 19, 2009, at 7:41 PM, Seven Du wrote:

> not sure about this, but did you try send dtmf to uuid   723f3dbb- 
> b87b-4cd4-98fc-
> 698eed7f2bdb other than cced4b9a-b6de-4be1-8c12-
> 3d18cc6e8454 ?

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From brian at freeswitch.org  Mon Oct 19 18:05:09 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 19 Oct 2009 20:05:09 -0500
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <7b8592a0910191351g14cbb47dj3f19362ff6197157@mail.gmail.com>
References: <7b8592a0910190516rafa4398q62c7a99ef703540f@mail.gmail.com>
	<5B924C57-B658-44C0-A1CE-00A3ECB5AA07@freeswitch.org>
	<7b8592a0910191351g14cbb47dj3f19362ff6197157@mail.gmail.com>
Message-ID: <FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>

Bill has confirmed it works fine in latest trunk.  So i'm not sure  
what exactly you're doing wrong can you provide me some debug output  
of what you're doing in your dialplan and the console debug?

/b

On Oct 19, 2009, at 3:51 PM, ineya ineya wrote:

> Here is internal.xml
> http://pastebin.freeswitch.org/10766
>
> and internal-ipv6.xml
> http://pastebin.freeswitch.org/10767

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From djbinter at yahoo.com  Mon Oct 19 19:13:43 2009
From: djbinter at yahoo.com (DJB)
Date: Mon, 19 Oct 2009 19:13:43 -0700 (PDT)
Subject: [Freeswitch-users] CPU Usage Question
Message-ID: <144734.94454.qm@web37508.mail.mud.yahoo.com>

We have about 1500 concurrent calls on the server.  And, the CPU usage is shown below:

02:06:46 AM  CPU   %user   %nice    %sys %iowait    %irq   %soft  %steal   %idle    intr/s
02:06:46 AM  all    0.06    0.00    0.03    0.44    0.00    0.00    0.00   99.47   1028.25
02:06:46 AM    0    0.08    0.03    0.03    0.01    0.00    0.01    0.00   99.84   1007.07
02:06:46 AM    1    0.06    0.00    0.02    0.01    0.00    0.00    0.00   99.91      0.00
02:06:46 AM    2    0.07    0.00    0.08    3.49    0.02    0.00    0.00   96.34     21.18
02:06:46 AM    3    0.05    0.00    0.01    0.01    0.00    0.00    0.00   99.94      0.00
02:06:46 AM    4    0.04    0.00    0.02    0.00    0.00    0.00    0.00   99.93      0.00
02:06:46 AM    5    0.05    0.00    0.02    0.00    0.00    0.00    0.00   99.92      0.00
02:06:46 AM    6    0.05    0.00    0.02    0.02    0.00    0.00    0.00   99.91      0.00
02:06:46 AM    7    0.05    0.00    0.02    0.01    0.00    0.00    0.00   99.93      0.00


1) I am wondering whether it is normal that every time when i checked, only CPU 2 always have the highest %iowait (in this case 3.49%).

2) Shoudn't it spread out and balance them to utilize all CPUs evenly?

Thank you.



      



From gabe at gundy.org  Mon Oct 19 19:17:28 2009
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 19 Oct 2009 20:17:28 -0600
Subject: [Freeswitch-users] Is anyone running Ubuntu 8.04/Hardy?
In-Reply-To: <4ADCA34E.5000700@marketelsystems.com>
References: <4ADCA34E.5000700@marketelsystems.com>
Message-ID: <903da5680910191917m79bf5ecfvdc8001952dbcc339@mail.gmail.com>

On Mon, Oct 19, 2009 at 11:35 AM, Mark Sobkow
<m.sobkow at marketelsystems.com> wrote:
> Everyone I've emailed with on the dev list is running the current
> release of Ubuntu, not 8.04/Hardy.

Well, what issues?

Gabe



From bottleman at icf.org.ru  Mon Oct 19 22:14:13 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Tue, 20 Oct 2009 09:14:13 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910140215y357fede1l37f6cba9725ddd89@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<65d96fc80910130014hc5f5fcet677f87b7fb57559e@mail.gmail.com>
	<65d96fc80910132359i6d53d8ban10f943795481bf7c@mail.gmail.com>
	<Pine.LNX.4.64.0910141134270.11620@srv.icf.org.ru>
	<65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>
	<Pine.LNX.4.64.0910141202500.21535@srv.icf.org.ru>
	<65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>
	<65d96fc80910140215y357fede1l37f6cba9725ddd89@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>

On 2009-10-14 11:15 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>On Wed, Oct 14, 2009 at 10:16 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
TC>
TC>>
TC>>
TC>> 2009/10/14 Georgiewskiy Yuriy <bottleman at icf.org.ru>
TC>>
TC>>> On 2009-10-14 09:58 +0200, Tihomir Culjaga wrote
TC>>> freeswitch-users at lists.fre...:
TC>>>
TC>>> i need trace level 4 from mod_h323 and debug log of entire call, tcpdump
TC>>> may be needed later, i have no way
TC>>> to test it on this time, i do it later.
TC>>>
TC>>>
TC>
TC>Here are the logs:
TC>
TC>call flow is H323_endpoint => FS => SIP_user
TC>
TC>slow start                => http://pastebin.freeswitch.org/10693
TC>slow start w eH245            => http://pastebin.freeswitch.org/10694
TC>fastStart w tunneling true         => http://pastebin.freeswitch.org/10701
TC>fastStart w tunneling false        => http://pastebin.freeswitch.org/10702
TC>fastStart w tunneling true eH245    => http://pastebin.freeswitch.org/10699
TC>fastStart w tunneling false eH245    => http://pastebin.freeswitch.org/10700
TC>
TC>
TC>In all cases, there is no audio on SIP User side... i see the SIP Phone
TC>(terminator) sending RTP to FS but FS is not forwarding it back to the
TC>originator. Also, i see originator (H323) sending RTP to FS but FS doesn't
TC>forward it to the terminator (SIP).
TC>
TC>
TC>
TC>call flow is SIP_user => FS => H323_endpoint is failing .. coredumped....
TC>http://pastebin.freeswitch.org/10703

i fix some bugs now, ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/mod_h323.tar.bz2 this is 
updated version, try it, if you experience no audio try enable rtp proxy in you sip profile.

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From bottleman at icf.org.ru  Mon Oct 19 22:17:07 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Tue, 20 Oct 2009 09:17:07 +0400 (MSD)
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <5F65174C-2CE6-4007-B288-C81227776281@freeswitch.org>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910082129430.1684@srv.icf.org.ru>
	<65d96fc80910081132pc27560ejbfbf6f29ed2055c6@mail.gmail.com>
	<Pine.LNX.4.64.0910121728430.17531@srv.icf.org.ru>
	<39EFDD6C-5485-4D99-9234-A1C559B0732B@freeswitch.org>
	<Pine.LNX.4.64.0910121846480.28243@srv.icf.org.ru>
	<65d96fc80910120903m439c87dav8ee6364a7cd57a1c@mail.gmail.com>
	<Pine.LNX.4.64.0910122128130.10300@srv.icf.org.ru>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<FB994F3F-0DC9-4085-A1CC-3EA101CA0ABA@freeswitch.org>
	<65d96fc80910122327u54505b44r4ff53e89293bcf9a@mail.gmail.com>
	<5F65174C-2CE6-4007-B288-C81227776281@freeswitch.org>
Message-ID: <Pine.LNX.4.64.0910200915170.2416@srv.icf.org.ru>

On 2009-10-12 23:31 -0700, Brian West wrote freeswitch-users at lists.freeswit...:

I think now is time to start host it in our svn, there is updated version with some 
fixes and improvements - ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/mod_h323.tar.bz2.

BW>I wouldn't call it donating per se... Its just giving it a place to  
BW>live with easy access for end users without having to do anything  
BW>extra go get it!  ;)
BW>
BW>/b
BW>
BW>On Oct 12, 2009, at 11:27 PM, Tihomir Culjaga wrote:
BW>
BW>> this will be perfect ... but it is up to Yuriy if he is willing to  
BW>> donate his work...
BW>>
BW>> T.
BW>
BW>
BW>_______________________________________________
BW>FreeSWITCH-users mailing list
BW>FreeSWITCH-users at lists.freeswitch.org
BW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
BW>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
BW>http://www.freeswitch.org
BW>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From brian at freeswitch.org  Mon Oct 19 22:28:26 2009
From: brian at freeswitch.org (=?utf-8?B?YnJpYW5AZnJlZXN3aXRjaC5vcmc=?=)
Date: Mon, 19 Oct 2009 22:28:26 -0700 (PDT)
Subject: [Freeswitch-users] =?utf-8?q?CPU_Usage_Question?=
Message-ID: <4add4a7a.1408c00a.4283.ffffc11f@mx.google.com>

It's because Sofia used one thread per profile so depending on load you will see one CPU used more.

/b 

Sent from my HTC on the Now Network from Sprint!

----- Reply message -----
From: "DJB" <djbinter at yahoo.com>
Date: Mon, Oct 19, 2009 9:13 PM
Subject: [Freeswitch-users] CPU Usage Question
To: "FREESWITCH-USERS MAILING LIST" <freeswitch-users at lists.freeswitch.org>

We have about 1500 concurrent calls on the server.  And, the CPU usage is shown below:

02:06:46 AM  CPU   %user   %nice    %sys %iowait    %irq   %soft  %steal   %idle    intr/s
02:06:46 AM  all    0.06    0.00    0.03    0.44    0.00    0.00    0.00   99.47   1028.25
02:06:46 AM    0    0.08    0.03    0.03    0.01    0.00    0.01    0.00   99.84   1007.07
02:06:46 AM    1    0.06    0.00    0.02    0.01    0.00    0.00    0.00   99.91      0.00
02:06:46 AM    2    0.07    0.00    0.08    3.49    0.02    0.00    0.00   96.34     21.18
02:06:46 AM    3    0.05    0.00    0.01    0.01    0.00    0.00    0.00   99.94      0.00
02:06:46 AM    4    0.04    0.00    0.02    0.00    0.00    0.00    0.00   99.93      0.00
02:06:46 AM    5    0.05    0.00    0.02    0.00    0.00    0.00    0.00   99.92      0.00
02:06:46 AM    6    0.05    0.00    0.02    0.02    0.00    0.00    0.00   99.91      0.00
02:06:46 AM    7    0.05    0.00    0.02    0.01    0.00    0.00    0.00   99.93      0.00


1) I am wondering whether it is normal that every time when i checked, only CPU 2 always have the highest %iowait (in this case 3.49%).

2) Shoudn't it spread out and balance them to utilize all CPUs evenly?

Thank you.



      

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From ineyaa at gmail.com  Mon Oct 19 22:53:11 2009
From: ineyaa at gmail.com (ineya ineya)
Date: Tue, 20 Oct 2009 07:53:11 +0200
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
References: <7b8592a0910190516rafa4398q62c7a99ef703540f@mail.gmail.com>
	<5B924C57-B658-44C0-A1CE-00A3ECB5AA07@freeswitch.org>
	<7b8592a0910191351g14cbb47dj3f19362ff6197157@mail.gmail.com>
	<FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
Message-ID: <7b8592a0910192253l1313d82j51c933cb5f55ad41@mail.gmail.com>

IP should be OK:

[root at franta /opt/freeswitch]# ifconfig eth1
eth1????? Link encap:Ethernet? HWaddr 00:4f:4e:62:ad:83
????????? inet addr:10.80.62.40? Bcast:10.80.62.255? Mask:255.255.255.0
????????? inet6 addr: 2000:2::1/32 Scope:Global
????????? inet6 addr: fe80::24f:4eff:fe62:ad83/64 Scope:Link
????????? UP BROADCAST RUNNING MULTICAST? MTU:1500? Metric:1
????????? RX packets:4310768 errors:0 dropped:0 overruns:0 frame:0
????????? TX packets:2537585 errors:0 dropped:0 overruns:0 carrier:0
????????? collisions:114862 txqueuelen:1000
????????? RX bytes:1067582578 (1018.1 MiB)? TX bytes:616134369 (587.5 MiB)
????????? Interrupt:19 Base address:0x3400

Console debug: http://pastebin.freeswitch.org/10768

When I do "sofia status profile internal-ipv6", I can see both phones
as registered:
Call-ID:? ? ? ? 9cb5faf2a052867b
User:? ? ? ? ? ?1007 at franta.openstage.net
Contact:? ? ? ? 1007 <sip:1007@[2000:2::1002]:5062;transport=udp>

but when it comes to connecting the call:

EXECUTE sofia/internal-ipv6/1006 at franta.openstage.net
bridge(user/1007 at franta.openstage.net)
2009-10-20 07:36:12.749270 [DEBUG] switch_ivr_originate.c:1027
variable string 0 = [presence_id=1007 at franta.openstage.net]
2009-10-20 07:36:12.749270 [ERR] switch_ivr_originate.c:1510 Cannot
create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]

@Bill: Can you share your IPv6 test config?

On Tue, Oct 20, 2009 at 3:05 AM, Brian West <brian at freeswitch.org> wrote:
>
> Bill has confirmed it works fine in latest trunk. ?So i'm not sure what exactly you're doing wrong can you provide me some debug output of what you're doing in your dialplan and the console debug?
> /b
> On Oct 19, 2009, at 3:51 PM, ineya ineya wrote:
>
> Here is internal.xml
> http://pastebin.freeswitch.org/10766
>
> and internal-ipv6.xml
> http://pastebin.freeswitch.org/10767
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From lakindia89 at gmail.com  Tue Oct 20 00:18:47 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Tue, 20 Oct 2009 12:48:47 +0530
Subject: [Freeswitch-users] Digium TE220 wiki
Message-ID: <7d79b3930910200018s25184c01k838bff05ad308ffb@mail.gmail.com>

Hi all,
    I've created a wiki page, which contains the example configuration for
making Digium TE220 to work.
    I request you people to check this, and give feedbacks.

http://wiki.freeswitch.org/wiki/Configuration_OpenZap-DigiumTE220-Example
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From helmut.kuper at ewetel.de  Tue Oct 20 00:25:14 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 20 Oct 2009 09:25:14 +0200
Subject: [Freeswitch-users] Qustion about INFO messages
	after	Connect/Answer
In-Reply-To: <191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>
References: <4AD859A4.5000101@ewetel.de>	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>	<4ADC7C09.6070503@ewetel.de>	<191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>	<4ADC85C0.7060908@ewetel.de>
	<4ADC8B6E.7070404@ewetel.de>
	<191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>
Message-ID: <4ADD65DA.4050809@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello Anthony,


On 19.10.2009 22:07, Anthony Minessale wrote:
> please update and test trunk
> 
> 1) I changed the core to remove the excess data by default in your scenario
> 2) I added variables you can use to control it
> origination_callee_id_name origination_callee_id_number which belong in
> {} in the dial string eg {origination_callee_id_number=1234}openzap/1/a/1234

updated and testet for SIP calls. "origination_callee_id_number=1234"
works within the dial string and via using "export" application but not
via "set" application. Didn't test it for openzap, yet but I guess it
will work, too.

What left is that "unknown" thing on callee's display (SIP phones only I
guess) ... I would suggest to make sending "unknown" INFO message
optional, but to be honest, I have no idea for what it was invented, so
maybe my suggestion is nonsens.


regards
Helmut
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eZWafXUZn3LjC07q/1IcsvM=
=7N8O
-----END PGP SIGNATURE-----



From helmut.kuper at ewetel.de  Tue Oct 20 00:35:56 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 20 Oct 2009 09:35:56 +0200
Subject: [Freeswitch-users] Stereo Support
Message-ID: <4ADD685C.1000506@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

I wonder if there is real stereo support planned for FreeSWITCH in terms
of streaming music/video-audio to desktop? I ask, because I heared that
celt codec is supporting stereo.

regards
helmut
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)

iD8DBQFK3Whc4tZeNddg3dwRAkdEAKCLBSV4H8q5JN4NZ39fgSZPxr5PCgCgmpTq
PAu8qozLP5FaVKIwfCXVqjU=
=Zseo
-----END PGP SIGNATURE-----



From helmut.kuper at ewetel.de  Tue Oct 20 00:39:31 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 20 Oct 2009 09:39:31 +0200
Subject: [Freeswitch-users] Is anyone running Ubuntu 8.04/Hardy?
In-Reply-To: <4ADCA34E.5000700@marketelsystems.com>
References: <4ADCA34E.5000700@marketelsystems.com>
Message-ID: <4ADD6933.4040504@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Mark,

I confirm that it works fine. I use it for two years now and there were
no problems compiling or running FS on ubuntu 8.04 caused by the OS.

This is my system:

lsb_release -a
No LSB modules are available.
Distributor ID: Ubuntu
Description:    Ubuntu 8.04.3 LTS
Release:        8.04
Codename:       hardy

uname -a
Linux ippbx-prod-node0 2.6.24-22-server #1 SMP Mon Nov 24 19:14:19 UTC
2008 i686 GNU/Linux


On 19.10.2009 19:35, Mark Sobkow wrote:
> Everyone I've emailed with on the dev list is running the current 
> release of Ubuntu, not 8.04/Hardy.  We're having some problems with 
> loading mod_sofia on Hardy, so unless I get some progress/help by end of 
> day Wednesday we're going to have to rebuild our development server with 
> the most recent release of Ubuntu on Thursday and Friday.
> 

regards
helmut
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3b5Ya5ueoiEY7O4QIrni73U=
=95L5
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From tculjaga at gmail.com  Tue Oct 20 01:17:54 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 20 Oct 2009 10:17:54 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<65d96fc80910130014hc5f5fcet677f87b7fb57559e@mail.gmail.com>
	<65d96fc80910132359i6d53d8ban10f943795481bf7c@mail.gmail.com>
	<Pine.LNX.4.64.0910141134270.11620@srv.icf.org.ru>
	<65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>
	<Pine.LNX.4.64.0910141202500.21535@srv.icf.org.ru>
	<65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>
	<65d96fc80910140215y357fede1l37f6cba9725ddd89@mail.gmail.com>
	<Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>
Message-ID: <65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>

> TC>
> TC>call flow is SIP_user => FS => H323_endpoint is failing ..
> coredumped....
> TC>http://pastebin.freeswitch.org/10703
>
> i fix some bugs now,
> ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/mod_h323.tar.bz2 this
> is
> updated version, try it, if you experience no audio try enable rtp proxy in
> you sip profile.
>
>
Hi,  there are several issues... lets start with top 4 :)



1. I'm still stuck with no audio:

I have this parameter in the sip profile set: <param
name="inbound-proxy-media" value="true"/>
...tried with both with slow start and fast start... any idea ?

pls check: http://pastebin.freeswitch.org/10771





2. outgoing calls still failing in coredumps: what is your dialplan ? ...
how do you call bridge application?

2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3252 Found capability:
UserInput/PointDevice <14>
2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3248 FindCapability: 15
2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external thread
0xb6eb60a0 for id 3048876944
2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:65 Destroyed external thread
0xb6ebafa8 for id 3048876944
2009-10-20 10:08:18.426608 [DEBUG] tlibthrd.cxx:406 Destroyed thread
0xb6ebafa8 PExternalThread:0xb5ba2b90(id = b5ba2b90)
2009-10-20 10:08:18.426608 [DEBUG] h323caps.cxx:3252 Found capability:
UserInput/Modal <15>
2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0,
expiries=0
2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0,
expiries=0
2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external thread
0xb6eba910 for id 3048876944
2009-10-20 10:08:18.426608 [DEBUG] h4601.cxx:1725 Endpoint Attached
Segmentation fault (core dumped)
tculjaga at subZero:~/freeswitch-trunk$

pls check: http://pastebin.freeswitch.org/10769




3.  when you hangup from SIP side, the call is not released end-to-end (the
H323 endpoint doesn't get any releaseComplete message)

2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=2,
expiries=3
2009-10-20 10:10:51.264527 [DEBUG] h323neg.cxx:432 Received
MasterSlaveDeterminationAck: state=Incoming
2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=1,
expiries=3
2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=2,
expiries=4
2009-10-20 10:10:51.264527 [DEBUG] h323.cxx:4138
InternalEstablishedConnectionCheck: connectionState=EstablishedConnection
fastStartState=FastStartAcknowledged
2009-10-20 10:10:51.264527 [DEBUG] h323caps.cxx:3264 FindCapability: "T.120"
2009-10-20 10:10:51.264527 [DEBUG] h323.cxx:4138
InternalEstablishedConnectionCheck: connectionState=EstablishedConnection
fastStartState=FastStartAcknowledged
2009-10-20 10:10:51.264527 [DEBUG] tlibthrd.cxx:1023
PThread::PXBlockOnIO(45,0)
2009-10-20 10:10:54.405479 [NOTICE] sofia.c:328 Hangup sofia/internal/
sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [CS_CONSUME_MEDIA]
[NORMAL_CLEARING]
2009-10-20 10:10:54.405479 [DEBUG] switch_channel.c:1726 Send signal
sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [KILL]
2009-10-20 10:10:54.405479 [DEBUG] switch_core_session.c:932 Send signal
sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [BREAK]
2009-10-20 10:10:54.405479 [DEBUG] switch_core_state_machine.c:437 thread
mismatch skipping state handler.
2009-10-20 10:10:54.405479 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) Running State
Change CS_HANGUP
2009-10-20 10:10:54.406530 [DEBUG] switch_core_state_machine.c:464
(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State HANGUP
2009-10-20 10:10:54.406530 [DEBUG] mod_sofia.c:338 Channel sofia/internal/
sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> hanging up, cause:
NORMAL_CLEARING
2009-10-20 10:10:54.406530 [DEBUG] switch_core_state_machine.c:46
sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> Standard HANGUP,
cause: NORMAL_CLEARING


pls check: http://pastebin.freeswitch.org/10771




4. sometimes when i shutdown FS i get coredimps - from my experience it
looks like you don't wait for a FS thread to finish when you exit...

2009-10-20 10:05:59.493306 [CONSOLE] switch_event.c:508 Stopping queue
thread 2
2009-10-20 10:05:59.493339 [CONSOLE] switch_core.c:1693 Finalizing Shutdown.
2009-10-20 10:05:59.493379 [CONSOLE] switch_log.c:310 Logger Ended.
2009-10-20 10:05:59.494472 [CONSOLE] switch_core_memory.c:567 Stopping
memory pool queue.
Segmentation fault (core dumped)
tculjaga at subZero:~/freeswitch-trunk$
tculjaga at subZero:~/freeswitch-trunk$


Please advice your FS/mod_h323.conf.xml settings...


Tihomir.




> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129-RIPE
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From maciej.aniserowicz at gmail.com  Tue Oct 20 03:30:59 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Tue, 20 Oct 2009 03:30:59 -0700 (PDT)
Subject: [Freeswitch-users] "Can not record session. Media not enabled on
 channel."
Message-ID: <1256034659005-3857858.post@n2.nabble.com>


Hello,
I am using the same set of extensions for testing the system during
development, they include XLite, Cisco sip phone and several extensions that
just play some audio file.
Sometimes, very rarely, this message "Can not record session.  Media not
enabled on channel." appears on FS console.
Like I wrote before, I don't change any codec settings and always use the
same set of devices/emulators.
What can cause this message?

Thanks,
Maciej Aniserowicz
-- 
View this message in context: http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3857858.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From nbelov at abisoft.spb.ru  Tue Oct 20 04:49:05 2009
From: nbelov at abisoft.spb.ru (Nikita Belov)
Date: Tue, 20 Oct 2009 15:49:05 +0400
Subject: [Freeswitch-users] uuid_send_dtmf fails (was: conference call)
In-Reply-To: <D2161B4F-CFEC-4717-AE64-B447C751C3B6@freeswitch.org>
References: <064A5201C10D4D20AC76EB93CA5EA4B5@abisoft.biz><BE4EDDA84CB64AA98F18D412B51912FC@abisoft.biz><b39273530910150504l5c7411b9g39b9ed33c592e85f@mail.gmail.com><02C2EFD4CF8D4ACF8EEC0F77C4D75028@abisoft.biz><414DC781-A0F6-418E-9AAB-FA4307634932@jerris.com><14E8C50A8F4D41559AFCCD8838350F4F@abisoft.biz><5F579783-33FE-4520-9DC9-7C880DFB34EF@jerris.com><FFF06EC411464736B44C44B58646A8C6@abisoft.biz><01AAA3C0-54DE-4BE0-B1D4-35B5B9C493DF@freeswitch.org><3F5FD88C28BF428CAA935EDE7357B686@abisoft.biz><23f91030910191741u5c6093e4w73e3351cd8769aa6@mail.gmail.com>
	<D2161B4F-CFEC-4717-AE64-B447C751C3B6@freeswitch.org>
Message-ID: <D464AFA2661C49C68AF378F8A199998E@abisoft.biz>

Yeah. A call to B and C eavesdrops call. I send dtmf for C to talk with B,
but B can't hear C. Here, what I had done in details:

 

[root at centos4-4-vm ~]# telnet localhost 8021 Trying 127.0.0.1...

Connected to localhost.localdomain (127.0.0.1).

Escape character is '^]'.

Content-Type: auth/request

 

auth ClueCon

 

Content-Type: command/reply

Reply-Text: +OK accepted

 

api originate user/1007 at master.agent.starpoundtech.net &park()

 

Content-Type: api/response

Content-Length: 41

 

+OK bba3b45a-4cc1-48af-a15d-1052d5f11371

 

SendMsg bba3b45a-4cc1-48af-a15d-1052d5f11371

call-command: execute

execute-app-name: eavesdrop

execute-app-arg: cd99f999-9b47-457e-8439-1d366e015b8c

 

Content-Type: command/reply

Reply-Text: +OK

 

Here I had started to hear A and B. Here what I saw in FS log:

2009-10-18 03:22:47 [DEBUG] switch_core_session.c:706

switch_core_session_queue_private_event() Send signal
sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes [BREAK]

2009-10-18 03:22:47 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event()
sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes Command Execute

eavesdrop(cd99f999-9b47-457e-8439-1d366e015b8c)

2009-10-18 03:22:47 [DEBUG] switch_core_media_bug.c:297

switch_core_media_bug_add() Attaching BUG to
sofia/internal/1006 at master.agent.starpoundtech.net

2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234

switch_core_session_read_frame()

sofia/internal/1006 at master.agent.starpoundtech.net receive message
[TRANSCODING_NECESSARY]

2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234

switch_core_session_read_frame()

sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes receive message
[TRANSCODING_NECESSARY]

 

Then I run command:

api uuid_send_dtmf bba3b45a-4cc1-48af-a15d-1052d5f11371 1

 

Content-Type: api/response

Content-Length: 14

 

-ERR no reply

 

Log:

2009-10-18 03:24:01 [DEBUG] switch_core_io.c:1190

switch_core_session_send_dtmf_string()

sofia/internal/sip:1007 at 172.26.10.64:5060;fs_nat=yes send dtmf

digit=1 ms=250 samples=2000

2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1282 do_2833() Send start packet
for [1] ts=2241760 dur=160/160/2000 seq=21346

2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=320/320/2000 seq=21347

2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=480/480/2000 seq=21348

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=640/640/2000 seq=21349

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=800/800/2000 seq=21350

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=960/960/2000 seq=21351

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=1120/1120/2000 seq=21352

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=1280/1280/2000 seq=21353

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=1440/1440/2000 seq=21354

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=1600/1600/2000 seq=21355

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=1760/1760/2000 seq=21356

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet
for [1] ts=2241760 dur=1920/1920/2000 seq=21357

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end packet for
[1] ts=2241760 dur=2080/2080/2000 seq=21358

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end packet for
[1] ts=2241760 dur=2080/2080/2000 seq=21359

2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end packet for
[1] ts=2241760 dur=2080/2080/2000 seq=21360

 

But both A and B couldn't hear me. 

Btw, after I had send dtmf "1" manually from my phone. B started to hear me.

There was this record in log:

2009-10-18 03:47:55 [DEBUG] switch_rtp.c:1767 switch_rtp_dequeue_dtmf() RTP
RECV DTMF 1:2240

 

 

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Tuesday, October 20, 2009 4:54 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] uuid_send_dtmf fails (was: conference call)

 

OK what makes you think it failed?  The fact you don't hear it?

 

/b

 

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From durk.debeer at isp.solcon.nl  Tue Oct 20 04:55:46 2009
From: durk.debeer at isp.solcon.nl (Durk de Beer)
Date: Tue, 20 Oct 2009 13:55:46 +0200
Subject: [Freeswitch-users] How to retain the B-number in an Call forward
Message-ID: <mailman.14.1269220914.3324.freeswitch-users@lists.freeswitch.org>


 Hello,


 I us a call forward on freeswitch to forward calls to my mobile phone. If now Freeswitch forwards a call, the number information presented to my mobile phone is that of the original caller, because that is what there is appearing in its display. What I want
 to see is that Freeswitch is calling me so I would know it is a business call and not an private call. Now I have no means to distinguishes between them.



 Kind
 regards,



 Durk de Beer


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From ahmedmunir007 at gmail.com  Tue Oct 20 05:33:49 2009
From: ahmedmunir007 at gmail.com (Ahmed Munir)
Date: Tue, 20 Oct 2009 17:33:49 +0500
Subject: [Freeswitch-users] mod_curl + mod_nibblebill
Message-ID: <a60fed5e0910200533h57d3d123maf8fa651801c31b4@mail.gmail.com>

Hi,

Consider below mob_nibblebill configuration,


<configuration name="nibblebill.conf" description="Nibble Billing">
  <settings>

    <param name="db_username" value="root"/>
    <param name="db_password" value="password"/>
    <param name="db_dsn" value="MySQL-freeswitch"/>

    <param name="db_table" value="accounts"/>

    <param name="db_column_cash" value="cash"/>

    <param name="db_column_account" value="id"/>

    <param name="global_heartbeat" value="60"/>

    <param name="lowbal_amt" value="5"/>
    <param name="lowbal_action" value="play ding"/>

     <param name="nobal_amt" value="0"/>
    <param name="nobal_action" value="hangup"/>

     <param name="percall_max_amt" value="100"/>
    <param name="percall_action" value="hangup"/>

  </settings>
</configuration>


When i put this configuration in
/usr/local/freeswitch/conf/autoload_configs, everything works perfect as
expected.

But when i call this configuration using mod_xml_curl, freeswitch does not
loads mod_nibblebill configuration from it. I even debuged mod_xml_curl
using console command,

xml_curl debug_on

and reload mod_nibblebill, I get exact same configuration from xml_curl but
mod_nibblebill does not repect it and try to use default parameter values as
mentioned below,


2009-10-20 16:58:23.396710 [ERR] switch_odbc.c:188 STATE: IM002 CODE 0
ERROR: [unixODBC][Driver Manager]Data source name not found, and no default
driver specified

2009-10-20 16:58:23.396738 [CRIT] mod_nibblebill.c:221 Cannot connect to
ODBC driver/database bandwidth.com (user: bandwidth.com / pass dev)!
2009-10-20 16:58:23.396800 [CONSOLE] switch_loadable_module.c:889
Successfully Loaded [mod_nibblebill]

I am also loading various other configs from xml_curl without any problem,
its only mod_nibblebill that is not honoring configuration provided by
xml_curl.

Please advise if i am doing anything wrong.


-- 
Regards,

Ahmed Munir
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From dome at tel.co.th  Tue Oct 20 06:31:17 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Tue, 20 Oct 2009 20:31:17 +0700
Subject: [Freeswitch-users] mod_odbc_query share memory problem
Message-ID: <8ccbff060910200631l44cc7dcfm2aa9511d86798aa1@mail.gmail.com>

Dear Sir,
I'm using mod_odbc_query and mod_nibble_billing for my calling card
solutoin. i found mod_odbc_query cant work with high load  (200 calls
concurrent)
i got error
[STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory;
Error while executing the query
]
in FS console

How to fix it ?

BG
Dome C.



From dwhite at olp.net  Tue Oct 20 06:34:37 2009
From: dwhite at olp.net (Dan White)
Date: Tue, 20 Oct 2009 08:34:37 -0500
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <7b8592a0910192253l1313d82j51c933cb5f55ad41@mail.gmail.com>
References: <7b8592a0910190516rafa4398q62c7a99ef703540f@mail.gmail.com>
	<5B924C57-B658-44C0-A1CE-00A3ECB5AA07@freeswitch.org>
	<7b8592a0910191351g14cbb47dj3f19362ff6197157@mail.gmail.com>
	<FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
	<7b8592a0910192253l1313d82j51c933cb5f55ad41@mail.gmail.com>
Message-ID: <20091020133437.GA4941@dan.olp.net>

On 20/10/09?07:53?+0200, ineya ineya wrote:
>IP should be OK:
>
>[root at franta /opt/freeswitch]# ifconfig eth1
>eth1????? Link encap:Ethernet? HWaddr 00:4f:4e:62:ad:83
>????????? inet addr:10.80.62.40? Bcast:10.80.62.255? Mask:255.255.255.0
>????????? inet6 addr: 2000:2::1/32 Scope:Global

This address caught my eye. Although valid, a /32 address assignment is
atypical. Is that what you intended? Does assigning a /64 address on both
ends make any difference?

-- 
Dan White



From rob4manhere at gmail.com  Tue Oct 20 06:46:43 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Tue, 20 Oct 2009 08:46:43 -0500
Subject: [Freeswitch-users] mod_odbc_query share memory problem
In-Reply-To: <8ccbff060910200631l44cc7dcfm2aa9511d86798aa1@mail.gmail.com>
References: <8ccbff060910200631l44cc7dcfm2aa9511d86798aa1@mail.gmail.com>
Message-ID: <1BFD340C-EC22-434D-B521-8E4D1572A343@gmail.com>

Hi Dome,

Have you tried increasing global_heartbeat to reduce the frequency of  
odbc calls?  What is it currently set to?

Rob

On Oct 20, 2009, at 8:31 AM, Dome Charoenyost wrote:

> Dear Sir,
> I'm using mod_odbc_query and mod_nibble_billing for my calling card
> solutoin. i found mod_odbc_query cant work with high load  (200 calls
> concurrent)
> i got error
> [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory;
> Error while executing the query
> ]
> in FS console
>
> How to fix it ?
>
> BG
> Dome C.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From rob4manhere at gmail.com  Tue Oct 20 07:00:05 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Tue, 20 Oct 2009 09:00:05 -0500
Subject: [Freeswitch-users] How to retain the B-number in an Call forward
In-Reply-To: <4adda65c.4944f10a.7a9b.ffff9408SMTPIN_ADDED@mx.google.com>
References: <4adda65c.4944f10a.7a9b.ffff9408SMTPIN_ADDED@mx.google.com>
Message-ID: <7394C8E4-C8D8-42C2-8ECD-B95119B604E5@gmail.com>

Have you tried setting the effective_caller_id_number before  
bridging?  Such as:

<action application="set" data="effective_caller_id_number=9185551212"/>
Cheers,
Rob

On Oct 20, 2009, at 6:55 AM, Durk de Beer wrote:

> Hello,
> I us a call forward on freeswitch to forward calls to my mobile  
> phone. If now Freeswitch forwards a call, the number information  
> presented to my mobile phone is that of the original caller, because  
> that is what there is appearing in its display. What I want to see  
> is that Freeswitch is calling me so I would know it is a business  
> call and not an private call. Now I have no means to distinguishes  
> between them.
> Kind regards,
> Durk de Beer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From dome at tel.co.th  Tue Oct 20 07:00:22 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Tue, 20 Oct 2009 21:00:22 +0700
Subject: [Freeswitch-users] mod_odbc_query share memory problem
In-Reply-To: <1BFD340C-EC22-434D-B521-8E4D1572A343@gmail.com>
References: <8ccbff060910200631l44cc7dcfm2aa9511d86798aa1@mail.gmail.com>
	<1BFD340C-EC22-434D-B521-8E4D1572A343@gmail.com>
Message-ID: <8ccbff060910200700o5213a8c2mf95acd9603f96ea@mail.gmail.com>

2009/10/20 Rob Forman <rob4manhere at gmail.com>:
> Hi Dome,
>
> Have you tried increasing global_heartbeat to reduce the frequency of
> odbc calls? ?What is it currently set to?

Now 1 min

>
> Rob
>
> On Oct 20, 2009, at 8:31 AM, Dome Charoenyost wrote:
>
>> Dear Sir,
>> I'm using mod_odbc_query and mod_nibble_billing for my calling card
>> solutoin. i found mod_odbc_query cant work with high load ?(200 calls
>> concurrent)
>> i got error
>> [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory;
>> Error while executing the query
>> ]
>> in FS console
>>
>> How to fix it ?
>>
>> BG
>> Dome C.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From leon at scarlet-internet.nl  Tue Oct 20 07:10:30 2009
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Tue, 20 Oct 2009 16:10:30 +0200
Subject: [Freeswitch-users] mod_odbc_query share memory problem
In-Reply-To: <8ccbff060910200631l44cc7dcfm2aa9511d86798aa1@mail.gmail.com>
References: <8ccbff060910200631l44cc7dcfm2aa9511d86798aa1@mail.gmail.com>
Message-ID: <180CE701-D840-47E6-821C-BB79C9F1945A@scarlet-internet.nl>

Hi,

I haven't run into that problem yet, but did you try increasing the  
maximum shared memory in /proc/sys/kernel/shmmax (sysctl  
kernel.shmmax) ?

regards,

Leon



On Oct 20, 2009, at 3:31 PM, Dome Charoenyost wrote:

> Dear Sir,
> I'm using mod_odbc_query and mod_nibble_billing for my calling card
> solutoin. i found mod_odbc_query cant work with high load  (200 calls
> concurrent)
> i got error
> [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory;
> Error while executing the query
> ]
> in FS console
>
> How to fix it ?
>
> BG
> Dome C.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From rob4manhere at gmail.com  Tue Oct 20 07:11:11 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Tue, 20 Oct 2009 09:11:11 -0500
Subject: [Freeswitch-users] mod_odbc_query share memory problem
In-Reply-To: <8ccbff060910200700o5213a8c2mf95acd9603f96ea@mail.gmail.com>
References: <8ccbff060910200631l44cc7dcfm2aa9511d86798aa1@mail.gmail.com>
	<1BFD340C-EC22-434D-B521-8E4D1572A343@gmail.com>
	<8ccbff060910200700o5213a8c2mf95acd9603f96ea@mail.gmail.com>
Message-ID: <5D397B2B-335C-4460-AAA2-60230F6B8C6A@gmail.com>

Try 300 seconds (5 minutes) and see if it improves.


On Oct 20, 2009, at 9:00 AM, Dome Charoenyost wrote:

> 2009/10/20 Rob Forman <rob4manhere at gmail.com>:
>> Hi Dome,
>>
>> Have you tried increasing global_heartbeat to reduce the frequency of
>> odbc calls?  What is it currently set to?
>
> Now 1 min
>
>>
>> Rob
>>
>> On Oct 20, 2009, at 8:31 AM, Dome Charoenyost wrote:
>>
>>> Dear Sir,
>>> I'm using mod_odbc_query and mod_nibble_billing for my calling card
>>> solutoin. i found mod_odbc_query cant work with high load  (200  
>>> calls
>>> concurrent)
>>> i got error
>>> [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory;
>>> Error while executing the query
>>> ]
>>> in FS console
>>>
>>> How to fix it ?
>>>
>>> BG
>>> Dome C.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From ineyaa at gmail.com  Tue Oct 20 07:16:46 2009
From: ineyaa at gmail.com (ineya ineya)
Date: Tue, 20 Oct 2009 16:16:46 +0200
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <20091020133437.GA4941@dan.olp.net>
References: <7b8592a0910190516rafa4398q62c7a99ef703540f@mail.gmail.com>
	<5B924C57-B658-44C0-A1CE-00A3ECB5AA07@freeswitch.org>
	<7b8592a0910191351g14cbb47dj3f19362ff6197157@mail.gmail.com>
	<FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
	<7b8592a0910192253l1313d82j51c933cb5f55ad41@mail.gmail.com>
	<20091020133437.GA4941@dan.olp.net>
Message-ID: <7b8592a0910200716q4a7db93cyb955595e4993ac96@mail.gmail.com>

No, it's the same thing with /64 on both ends.
I tried to built SVN version and modified just IP addresses in
sip_profiles, but still can't call from one phone to another. It goes
straght to voicemail.

Is there a softphone, which you know works with freeswitch on IPv6?
Maybe the error is in phones I'm currently using.

On Tue, Oct 20, 2009 at 3:34 PM, Dan White <dwhite at olp.net> wrote:

> This address caught my eye. Although valid, a /32 address assignment is
> atypical. Is that what you intended? Does assigning a /64 address on both
> ends make any difference?
>
> --
> Dan White
>



From brian at freeswitch.org  Tue Oct 20 07:21:24 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 20 Oct 2009 09:21:24 -0500
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <7b8592a0910200716q4a7db93cyb955595e4993ac96@mail.gmail.com>
References: <7b8592a0910190516rafa4398q62c7a99ef703540f@mail.gmail.com>
	<5B924C57-B658-44C0-A1CE-00A3ECB5AA07@freeswitch.org>
	<7b8592a0910191351g14cbb47dj3f19362ff6197157@mail.gmail.com>
	<FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
	<7b8592a0910192253l1313d82j51c933cb5f55ad41@mail.gmail.com>
	<20091020133437.GA4941@dan.olp.net>
	<7b8592a0910200716q4a7db93cyb955595e4993ac96@mail.gmail.com>
Message-ID: <858EC056-1D84-4F30-A0B7-9A70FF98695D@freeswitch.org>

Well how are you trying to dial users?

/b

On Oct 20, 2009, at 9:16 AM, ineya ineya wrote:

> No, it's the same thing with /64 on both ends.
> I tried to built SVN version and modified just IP addresses in
> sip_profiles, but still can't call from one phone to another. It goes
> straght to voicemail.
>
> Is there a softphone, which you know works with freeswitch on IPv6?
> Maybe the error is in phones I'm currently using.




From m.sobkow at marketelsystems.com  Tue Oct 20 07:27:36 2009
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Tue, 20 Oct 2009 08:27:36 -0600
Subject: [Freeswitch-users] Is anyone running Ubuntu 8.04/Hardy?
In-Reply-To: <903da5680910191917m79bf5ecfvdc8001952dbcc339@mail.gmail.com>
References: <4ADCA34E.5000700@marketelsystems.com>
	<903da5680910191917m79bf5ecfvdc8001952dbcc339@mail.gmail.com>
Message-ID: <4ADDC8D8.7090401@marketelsystems.com>

Gabriel Gunderson wrote:
> On Mon, Oct 19, 2009 at 11:35 AM, Mark Sobkow
> <m.sobkow at marketelsystems.com> wrote:
>   
>> Everyone I've emailed with on the dev list is running the current
>> release of Ubuntu, not 8.04/Hardy.
>>     
>
> Well, what issues?
>
> Gabe
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   
We had problems with loading mod_sofia.  However, we were running an 
older release, then we'd tried upgrading to the source build from 
launchpad.net.

Yesterday we downloaded the current svn.freeswitch.org, and that 
particular problem has gone away.  Now we need to figure out why our 
Erlang component can synchronize from Windows, but not from Linux.

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 




From dome at tel.co.th  Tue Oct 20 07:30:38 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Tue, 20 Oct 2009 21:30:38 +0700
Subject: [Freeswitch-users] mod_odbc_query share memory problem
In-Reply-To: <5D397B2B-335C-4460-AAA2-60230F6B8C6A@gmail.com>
References: <8ccbff060910200631l44cc7dcfm2aa9511d86798aa1@mail.gmail.com>
	<1BFD340C-EC22-434D-B521-8E4D1572A343@gmail.com>
	<8ccbff060910200700o5213a8c2mf95acd9603f96ea@mail.gmail.com>
	<5D397B2B-335C-4460-AAA2-60230F6B8C6A@gmail.com>
Message-ID: <8ccbff060910200730w6a46ffc7u1ade7e5413b76cf6@mail.gmail.com>

2009/10/20 Rob Forman <rob4manhere at gmail.com>:
> Try 300 seconds (5 minutes) and see if it improves.
Ok. i'll try

>
>
> On Oct 20, 2009, at 9:00 AM, Dome Charoenyost wrote:
>
>> 2009/10/20 Rob Forman <rob4manhere at gmail.com>:
>>> Hi Dome,
>>>
>>> Have you tried increasing global_heartbeat to reduce the frequency of
>>> odbc calls? ?What is it currently set to?
>>
>> Now 1 min
>>
>>>
>>> Rob
>>>
>>> On Oct 20, 2009, at 8:31 AM, Dome Charoenyost wrote:
>>>
>>>> Dear Sir,
>>>> I'm using mod_odbc_query and mod_nibble_billing for my calling card
>>>> solutoin. i found mod_odbc_query cant work with high load ?(200
>>>> calls
>>>> concurrent)
>>>> i got error
>>>> [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory;
>>>> Error while executing the query
>>>> ]
>>>> in FS console
>>>>
>>>> How to fix it ?
>>>>
>>>> BG
>>>> Dome C.
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From dome at tel.co.th  Tue Oct 20 07:35:15 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Tue, 20 Oct 2009 21:35:15 +0700
Subject: [Freeswitch-users] mod_odbc_query share memory problem
In-Reply-To: <180CE701-D840-47E6-821C-BB79C9F1945A@scarlet-internet.nl>
References: <8ccbff060910200631l44cc7dcfm2aa9511d86798aa1@mail.gmail.com>
	<180CE701-D840-47E6-821C-BB79C9F1945A@scarlet-internet.nl>
Message-ID: <8ccbff060910200735h73e9459ex7460fcbeac457d59@mail.gmail.com>

2009/10/20 Leon de Rooij <leon at scarlet-internet.nl>:
> Hi,
>
> I haven't run into that problem yet, but did you try increasing the
> maximum shared memory in /proc/sys/kernel/shmmax (sysctl
> kernel.shmmax) ?
Recommend me please i have 4GB RAM.
is posible to increase to 50% RAM ?

Dome C.
>
> regards,
>
> Leon
>
>
>
> On Oct 20, 2009, at 3:31 PM, Dome Charoenyost wrote:
>
>> Dear Sir,
>> I'm using mod_odbc_query and mod_nibble_billing for my calling card
>> solutoin. i found mod_odbc_query cant work with high load ?(200 calls
>> concurrent)
>> i got error
>> [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory;
>> Error while executing the query
>> ]
>> in FS console
>>
>> How to fix it ?
>>
>> BG
>> Dome C.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From ngay01042005 at gmail.com  Tue Oct 20 07:53:01 2009
From: ngay01042005 at gmail.com (homqua)
Date: Tue, 20 Oct 2009 07:53:01 -0700 (PDT)
Subject: [Freeswitch-users] Connect PHP SOAP Web Server with SQLite database
 of FS
Message-ID: <25976554.post@talk.nabble.com>


Now I am building a PHP SOAP Web Service to access the database of FS. Anyone
has idea about how to access sqlite database of FS through PHP ? I have read
about socket event in FS, but I don't know whether it can response with the
query of database or not.
Thanks for your help.
-- 
View this message in context: http://www.nabble.com/Connect-PHP-SOAP-Web-Server-with-SQLite-database-of-FS-tp25976554p25976554.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From bottleman at icf.org.ru  Tue Oct 20 08:00:57 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Tue, 20 Oct 2009 19:00:57 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<65d96fc80910130014hc5f5fcet677f87b7fb57559e@mail.gmail.com>
	<65d96fc80910132359i6d53d8ban10f943795481bf7c@mail.gmail.com>
	<Pine.LNX.4.64.0910141134270.11620@srv.icf.org.ru>
	<65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>
	<Pine.LNX.4.64.0910141202500.21535@srv.icf.org.ru>
	<65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>
	<65d96fc80910140215y357fede1l37f6cba9725ddd89@mail.gmail.com>
	<Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>
	<65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910201841030.12352@srv.icf.org.ru>

On 2009-10-20 10:17 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>> TC>
TC>> TC>call flow is SIP_user => FS => H323_endpoint is failing ..
TC>> coredumped....
TC>> TC>http://pastebin.freeswitch.org/10703
TC>>
TC>> i fix some bugs now,
TC>> ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/mod_h323.tar.bz2 this
TC>> is
TC>> updated version, try it, if you experience no audio try enable rtp proxy in
TC>> you sip profile.
TC>>
TC>>
TC>Hi,  there are several issues... lets start with top 4 :)
TC>
TC>
TC>
TC>1. I'm still stuck with no audio:
TC>
TC>I have this parameter in the sip profile set: <param
TC>name="inbound-proxy-media" value="true"/>
TC>...tried with both with slow start and fast start... any idea ?
TC>
TC>pls check: http://pastebin.freeswitch.org/10771

try to sniff rtp traffic on fs, is it go thouch him.

TC>
TC>
TC>
TC>
TC>
TC>2. outgoing calls still failing in coredumps: what is your dialplan ? ...
TC>how do you call bridge application?

simple:


<action application="bridge" data="h323/${number}"/>

if fs not registered on gk then data="h323/${number}@xxx.xxx.xxx.xxx".
       

TC>
TC>2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3252 Found capability:
TC>UserInput/PointDevice <14>
TC>2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3248 FindCapability: 15
TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external thread
TC>0xb6eb60a0 for id 3048876944
TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:65 Destroyed external thread
TC>0xb6ebafa8 for id 3048876944
TC>2009-10-20 10:08:18.426608 [DEBUG] tlibthrd.cxx:406 Destroyed thread
TC>0xb6ebafa8 PExternalThread:0xb5ba2b90(id = b5ba2b90)
TC>2009-10-20 10:08:18.426608 [DEBUG] h323caps.cxx:3252 Found capability:
TC>UserInput/Modal <15>
TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0,
TC>expiries=0
TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0,
TC>expiries=0
TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external thread
TC>0xb6eba910 for id 3048876944
TC>2009-10-20 10:08:18.426608 [DEBUG] h4601.cxx:1725 Endpoint Attached
TC>Segmentation fault (core dumped)
TC>tculjaga at subZero:~/freeswitch-trunk$
TC>
TC>pls check: http://pastebin.freeswitch.org/10769

look strange, what version of libpt/h323plus you use and freeswitch itself ?

TC>
TC>
TC>
TC>3.  when you hangup from SIP side, the call is not released end-to-end (the
TC>H323 endpoint doesn't get any releaseComplete message)
TC>
TC>2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=2,
TC>expiries=3
TC>2009-10-20 10:10:51.264527 [DEBUG] h323neg.cxx:432 Received
TC>MasterSlaveDeterminationAck: state=Incoming
TC>2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=1,
TC>expiries=3
TC>2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=2,
TC>expiries=4
TC>2009-10-20 10:10:51.264527 [DEBUG] h323.cxx:4138
TC>InternalEstablishedConnectionCheck: connectionState=EstablishedConnection
TC>fastStartState=FastStartAcknowledged
TC>2009-10-20 10:10:51.264527 [DEBUG] h323caps.cxx:3264 FindCapability: "T.120"
TC>2009-10-20 10:10:51.264527 [DEBUG] h323.cxx:4138
TC>InternalEstablishedConnectionCheck: connectionState=EstablishedConnection
TC>fastStartState=FastStartAcknowledged
TC>2009-10-20 10:10:51.264527 [DEBUG] tlibthrd.cxx:1023
TC>PThread::PXBlockOnIO(45,0)
TC>2009-10-20 10:10:54.405479 [NOTICE] sofia.c:328 Hangup sofia/internal/
TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [CS_CONSUME_MEDIA]
TC>[NORMAL_CLEARING]
TC>2009-10-20 10:10:54.405479 [DEBUG] switch_channel.c:1726 Send signal
TC>sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [KILL]
TC>2009-10-20 10:10:54.405479 [DEBUG] switch_core_session.c:932 Send signal
TC>sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [BREAK]
TC>2009-10-20 10:10:54.405479 [DEBUG] switch_core_state_machine.c:437 thread
TC>mismatch skipping state handler.
TC>2009-10-20 10:10:54.405479 [DEBUG] switch_core_state_machine.c:306
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) Running State
TC>Change CS_HANGUP
TC>2009-10-20 10:10:54.406530 [DEBUG] switch_core_state_machine.c:464
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State HANGUP
TC>2009-10-20 10:10:54.406530 [DEBUG] mod_sofia.c:338 Channel sofia/internal/
TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> hanging up, cause:
TC>NORMAL_CLEARING
TC>2009-10-20 10:10:54.406530 [DEBUG] switch_core_state_machine.c:46
TC>sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> Standard HANGUP,
TC>cause: NORMAL_CLEARING
TC>
TC>
TC>pls check: http://pastebin.freeswitch.org/10771

hm, according this log it send release complete:
2009-10-20 10:11:03.995707 [DEBUG] h323pdu.cxx:609 Sending PDU [ip$10.4.62.7:1720/ip$10.1.14.153:3341]
i think please try to dump full traffic of this call from both endpoints and send me.

TC>
TC>
TC>
TC>
TC>4. sometimes when i shutdown FS i get coredimps - from my experience it
TC>looks like you don't wait for a FS thread to finish when you exit...
TC>
TC>2009-10-20 10:05:59.493306 [CONSOLE] switch_event.c:508 Stopping queue
TC>thread 2
TC>2009-10-20 10:05:59.493339 [CONSOLE] switch_core.c:1693 Finalizing Shutdown.
TC>2009-10-20 10:05:59.493379 [CONSOLE] switch_log.c:310 Logger Ended.
TC>2009-10-20 10:05:59.494472 [CONSOLE] switch_core_memory.c:567 Stopping
TC>memory pool queue.
TC>Segmentation fault (core dumped)
TC>tculjaga at subZero:~/freeswitch-trunk$
TC>tculjaga at subZero:~/freeswitch-trunk$

may be, i don't look on this so close at this time.

TC>Please advice your FS/mod_h323.conf.xml settings...

<configuration name="h323.conf" description="H323 Endpoints">
<settings>
<param name="trace-level" value="4"/>
<param name="context" value="default"/>
<param name="dialplan" value="XML"/>
<param name="codec-prefs" value="PCMA,PCMU,GSM,G729,G726,LPC"/>
<param name="gk-address" value="91.195.204.1"/>    <!-- empty to disable, "*" to search LAN -->
<param name="gk-identifer" value="ICFGK"/>  <!-- optional name of gk -->
<param name="gk-interface" value=""/>  <!-- optional listener interface name -->
</settings>
<listeners>
<listener name="default">
<param name="h323-ip" value="$${local_ip_v4}"/>
<param name="h323-port" value="1720"/>
</listener>
</listeners>
</configuration>

this my current settings.
							

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From bottleman at icf.org.ru  Tue Oct 20 08:01:05 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Tue, 20 Oct 2009 19:01:05 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<65d96fc80910130014hc5f5fcet677f87b7fb57559e@mail.gmail.com>
	<65d96fc80910132359i6d53d8ban10f943795481bf7c@mail.gmail.com>
	<Pine.LNX.4.64.0910141134270.11620@srv.icf.org.ru>
	<65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>
	<Pine.LNX.4.64.0910141202500.21535@srv.icf.org.ru>
	<65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>
	<65d96fc80910140215y357fede1l37f6cba9725ddd89@mail.gmail.com>
	<Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>
	<65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910201900570.12352@srv.icf.org.ru>

On 2009-10-20 10:17 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>> TC>
TC>> TC>call flow is SIP_user => FS => H323_endpoint is failing ..
TC>> coredumped....
TC>> TC>http://pastebin.freeswitch.org/10703
TC>>
TC>> i fix some bugs now,
TC>> ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/mod_h323.tar.bz2 this
TC>> is
TC>> updated version, try it, if you experience no audio try enable rtp proxy in
TC>> you sip profile.
TC>>
TC>>
TC>Hi,  there are several issues... lets start with top 4 :)
TC>
TC>
TC>
TC>1. I'm still stuck with no audio:
TC>
TC>I have this parameter in the sip profile set: <param
TC>name="inbound-proxy-media" value="true"/>
TC>...tried with both with slow start and fast start... any idea ?
TC>
TC>pls check: http://pastebin.freeswitch.org/10771

try to sniff rtp traffic on fs, is it go thouch him.

TC>
TC>
TC>
TC>
TC>
TC>2. outgoing calls still failing in coredumps: what is your dialplan ? ...
TC>how do you call bridge application?

simple:


<action application="bridge" data="h323/${number}"/>

if fs not registered on gk then data="h323/${number}@xxx.xxx.xxx.xxx".
       

TC>
TC>2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3252 Found capability:
TC>UserInput/PointDevice <14>
TC>2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3248 FindCapability: 15
TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external thread
TC>0xb6eb60a0 for id 3048876944
TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:65 Destroyed external thread
TC>0xb6ebafa8 for id 3048876944
TC>2009-10-20 10:08:18.426608 [DEBUG] tlibthrd.cxx:406 Destroyed thread
TC>0xb6ebafa8 PExternalThread:0xb5ba2b90(id = b5ba2b90)
TC>2009-10-20 10:08:18.426608 [DEBUG] h323caps.cxx:3252 Found capability:
TC>UserInput/Modal <15>
TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0,
TC>expiries=0
TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0,
TC>expiries=0
TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external thread
TC>0xb6eba910 for id 3048876944
TC>2009-10-20 10:08:18.426608 [DEBUG] h4601.cxx:1725 Endpoint Attached
TC>Segmentation fault (core dumped)
TC>tculjaga at subZero:~/freeswitch-trunk$
TC>
TC>pls check: http://pastebin.freeswitch.org/10769

look strange, what version of libpt/h323plus you use and freeswitch itself ?

TC>
TC>
TC>
TC>3.  when you hangup from SIP side, the call is not released end-to-end (the
TC>H323 endpoint doesn't get any releaseComplete message)
TC>
TC>2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=2,
TC>expiries=3
TC>2009-10-20 10:10:51.264527 [DEBUG] h323neg.cxx:432 Received
TC>MasterSlaveDeterminationAck: state=Incoming
TC>2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=1,
TC>expiries=3
TC>2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=2,
TC>expiries=4
TC>2009-10-20 10:10:51.264527 [DEBUG] h323.cxx:4138
TC>InternalEstablishedConnectionCheck: connectionState=EstablishedConnection
TC>fastStartState=FastStartAcknowledged
TC>2009-10-20 10:10:51.264527 [DEBUG] h323caps.cxx:3264 FindCapability: "T.120"
TC>2009-10-20 10:10:51.264527 [DEBUG] h323.cxx:4138
TC>InternalEstablishedConnectionCheck: connectionState=EstablishedConnection
TC>fastStartState=FastStartAcknowledged
TC>2009-10-20 10:10:51.264527 [DEBUG] tlibthrd.cxx:1023
TC>PThread::PXBlockOnIO(45,0)
TC>2009-10-20 10:10:54.405479 [NOTICE] sofia.c:328 Hangup sofia/internal/
TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [CS_CONSUME_MEDIA]
TC>[NORMAL_CLEARING]
TC>2009-10-20 10:10:54.405479 [DEBUG] switch_channel.c:1726 Send signal
TC>sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [KILL]
TC>2009-10-20 10:10:54.405479 [DEBUG] switch_core_session.c:932 Send signal
TC>sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [BREAK]
TC>2009-10-20 10:10:54.405479 [DEBUG] switch_core_state_machine.c:437 thread
TC>mismatch skipping state handler.
TC>2009-10-20 10:10:54.405479 [DEBUG] switch_core_state_machine.c:306
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) Running State
TC>Change CS_HANGUP
TC>2009-10-20 10:10:54.406530 [DEBUG] switch_core_state_machine.c:464
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State HANGUP
TC>2009-10-20 10:10:54.406530 [DEBUG] mod_sofia.c:338 Channel sofia/internal/
TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> hanging up, cause:
TC>NORMAL_CLEARING
TC>2009-10-20 10:10:54.406530 [DEBUG] switch_core_state_machine.c:46
TC>sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> Standard HANGUP,
TC>cause: NORMAL_CLEARING
TC>
TC>
TC>pls check: http://pastebin.freeswitch.org/10771

hm, according this log it send release complete:
2009-10-20 10:11:03.995707 [DEBUG] h323pdu.cxx:609 Sending PDU [ip$10.4.62.7:1720/ip$10.1.14.153:3341]
i think please try to dump full traffic of this call from both endpoints and send me.

TC>
TC>
TC>
TC>
TC>4. sometimes when i shutdown FS i get coredimps - from my experience it
TC>looks like you don't wait for a FS thread to finish when you exit...
TC>
TC>2009-10-20 10:05:59.493306 [CONSOLE] switch_event.c:508 Stopping queue
TC>thread 2
TC>2009-10-20 10:05:59.493339 [CONSOLE] switch_core.c:1693 Finalizing Shutdown.
TC>2009-10-20 10:05:59.493379 [CONSOLE] switch_log.c:310 Logger Ended.
TC>2009-10-20 10:05:59.494472 [CONSOLE] switch_core_memory.c:567 Stopping
TC>memory pool queue.
TC>Segmentation fault (core dumped)
TC>tculjaga at subZero:~/freeswitch-trunk$
TC>tculjaga at subZero:~/freeswitch-trunk$

may be, i don't look on this so close at this time.

TC>Please advice your FS/mod_h323.conf.xml settings...

<configuration name="h323.conf" description="H323 Endpoints">
<settings>
<param name="trace-level" value="4"/>
<param name="context" value="default"/>
<param name="dialplan" value="XML"/>
<param name="codec-prefs" value="PCMA,PCMU,GSM,G729,G726,LPC"/>
<param name="gk-address" value="91.195.204.1"/>    <!-- empty to disable, "*" to search LAN -->
<param name="gk-identifer" value="ICFGK"/>  <!-- optional name of gk -->
<param name="gk-interface" value=""/>  <!-- optional listener interface name -->
</settings>
<listeners>
<listener name="default">
<param name="h323-ip" value="$${local_ip_v4}"/>
<param name="h323-port" value="1720"/>
</listener>
</listeners>
</configuration>

this my current settings.
							

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From rob4manhere at gmail.com  Tue Oct 20 08:03:36 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Tue, 20 Oct 2009 10:03:36 -0500
Subject: [Freeswitch-users] mod_odbc_query share memory problem
In-Reply-To: <8ccbff060910200735h73e9459ex7460fcbeac457d59@mail.gmail.com>
References: <8ccbff060910200631l44cc7dcfm2aa9511d86798aa1@mail.gmail.com>
	<180CE701-D840-47E6-821C-BB79C9F1945A@scarlet-internet.nl>
	<8ccbff060910200735h73e9459ex7460fcbeac457d59@mail.gmail.com>
Message-ID: <85AF1814-9B1A-439C-83F5-D46F9BFEC140@gmail.com>

You could, but I would try just doubling whatever it is to see if  
thats improves the issue first.  The default is 32MB.  You could  
double it to 64MB and test again.

What is it currently set to (run: sysctl kernel.shmmax)?  You can  
change it on the fly with sysctl.  Once you're done testing and want  
it to persist across reboots, you'll want to set in /etc/sysctl.conf .

Cheers,
Rob

On Oct 20, 2009, at 9:35 AM, Dome Charoenyost wrote:

> 2009/10/20 Leon de Rooij <leon at scarlet-internet.nl>:
>> Hi,
>>
>> I haven't run into that problem yet, but did you try increasing the
>> maximum shared memory in /proc/sys/kernel/shmmax (sysctl
>> kernel.shmmax) ?
> Recommend me please i have 4GB RAM.
> is posible to increase to 50% RAM ?
>
> Dome C.
>>
>> regards,
>>
>> Leon
>>
>>
>>
>> On Oct 20, 2009, at 3:31 PM, Dome Charoenyost wrote:
>>
>>> Dear Sir,
>>> I'm using mod_odbc_query and mod_nibble_billing for my calling card
>>> solutoin. i found mod_odbc_query cant work with high load  (200  
>>> calls
>>> concurrent)
>>> i got error
>>> [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory;
>>> Error while executing the query
>>> ]
>>> in FS console
>>>
>>> How to fix it ?
>>>
>>> BG
>>> Dome C.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From dome at tel.co.th  Tue Oct 20 08:14:59 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Tue, 20 Oct 2009 22:14:59 +0700
Subject: [Freeswitch-users] mod_odbc_query share memory problem
In-Reply-To: <85AF1814-9B1A-439C-83F5-D46F9BFEC140@gmail.com>
References: <8ccbff060910200631l44cc7dcfm2aa9511d86798aa1@mail.gmail.com>
	<180CE701-D840-47E6-821C-BB79C9F1945A@scarlet-internet.nl>
	<8ccbff060910200735h73e9459ex7460fcbeac457d59@mail.gmail.com>
	<85AF1814-9B1A-439C-83F5-D46F9BFEC140@gmail.com>
Message-ID: <8ccbff060910200814w2fa37866m8e211a73610f03c4@mail.gmail.com>

2009/10/20 Rob Forman <rob4manhere at gmail.com>:
> You could, but I would try just doubling whatever it is to see if
> thats improves the issue first. ?The default is 32MB. ?You could
> double it to 64MB and test again.
Ok' let's me try 64MB first
i found some information
https://www.ishoof.org/project/wiki/index.php/ORACLE_11_G_-_Gentoo_Installation

>
> What is it currently set to (run: sysctl kernel.shmmax)? ?You can
> change it on the fly with sysctl. ?Once you're done testing and want
> it to persist across reboots, you'll want to set in /etc/sysctl.conf .
>
> Cheers,
> Rob
>
> On Oct 20, 2009, at 9:35 AM, Dome Charoenyost wrote:
>
>> 2009/10/20 Leon de Rooij <leon at scarlet-internet.nl>:
>>> Hi,
>>>
>>> I haven't run into that problem yet, but did you try increasing the
>>> maximum shared memory in /proc/sys/kernel/shmmax (sysctl
>>> kernel.shmmax) ?
>> Recommend me please i have 4GB RAM.
>> is posible to increase to 50% RAM ?
>>
>> Dome C.
>>>
>>> regards,
>>>
>>> Leon
>>>
>>>
>>>
>>> On Oct 20, 2009, at 3:31 PM, Dome Charoenyost wrote:
>>>
>>>> Dear Sir,
>>>> I'm using mod_odbc_query and mod_nibble_billing for my calling card
>>>> solutoin. i found mod_odbc_query cant work with high load ?(200
>>>> calls
>>>> concurrent)
>>>> i got error
>>>> [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory;
>>>> Error while executing the query
>>>> ]
>>>> in FS console
>>>>
>>>> How to fix it ?
>>>>
>>>> BG
>>>> Dome C.
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From bottleman at icf.org.ru  Tue Oct 20 08:17:37 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Tue, 20 Oct 2009 19:17:37 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910121443g996612dha90669349457b9e8@mail.gmail.com>
	<65d96fc80910130014hc5f5fcet677f87b7fb57559e@mail.gmail.com>
	<65d96fc80910132359i6d53d8ban10f943795481bf7c@mail.gmail.com>
	<Pine.LNX.4.64.0910141134270.11620@srv.icf.org.ru>
	<65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>
	<Pine.LNX.4.64.0910141202500.21535@srv.icf.org.ru>
	<65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>
	<65d96fc80910140215y357fede1l37f6cba9725ddd89@mail.gmail.com>
	<Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>
	<65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910201901050.12352@srv.icf.org.ru>

On 2009-10-20 10:17 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>> TC>
TC>> TC>call flow is SIP_user => FS => H323_endpoint is failing ..
TC>> coredumped....
TC>> TC>http://pastebin.freeswitch.org/10703
TC>>
TC>> i fix some bugs now,
TC>> ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/mod_h323.tar.bz2 this
TC>> is
TC>> updated version, try it, if you experience no audio try enable rtp proxy in
TC>> you sip profile.
TC>>
TC>>
TC>Hi,  there are several issues... lets start with top 4 :)
TC>
TC>
TC>
TC>1. I'm still stuck with no audio:
TC>
TC>I have this parameter in the sip profile set: <param
TC>name="inbound-proxy-media" value="true"/>
TC>...tried with both with slow start and fast start... any idea ?
TC>
TC>pls check: http://pastebin.freeswitch.org/10771

try to sniff rtp traffic on fs, is it go thouch him.

TC>
TC>
TC>
TC>
TC>
TC>2. outgoing calls still failing in coredumps: what is your dialplan ? ...
TC>how do you call bridge application?

simple:


<action application="bridge" data="h323/${number}"/>

if fs not registered on gk then data="h323/${number}@xxx.xxx.xxx.xxx".
       

TC>
TC>2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3252 Found capability:
TC>UserInput/PointDevice <14>
TC>2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3248 FindCapability: 15
TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external thread
TC>0xb6eb60a0 for id 3048876944
TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:65 Destroyed external thread
TC>0xb6ebafa8 for id 3048876944
TC>2009-10-20 10:08:18.426608 [DEBUG] tlibthrd.cxx:406 Destroyed thread
TC>0xb6ebafa8 PExternalThread:0xb5ba2b90(id = b5ba2b90)
TC>2009-10-20 10:08:18.426608 [DEBUG] h323caps.cxx:3252 Found capability:
TC>UserInput/Modal <15>
TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0,
TC>expiries=0
TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0,
TC>expiries=0
TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external thread
TC>0xb6eba910 for id 3048876944
TC>2009-10-20 10:08:18.426608 [DEBUG] h4601.cxx:1725 Endpoint Attached
TC>Segmentation fault (core dumped)
TC>tculjaga at subZero:~/freeswitch-trunk$
TC>
TC>pls check: http://pastebin.freeswitch.org/10769

look strange, what version of libpt/h323plus you use and freeswitch itself ?

TC>
TC>
TC>
TC>3.  when you hangup from SIP side, the call is not released end-to-end (the
TC>H323 endpoint doesn't get any releaseComplete message)
TC>
TC>2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=2,
TC>expiries=3
TC>2009-10-20 10:10:51.264527 [DEBUG] h323neg.cxx:432 Received
TC>MasterSlaveDeterminationAck: state=Incoming
TC>2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=1,
TC>expiries=3
TC>2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=2,
TC>expiries=4
TC>2009-10-20 10:10:51.264527 [DEBUG] h323.cxx:4138
TC>InternalEstablishedConnectionCheck: connectionState=EstablishedConnection
TC>fastStartState=FastStartAcknowledged
TC>2009-10-20 10:10:51.264527 [DEBUG] h323caps.cxx:3264 FindCapability: "T.120"
TC>2009-10-20 10:10:51.264527 [DEBUG] h323.cxx:4138
TC>InternalEstablishedConnectionCheck: connectionState=EstablishedConnection
TC>fastStartState=FastStartAcknowledged
TC>2009-10-20 10:10:51.264527 [DEBUG] tlibthrd.cxx:1023
TC>PThread::PXBlockOnIO(45,0)
TC>2009-10-20 10:10:54.405479 [NOTICE] sofia.c:328 Hangup sofia/internal/
TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [CS_CONSUME_MEDIA]
TC>[NORMAL_CLEARING]
TC>2009-10-20 10:10:54.405479 [DEBUG] switch_channel.c:1726 Send signal
TC>sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [KILL]
TC>2009-10-20 10:10:54.405479 [DEBUG] switch_core_session.c:932 Send signal
TC>sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [BREAK]
TC>2009-10-20 10:10:54.405479 [DEBUG] switch_core_state_machine.c:437 thread
TC>mismatch skipping state handler.
TC>2009-10-20 10:10:54.405479 [DEBUG] switch_core_state_machine.c:306
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) Running State
TC>Change CS_HANGUP
TC>2009-10-20 10:10:54.406530 [DEBUG] switch_core_state_machine.c:464
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State HANGUP
TC>2009-10-20 10:10:54.406530 [DEBUG] mod_sofia.c:338 Channel sofia/internal/
TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> hanging up, cause:
TC>NORMAL_CLEARING
TC>2009-10-20 10:10:54.406530 [DEBUG] switch_core_state_machine.c:46
TC>sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> Standard HANGUP,
TC>cause: NORMAL_CLEARING
TC>
TC>
TC>pls check: http://pastebin.freeswitch.org/10771

hm, according this log it send release complete:
2009-10-20 10:11:03.995707 [DEBUG] h323pdu.cxx:609 Sending PDU [ip$10.4.62.7:1720/ip$10.1.14.153:3341]
i think please try to dump full traffic of this call from both endpoints and send me.

TC>
TC>
TC>
TC>
TC>4. sometimes when i shutdown FS i get coredimps - from my experience it
TC>looks like you don't wait for a FS thread to finish when you exit...
TC>
TC>2009-10-20 10:05:59.493306 [CONSOLE] switch_event.c:508 Stopping queue
TC>thread 2
TC>2009-10-20 10:05:59.493339 [CONSOLE] switch_core.c:1693 Finalizing Shutdown.
TC>2009-10-20 10:05:59.493379 [CONSOLE] switch_log.c:310 Logger Ended.
TC>2009-10-20 10:05:59.494472 [CONSOLE] switch_core_memory.c:567 Stopping
TC>memory pool queue.
TC>Segmentation fault (core dumped)
TC>tculjaga at subZero:~/freeswitch-trunk$
TC>tculjaga at subZero:~/freeswitch-trunk$

may be, i don't look on this so close at this time.

TC>Please advice your FS/mod_h323.conf.xml settings...

<configuration name="h323.conf" description="H323 Endpoints">
<settings>
<param name="trace-level" value="4"/>
<param name="context" value="default"/>
<param name="dialplan" value="XML"/>
<param name="codec-prefs" value="PCMA,PCMU,GSM,G729,G726,LPC"/>
<param name="gk-address" value="91.195.204.1"/>    <!-- empty to disable, "*" to search LAN -->
<param name="gk-identifer" value="ICFGK"/>  <!-- optional name of gk -->
<param name="gk-interface" value=""/>  <!-- optional listener interface name -->
</settings>
<listeners>
<listener name="default">
<param name="h323-ip" value="$${local_ip_v4}"/>
<param name="h323-port" value="1720"/>
</listener>
</listeners>
</configuration>

this my current settings.
							

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From ineyaa at gmail.com  Tue Oct 20 08:21:09 2009
From: ineyaa at gmail.com (ineya ineya)
Date: Tue, 20 Oct 2009 17:21:09 +0200
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <858EC056-1D84-4F30-A0B7-9A70FF98695D@freeswitch.org>
References: <7b8592a0910190516rafa4398q62c7a99ef703540f@mail.gmail.com>
	<5B924C57-B658-44C0-A1CE-00A3ECB5AA07@freeswitch.org>
	<7b8592a0910191351g14cbb47dj3f19362ff6197157@mail.gmail.com>
	<FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
	<7b8592a0910192253l1313d82j51c933cb5f55ad41@mail.gmail.com>
	<20091020133437.GA4941@dan.olp.net>
	<7b8592a0910200716q4a7db93cyb955595e4993ac96@mail.gmail.com>
	<858EC056-1D84-4F30-A0B7-9A70FF98695D@freeswitch.org>
Message-ID: <7b8592a0910200821m18fe9b4dg73dda05a10699af2@mail.gmail.com>

By numbers, I have 2 numbers registered, so I dial "1006" or "1007".
This worked for IPv4, so I haven't thought about doing it differently for IPv6.
SIP messages looked OK to me - you just gave me an idea to compare SIP
messages for IPv4 and IPv6. I'll try that tommorow.


On Tue, Oct 20, 2009 at 4:21 PM, Brian West <brian at freeswitch.org> wrote:
> Well how are you trying to dial users?
>
> /b
>



From rob4manhere at gmail.com  Tue Oct 20 08:23:24 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Tue, 20 Oct 2009 10:23:24 -0500
Subject: [Freeswitch-users] mod_odbc_query share memory problem
In-Reply-To: <8ccbff060910200814w2fa37866m8e211a73610f03c4@mail.gmail.com>
References: <8ccbff060910200631l44cc7dcfm2aa9511d86798aa1@mail.gmail.com>
	<180CE701-D840-47E6-821C-BB79C9F1945A@scarlet-internet.nl>
	<8ccbff060910200735h73e9459ex7460fcbeac457d59@mail.gmail.com>
	<85AF1814-9B1A-439C-83F5-D46F9BFEC140@gmail.com>
	<8ccbff060910200814w2fa37866m8e211a73610f03c4@mail.gmail.com>
Message-ID: <7B94B9BF-B15F-4690-B0E6-74776BD26898@gmail.com>

Those are recommended values for Oracle.  Freeswitch != Oracle.  They  
behave and use a server's resources very differently.

I wouldn't change more than you need for now.  Tweak shmmax then go  
from there.

Cheers,
Rob

On Oct 20, 2009, at 10:14 AM, Dome Charoenyost wrote:

> 2009/10/20 Rob Forman <rob4manhere at gmail.com>:
>> You could, but I would try just doubling whatever it is to see if
>> thats improves the issue first.  The default is 32MB.  You could
>> double it to 64MB and test again.
> Ok' let's me try 64MB first
> i found some information
> https://www.ishoof.org/project/wiki/index.php/ORACLE_11_G_-_Gentoo_Installation
>
>>
>> What is it currently set to (run: sysctl kernel.shmmax)?  You can
>> change it on the fly with sysctl.  Once you're done testing and want
>> it to persist across reboots, you'll want to set in /etc/ 
>> sysctl.conf .
>>
>> Cheers,
>> Rob
>>
>> On Oct 20, 2009, at 9:35 AM, Dome Charoenyost wrote:
>>
>>> 2009/10/20 Leon de Rooij <leon at scarlet-internet.nl>:
>>>> Hi,
>>>>
>>>> I haven't run into that problem yet, but did you try increasing the
>>>> maximum shared memory in /proc/sys/kernel/shmmax (sysctl
>>>> kernel.shmmax) ?
>>> Recommend me please i have 4GB RAM.
>>> is posible to increase to 50% RAM ?
>>>
>>> Dome C.
>>>>
>>>> regards,
>>>>
>>>> Leon
>>>>
>>>>
>>>>
>>>> On Oct 20, 2009, at 3:31 PM, Dome Charoenyost wrote:
>>>>
>>>>> Dear Sir,
>>>>> I'm using mod_odbc_query and mod_nibble_billing for my calling  
>>>>> card
>>>>> solutoin. i found mod_odbc_query cant work with high load  (200
>>>>> calls
>>>>> concurrent)
>>>>> i got error
>>>>> [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory;
>>>>> Error while executing the query
>>>>> ]
>>>>> in FS console
>>>>>
>>>>> How to fix it ?
>>>>>
>>>>> BG
>>>>> Dome C.
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From gmaruzz at celliax.org  Tue Oct 20 08:25:09 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 20 Oct 2009 17:25:09 +0200
Subject: [Freeswitch-users] Is anyone running Ubuntu 8.04/Hardy?
In-Reply-To: <4ADDC8D8.7090401@marketelsystems.com>
References: <4ADCA34E.5000700@marketelsystems.com>
	<903da5680910191917m79bf5ecfvdc8001952dbcc339@mail.gmail.com>
	<4ADDC8D8.7090401@marketelsystems.com>
Message-ID: <7b197bef0910200825v64e1239emf717f357e423794@mail.gmail.com>

maybe next time test and/or search the mailing list before asking. I
was a little worried when I read that it do not works on Hardy. Good
to be reassured, it works. :-)



On 10/20/09, Mark Sobkow <m.sobkow at marketelsystems.com> wrote:
> Gabriel Gunderson wrote:
>> On Mon, Oct 19, 2009 at 11:35 AM, Mark Sobkow
>> <m.sobkow at marketelsystems.com> wrote:
>>
>>> Everyone I've emailed with on the dev list is running the current
>>> release of Ubuntu, not 8.04/Hardy.
>>>
>>
>> Well, what issues?
>>
>> Gabe
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> We had problems with loading mod_sofia.  However, we were running an
> older release, then we'd tried upgrading to the source build from
> launchpad.net.
>
> Yesterday we downloaded the current svn.freeswitch.org, and that
> particular problem has gone away.  Now we need to figure out why our
> Erlang component can synchronize from Windows, but not from Linux.
>
> --
> Mark Sobkow
> Senior Developer
> MarkeTel Multi-Line Dialing Systems LTD.
> 428 Victoria Ave
> Regina, SK S4N-0P6
> Toll-Free: 800-289-8616-X533
> Local: 306-359-6893-X533
> Fax: 306-359-6879
> Email: m.sobkow at marketelsystems.com
> Web: http://www.marketelsystems.com
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From abovebluesky at gmail.com  Tue Oct 20 08:25:48 2009
From: abovebluesky at gmail.com (Uncle Johny)
Date: Tue, 20 Oct 2009 08:25:48 -0700 (PDT)
Subject: [Freeswitch-users] bind_meta_app problem with custome extension
In-Reply-To: <25963971.post@talk.nabble.com>
References: <25963971.post@talk.nabble.com>
Message-ID: <25977224.post@talk.nabble.com>


OMG such a stupid user error!

I have to write
<action application="bind_meta_app" data="7 ab s
/usr/local/freeswitch/conf/dialplan/execute_extension::roar XML features"/>

with double colon instead of a single one.

Thanks everyone for help!


Uncle Johny wrote:
> 
> Hi guys,
> 
> I hope you can help me with this one since it looks like some FS bug or my
> user error :) .I have a simple dialplan in FS which allows to transfer a
> person to some 3rd party phone but it works only if you call it from
> default context.
> If it is not called from default context I get this error all the time:
> 2009-09-12 11:58:17.484680 [ERR] mod_native_file.c:68 Error opening
> /usr/local/freeswitch/conf/dialplan/execute_extension:roar XML
> features.PCMU
> 
> With the following configuration 2 parties will be connected but whenever
> you press *7 it just does not work and FS splits the message. Very weird
> that this works if I put the extension into default.xml.
> 
> Any ideas what is wrong here?
> 
> My config:
> 
> I tried to create my own context and call it pstn.
> So when FS gets a call from proxy it goes to pstn context, since I have
> this line in sip_proxies/internal.xml
> 
> 
> in dialplan directory I have pstn.xml
> <include>
>   <context name="pstn">
>    <extension name="pstn_enum">
>     <condition field="destination_number" expression="^(.*)$">
>                 <action application="answer"/>
>                 <action application="bind_meta_app" data="7 ab s
> /usr/local/freeswitch/conf/dialplan/execute_extension:roar XML features"/>
>                 <action application="set"
> data="hangup_after_bridge=true"/>
>                 <action application="bridge"
> data="sofia/external/${destination_number}@207.88.122.16"/>
>     </condition>
>   </extension>
> </context>
> </include>
> 
> In features.xml (It can be anything there since FS never reaches this code
> and dies before executing it):
> <extension name="roar">
>      <condition field="destination_number" expression="^roar$">
>         <action application="set" data="continue_on_fail=true"/>
>         <action application="log" data="INFO transfer context" />
>         <action application="read" data="1 1 /root/1.gsm attxfer_callthis
> 1000 #"/>
>         <action application="log" data="INFO blah ${attxfer_callthis}"/>
>         <action application="set" data="origination_cancel_key=#"/>
>         <action application="log" data="INFO after read"/>
>      </condition>
> </extension>
> 
>  
> 

-- 
View this message in context: http://www.nabble.com/bind_meta_app-problem-with-custome-extension-tp25963971p25977224.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From anthony.minessale at gmail.com  Tue Oct 20 08:31:49 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 20 Oct 2009 10:31:49 -0500
Subject: [Freeswitch-users] Stereo Support
In-Reply-To: <4ADD685C.1000506@ewetel.de>
References: <4ADD685C.1000506@ewetel.de>
Message-ID: <191c3a030910200831h59a58e2aqf1aee92cc4859786@mail.gmail.com>

There is not enough spec/devices using spec on stereo to try to implement it
at this time.
Maybe some day.


On Tue, Oct 20, 2009 at 2:35 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> I wonder if there is real stereo support planned for FreeSWITCH in terms
> of streaming music/video-audio to desktop? I ask, because I heared that
> celt codec is supporting stereo.
>
> regards
> helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
>
> iD8DBQFK3Whc4tZeNddg3dwRAkdEAKCLBSV4H8q5JN4NZ39fgSZPxr5PCgCgmpTq
> PAu8qozLP5FaVKIwfCXVqjU=
> =Zseo
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From dome at tel.co.th  Tue Oct 20 08:33:25 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Tue, 20 Oct 2009 22:33:25 +0700
Subject: [Freeswitch-users] mod_odbc_query share memory problem
In-Reply-To: <7B94B9BF-B15F-4690-B0E6-74776BD26898@gmail.com>
References: <8ccbff060910200631l44cc7dcfm2aa9511d86798aa1@mail.gmail.com>
	<180CE701-D840-47E6-821C-BB79C9F1945A@scarlet-internet.nl>
	<8ccbff060910200735h73e9459ex7460fcbeac457d59@mail.gmail.com>
	<85AF1814-9B1A-439C-83F5-D46F9BFEC140@gmail.com>
	<8ccbff060910200814w2fa37866m8e211a73610f03c4@mail.gmail.com>
	<7B94B9BF-B15F-4690-B0E6-74776BD26898@gmail.com>
Message-ID: <8ccbff060910200833g892d35fr69c2a4ea5c16562b@mail.gmail.com>

Now i'm setting to  64MB
I found channel problem also. many channel not disconnect
freeswitch at internal> show calls count
87 total.

freeswitch at internal> show channels count
466 total.

freeswitch at internal> status
UP 0 years, 0 days, 1 hour, 53 minutes, 23 seconds, 872 milliseconds,
0 microseconds
11641 session(s) since startup
154 session(s) 0/30
1000 session(s) max

I think channels should be 154 (same session) but 466 is to high. may
be some channels not disconnect

Dome C.



2009/10/20 Rob Forman <rob4manhere at gmail.com>:
> Those are recommended values for Oracle. ?Freeswitch != Oracle. ?They
> behave and use a server's resources very differently.
>
> I wouldn't change more than you need for now. ?Tweak shmmax then go
> from there.
>
> Cheers,
> Rob
>
> On Oct 20, 2009, at 10:14 AM, Dome Charoenyost wrote:
>
>> 2009/10/20 Rob Forman <rob4manhere at gmail.com>:
>>> You could, but I would try just doubling whatever it is to see if
>>> thats improves the issue first. ?The default is 32MB. ?You could
>>> double it to 64MB and test again.
>> Ok' let's me try 64MB first
>> i found some information
>> https://www.ishoof.org/project/wiki/index.php/ORACLE_11_G_-_Gentoo_Installation
>>
>>>
>>> What is it currently set to (run: sysctl kernel.shmmax)? ?You can
>>> change it on the fly with sysctl. ?Once you're done testing and want
>>> it to persist across reboots, you'll want to set in /etc/
>>> sysctl.conf .
>>>
>>> Cheers,
>>> Rob
>>>
>>> On Oct 20, 2009, at 9:35 AM, Dome Charoenyost wrote:
>>>
>>>> 2009/10/20 Leon de Rooij <leon at scarlet-internet.nl>:
>>>>> Hi,
>>>>>
>>>>> I haven't run into that problem yet, but did you try increasing the
>>>>> maximum shared memory in /proc/sys/kernel/shmmax (sysctl
>>>>> kernel.shmmax) ?
>>>> Recommend me please i have 4GB RAM.
>>>> is posible to increase to 50% RAM ?
>>>>
>>>> Dome C.
>>>>>
>>>>> regards,
>>>>>
>>>>> Leon
>>>>>
>>>>>
>>>>>
>>>>> On Oct 20, 2009, at 3:31 PM, Dome Charoenyost wrote:
>>>>>
>>>>>> Dear Sir,
>>>>>> I'm using mod_odbc_query and mod_nibble_billing for my calling
>>>>>> card
>>>>>> solutoin. i found mod_odbc_query cant work with high load ?(200
>>>>>> calls
>>>>>> concurrent)
>>>>>> i got error
>>>>>> [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory;
>>>>>> Error while executing the query
>>>>>> ]
>>>>>> in FS console
>>>>>>
>>>>>> How to fix it ?
>>>>>>
>>>>>> BG
>>>>>> Dome C.
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From brian at freeswitch.org  Tue Oct 20 08:34:21 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 20 Oct 2009 10:34:21 -0500
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <7b8592a0910200821m18fe9b4dg73dda05a10699af2@mail.gmail.com>
References: <7b8592a0910190516rafa4398q62c7a99ef703540f@mail.gmail.com>
	<5B924C57-B658-44C0-A1CE-00A3ECB5AA07@freeswitch.org>
	<7b8592a0910191351g14cbb47dj3f19362ff6197157@mail.gmail.com>
	<FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
	<7b8592a0910192253l1313d82j51c933cb5f55ad41@mail.gmail.com>
	<20091020133437.GA4941@dan.olp.net>
	<7b8592a0910200716q4a7db93cyb955595e4993ac96@mail.gmail.com>
	<858EC056-1D84-4F30-A0B7-9A70FF98695D@freeswitch.org>
	<7b8592a0910200821m18fe9b4dg73dda05a10699af2@mail.gmail.com>
Message-ID: <642B6CED-CDF3-4D40-B493-0F2D6222C30B@freeswitch.org>

The defaults will NOT work with ipv6 out of the box because the  
sofia_contact on the directory only looks at the internal profile NOT  
the internal-ipv6 profile... open up the directory default and change  
the sofia_contact to prepend the internal-ipv6/user at domain

/b

On Oct 20, 2009, at 10:21 AM, ineya ineya wrote:

> By numbers, I have 2 numbers registered, so I dial "1006" or "1007".
> This worked for IPv4, so I haven't thought about doing it  
> differently for IPv6.
> SIP messages looked OK to me - you just gave me an idea to compare SIP
> messages for IPv4 and IPv6. I'll try that tommorow.




From dome at tel.co.th  Tue Oct 20 08:47:37 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Tue, 20 Oct 2009 22:47:37 +0700
Subject: [Freeswitch-users] CS_REPORTING Channel event state
Message-ID: <8ccbff060910200847y5e7ccab6o84d431836970bcb4@mail.gmail.com>

Dear All
What's CS_REPORTING state ?
I found many channels not hang up  ans state is CS_REPORTING
e264f84a-bd87-11de-9a90-2320c02172de,outbound,2009-10-20
21:50:26,1256050226,sofia/external/xxxxxxxxxxxx at xxx.xxx.xx.191:7050,CS_REPORTING,FreeSWITCH,xxxxxxxxxxxxxxxxxxxx,xx.xxx.xxx.xxx,7050,,,XML,public,,,,,

BG
Dome C.



From msc at freeswitch.org  Tue Oct 20 09:19:45 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 20 Oct 2009 09:19:45 -0700
Subject: [Freeswitch-users] Qustion about INFO messages after
	Connect/Answer
In-Reply-To: <4ADD65DA.4050809@ewetel.de>
References: <4AD859A4.5000101@ewetel.de>
	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>
	<4ADC7C09.6070503@ewetel.de>
	<191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>
	<4ADC85C0.7060908@ewetel.de> <4ADC8B6E.7070404@ewetel.de>
	<191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>
	<4ADD65DA.4050809@ewetel.de>
Message-ID: <87f2f3b90910200919i62dac94fq1b6c9289d9fb357c@mail.gmail.com>

On Tue, Oct 20, 2009 at 12:25 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello Anthony,
>
>
> On 19.10.2009 22:07, Anthony Minessale wrote:
> > please update and test trunk
> >
> > 1) I changed the core to remove the excess data by default in your
> scenario
> > 2) I added variables you can use to control it
> > origination_callee_id_name origination_callee_id_number which belong in
> > {} in the dial string eg
> {origination_callee_id_number=1234}openzap/1/a/1234
>
> updated and testet for SIP calls. "origination_callee_id_number=1234"
> works within the dial string and via using "export" application but not
> via "set" application. Didn't test it for openzap, yet but I guess it
> will work, too.
>

Thanks for testing. BTW, according to Tony's instructions the user needs to
put the variable definition inside the {} of the dialstring. This means that
the vars are designed for the new leg, not the local leg, which means that
"export" would work but "set" would not. (Remember, "set" is to set a chan
var on the local channel, "export" will set a chan var locally *and* on the
other call leg. See also the "nolocal" option of export.)

>
> What left is that "unknown" thing on callee's display (SIP phones only I
> guess) ... I would suggest to make sending "unknown" INFO message
> optional, but to be honest, I have no idea for what it was invented, so
> maybe my suggestion is nonsens.
>
>
Under what conditions did you see "unknown"? I'm wondering if the user can
just pick a default other than "unknown" if he wants something else to be
displayed.

Thoughts?
-MC
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From brian at freeswitch.org  Tue Oct 20 09:41:48 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 20 Oct 2009 11:41:48 -0500
Subject: [Freeswitch-users] Qustion about INFO messages after
	Connect/Answer
In-Reply-To: <87f2f3b90910200919i62dac94fq1b6c9289d9fb357c@mail.gmail.com>
References: <4AD859A4.5000101@ewetel.de>
	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>
	<4ADC7C09.6070503@ewetel.de>
	<191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>
	<4ADC85C0.7060908@ewetel.de> <4ADC8B6E.7070404@ewetel.de>
	<191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>
	<4ADD65DA.4050809@ewetel.de>
	<87f2f3b90910200919i62dac94fq1b6c9289d9fb357c@mail.gmail.com>
Message-ID: <CBC6759C-5192-4731-BACA-79C90CD4EE27@freeswitch.org>

Or just set the var to what you want it to say?

/b

On Oct 20, 2009, at 11:19 AM, Michael Collins wrote:

>
> Under what conditions did you see "unknown"? I'm wondering if the  
> user can just pick a default other than "unknown" if he wants  
> something else to be displayed.
>
> Thoughts?
> -MC




From ineyaa at gmail.com  Tue Oct 20 09:47:33 2009
From: ineyaa at gmail.com (ineya ineya)
Date: Tue, 20 Oct 2009 18:47:33 +0200
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <642B6CED-CDF3-4D40-B493-0F2D6222C30B@freeswitch.org>
References: <7b8592a0910190516rafa4398q62c7a99ef703540f@mail.gmail.com>
	<5B924C57-B658-44C0-A1CE-00A3ECB5AA07@freeswitch.org>
	<7b8592a0910191351g14cbb47dj3f19362ff6197157@mail.gmail.com>
	<FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
	<7b8592a0910192253l1313d82j51c933cb5f55ad41@mail.gmail.com>
	<20091020133437.GA4941@dan.olp.net>
	<7b8592a0910200716q4a7db93cyb955595e4993ac96@mail.gmail.com>
	<858EC056-1D84-4F30-A0B7-9A70FF98695D@freeswitch.org>
	<7b8592a0910200821m18fe9b4dg73dda05a10699af2@mail.gmail.com>
	<642B6CED-CDF3-4D40-B493-0F2D6222C30B@freeswitch.org>
Message-ID: <7b8592a0910200947k6ca5a391l35786f0e855ae388@mail.gmail.com>

Hmm, I didn't about this.

so in directory/default.xml I have:
 <param name="dial-string"
value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>

and the modified version for IPv6 would be ?...
 <param name="dial-string"
value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(internal-ipv6/${dialed_user}@${dialed_domain})}"/>



On Tue, Oct 20, 2009 at 5:34 PM, Brian West <brian at freeswitch.org> wrote:
> The defaults will NOT work with ipv6 out of the box because the
> sofia_contact on the directory only looks at the internal profile NOT
> the internal-ipv6 profile... open up the directory default and change
> the sofia_contact to prepend the internal-ipv6/user at domain
>
> /b
>



From msc at freeswitch.org  Tue Oct 20 10:42:07 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 20 Oct 2009 10:42:07 -0700
Subject: [Freeswitch-users] Janitorial Project Update - Wiki Cleanup
Message-ID: <87f2f3b90910201042o75b93127v619d61b84d962677@mail.gmail.com>

Hello all,

As you may know we have a weekly conference call each Friday. This week's
agenda has several wiki cleanup sub-projects. I wanted to crowdsource this
because, as the saying goes, many hands makes the work light. In other
words, if everyone can help a little bit then we won't be dumping a lot of
work on just a few people.

HOW YOU CAN HELP

If you don't already have a wiki account then please go to
http://wiki.freeswitch.org and create one. All wiki users have editing
rights and can make changes to wiki content. Don't worry about making
mistakes! We can always revert a change. Brian West and I review all wiki
edits each day. Standard users can't accidentally delete a page or
permanently remove content. You're totally safe to make edits. By the way,
when you sign in to the wiki you won't see the ads on the left-hand side of
the page and you won't have to scroll down so far to locate the search box.

There are several ways that you can assist with the wiki cleanup. One way is
to help users like Diego Viola who are quite literally going page to page
looking for spelling errors, grammar, etc. If you see an obvious error
please by all means correct it.

Another thing you can do is visit this page and look at the janitorial
section:
http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_23

There are several areas where we can clean up and improve the wiki:

*Dead end pages - these are pages that are linked to by other wiki pages but
do not themselves have links back to anywhere. At the very least each page
should have a link back to some other "see also" kind of content.

*Double redirects - these are redirect pages that point to other redirect
pages. These need to be pointed to the ultimate destination page.

*Orphaned pages - orphans are pages that are not linked to by other pages.
The only way to access these pages is via a search or if you already know
the URL. We need people to review the orphaned pages and add links to them
from other pages.

*Wanted pages and categories - these are pages and categories that have
links but no content. We need people review them and add content or
redirects as appropriate.

*Uncategorized pages - all pages belong to some category and these pages
haven't been added to a category yet. Categories are handy because they act
as natural indexes. Please review pages that have no categories. If you have
an idea for a category that does not yet exist please email me and let me
know. We don't want to have too many categories.

The best way to learn wiki markup is to click the edit button on a page and
see what wiki markup does. You can change the markup and then click Preview
to see what it doesn. Click the cancel link to discard any changes. Play
around and learn!

Thank you for your help. Please feel free to ask for assistance with editing
the wiki. It is a community resource and the more people who know how to add
content, the better off we all are.

Thanks,
Michael
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From dmitry.bely at gmail.com  Tue Oct 20 10:43:03 2009
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Tue, 20 Oct 2009 21:43:03 +0400
Subject: [Freeswitch-users] Status of ubuntu/debian packages.
In-Reply-To: <4ABA47FB.2050100@gmail.com>
References: <4ABA47FB.2050100@gmail.com>
Message-ID: <90823c940910201043g55f51005kc3b90d6b8c8a4f6f@mail.gmail.com>

William,

Where is mod_skypiax.so? Isn't freeswitch-skypiax package expected to
contain it?

- Dmitry Bely



From msc at freeswitch.org  Tue Oct 20 10:44:10 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 20 Oct 2009 10:44:10 -0700
Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Agenda - Oct
	23rd
Message-ID: <87f2f3b90910201044v31a281d0j1c441425ba993786@mail.gmail.com>

FYI,

The conference call agenda for this week is online:
http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_23

Also, we will be discussing whether or not Friday is the best time to be
doing this call. Just remember that the FS devs have lots of work to do and
they can't always accommodate everyone's schedules. Still, they'll do their
best to be flexible wherever possible.

Thanks for supporting the weekly conference call.
-Michael
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From chris at cloudtel.com  Tue Oct 20 11:23:35 2009
From: chris at cloudtel.com (Chris Burns)
Date: Tue, 20 Oct 2009 14:23:35 -0400
Subject: [Freeswitch-users] Connect PHP SOAP Web Server with SQLite
	database of FS
In-Reply-To: <25976554.post@talk.nabble.com>
References: <25976554.post@talk.nabble.com>
Message-ID: <200910201423.35419.chris@cloudtel.com>

If you really wanted: http://php.net/manual/en/book.sqlite.php

But I would recommend you make use of ODBC to use a client/server RDBMS. 
Here's some good reading: 
http://www.sqlite.org/cvstrac/wiki?p=WhenToUseSqlite

On October 20, 2009 10:53:01 am homqua wrote:
> Now I am building a PHP SOAP Web Service to access the database of FS.
> Anyone has idea about how to access sqlite database of FS through PHP ? I
> have read about socket event in FS, but I don't know whether it can
> response with the query of database or not.
> Thanks for your help.





From msc at freeswitch.org  Tue Oct 20 11:59:48 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 20 Oct 2009 11:59:48 -0700
Subject: [Freeswitch-users] Digium TE220 wiki
In-Reply-To: <7d79b3930910200018s25184c01k838bff05ad308ffb@mail.gmail.com>
References: <7d79b3930910200018s25184c01k838bff05ad308ffb@mail.gmail.com>
Message-ID: <87f2f3b90910201159h1721eqe363e9ed397d634b@mail.gmail.com>

Thanks. I added minor changes, plus a category and see also. Thanks for
adding this content!
-MC

On Tue, Oct 20, 2009 at 12:18 AM, lakshmanan ganapathy <lakindia89 at gmail.com
> wrote:

> Hi all,
>     I've created a wiki page, which contains the example configuration for
> making Digium TE220 to work.
>     I request you people to check this, and give feedbacks.
>
> http://wiki.freeswitch.org/wiki/Configuration_OpenZap-DigiumTE220-Example
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kristian.kielhofner at gmail.com  Tue Oct 20 12:56:15 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Tue, 20 Oct 2009 15:56:15 -0400
Subject: [Freeswitch-users] Troubles with proxy media mode
Message-ID: <2d9149cd0910201256j39b01228ufa34f977787f0c79@mail.gmail.com>

Hello everyone,

  I'm trying to use proxy media across two profiles.  The codec
settings are identical, they both have late negotiation enabled, and
they both have inbound-proxy-media set to true (I also tried setting
proxy_media from the dialplan).

  FreeSWITCH ends up clearing the call with TRANSCODING_NECESSARY but
I can't figure out why it thinks it needs to transcode for this call.
I've attached a level 7 debug and an ngrep siptrace showing traffic
from both profiles.

  FreeSWITCH trunk rev. 15180 running on Debian 5.0.2.

  See anything interesting?

Thanks!

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com
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From anthony.minessale at gmail.com  Tue Oct 20 13:11:37 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 20 Oct 2009 15:11:37 -0500
Subject: [Freeswitch-users] Troubles with proxy media mode
In-Reply-To: <2d9149cd0910201256j39b01228ufa34f977787f0c79@mail.gmail.com>
References: <2d9149cd0910201256j39b01228ufa34f977787f0c79@mail.gmail.com>
Message-ID: <191c3a030910201311t31806073y936fac4c6c4ef9c@mail.gmail.com>

Fixed in 15181 =D 1 revision higher doh =D


On Tue, Oct 20, 2009 at 2:56 PM, Kristian Kielhofner <
kristian.kielhofner at gmail.com> wrote:

> Hello everyone,
>
>  I'm trying to use proxy media across two profiles.  The codec
> settings are identical, they both have late negotiation enabled, and
> they both have inbound-proxy-media set to true (I also tried setting
> proxy_media from the dialplan).
>
>  FreeSWITCH ends up clearing the call with TRANSCODING_NECESSARY but
> I can't figure out why it thinks it needs to transcode for this call.
> I've attached a level 7 debug and an ngrep siptrace showing traffic
> from both profiles.
>
>  FreeSWITCH trunk rev. 15180 running on Debian 5.0.2.
>
>  See anything interesting?
>
> Thanks!
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From tculjaga at gmail.com  Tue Oct 20 13:16:42 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 20 Oct 2009 22:16:42 +0200
Subject: [Freeswitch-users] Troubles with proxy media mode
In-Reply-To: <2d9149cd0910201256j39b01228ufa34f977787f0c79@mail.gmail.com>
References: <2d9149cd0910201256j39b01228ufa34f977787f0c79@mail.gmail.com>
Message-ID: <65d96fc80910201316i3e0c4d73q9bf269176a420045@mail.gmail.com>

you are making FS to play wav file when sending a call in G711 or GSM or
some other codec.

you might use mod_native_file<http://wiki.freeswitch.org/wiki/Mod_native_file>to
avoid transcoding.

T.



On Tue, Oct 20, 2009 at 9:56 PM, Kristian Kielhofner <
kristian.kielhofner at gmail.com> wrote:

> Hello everyone,
>
>  I'm trying to use proxy media across two profiles.  The codec
> settings are identical, they both have late negotiation enabled, and
> they both have inbound-proxy-media set to true (I also tried setting
> proxy_media from the dialplan).
>
>  FreeSWITCH ends up clearing the call with TRANSCODING_NECESSARY but
> I can't figure out why it thinks it needs to transcode for this call.
> I've attached a level 7 debug and an ngrep siptrace showing traffic
> from both profiles.
>
>  FreeSWITCH trunk rev. 15180 running on Debian 5.0.2.
>
>  See anything interesting?
>
> Thanks!
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kristian.kielhofner at gmail.com  Tue Oct 20 13:37:24 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Tue, 20 Oct 2009 16:37:24 -0400
Subject: [Freeswitch-users] Troubles with proxy media mode
In-Reply-To: <191c3a030910201311t31806073y936fac4c6c4ef9c@mail.gmail.com>
References: <2d9149cd0910201256j39b01228ufa34f977787f0c79@mail.gmail.com>
	<191c3a030910201311t31806073y936fac4c6c4ef9c@mail.gmail.com>
Message-ID: <2d9149cd0910201337m22a03d1fr9538cc54fbd22b98@mail.gmail.com>

Updated and now working.

Thanks!

On Tue, Oct 20, 2009 at 4:11 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Fixed in 15181 =D 1 revision higher doh =D
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From kristian.kielhofner at gmail.com  Tue Oct 20 13:57:41 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Tue, 20 Oct 2009 16:57:41 -0400
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
Message-ID: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>

Hello everyone,

  Now that proxy media mode is working again I'm trying to figure out
why T.38 with re-INVITE doesn't...

  Everything goes well until my end tries to re-INVITE to T.38:

U 10.16.5.129:5060 -> 65.196.170.191:5060
INVITE sip:mod_sofia at 65.196.170.191:5060 SIP/2.0.
Via: SIP/2.0/UDP 10.16.5.129:5060;branch=z9hG4bK-2656bddb.
From: <sip:fax_9186 at 10.16.5.129:5060>;tag=9f75a042a3ca1f17i0.
To: "WIRELESS CALLER" <sip:19412848354 at 65.196.170.191>;tag=4eUH874j0eUZF.
Remote-Party-ID: fax_9186 <sip:fax_9186 at 65.196.170.191>;screen=yes;party=called.
Call-ID: 2889ab05-385d-122d-07b2-0014221c0392.
CSeq: 101 INVITE.
Max-Forwards: 70.
Contact: fax_9186 <sip:fax_9186 at 10.16.5.129:5060>.
Expires: 30.
User-Agent: Linksys/SPA3102-5.1.10(GW).
Content-Length: 261.
Content-Type: application/sdp.
.
v=0.
o=- 57670 57670 IN IP4 10.16.5.129.
s=-.
c=IN IP4 10.16.5.129.
t=0 0.
m=image 16454 udptl t38.
a=T38FaxVersion:0.
a=T38MaxBitRate:14400.
a=T38FaxRateManagement:transferredTCF.
a=T38FaxMaxBuffer:200.
a=T38FaxMaxDatagram:200.
a=T38FaxUdpEC:t38UDPRedundancy.


U 65.196.170.191:5060 -> 10.16.5.129:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 10.16.5.129:5060;branch=z9hG4bK-2656bddb.
From: <sip:fax_9186 at 10.16.5.129:5060>;tag=9f75a042a3ca1f17i0.
To: "WIRELESS CALLER" <sip:19412848354 at 65.196.170.191>;tag=4eUH874j0eUZF.
Call-ID: 2889ab05-385d-122d-07b2-0014221c0392.
CSeq: 101 INVITE.
User-Agent: Star2Star Media.
Content-Length: 0.
.

U 65.196.170.191:5060 -> 10.16.5.129:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 10.16.5.129:5060;branch=z9hG4bK-2656bddb.
From: <sip:fax_9186 at 10.16.5.129:5060>;tag=9f75a042a3ca1f17i0.
To: "WIRELESS CALLER" <sip:19412848354 at 65.196.170.191>;tag=4eUH874j0eUZF.
Call-ID: 2889ab05-385d-122d-07b2-0014221c0392.
CSeq: 101 INVITE.
Contact: <sip:mod_sofia at 65.196.170.191:5060>.
User-Agent: Star2Star Media.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, precondition, path, replaces.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 242.
.
v=0.
o=Sonus_UAC 881416742673069109 186840485790971511 IN IP4 65.196.170.129.
s=SIP Media Capabilities.
c=IN IP4 65.196.170.191.
t=0 0.
m=audio 19646 RTP/AVP 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.


U 10.16.5.129:5060 -> 65.196.170.191:5060
ACK sip:mod_sofia at 65.196.170.191:5060 SIP/2.0.
Via: SIP/2.0/UDP 10.16.5.129:5060;branch=z9hG4bK-842177a7.
From: <sip:fax_9186 at 10.16.5.129:5060>;tag=9f75a042a3ca1f17i0.
To: "WIRELESS CALLER" <sip:19412848354 at 65.196.170.191>;tag=4eUH874j0eUZF.
Call-ID: 2889ab05-385d-122d-07b2-0014221c0392.
CSeq: 101 ACK.
Max-Forwards: 70.
Contact: fax_9186 <sip:fax_9186 at 10.16.5.129:5060>.
User-Agent: Linksys/SPA3102-5.1.10(GW).
Content-Length: 0.
.

U 10.16.5.129:5060 -> 65.196.170.191:5060
BYE sip:mod_sofia at 65.196.170.191:5060 SIP/2.0.
Via: SIP/2.0/UDP 10.16.5.129:5060;branch=z9hG4bK-9e6b284b.
From: <sip:fax_9186 at 10.16.5.129:5060>;tag=9f75a042a3ca1f17i0.
To: "WIRELESS CALLER" <sip:19412848354 at 65.196.170.191>;tag=4eUH874j0eUZF.
Call-ID: 2889ab05-385d-122d-07b2-0014221c0392.
CSeq: 102 BYE.
Max-Forwards: 70.
User-Agent: Linksys/SPA3102-5.1.10(GW).
Content-Length: 0.
.

U 65.196.170.191:5060 -> 10.16.5.129:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 10.16.5.129:5060;branch=z9hG4bK-9e6b284b.
From: <sip:fax_9186 at 10.16.5.129:5060>;tag=9f75a042a3ca1f17i0.
To: "WIRELESS CALLER" <sip:19412848354 at 65.196.170.191>;tag=4eUH874j0eUZF.
Call-ID: 2889ab05-385d-122d-07b2-0014221c0392.
CSeq: 102 BYE.
User-Agent: Star2Star Media.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, precondition, path, replaces.
Content-Length: 0.
.

U 65.196.170.191:5080 -> 65.196.170.129:5060
BYE sip:gw+sip0 at 65.196.170.129:5080;transport=udp SIP/2.0.
Via: SIP/2.0/UDP 65.196.170.191:5080;rport;branch=z9hG4bKpv9pFF0S5r9ca.
Route: <sip:65.196.170.129;lr=on;ftag=3D535BNy2tS9e;did=699.54d14392>.
Max-Forwards: 70.
From: <sip:+15673861009 at 65.196.170.129>;tag=p4H9tpXS1Qv8m.
To: "WIRELESS CALLER" <sip:+19412848354 at 65.196.170.129>;tag=3D535BNy2tS9e.
Call-ID: 2d4c8faa-385d-122d-a69b-003018ae0bd3.
CSeq: 121917605 BYE.
Contact: <sip:+15673861009 at 65.196.170.191:5080;transport=udp>.
User-Agent: Star2Star Media.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY.
Supported: timer, precondition, path, replaces.
Reason: Q.850;cause=16;text="NORMAL_CLEARING".
Content-Length: 0.
.

  It's bizarre because the re-INVITE never gets transmitted to the
other leg and FreeSWITCH responds with an SDP that only supports
RFC2833.  Here is the FreeSWITCH level 7 log:

2009-10-20 20:54:15.577191 [NOTICE] switch_channel.c:613 New Channel
sofia/s2s/+19412848354 at 65.196.170.129
[ba5b3890-bdba-11de-b004-d7a35c351f6f]
2009-10-20 20:54:15.577191 [DEBUG] switch_core_state_machine.c:306
(sofia/s2s/+19412848354 at 65.196.170.129) Running State Change CS_NEW
2009-10-20 20:54:15.577191 [DEBUG] sofia.c:3493 Channel
sofia/s2s/+19412848354 at 65.196.170.129 entering state [received][100]
2009-10-20 20:54:15.577191 [DEBUG] sofia.c:3500 Remote SDP:
v=0
o=Sonus_UAC 7412556365376427063 4600708733824463060 IN IP4 65.196.170.129
s=SIP Media Capabilities
c=IN IP4 4.55.0.130
t=0 0
m=audio 26636 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20

2009-10-20 20:54:15.577191 [DEBUG] switch_core_state_machine.c:312
(sofia/s2s/+19412848354 at 65.196.170.129) State NEW
2009-10-20 20:54:15.577191 [DEBUG] sofia.c:3615
(sofia/s2s/+19412848354 at 65.196.170.129) State Change CS_NEW -> CS_INIT
2009-10-20 20:54:15.577191 [DEBUG] switch_core_session.c:985 Send
signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:306
(sofia/s2s/+19412848354 at 65.196.170.129) Running State Change CS_INIT
2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:330
(sofia/s2s/+19412848354 at 65.196.170.129) State INIT
2009-10-20 20:54:15.581191 [DEBUG] mod_sofia.c:83
sofia/s2s/+19412848354 at 65.196.170.129 SOFIA INIT
2009-10-20 20:54:15.581191 [DEBUG] mod_sofia.c:111
(sofia/s2s/+19412848354 at 65.196.170.129) State Change CS_INIT ->
CS_ROUTING
2009-10-20 20:54:15.581191 [DEBUG] switch_core_session.c:985 Send
signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:330
(sofia/s2s/+19412848354 at 65.196.170.129) State INIT going to sleep
2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:306
(sofia/s2s/+19412848354 at 65.196.170.129) Running State Change
CS_ROUTING
2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:333
(sofia/s2s/+19412848354 at 65.196.170.129) State ROUTING
2009-10-20 20:54:15.581191 [DEBUG] mod_sofia.c:130
sofia/s2s/+19412848354 at 65.196.170.129 SOFIA ROUTING
2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:78
sofia/s2s/+19412848354 at 65.196.170.129 Standard ROUTING
2009-10-20 20:54:15.581191 [INFO] mod_dialplan_xml.c:391 Processing
WIRELESS CALLER->fax_9186 in context s2s-in
Dialplan: sofia/s2s/+19412848354 at 65.196.170.129 parsing
[s2s-in->unloop] continue=false
Dialplan: sofia/s2s/+19412848354 at 65.196.170.129 Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/s2s/+19412848354 at 65.196.170.129 Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/s2s/+19412848354 at 65.196.170.129 parsing
[s2s-in->faxpass] continue=false
Dialplan: sofia/s2s/+19412848354 at 65.196.170.129 Absolute Condition [faxpass]
Dialplan: sofia/s2s/+19412848354 at 65.196.170.129 Action
bridge(sofia/rmt/fax_9186%rmt.star2star.com)
2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:114
(sofia/s2s/+19412848354 at 65.196.170.129) State Change CS_ROUTING ->
CS_EXECUTE
2009-10-20 20:54:15.581191 [DEBUG] switch_core_session.c:985 Send
signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:333
(sofia/s2s/+19412848354 at 65.196.170.129) State ROUTING going to sleep
2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:306
(sofia/s2s/+19412848354 at 65.196.170.129) Running State Change
CS_EXECUTE
2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:340
(sofia/s2s/+19412848354 at 65.196.170.129) State EXECUTE
2009-10-20 20:54:15.581191 [DEBUG] mod_sofia.c:173
sofia/s2s/+19412848354 at 65.196.170.129 SOFIA EXECUTE
2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:151
sofia/s2s/+19412848354 at 65.196.170.129 Standard EXECUTE
EXECUTE sofia/s2s/+19412848354 at 65.196.170.129
bridge(sofia/rmt/fax_9186%rmt.star2star.com)
2009-10-20 20:54:15.585206 [NOTICE] switch_channel.c:613 New Channel
sofia/rmt/fax_9186 [ba5baee2-bdba-11de-b004-d7a35c351f6f]
2009-10-20 20:54:15.585206 [DEBUG] mod_sofia.c:2981
(sofia/rmt/fax_9186) State Change CS_NEW -> CS_INIT
2009-10-20 20:54:15.585206 [DEBUG] switch_core_session.c:985 Send
signal sofia/rmt/fax_9186 [BREAK]
2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:306
(sofia/rmt/fax_9186) Running State Change CS_INIT
2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:330
(sofia/rmt/fax_9186) State INIT
2009-10-20 20:54:15.585206 [DEBUG] mod_sofia.c:83 sofia/rmt/fax_9186 SOFIA INIT
2009-10-20 20:54:15.585206 [DEBUG] sofia_glue.c:1328
sofia/rmt/fax_9186 Patched SDP
---
v=0
o=Sonus_UAC 7412556365376427063 4600708733824463060 IN IP4 65.196.170.129
s=SIP Media Capabilities
c=IN IP4 4.55.0.130
t=0 0
m=audio 26636 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20

+++
v=0
o=Sonus_UAC 7412556365376427063 4600708733824463060 IN IP4 65.196.170.129
s=SIP Media Capabilities
c=IN IP4 65.196.170.191
t=0 0
m=audio 11842 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20

2009-10-20 20:54:15.585206 [DEBUG] sofia_glue.c:1761
sip:fax_9186 at 10.16.5.129:5060 Setting proxy route to
sofia/rmt/fax_9186
2009-10-20 20:54:15.585206 [DEBUG] mod_sofia.c:111
(sofia/rmt/fax_9186) State Change CS_INIT -> CS_ROUTING
2009-10-20 20:54:15.585206 [DEBUG] switch_core_session.c:985 Send
signal sofia/rmt/fax_9186 [BREAK]
2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:330
(sofia/rmt/fax_9186) State INIT going to sleep
2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:306
(sofia/rmt/fax_9186) Running State Change CS_ROUTING
2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:333
(sofia/rmt/fax_9186) State ROUTING
2009-10-20 20:54:15.585206 [DEBUG] mod_sofia.c:130 sofia/rmt/fax_9186
SOFIA ROUTING
2009-10-20 20:54:15.585206 [DEBUG] switch_ivr_originate.c:66
(sofia/rmt/fax_9186) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2009-10-20 20:54:15.585206 [DEBUG] switch_core_session.c:985 Send
signal sofia/rmt/fax_9186 [BREAK]
2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:333
(sofia/rmt/fax_9186) State ROUTING going to sleep
2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:306
(sofia/rmt/fax_9186) Running State Change CS_CONSUME_MEDIA
2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:352
(sofia/rmt/fax_9186) State CONSUME_MEDIA
2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:352
(sofia/rmt/fax_9186) State CONSUME_MEDIA going to sleep
2009-10-20 20:54:15.585206 [DEBUG] sofia.c:3493 Channel
sofia/rmt/fax_9186 entering state [calling][0]
2009-10-20 20:54:15.601186 [DEBUG] sofia.c:3493 Channel
sofia/rmt/fax_9186 entering state [proceeding][180]
2009-10-20 20:54:15.601186 [NOTICE] sofia.c:3557 Ring-Ready sofia/rmt/fax_9186!
2009-10-20 20:54:15.606802 [DEBUG] switch_ivr_originate.c:388
sofia/s2s/+19412848354 at 65.196.170.129 receive message [RINGING]
2009-10-20 20:54:15.606802 [NOTICE] mod_sofia.c:1525 Ring-Ready
sofia/s2s/+19412848354 at 65.196.170.129!
2009-10-20 20:54:15.606802 [DEBUG] switch_core_session.c:653 Send
signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:15.606802 [NOTICE] switch_ivr_originate.c:388 Ring
Ready sofia/s2s/+19412848354 at 65.196.170.129!
2009-10-20 20:54:15.609957 [DEBUG] sofia.c:3493 Channel
sofia/s2s/+19412848354 at 65.196.170.129 entering state [early][180]
2009-10-20 20:54:16.169195 [DEBUG] sofia.c:3493 Channel
sofia/rmt/fax_9186 entering state [ready][200]
2009-10-20 20:54:16.169195 [DEBUG] sofia.c:3500 Remote SDP:
v=0
o=- 74979 74979 IN IP4 10.16.5.129
s=-
c=IN IP4 10.16.5.129
t=0 0
m=audio 16458 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

2009-10-20 20:54:16.169195 [DEBUG] switch_channel.c:2105 Send signal
sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:16.169195 [NOTICE] sofia.c:3948 Channel
[sofia/rmt/fax_9186] has been answered
2009-10-20 20:54:16.169195 [DEBUG] sofia_glue.c:2102 Set Codec
sofia/rmt/fax_9186 PROXY/8000 20 ms 160 samples
2009-10-20 20:54:16.175057 [DEBUG] sofia_glue.c:2379 PROXY AUDIO RTP
[sofia/rmt/fax_9186] 65.196.170.191:11842->10.16.5.129:16458 codec: 0
ms: 20
2009-10-20 20:54:16.175057 [DEBUG] switch_rtp.c:1163 Not using a timer
2009-10-20 20:54:16.175057 [DEBUG] sofia.c:3961
sofia/s2s/+19412848354 at 65.196.170.129 receive message [ANSWER]
2009-10-20 20:54:16.175057 [DEBUG] sofia_glue.c:1328
sofia/s2s/+19412848354 at 65.196.170.129 Patched SDP
---
v=0
o=- 74979 74979 IN IP4 10.16.5.129
s=-
c=IN IP4 10.16.5.129
t=0 0
m=audio 16458 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

+++
v=0
o=- 74979 74979 IN IP4 10.16.5.129
s=-
c=IN IP4 65.196.170.191
t=0 0
m=audio 11186 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

2009-10-20 20:54:16.175057 [DEBUG] sofia_glue.c:2102 Set Codec
sofia/s2s/+19412848354 at 65.196.170.129 PROXY/8000 20 ms 160 samples
2009-10-20 20:54:16.177257 [DEBUG] sofia_glue.c:2379 PROXY AUDIO RTP
[sofia/s2s/+19412848354 at 65.196.170.129]
65.196.170.191:11186->4.55.0.130:26636 codec: 0 ms: 20
2009-10-20 20:54:16.177257 [DEBUG] switch_rtp.c:1163 Not using a timer
2009-10-20 20:54:16.177257 [DEBUG] switch_ivr_originate.c:2324
sofia/s2s/+19412848354 at 65.196.170.129 receive message [ANSWER]
2009-10-20 20:54:16.182190 [DEBUG] switch_core_session.c:712 Send
signal sofia/rmt/fax_9186 [BREAK]
2009-10-20 20:54:16.182190 [DEBUG] switch_core_session.c:653 Send
signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:16.182190 [DEBUG] sofia.c:407 sofia/rmt/fax_9186
receive message [DISPLAY]
2009-10-20 20:54:16.182190 [DEBUG] switch_core_session.c:712 Send
signal sofia/rmt/fax_9186 [BREAK]
2009-10-20 20:54:16.182190 [DEBUG] switch_core_session.c:653 Send
signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:16.182190 [NOTICE] sofia.c:3961 Channel
[sofia/s2s/+19412848354 at 65.196.170.129] has been answered
2009-10-20 20:54:16.185443 [DEBUG] switch_channel.c:182
sofia/s2s/+19412848354 at 65.196.170.129 receive message [AUDIO_SYNC]
2009-10-20 20:54:16.185443 [DEBUG] switch_ivr_originate.c:2368
Originate Resulted in Success: [sofia/rmt/fax_9186]
2009-10-20 20:54:16.185443 [DEBUG] switch_channel.c:182
sofia/rmt/fax_9186 receive message [AUDIO_SYNC]
2009-10-20 20:54:16.185443 [DEBUG] switch_channel.c:182
sofia/s2s/+19412848354 at 65.196.170.129 receive message [AUDIO_SYNC]
2009-10-20 20:54:16.185443 [DEBUG] sofia.c:407 sofia/rmt/fax_9186
receive message [DISPLAY]
2009-10-20 20:54:16.185443 [DEBUG] mod_sofia.c:1328 Not sending same
id again "unknown" <fax_9186>
2009-10-20 20:54:16.185443 [DEBUG] switch_ivr_bridge.c:975
sofia/rmt/fax_9186 receive message [BRIDGE]
2009-10-20 20:54:16.185443 [DEBUG] switch_core_session.c:653 Send
signal sofia/rmt/fax_9186 [BREAK]
2009-10-20 20:54:16.185443 [DEBUG] sofia.c:3493 Channel
sofia/s2s/+19412848354 at 65.196.170.129 entering state [completed][200]
2009-10-20 20:54:16.185443 [DEBUG] switch_ivr_bridge.c:982
sofia/s2s/+19412848354 at 65.196.170.129 receive message [BRIDGE]
2009-10-20 20:54:16.185443 [DEBUG] switch_core_session.c:653 Send
signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:16.185443 [DEBUG] switch_ivr_bridge.c:1026
(sofia/rmt/fax_9186) State Change CS_CONSUME_MEDIA ->
CS_EXCHANGE_MEDIA
2009-10-20 20:54:16.185443 [DEBUG] switch_core_session.c:985 Send
signal sofia/rmt/fax_9186 [BREAK]
2009-10-20 20:54:16.185443 [DEBUG] switch_core_state_machine.c:306
(sofia/rmt/fax_9186) Running State Change CS_EXCHANGE_MEDIA
2009-10-20 20:54:16.185443 [DEBUG] switch_core_state_machine.c:343
(sofia/rmt/fax_9186) State EXCHANGE_MEDIA
2009-10-20 20:54:16.185443 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK
2009-10-20 20:54:16.242188 [DEBUG] sofia.c:3493 Channel
sofia/s2s/+19412848354 at 65.196.170.129 entering state [ready][200]
2009-10-20 20:54:16.310188 [DEBUG] switch_core_session.c:712 Send
signal sofia/rmt/fax_9186 [BREAK]
2009-10-20 20:54:16.310188 [DEBUG] switch_core_session.c:712 Send
signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:16.310188 [DEBUG] switch_ivr_bridge.c:122
sofia/rmt/fax_9186 receive message [DISPLAY]
2009-10-20 20:54:16.313084 [DEBUG] switch_ivr_bridge.c:122
sofia/s2s/+19412848354 at 65.196.170.129 receive message [DISPLAY]
2009-10-20 20:54:18.993185 [DEBUG] sofia.c:3493 Channel
sofia/rmt/fax_9186 entering state [received][100]
2009-10-20 20:54:18.993185 [DEBUG] sofia.c:3500 Remote SDP:
v=0
o=- 75318 75318 IN IP4 10.16.5.129
s=-
c=IN IP4 10.16.5.129
t=0 0
m=image 16458 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:200
a=T38FaxUdpEC:t38UDPRedundancy

2009-10-20 20:54:18.993185 [DEBUG] switch_core_session.c:861 Send
signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:18.993185 [DEBUG] switch_ivr_bridge.c:301
sofia/s2s/+19412848354 at 65.196.170.129 receive message [UNBRIDGE]
2009-10-20 20:54:18.993185 [DEBUG] switch_core_session.c:653 Send
signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:18.993185 [DEBUG] sofia.c:3493 Channel
sofia/rmt/fax_9186 entering state [completed][200]
2009-10-20 20:54:19.005192 [DEBUG] sofia.c:3493 Channel
sofia/rmt/fax_9186 entering state [ready][200]
2009-10-20 20:54:19.005192 [NOTICE] sofia.c:328 Hangup
sofia/rmt/fax_9186 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2009-10-20 20:54:19.005192 [DEBUG] switch_channel.c:1896 Send signal
sofia/rmt/fax_9186 [KILL]
2009-10-20 20:54:19.005192 [DEBUG] switch_core_session.c:985 Send
signal sofia/rmt/fax_9186 [BREAK]
2009-10-20 20:54:19.005192 [DEBUG] switch_core_state_machine.c:451
thread mismatch skipping state handler.
2009-10-20 20:54:19.009120 [DEBUG] switch_ivr_bridge.c:528 BRIDGE
THREAD DONE [sofia/rmt/fax_9186]
2009-10-20 20:54:19.009120 [DEBUG] switch_ivr_bridge.c:530 Send signal
sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:343
(sofia/rmt/fax_9186) State EXCHANGE_MEDIA going to sleep
2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:306
(sofia/rmt/fax_9186) Running State Change CS_HANGUP
2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:478
(sofia/rmt/fax_9186) State HANGUP
2009-10-20 20:54:19.009120 [DEBUG] mod_sofia.c:344 Channel
sofia/rmt/fax_9186 hanging up, cause: NORMAL_CLEARING
2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:46
sofia/rmt/fax_9186 Standard HANGUP, cause: NORMAL_CLEARING
2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:478
(sofia/rmt/fax_9186) State HANGUP going to sleep
2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:325
(sofia/rmt/fax_9186) State Change CS_HANGUP -> CS_REPORTING
2009-10-20 20:54:19.009120 [DEBUG] switch_core_session.c:985 Send
signal sofia/rmt/fax_9186 [BREAK]
2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:306
(sofia/rmt/fax_9186) Running State Change CS_REPORTING
2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:569
(sofia/rmt/fax_9186) State REPORTING
2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:53
sofia/rmt/fax_9186 Standard REPORTING, cause: NORMAL_CLEARING
2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:569
(sofia/rmt/fax_9186) State REPORTING going to sleep
2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:319
(sofia/rmt/fax_9186) State Change CS_REPORTING -> CS_DESTROY
2009-10-20 20:54:19.009120 [DEBUG] switch_core_session.c:985 Send
signal sofia/rmt/fax_9186 [BREAK]
2009-10-20 20:54:19.009120 [DEBUG] switch_core_session.c:1122 Session
16 (sofia/rmt/fax_9186) Locked, Waiting on external entities
2009-10-20 20:54:19.110187 [DEBUG] switch_ivr.c:540
sofia/s2s/+19412848354 at 65.196.170.129 Command Execute
playback(local_stream://moh)
EXECUTE sofia/s2s/+19412848354 at 65.196.170.129 playback(local_stream://moh)
2009-10-20 20:54:19.112927 [ERR] switch_core_file.c:116 Invalid file
format [local_stream] for [moh]!
2009-10-20 20:54:19.112927 [DEBUG] switch_ivr_bridge.c:304
sofia/s2s/+19412848354 at 65.196.170.129 receive message [BRIDGE]
2009-10-20 20:54:19.112927 [DEBUG] switch_core_session.c:653 Send
signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:19.112927 [DEBUG] switch_ivr_bridge.c:306 Send signal
sofia/rmt/fax_9186 [BREAK]
2009-10-20 20:54:19.112927 [DEBUG] switch_ivr_bridge.c:502
sofia/s2s/+19412848354 at 65.196.170.129 receive message [UNBRIDGE]
2009-10-20 20:54:19.112927 [DEBUG] switch_core_session.c:653 Send
signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:19.112927 [DEBUG] switch_ivr_bridge.c:528 BRIDGE
THREAD DONE [sofia/s2s/+19412848354 at 65.196.170.129]
2009-10-20 20:54:19.112927 [DEBUG] switch_ivr_bridge.c:530 Send signal
sofia/rmt/fax_9186 [BREAK]
2009-10-20 20:54:19.112927 [NOTICE] switch_core_state_machine.c:179
Hangup sofia/s2s/+19412848354 at 65.196.170.129 [CS_EXECUTE]
[NORMAL_CLEARING]
2009-10-20 20:54:19.112927 [DEBUG] switch_channel.c:1896 Send signal
sofia/s2s/+19412848354 at 65.196.170.129 [KILL]
2009-10-20 20:54:19.112927 [DEBUG] switch_core_session.c:985 Send
signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:478
(sofia/s2s/+19412848354 at 65.196.170.129) State HANGUP
2009-10-20 20:54:19.112927 [DEBUG] mod_sofia.c:344 Channel
sofia/s2s/+19412848354 at 65.196.170.129 hanging up, cause:
NORMAL_CLEARING
2009-10-20 20:54:19.112927 [DEBUG] mod_sofia.c:382 Sending BYE to
sofia/s2s/+19412848354 at 65.196.170.129
2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:46
sofia/s2s/+19412848354 at 65.196.170.129 Standard HANGUP, cause:
NORMAL_CLEARING
2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:478
(sofia/s2s/+19412848354 at 65.196.170.129) State HANGUP going to sleep
2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:340
(sofia/s2s/+19412848354 at 65.196.170.129) State EXECUTE going to sleep
2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:306
(sofia/s2s/+19412848354 at 65.196.170.129) Running State Change CS_HANGUP
2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:456
handler already called, skipping state handler.
2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:325
(sofia/s2s/+19412848354 at 65.196.170.129) State Change CS_HANGUP ->
CS_REPORTING
2009-10-20 20:54:19.112927 [DEBUG] switch_core_session.c:985 Send
signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:306
(sofia/s2s/+19412848354 at 65.196.170.129) Running State Change
CS_REPORTING
2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:569
(sofia/s2s/+19412848354 at 65.196.170.129) State REPORTING
2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:53
sofia/s2s/+19412848354 at 65.196.170.129 Standard REPORTING, cause:
NORMAL_CLEARING
2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:569
(sofia/s2s/+19412848354 at 65.196.170.129) State REPORTING going to sleep
2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:319
(sofia/s2s/+19412848354 at 65.196.170.129) State Change CS_REPORTING ->
CS_DESTROY
2009-10-20 20:54:19.112927 [DEBUG] switch_core_session.c:985 Send
signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
2009-10-20 20:54:19.112927 [DEBUG] switch_core_session.c:1122 Session
15 (sofia/s2s/+19412848354 at 65.196.170.129) Locked, Waiting on external
entities
2009-10-20 20:54:19.242188 [NOTICE] switch_core_session.c:1140 Session
16 (sofia/rmt/fax_9186) Ended
2009-10-20 20:54:19.242188 [NOTICE] switch_core_session.c:1142 Close
Channel sofia/rmt/fax_9186 [CS_DESTROY]
2009-10-20 20:54:19.242188 [DEBUG] switch_core_state_machine.c:415
(sofia/rmt/fax_9186) Running State Change CS_DESTROY
2009-10-20 20:54:19.245025 [DEBUG] switch_core_state_machine.c:426
(sofia/rmt/fax_9186) State DESTROY
2009-10-20 20:54:19.245025 [DEBUG] mod_sofia.c:261 sofia/rmt/fax_9186
SOFIA DESTROY
2009-10-20 20:54:19.245025 [DEBUG] switch_core_state_machine.c:60
sofia/rmt/fax_9186 Standard DESTROY
2009-10-20 20:54:19.245025 [DEBUG] switch_core_state_machine.c:426
(sofia/rmt/fax_9186) State DESTROY going to sleep
2009-10-20 20:54:19.245025 [NOTICE] switch_core_session.c:1140 Session
15 (sofia/s2s/+19412848354 at 65.196.170.129) Ended
2009-10-20 20:54:19.245025 [NOTICE] switch_core_session.c:1142 Close
Channel sofia/s2s/+19412848354 at 65.196.170.129 [CS_DESTROY]
2009-10-20 20:54:19.245025 [DEBUG] switch_core_state_machine.c:415
(sofia/s2s/+19412848354 at 65.196.170.129) Running State Change
CS_DESTROY
2009-10-20 20:54:19.245025 [DEBUG] switch_core_state_machine.c:426
(sofia/s2s/+19412848354 at 65.196.170.129) State DESTROY
2009-10-20 20:54:19.245025 [DEBUG] mod_sofia.c:261
sofia/s2s/+19412848354 at 65.196.170.129 SOFIA DESTROY
2009-10-20 20:54:19.245025 [DEBUG] switch_core_state_machine.c:60
sofia/s2s/+19412848354 at 65.196.170.129 Standard DESTROY
2009-10-20 20:54:19.245025 [DEBUG] switch_core_state_machine.c:426
(sofia/s2s/+19412848354 at 65.196.170.129) State DESTROY going to sleep


-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From anthony.minessale at gmail.com  Tue Oct 20 14:34:38 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 20 Oct 2009 16:34:38 -0500
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
In-Reply-To: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>
References: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>
Message-ID: <191c3a030910201434q4c6e65e1h58c5f0a82addc2d3@mail.gmail.com>

issue:
console loglevel debug
sofia profile internal siptrace on

and put it on pastebin
http://pastebin.freeswitch.org


On Tue, Oct 20, 2009 at 3:57 PM, Kristian Kielhofner <
kristian.kielhofner at gmail.com> wrote:

> Hello everyone,
>
>  Now that proxy media mode is working again I'm trying to figure out
> why T.38 with re-INVITE doesn't...
>
>  Everything goes well until my end tries to re-INVITE to T.38:
>
> U 10.16.5.129:5060 -> 65.196.170.191:5060
> INVITE sip:mod_sofia at 65.196.170.191:5060 SIP/2.0.
> Via: SIP/2.0/UDP 10.16.5.129:5060;branch=z9hG4bK-2656bddb.
> From: <sip:fax_9186 at 10.16.5.129:5060>;tag=9f75a042a3ca1f17i0.
> To: "WIRELESS CALLER" <sip:19412848354 at 65.196.170.191<sip%3A19412848354 at 65.196.170.191>
> >;tag=4eUH874j0eUZF.
> Remote-Party-ID: fax_9186 <sip:fax_9186 at 65.196.170.191<sip%3Afax_9186 at 65.196.170.191>
> >;screen=yes;party=called.
> Call-ID: 2889ab05-385d-122d-07b2-0014221c0392.
> CSeq: 101 INVITE.
> Max-Forwards: 70.
> Contact: fax_9186 <sip:fax_9186 at 10.16.5.129:5060>.
> Expires: 30.
> User-Agent: Linksys/SPA3102-5.1.10(GW).
> Content-Length: 261.
> Content-Type: application/sdp.
> .
> v=0.
> o=- 57670 57670 IN IP4 10.16.5.129.
> s=-.
> c=IN IP4 10.16.5.129.
> t=0 0.
> m=image 16454 udptl t38.
> a=T38FaxVersion:0.
> a=T38MaxBitRate:14400.
> a=T38FaxRateManagement:transferredTCF.
> a=T38FaxMaxBuffer:200.
> a=T38FaxMaxDatagram:200.
> a=T38FaxUdpEC:t38UDPRedundancy.
>
>
> U 65.196.170.191:5060 -> 10.16.5.129:5060
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP 10.16.5.129:5060;branch=z9hG4bK-2656bddb.
> From: <sip:fax_9186 at 10.16.5.129:5060>;tag=9f75a042a3ca1f17i0.
> To: "WIRELESS CALLER" <sip:19412848354 at 65.196.170.191<sip%3A19412848354 at 65.196.170.191>
> >;tag=4eUH874j0eUZF.
> Call-ID: 2889ab05-385d-122d-07b2-0014221c0392.
> CSeq: 101 INVITE.
> User-Agent: Star2Star Media.
> Content-Length: 0.
> .
>
> U 65.196.170.191:5060 -> 10.16.5.129:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 10.16.5.129:5060;branch=z9hG4bK-2656bddb.
> From: <sip:fax_9186 at 10.16.5.129:5060>;tag=9f75a042a3ca1f17i0.
> To: "WIRELESS CALLER" <sip:19412848354 at 65.196.170.191<sip%3A19412848354 at 65.196.170.191>
> >;tag=4eUH874j0eUZF.
> Call-ID: 2889ab05-385d-122d-07b2-0014221c0392.
> CSeq: 101 INVITE.
> Contact: <sip:mod_sofia at 65.196.170.191:5060>.
> User-Agent: Star2Star Media.
> Accept: application/sdp.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
> REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE.
> Supported: timer, precondition, path, replaces.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 242.
> .
> v=0.
> o=Sonus_UAC 881416742673069109 186840485790971511 IN IP4 65.196.170.129.
> s=SIP Media Capabilities.
> c=IN IP4 65.196.170.191.
> t=0 0.
> m=audio 19646 RTP/AVP 101.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=silenceSupp:off - - - -.
>
>
> U 10.16.5.129:5060 -> 65.196.170.191:5060
> ACK sip:mod_sofia at 65.196.170.191:5060 SIP/2.0.
> Via: SIP/2.0/UDP 10.16.5.129:5060;branch=z9hG4bK-842177a7.
> From: <sip:fax_9186 at 10.16.5.129:5060>;tag=9f75a042a3ca1f17i0.
> To: "WIRELESS CALLER" <sip:19412848354 at 65.196.170.191<sip%3A19412848354 at 65.196.170.191>
> >;tag=4eUH874j0eUZF.
> Call-ID: 2889ab05-385d-122d-07b2-0014221c0392.
> CSeq: 101 ACK.
> Max-Forwards: 70.
> Contact: fax_9186 <sip:fax_9186 at 10.16.5.129:5060>.
> User-Agent: Linksys/SPA3102-5.1.10(GW).
> Content-Length: 0.
> .
>
> U 10.16.5.129:5060 -> 65.196.170.191:5060
> BYE sip:mod_sofia at 65.196.170.191:5060 SIP/2.0.
> Via: SIP/2.0/UDP 10.16.5.129:5060;branch=z9hG4bK-9e6b284b.
> From: <sip:fax_9186 at 10.16.5.129:5060>;tag=9f75a042a3ca1f17i0.
> To: "WIRELESS CALLER" <sip:19412848354 at 65.196.170.191<sip%3A19412848354 at 65.196.170.191>
> >;tag=4eUH874j0eUZF.
> Call-ID: 2889ab05-385d-122d-07b2-0014221c0392.
> CSeq: 102 BYE.
> Max-Forwards: 70.
> User-Agent: Linksys/SPA3102-5.1.10(GW).
> Content-Length: 0.
> .
>
> U 65.196.170.191:5060 -> 10.16.5.129:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 10.16.5.129:5060;branch=z9hG4bK-9e6b284b.
> From: <sip:fax_9186 at 10.16.5.129:5060>;tag=9f75a042a3ca1f17i0.
> To: "WIRELESS CALLER" <sip:19412848354 at 65.196.170.191<sip%3A19412848354 at 65.196.170.191>
> >;tag=4eUH874j0eUZF.
> Call-ID: 2889ab05-385d-122d-07b2-0014221c0392.
> CSeq: 102 BYE.
> User-Agent: Star2Star Media.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
> REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE.
> Supported: timer, precondition, path, replaces.
> Content-Length: 0.
> .
>
> U 65.196.170.191:5080 -> 65.196.170.129:5060
> BYE sip:gw+sip0 at 65.196.170.129:5080;transport=udp SIP/2.0.
> Via: SIP/2.0/UDP 65.196.170.191:5080;rport;branch=z9hG4bKpv9pFF0S5r9ca.
> Route: <sip:65.196.170.129;lr=on;ftag=3D535BNy2tS9e;did=699.54d14392>.
> Max-Forwards: 70.
> From: <sip:+15673861009 at 65.196.170.129<sip%3A%2B15673861009 at 65.196.170.129>
> >;tag=p4H9tpXS1Qv8m.
> To: "WIRELESS CALLER" <sip:+19412848354 at 65.196.170.129<sip%3A%2B19412848354 at 65.196.170.129>
> >;tag=3D535BNy2tS9e.
> Call-ID: 2d4c8faa-385d-122d-a69b-003018ae0bd3.
> CSeq: 121917605 BYE.
> Contact: <sip:+15673861009 at 65.196.170.191:5080;transport=udp>.
> User-Agent: Star2Star Media.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
> REFER, UPDATE, NOTIFY.
> Supported: timer, precondition, path, replaces.
> Reason: Q.850;cause=16;text="NORMAL_CLEARING".
> Content-Length: 0.
> .
>
>  It's bizarre because the re-INVITE never gets transmitted to the
> other leg and FreeSWITCH responds with an SDP that only supports
> RFC2833.  Here is the FreeSWITCH level 7 log:
>
> 2009-10-20 20:54:15.577191 [NOTICE] switch_channel.c:613 New Channel
> sofia/s2s/+19412848354 at 65.196.170.129
> [ba5b3890-bdba-11de-b004-d7a35c351f6f]
> 2009-10-20 20:54:15.577191 [DEBUG] switch_core_state_machine.c:306
> (sofia/s2s/+19412848354 at 65.196.170.129) Running State Change CS_NEW
> 2009-10-20 20:54:15.577191 [DEBUG] sofia.c:3493 Channel
> sofia/s2s/+19412848354 at 65.196.170.129 entering state [received][100]
> 2009-10-20 20:54:15.577191 [DEBUG] sofia.c:3500 Remote SDP:
> v=0
> o=Sonus_UAC 7412556365376427063 4600708733824463060 IN IP4 65.196.170.129
> s=SIP Media Capabilities
> c=IN IP4 4.55.0.130
> t=0 0
> m=audio 26636 RTP/AVP 0 8 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=maxptime:20
>
> 2009-10-20 20:54:15.577191 [DEBUG] switch_core_state_machine.c:312
> (sofia/s2s/+19412848354 at 65.196.170.129) State NEW
> 2009-10-20 20:54:15.577191 [DEBUG] sofia.c:3615
> (sofia/s2s/+19412848354 at 65.196.170.129) State Change CS_NEW -> CS_INIT
> 2009-10-20 20:54:15.577191 [DEBUG] switch_core_session.c:985 Send
> signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:306
> (sofia/s2s/+19412848354 at 65.196.170.129) Running State Change CS_INIT
> 2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:330
> (sofia/s2s/+19412848354 at 65.196.170.129) State INIT
> 2009-10-20 20:54:15.581191 [DEBUG] mod_sofia.c:83
> sofia/s2s/+19412848354 at 65.196.170.129 SOFIA INIT
> 2009-10-20 20:54:15.581191 [DEBUG] mod_sofia.c:111
> (sofia/s2s/+19412848354 at 65.196.170.129) State Change CS_INIT ->
> CS_ROUTING
> 2009-10-20 20:54:15.581191 [DEBUG] switch_core_session.c:985 Send
> signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:330
> (sofia/s2s/+19412848354 at 65.196.170.129) State INIT going to sleep
> 2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:306
> (sofia/s2s/+19412848354 at 65.196.170.129) Running State Change
> CS_ROUTING
> 2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:333
> (sofia/s2s/+19412848354 at 65.196.170.129) State ROUTING
> 2009-10-20 20:54:15.581191 [DEBUG] mod_sofia.c:130
> sofia/s2s/+19412848354 at 65.196.170.129 SOFIA ROUTING
> 2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:78
> sofia/s2s/+19412848354 at 65.196.170.129 Standard ROUTING
> 2009-10-20 20:54:15.581191 [INFO] mod_dialplan_xml.c:391 Processing
> WIRELESS CALLER->fax_9186 in context s2s-in
> Dialplan: sofia/s2s/+19412848354 at 65.196.170.129 parsing
> [s2s-in->unloop] continue=false
> Dialplan: sofia/s2s/+19412848354 at 65.196.170.129 Regex (PASS) [unloop]
> ${unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/s2s/+19412848354 at 65.196.170.129 Regex (FAIL) [unloop]
> ${sip_looped_call}() =~ /^true$/ break=on-false
> Dialplan: sofia/s2s/+19412848354 at 65.196.170.129 parsing
> [s2s-in->faxpass] continue=false
> Dialplan: sofia/s2s/+19412848354 at 65.196.170.129 Absolute Condition
> [faxpass]
> Dialplan: sofia/s2s/+19412848354 at 65.196.170.129 Action
> bridge(sofia/rmt/fax_9186%rmt.star2star.com)
> 2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:114
> (sofia/s2s/+19412848354 at 65.196.170.129) State Change CS_ROUTING ->
> CS_EXECUTE
> 2009-10-20 20:54:15.581191 [DEBUG] switch_core_session.c:985 Send
> signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:333
> (sofia/s2s/+19412848354 at 65.196.170.129) State ROUTING going to sleep
> 2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:306
> (sofia/s2s/+19412848354 at 65.196.170.129) Running State Change
> CS_EXECUTE
> 2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:340
> (sofia/s2s/+19412848354 at 65.196.170.129) State EXECUTE
> 2009-10-20 20:54:15.581191 [DEBUG] mod_sofia.c:173
> sofia/s2s/+19412848354 at 65.196.170.129 SOFIA EXECUTE
> 2009-10-20 20:54:15.581191 [DEBUG] switch_core_state_machine.c:151
> sofia/s2s/+19412848354 at 65.196.170.129 Standard EXECUTE
> EXECUTE sofia/s2s/+19412848354 at 65.196.170.129
> bridge(sofia/rmt/fax_9186%rmt.star2star.com)
> 2009-10-20 20:54:15.585206 [NOTICE] switch_channel.c:613 New Channel
> sofia/rmt/fax_9186 [ba5baee2-bdba-11de-b004-d7a35c351f6f]
> 2009-10-20 20:54:15.585206 [DEBUG] mod_sofia.c:2981
> (sofia/rmt/fax_9186) State Change CS_NEW -> CS_INIT
> 2009-10-20 20:54:15.585206 [DEBUG] switch_core_session.c:985 Send
> signal sofia/rmt/fax_9186 [BREAK]
> 2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:306
> (sofia/rmt/fax_9186) Running State Change CS_INIT
> 2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:330
> (sofia/rmt/fax_9186) State INIT
> 2009-10-20 20:54:15.585206 [DEBUG] mod_sofia.c:83 sofia/rmt/fax_9186 SOFIA
> INIT
> 2009-10-20 20:54:15.585206 [DEBUG] sofia_glue.c:1328
> sofia/rmt/fax_9186 Patched SDP
> ---
> v=0
> o=Sonus_UAC 7412556365376427063 4600708733824463060 IN IP4 65.196.170.129
> s=SIP Media Capabilities
> c=IN IP4 4.55.0.130
> t=0 0
> m=audio 26636 RTP/AVP 0 8 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=maxptime:20
>
> +++
> v=0
> o=Sonus_UAC 7412556365376427063 4600708733824463060 IN IP4 65.196.170.129
> s=SIP Media Capabilities
> c=IN IP4 65.196.170.191
> t=0 0
> m=audio 11842 RTP/AVP 0 8 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=maxptime:20
>
> 2009-10-20 20:54:15.585206 [DEBUG] sofia_glue.c:1761
> sip:fax_9186 at 10.16.5.129:5060 Setting proxy route to
> sofia/rmt/fax_9186
> 2009-10-20 20:54:15.585206 [DEBUG] mod_sofia.c:111
> (sofia/rmt/fax_9186) State Change CS_INIT -> CS_ROUTING
> 2009-10-20 20:54:15.585206 [DEBUG] switch_core_session.c:985 Send
> signal sofia/rmt/fax_9186 [BREAK]
> 2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:330
> (sofia/rmt/fax_9186) State INIT going to sleep
> 2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:306
> (sofia/rmt/fax_9186) Running State Change CS_ROUTING
> 2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:333
> (sofia/rmt/fax_9186) State ROUTING
> 2009-10-20 20:54:15.585206 [DEBUG] mod_sofia.c:130 sofia/rmt/fax_9186
> SOFIA ROUTING
> 2009-10-20 20:54:15.585206 [DEBUG] switch_ivr_originate.c:66
> (sofia/rmt/fax_9186) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2009-10-20 20:54:15.585206 [DEBUG] switch_core_session.c:985 Send
> signal sofia/rmt/fax_9186 [BREAK]
> 2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:333
> (sofia/rmt/fax_9186) State ROUTING going to sleep
> 2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:306
> (sofia/rmt/fax_9186) Running State Change CS_CONSUME_MEDIA
> 2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:352
> (sofia/rmt/fax_9186) State CONSUME_MEDIA
> 2009-10-20 20:54:15.585206 [DEBUG] switch_core_state_machine.c:352
> (sofia/rmt/fax_9186) State CONSUME_MEDIA going to sleep
> 2009-10-20 20:54:15.585206 [DEBUG] sofia.c:3493 Channel
> sofia/rmt/fax_9186 entering state [calling][0]
> 2009-10-20 20:54:15.601186 [DEBUG] sofia.c:3493 Channel
> sofia/rmt/fax_9186 entering state [proceeding][180]
> 2009-10-20 20:54:15.601186 [NOTICE] sofia.c:3557 Ring-Ready
> sofia/rmt/fax_9186!
> 2009-10-20 20:54:15.606802 [DEBUG] switch_ivr_originate.c:388
> sofia/s2s/+19412848354 at 65.196.170.129 receive message [RINGING]
> 2009-10-20 20:54:15.606802 [NOTICE] mod_sofia.c:1525 Ring-Ready
> sofia/s2s/+19412848354 at 65.196.170.129!
> 2009-10-20 20:54:15.606802 [DEBUG] switch_core_session.c:653 Send
> signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:15.606802 [NOTICE] switch_ivr_originate.c:388 Ring
> Ready sofia/s2s/+19412848354 at 65.196.170.129!
> 2009-10-20 20:54:15.609957 [DEBUG] sofia.c:3493 Channel
> sofia/s2s/+19412848354 at 65.196.170.129 entering state [early][180]
> 2009-10-20 20:54:16.169195 [DEBUG] sofia.c:3493 Channel
> sofia/rmt/fax_9186 entering state [ready][200]
> 2009-10-20 20:54:16.169195 [DEBUG] sofia.c:3500 Remote SDP:
> v=0
> o=- 74979 74979 IN IP4 10.16.5.129
> s=-
> c=IN IP4 10.16.5.129
> t=0 0
> m=audio 16458 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
>
> 2009-10-20 20:54:16.169195 [DEBUG] switch_channel.c:2105 Send signal
> sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:16.169195 [NOTICE] sofia.c:3948 Channel
> [sofia/rmt/fax_9186] has been answered
> 2009-10-20 20:54:16.169195 [DEBUG] sofia_glue.c:2102 Set Codec
> sofia/rmt/fax_9186 PROXY/8000 20 ms 160 samples
> 2009-10-20 20:54:16.175057 [DEBUG] sofia_glue.c:2379 PROXY AUDIO RTP
> [sofia/rmt/fax_9186] 65.196.170.191:11842->10.16.5.129:16458 codec: 0
> ms: 20
> 2009-10-20 20:54:16.175057 [DEBUG] switch_rtp.c:1163 Not using a timer
> 2009-10-20 20:54:16.175057 [DEBUG] sofia.c:3961
> sofia/s2s/+19412848354 at 65.196.170.129 receive message [ANSWER]
> 2009-10-20 20:54:16.175057 [DEBUG] sofia_glue.c:1328
> sofia/s2s/+19412848354 at 65.196.170.129 Patched SDP
> ---
> v=0
> o=- 74979 74979 IN IP4 10.16.5.129
> s=-
> c=IN IP4 10.16.5.129
> t=0 0
> m=audio 16458 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
>
> +++
> v=0
> o=- 74979 74979 IN IP4 10.16.5.129
> s=-
> c=IN IP4 65.196.170.191
> t=0 0
> m=audio 11186 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
>
> 2009-10-20 20:54:16.175057 [DEBUG] sofia_glue.c:2102 Set Codec
> sofia/s2s/+19412848354 at 65.196.170.129 PROXY/8000 20 ms 160 samples
> 2009-10-20 20:54:16.177257 [DEBUG] sofia_glue.c:2379 PROXY AUDIO RTP
> [sofia/s2s/+19412848354 at 65.196.170.129]
> 65.196.170.191:11186->4.55.0.130:26636 codec: 0 ms: 20
> 2009-10-20 20:54:16.177257 [DEBUG] switch_rtp.c:1163 Not using a timer
> 2009-10-20 20:54:16.177257 [DEBUG] switch_ivr_originate.c:2324
> sofia/s2s/+19412848354 at 65.196.170.129 receive message [ANSWER]
> 2009-10-20 20:54:16.182190 [DEBUG] switch_core_session.c:712 Send
> signal sofia/rmt/fax_9186 [BREAK]
> 2009-10-20 20:54:16.182190 [DEBUG] switch_core_session.c:653 Send
> signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:16.182190 [DEBUG] sofia.c:407 sofia/rmt/fax_9186
> receive message [DISPLAY]
> 2009-10-20 20:54:16.182190 [DEBUG] switch_core_session.c:712 Send
> signal sofia/rmt/fax_9186 [BREAK]
> 2009-10-20 20:54:16.182190 [DEBUG] switch_core_session.c:653 Send
> signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:16.182190 [NOTICE] sofia.c:3961 Channel
> [sofia/s2s/+19412848354 at 65.196.170.129] has been answered
> 2009-10-20 20:54:16.185443 [DEBUG] switch_channel.c:182
> sofia/s2s/+19412848354 at 65.196.170.129 receive message [AUDIO_SYNC]
> 2009-10-20 20:54:16.185443 [DEBUG] switch_ivr_originate.c:2368
> Originate Resulted in Success: [sofia/rmt/fax_9186]
> 2009-10-20 20:54:16.185443 [DEBUG] switch_channel.c:182
> sofia/rmt/fax_9186 receive message [AUDIO_SYNC]
> 2009-10-20 20:54:16.185443 [DEBUG] switch_channel.c:182
> sofia/s2s/+19412848354 at 65.196.170.129 receive message [AUDIO_SYNC]
> 2009-10-20 20:54:16.185443 [DEBUG] sofia.c:407 sofia/rmt/fax_9186
> receive message [DISPLAY]
> 2009-10-20 20:54:16.185443 [DEBUG] mod_sofia.c:1328 Not sending same
> id again "unknown" <fax_9186>
> 2009-10-20 20:54:16.185443 [DEBUG] switch_ivr_bridge.c:975
> sofia/rmt/fax_9186 receive message [BRIDGE]
> 2009-10-20 20:54:16.185443 [DEBUG] switch_core_session.c:653 Send
> signal sofia/rmt/fax_9186 [BREAK]
> 2009-10-20 20:54:16.185443 [DEBUG] sofia.c:3493 Channel
> sofia/s2s/+19412848354 at 65.196.170.129 entering state [completed][200]
> 2009-10-20 20:54:16.185443 [DEBUG] switch_ivr_bridge.c:982
> sofia/s2s/+19412848354 at 65.196.170.129 receive message [BRIDGE]
> 2009-10-20 20:54:16.185443 [DEBUG] switch_core_session.c:653 Send
> signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:16.185443 [DEBUG] switch_ivr_bridge.c:1026
> (sofia/rmt/fax_9186) State Change CS_CONSUME_MEDIA ->
> CS_EXCHANGE_MEDIA
> 2009-10-20 20:54:16.185443 [DEBUG] switch_core_session.c:985 Send
> signal sofia/rmt/fax_9186 [BREAK]
> 2009-10-20 20:54:16.185443 [DEBUG] switch_core_state_machine.c:306
> (sofia/rmt/fax_9186) Running State Change CS_EXCHANGE_MEDIA
> 2009-10-20 20:54:16.185443 [DEBUG] switch_core_state_machine.c:343
> (sofia/rmt/fax_9186) State EXCHANGE_MEDIA
> 2009-10-20 20:54:16.185443 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK
> 2009-10-20 20:54:16.242188 [DEBUG] sofia.c:3493 Channel
> sofia/s2s/+19412848354 at 65.196.170.129 entering state [ready][200]
> 2009-10-20 20:54:16.310188 [DEBUG] switch_core_session.c:712 Send
> signal sofia/rmt/fax_9186 [BREAK]
> 2009-10-20 20:54:16.310188 [DEBUG] switch_core_session.c:712 Send
> signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:16.310188 [DEBUG] switch_ivr_bridge.c:122
> sofia/rmt/fax_9186 receive message [DISPLAY]
> 2009-10-20 20:54:16.313084 [DEBUG] switch_ivr_bridge.c:122
> sofia/s2s/+19412848354 at 65.196.170.129 receive message [DISPLAY]
> 2009-10-20 20:54:18.993185 [DEBUG] sofia.c:3493 Channel
> sofia/rmt/fax_9186 entering state [received][100]
> 2009-10-20 20:54:18.993185 [DEBUG] sofia.c:3500 Remote SDP:
> v=0
> o=- 75318 75318 IN IP4 10.16.5.129
> s=-
> c=IN IP4 10.16.5.129
> t=0 0
> m=image 16458 udptl t38
> a=T38FaxVersion:0
> a=T38MaxBitRate:14400
> a=T38FaxRateManagement:transferredTCF
> a=T38FaxMaxBuffer:200
> a=T38FaxMaxDatagram:200
> a=T38FaxUdpEC:t38UDPRedundancy
>
> 2009-10-20 20:54:18.993185 [DEBUG] switch_core_session.c:861 Send
> signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:18.993185 [DEBUG] switch_ivr_bridge.c:301
> sofia/s2s/+19412848354 at 65.196.170.129 receive message [UNBRIDGE]
> 2009-10-20 20:54:18.993185 [DEBUG] switch_core_session.c:653 Send
> signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:18.993185 [DEBUG] sofia.c:3493 Channel
> sofia/rmt/fax_9186 entering state [completed][200]
> 2009-10-20 20:54:19.005192 [DEBUG] sofia.c:3493 Channel
> sofia/rmt/fax_9186 entering state [ready][200]
> 2009-10-20 20:54:19.005192 [NOTICE] sofia.c:328 Hangup
> sofia/rmt/fax_9186 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
> 2009-10-20 20:54:19.005192 [DEBUG] switch_channel.c:1896 Send signal
> sofia/rmt/fax_9186 [KILL]
> 2009-10-20 20:54:19.005192 [DEBUG] switch_core_session.c:985 Send
> signal sofia/rmt/fax_9186 [BREAK]
> 2009-10-20 20:54:19.005192 [DEBUG] switch_core_state_machine.c:451
> thread mismatch skipping state handler.
> 2009-10-20 20:54:19.009120 [DEBUG] switch_ivr_bridge.c:528 BRIDGE
> THREAD DONE [sofia/rmt/fax_9186]
> 2009-10-20 20:54:19.009120 [DEBUG] switch_ivr_bridge.c:530 Send signal
> sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:343
> (sofia/rmt/fax_9186) State EXCHANGE_MEDIA going to sleep
> 2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:306
> (sofia/rmt/fax_9186) Running State Change CS_HANGUP
> 2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:478
> (sofia/rmt/fax_9186) State HANGUP
> 2009-10-20 20:54:19.009120 [DEBUG] mod_sofia.c:344 Channel
> sofia/rmt/fax_9186 hanging up, cause: NORMAL_CLEARING
> 2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:46
> sofia/rmt/fax_9186 Standard HANGUP, cause: NORMAL_CLEARING
> 2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:478
> (sofia/rmt/fax_9186) State HANGUP going to sleep
> 2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:325
> (sofia/rmt/fax_9186) State Change CS_HANGUP -> CS_REPORTING
> 2009-10-20 20:54:19.009120 [DEBUG] switch_core_session.c:985 Send
> signal sofia/rmt/fax_9186 [BREAK]
> 2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:306
> (sofia/rmt/fax_9186) Running State Change CS_REPORTING
> 2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:569
> (sofia/rmt/fax_9186) State REPORTING
> 2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:53
> sofia/rmt/fax_9186 Standard REPORTING, cause: NORMAL_CLEARING
> 2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:569
> (sofia/rmt/fax_9186) State REPORTING going to sleep
> 2009-10-20 20:54:19.009120 [DEBUG] switch_core_state_machine.c:319
> (sofia/rmt/fax_9186) State Change CS_REPORTING -> CS_DESTROY
> 2009-10-20 20:54:19.009120 [DEBUG] switch_core_session.c:985 Send
> signal sofia/rmt/fax_9186 [BREAK]
> 2009-10-20 20:54:19.009120 [DEBUG] switch_core_session.c:1122 Session
> 16 (sofia/rmt/fax_9186) Locked, Waiting on external entities
> 2009-10-20 20:54:19.110187 [DEBUG] switch_ivr.c:540
> sofia/s2s/+19412848354 at 65.196.170.129 Command Execute
> playback(local_stream://moh)
> EXECUTE sofia/s2s/+19412848354 at 65.196.170.129 playback(local_stream://moh)
> 2009-10-20 20:54:19.112927 [ERR] switch_core_file.c:116 Invalid file
> format [local_stream] for [moh]!
> 2009-10-20 20:54:19.112927 [DEBUG] switch_ivr_bridge.c:304
> sofia/s2s/+19412848354 at 65.196.170.129 receive message [BRIDGE]
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_session.c:653 Send
> signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:19.112927 [DEBUG] switch_ivr_bridge.c:306 Send signal
> sofia/rmt/fax_9186 [BREAK]
> 2009-10-20 20:54:19.112927 [DEBUG] switch_ivr_bridge.c:502
> sofia/s2s/+19412848354 at 65.196.170.129 receive message [UNBRIDGE]
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_session.c:653 Send
> signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:19.112927 [DEBUG] switch_ivr_bridge.c:528 BRIDGE
> THREAD DONE [sofia/s2s/+19412848354 at 65.196.170.129]
> 2009-10-20 20:54:19.112927 [DEBUG] switch_ivr_bridge.c:530 Send signal
> sofia/rmt/fax_9186 [BREAK]
> 2009-10-20 20:54:19.112927 [NOTICE] switch_core_state_machine.c:179
> Hangup sofia/s2s/+19412848354 at 65.196.170.129 [CS_EXECUTE]
> [NORMAL_CLEARING]
> 2009-10-20 20:54:19.112927 [DEBUG] switch_channel.c:1896 Send signal
> sofia/s2s/+19412848354 at 65.196.170.129 [KILL]
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_session.c:985 Send
> signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:478
> (sofia/s2s/+19412848354 at 65.196.170.129) State HANGUP
> 2009-10-20 20:54:19.112927 [DEBUG] mod_sofia.c:344 Channel
> sofia/s2s/+19412848354 at 65.196.170.129 hanging up, cause:
> NORMAL_CLEARING
> 2009-10-20 20:54:19.112927 [DEBUG] mod_sofia.c:382 Sending BYE to
> sofia/s2s/+19412848354 at 65.196.170.129
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:46
> sofia/s2s/+19412848354 at 65.196.170.129 Standard HANGUP, cause:
> NORMAL_CLEARING
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:478
> (sofia/s2s/+19412848354 at 65.196.170.129) State HANGUP going to sleep
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:340
> (sofia/s2s/+19412848354 at 65.196.170.129) State EXECUTE going to sleep
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:306
> (sofia/s2s/+19412848354 at 65.196.170.129) Running State Change CS_HANGUP
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:456
> handler already called, skipping state handler.
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:325
> (sofia/s2s/+19412848354 at 65.196.170.129) State Change CS_HANGUP ->
> CS_REPORTING
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_session.c:985 Send
> signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:306
> (sofia/s2s/+19412848354 at 65.196.170.129) Running State Change
> CS_REPORTING
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:569
> (sofia/s2s/+19412848354 at 65.196.170.129) State REPORTING
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:53
> sofia/s2s/+19412848354 at 65.196.170.129 Standard REPORTING, cause:
> NORMAL_CLEARING
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:569
> (sofia/s2s/+19412848354 at 65.196.170.129) State REPORTING going to sleep
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_state_machine.c:319
> (sofia/s2s/+19412848354 at 65.196.170.129) State Change CS_REPORTING ->
> CS_DESTROY
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_session.c:985 Send
> signal sofia/s2s/+19412848354 at 65.196.170.129 [BREAK]
> 2009-10-20 20:54:19.112927 [DEBUG] switch_core_session.c:1122 Session
> 15 (sofia/s2s/+19412848354 at 65.196.170.129) Locked, Waiting on external
> entities
> 2009-10-20 20:54:19.242188 [NOTICE] switch_core_session.c:1140 Session
> 16 (sofia/rmt/fax_9186) Ended
> 2009-10-20 20:54:19.242188 [NOTICE] switch_core_session.c:1142 Close
> Channel sofia/rmt/fax_9186 [CS_DESTROY]
> 2009-10-20 20:54:19.242188 [DEBUG] switch_core_state_machine.c:415
> (sofia/rmt/fax_9186) Running State Change CS_DESTROY
> 2009-10-20 20:54:19.245025 [DEBUG] switch_core_state_machine.c:426
> (sofia/rmt/fax_9186) State DESTROY
> 2009-10-20 20:54:19.245025 [DEBUG] mod_sofia.c:261 sofia/rmt/fax_9186
> SOFIA DESTROY
> 2009-10-20 20:54:19.245025 [DEBUG] switch_core_state_machine.c:60
> sofia/rmt/fax_9186 Standard DESTROY
> 2009-10-20 20:54:19.245025 [DEBUG] switch_core_state_machine.c:426
> (sofia/rmt/fax_9186) State DESTROY going to sleep
> 2009-10-20 20:54:19.245025 [NOTICE] switch_core_session.c:1140 Session
> 15 (sofia/s2s/+19412848354 at 65.196.170.129) Ended
> 2009-10-20 20:54:19.245025 [NOTICE] switch_core_session.c:1142 Close
> Channel sofia/s2s/+19412848354 at 65.196.170.129 [CS_DESTROY]
> 2009-10-20 20:54:19.245025 [DEBUG] switch_core_state_machine.c:415
> (sofia/s2s/+19412848354 at 65.196.170.129) Running State Change
> CS_DESTROY
> 2009-10-20 20:54:19.245025 [DEBUG] switch_core_state_machine.c:426
> (sofia/s2s/+19412848354 at 65.196.170.129) State DESTROY
> 2009-10-20 20:54:19.245025 [DEBUG] mod_sofia.c:261
> sofia/s2s/+19412848354 at 65.196.170.129 SOFIA DESTROY
> 2009-10-20 20:54:19.245025 [DEBUG] switch_core_state_machine.c:60
> sofia/s2s/+19412848354 at 65.196.170.129 Standard DESTROY
> 2009-10-20 20:54:19.245025 [DEBUG] switch_core_state_machine.c:426
> (sofia/s2s/+19412848354 at 65.196.170.129) State DESTROY going to sleep
>
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From brian at freeswitch.org  Tue Oct 20 14:35:17 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 20 Oct 2009 16:35:17 -0500
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
In-Reply-To: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>
References: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>
Message-ID: <6F7186AC-AF0F-4594-8555-7071A758AF39@freeswitch.org>

Fix this or set it to silence.

/b

On Oct 20, 2009, at 3:57 PM, Kristian Kielhofner wrote:

> EXECUTE sofia/s2s/+19412848354 at 65.196.170.129 playback 
> (local_stream://moh)
> 2009-10-20 20:54:19.112927 [ERR] switch_core_file.c:116 Invalid file
> format [local_stream] for [moh]!




From mcampbellsmith at gmail.com  Tue Oct 20 16:07:32 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Wed, 21 Oct 2009 10:07:32 +1100
Subject: [Freeswitch-users] 2 voicemail questions
In-Reply-To: <33c87fa30907100257w29c628cbnb641d09a75bc22b8@mail.gmail.com>
References: <33c87fa30907100257w29c628cbnb641d09a75bc22b8@mail.gmail.com>
Message-ID: <33c87fa30910201607x16eeaf0ek823084bbf14dbb6c@mail.gmail.com>

1. Can I email the voicemail message to multiple email addresses?

I revisited this again after not requiring it for a while.  A comma
separated list in the extension.xml file does work.  The problem was
the template file.  Once I removed < and > in the To: field, I can
send to multple emails no problems:

In 1000.xml (for example)
      <param name="vm-mailto" value="email at number1.com,email at number2.com"/>

And then I editied voicemail.tpl and changed the second line from:
To: <${voicemail_email}>

TO:
To: ${voicemail_email}

Not sure if this is a fault in the mail server or not.  Maybe the
default templates should be changed to handle this?


On Fri, Jul 10, 2009 at 8:57 PM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> Hi!
>>> 1. Can I email the voicemail message to multiple email addresses?
> A comma separated list does not work in the extensions.xml file
> (1000.xml), but it does work if I hard code the email addresses into
> the notify-voicemail.tpl file.
>
> Could this be added to the switch so that it can handle comma separated lists?
>
>>> 2. How can I make Freeswitch dial a number AFTER a voicemail is left?
>
>>api Hangup hook?
>
> i want the 'voicemail' application to appear to call the extension to
> notify the user that there is a waiting message. ?This is an extract
> from my dialplan.xml:
>
> ? ? ?<condition field="destination_number" expression="^(10[01][0-9])$">
> ? ? ? ?<action application="set" data="dialed_extension=$1"/>
> ? ? ? ?<action application="export" data="dialed_extension=$1"/>
> ? ? ? ?<!-- <action application="set" data="ringback=${us-ring}"/> -->
> ? ? ? ?<action application="set" data="call_timeout=30"/>
> ? ? ? ?<action application="set" data="hangup_after_bridge=true"/>
> ? ? ? ?<action application="set" data="continue_on_fail=true"/>
> ? ? ? ?<action application="bridge"
> data="user/${dialed_extension}@${domain_name}"/>
> ? ? ? ?<action application="answer"/>
> ? ? ? ?<action application="sleep" data="1000"/>
> ? ? ? ?<action application="voicemail" data="default ${domain_name}
> ${dialed_extension}"/>
> ? ? ?</condition>
> ? ? ?<condition
> field="${vm_boxcount(${destination_number}@${domain_name})}"
> expression="^(1)$">
> ? ? ? ? <action application="log" data="MSGBOX
> ${vm_boxcount(${dialed_extension}@${domain_name})}"/>
> ? ? ? ?<action application="set" data="api_hangup_hook=originate
> sofia/internal_nat/${dialed_etension}%${domain_name} default default
> Message 4000 4000 3"/>
>
> This only works if the B leg (ie voicemail application) hangs up
> first. ?This would be an unusual situation and does not achieve what I
> want... is there any other way to achieve this?
>
> Thanks
>
>> Hi!
>>
>> I have 2 questions regarding voicemail ...
>>
>> 1. Can I email the voicemail message to multiple email addresses? ?If
>> so, what format is this in?
>> ? ? ?<param name="vm-mailto" value="me at myemail.com"/>
>
> Try a comma sep. list. ?Not sure if it will work.
>
>>
>> 2. How can I make Freeswitch dial a number AFTER a voicemail is left?
>
> api Hangup hook?
>
> I g
>> From: Brian West <brian at freeswitch.org>
>
>> On Jul 7, 2009, at 9:11 AM, Mark Campbell-Smith wrote:
>
>> Hi!
>>
>> I have 2 questions regarding voicemail ...
>>
>> 1. Can I email the voicemail message to multiple email addresses? ?If
>> so, what format is this in?
>> ? ? ?<param name="vm-mailto" value="me at myemail.com"/>
>
>> Try a comma sep. list. ?Not sure if it will work.
>



From msc at freeswitch.org  Tue Oct 20 17:46:27 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 20 Oct 2009 17:46:27 -0700
Subject: [Freeswitch-users] "Can not record session. Media not enabled
	on channel."
In-Reply-To: <1256034659005-3857858.post@n2.nabble.com>
References: <1256034659005-3857858.post@n2.nabble.com>
Message-ID: <87f2f3b90910201746h43a31b32m5c163fd98373f915@mail.gmail.com>

What's in the dialplan for this channel? Is bypass-media or proxy-media set
to true? Do a debug trace and post it in pastebin.
-MC

On Tue, Oct 20, 2009 at 3:30 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> Hello,
> I am using the same set of extensions for testing the system during
> development, they include XLite, Cisco sip phone and several extensions
> that
> just play some audio file.
> Sometimes, very rarely, this message "Can not record session.  Media not
> enabled on channel." appears on FS console.
> Like I wrote before, I don't change any codec settings and always use the
> same set of devices/emulators.
> What can cause this message?
>
> Thanks,
> Maciej Aniserowicz
>
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From mike at jerris.com  Tue Oct 20 18:48:56 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 20 Oct 2009 21:48:56 -0400
Subject: [Freeswitch-users] CS_REPORTING Channel event state
In-Reply-To: <8ccbff060910200847y5e7ccab6o84d431836970bcb4@mail.gmail.com>
References: <8ccbff060910200847y5e7ccab6o84d431836970bcb4@mail.gmail.com>
Message-ID: <08C886FF-6D2E-46CD-AC7D-2AF37F18B170@jerris.com>

REPORTING is the state that it writes to CDR.  If you have calls stuck  
in this state, take one and try to use uuid_kill on it and see if it  
goes away, then get a core off of it and pastebin the thread apply all  
bt (with no other calls up).  What modules are you using for cdr and  
with what configuration?

Mike

On Oct 20, 2009, at 11:47 AM, Dome Charoenyost wrote:

> Dear All
> What's CS_REPORTING state ?
> I found many channels not hang up  ans state is CS_REPORTING
> e264f84a-bd87-11de-9a90-2320c02172de,outbound,2009-10-20
> 21:50:26,1256050226,sofia/external/xxxxxxxxxxxx at xxx.xxx.xx. 
> 191:7050,CS_REPORTING,FreeSWITCH,xxxxxxxxxxxxxxxxxxxx,xx.xxx.xxx.xxx, 
> 7050,,,XML,public,,,,,




From mike at jerris.com  Tue Oct 20 18:56:32 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 20 Oct 2009 21:56:32 -0400
Subject: [Freeswitch-users] Connect PHP SOAP Web Server with SQLite
	database of FS
In-Reply-To: <200910201423.35419.chris@cloudtel.com>
References: <25976554.post@talk.nabble.com>
	<200910201423.35419.chris@cloudtel.com>
Message-ID: <A5C02801-C96A-411B-A6D1-55D6D28617DA@jerris.com>

If you really want to access this information outside I would strongly  
recommend using odbc instead of the internal sqlite db, it does not  
handle locking contention well.  If you need access to things in the  
core db (like show calls and show channels information) you will need  
to write a small daemon that listens on events socket and puts that  
information into a database.

Mike

On Oct 20, 2009, at 2:23 PM, Chris Burns wrote:

> If you really wanted: http://php.net/manual/en/book.sqlite.php
>
> But I would recommend you make use of ODBC to use a client/server  
> RDBMS.
> Here's some good reading:
> http://www.sqlite.org/cvstrac/wiki?p=WhenToUseSqlite
>
> On October 20, 2009 10:53:01 am homqua wrote:
>> Now I am building a PHP SOAP Web Service to access the database of  
>> FS.
>> Anyone has idea about how to access sqlite database of FS through  
>> PHP ? I
>> have read about socket event in FS, but I don't know whether it can
>> response with the query of database or not.
>> Thanks for your help.
>
>




From kristian.kielhofner at gmail.com  Tue Oct 20 19:37:01 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Tue, 20 Oct 2009 22:37:01 -0400
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
In-Reply-To: <6F7186AC-AF0F-4594-8555-7071A758AF39@freeswitch.org>
References: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>
	<6F7186AC-AF0F-4594-8555-7071A758AF39@freeswitch.org>
Message-ID: <2d9149cd0910201937t4ab718b1yfbeb2a1f3d33f893@mail.gmail.com>

Brian,

  It's already set to silence :).

  My guess here is that Sofia is mis-interpreting the re-INVITE as not
having any media (a la RFC 2543 hold).

  I don't want to place the call on hold.  I want to negotiate T.38 ;).

On Tue, Oct 20, 2009 at 5:35 PM, Brian West <brian at freeswitch.org> wrote:
> Fix this or set it to silence.
>
> /b
>
> On Oct 20, 2009, at 3:57 PM, Kristian Kielhofner wrote:
>
>> EXECUTE sofia/s2s/+19412848354 at 65.196.170.129 playback
>> (local_stream://moh)
>> 2009-10-20 20:54:19.112927 [ERR] switch_core_file.c:116 Invalid file
>> format [local_stream] for [moh]!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From kristian.kielhofner at gmail.com  Tue Oct 20 19:37:50 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Tue, 20 Oct 2009 22:37:50 -0400
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
In-Reply-To: <191c3a030910201434q4c6e65e1h58c5f0a82addc2d3@mail.gmail.com>
References: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>
	<191c3a030910201434q4c6e65e1h58c5f0a82addc2d3@mail.gmail.com>
Message-ID: <2d9149cd0910201937v282d7ae9ve137ac396cfe81f3@mail.gmail.com>

Anthony,

  I will the next chance I get.

Thanks!

On Tue, Oct 20, 2009 at 5:34 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> issue:
> console loglevel debug
> sofia profile internal siptrace on
>
> and put it on pastebin
> http://pastebin.freeswitch.org
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From kristian.kielhofner at gmail.com  Tue Oct 20 20:50:55 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Tue, 20 Oct 2009 23:50:55 -0400
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
In-Reply-To: <191c3a030910201434q4c6e65e1h58c5f0a82addc2d3@mail.gmail.com>
References: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>
	<191c3a030910201434q4c6e65e1h58c5f0a82addc2d3@mail.gmail.com>
Message-ID: <2d9149cd0910202050i36a7ea82nc7253d87e38f990a@mail.gmail.com>

Done.

On Tue, Oct 20, 2009 at 5:34 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> issue:
> console loglevel debug
> sofia profile internal siptrace on
>
> and put it on pastebin
> http://pastebin.freeswitch.org
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From jason at jasonjgw.net  Tue Oct 20 21:03:43 2009
From: jason at jasonjgw.net (Jason White)
Date: Wed, 21 Oct 2009 15:03:43 +1100
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <7b8592a0910200947k6ca5a391l35786f0e855ae388@mail.gmail.com>
References: <5B924C57-B658-44C0-A1CE-00A3ECB5AA07@freeswitch.org>
	<7b8592a0910191351g14cbb47dj3f19362ff6197157@mail.gmail.com>
	<FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
	<7b8592a0910192253l1313d82j51c933cb5f55ad41@mail.gmail.com>
	<20091020133437.GA4941@dan.olp.net>
	<7b8592a0910200716q4a7db93cyb955595e4993ac96@mail.gmail.com>
	<858EC056-1D84-4F30-A0B7-9A70FF98695D@freeswitch.org>
	<7b8592a0910200821m18fe9b4dg73dda05a10699af2@mail.gmail.com>
	<642B6CED-CDF3-4D40-B493-0F2D6222C30B@freeswitch.org>
	<7b8592a0910200947k6ca5a391l35786f0e855ae388@mail.gmail.com>
Message-ID: <20091021040343.GA9440@jdc.jasonjgw.net>

ineya ineya <ineyaa at gmail.com> wrote:
> Hmm, I didn't about this.
> 
> so in directory/default.xml I have:
>  <param name="dial-string"
> value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
> 
> and the modified version for IPv6 would be ?...
>  <param name="dial-string"
> value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(internal-ipv6/${dialed_user}@${dialed_domain})}"/>

Yes. Are you suggesting it didn't work?




From q.edward at gmail.com  Tue Oct 20 21:20:28 2009
From: q.edward at gmail.com (Edward Q.)
Date: Wed, 21 Oct 2009 00:20:28 -0400
Subject: [Freeswitch-users] Hi Guys
Message-ID: <89313a90910202120v6c563660s61f2efba22f916f@mail.gmail.com>

Hi to all of you guys (and ladies too).
I would like to bother all of you and take just a little of your time.
I am a total NOOB on freeswitch and would like a little help.
I need to know from all of you guys which would be the hardware recommended
from all of you guys to be able to deal with a DS3.
The thing is i have a CANTATA switch already with the DS3 and making calls.
But i will have to communicate the company international headquarters
through there which in turn there are 21 offices around the world and at a
certain time all of them will be passing from freeswitch server to the land
lines. So could you pleae help out here.
Thanks to all of you and the ones who take the time to reply.
Ed

PS All the calls will be comming from the web through SIP protocol into the
freeswitch going out to cell phones and land lines for the executives to
handle the calls.
Call flow as this.  A sip client on the webserver connects to the
freeswitch, registers and makes the call to the support agent or the
executive. Thats what we are implementing.
So i guess the web server has to be another little animal to handle the
traffic too. Thanks again.
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From sicfslist at gmail.com  Tue Oct 20 22:55:19 2009
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Wed, 21 Oct 2009 00:55:19 -0500
Subject: [Freeswitch-users] Hi Guys
In-Reply-To: <89313a90910202120v6c563660s61f2efba22f916f@mail.gmail.com>
References: <89313a90910202120v6c563660s61f2efba22f916f@mail.gmail.com>
Message-ID: <35b355e90910202255x680232f9x45319ab835f5846b@mail.gmail.com>

Question:

Will the call flow look like this (above was not very clear):

web --> FS --> Cantata --> PSTN (via TDM circuits)

Or are you trying to replace the Cantata?

SDR
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From nagalenoj at gmail.com  Tue Oct 20 23:06:31 2009
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Wed, 21 Oct 2009 11:36:31 +0530
Subject: [Freeswitch-users] UUID of the newly originated call?
Message-ID: <fe67a3340910202306i55a0b9ceve92ab19e000edf11@mail.gmail.com>

I've tried with origination_uuid.

First, I tried with SIP and my program executes successfully as what I
expected. This program initiates a new call when a call comes and let the
new call to eavesdrop the landed call.
When, I experimented with PRI(openzap), I'm facing the following error. And,
I am unable to make any calls(even from CLI). It is reporting the same error
for the subsequent calls.
Error:        2009-10-21 11:08:31.661252 [ERR] mod_openzap.c:1760 Session
for channel 1:1 not found [UUID: 8364eb30-be03-11de-aa07-07e76b7651a9]

Full log is in http://pastebin.freeswitch.org/10780
My script is here, http://pastebin.freeswitch.org/10781

What is this error for and how to avoid this?

Is there any other way to get the uuid of the originated call except
explicitly defining(origination_uuid).?!

-- 
Regards,
Nagalenoj H.
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From q.edward at gmail.com  Tue Oct 20 23:09:12 2009
From: q.edward at gmail.com (Edward Q.)
Date: Wed, 21 Oct 2009 02:09:12 -0400
Subject: [Freeswitch-users] Hi Guys
In-Reply-To: <35b355e90910202255x680232f9x45319ab835f5846b@mail.gmail.com>
References: <89313a90910202120v6c563660s61f2efba22f916f@mail.gmail.com> 
	<35b355e90910202255x680232f9x45319ab835f5846b@mail.gmail.com>
Message-ID: <89313a90910202309r6751b242q6b96a56cd1bfe57@mail.gmail.com>

It will flow exactly like that My friends.Thank you for your help and
clarifying that ..
Ed

On Wed, Oct 21, 2009 at 1:55 AM, Shelby Ramsey <sicfslist at gmail.com> wrote:

> Question:
>
> Will the call flow look like this (above was not very clear):
>
> web --> FS --> Cantata --> PSTN (via TDM circuits)
>
> Or are you trying to replace the Cantata?
>
> SDR
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ineyaa at gmail.com  Tue Oct 20 23:36:10 2009
From: ineyaa at gmail.com (ineya ineya)
Date: Wed, 21 Oct 2009 08:36:10 +0200
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <20091021040343.GA9440@jdc.jasonjgw.net>
References: <5B924C57-B658-44C0-A1CE-00A3ECB5AA07@freeswitch.org>
	<FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
	<7b8592a0910192253l1313d82j51c933cb5f55ad41@mail.gmail.com>
	<20091020133437.GA4941@dan.olp.net>
	<7b8592a0910200716q4a7db93cyb955595e4993ac96@mail.gmail.com>
	<858EC056-1D84-4F30-A0B7-9A70FF98695D@freeswitch.org>
	<7b8592a0910200821m18fe9b4dg73dda05a10699af2@mail.gmail.com>
	<642B6CED-CDF3-4D40-B493-0F2D6222C30B@freeswitch.org>
	<7b8592a0910200947k6ca5a391l35786f0e855ae388@mail.gmail.com>
	<20091021040343.GA9440@jdc.jasonjgw.net>
Message-ID: <7b8592a0910202336x1bf2a8a2nd636219ec7031c5a@mail.gmail.com>

Sort of. I have different error now:

2009-10-21 08:34:08.446939 [NOTICE] mod_sofia.c:1521 Pre-Answer
sofia/internal-ipv6/1006 at franta.openstage.net!
2009-10-21 08:34:08.462967 [NOTICE] sofia.c:3925 Hangup
sofia/internal-ipv6/sip:1007@[2000:2::1002]:5062 [CS_CONSUME_MEDIA]
[INCOMPATIBLE_DESTINATION]

Here is the complete log:
http://pastebin.freeswitch.org/10782

> Yes. Are you suggesting it didn't work?
>
>



From sicfslist at gmail.com  Tue Oct 20 23:49:55 2009
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Wed, 21 Oct 2009 01:49:55 -0500
Subject: [Freeswitch-users] Hi Guys
In-Reply-To: <89313a90910202309r6751b242q6b96a56cd1bfe57@mail.gmail.com>
References: <89313a90910202120v6c563660s61f2efba22f916f@mail.gmail.com>
	<35b355e90910202255x680232f9x45319ab835f5846b@mail.gmail.com>
	<89313a90910202309r6751b242q6b96a56cd1bfe57@mail.gmail.com>
Message-ID: <35b355e90910202349q31bc8f7as6e277008600d81b0@mail.gmail.com>

Well ... the question is pretty generic.

But based on these assumptions:
  -- no media (bypass media)
  -- routing done via XML dialplan

Something along the lines of a quad core machine with 4 gigs of ram would be
overkill for 692 calls.

Things to remember:
  -- the more cores the better (FS is heavily threaded)
  -- the more memory the better
  -- 64 bit is way better than 32 bit

SDR





>
>
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From jason at jasonjgw.net  Tue Oct 20 23:58:41 2009
From: jason at jasonjgw.net (Jason White)
Date: Wed, 21 Oct 2009 17:58:41 +1100
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <7b8592a0910202336x1bf2a8a2nd636219ec7031c5a@mail.gmail.com>
References: <FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
	<7b8592a0910192253l1313d82j51c933cb5f55ad41@mail.gmail.com>
	<20091020133437.GA4941@dan.olp.net>
	<7b8592a0910200716q4a7db93cyb955595e4993ac96@mail.gmail.com>
	<858EC056-1D84-4F30-A0B7-9A70FF98695D@freeswitch.org>
	<7b8592a0910200821m18fe9b4dg73dda05a10699af2@mail.gmail.com>
	<642B6CED-CDF3-4D40-B493-0F2D6222C30B@freeswitch.org>
	<7b8592a0910200947k6ca5a391l35786f0e855ae388@mail.gmail.com>
	<20091021040343.GA9440@jdc.jasonjgw.net>
	<7b8592a0910202336x1bf2a8a2nd636219ec7031c5a@mail.gmail.com>
Message-ID: <20091021065841.GA13150@jdc.jasonjgw.net>

ineya ineya <ineyaa at gmail.com> wrote:
> Sort of. I have different error now:
> 
> 2009-10-21 08:34:08.446939 [NOTICE] mod_sofia.c:1521 Pre-Answer
> sofia/internal-ipv6/1006 at franta.openstage.net!
> 2009-10-21 08:34:08.462967 [NOTICE] sofia.c:3925 Hangup
> sofia/internal-ipv6/sip:1007@[2000:2::1002]:5062 [CS_CONSUME_MEDIA]
> [INCOMPATIBLE_DESTINATION]

I suspect the codec negotiation. Make sure that both ends are offering a
common codec.




From tculjaga at gmail.com  Wed Oct 21 00:34:52 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 21 Oct 2009 09:34:52 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910201841030.12352@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910132359i6d53d8ban10f943795481bf7c@mail.gmail.com>
	<Pine.LNX.4.64.0910141134270.11620@srv.icf.org.ru>
	<65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>
	<Pine.LNX.4.64.0910141202500.21535@srv.icf.org.ru>
	<65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>
	<65d96fc80910140215y357fede1l37f6cba9725ddd89@mail.gmail.com>
	<Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>
	<65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
	<Pine.LNX.4.64.0910201841030.12352@srv.icf.org.ru>
Message-ID: <65d96fc80910210034nd7e8864hfcc421ec6a6c52f@mail.gmail.com>

>
>
>
> simple:
>
>
> <action application="bridge" data="h323/${number}"/>
>
> if fs not registered on gk then data="h323/${number}@xxx.xxx.xxx.xxx".
>
>
> TC>
> TC>2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3252 Found capability:
> TC>UserInput/PointDevice <14>
> TC>2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3248 FindCapability: 15
> TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external
> thread
> TC>0xb6eb60a0 for id 3048876944
> TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:65 Destroyed external
> thread
> TC>0xb6ebafa8 for id 3048876944
> TC>2009-10-20 10:08:18.426608 [DEBUG] tlibthrd.cxx:406 Destroyed thread
> TC>0xb6ebafa8 PExternalThread:0xb5ba2b90(id = b5ba2b90)
> TC>2009-10-20 10:08:18.426608 [DEBUG] h323caps.cxx:3252 Found capability:
> TC>UserInput/Modal <15>
> TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0,
> TC>expiries=0
> TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0,
> TC>expiries=0
> TC>2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external
> thread
> TC>0xb6eba910 for id 3048876944
> TC>2009-10-20 10:08:18.426608 [DEBUG] h4601.cxx:1725 Endpoint Attached
> TC>Segmentation fault (core dumped)
> TC>tculjaga at subZero:~/freeswitch-trunk$
> TC>
> TC>pls check: http://pastebin.freeswitch.org/10769
>
> look strange, what version of libpt/h323plus you use and freeswitch itself
> ?
>
> TC>
> TC>
>



I was using latest libpt.so.2.7-beta1.

Now I went back to libpt.so.2.6-beta6 (can't find 2.6.5 you mentioned...)
and FS is crashing on every call :P .. regardless if it is inbound or
outbound...

FreeSWITCH Version 1.0.trunk (15079M)

H323Plus is from cvs





so, what i did is:


create a directory e.g. h323

mkdir -p ~/h323
cd ~/h323

  svn co
http://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/branches/v2_6ptlib-2.6

  export PKG_CONFIG_PATH=/usr/local/lib/pkgconfig
  export LD_LIBRARY_PATH=/usr/local/lib

cd ptlib-2.6

  ./configure
  make
  sudo make install



cd ~/h323

  cvs -d:pserver:anonymous at h323plus.cvs.sourceforge.net:/cvsroot/h323plus
checkout h323plus

  export PTLIBDIR=~/h323plus/ptlib


cd h323plus

  ./configure
  make
  sudo make install



assuming you have FS src in your home

cd ~/freeswitch-trunk

  make mod_h323-clean
  make mod_h323
  sudo make mod_h323-install



cd /usr/local/freeswitch/lib/

  sudo ln -sf /usr/local/lib/libpt.so.2.6-beta6 libpt.so.2.6-beta6




start FS and load mod_h323





Please, can you advice what exact revisions of ptlib you are using so i can
do "svn so -r xxx", also what exact revision of freeswitch and H323Plus you
are using ?





Now with ptlib-2.6-beta6 can't even.


T.
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From helmut.kuper at ewetel.de  Wed Oct 21 01:26:45 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Wed, 21 Oct 2009 10:26:45 +0200
Subject: [Freeswitch-users] Qustion about INFO messages
	after	Connect/Answer
In-Reply-To: <CBC6759C-5192-4731-BACA-79C90CD4EE27@freeswitch.org>
References: <4AD859A4.5000101@ewetel.de>	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>	<4ADC7C09.6070503@ewetel.de>	<191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>	<4ADC85C0.7060908@ewetel.de>
	<4ADC8B6E.7070404@ewetel.de>	<191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>	<4ADD65DA.4050809@ewetel.de>	<87f2f3b90910200919i62dac94fq1b6c9289d9fb357c@mail.gmail.com>
	<CBC6759C-5192-4731-BACA-79C90CD4EE27@freeswitch.org>
Message-ID: <4ADEC5C5.1060306@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Brian, hi Mike,

On 20.10.2009 18:41, Brian West wrote:
> Or just set the var to what you want it to say?

Yes, understood and it works so far. This means that I must enhance my
dialplan to set this new variable to preserve old behaviour. No big
deal, but at least I have to know it.

> 
> /b
> 
> On Oct 20, 2009, at 11:19 AM, Michael Collins wrote:
> 
>>
>> Under what conditions did you see "unknown"? I'm wondering if the  
>> user can just pick a default other than "unknown" if he wants  
>> something else to be displayed.

I get it for internal calls from Snom to Snom. It seems to be the
default configuration. The sip flow shows two INFO messages sent from FS
to caller after callee picked up. The first INFO messages set the
callee's name to "unknown" on caller's side. The second changed it back
to callee's number.

Maybe there is a plan behind it ... by now it is simply increasing the
sip signalling load.

Any ideas for what the first INFO message is?

regards
helmut
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y0JZzsSKrGyPXTnPypy+qqQ=
=jNtK
-----END PGP SIGNATURE-----



From red.rain.seven at gmail.com  Wed Oct 21 02:44:53 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Wed, 21 Oct 2009 17:44:53 +0800
Subject: [Freeswitch-users] sched_api doesn't get launched
In-Reply-To: <87f2f3b90910161237h356b7aa2q6ec12c6ecf78dc34@mail.gmail.com>
References: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
	<3F1747F9-C543-4C6F-BB01-4DD5FA5B7BC5@jerris.com>
	<59ad9ca10910161153h5d531533vd844e49b9ca6d896@mail.gmail.com>
	<87f2f3b90910161237h356b7aa2q6ec12c6ecf78dc34@mail.gmail.com>
Message-ID: <59ad9ca10910210244q5410f2bdq3b93c25f95673d11@mail.gmail.com>

I can't seem to find the right thing to use in mod_java to execute api
commands, only api_after_bridge

2009-10-21 17:42:46.593094 [NOTICE] mod_sofia.c:1509 Pre-Answer
sofia/internal/1688003 at 192.168.1.66!
#
# A fatal error has been detected by the Java Runtime Environment:
#
#  SIGSEGV (0xb) at pc=0x004e4480, pid=1927, tid=16116624
#
# JRE version: 6.0_16-b01
# Java VM: Java HotSpot(TM) Client VM (14.2-b01 mixed mode linux-x86 )
# Problematic frame:
# C  [libc.so.6+0x6f480]  strcpy+0x10
#
# An error report file with more information is saved as:
# /usr/local/freeswitch/bin/hs_err_pid1927.log
2009-10-21 17:42:59.883729 [ERR] switch_core_session.c:1374 Invalid
Application sched_api
2009-10-21 17:42:59.883729 [NOTICE] switch_core_session.c:1375 Hangup
sofia/internal/1688003 at 192.168.1.66 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
#
# If you would like to submit a bug report, please visit:
#   http://java.sun.com/webapps/bugreport/crash.jsp
# The crash happened outside the Java Virtual Machine in native code.
# See problematic frame for where to report the bug.


On Sat, Oct 17, 2009 at 3:37 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Oct 16, 2009 at 11:53 AM, Henry Huang <red.rain.seven at gmail.com>wrote:
>
>> So how would you trigger it from a script dialplan? The only time it
>> seemed to work is when I did setVariable("api_after_bridge", "sched_api blah
>> blah blah");
>> but then it gets executed after the channel's been teared down. I thought
>> api_after_bridge means right after the call gets connected.
>>
>> I need something to execute an api command right before or right after the
>> call gets bridged.
>>
>> api_after_bridge is a channel variable, so using setVariable works just
> fine. If you need to sched_api is an API only. Check these out:
> http://wiki.freeswitch.org/wiki/Mod_commands#Misc._Commands
>
> So you need an API object in order to use it. I don't know the syntax for
> creating an api obj in Java but in Lua it goes like this:
> api = freeswitch.API();
> res = api:execute("sched_api","+300 none my_api my_api_args")
>
> Remember, if the method you are using isn't found in the dial plan tools
> then it isn't a dial plan application. Make sure it's on the list:
> http://wiki.freeswitch.org/wiki/Mod_dptools
>
> On the other hand, API commands are listed here:
> http://wiki.freeswitch.org/wiki/Mod_commands
>
> dptools require a session object, api commands require an api object...
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From bottleman at icf.org.ru  Wed Oct 21 03:32:43 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Wed, 21 Oct 2009 14:32:43 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910210034nd7e8864hfcc421ec6a6c52f@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910132359i6d53d8ban10f943795481bf7c@mail.gmail.com>
	<Pine.LNX.4.64.0910141134270.11620@srv.icf.org.ru>
	<65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>
	<Pine.LNX.4.64.0910141202500.21535@srv.icf.org.ru>
	<65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>
	<65d96fc80910140215y357fede1l37f6cba9725ddd89@mail.gmail.com>
	<Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>
	<65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
	<Pine.LNX.4.64.0910201841030.12352@srv.icf.org.ru>
	<65d96fc80910210034nd7e8864hfcc421ec6a6c52f@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910211427520.9175@srv.icf.org.ru>

On 2009-10-21 09:34 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:


TC>
TC>
TC>
TC>I was using latest libpt.so.2.7-beta1.
TC>
TC>Now I went back to libpt.so.2.6-beta6 (can't find 2.6.5 you mentioned...)
TC>and FS is crashing on every call :P .. regardless if it is inbound or
TC>outbound...

http://www.opalvoip.org/ first link into Lalande Stable 5 Released. 
On some version of cvs ptlib i get crash on module loading:)

TC>
TC>FreeSWITCH Version 1.0.trunk (15079M)

hm, i don't test it on trunk, may be there some isues, try get stack backtrace from core file to 
see where it crash. I use 1.0.4


TC>H323Plus is from cvs

it's ok.

TC>
TC>so, what i did is:
TC>
TC>
TC>create a directory e.g. h323
TC>
TC>mkdir -p ~/h323
TC>cd ~/h323
TC>
TC>  svn co
TC>http://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/branches/v2_6ptlib-2.6
TC>
TC>  export PKG_CONFIG_PATH=/usr/local/lib/pkgconfig
TC>  export LD_LIBRARY_PATH=/usr/local/lib
TC>
TC>cd ptlib-2.6
TC>
TC>  ./configure
TC>  make
TC>  sudo make install
TC>
TC>
TC>
TC>cd ~/h323
TC>
TC>  cvs -d:pserver:anonymous at h323plus.cvs.sourceforge.net:/cvsroot/h323plus
TC>checkout h323plus
TC>
TC>  export PTLIBDIR=~/h323plus/ptlib
TC>
TC>
TC>cd h323plus
TC>
TC>  ./configure
TC>  make
TC>  sudo make install
TC>
TC>
TC>
TC>assuming you have FS src in your home
TC>
TC>cd ~/freeswitch-trunk
TC>
TC>  make mod_h323-clean
TC>  make mod_h323
TC>  sudo make mod_h323-install
TC>
TC>
TC>
TC>cd /usr/local/freeswitch/lib/
TC>
TC>  sudo ln -sf /usr/local/lib/libpt.so.2.6-beta6 libpt.so.2.6-beta6
TC>
TC>
TC>
TC>
TC>start FS and load mod_h323
TC>
TC>
TC>
TC>
TC>
TC>Please, can you advice what exact revisions of ptlib you are using so i can
TC>do "svn so -r xxx", also what exact revision of freeswitch and H323Plus you
TC>are using ?
TC>
TC>
TC>
TC>
TC>
TC>Now with ptlib-2.6-beta6 can't even.
TC>
TC>
TC>T.
TC>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From ahmedmunir007 at gmail.com  Wed Oct 21 03:37:15 2009
From: ahmedmunir007 at gmail.com (Ahmed Munir)
Date: Wed, 21 Oct 2009 15:37:15 +0500
Subject: [Freeswitch-users] Call custom variable in condition
Message-ID: <a60fed5e0910210337n45ffefe2wc3e3d9d0b403a568@mail.gmail.com>

Hi,

I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I want
to use it in condition, as I'm listing down the configuration below;

<context name="SIP_incoming">
   <extension name="call-sip-extensions">
      <condition field="destination_number" expression="^(\d+)$">
          <action application="set" data="AUTHENTICATION_STATUS=0"/>
           <action application="transfer" data="${AUTHENTICATION_STATUS} XML
Authen_Status"/>
      </condition>
   </extension>
</context>

<context name="Authen_Status">
     <extension name="exten-auth-status">
       <condition field="AUTHENTICATION_STATUS" expression="^0$">
          <action application="answer"/>
          <action application="playback" data="play.wav"/>
      </condition>
    </extension>
  </context>




 But unfortunately it is not working. Kindly advise me how to do implement
it(Note: I don't want to call script). And one more thing to ask how can I
transfer the values within the same context?

-- 
Regards,

Ahmed Munir
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From mustafa.pk at gmail.com  Wed Oct 21 03:52:24 2009
From: mustafa.pk at gmail.com (Ghulam Mustafa)
Date: Wed, 21 Oct 2009 15:52:24 +0500
Subject: [Freeswitch-users] Call custom variable in condition
In-Reply-To: <a60fed5e0910210337n45ffefe2wc3e3d9d0b403a568@mail.gmail.com>
References: <a60fed5e0910210337n45ffefe2wc3e3d9d0b403a568@mail.gmail.com>
Message-ID: <4ADEE7E8.9000008@gmail.com>

Ahmed,

you can't use variables set by "set" application within a condition, 
though it doesn't make sense. wondering if there is any logic behind 
this or it's just a simple missing feature. anyone?

-m

Ahmed Munir wrote:
> Hi,
>
> I've declared a variable named AUTHENTICATION_STATUS in dialplan, and 
> I want to use it in condition, as I'm listing down the configuration 
> below;
>
> <context name="SIP_incoming">
>    <extension name="call-sip-extensions">
>       <condition field="destination_number" expression="^(\d+)$">
>           <action application="set" data="AUTHENTICATION_STATUS=0"/>
>            <action application="transfer" 
> data="${AUTHENTICATION_STATUS} XML Authen_Status"/>
>       </condition>
>    </extension>
> </context>
>
> <context name="Authen_Status">
>      <extension name="exten-auth-status">
>        <condition field="AUTHENTICATION_STATUS" expression="^0$">
>           <action application="answer"/>
>           <action application="playback" data="play.wav"/>
>       </condition>
>     </extension>
>   </context>
>
>
>
>
>  But unfortunately it is not working. Kindly advise me how to do 
> implement it(Note: I don't want to call script). And one more thing to 
> ask how can I transfer the values within the same context?
>
> -- 
> Regards,
>
> Ahmed Munir
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   




From tculjaga at gmail.com  Wed Oct 21 04:13:13 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 21 Oct 2009 13:13:13 +0200
Subject: [Freeswitch-users] Call custom variable in condition
In-Reply-To: <a60fed5e0910210337n45ffefe2wc3e3d9d0b403a568@mail.gmail.com>
References: <a60fed5e0910210337n45ffefe2wc3e3d9d0b403a568@mail.gmail.com>
Message-ID: <65d96fc80910210413t1123dbc0xd5740bc0673d638c@mail.gmail.com>

consider this:



<context name="SIP_incoming">
   <extension name="call-sip-extensions">
      <condition field="destination_number" expression="^(\d+)$">
          <action application="set" data="AUTHENTICATION_STATUS=0"/>
           <action application="transfer" data="${AUTHENTICATION_STATUS} XML
Authen_Status"/>
      </condition>
   </extension>
</context>



<context name="Authen_Status">
     <extension name="exten-auth-status">
       <condition field="${AUTHENTICATION_STATUS}" expression="^0$">
          <action application="answer"/>
          <action application="playback" data="play.wav"/>
      </condition>
    </extension>
  </context>





here is one of my dialplan. I'm using execute_extension but it is quite the
same...



   <extension name="ServiceLookup">
      <condition field="destination_number" expression="(^300030)(.*)">
         <action application="lookup_service_destination" data="in
${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, in $
1, in ${network_addr}:5060, out red_contact, out authResult"/>
         <action application="log" data="INFO ########################
ServiceLookup ########################\n"/>
         <action application="log" data="INFO ########################
contact = '${red_contact}' ##############\n"/>
         <action application="log" data="INFO ########################
CallerNum = '${caller_id_number:6:16}' ##########\n"/>
         <action application="log" data="INFO ########################
RADIUS auth = '${authResult}' ##########\n"/>

         <action application="execute_extension" data="doRedirect XML
public"/>
        </condition>
   </extension>


   <extension name="doRedirect">
      <condition field="destination_number" expression="^doRedirect$"/>
      <condition field="${authResult}" expression="^0$|^60$">
         <action application="log" data="INFO ########################
RADIUS auth OK!!!' ##########\n"/>
         <action application="redirect" data="${red_contact}"/>
         <anti-action application="log" data="INFO ########################
RADIUS auth NOK!! ##########\n"/>
         <anti-action application="respond" data="403 Forbidden"/>
      </condition>

   </extension>




On Wed, Oct 21, 2009 at 12:37 PM, Ahmed Munir <ahmedmunir007 at gmail.com>wrote:

> Hi,
>
> I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I
> want to use it in condition, as I'm listing down the configuration below;
>
> <context name="SIP_incoming">
>    <extension name="call-sip-extensions">
>       <condition field="destination_number" expression="^(\d+)$">
>           <action application="set" data="AUTHENTICATION_STATUS=0"/>
>            <action application="transfer" data="${AUTHENTICATION_STATUS}
> XML Authen_Status"/>
>       </condition>
>    </extension>
> </context>
>
> <context name="Authen_Status">
>      <extension name="exten-auth-status">
>        <condition field="AUTHENTICATION_STATUS" expression="^0$">
>           <action application="answer"/>
>           <action application="playback" data="play.wav"/>
>       </condition>
>     </extension>
>   </context>
>
>
>
>
>  But unfortunately it is not working. Kindly advise me how to do implement
> it(Note: I don't want to call script). And one more thing to ask how can I
> transfer the values within the same context?
>
> --
> Regards,
>
> Ahmed Munir
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mcampbellsmith at gmail.com  Wed Oct 21 04:34:55 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Wed, 21 Oct 2009 22:34:55 +1100
Subject: [Freeswitch-users] NOT in dialplan expression
Message-ID: <33c87fa30910210434t268e8026pfd2b5439f26069fd@mail.gmail.com>

Hi!

How do I do a NOT equal to in a dialplan expression

Normaly in regex I would use the ! character.  This doesn't seem to work in FS..

ie
      <condition field="${variable}" expression="!^1">

Shouldn't that match when the variable is not starting with one?



From tculjaga at gmail.com  Wed Oct 21 04:39:17 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 21 Oct 2009 13:39:17 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910211427520.9175@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>
	<Pine.LNX.4.64.0910141202500.21535@srv.icf.org.ru>
	<65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>
	<65d96fc80910140215y357fede1l37f6cba9725ddd89@mail.gmail.com>
	<Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>
	<65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
	<Pine.LNX.4.64.0910201841030.12352@srv.icf.org.ru>
	<65d96fc80910210034nd7e8864hfcc421ec6a6c52f@mail.gmail.com>
	<Pine.LNX.4.64.0910211427520.9175@srv.icf.org.ru>
Message-ID: <65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>

2009/10/21 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-21 09:34 +0200, Tihomir Culjaga wrote
> freeswitch-users at lists.fre...:
>
>
> TC>
> TC>
> TC>
> TC>I was using latest libpt.so.2.7-beta1.
> TC>
> TC>Now I went back to libpt.so.2.6-beta6 (can't find 2.6.5 you
> mentioned...)
> TC>and FS is crashing on every call :P .. regardless if it is inbound or
> TC>outbound...
>
> http://www.opalvoip.org/ first link into Lalande Stable 5 Released.
> On some version of cvs ptlib i get crash on module loading:)
>
> TC>
> TC>FreeSWITCH Version 1.0.trunk (15079M)
>
> hm, i don't test it on trunk, may be there some isues, try get stack
> backtrace from core file to
> see where it crash. I use 1.0.4
>
>
>

module load crash: http://pastebin.freeswitch.org/10783
FreeSWITCH backtrace: http://pastebin.freeswitch.org/10784


now, the only different thing is FS trunk ...

:P
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From tculjaga at gmail.com  Wed Oct 21 04:48:24 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 21 Oct 2009 13:48:24 +0200
Subject: [Freeswitch-users] NOT in dialplan expression
In-Reply-To: <33c87fa30910210434t268e8026pfd2b5439f26069fd@mail.gmail.com>
References: <33c87fa30910210434t268e8026pfd2b5439f26069fd@mail.gmail.com>
Message-ID: <65d96fc80910210448w14d52508j7793fafd930cc8dc@mail.gmail.com>

it depends of what you are trying to acheave.... one approach is with regex

check this: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_regex

you can set a different variable and have it true or false ... than you can
compare for false state...


well .. it is up to you ...

T.


On Wed, Oct 21, 2009 at 1:34 PM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> Hi!
>
> How do I do a NOT equal to in a dialplan expression
>
> Normaly in regex I would use the ! character.  This doesn't seem to work in
> FS..
>
> ie
>      <condition field="${variable}" expression="!^1">
>
> Shouldn't that match when the variable is not starting with one?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ineyaa at gmail.com  Wed Oct 21 04:58:36 2009
From: ineyaa at gmail.com (ineya ineya)
Date: Wed, 21 Oct 2009 13:58:36 +0200
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <20091021065841.GA13150@jdc.jasonjgw.net>
References: <FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
	<20091020133437.GA4941@dan.olp.net>
	<7b8592a0910200716q4a7db93cyb955595e4993ac96@mail.gmail.com>
	<858EC056-1D84-4F30-A0B7-9A70FF98695D@freeswitch.org>
	<7b8592a0910200821m18fe9b4dg73dda05a10699af2@mail.gmail.com>
	<642B6CED-CDF3-4D40-B493-0F2D6222C30B@freeswitch.org>
	<7b8592a0910200947k6ca5a391l35786f0e855ae388@mail.gmail.com>
	<20091021040343.GA9440@jdc.jasonjgw.net>
	<7b8592a0910202336x1bf2a8a2nd636219ec7031c5a@mail.gmail.com>
	<20091021065841.GA13150@jdc.jasonjgw.net>
Message-ID: <7b8592a0910210458s28ddffbdi77cf116d33c28405@mail.gmail.com>

Codecs are fine. I spent much time experimenting with codecs and
completely missed, that freeswitch is modifiyng the SDP record.

When phone A is making a call the SDP contains candidate media attributes:

a=candidate:123abc 1 UDP 9 2000:2::1001 5012 typ host
a=candidate:123abc 1 UDP 8 10.80.62.92 5010 typ host

But when freeswitch makes the INVITE on phone B, these 2 are missing
and phone is looking for it, so the INVITE gets rejected by phone with
"448 Not acceptable here"

So the question is, how can I make the freeswitch to pass these
candidate media attributes?

On Wed, Oct 21, 2009 at 8:58 AM, Jason White <jason at jasonjgw.net> wrote:

>
> I suspect the codec negotiation. Make sure that both ends are offering a
> common codec.
>



From leon at scarlet-internet.nl  Wed Oct 21 05:00:05 2009
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Wed, 21 Oct 2009 14:00:05 +0200
Subject: [Freeswitch-users] NOT in dialplan expression
In-Reply-To: <33c87fa30910210434t268e8026pfd2b5439f26069fd@mail.gmail.com>
References: <33c87fa30910210434t268e8026pfd2b5439f26069fd@mail.gmail.com>
Message-ID: <F813C725-7F27-4FDC-94AE-105F36BF6258@scarlet-internet.nl>

Hi,

Negating is done with [^...] in a regex, so 'not 1' is matched with:

/^[^1]$/

If you want to match on a longer sequence, you can do that with  
negative lookahead, for example 'not 123' can be matched like this:

/^(?!123$)\d{3}$/

regards,

Leon


On Oct 21, 2009, at 1:34 PM, Mark Campbell-Smith wrote:

> Hi!
>
> How do I do a NOT equal to in a dialplan expression
>
> Normaly in regex I would use the ! character.  This doesn't seem to  
> work in FS..
>
> ie
>      <condition field="${variable}" expression="!^1">
>
> Shouldn't that match when the variable is not starting with one?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From bottleman at icf.org.ru  Wed Oct 21 05:24:00 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Wed, 21 Oct 2009 16:24:00 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>
	<Pine.LNX.4.64.0910141202500.21535@srv.icf.org.ru>
	<65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>
	<65d96fc80910140215y357fede1l37f6cba9725ddd89@mail.gmail.com>
	<Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>
	<65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
	<Pine.LNX.4.64.0910201841030.12352@srv.icf.org.ru>
	<65d96fc80910210034nd7e8864hfcc421ec6a6c52f@mail.gmail.com>
	<Pine.LNX.4.64.0910211427520.9175@srv.icf.org.ru>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>

On 2009-10-21 13:39 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>2009/10/21 Georgiewskiy Yuriy <bottleman at icf.org.ru>
TC>
TC>> On 2009-10-21 09:34 +0200, Tihomir Culjaga wrote
TC>> freeswitch-users at lists.fre...:
TC>>
TC>>
TC>> TC>
TC>> TC>
TC>> TC>
TC>> TC>I was using latest libpt.so.2.7-beta1.
TC>> TC>
TC>> TC>Now I went back to libpt.so.2.6-beta6 (can't find 2.6.5 you
TC>> mentioned...)
TC>> TC>and FS is crashing on every call :P .. regardless if it is inbound or
TC>> TC>outbound...
TC>>
TC>> http://www.opalvoip.org/ first link into Lalande Stable 5 Released.
TC>> On some version of cvs ptlib i get crash on module loading:)
TC>>
TC>> TC>
TC>> TC>FreeSWITCH Version 1.0.trunk (15079M)
TC>>
TC>> hm, i don't test it on trunk, may be there some isues, try get stack
TC>> backtrace from core file to
TC>> see where it crash. I use 1.0.4
TC>>
TC>>
TC>>
TC>
TC>module load crash: http://pastebin.freeswitch.org/10783
TC>FreeSWITCH backtrace: http://pastebin.freeswitch.org/10784
TC>
TC>
TC>now, the only different thing is FS trunk ...

i have no trunk at this time and cannot test it, have you enabled crash-protection in fs?

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From woodydickson at gmail.com  Wed Oct 21 06:26:57 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Wed, 21 Oct 2009 21:26:57 +0800
Subject: [Freeswitch-users] check sip client availability
Message-ID: <a44688a0910210626rfd6de48m1232862f026d6b3c@mail.gmail.com>

Hi,

Is there any API to tell freeswitch to send a SIP OPTION message to check
the availability of a SIP client?

thanks,
woody
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From brian at freeswitch.org  Wed Oct 21 07:25:17 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Oct 2009 09:25:17 -0500
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <7b8592a0910210458s28ddffbdi77cf116d33c28405@mail.gmail.com>
References: <FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
	<20091020133437.GA4941@dan.olp.net>
	<7b8592a0910200716q4a7db93cyb955595e4993ac96@mail.gmail.com>
	<858EC056-1D84-4F30-A0B7-9A70FF98695D@freeswitch.org>
	<7b8592a0910200821m18fe9b4dg73dda05a10699af2@mail.gmail.com>
	<642B6CED-CDF3-4D40-B493-0F2D6222C30B@freeswitch.org>
	<7b8592a0910200947k6ca5a391l35786f0e855ae388@mail.gmail.com>
	<20091021040343.GA9440@jdc.jasonjgw.net>
	<7b8592a0910202336x1bf2a8a2nd636219ec7031c5a@mail.gmail.com>
	<20091021065841.GA13150@jdc.jasonjgw.net>
	<7b8592a0910210458s28ddffbdi77cf116d33c28405@mail.gmail.com>
Message-ID: <BCA761FE-0B4B-4B3C-9BC4-607A92A99E22@freeswitch.org>

What are you using to make this call?

/b

On Oct 21, 2009, at 6:58 AM, ineya ineya wrote:

> Codecs are fine. I spent much time experimenting with codecs and
> completely missed, that freeswitch is modifiyng the SDP record.
>
> When phone A is making a call the SDP contains candidate media  
> attributes:
>
> a=candidate:123abc 1 UDP 9 2000:2::1001 5012 typ host
> a=candidate:123abc 1 UDP 8 10.80.62.92 5010 typ host
>
> But when freeswitch makes the INVITE on phone B, these 2 are missing
> and phone is looking for it, so the INVITE gets rejected by phone with
> "448 Not acceptable here"
>
> So the question is, how can I make the freeswitch to pass these
> candidate media attributes?




From brian at freeswitch.org  Wed Oct 21 07:26:03 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Oct 2009 09:26:03 -0500
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>
	<Pine.LNX.4.64.0910141202500.21535@srv.icf.org.ru>
	<65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>
	<65d96fc80910140215y357fede1l37f6cba9725ddd89@mail.gmail.com>
	<Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>
	<65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
	<Pine.LNX.4.64.0910201841030.12352@srv.icf.org.ru>
	<65d96fc80910210034nd7e8864hfcc421ec6a6c52f@mail.gmail.com>
	<Pine.LNX.4.64.0910211427520.9175@srv.icf.org.ru>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
Message-ID: <DE165593-BA7B-4B0A-AAC2-429831D2ECDC@freeswitch.org>

Your SVN Account will be done soon and the directory in endpoints is  
already created for you to start importing your work.

Thanks,

/b

On Oct 21, 2009, at 7:24 AM, Georgiewskiy Yuriy wrote:

> On 2009-10-21 13:39 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre 
> ...:
>
> TC>2009/10/21 Georgiewskiy Yuriy <bottleman at icf.org.ru>
> TC>
> TC>> On 2009-10-21 09:34 +0200, Tihomir Culjaga wrote
> TC>> freeswitch-users at lists.fre...:
> TC>>
> TC>>
> TC>> TC>
> TC>> TC>
> TC>> TC>
> TC>> TC>I was using latest libpt.so.2.7-beta1.
> TC>> TC>
> TC>> TC>Now I went back to libpt.so.2.6-beta6 (can't find 2.6.5 you
> TC>> mentioned...)
> TC>> TC>and FS is crashing on every call :P .. regardless if it is  
> inbound or
> TC>> TC>outbound...
> TC>>
> TC>> http://www.opalvoip.org/ first link into Lalande Stable 5  
> Released.
> TC>> On some version of cvs ptlib i get crash on module loading:)
> TC>>
> TC>> TC>
> TC>> TC>FreeSWITCH Version 1.0.trunk (15079M)
> TC>>
> TC>> hm, i don't test it on trunk, may be there some isues, try get  
> stack
> TC>> backtrace from core file to
> TC>> see where it crash. I use 1.0.4
> TC>>
> TC>>
> TC>>
> TC>
> TC>module load crash: http://pastebin.freeswitch.org/10783
> TC>FreeSWITCH backtrace: http://pastebin.freeswitch.org/10784
> TC>
> TC>
> TC>now, the only different thing is FS trunk ...
>
> i have no trunk at this time and cannot test it, have you enabled  
> crash-protection in fs?
>
> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129- 
> RIPE_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From bottleman at icf.org.ru  Wed Oct 21 07:37:01 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Wed, 21 Oct 2009 18:37:01 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <DE165593-BA7B-4B0A-AAC2-429831D2ECDC@freeswitch.org>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910140058x3750f4bax7de05caf697ed30b@mail.gmail.com>
	<Pine.LNX.4.64.0910141202500.21535@srv.icf.org.ru>
	<65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>
	<65d96fc80910140215y357fede1l37f6cba9725ddd89@mail.gmail.com>
	<Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>
	<65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
	<Pine.LNX.4.64.0910201841030.12352@srv.icf.org.ru>
	<65d96fc80910210034nd7e8864hfcc421ec6a6c52f@mail.gmail.com>
	<Pine.LNX.4.64.0910211427520.9175@srv.icf.org.ru>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
	<DE165593-BA7B-4B0A-AAC2-429831D2ECDC@freeswitch.org>
Message-ID: <Pine.LNX.4.64.0910211836450.30865@srv.icf.org.ru>

On 2009-10-21 09:26 -0500, Brian West wrote freeswitch-users at lists.freeswit...:

??.

BW>Your SVN Account will be done soon and the directory in endpoints is  
BW>already created for you to start importing your work.
BW>
BW>Thanks,
BW>
BW>/b
BW>
BW>On Oct 21, 2009, at 7:24 AM, Georgiewskiy Yuriy wrote:
BW>
BW>> On 2009-10-21 13:39 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre 
BW>> ...:
BW>>
BW>> TC>2009/10/21 Georgiewskiy Yuriy <bottleman at icf.org.ru>
BW>> TC>
BW>> TC>> On 2009-10-21 09:34 +0200, Tihomir Culjaga wrote
BW>> TC>> freeswitch-users at lists.fre...:
BW>> TC>>
BW>> TC>>
BW>> TC>> TC>
BW>> TC>> TC>
BW>> TC>> TC>
BW>> TC>> TC>I was using latest libpt.so.2.7-beta1.
BW>> TC>> TC>
BW>> TC>> TC>Now I went back to libpt.so.2.6-beta6 (can't find 2.6.5 you
BW>> TC>> mentioned...)
BW>> TC>> TC>and FS is crashing on every call :P .. regardless if it is  
BW>> inbound or
BW>> TC>> TC>outbound...
BW>> TC>>
BW>> TC>> http://www.opalvoip.org/ first link into Lalande Stable 5  
BW>> Released.
BW>> TC>> On some version of cvs ptlib i get crash on module loading:)
BW>> TC>>
BW>> TC>> TC>
BW>> TC>> TC>FreeSWITCH Version 1.0.trunk (15079M)
BW>> TC>>
BW>> TC>> hm, i don't test it on trunk, may be there some isues, try get  
BW>> stack
BW>> TC>> backtrace from core file to
BW>> TC>> see where it crash. I use 1.0.4
BW>> TC>>
BW>> TC>>
BW>> TC>>
BW>> TC>
BW>> TC>module load crash: http://pastebin.freeswitch.org/10783
BW>> TC>FreeSWITCH backtrace: http://pastebin.freeswitch.org/10784
BW>> TC>
BW>> TC>
BW>> TC>now, the only different thing is FS trunk ...
BW>>
BW>> i have no trunk at this time and cannot test it, have you enabled  
BW>> crash-protection in fs?
BW>>
BW>> C ?????????                       With Best Regards
BW>> ???????????? ????.                Georgiewskiy Yuriy
BW>> +7 4872 711666                    +7 4872 711666
BW>> ???? +7 4872 711143               fax +7 4872 711143
BW>> ???????? ??? "?? ?? ??????"       IT Service Ltd
BW>> http://nkoort.ru                  http://nkoort.ru
BW>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
BW>> YG129-RIPE                        YG129- 
BW>> RIPE_______________________________________________
BW>> FreeSWITCH-users mailing list
BW>> FreeSWITCH-users at lists.freeswitch.org
BW>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
BW>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
BW>> users
BW>> http://www.freeswitch.org
BW>
BW>
BW>_______________________________________________
BW>FreeSWITCH-users mailing list
BW>FreeSWITCH-users at lists.freeswitch.org
BW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
BW>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
BW>http://www.freeswitch.org
BW>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From tculjaga at gmail.com  Wed Oct 21 07:43:53 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 21 Oct 2009 16:43:53 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>
	<65d96fc80910140215y357fede1l37f6cba9725ddd89@mail.gmail.com>
	<Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>
	<65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
	<Pine.LNX.4.64.0910201841030.12352@srv.icf.org.ru>
	<65d96fc80910210034nd7e8864hfcc421ec6a6c52f@mail.gmail.com>
	<Pine.LNX.4.64.0910211427520.9175@srv.icf.org.ru>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
Message-ID: <65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>

>
> TC>
> TC>module load crash: http://pastebin.freeswitch.org/10783
> TC>FreeSWITCH backtrace: http://pastebin.freeswitch.org/10784
> TC>
> TC>
> TC>now, the only different thing is FS trunk ...
>
> i have no trunk at this time and cannot test it, have you enabled
> crash-protection in fs?
>
>
Ok i moved back FS to 1.0.4 and i'm back on the first issues,

1. no audio after i answer the call (call flow is H323 => FS => SIP).. i
hear the ringback on the H323 side bit when i answer the call, nothing!

FS = 10.4.62.7
SIP phone = 10.4.62.89
H323 endpoint = 10.1.14.153



 53.696259  10.1.14.153 -> 10.4.62.7    H.225.0 CS: setup
 53.705415    10.4.62.7 -> 10.1.14.153  H.225.0 CS: callProceeding
 53.716368    10.4.62.7 -> 10.4.62.89   SIP/SDP Request: INVITE
sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>;transport=udp, with session
description
 53.717697    10.4.62.7 -> 10.1.14.153  H.225.0 CS: alerting
 53.720132    10.4.62.7 -> 10.1.14.153  H.225.0 CS: facility
 53.725453    10.4.62.7 -> 10.1.14.153  H.245 terminalCapabilitySet
 53.725957    10.4.62.7 -> 10.1.14.153  H.245 masterSlaveDetermination
 53.728129   10.4.62.89 -> 10.4.62.7    SIP Status: 100 Trying
 53.741112   10.4.62.89 -> 10.4.62.7    SIP Status: 180 Ringing
 53.743199  10.1.14.153 -> 10.4.62.7    H.245 terminalCapabilitySet
 53.744221  10.1.14.153 -> 10.4.62.7    H.245 masterSlaveDetermination
 53.749443    10.4.62.7 -> 10.1.14.153  H.245 terminalCapabilitySetAck
 53.751624    10.4.62.7 -> 10.1.14.153  H.245 masterSlaveDeterminationAck
 53.758710  10.1.14.153 -> 10.4.62.7    H.245 terminalCapabilitySetAck
 53.761241  10.1.14.153 -> 10.4.62.7    H.245 masterSlaveDeterminationAck
 53.763919    10.4.62.7 -> 10.1.14.153  H.245 openLogicalChannel (g711A)
#3712: OLC found 10.1.14.153/10.4.62.7/101
 53.777464  10.1.14.153 -> 10.4.62.7    H.245 openLogicalChannelAck
 53.799864    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1305, Time=240, Mark
 53.828940    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1306, Time=480
 53.859180    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1307, Time=720
 53.889379    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1308, Time=960
 53.919611    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1309, Time=1200
 53.949833    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1310, Time=1440
 53.978964    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1311, Time=1680
 54.009218    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1312, Time=1920
 54.039424    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1313, Time=2160
 54.069640    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1314, Time=2400
 54.099795    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1315, Time=2640
 54.129064    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1316, Time=2880

<------------------ snip ----------------> this is a ringback sent from FS
=> H323endpoint

 60.189074    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1518, Time=51360
 60.219197    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1519, Time=51600
 60.249451    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1520, Time=51840
 60.279684    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1521, Time=52080
 60.309689    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1522, Time=52320
 60.338812    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1523, Time=52560
 60.368997    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1524, Time=52800
 60.399231    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1525, Time=53040
 60.429445    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1526, Time=53280
 60.459679    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1527, Time=53520
 60.488917    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1528, Time=53760
 60.494975   10.4.62.89 -> 10.4.62.7    SIP/SDP Status: 200 OK, with session
description
 60.495478    10.4.62.7 -> 10.4.62.89   SIP Request: ACK
sip:1001 at 10.4.62.89<sip%3A1001 at 10.4.62.89>
;transport=udp
 60.496114    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 1
 60.519467    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
SSRC=0xBE241F, Seq=1529, Time=54000
 60.520574    10.4.62.7 -> 10.1.14.153  H.225.0 CS: connect
 60.531284  10.1.14.153 -> 10.4.62.7    H.245 openLogicalChannel (g711A)
#4045: OLC found 10.4.62.7/10.1.14.153/101
 60.535298    10.4.62.7 -> 10.1.14.153  H.245 openLogicalChannelAck
 60.595098  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=39691, Time=160, Mark
 60.598967  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=39692, Time=320
 60.616538  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=39693, Time=480
 60.639583  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=39694, Time=640
 60.657185  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=39695, Time=800
 60.673118   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21152, Time=98705487
 60.680623  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=39696, Time=960
 60.693033   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21153, Time=98705647
 60.698727  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=39697, Time=1120
 60.712984   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21154, Time=98705807
 60.716184  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=39698, Time=1280
 60.733016   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21155, Time=98705967
 60.739264  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=39699, Time=1440
 60.752987   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21156, Time=98706127
 60.756820  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=39700, Time=1600
 60.772977   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21157, Time=98706287
 60.780507  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=39701, Time=1760
 60.793029   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21158, Time=98706447
 60.798172  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=39702, Time=1920
 60.812980   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21159, Time=98706607
 60.815755  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=39703, Time=2080
 60.833107   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21160, Time=98706767
 60.838943  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=39704, Time=2240
 60.852999   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21161, Time=98706927
 60.856476  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=39705, Time=2400

<------------------ snip ----------------> this is after after SIP phone
answered

 68.245488  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=40079, Time=62240
 68.252993   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21531, Time=98766127
 68.268524  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=40080, Time=62400
 68.272988   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21532, Time=98766287
 68.286129  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=40081, Time=62560
 68.292978   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21533, Time=98766447
 68.309595  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=40082, Time=62720
 68.312988   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21534, Time=98766607
 68.327216  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=40083, Time=62880
 68.332982   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21535, Time=98766767
 68.350940  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=40084, Time=63040
 68.352977   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xC2B5E0D, Seq=21536, Time=98766927
 68.368506  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=40085, Time=63200
 68.374916   10.4.62.89 -> 10.4.62.7    SIP Request: BYE
sip:mod_sofia at 10.4.62.7:5060
 68.375342    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 3
 68.375620    10.4.62.7 -> 10.4.62.89   SIP Status: 200 OK
 68.385785  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=40086, Time=63360
 68.409172  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=40087, Time=63520

<------------------ snip ----------------> This is after SIP Phone released
the call

 70.495427  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=40193, Time=80480
 70.518592  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=40194, Time=80640
 70.536159  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xA994332E, Seq=40195, Time=80800
 70.552157  10.1.14.153 -> 10.4.62.7    H.245 endSessionCommand
 70.552401  10.1.14.153 -> 10.4.62.7    H.225.0 CS: releaseComplete
 70.553981    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 3
 70.554488    10.4.62.7 -> 10.1.14.153  H.245 endSessionCommand
 70.555056    10.4.62.7 -> 10.1.14.153  H.225.0 CS: releaseComplete



Well, something gets wrong hrer ...




2. hangup from sip side doesn't release the h323 leg (now the difference is
that FS is not complaining about thread mismatch ant it looks clean but FS
doesn't send any releasecomplete message... strange)
3. coredumps when i place outgoing calls
-------------- next part --------------
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From rupa at rupa.com  Wed Oct 21 08:07:46 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 21 Oct 2009 10:07:46 -0500
Subject: [Freeswitch-users] NOT in dialplan expression
In-Reply-To: <33c87fa30910210434t268e8026pfd2b5439f26069fd@mail.gmail.com>
References: <33c87fa30910210434t268e8026pfd2b5439f26069fd@mail.gmail.com>
Message-ID: <b39273530910210807j1f8a2fc6l4ff2bf1c154497b9@mail.gmail.com>

You can also look at using anti-action rather than action after the condition.

condition == if
action == then
anti-action == else

On Wed, Oct 21, 2009 at 6:34 AM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> Hi!
>
> How do I do a NOT equal to in a dialplan expression
>
> Normaly in regex I would use the ! character. ?This doesn't seem to work in FS..
>
> ie
> ? ? ?<condition field="${variable}" expression="!^1">
>
> Shouldn't that match when the variable is not starting with one?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa



From rupa at rupa.com  Wed Oct 21 08:09:35 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 21 Oct 2009 10:09:35 -0500
Subject: [Freeswitch-users] check sip client availability
In-Reply-To: <a44688a0910210626rfd6de48m1232862f026d6b3c@mail.gmail.com>
References: <a44688a0910210626rfd6de48m1232862f026d6b3c@mail.gmail.com>
Message-ID: <b39273530910210809g22cf8829mc4918590c849c669@mail.gmail.com>

This applies only to clients that are detected as nat, but maybe:

http://wiki.freeswitch.org/wiki/Sofia.conf.xml#nat-options-ping

This is applied to the profile, not an app or a var setting.

On Wed, Oct 21, 2009 at 8:26 AM, Woody Dickson <woodydickson at gmail.com> wrote:
> Hi,
>
> Is there any API to tell freeswitch to send a SIP OPTION message to check
> the availability of a SIP client?
>
> thanks,
> woody
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa



From bottleman at icf.org.ru  Wed Oct 21 08:23:59 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Wed, 21 Oct 2009 19:23:59 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910140116i12eb13fcj81792ab53cef14ca@mail.gmail.com>
	<65d96fc80910140215y357fede1l37f6cba9725ddd89@mail.gmail.com>
	<Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>
	<65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
	<Pine.LNX.4.64.0910201841030.12352@srv.icf.org.ru>
	<65d96fc80910210034nd7e8864hfcc421ec6a6c52f@mail.gmail.com>
	<Pine.LNX.4.64.0910211427520.9175@srv.icf.org.ru>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>

On 2009-10-21 16:43 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>>
TC>> TC>
TC>> TC>module load crash: http://pastebin.freeswitch.org/10783
TC>> TC>FreeSWITCH backtrace: http://pastebin.freeswitch.org/10784
TC>> TC>
TC>> TC>
TC>> TC>now, the only different thing is FS trunk ...
TC>>
TC>> i have no trunk at this time and cannot test it, have you enabled
TC>> crash-protection in fs?
TC>>
TC>>
TC>Ok i moved back FS to 1.0.4 and i'm back on the first issues,
TC>
TC>1. no audio after i answer the call (call flow is H323 => FS => SIP).. i
TC>hear the ringback on the H323 side bit when i answer the call, nothing!

TC>FS = 10.4.62.7
TC>SIP phone = 10.4.62.89
TC>H323 endpoint = 10.1.14.153

TC> 53.696259  10.1.14.153 -> 10.4.62.7    H.225.0 CS: setup
TC> 53.705415    10.4.62.7 -> 10.1.14.153  H.225.0 CS: callProceeding
TC> 53.716368    10.4.62.7 -> 10.4.62.89   SIP/SDP Request: INVITE
TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>;transport=udp, with session
TC>description
TC> 53.717697    10.4.62.7 -> 10.1.14.153  H.225.0 CS: alerting
TC> 53.720132    10.4.62.7 -> 10.1.14.153  H.225.0 CS: facility
TC> 53.725453    10.4.62.7 -> 10.1.14.153  H.245 terminalCapabilitySet
TC> 53.725957    10.4.62.7 -> 10.1.14.153  H.245 masterSlaveDetermination
TC> 53.728129   10.4.62.89 -> 10.4.62.7    SIP Status: 100 Trying
TC> 53.741112   10.4.62.89 -> 10.4.62.7    SIP Status: 180 Ringing
TC> 53.743199  10.1.14.153 -> 10.4.62.7    H.245 terminalCapabilitySet
TC> 53.744221  10.1.14.153 -> 10.4.62.7    H.245 masterSlaveDetermination
TC> 53.749443    10.4.62.7 -> 10.1.14.153  H.245 terminalCapabilitySetAck
TC> 53.751624    10.4.62.7 -> 10.1.14.153  H.245 masterSlaveDeterminationAck
TC> 53.758710  10.1.14.153 -> 10.4.62.7    H.245 terminalCapabilitySetAck
TC> 53.761241  10.1.14.153 -> 10.4.62.7    H.245 masterSlaveDeterminationAck
TC> 53.763919    10.4.62.7 -> 10.1.14.153  H.245 openLogicalChannel (g711A)
TC>#3712: OLC found 10.1.14.153/10.4.62.7/101
TC> 53.777464  10.1.14.153 -> 10.4.62.7    H.245 openLogicalChannelAck
TC> 53.799864    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1305, Time=240, Mark
TC> 53.828940    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1306, Time=480
TC> 53.859180    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1307, Time=720
TC> 53.889379    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1308, Time=960
TC> 53.919611    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1309, Time=1200
TC> 53.949833    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1310, Time=1440
TC> 53.978964    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1311, Time=1680
TC> 54.009218    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1312, Time=1920
TC> 54.039424    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1313, Time=2160
TC> 54.069640    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1314, Time=2400
TC> 54.099795    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1315, Time=2640
TC> 54.129064    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1316, Time=2880
TC>
TC><------------------ snip ----------------> this is a ringback sent from FS
TC>=> H323endpoint
TC>
TC> 60.189074    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1518, Time=51360
TC> 60.219197    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1519, Time=51600
TC> 60.249451    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1520, Time=51840
TC> 60.279684    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1521, Time=52080
TC> 60.309689    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1522, Time=52320
TC> 60.338812    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1523, Time=52560
TC> 60.368997    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1524, Time=52800
TC> 60.399231    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1525, Time=53040
TC> 60.429445    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1526, Time=53280
TC> 60.459679    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1527, Time=53520
TC> 60.488917    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1528, Time=53760
TC> 60.494975   10.4.62.89 -> 10.4.62.7    SIP/SDP Status: 200 OK, with session
TC>description
TC> 60.495478    10.4.62.7 -> 10.4.62.89   SIP Request: ACK
TC>sip:1001 at 10.4.62.89<sip%3A1001 at 10.4.62.89>
TC>;transport=udp
TC> 60.496114    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 1
TC> 60.519467    10.4.62.7 -> 10.1.14.153  RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xBE241F, Seq=1529, Time=54000
TC> 60.520574    10.4.62.7 -> 10.1.14.153  H.225.0 CS: connect
TC> 60.531284  10.1.14.153 -> 10.4.62.7    H.245 openLogicalChannel (g711A)
TC>#4045: OLC found 10.4.62.7/10.1.14.153/101
TC> 60.535298    10.4.62.7 -> 10.1.14.153  H.245 openLogicalChannelAck
TC> 60.595098  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=39691, Time=160, Mark
TC> 60.598967  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=39692, Time=320
TC> 60.616538  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=39693, Time=480
TC> 60.639583  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=39694, Time=640
TC> 60.657185  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=39695, Time=800
TC> 60.673118   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21152, Time=98705487
TC> 60.680623  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=39696, Time=960
TC> 60.693033   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21153, Time=98705647
TC> 60.698727  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=39697, Time=1120
TC> 60.712984   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21154, Time=98705807
TC> 60.716184  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=39698, Time=1280
TC> 60.733016   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21155, Time=98705967
TC> 60.739264  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=39699, Time=1440
TC> 60.752987   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21156, Time=98706127
TC> 60.756820  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=39700, Time=1600
TC> 60.772977   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21157, Time=98706287
TC> 60.780507  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=39701, Time=1760
TC> 60.793029   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21158, Time=98706447
TC> 60.798172  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=39702, Time=1920
TC> 60.812980   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21159, Time=98706607
TC> 60.815755  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=39703, Time=2080
TC> 60.833107   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21160, Time=98706767
TC> 60.838943  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=39704, Time=2240
TC> 60.852999   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21161, Time=98706927
TC> 60.856476  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=39705, Time=2400
TC>
TC><------------------ snip ----------------> this is after after SIP phone
TC>answered
TC>
TC> 68.245488  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=40079, Time=62240
TC> 68.252993   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21531, Time=98766127
TC> 68.268524  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=40080, Time=62400
TC> 68.272988   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21532, Time=98766287
TC> 68.286129  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=40081, Time=62560
TC> 68.292978   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21533, Time=98766447
TC> 68.309595  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=40082, Time=62720
TC> 68.312988   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21534, Time=98766607
TC> 68.327216  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=40083, Time=62880
TC> 68.332982   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21535, Time=98766767
TC> 68.350940  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=40084, Time=63040
TC> 68.352977   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xC2B5E0D, Seq=21536, Time=98766927
TC> 68.368506  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=40085, Time=63200
TC> 68.374916   10.4.62.89 -> 10.4.62.7    SIP Request: BYE
TC>sip:mod_sofia at 10.4.62.7:5060
TC> 68.375342    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 3
TC> 68.375620    10.4.62.7 -> 10.4.62.89   SIP Status: 200 OK
TC> 68.385785  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=40086, Time=63360
TC> 68.409172  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=40087, Time=63520
TC>
TC><------------------ snip ----------------> This is after SIP Phone released
TC>the call
TC>
TC> 70.495427  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=40193, Time=80480
TC> 70.518592  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=40194, Time=80640
TC> 70.536159  10.1.14.153 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
TC>SSRC=0xA994332E, Seq=40195, Time=80800
TC> 70.552157  10.1.14.153 -> 10.4.62.7    H.245 endSessionCommand
TC> 70.552401  10.1.14.153 -> 10.4.62.7    H.225.0 CS: releaseComplete
TC> 70.553981    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 3
TC> 70.554488    10.4.62.7 -> 10.1.14.153  H.245 endSessionCommand
TC> 70.555056    10.4.62.7 -> 10.1.14.153  H.225.0 CS: releaseComplete
TC>
TC>
TC>
TC>Well, something gets wrong hrer ...
TC>
TC>
TC>
TC>
TC>2. hangup from sip side doesn't release the h323 leg (now the difference is
TC>that FS is not complaining about thread mismatch ant it looks clean but FS
TC>doesn't send any releasecomplete message... strange)
TC>3. coredumps when i place outgoing calls

btw, 

TC> 70.552157  10.1.14.153 -> 10.4.62.7    H.245 endSessionCommand                                             
TC> 70.552401  10.1.14.153 -> 10.4.62.7    H.225.0 CS: releaseComplete                                         
TC> 70.553981    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 3                                           
TC> 70.554488    10.4.62.7 -> 10.1.14.153  H.245 endSessionCommand                                             
TC> 70.555056    10.4.62.7 -> 10.1.14.153  H.225.0 CS: releaseComplete 

it send, now i have no way to test h323->sip transit, i will have it tomorow.
sip->h323 for me work fine now, give backtrace from code dump of 1.0.4 where it die?

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From ineyaa at gmail.com  Wed Oct 21 08:40:13 2009
From: ineyaa at gmail.com (ineya ineya)
Date: Wed, 21 Oct 2009 17:40:13 +0200
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <BCA761FE-0B4B-4B3C-9BC4-607A92A99E22@freeswitch.org>
References: <FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
	<858EC056-1D84-4F30-A0B7-9A70FF98695D@freeswitch.org>
	<7b8592a0910200821m18fe9b4dg73dda05a10699af2@mail.gmail.com>
	<642B6CED-CDF3-4D40-B493-0F2D6222C30B@freeswitch.org>
	<7b8592a0910200947k6ca5a391l35786f0e855ae388@mail.gmail.com>
	<20091021040343.GA9440@jdc.jasonjgw.net>
	<7b8592a0910202336x1bf2a8a2nd636219ec7031c5a@mail.gmail.com>
	<20091021065841.GA13150@jdc.jasonjgw.net>
	<7b8592a0910210458s28ddffbdi77cf116d33c28405@mail.gmail.com>
	<BCA761FE-0B4B-4B3C-9BC4-607A92A99E22@freeswitch.org>
Message-ID: <7b8592a0910210840u57a664f3l9c65ee75f457ea6a@mail.gmail.com>

the phone is called OpenStage

On Wed, Oct 21, 2009 at 4:25 PM, Brian West <brian at freeswitch.org> wrote:
> What are you using to make this call?
>
> /b
>



From msc at freeswitch.org  Wed Oct 21 08:43:58 2009
From: msc at freeswitch.org (Michael S Collins)
Date: Wed, 21 Oct 2009 08:43:58 -0700
Subject: [Freeswitch-users] sched_api doesn't get launched
In-Reply-To: <59ad9ca10910210244q5410f2bdq3b93c25f95673d11@mail.gmail.com>
References: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
	<3F1747F9-C543-4C6F-BB01-4DD5FA5B7BC5@jerris.com>
	<59ad9ca10910161153h5d531533vd844e49b9ca6d896@mail.gmail.com>
	<87f2f3b90910161237h356b7aa2q6ec12c6ecf78dc34@mail.gmail.com>
	<59ad9ca10910210244q5410f2bdq3b93c25f95673d11@mail.gmail.com>
Message-ID: <38B84C14-FFC3-43F4-BAAE-61B1F1D762C7@freeswitch.org>

Judging by this error I would assume that you're still calling  
sched_api as a Dialplan application and not as an FS API command.

You need to figure out how to create an API obj in java and call  
sched_api from that object.

-MC

Sent from my iPhone

On Oct 21, 2009, at 2:44 AM, Henry Huang <red.rain.seven at gmail.com>  
wrote:

> I can't seem to find the right thing to use in mod_java to execute  
> api commands, only api_after_bridge
>
>
> 2009-10-21 17:42:46.593094 [NOTICE] mod_sofia.c:1509 Pre-Answer  
> sofia/internal/1688003 at 192.168.1.66!
> #
> # A fatal error has been detected by the Java Runtime Environment:
> #
> #  SIGSEGV (0xb) at pc=0x004e4480, pid=1927, tid=16116624
> #
> # JRE version: 6.0_16-b01
> # Java VM: Java HotSpot(TM) Client VM (14.2-b01 mixed mode linux-x86 )
> # Problematic frame:
> # C  [libc.so.6+0x6f480]  strcpy+0x10
> #
> # An error report file with more information is saved as:
> # /usr/local/freeswitch/bin/hs_err_pid1927.log
> 2009-10-21 17:42:59.883729 [ERR] switch_core_session.c:1374 Invalid  
> Application sched_api
> 2009-10-21 17:42:59.883729 [NOTICE] switch_core_session.c:1375  
> Hangup sofia/internal/1688003 at 192.168.1.66 [CS_EXECUTE]  
> [DESTINATION_OUT_OF_ORDER]
> #
> # If you would like to submit a bug report, please visit:
> #   http://java.sun.com/webapps/bugreport/crash.jsp
> # The crash happened outside the Java Virtual Machine in native code.
> # See problematic frame for where to report the bug.
>
>
> On Sat, Oct 17, 2009 at 3:37 AM, Michael Collins  
> <msc at freeswitch.org> wrote:
>
>
> On Fri, Oct 16, 2009 at 11:53 AM, Henry Huang <red.rain.seven at gmail.com 
> > wrote:
> So how would you trigger it from a script dialplan? The only time it  
> seemed to work is when I did setVariable("api_after_bridge",  
> "sched_api blah blah blah");
> but then it gets executed after the channel's been teared down. I  
> thought api_after_bridge means right after the call gets connected.
>
> I need something to execute an api command right before or right  
> after the call gets bridged.
>
> api_after_bridge is a channel variable, so using setVariable works  
> just fine. If you need to sched_api is an API only. Check these out:
> http://wiki.freeswitch.org/wiki/Mod_commands#Misc._Commands
>
> So you need an API object in order to use it. I don't know the  
> syntax for creating an api obj in Java but in Lua it goes like this:
> api = freeswitch.API();
> res = api:execute("sched_api","+300 none my_api my_api_args")
>
> Remember, if the method you are using isn't found in the dial plan  
> tools then it isn't a dial plan application. Make sure it's on the  
> list:
> http://wiki.freeswitch.org/wiki/Mod_dptools
>
> On the other hand, API commands are listed here:
>
> http://wiki.freeswitch.org/wiki/Mod_commands
>
> dptools require a session object, api commands require an api  
> object...
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Henry Huang
> UniC Solution - Communication Unified
> VoIP & Open Source software Consultant
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From tculjaga at gmail.com  Wed Oct 21 08:48:47 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 21 Oct 2009 17:48:47 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>
	<65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
	<Pine.LNX.4.64.0910201841030.12352@srv.icf.org.ru>
	<65d96fc80910210034nd7e8864hfcc421ec6a6c52f@mail.gmail.com>
	<Pine.LNX.4.64.0910211427520.9175@srv.icf.org.ru>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
Message-ID: <65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>

FS = 10.4.62.7
SIP phone = 10.4.62.89
H323 endpoint = 10.1.14.153



> TC>2. hangup from sip side doesn't release the h323 leg (now the difference
> is
> TC>that FS is not complaining about thread mismatch ant it looks clean but
> FS
> TC>doesn't send any releasecomplete message... strange)
> TC>3. coredumps when i place outgoing calls
>
> btw,
>
> TC> 70.552157  10.1.14.153 -> 10.4.62.7    H.245 endSessionCommand
> TC> 70.552401  10.1.14.153 -> 10.4.62.7    H.225.0 CS: releaseComplete
> TC> 70.553981    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 3
> TC> 70.554488    10.4.62.7 -> 10.1.14.153  H.245 endSessionCommand
> TC> 70.555056    10.4.62.7 -> 10.1.14.153  H.225.0 CS: releaseComplete
>
> it send, now i have no way to test h323->sip transit, i will have it
> tomorow.
> sip->h323 for me work fine now, give backtrace from code dump of 1.0.4
> where it die?
>
>

this endSession is when i hangup from H232 side as well :P ... if i don't
hangup on H323 side the H323 leg is not released. Pls chec the time the
packets were sent ...



Here i hangup on the SIP Phone:

68.374916   10.4.62.89 -> 10.4.62.7    SIP Request: BYE
sip:mod_sofia at 10.4.62.7:5060
68.375342    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 3
68.375620    10.4.62.7 -> 10.4.62.89   SIP Status: 200 OK

2 sec delay

Here i hangup on the H323 endpoint (releaseComplete comes from H323 endpoint
first here )
70.552157  10.1.14.153 -> 10.4.62.7    H.245 endSessionCommand
70.552401  10.1.14.153 -> 10.4.62.7    H.225.0 CS: releaseComplete


FS just acknowlages it here:
70.554488    10.4.62.7 -> 10.1.14.153  H.245 endSessionCommand
70.555056    10.4.62.7 -> 10.1.14.153  H.225.0 CS: releaseComplete



I have enabled crash-protection and when i do SIP => H323 call it doesn't
generate coredumps... it is just this thread that is crashing ... pls check
the log downbelow:

Dialplan: sofia/internal/1001 at singtel Regex (FAIL) [EMERGENCY]
destination_number(05492122) =~ /^0(112|9[23456])$/ break=on-false
Dialplan: sofia/internal/1001 at singtel parsing [default->SPECIAL_SERVICES]
continue=false
Dialplan: sofia/internal/1001 at singtel Regex (FAIL) [SPECIAL_SERVICES]
destination_number(05492122) =~ /^0(9[01789]\d{3,4})$/ break=on-false
Dialplan: sofia/internal/1001 at singtel parsing [default->ENYTHING_ELSE]
continue=false
Dialplan: sofia/internal/1001 at singtel Regex (PASS) [ENYTHING_ELSE]
destination_number(05492122) =~
/^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$/ break=on-false
Dialplan: sofia/internal/1001 at singtel Action
set(effective_caller_id_number=1001282122)
Dialplan: sofia/internal/1001 at singtel Action set(NCX_IP=10.4.4.254)
Dialplan: sofia/internal/1001 at singtel Action set(call_timeout=30)
Dialplan: sofia/internal/1001 at singtel Action set(hangup_after_bridge=true)
Dialplan: sofia/internal/1001 at singtel Action set(bypass_media=false)
Dialplan: sofia/internal/1001 at singtel Action set(proxy_media=true)
Dialplan: sofia/internal/1001 at singtel Action bridge(h323/05492122@${NCX_IP})
2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:114
(sofia/internal/1001 at singtel) State Change CS_ROUTING -> CS_EXECUTE
2009-10-21 17:35:28.682475 [DEBUG] switch_core_session.c:932 Send signal
sofia/internal/1001 at singtel [BREAK]
2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:484
(sofia/internal/1001 at singtel) State ROUTING going to sleep
2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/1001 at singtel) Running State Change CS_EXECUTE
2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:491
(sofia/internal/1001 at singtel) State EXECUTE
2009-10-21 17:35:28.682475 [DEBUG] mod_sofia.c:173
sofia/internal/1001 at singtel SOFIA EXECUTE
2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:151
sofia/internal/1001 at singtel Standard EXECUTE
EXECUTE sofia/internal/1001 at singtel set(open=true)
2009-10-21 17:35:28.682475 [DEBUG] mod_dptools.c:748
sofia/internal/1001 at singtel SET [open]=[true]
EXECUTE sofia/internal/1001 at singtelhash(insert/10.4.62.7-spymap/1001/5c3ebda2-be57-11de-a6dd-e7de0b74bdc9)
EXECUTE sofia/internal/1001 at singtelhash(insert/10.4.62.7-last_dial/1001/05492122)
EXECUTE sofia/internal/1001 at singtelhash(insert/10.4.62.7-last_dial/global/5c3ebda2-be57-11de-a6dd-e7de0b74bdc9)
EXECUTE sofia/internal/1001 at singtelset(effective_caller_id_number=1001282122)
2009-10-21 17:35:28.685219 [DEBUG] mod_dptools.c:748
sofia/internal/1001 at singtel SET [effective_caller_id_number]=[1001282122]
EXECUTE sofia/internal/1001 at singtel set(NCX_IP=10.4.4.254)
2009-10-21 17:35:28.685219 [DEBUG] mod_dptools.c:748
sofia/internal/1001 at singtel SET [NCX_IP]=[10.4.4.254]
EXECUTE sofia/internal/1001 at singtel set(call_timeout=30)
2009-10-21 17:35:28.686292 [DEBUG] mod_dptools.c:748
sofia/internal/1001 at singtel SET [call_timeout]=[30]
EXECUTE sofia/internal/1001 at singtel set(hangup_after_bridge=true)
2009-10-21 17:35:28.686292 [DEBUG] mod_dptools.c:748
sofia/internal/1001 at singtel SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/1001 at singtel set(bypass_media=false)
2009-10-21 17:35:28.686292 [DEBUG] mod_dptools.c:748
sofia/internal/1001 at singtel SET [bypass_media]=[false]
EXECUTE sofia/internal/1001 at singtel set(proxy_media=true)
2009-10-21 17:35:28.686292 [DEBUG] mod_dptools.c:748
sofia/internal/1001 at singtel SET [proxy_media]=[true]
EXECUTE sofia/internal/1001 at singtel bridge(h323/05492122 at 10.4.4.254)
2009-10-21 17:35:28.687357 [DEBUG] mod_h323.cpp:1380
======>create_outgoing_channel DST NUMBER = 05492122 at 10.4.4.254
2009-10-21 17:35:28.687357 [DEBUG] mod_h323.cpp:1336
======>FSH323EndPoint::FSMakeCall DST NUMBER =
05492122 at 10.4.4.254[FSH323EndPoint]
2009-10-21 17:35:28.687357 [DEBUG] mod_h323.cpp:1348 ----> Using 10.4.62.7
for outbound call
2009-10-21 17:35:28.687357 [DEBUG] pdns.cxx:498 SRV Lookup 10.4.4.254
service _h323cs._tcp.
2009-10-21 17:35:28.689538 [DEBUG] pdns.cxx:471 SRV Lookup Fail no domain
h323:05492122 at 10.4.4.254 <h323%3A05492122 at 10.4.4.254>
2009-10-21 17:35:28.689538 [DEBUG] h323ep.cxx:2315 DNS SRV Cannot resolve
remote party 05492122 at 10.4.4.254
2009-10-21 17:35:28.689538 [DEBUG] h323ep.cxx:2103 Making call to:
05492122 at 10.4.4.254
2009-10-21 17:35:28.689538 [DEBUG] mod_h323.cpp:483
======>FSH323EndPoint::CreateConnection callReference = 12651 userDate =
0xb5e0dac0 [FSH323EndPoint]
2009-10-21 17:35:28.689538 [DEBUG] mod_h323.cpp:486 ------>
SWITCH_CALL_DIRECTION_OUTBOUND
2009-10-21 17:35:28.689538 [DEBUG] mod_h323.cpp:496 ------> fsSession =
0xb7569548
2009-10-21 17:35:28.689538 [DEBUG] h323caps.cxx:3186 Added capability:
G.711-ALaw-64k <1>
2009-10-21 17:35:28.689538 [DEBUG] h323caps.cxx:3186 Added capability:
G.711-uLaw-64k <2>
2009-10-21 17:35:28.689538 [DEBUG] h323caps.cxx:3186 Added capability:
GSM-06.10{sw} <3>
2009-10-21 17:35:28.689538 [DEBUG] h323caps.cxx:3186 Added capability:
G.729A/B{sw} <4>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
G.729A{sw} <5>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
G.729B{sw} <6>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
G.729{sw} <7>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
UserInput/hookflash <8>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
UserInput/basicString <9>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
UserInput/dtmf <10>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
UserInput/RFC2833 <11>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
UserInput/Navigation <12>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
UserInput/Softkey <13>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
UserInput/PointDevice <14>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
UserInput/Modal <15>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 1
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
G.711-ALaw-64k <1>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 2
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
G.711-uLaw-64k <2>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 3
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
GSM-06.10{sw} <3>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 4
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
G.729A/B{sw} <4>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 5
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
G.729A{sw} <5>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 6
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
G.729B{sw} <6>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 7
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
G.729{sw} <7>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 8
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
UserInput/hookflash <8>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 9
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
UserInput/basicString <9>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 10
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
UserInput/dtmf <10>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 11
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
UserInput/RFC2833 <11>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 12
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
UserInput/Navigation <12>
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 13
2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
UserInput/Softkey <13>
2009-10-21 17:35:28.691688 [DEBUG] h323caps.cxx:3248 FindCapability: 14
2009-10-21 17:35:28.691688 [DEBUG] h323caps.cxx:3252 Found capability:
UserInput/PointDevice <14>
2009-10-21 17:35:28.691688 [DEBUG] h323caps.cxx:3248 FindCapability: 15
2009-10-21 17:35:28.691688 [DEBUG] h323caps.cxx:3252 Found capability:
UserInput/Modal <15>
2009-10-21 17:35:28.691688 [DEBUG] h4601.cxx:1725 Endpoint Attached
2009-10-21 17:35:28.691688 [DEBUG] rfc2833.cxx:87 Handler created
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:525
======>FSH323Connection::FSH323Connection FSH323Connection
2009-10-21 17:35:28.691688 [NOTICE] switch_channel.c:602 New Channel h323/
05492122 at 10.4.4.254 [5c413af0-be57-11de-a6dd-e7de0b74bdc9]
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:546 (h323/
05492122 at 10.4.4.254) State Change CS_NEW -> CS_INIT
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:958
======>FSH323Connection::state_change FSH323Connection
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:959 State changed on
connection FSH323Connection
2009-10-21 17:35:28.691688 [DEBUG] switch_core_session.c:932 Send signal
h323/05492122 at 10.4.4.254 [BREAK]
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:831
======>FSH323Connection::kill_channel FSH323Connection
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:832 Kill 3 on connection
FSH323Connection
2009-10-21 17:35:28.691688 [DEBUG] h323ep.cxx:2183 Created new connection:
ip$localhost/12651
2009-10-21 17:35:28.691688 [DEBUG] osutil.cxx:188 File handle high water
mark set: 58 Thread unblock pipe
2009-10-21 17:35:28.691688 [DEBUG] h323ep.cxx:1101 Started call thread
2009-10-21 17:35:28.691688 [CRIT] switch_core_state_machine.c:239 Obtained
10 stack frames.
/usr/local/freeswitch/lib/libfreeswitch.so.1 [0xb7f511f5]
[0xb80be400]
/usr/local/freeswitch/mod/mod_h323.so [0xb6953cc9]
/usr/local/freeswitch/lib/libfreeswitch.so.1(switch_core_session_outgoing_channel+0x1ad)
[0xb7f4f59d]
/usr/local/freeswitch/lib/libfreeswitch.so.1(switch_ivr_originate+0x1b2b)
[0xb7f7aa1b]
/usr/local/freeswitch/mod/mod_dptools.so [0xb70ea2d5]
/usr/local/freeswitch/lib/libfreeswitch.so.1(switch_core_session_exec+0x284)
[0xb7f4c5a4]
/usr/local/freeswitch/lib/libfreeswitch.so.1(switch_core_session_execute_application+0x8b)
[0xb7f4cacb]
/usr/local/freeswitch/lib/libfreeswitch.so.1(switch_core_session_run+0x12b8)
[0xb7f52658]
/usr/local/freeswitch/lib/libfreeswitch.so.1 [0xb7f4e6c5]
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:509
======>FSH323EndPoint::OnSetGatewayPrefixes FSH323EndPoint
2009-10-21 17:35:28.691688 [CRIT] switch_core_state_machine.c:352 Thread has
crashed for channel sofia/internal/1001 at singtel
2009-10-21 17:35:28.691688 [NOTICE] switch_core_state_machine.c:353 Hangup
sofia/internal/1001 at singtel [CS_EXECUTE] [CRASH]
2009-10-21 17:35:28.691688 [DEBUG] transports.cxx:1828 Connecting to
10.4.4.254:1720 (local port=0)
2009-10-21 17:35:28.691688 [DEBUG] osutil.cxx:188 File handle high water
mark set: 59 PTCPSocket
2009-10-21 17:35:28.691688 [DEBUG] switch_channel.c:1683 Send signal
sofia/internal/1001 at singtel [KILL]
2009-10-21 17:35:28.691688 [DEBUG] switch_core_session.c:932 Send signal
sofia/internal/1001 at singtel [BREAK]
2009-10-21 17:35:28.691688 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/1001 at singtel) Running State Change CS_HANGUP
2009-10-21 17:35:28.691688 [DEBUG] transports.cxx:1681 Started connection:
host=10.4.4.254:1720, if=10.4.62.7:52984, handle=59
2009-10-21 17:35:28.691688 [DEBUG] h323.cxx:2680 Sending Setup PDU
2009-10-21 17:35:28.691688 [DEBUG] switch_core_state_machine.c:434
(sofia/internal/1001 at singtel) State HANGUP
2009-10-21 17:35:28.691688 [DEBUG] mod_sofia.c:338 Channel
sofia/internal/1001 at singtel hanging up, cause: CRASH
2009-10-21 17:35:28.691688 [DEBUG] mod_sofia.c:417 Responding to INVITE
with: 480
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:559
======>FSH323Connection::OnSetLocalCapabilities() [FSH323Connection]
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:565
======>FSH323Connection::SetLocalCapabilities() Size local capability = 15
[FSH323Connection]
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
======>FSH323Connection::decodeCapability
2009-10-21 17:35:28.691688 [INFO] mod_h323.cpp:619 capability
'G.711-ALaw-64k' format 'PCMU' payload 8
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
======>FSH323Connection::decodeCapability
2009-10-21 17:35:28.691688 [INFO] mod_h323.cpp:619 capability
'G.711-uLaw-64k' format 'PCMA' payload 0
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
======>FSH323Connection::decodeCapability
2009-10-21 17:35:28.691688 [INFO] mod_h323.cpp:619 capability 'GSM-06.10'
format 'GSM' payload 3
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
======>FSH323Connection::decodeCapability
2009-10-21 17:35:28.691688 [INFO] mod_h323.cpp:619 capability 'G.729A/B'
format 'G729' payload 18
2009-10-21 17:35:28.691688 [DEBUG] switch_core_state_machine.c:46
sofia/internal/1001 at singtel Standard HANGUP, cause: CRASH
2009-10-21 17:35:28.691688 [DEBUG] switch_core_state_machine.c:434
(sofia/internal/1001 at singtel) State HANGUP going to sleep
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
======>FSH323Connection::decodeCapability
2009-10-21 17:35:28.691688 [INFO] mod_h323.cpp:619 capability 'G.729A'
format 'G729' payload 18
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
======>FSH323Connection::decodeCapability
2009-10-21 17:35:28.691688 [INFO] mod_h323.cpp:619 capability 'G.729B'
format 'G729b' payload 18
2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
======>FSH323Connection::decodeCapability



Well, I'm not sure if the backtrace is from 1.0.4 ... i will disable
crass-protection and will send new logs to you.


Also, if you like i can give you access to the machine itself...

T.
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From jerry.richards at teotech.com  Wed Oct 21 09:11:04 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 21 Oct 2009 09:11:04 -0700
Subject: [Freeswitch-users] Stop/Restart of Freeswitch Causes Crash
Message-ID: <08086CB11D744A97A3FD87B383F43EA6@greyhawk.tonecommander.com>


Sometimes if I stop (using "..." command) and then restart freeswitch (using
"./freeswitch" command), the program will crash and return to the Linux
(CentOS 5.3) prompt.  I am using version 1.0.4.

I just pasted the freeswitch/terminal log into the Pastebin.

Best Regards,
Jerry




From diego.viola at gmail.com  Wed Oct 21 09:31:32 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Wed, 21 Oct 2009 16:31:32 +0000
Subject: [Freeswitch-users] Stop/Restart of Freeswitch Causes Crash
In-Reply-To: <08086CB11D744A97A3FD87B383F43EA6@greyhawk.tonecommander.com>
References: <08086CB11D744A97A3FD87B383F43EA6@greyhawk.tonecommander.com>
Message-ID: <86a32abc0910210931l36072b47ic92d8550f190bdeb@mail.gmail.com>

You should debug FreeSWITCH, check this out:

http://wiki.freeswitch.org/wiki/Debugging_Freeswitch

And then open a Jira.

Diego

On Wed, Oct 21, 2009 at 4:11 PM, Jerry Richards
<jerry.richards at teotech.com> wrote:
>
> Sometimes if I stop (using "..." command) and then restart freeswitch (using
> "./freeswitch" command), the program will crash and return to the Linux
> (CentOS 5.3) prompt. ?I am using version 1.0.4.
>
> I just pasted the freeswitch/terminal log into the Pastebin.
>
> Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From anthony.minessale at gmail.com  Wed Oct 21 09:33:55 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Oct 2009 11:33:55 -0500
Subject: [Freeswitch-users] Stop/Restart of Freeswitch Causes Crash
In-Reply-To: <08086CB11D744A97A3FD87B383F43EA6@greyhawk.tonecommander.com>
References: <08086CB11D744A97A3FD87B383F43EA6@greyhawk.tonecommander.com>
Message-ID: <191c3a030910210933l6217179bk866fedef1007d716@mail.gmail.com>

That problem is fixed if you update to SVN trunk or latest snapshot.


On Wed, Oct 21, 2009 at 11:11 AM, Jerry Richards <jerry.richards at teotech.com
> wrote:

>
> Sometimes if I stop (using "..." command) and then restart freeswitch
> (using
> "./freeswitch" command), the program will crash and return to the Linux
> (CentOS 5.3) prompt.  I am using version 1.0.4.
>
> I just pasted the freeswitch/terminal log into the Pastebin.
>
> Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Wed Oct 21 09:39:31 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Oct 2009 11:39:31 -0500
Subject: [Freeswitch-users] Call custom variable in condition
In-Reply-To: <65d96fc80910210413t1123dbc0xd5740bc0673d638c@mail.gmail.com>
References: <a60fed5e0910210337n45ffefe2wc3e3d9d0b403a568@mail.gmail.com>
	<65d96fc80910210413t1123dbc0xd5740bc0673d638c@mail.gmail.com>
Message-ID: <191c3a030910210939k7dbf1494gf713df8770356450@mail.gmail.com>

It not only makes sense it's well documented on the wiki page.
The set line is not happening right when it's encountered, the set line is
copied into the channel and executed later after the whole dialplan is
parsed.  The dialplan is a pre-processor not a runtime engine.

Here is a new feature in pre-1.0.5 (svn trunk)

Some applications like set can now be executed within the dialplan but you
should use it sparingly.
<action application="set" data="testing=true" inline="true"/>

The inline=true makes it execute inside the dialplan and it's never copied
into your resulting extension because it's executed immediately.



On Wed, Oct 21, 2009 at 6:13 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> consider this:
>
>
>
> <context name="SIP_incoming">
>    <extension name="call-sip-extensions">
>       <condition field="destination_number" expression="^(\d+)$">
>           <action application="set" data="AUTHENTICATION_STATUS=0"/>
>            <action application="transfer" data="${AUTHENTICATION_STATUS}
> XML Authen_Status"/>
>       </condition>
>    </extension>
> </context>
>
>
>
> <context name="Authen_Status">
>      <extension name="exten-auth-status">
>        <condition field="${AUTHENTICATION_STATUS}" expression="^0$">
>           <action application="answer"/>
>           <action application="playback" data="play.wav"/>
>       </condition>
>     </extension>
>   </context>
>
>
>
>
>
> here is one of my dialplan. I'm using execute_extension but it is quite the
> same...
>
>
>
>    <extension name="ServiceLookup">
>       <condition field="destination_number" expression="(^300030)(.*)">
>          <action application="lookup_service_destination" data="in
> ${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, in $
> 1, in ${network_addr}:5060, out red_contact, out authResult"/>
>          <action application="log" data="INFO ########################
> ServiceLookup ########################\n"/>
>          <action application="log" data="INFO ########################
> contact = '${red_contact}' ##############\n"/>
>          <action application="log" data="INFO ########################
> CallerNum = '${caller_id_number:6:16}' ##########\n"/>
>          <action application="log" data="INFO ########################
> RADIUS auth = '${authResult}' ##########\n"/>
>
>          <action application="execute_extension" data="doRedirect XML
> public"/>
>         </condition>
>    </extension>
>
>
>    <extension name="doRedirect">
>       <condition field="destination_number" expression="^doRedirect$"/>
>       <condition field="${authResult}" expression="^0$|^60$">
>          <action application="log" data="INFO ########################
> RADIUS auth OK!!!' ##########\n"/>
>          <action application="redirect" data="${red_contact}"/>
>          <anti-action application="log" data="INFO ########################
> RADIUS auth NOK!! ##########\n"/>
>          <anti-action application="respond" data="403 Forbidden"/>
>       </condition>
>
>    </extension>
>
>
>
>
> On Wed, Oct 21, 2009 at 12:37 PM, Ahmed Munir <ahmedmunir007 at gmail.com>wrote:
>
>> Hi,
>>
>> I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I
>> want to use it in condition, as I'm listing down the configuration below;
>>
>> <context name="SIP_incoming">
>>    <extension name="call-sip-extensions">
>>       <condition field="destination_number" expression="^(\d+)$">
>>           <action application="set" data="AUTHENTICATION_STATUS=0"/>
>>            <action application="transfer" data="${AUTHENTICATION_STATUS}
>> XML Authen_Status"/>
>>       </condition>
>>    </extension>
>> </context>
>>
>> <context name="Authen_Status">
>>      <extension name="exten-auth-status">
>>        <condition field="AUTHENTICATION_STATUS" expression="^0$">
>>           <action application="answer"/>
>>           <action application="playback" data="play.wav"/>
>>       </condition>
>>     </extension>
>>   </context>
>>
>>
>>
>>
>>  But unfortunately it is not working. Kindly advise me how to do implement
>> it(Note: I don't want to call script). And one more thing to ask how can I
>> transfer the values within the same context?
>>
>> --
>> Regards,
>>
>> Ahmed Munir
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From q.edward at gmail.com  Wed Oct 21 09:48:52 2009
From: q.edward at gmail.com (Edward Q.)
Date: Wed, 21 Oct 2009 12:48:52 -0400
Subject: [Freeswitch-users] Hi Guys
In-Reply-To: <35b355e90910202349q31bc8f7as6e277008600d81b0@mail.gmail.com>
References: <89313a90910202120v6c563660s61f2efba22f916f@mail.gmail.com> 
	<35b355e90910202255x680232f9x45319ab835f5846b@mail.gmail.com> 
	<89313a90910202309r6751b242q6b96a56cd1bfe57@mail.gmail.com> 
	<35b355e90910202349q31bc8f7as6e277008600d81b0@mail.gmail.com>
Message-ID: <89313a90910210948w5410e16ai1b7b48ee8fe1ac25@mail.gmail.com>

Thank you guysEd

On Wed, Oct 21, 2009 at 2:49 AM, Shelby Ramsey <sicfslist at gmail.com> wrote:

> Well ... the question is pretty generic.
>
> But based on these assumptions:
>   -- no media (bypass media)
>   -- routing done via XML dialplan
>
> Something along the lines of a quad core machine with 4 gigs of ram would
> be overkill for 692 calls.
>
> Things to remember:
>   -- the more cores the better (FS is heavily threaded)
>   -- the more memory the better
>   -- 64 bit is way better than 32 bit
>
> SDR
>
>
>
>
>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Wed Oct 21 09:54:34 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Oct 2009 11:54:34 -0500
Subject: [Freeswitch-users] Stop/Restart of Freeswitch Causes Crash
In-Reply-To: <86a32abc0910210931l36072b47ic92d8550f190bdeb@mail.gmail.com>
References: <08086CB11D744A97A3FD87B383F43EA6@greyhawk.tonecommander.com>
	<86a32abc0910210931l36072b47ic92d8550f190bdeb@mail.gmail.com>
Message-ID: <191c3a030910210954g74c2d5f8p3451841a8836147@mail.gmail.com>

no you shouldnt
we don't take jiras about release revisions only trunk.


On Wed, Oct 21, 2009 at 11:31 AM, Diego Viola <diego.viola at gmail.com> wrote:

> You should debug FreeSWITCH, check this out:
>
> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch
>
> And then open a Jira.
>
> Diego
>
> On Wed, Oct 21, 2009 at 4:11 PM, Jerry Richards
> <jerry.richards at teotech.com> wrote:
> >
> > Sometimes if I stop (using "..." command) and then restart freeswitch
> (using
> > "./freeswitch" command), the program will crash and return to the Linux
> > (CentOS 5.3) prompt.  I am using version 1.0.4.
> >
> > I just pasted the freeswitch/terminal log into the Pastebin.
> >
> > Best Regards,
> > Jerry
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From kaosfrack at gmail.com  Wed Oct 21 04:11:47 2009
From: kaosfrack at gmail.com (Joey Carter)
Date: Wed, 21 Oct 2009 16:11:47 +0500
Subject: [Freeswitch-users] Is background sound|music during call possible?
Message-ID: <700164f40910210411k6ce2795cud47202a25ce27f87@mail.gmail.com>

Hello

How can I add background music that will play during call?

Thanks



From keith at laaks.com  Wed Oct 21 09:45:19 2009
From: keith at laaks.com (Keith Laaks)
Date: Wed, 21 Oct 2009 18:45:19 +0200
Subject: [Freeswitch-users] Calls dropping on SIP timer expiry due to
	UPDATE's being ignored.
Message-ID: <1256143519.7836.107.camel@keithl-lt>

Hi,

Hope someone knows how I am able to get around this one. Here goes...

Did an upgrade to trunk (from a July vintage build) last week and
noticed calls out to a provider were now failing after about 30 seconds
or so - post answer. Tried latest (15183) - same thing.

Analysing, I see that I have multiple UPDATE messages now being sent to
the provider, but no response being sent back to FS. So FS times out and
eventually kills the call.
Interestingly, it only drops the A-leg; the B-leg remains up till the B
party hangs up.

I cant recall seeing these UPDATE messages before...

The intent of the UPDATE seems to be to send the callee name & number to
the B-leg.

If its the provider's sip stack that's broken w.r.t. handling UPDATE -
is there any way to get around it by doing something in my config to
ensure these UPDATE's are not 'triggered' ?


Some traces below. 


Any suggestions welcomed...

Best Regards

Keith
Pretoria, South Africa.

--------------------------------------------------------------------------------------------------------------------------------------------------


send 1048 bytes to udp/[196.10.11.12]:5060 at 13:24:04.249269:

------------------------------------------------------------------------
   INVITE sip:27835551111 at 196.10.11.12 SIP/2.0
   Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bKyQepDXQ5H8g5m
   Max-Forwards: 67
   From: "Keith PhoneADSL" <sip:878050000 at 10.17.10.10>;tag=Upa3NvXpBB1eF
   To: <sip:27835551111 at 196.10.11.12>
   Call-ID: d821359d-38e7-122d-a38e-002264cc9b93
   CSeq: 121947386 INVITE
   Contact: <sip:gw+vprov at 10.17.10.10:5060;transport=udp;gw=vprov>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 247
   X-Actually-Support: UPDATE
   Remote-Party-ID: "Keith PhoneADSL"
<sip:878050000 at 10.17.10.10>;party=calling;screen=yes;privacy=off
   
   v=0
   o=FreeSWITCH 1256118582 1256118583 IN IP4 10.17.10.10
   s=FreeSWITCH
   c=IN IP4 10.17.10.10
   t=0 0
   m=audio 12862 RTP/AVP 18 101
   a=rtpmap:18 G729/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20

------------------------------------------------------------------------
2009-10-21 15:24:04.248448 [DEBUG] sofia.c:3493 Channel
sofia/vvrf/27835551111 entering state [calling][0]
recv 601 bytes from udp/[196.10.11.12]:5060 at 13:24:04.307690:

------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bKyQepDXQ5H8g5m
   From: "Keith PhoneADSL" <sip:878050000 at 10.17.10.10>;tag=Upa3NvXpBB1eF
   To: <sip:27835551111 at 196.10.11.12>;tag=GR52RWG346-34
   Call-ID: d821359d-38e7-122d-a38e-002264cc9b93
   CSeq: 121947386 INVITE
   Contact: "vprov C5CM" <sip:196.10.11.12:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M
   Allow-Events: talk
   Allow-Events: refer
   Content-Disposition: session
   X-Actually-Support: UPDATE
   Remote-Party-ID: "Keith PhoneADSL"
<sip:878050000 at 10.17.10.10>;party=calling;screen=yes;privacy=off
   Content-Length: 0
   

------------------------------------------------------------------------
recv 879 bytes from udp/[196.10.11.12]:5060 at 13:24:08.508162:

------------------------------------------------------------------------
   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bKyQepDXQ5H8g5m
   From: "Keith PhoneADSL" <sip:878050000 at 10.17.10.10>;tag=Upa3NvXpBB1eF
   To: <sip:27835551111 at 196.10.11.12>;tag=GR52RWG346-34
   Call-ID: d821359d-38e7-122d-a38e-002264cc9b93
   CSeq: 121947386 INVITE
   Contact: "vprov C5CM" <sip:196.10.11.12:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M
   Allow-Events: talk
   Allow-Events: refer
   Content-Disposition: session
   X-Actually-Support: UPDATE
   Remote-Party-ID: "Keith PhoneADSL"
<sip:878050000 at 10.17.10.10>;party=calling;screen=yes;privacy=off
   Content-Type: application/sdp
   Content-Length:   233
   
   v=0
   o=Clarent 152602 152603 IN IP4 196.10.11.15
   s=Clarent C5CM
   c=IN IP4 196.10.11.15
   t=0 0
   m=audio 5230 RTP/AVP 18 101
   a=rtpmap:18 G729/8000
   a=ptime:20
   a=fmtp:18 annexb=no
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15

------------------------------------------------------------------------
2009-10-21 15:24:08.507506 [DEBUG] sofia.c:3493 Channel
sofia/vvrf/27835551111 entering state [proceeding][183]
2009-10-21 15:24:08.507506 [DEBUG] sofia.c:3500 Remote SDP:
v=0
o=Clarent 152602 152603 IN IP4 196.10.11.15
s=Clarent C5CM
c=IN IP4 196.10.11.15
t=0 0
m=audio 5230 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

2009-10-21 15:24:08.507506 [DEBUG] sofia_glue.c:3144 Audio Codec Compare
[G729:18:8000:20]/[G729:18:8000:20]
2009-10-21 15:24:08.508561 [DEBUG] sofia_glue.c:2102 Set Codec
sofia/vvrf/27835551111 G729/8000 20 ms 160 samples
2009-10-21 15:24:08.508561 [DEBUG] sofia_glue.c:3104 Set 2833 dtmf
payload to 101
2009-10-21 15:24:08.508561 [DEBUG] sofia_glue.c:2336 AUDIO RTP
[sofia/vvrf/27835551111] 10.17.10.10 port 12862 -> 196.10.11.15 port
5230 codec: 18 ms: 20
2009-10-21 15:24:08.508561 [DEBUG] switch_rtp.c:1155 Starting timer
[soft] 160 bytes per 20ms
2009-10-21 15:24:08.509616 [NOTICE] sofia_glue.c:2771 Pre-Answer
sofia/vvrf/27835551111!
2009-10-21 15:24:08.509616 [DEBUG] switch_channel.c:1992 Send signal
sofia/inetticky/27878050000 at 196.222.3.4 [BREAK]
2009-10-21 15:24:08.510671 [DEBUG] switch_ivr_originate.c:615 Setting
codec string on sofia/inetticky/27878050000 at 196.222.3.4 to
G729 at 8000h@20i
2009-10-21 15:24:08.510671 [DEBUG] switch_ivr_originate.c:2326
sofia/inetticky/27878050000 at 196.222.3.4 receive message [PROGRESS]
2009-10-21 15:24:08.510671 [INFO] switch_ivr_originate.c:2326 Sending
early media
2009-10-21 15:24:08.510671 [DEBUG] sofia_glue.c:3144 Audio Codec Compare
[G729:18:8000:0]/[G729:18:8000:20]
2009-10-21 15:24:08.510671 [DEBUG] sofia_glue.c:2102 Set Codec
sofia/inetticky/27878050000 at 196.222.3.4 G729/8000 20 ms 160 samples
2009-10-21 15:24:08.510671 [DEBUG] sofia_glue.c:3104 Set 2833 dtmf
payload to 101
2009-10-21 15:24:08.510671 [DEBUG] sofia_glue.c:2336 AUDIO RTP
[sofia/inetticky/27878050000 at 196.222.3.4] 196.222.3.4 port 16840 ->
41.242.6.167 port 62936 codec: 18 ms: 20
2009-10-21 15:24:08.510671 [DEBUG] switch_rtp.c:1155 Starting timer
[soft] 160 bytes per 20ms
2009-10-21 15:24:08.511769 [NOTICE] sofia_glue.c:2771 Pre-Answer
sofia/inetticky/27878050000 at 196.222.3.4!
2009-10-21 15:24:08.511769 [INFO] mod_sofia.c:1582 Ring SDP:
v=0
o=FreeSWITCH 1256114608 1256114609 IN IP4 196.222.3.4
s=FreeSWITCH
c=IN IP4 196.222.3.4
t=0 0
m=audio 16840 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2009-10-21 15:24:08.511769 [DEBUG] switch_core_session.c:712 Send signal
sofia/vvrf/27835551111 [BREAK]
2009-10-21 15:24:08.511769 [DEBUG] switch_core_session.c:653 Send signal
sofia/inetticky/27878050000 at 196.222.3.4 [BREAK]
2009-10-21 15:24:08.511769 [DEBUG] sofia.c:407 sofia/vvrf/27835551111
receive message [DISPLAY]
2009-10-21 15:24:08.511769 [DEBUG] switch_ivr_originate.c:2368 Originate
Resulted in Success: [sofia/vvrf/27835551111]
2009-10-21 15:24:08.511769 [DEBUG] switch_channel.c:182
sofia/vvrf/27835551111 receive message [AUDIO_SYNC]
2009-10-21 15:24:08.511769 [DEBUG] switch_channel.c:182
sofia/inetticky/27878050000 at 196.222.3.4 receive message [AUDIO_SYNC]
2009-10-21 15:24:08.512832 [DEBUG] switch_ivr_bridge.c:975
sofia/vvrf/27835551111 receive message [BRIDGE]
2009-10-21 15:24:08.512832 [DEBUG] switch_core_session.c:653 Send signal
sofia/vvrf/27835551111 [BREAK]
2009-10-21 15:24:08.512832 [DEBUG] switch_ivr_bridge.c:982
sofia/inetticky/27878050000 at 196.222.3.4 receive message [BRIDGE]
send 973 bytes to udp/[41.242.6.167]:62928 at 13:24:08.512974:

------------------------------------------------------------------------
   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP
41.242.6.167:56675;branch=z9hG4bK2234c0462cdad03fc318bd6c7438282;rport=629282009-10-21 15:24:08.512832 [DEBUG] switch_core_session.c:653 Send signal sofia/inetticky/27878050000 at 196.222.3.4 [BREAK]
2009-10-21 15:24:08.512832 [DEBUG] switch_ivr_bridge.c:1026
(sofia/vvrf/27835551111) State Change CS_CONSUME_MEDIA ->
CS_EXCHANGE_MEDIA
2009-10-21 15:24:08.512832 [DEBUG] switch_core_session.c:985 Send signal
sofia/vvrf/27835551111 [BREAK]
2009-10-21 15:24:08.512832 [DEBUG] switch_core_state_machine.c:306
(sofia/vvrf/27835551111) Running State Change CS_EXCHANGE_MEDIA
2009-10-21 15:24:08.512832 [DEBUG] switch_core_state_machine.c:343
(sofia/vvrf/27835551111) State EXCHANGE_MEDIA
2009-10-21 15:24:08.512832 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK

   From: "Keith PhoneADSL" <sip:27878050000 at 196.222.3.4>;tag=3283506127
   To: <sip:0835551111 at 196.222.3.4>;user=phone;tag=g2DFFpS0QN4cc
   Call-ID: 4252084844 at 192_168_1_65
   CSeq: 3 INVITE
   Contact: <sip:0835551111 at 196.222.3.4:5060;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 249
   X-Actually-Support: UPDATE
   
   v=0
   o=FreeSWITCH 1256114608 1256114609 IN IP4 196.222.3.4
   s=FreeSWITCH
   c=IN IP4 196.222.3.4
   t=0 0
   m=audio 16840 RTP/AVP 18 101
   a=rtpmap:18 G729/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20

------------------------------------------------------------------------
2009-10-21 15:24:08.512832 [DEBUG] sofia.c:3493 Channel
sofia/inetticky/27878050000 at 196.222.3.4 entering state [early][183]
recv 928 bytes from udp/[196.10.11.12]:5060 at 13:24:14.188781:

------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bKyQepDXQ5H8g5m
   From: "Keith PhoneADSL" <sip:878050000 at 10.17.10.10>;tag=Upa3NvXpBB1eF
   To: <sip:27835551111 at 196.10.11.12>;tag=GR52RWG346-34
   Call-ID: d821359d-38e7-122d-a38e-002264cc9b93
   CSeq: 121947386 INVITE
   Contact: "vprov C5CM" <sip:196.10.11.12:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M
   Allow-Events: talk
   Allow-Events: refer
   Content-Disposition: session
   X-Actually-Support: UPDATE
   Remote-Party-ID: "Keith PhoneADSL"
<sip:878050000 at 10.17.10.10>;party=calling;screen=yes;privacy=off
   Allow: ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE
   Content-Type: application/sdp
   Content-Length:   233
   
   v=0
   o=Clarent 153288 153288 IN IP4 196.10.11.15
   s=Clarent C5CM
   c=IN IP4 196.10.11.15
   t=0 0
   m=audio 5230 RTP/AVP 18 101
   a=rtpmap:18 G729/8000
   a=ptime:20
   a=fmtp:18 annexb=no
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15

------------------------------------------------------------------------
send 396 bytes to udp/[196.10.11.12]:5060 at 13:24:14.189032:

------------------------------------------------------------------------
   ACK sip:196.10.11.12:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bKZ07eFr88eH7Qg
   Max-Forwards: 70
   From: "Keith PhoneADSL" <sip:878050000 at 10.17.10.10>;tag=Upa3NvXpBB1eF
   To: <sip:27835551111 at 196.10.11.12>;tag=GR52RWG346-34
   Call-ID: d821359d-38e7-122d-a38e-002264cc9b93
   CSeq: 121947386 ACK
   Contact: <sip:gw+vprov at 10.17.10.10:5060;transport=udp;gw=vprov>
   Content-Length: 0
   

------------------------------------------------------------------------
2009-10-21 15:24:14.188910 [DEBUG] sofia.c:3493 Channel
sofia/vvrf/27835551111 entering state [ready][200]
2009-10-21 15:24:14.188910 [DEBUG] switch_channel.c:2105 Send signal
sofia/inetticky/27878050000 at 196.222.3.4 [BREAK]
2009-10-21 15:24:14.188910 [NOTICE] sofia.c:3928 Channel
[sofia/vvrf/27835551111] has been answered
2009-10-21 15:24:14.188910 [DEBUG] switch_channel.c:182
sofia/vvrf/27835551111 receive message [AUDIO_SYNC]
2009-10-21 15:24:14.204823 [DEBUG] switch_ivr_bridge.c:378
sofia/inetticky/27878050000 at 196.222.3.4 receive message [ANSWER]
2009-10-21 15:24:14.204823 [DEBUG] mod_sofia.c:543 Local SDP
sofia/inetticky/27878050000 at 196.222.3.4:
v=0
o=FreeSWITCH 1256114608 1256114610 IN IP4 196.222.3.4
s=FreeSWITCH
c=IN IP4 196.222.3.4
t=0 0
m=audio 16840 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2009-10-21 15:24:14.204823 [DEBUG] switch_core_session.c:712 Send signal
sofia/vvrf/27835551111 [BREAK]
2009-10-21 15:24:14.204823 [DEBUG] switch_core_session.c:653 Send signal
sofia/inetticky/27878050000 at 196.222.3.4 [BREAK]
2009-10-21 15:24:14.204823 [NOTICE] switch_ivr_bridge.c:378 Channel
[sofia/inetticky/27878050000 at 196.222.3.4] has been answered
2009-10-21 15:24:14.204823 [DEBUG] switch_channel.c:182
sofia/inetticky/27878050000 at 196.222.3.4 receive message [AUDIO_SYNC]
send 983 bytes to udp/[41.242.6.167]:62928 at 13:24:14.205173:

------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP
41.242.6.167:56675;branch=z9hG4bK2234c0462cdad03fc318bd6c7438282;rport=62928
   From: "Keith PhoneADSL" <sip:27878050000 at 196.222.3.4>;tag=3283506127
   To: <sip:0835551111 at 196.222.3.4>;user=phone;tag=g2DFFpS0QN4cc
   Call-ID: 4252084844 at 192_168_1_65
   CSeq: 3 INVITE
   Contact: <sip:0835551111 at 196.222.3.4:5060;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, refer
   Session-Expires: 120;refresher=uas
   Min-SE: 120
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 249
   X-Actually-Support: UPDATE
   
   v=0
   o=FreeSWITCH 1256114608 1256114609 IN IP4 196.222.3.4
   s=FreeSWITCH
   c=IN IP4 196.222.3.4
   t=0 0
   m=audio 16840 RTP/AVP 18 101
   a=rtpmap:18 G729/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20

------------------------------------------------------------------------
2009-10-21 15:24:14.204823 [DEBUG] sofia.c:3493 Channel
sofia/inetticky/27878050000 at 196.222.3.4 entering state [completed][200]
2009-10-21 15:24:14.224959 [DEBUG] sofia.c:407 sofia/vvrf/27835551111
receive message [DISPLAY]
send 947 bytes to udp/[196.10.11.12]:5060 at 13:24:14.225162:

------------------------------------------------------------------------
   UPDATE sip:196.10.11.12:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bK0907gKScctXac
   Max-Forwards: 70
   From: "Keith PhoneADSL" <sip:878050000 at 10.17.10.10>;tag=Upa3NvXpBB1eF
   To: <sip:27835551111 at 196.10.11.12>;tag=GR52RWG346-34
   Call-ID: d821359d-38e7-122d-a38e-002264cc9b93
   CSeq: 121947387 UPDATE
   Contact: <sip:gw+vprov at 10.17.10.10:5060;transport=udp;gw=vprov>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY
   Supported: timer, precondition, path, replaces
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 247
   P-Asserted-Identity: "unknown" <27835551111>
   
   v=0
   o=FreeSWITCH 1256118582 1256118583 IN IP4 10.17.10.10
   s=FreeSWITCH
   c=IN IP4 10.17.10.10
   t=0 0
   m=audio 12862 RTP/AVP 18 101
   a=rtpmap:18 G729/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20

------------------------------------------------------------------------
2009-10-21 15:24:14.224959 [DEBUG] sofia.c:3493 Channel
sofia/vvrf/27835551111 entering state [calling][0]
recv 733 bytes from udp/[41.242.6.167]:62928 at 13:24:14.533488:

------------------------------------------------------------------------
   ACK sip:0835551111 at 196.222.3.4:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP
41.242.6.167:56675;branch=z9hG4bKd2c91e1afbf0880332fbeba5fc9c04a6;rport
   From: "Keith PhoneADSL" <sip:27878050000 at 196.222.3.4>;tag=3283506127
   To: <sip:0835551111 at 196.222.3.4>;user=phone;tag=g2DFFpS0QN4cc
   Call-ID: 4252084844 at 192_168_1_65
   CSeq: 3 ACK
   Contact: <sip:27878050000 at 41.242.6.167:56675>
   Proxy-Authorization: Digest username="27878050000",
realm="196.222.3.4", qop=auth, algorithm=MD5,
uri="sip:0835551111 at 196.222.3.4",
nonce="0092f192-be45-11de-a2c1-a70e5037c20e", nc=00000001,
cnonce="ab2d49987adb74997fc6b236fc16d23",
response="445ae74c1f910e45d9aa1043402b4dbb"
   Max-Forwards: 70
   User-Agent: C455 IP021910000000
   Content-Length: 0
   

------------------------------------------------------------------------
2009-10-21 15:24:14.533538 [DEBUG] sofia.c:3493 Channel
sofia/inetticky/27878050000 at 196.222.3.4 entering state [ready][200]
2009-10-21 15:24:14.545161 [DEBUG] switch_core_session.c:712 Send signal
sofia/vvrf/27835551111 [BREAK]
2009-10-21 15:24:14.545161 [DEBUG] switch_core_session.c:712 Send signal
sofia/inetticky/27878050000 at 196.222.3.4 [BREAK]
2009-10-21 15:24:14.565231 [DEBUG] switch_ivr_bridge.c:122
sofia/vvrf/27835551111 receive message [DISPLAY]
2009-10-21 15:24:14.565231 [DEBUG] switch_ivr_bridge.c:122
sofia/inetticky/27878050000 at 196.222.3.4 receive message [DISPLAY]
send 947 bytes to udp/[196.10.11.12]:5060 at 13:24:14.727678:

------------------------------------------------------------------------
   UPDATE sip:196.10.11.12:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bK0907gKScctXac
   Max-Forwards: 70
   From: "Keith PhoneADSL" <sip:878050000 at 10.17.10.10>;tag=Upa3NvXpBB1eF
   To: <sip:27835551111 at 196.10.11.12>;tag=GR52RWG346-34
   Call-ID: d821359d-38e7-122d-a38e-002264cc9b93
   CSeq: 121947387 UPDATE
   Contact: <sip:gw+vprov at 10.17.10.10:5060;transport=udp;gw=vprov>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY
   Supported: timer, precondition, path, replaces
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 247
   P-Asserted-Identity: "unknown" <27835551111>
   
   v=0
   o=FreeSWITCH 1256118582 1256118583 IN IP4 10.17.10.10
   s=FreeSWITCH
   c=IN IP4 10.17.10.10
   t=0 0
   m=audio 12862 RTP/AVP 18 101
   a=rtpmap:18 G729/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20

------------------------------------------------------------------------
send 947 bytes to udp/[196.10.11.12]:5060 at 13:24:15.728255:

------------------------------------------------------------------------
   UPDATE sip:196.10.11.12:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bK0907gKScctXac
   Max-Forwards: 70
   From: "Keith PhoneADSL" <sip:878050000 at 10.17.10.10>;tag=Upa3NvXpBB1eF
   To: <sip:27835551111 at 196.10.11.12>;tag=GR52RWG346-34
   Call-ID: d821359d-38e7-122d-a38e-002264cc9b93
   CSeq: 121947387 UPDATE
   Contact: <sip:gw+vprov at 10.17.10.10:5060;transport=udp;gw=vprov>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY
   Supported: timer, precondition, path, replaces
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 247
   P-Asserted-Identity: "unknown" <27835551111>
   
   v=0
   o=FreeSWITCH 1256118582 1256118583 IN IP4 10.17.10.10
   s=FreeSWITCH
   c=IN IP4 10.17.10.10
   t=0 0
   m=audio 12862 RTP/AVP 18 101
   a=rtpmap:18 G729/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20

[Many more copies of the UPDATE's continue  ... then ]

2009-10-21 15:24:46.228024 [DEBUG] sofia.c:3493 Channel
sofia/vvrf/27835551111 entering state [terminated][408]
2009-10-21 15:24:46.228024 [NOTICE] sofia.c:4039 Hangup
sofia/vvrf/27835551111 [CS_EXCHANGE_MEDIA] [RECOVERY_ON_TIMER_EXPIRE]
2009-10-21 15:24:46.228024 [DEBUG] switch_channel.c:1896 Send signal
sofia/vvrf/27835551111 [KILL]  

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From msc at freeswitch.org  Wed Oct 21 10:02:48 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 21 Oct 2009 10:02:48 -0700
Subject: [Freeswitch-users] Call custom variable in condition
In-Reply-To: <191c3a030910210939k7dbf1494gf713df8770356450@mail.gmail.com>
References: <a60fed5e0910210337n45ffefe2wc3e3d9d0b403a568@mail.gmail.com>
	<65d96fc80910210413t1123dbc0xd5740bc0673d638c@mail.gmail.com>
	<191c3a030910210939k7dbf1494gf713df8770356450@mail.gmail.com>
Message-ID: <87f2f3b90910211002y7c35c622g4c76472fd8c7bada@mail.gmail.com>

On Wed, Oct 21, 2009 at 9:39 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> It not only makes sense it's well documented on the wiki page.
> The set line is not happening right when it's encountered, the set line is
> copied into the channel and executed later after the whole dialplan is
> parsed.  The dialplan is a pre-processor not a runtime engine.
>
> Here is a new feature in pre-1.0.5 (svn trunk)
>
> Some applications like set can now be executed within the dialplan but you
> should use it sparingly.
> <action application="set" data="testing=true" inline="true"/>
>

I'm getting ready to document this feature. For the sake of edification, why
is it best to use this sparingly, other than wide-spread use making
dialplans all cluttered?
-MC


>
> The inline=true makes it execute inside the dialplan and it's never copied
> into your resulting extension because it's executed immediately.
>
>
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From diego.viola at gmail.com  Wed Oct 21 10:04:07 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Wed, 21 Oct 2009 17:04:07 +0000
Subject: [Freeswitch-users] Stop/Restart of Freeswitch Causes Crash
In-Reply-To: <191c3a030910210954g74c2d5f8p3451841a8836147@mail.gmail.com>
References: <08086CB11D744A97A3FD87B383F43EA6@greyhawk.tonecommander.com>
	<86a32abc0910210931l36072b47ic92d8550f190bdeb@mail.gmail.com>
	<191c3a030910210954g74c2d5f8p3451841a8836147@mail.gmail.com>
Message-ID: <86a32abc0910211004u1acfb9a2hb963d63e0c7eeab9@mail.gmail.com>

Oh ok, I didn't knew that Anthony, sorry.

Do as Anthony said, update to trunk :).

On Wed, Oct 21, 2009 at 4:54 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> no you shouldnt
> we don't take jiras about release revisions only trunk.
>
>
> On Wed, Oct 21, 2009 at 11:31 AM, Diego Viola <diego.viola at gmail.com> wrote:
>>
>> You should debug FreeSWITCH, check this out:
>>
>> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch
>>
>> And then open a Jira.
>>
>> Diego
>>
>> On Wed, Oct 21, 2009 at 4:11 PM, Jerry Richards
>> <jerry.richards at teotech.com> wrote:
>> >
>> > Sometimes if I stop (using "..." command) and then restart freeswitch
>> > (using
>> > "./freeswitch" command), the program will crash and return to the Linux
>> > (CentOS 5.3) prompt. ?I am using version 1.0.4.
>> >
>> > I just pasted the freeswitch/terminal log into the Pastebin.
>> >
>> > Best Regards,
>> > Jerry
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From anthony.minessale at gmail.com  Wed Oct 21 10:10:08 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Oct 2009 12:10:08 -0500
Subject: [Freeswitch-users] Call custom variable in condition
In-Reply-To: <87f2f3b90910211002y7c35c622g4c76472fd8c7bada@mail.gmail.com>
References: <a60fed5e0910210337n45ffefe2wc3e3d9d0b403a568@mail.gmail.com>
	<65d96fc80910210413t1123dbc0xd5740bc0673d638c@mail.gmail.com>
	<191c3a030910210939k7dbf1494gf713df8770356450@mail.gmail.com>
	<87f2f3b90910211002y7c35c622g4c76472fd8c7bada@mail.gmail.com>
Message-ID: <191c3a030910211010sfb5fceco31775d3dbacfa181@mail.gmail.com>

you should not abuse it is all i mean, we have measures to limit what apps
you can use in this manner but usually requiring a more complicated dialplan
is a hint you are doing something wrong ;)

On Wed, Oct 21, 2009 at 12:02 PM, Michael Collins <msc at freeswitch.org>wrote:

>
>
> On Wed, Oct 21, 2009 at 9:39 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> It not only makes sense it's well documented on the wiki page.
>> The set line is not happening right when it's encountered, the set line is
>> copied into the channel and executed later after the whole dialplan is
>> parsed.  The dialplan is a pre-processor not a runtime engine.
>>
>> Here is a new feature in pre-1.0.5 (svn trunk)
>>
>> Some applications like set can now be executed within the dialplan but you
>> should use it sparingly.
>> <action application="set" data="testing=true" inline="true"/>
>>
>
> I'm getting ready to document this feature. For the sake of edification,
> why is it best to use this sparingly, other than wide-spread use making
> dialplans all cluttered?
> -MC
>
>
>>
>> The inline=true makes it execute inside the dialplan and it's never copied
>> into your resulting extension because it's executed immediately.
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mike at jerris.com  Wed Oct 21 10:29:32 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 21 Oct 2009 13:29:32 -0400
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <7b8592a0910210458s28ddffbdi77cf116d33c28405@mail.gmail.com>
References: <FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
	<20091020133437.GA4941@dan.olp.net>
	<7b8592a0910200716q4a7db93cyb955595e4993ac96@mail.gmail.com>
	<858EC056-1D84-4F30-A0B7-9A70FF98695D@freeswitch.org>
	<7b8592a0910200821m18fe9b4dg73dda05a10699af2@mail.gmail.com>
	<642B6CED-CDF3-4D40-B493-0F2D6222C30B@freeswitch.org>
	<7b8592a0910200947k6ca5a391l35786f0e855ae388@mail.gmail.com>
	<20091021040343.GA9440@jdc.jasonjgw.net>
	<7b8592a0910202336x1bf2a8a2nd636219ec7031c5a@mail.gmail.com>
	<20091021065841.GA13150@jdc.jasonjgw.net>
	<7b8592a0910210458s28ddffbdi77cf116d33c28405@mail.gmail.com>
Message-ID: <67234FAF-73DE-462E-B7E8-4BDE922D7CB8@jerris.com>

This appears to be some sort of ice implementation?  We don't support  
sip ice at this time.

Mike

On Oct 21, 2009, at 7:58 AM, ineya ineya wrote:

> Codecs are fine. I spent much time experimenting with codecs and
> completely missed, that freeswitch is modifiyng the SDP record.
>
> When phone A is making a call the SDP contains candidate media  
> attributes:
>
> a=candidate:123abc 1 UDP 9 2000:2::1001 5012 typ host
> a=candidate:123abc 1 UDP 8 10.80.62.92 5010 typ host
>
> But when freeswitch makes the INVITE on phone B, these 2 are missing
> and phone is looking for it, so the INVITE gets rejected by phone with
> "448 Not acceptable here"
>
> So the question is, how can I make the freeswitch to pass these
> candidate media attributes?
>
> On Wed, Oct 21, 2009 at 8:58 AM, Jason White <jason at jasonjgw.net>  
> wrote:
>
>>
>> I suspect the codec negotiation. Make sure that both ends are  
>> offering a
>> common codec.
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From bottleman at icf.org.ru  Wed Oct 21 10:28:51 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Wed, 21 Oct 2009 21:28:51 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910200908460.2416@srv.icf.org.ru>
	<65d96fc80910200117j4e88e7ax19dc442ae0bb441d@mail.gmail.com>
	<Pine.LNX.4.64.0910201841030.12352@srv.icf.org.ru>
	<65d96fc80910210034nd7e8864hfcc421ec6a6c52f@mail.gmail.com>
	<Pine.LNX.4.64.0910211427520.9175@srv.icf.org.ru>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>

On 2009-10-21 17:48 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>FS = 10.4.62.7
TC>SIP phone = 10.4.62.89
TC>H323 endpoint = 10.1.14.153
TC>
TC>
TC>
TC>> TC>2. hangup from sip side doesn't release the h323 leg (now the difference
TC>> is
TC>> TC>that FS is not complaining about thread mismatch ant it looks clean but
TC>> FS
TC>> TC>doesn't send any releasecomplete message... strange)
TC>> TC>3. coredumps when i place outgoing calls
TC>>
TC>> btw,
TC>>
TC>> TC> 70.552157  10.1.14.153 -> 10.4.62.7    H.245 endSessionCommand
TC>> TC> 70.552401  10.1.14.153 -> 10.4.62.7    H.225.0 CS: releaseComplete
TC>> TC> 70.553981    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 3
TC>> TC> 70.554488    10.4.62.7 -> 10.1.14.153  H.245 endSessionCommand
TC>> TC> 70.555056    10.4.62.7 -> 10.1.14.153  H.225.0 CS: releaseComplete
TC>>
TC>> it send, now i have no way to test h323->sip transit, i will have it
TC>> tomorow.
TC>> sip->h323 for me work fine now, give backtrace from code dump of 1.0.4
TC>> where it die?
TC>>
TC>>
TC>
TC>this endSession is when i hangup from H232 side as well :P ... if i don't
TC>hangup on H323 side the H323 leg is not released. Pls chec the time the
TC>packets were sent ...
TC>
TC>
TC>
TC>Here i hangup on the SIP Phone:
TC>
TC>68.374916   10.4.62.89 -> 10.4.62.7    SIP Request: BYE
TC>sip:mod_sofia at 10.4.62.7:5060
TC>68.375342    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 3
TC>68.375620    10.4.62.7 -> 10.4.62.89   SIP Status: 200 OK
TC>
TC>2 sec delay
TC>
TC>Here i hangup on the H323 endpoint (releaseComplete comes from H323 endpoint
TC>first here )
TC>70.552157  10.1.14.153 -> 10.4.62.7    H.245 endSessionCommand
TC>70.552401  10.1.14.153 -> 10.4.62.7    H.225.0 CS: releaseComplete
TC>
TC>
TC>FS just acknowlages it here:
TC>70.554488    10.4.62.7 -> 10.1.14.153  H.245 endSessionCommand
TC>70.555056    10.4.62.7 -> 10.1.14.153  H.225.0 CS: releaseComplete
TC>
TC>
TC>
TC>I have enabled crash-protection and when i do SIP => H323 call it doesn't
TC>generate coredumps... it is just this thread that is crashing ... pls check
TC>the log downbelow:

core dump in case enabled crash-protection may be unusable, i have a case then 
my module crash silently, after this crash-protection is killing sip leg and after
this i get core dump where backtrace show me segfault in libc6, i spent one day to
understand this situation, and then i disable crash-protection i see there is actualy 
it crashes. disable crash-protection and show backtrace of crash, i think result will
be different.

TC>
TC>Dialplan: sofia/internal/1001 at singtel Regex (FAIL) [EMERGENCY]
TC>destination_number(05492122) =~ /^0(112|9[23456])$/ break=on-false
TC>Dialplan: sofia/internal/1001 at singtel parsing [default->SPECIAL_SERVICES]
TC>continue=false
TC>Dialplan: sofia/internal/1001 at singtel Regex (FAIL) [SPECIAL_SERVICES]
TC>destination_number(05492122) =~ /^0(9[01789]\d{3,4})$/ break=on-false
TC>Dialplan: sofia/internal/1001 at singtel parsing [default->ENYTHING_ELSE]
TC>continue=false
TC>Dialplan: sofia/internal/1001 at singtel Regex (PASS) [ENYTHING_ELSE]
TC>destination_number(05492122) =~
TC>/^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$/ break=on-false
TC>Dialplan: sofia/internal/1001 at singtel Action
TC>set(effective_caller_id_number=1001282122)
TC>Dialplan: sofia/internal/1001 at singtel Action set(NCX_IP=10.4.4.254)
TC>Dialplan: sofia/internal/1001 at singtel Action set(call_timeout=30)
TC>Dialplan: sofia/internal/1001 at singtel Action set(hangup_after_bridge=true)
TC>Dialplan: sofia/internal/1001 at singtel Action set(bypass_media=false)
TC>Dialplan: sofia/internal/1001 at singtel Action set(proxy_media=true)
TC>Dialplan: sofia/internal/1001 at singtel Action bridge(h323/05492122@${NCX_IP})
TC>2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:114
TC>(sofia/internal/1001 at singtel) State Change CS_ROUTING -> CS_EXECUTE
TC>2009-10-21 17:35:28.682475 [DEBUG] switch_core_session.c:932 Send signal
TC>sofia/internal/1001 at singtel [BREAK]
TC>2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:484
TC>(sofia/internal/1001 at singtel) State ROUTING going to sleep
TC>2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:398
TC>(sofia/internal/1001 at singtel) Running State Change CS_EXECUTE
TC>2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:491
TC>(sofia/internal/1001 at singtel) State EXECUTE
TC>2009-10-21 17:35:28.682475 [DEBUG] mod_sofia.c:173
TC>sofia/internal/1001 at singtel SOFIA EXECUTE
TC>2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:151
TC>sofia/internal/1001 at singtel Standard EXECUTE
TC>EXECUTE sofia/internal/1001 at singtel set(open=true)
TC>2009-10-21 17:35:28.682475 [DEBUG] mod_dptools.c:748
TC>sofia/internal/1001 at singtel SET [open]=[true]
TC>EXECUTE sofia/internal/1001 at singtelhash(insert/10.4.62.7-spymap/1001/5c3ebda2-be57-11de-a6dd-e7de0b74bdc9)
TC>EXECUTE sofia/internal/1001 at singtelhash(insert/10.4.62.7-last_dial/1001/05492122)
TC>EXECUTE sofia/internal/1001 at singtelhash(insert/10.4.62.7-last_dial/global/5c3ebda2-be57-11de-a6dd-e7de0b74bdc9)
TC>EXECUTE sofia/internal/1001 at singtelset(effective_caller_id_number=1001282122)
TC>2009-10-21 17:35:28.685219 [DEBUG] mod_dptools.c:748
TC>sofia/internal/1001 at singtel SET [effective_caller_id_number]=[1001282122]
TC>EXECUTE sofia/internal/1001 at singtel set(NCX_IP=10.4.4.254)
TC>2009-10-21 17:35:28.685219 [DEBUG] mod_dptools.c:748
TC>sofia/internal/1001 at singtel SET [NCX_IP]=[10.4.4.254]
TC>EXECUTE sofia/internal/1001 at singtel set(call_timeout=30)
TC>2009-10-21 17:35:28.686292 [DEBUG] mod_dptools.c:748
TC>sofia/internal/1001 at singtel SET [call_timeout]=[30]
TC>EXECUTE sofia/internal/1001 at singtel set(hangup_after_bridge=true)
TC>2009-10-21 17:35:28.686292 [DEBUG] mod_dptools.c:748
TC>sofia/internal/1001 at singtel SET [hangup_after_bridge]=[true]
TC>EXECUTE sofia/internal/1001 at singtel set(bypass_media=false)
TC>2009-10-21 17:35:28.686292 [DEBUG] mod_dptools.c:748
TC>sofia/internal/1001 at singtel SET [bypass_media]=[false]
TC>EXECUTE sofia/internal/1001 at singtel set(proxy_media=true)
TC>2009-10-21 17:35:28.686292 [DEBUG] mod_dptools.c:748
TC>sofia/internal/1001 at singtel SET [proxy_media]=[true]
TC>EXECUTE sofia/internal/1001 at singtel bridge(h323/05492122 at 10.4.4.254)
TC>2009-10-21 17:35:28.687357 [DEBUG] mod_h323.cpp:1380
TC>======>create_outgoing_channel DST NUMBER = 05492122 at 10.4.4.254
TC>2009-10-21 17:35:28.687357 [DEBUG] mod_h323.cpp:1336
TC>======>FSH323EndPoint::FSMakeCall DST NUMBER =
TC>05492122 at 10.4.4.254[FSH323EndPoint]
TC>2009-10-21 17:35:28.687357 [DEBUG] mod_h323.cpp:1348 ----> Using 10.4.62.7
TC>for outbound call
TC>2009-10-21 17:35:28.687357 [DEBUG] pdns.cxx:498 SRV Lookup 10.4.4.254
TC>service _h323cs._tcp.
TC>2009-10-21 17:35:28.689538 [DEBUG] pdns.cxx:471 SRV Lookup Fail no domain
TC>h323:05492122 at 10.4.4.254 <h323%3A05492122 at 10.4.4.254>
TC>2009-10-21 17:35:28.689538 [DEBUG] h323ep.cxx:2315 DNS SRV Cannot resolve
TC>remote party 05492122 at 10.4.4.254
TC>2009-10-21 17:35:28.689538 [DEBUG] h323ep.cxx:2103 Making call to:
TC>05492122 at 10.4.4.254
TC>2009-10-21 17:35:28.689538 [DEBUG] mod_h323.cpp:483
TC>======>FSH323EndPoint::CreateConnection callReference = 12651 userDate =
TC>0xb5e0dac0 [FSH323EndPoint]
TC>2009-10-21 17:35:28.689538 [DEBUG] mod_h323.cpp:486 ------>
TC>SWITCH_CALL_DIRECTION_OUTBOUND
TC>2009-10-21 17:35:28.689538 [DEBUG] mod_h323.cpp:496 ------> fsSession =
TC>0xb7569548
TC>2009-10-21 17:35:28.689538 [DEBUG] h323caps.cxx:3186 Added capability:
TC>G.711-ALaw-64k <1>
TC>2009-10-21 17:35:28.689538 [DEBUG] h323caps.cxx:3186 Added capability:
TC>G.711-uLaw-64k <2>
TC>2009-10-21 17:35:28.689538 [DEBUG] h323caps.cxx:3186 Added capability:
TC>GSM-06.10{sw} <3>
TC>2009-10-21 17:35:28.689538 [DEBUG] h323caps.cxx:3186 Added capability:
TC>G.729A/B{sw} <4>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
TC>G.729A{sw} <5>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
TC>G.729B{sw} <6>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
TC>G.729{sw} <7>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
TC>UserInput/hookflash <8>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
TC>UserInput/basicString <9>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
TC>UserInput/dtmf <10>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
TC>UserInput/RFC2833 <11>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
TC>UserInput/Navigation <12>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
TC>UserInput/Softkey <13>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
TC>UserInput/PointDevice <14>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3186 Added capability:
TC>UserInput/Modal <15>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 1
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
TC>G.711-ALaw-64k <1>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 2
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
TC>G.711-uLaw-64k <2>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 3
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
TC>GSM-06.10{sw} <3>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 4
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
TC>G.729A/B{sw} <4>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 5
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
TC>G.729A{sw} <5>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 6
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
TC>G.729B{sw} <6>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 7
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
TC>G.729{sw} <7>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 8
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
TC>UserInput/hookflash <8>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 9
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
TC>UserInput/basicString <9>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 10
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
TC>UserInput/dtmf <10>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 11
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
TC>UserInput/RFC2833 <11>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 12
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
TC>UserInput/Navigation <12>
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3248 FindCapability: 13
TC>2009-10-21 17:35:28.690630 [DEBUG] h323caps.cxx:3252 Found capability:
TC>UserInput/Softkey <13>
TC>2009-10-21 17:35:28.691688 [DEBUG] h323caps.cxx:3248 FindCapability: 14
TC>2009-10-21 17:35:28.691688 [DEBUG] h323caps.cxx:3252 Found capability:
TC>UserInput/PointDevice <14>
TC>2009-10-21 17:35:28.691688 [DEBUG] h323caps.cxx:3248 FindCapability: 15
TC>2009-10-21 17:35:28.691688 [DEBUG] h323caps.cxx:3252 Found capability:
TC>UserInput/Modal <15>
TC>2009-10-21 17:35:28.691688 [DEBUG] h4601.cxx:1725 Endpoint Attached
TC>2009-10-21 17:35:28.691688 [DEBUG] rfc2833.cxx:87 Handler created
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:525
TC>======>FSH323Connection::FSH323Connection FSH323Connection
TC>2009-10-21 17:35:28.691688 [NOTICE] switch_channel.c:602 New Channel h323/
TC>05492122 at 10.4.4.254 [5c413af0-be57-11de-a6dd-e7de0b74bdc9]
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:546 (h323/
TC>05492122 at 10.4.4.254) State Change CS_NEW -> CS_INIT
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:958
TC>======>FSH323Connection::state_change FSH323Connection
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:959 State changed on
TC>connection FSH323Connection
TC>2009-10-21 17:35:28.691688 [DEBUG] switch_core_session.c:932 Send signal
TC>h323/05492122 at 10.4.4.254 [BREAK]
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:831
TC>======>FSH323Connection::kill_channel FSH323Connection
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:832 Kill 3 on connection
TC>FSH323Connection
TC>2009-10-21 17:35:28.691688 [DEBUG] h323ep.cxx:2183 Created new connection:
TC>ip$localhost/12651
TC>2009-10-21 17:35:28.691688 [DEBUG] osutil.cxx:188 File handle high water
TC>mark set: 58 Thread unblock pipe
TC>2009-10-21 17:35:28.691688 [DEBUG] h323ep.cxx:1101 Started call thread
TC>2009-10-21 17:35:28.691688 [CRIT] switch_core_state_machine.c:239 Obtained
TC>10 stack frames.
TC>/usr/local/freeswitch/lib/libfreeswitch.so.1 [0xb7f511f5]
TC>[0xb80be400]
TC>/usr/local/freeswitch/mod/mod_h323.so [0xb6953cc9]
TC>/usr/local/freeswitch/lib/libfreeswitch.so.1(switch_core_session_outgoing_channel+0x1ad)
TC>[0xb7f4f59d]
TC>/usr/local/freeswitch/lib/libfreeswitch.so.1(switch_ivr_originate+0x1b2b)
TC>[0xb7f7aa1b]
TC>/usr/local/freeswitch/mod/mod_dptools.so [0xb70ea2d5]
TC>/usr/local/freeswitch/lib/libfreeswitch.so.1(switch_core_session_exec+0x284)
TC>[0xb7f4c5a4]
TC>/usr/local/freeswitch/lib/libfreeswitch.so.1(switch_core_session_execute_application+0x8b)
TC>[0xb7f4cacb]
TC>/usr/local/freeswitch/lib/libfreeswitch.so.1(switch_core_session_run+0x12b8)
TC>[0xb7f52658]
TC>/usr/local/freeswitch/lib/libfreeswitch.so.1 [0xb7f4e6c5]
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:509
TC>======>FSH323EndPoint::OnSetGatewayPrefixes FSH323EndPoint
TC>2009-10-21 17:35:28.691688 [CRIT] switch_core_state_machine.c:352 Thread has
TC>crashed for channel sofia/internal/1001 at singtel
TC>2009-10-21 17:35:28.691688 [NOTICE] switch_core_state_machine.c:353 Hangup
TC>sofia/internal/1001 at singtel [CS_EXECUTE] [CRASH]
TC>2009-10-21 17:35:28.691688 [DEBUG] transports.cxx:1828 Connecting to
TC>10.4.4.254:1720 (local port=0)
TC>2009-10-21 17:35:28.691688 [DEBUG] osutil.cxx:188 File handle high water
TC>mark set: 59 PTCPSocket
TC>2009-10-21 17:35:28.691688 [DEBUG] switch_channel.c:1683 Send signal
TC>sofia/internal/1001 at singtel [KILL]
TC>2009-10-21 17:35:28.691688 [DEBUG] switch_core_session.c:932 Send signal
TC>sofia/internal/1001 at singtel [BREAK]
TC>2009-10-21 17:35:28.691688 [DEBUG] switch_core_state_machine.c:398
TC>(sofia/internal/1001 at singtel) Running State Change CS_HANGUP
TC>2009-10-21 17:35:28.691688 [DEBUG] transports.cxx:1681 Started connection:
TC>host=10.4.4.254:1720, if=10.4.62.7:52984, handle=59
TC>2009-10-21 17:35:28.691688 [DEBUG] h323.cxx:2680 Sending Setup PDU
TC>2009-10-21 17:35:28.691688 [DEBUG] switch_core_state_machine.c:434
TC>(sofia/internal/1001 at singtel) State HANGUP
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_sofia.c:338 Channel
TC>sofia/internal/1001 at singtel hanging up, cause: CRASH
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_sofia.c:417 Responding to INVITE
TC>with: 480
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:559
TC>======>FSH323Connection::OnSetLocalCapabilities() [FSH323Connection]
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:565
TC>======>FSH323Connection::SetLocalCapabilities() Size local capability = 15
TC>[FSH323Connection]
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
TC>======>FSH323Connection::decodeCapability
TC>2009-10-21 17:35:28.691688 [INFO] mod_h323.cpp:619 capability
TC>'G.711-ALaw-64k' format 'PCMU' payload 8
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
TC>======>FSH323Connection::decodeCapability
TC>2009-10-21 17:35:28.691688 [INFO] mod_h323.cpp:619 capability
TC>'G.711-uLaw-64k' format 'PCMA' payload 0
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
TC>======>FSH323Connection::decodeCapability
TC>2009-10-21 17:35:28.691688 [INFO] mod_h323.cpp:619 capability 'GSM-06.10'
TC>format 'GSM' payload 3
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
TC>======>FSH323Connection::decodeCapability
TC>2009-10-21 17:35:28.691688 [INFO] mod_h323.cpp:619 capability 'G.729A/B'
TC>format 'G729' payload 18
TC>2009-10-21 17:35:28.691688 [DEBUG] switch_core_state_machine.c:46
TC>sofia/internal/1001 at singtel Standard HANGUP, cause: CRASH
TC>2009-10-21 17:35:28.691688 [DEBUG] switch_core_state_machine.c:434
TC>(sofia/internal/1001 at singtel) State HANGUP going to sleep
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
TC>======>FSH323Connection::decodeCapability
TC>2009-10-21 17:35:28.691688 [INFO] mod_h323.cpp:619 capability 'G.729A'
TC>format 'G729' payload 18
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
TC>======>FSH323Connection::decodeCapability
TC>2009-10-21 17:35:28.691688 [INFO] mod_h323.cpp:619 capability 'G.729B'
TC>format 'G729b' payload 18
TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
TC>======>FSH323Connection::decodeCapability
TC>
TC>
TC>
TC>Well, I'm not sure if the backtrace is from 1.0.4 ... i will disable
TC>crass-protection and will send new logs to you.
TC>
TC>
TC>Also, if you like i can give you access to the machine itself...
TC>
TC>T.
TC>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From brian at freeswitch.org  Wed Oct 21 10:30:59 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Oct 2009 12:30:59 -0500
Subject: [Freeswitch-users] Calls dropping on SIP timer expiry due to
	UPDATE's being ignored.
In-Reply-To: <1256143519.7836.107.camel@keithl-lt>
References: <1256143519.7836.107.camel@keithl-lt>
Message-ID: <4FEF9AF8-14F2-4248-ACD5-E711CDE7D017@freeswitch.org>

This will be fixed soon. Watch SVN.

/b

On Oct 21, 2009, at 11:45 AM, Keith Laaks wrote:

> Hi,



From brian at freeswitch.org  Wed Oct 21 10:31:25 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Oct 2009 12:31:25 -0500
Subject: [Freeswitch-users] Is background sound|music during call
	possible?
In-Reply-To: <700164f40910210411k6ce2795cud47202a25ce27f87@mail.gmail.com>
References: <700164f40910210411k6ce2795cud47202a25ce27f87@mail.gmail.com>
Message-ID: <3893EF02-247C-4C38-8F56-15E2C6A7D19D@freeswitch.org>

uuid_displace with mux option.

/b

On Oct 21, 2009, at 6:11 AM, Joey Carter wrote:

> Hello
>
> How can I add background music that will play during call?
>
> Thanks




From helmut.kuper at ewetel.de  Wed Oct 21 10:43:23 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Wed, 21 Oct 2009 19:43:23 +0200
Subject: [Freeswitch-users] Qustion about INFO messages
	after	Connect/Answer
In-Reply-To: <87f2f3b90910191435s7528e256r85df5f12fc634f24@mail.gmail.com>
References: <4AD859A4.5000101@ewetel.de>	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>	<4ADC7C09.6070503@ewetel.de>	<191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>	<4ADC85C0.7060908@ewetel.de>
	<4ADC8B6E.7070404@ewetel.de>	<191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>
	<87f2f3b90910191435s7528e256r85df5f12fc634f24@mail.gmail.com>
Message-ID: <4ADF483B.4050807@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello Mike,

just updated my prod system. The "1/a/" problem is solved with Anthony's
"originate_callee_id_name" chvar.

thanks alot :)

So, last thing of this thread is still the "unknown" thing on callee's
display, which is (by now) NOT affected by the new chvars.

regards
Helmut


On 19.10.2009 23:35, Michael Collins wrote:
> 
> 
> On Mon, Oct 19, 2009 at 1:07 PM, Anthony Minessale
> <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>> wrote:
> 
>     please update and test trunk
> 
>     1) I changed the core to remove the excess data by default in your
>     scenario
>     2) I added variables you can use to control it
>     origination_callee_id_name origination_callee_id_number which belong
>     in {} in the dial string eg
>     {origination_callee_id_number=1234}openzap/1/a/1234
> 
> 
> After you test, please confirm the behavior and then we'll update the
> wiki on these two new chan vars.
> -MC
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)

iD8DBQFK30g74tZeNddg3dwRAjaGAKDDNnxPPg+lmlCSs33MCw/V191q3ACdFlpv
Alf3NeoCA8Qbm2PZ1k2HHOg=
=hzVn
-----END PGP SIGNATURE-----



From msc at freeswitch.org  Wed Oct 21 10:44:58 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 21 Oct 2009 10:44:58 -0700
Subject: [Freeswitch-users] NOT in dialplan expression
In-Reply-To: <b39273530910210807j1f8a2fc6l4ff2bf1c154497b9@mail.gmail.com>
References: <33c87fa30910210434t268e8026pfd2b5439f26069fd@mail.gmail.com>
	<b39273530910210807j1f8a2fc6l4ff2bf1c154497b9@mail.gmail.com>
Message-ID: <87f2f3b90910211044t1a02a653ya3bb9e1911099daa@mail.gmail.com>

On Wed, Oct 21, 2009 at 8:07 AM, Rupa Schomaker <rupa at rupa.com> wrote:

> You can also look at using anti-action rather than action after the
> condition.
>
> condition == if
> action == then
> anti-action == else
>

Rupa & Leon,

Nice job of explaining the options. Sometimes we forget about the powerful
constructs that are available in FreeSWITCH and PCRE. karma++ for both of
you. :)
-MC
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From msc at freeswitch.org  Wed Oct 21 10:48:06 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 21 Oct 2009 10:48:06 -0700
Subject: [Freeswitch-users] Qustion about INFO messages after
	Connect/Answer
In-Reply-To: <4ADF483B.4050807@ewetel.de>
References: <4AD859A4.5000101@ewetel.de>
	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>
	<4ADC7C09.6070503@ewetel.de>
	<191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>
	<4ADC85C0.7060908@ewetel.de> <4ADC8B6E.7070404@ewetel.de>
	<191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>
	<87f2f3b90910191435s7528e256r85df5f12fc634f24@mail.gmail.com>
	<4ADF483B.4050807@ewetel.de>
Message-ID: <87f2f3b90910211048s208daf1eyfae501091ab75530@mail.gmail.com>

On Wed, Oct 21, 2009 at 10:43 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello Mike,
>
> just updated my prod system. The "1/a/" problem is solved with Anthony's
> "originate_callee_id_name" chvar.
>
> thanks alot :)
>
> So, last thing of this thread is still the "unknown" thing on callee's
> display, which is (by now) NOT affected by the new chvars.
>
Okay, you are able to reproduce that "unknown" thing? Can you pastebin a
fresh debug log w/ SIP trace on, plus and relevant dp changes from the
default dialplan?
Thanks,
MC
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From jerry.richards at teotech.com  Wed Oct 21 11:37:46 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 21 Oct 2009 11:37:46 -0700
Subject: [Freeswitch-users] 3rd Party Dial Plan Tool
Message-ID: <840EFA6312584A7498E5AB88459851DC@greyhawk.tonecommander.com>


Can anyone recommend a good 3rd party dialplan tool that will work with
Freeswitch?

Best Regards,
Jerry




From anthony.minessale at gmail.com  Wed Oct 21 11:54:17 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Oct 2009 13:54:17 -0500
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
In-Reply-To: <2d9149cd0910202050i36a7ea82nc7253d87e38f990a@mail.gmail.com>
References: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>
	<191c3a030910201434q4c6e65e1h58c5f0a82addc2d3@mail.gmail.com>
	<2d9149cd0910202050i36a7ea82nc7253d87e38f990a@mail.gmail.com>
Message-ID: <191c3a030910211154gc8412b5gd999fbedffa02b53@mail.gmail.com>

can you try trunk and let me know right away,
if it's still not working i may need ssh access and call you on the phone.


On Tue, Oct 20, 2009 at 10:50 PM, Kristian Kielhofner <
kristian.kielhofner at gmail.com> wrote:

> Done.
>
> On Tue, Oct 20, 2009 at 5:34 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > issue:
> > console loglevel debug
> > sofia profile internal siptrace on
> >
> > and put it on pastebin
> > http://pastebin.freeswitch.org
> >
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From brian at freeswitch.org  Wed Oct 21 12:00:12 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Oct 2009 14:00:12 -0500
Subject: [Freeswitch-users] 3rd Party Dial Plan Tool
In-Reply-To: <840EFA6312584A7498E5AB88459851DC@greyhawk.tonecommander.com>
References: <840EFA6312584A7498E5AB88459851DC@greyhawk.tonecommander.com>
Message-ID: <5067C824-5FB3-412A-9653-9A872DA9204A@freeswitch.org>

none specifically exist... good ole trusty editor?

/b
PS: 


On Oct 21, 2009, at 1:37 PM, Jerry Richards wrote:

>
> Can anyone recommend a good 3rd party dialplan tool that will work  
> with
> Freeswitch?
>
> Best Regards,
> Jerry




From sprice at gmail.com  Wed Oct 21 12:10:25 2009
From: sprice at gmail.com (SP)
Date: Wed, 21 Oct 2009 14:10:25 -0500
Subject: [Freeswitch-users] 3rd Party Dial Plan Tool
In-Reply-To: <5067C824-5FB3-412A-9653-9A872DA9204A@freeswitch.org>
References: <840EFA6312584A7498E5AB88459851DC@greyhawk.tonecommander.com>
	<5067C824-5FB3-412A-9653-9A872DA9204A@freeswitch.org>
Message-ID: <7e2ac3270910211210r2ea53cafr66058b38c393fe92@mail.gmail.com>

You could also try consulting at freeswitch.org

On Wed, Oct 21, 2009 at 14:00, Brian West <brian at freeswitch.org> wrote:
> none specifically exist... good ole trusty editor?
>
> /b
> PS: 
>
>
> On Oct 21, 2009, at 1:37 PM, Jerry Richards wrote:
>
>>
>> Can anyone recommend a good 3rd party dialplan tool that will work
>> with
>> Freeswitch?
>>
>> Best Regards,
>> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Shannon



From mike at jerris.com  Wed Oct 21 12:50:51 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 21 Oct 2009 15:50:51 -0400
Subject: [Freeswitch-users] sched_api doesn't get launched
In-Reply-To: <59ad9ca10910210244q5410f2bdq3b93c25f95673d11@mail.gmail.com>
References: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
	<3F1747F9-C543-4C6F-BB01-4DD5FA5B7BC5@jerris.com>
	<59ad9ca10910161153h5d531533vd844e49b9ca6d896@mail.gmail.com>
	<87f2f3b90910161237h356b7aa2q6ec12c6ecf78dc34@mail.gmail.com>
	<59ad9ca10910210244q5410f2bdq3b93c25f95673d11@mail.gmail.com>
Message-ID: <033669F7-EE8B-455C-A8DE-FD4EE564F34C@jerris.com>

The syntax is different, but the api is the same as lua:

> So you need an API object in order to use it. I don't know the  
> syntax for creating an api obj in Java but in Lua it goes like this:
> api = freeswitch.API();
> res = api:execute("sched_api","+300 none my_api my_api_args")

create the API object and use the execute method of it.

Mike


On Oct 21, 2009, at 5:44 AM, Henry Huang wrote:

> I can't seem to find the right thing to use in mod_java to execute  
> api commands, only api_after_bridge
>
>
> 2009-10-21 17:42:46.593094 [NOTICE] mod_sofia.c:1509 Pre-Answer  
> sofia/internal/1688003 at 192.168.1.66!
> #
> # A fatal error has been detected by the Java Runtime Environment:
> #
> #  SIGSEGV (0xb) at pc=0x004e4480, pid=1927, tid=16116624
> #
> # JRE version: 6.0_16-b01
> # Java VM: Java HotSpot(TM) Client VM (14.2-b01 mixed mode linux-x86 )
> # Problematic frame:
> # C  [libc.so.6+0x6f480]  strcpy+0x10
> #
> # An error report file with more information is saved as:
> # /usr/local/freeswitch/bin/hs_err_pid1927.log
> 2009-10-21 17:42:59.883729 [ERR] switch_core_session.c:1374 Invalid  
> Application sched_api
> 2009-10-21 17:42:59.883729 [NOTICE] switch_core_session.c:1375  
> Hangup sofia/internal/1688003 at 192.168.1.66 [CS_EXECUTE]  
> [DESTINATION_OUT_OF_ORDER]
> #
> # If you would like to submit a bug report, please visit:
> #   http://java.sun.com/webapps/bugreport/crash.jsp
> # The crash happened outside the Java Virtual Machine in native code.
> # See problematic frame for where to report the bug.
>
>
> On Sat, Oct 17, 2009 at 3:37 AM, Michael Collins  
> <msc at freeswitch.org> wrote:
>
>
> On Fri, Oct 16, 2009 at 11:53 AM, Henry Huang <red.rain.seven at gmail.com 
> > wrote:
> So how would you trigger it from a script dialplan? The only time it  
> seemed to work is when I did setVariable("api_after_bridge",  
> "sched_api blah blah blah");
> but then it gets executed after the channel's been teared down. I  
> thought api_after_bridge means right after the call gets connected.
>
> I need something to execute an api command right before or right  
> after the call gets bridged.
>
> api_after_bridge is a channel variable, so using setVariable works  
> just fine. If you need to sched_api is an API only. Check these out:
> http://wiki.freeswitch.org/wiki/Mod_commands#Misc._Commands
>
> So you need an API object in order to use it. I don't know the  
> syntax for creating an api obj in Java but in Lua it goes like this:
> api = freeswitch.API();
> res = api:execute("sched_api","+300 none my_api my_api_args")
>
> Remember, if the method you are using isn't found in the dial plan  
> tools then it isn't a dial plan application. Make sure it's on the  
> list:
> http://wiki.freeswitch.org/wiki/Mod_dptools
>
> On the other hand, API commands are listed here:
>
> http://wiki.freeswitch.org/wiki/Mod_commands
>
> dptools require a session object, api commands require an api  
> object...
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Henry Huang
> UniC Solution - Communication Unified
> VoIP & Open Source software Consultant
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From ineyaa at gmail.com  Wed Oct 21 13:08:20 2009
From: ineyaa at gmail.com (ineya ineya)
Date: Wed, 21 Oct 2009 22:08:20 +0200
Subject: [Freeswitch-users] can't dial from IPv6 to IPv6
In-Reply-To: <67234FAF-73DE-462E-B7E8-4BDE922D7CB8@jerris.com>
References: <FD95F391-2470-4663-B627-59CF6B9F84D5@freeswitch.org>
	<858EC056-1D84-4F30-A0B7-9A70FF98695D@freeswitch.org>
	<7b8592a0910200821m18fe9b4dg73dda05a10699af2@mail.gmail.com>
	<642B6CED-CDF3-4D40-B493-0F2D6222C30B@freeswitch.org>
	<7b8592a0910200947k6ca5a391l35786f0e855ae388@mail.gmail.com>
	<20091021040343.GA9440@jdc.jasonjgw.net>
	<7b8592a0910202336x1bf2a8a2nd636219ec7031c5a@mail.gmail.com>
	<20091021065841.GA13150@jdc.jasonjgw.net>
	<7b8592a0910210458s28ddffbdi77cf116d33c28405@mail.gmail.com>
	<67234FAF-73DE-462E-B7E8-4BDE922D7CB8@jerris.com>
Message-ID: <7b8592a0910211308w597c29a3m3afc9953266ca41b@mail.gmail.com>

Yes, I think you are right, the sip stack comes from company called
M5T as far as I know, and there are references to ICE on their site.

On Wed, Oct 21, 2009 at 7:29 PM, Michael Jerris <mike at jerris.com> wrote:
> This appears to be some sort of ice implementation? ?We don't support
> sip ice at this time.
>



From anthony.minessale at gmail.com  Wed Oct 21 13:09:24 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Oct 2009 15:09:24 -0500
Subject: [Freeswitch-users] sched_api doesn't get launched
In-Reply-To: <59ad9ca10910210244q5410f2bdq3b93c25f95673d11@mail.gmail.com>
References: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
	<3F1747F9-C543-4C6F-BB01-4DD5FA5B7BC5@jerris.com>
	<59ad9ca10910161153h5d531533vd844e49b9ca6d896@mail.gmail.com>
	<87f2f3b90910161237h356b7aa2q6ec12c6ecf78dc34@mail.gmail.com>
	<59ad9ca10910210244q5410f2bdq3b93c25f95673d11@mail.gmail.com>
Message-ID: <191c3a030910211309sb727b37g6324e1833f2a1e8c@mail.gmail.com>

Yes you need an API object as described in other email.

Which line of code from java caused that segfault
It looks like a simple NULL string issue that we may want to hunt down.

On Wed, Oct 21, 2009 at 4:44 AM, Henry Huang <red.rain.seven at gmail.com>wrote:

> I can't seem to find the right thing to use in mod_java to execute api
> commands, only api_after_bridge
>
> 2009-10-21 17:42:46.593094 [NOTICE] mod_sofia.c:1509 Pre-Answer
> sofia/internal/1688003 at 192.168.1.66!
> #
> # A fatal error has been detected by the Java Runtime Environment:
> #
> #  SIGSEGV (0xb) at pc=0x004e4480, pid=1927, tid=16116624
> #
> # JRE version: 6.0_16-b01
> # Java VM: Java HotSpot(TM) Client VM (14.2-b01 mixed mode linux-x86 )
> # Problematic frame:
> # C  [libc.so.6+0x6f480]  strcpy+0x10
> #
> # An error report file with more information is saved as:
> # /usr/local/freeswitch/bin/hs_err_pid1927.log
> 2009-10-21 17:42:59.883729 [ERR] switch_core_session.c:1374 Invalid
> Application sched_api
> 2009-10-21 17:42:59.883729 [NOTICE] switch_core_session.c:1375 Hangup
> sofia/internal/1688003 at 192.168.1.66 [CS_EXECUTE]
> [DESTINATION_OUT_OF_ORDER]
> #
> # If you would like to submit a bug report, please visit:
> #   http://java.sun.com/webapps/bugreport/crash.jsp
> # The crash happened outside the Java Virtual Machine in native code.
> # See problematic frame for where to report the bug.
>
>
> On Sat, Oct 17, 2009 at 3:37 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Fri, Oct 16, 2009 at 11:53 AM, Henry Huang <red.rain.seven at gmail.com>wrote:
>>
>>> So how would you trigger it from a script dialplan? The only time it
>>> seemed to work is when I did setVariable("api_after_bridge", "sched_api blah
>>> blah blah");
>>> but then it gets executed after the channel's been teared down. I thought
>>> api_after_bridge means right after the call gets connected.
>>>
>>> I need something to execute an api command right before or right after
>>> the call gets bridged.
>>>
>>> api_after_bridge is a channel variable, so using setVariable works just
>> fine. If you need to sched_api is an API only. Check these out:
>> http://wiki.freeswitch.org/wiki/Mod_commands#Misc._Commands
>>
>> So you need an API object in order to use it. I don't know the syntax for
>> creating an api obj in Java but in Lua it goes like this:
>> api = freeswitch.API();
>> res = api:execute("sched_api","+300 none my_api my_api_args")
>>
>> Remember, if the method you are using isn't found in the dial plan tools
>> then it isn't a dial plan application. Make sure it's on the list:
>> http://wiki.freeswitch.org/wiki/Mod_dptools
>>
>> On the other hand, API commands are listed here:
>> http://wiki.freeswitch.org/wiki/Mod_commands
>>
>> dptools require a session object, api commands require an api object...
>>
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Henry Huang
> UniC Solution - Communication Unified
> VoIP & Open Source software Consultant
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From mike at jerris.com  Wed Oct 21 15:05:56 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 21 Oct 2009 18:05:56 -0400
Subject: [Freeswitch-users] Calls dropping on SIP timer expiry due to
	UPDATE's being ignored.
In-Reply-To: <1256143519.7836.107.camel@keithl-lt>
References: <1256143519.7836.107.camel@keithl-lt>
Message-ID: <94C69E51-6577-43A7-AB5B-70BF1AFE12DD@jerris.com>

This should now be fixed in latest svn trunk.

Mike

On Oct 21, 2009, at 12:45 PM, Keith Laaks wrote:

> Hi,
>
> Hope someone knows how I am able to get around this one. Here goes...
>
> Did an upgrade to trunk (from a July vintage build) last week and  
> noticed calls out to a provider were now failing after about 30  
> seconds or so - post answer. Tried latest (15183) - same thing.
>
> Analysing, I see that I have multiple UPDATE messages now being sent  
> to the provider, but no response being sent back to FS. So FS times  
> out and eventually kills the call.
> Interestingly, it only drops the A-leg; the B-leg remains up till  
> the B party hangs up.
>
> I cant recall seeing these UPDATE messages before...
>
> The intent of the UPDATE seems to be to send the callee name &  
> number to the B-leg.
>
> If its the provider's sip stack that's broken w.r.t. handling UPDATE  
> - is there any way to get around it by doing something in my config  
> to ensure these UPDATE's are not 'triggered' ?




From mcampbellsmith at gmail.com  Wed Oct 21 15:51:47 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 22 Oct 2009 09:51:47 +1100
Subject: [Freeswitch-users] NOT in dialplan expression
In-Reply-To: <87f2f3b90910211044t1a02a653ya3bb9e1911099daa@mail.gmail.com>
References: <33c87fa30910210434t268e8026pfd2b5439f26069fd@mail.gmail.com>
	<b39273530910210807j1f8a2fc6l4ff2bf1c154497b9@mail.gmail.com>
	<87f2f3b90910211044t1a02a653ya3bb9e1911099daa@mail.gmail.com>
Message-ID: <33c87fa30910211551m40a21daai3405b7c32da6655d@mail.gmail.com>

Thanks Guys for your suggestions!  Very much appreciated.

The reason I asked was that I want to check if a variable is set or
not.  I don't think your suggestions will work because I actually have
to check to see an undefined variable.  Any way to do this?

I want to have the same extension for checking digits read from DTMF.
The problem is that the digits are not set to the variable until the
transfer statement, which means I would require 2 extensions.   It
will make things simpler if I can have the same extension that reads
the digits and then checks them.

For example:

 <extension name="checkdigits">
      <condition field="destination_number" expression="^checkdigits$"/>
       <!--The next condition should be true if the variable $digits
is not set/undefined -->
        <condition field="${digits}" expression="^[^.+]$" />
-----<<<< what should be here to check for undefined variable $digits
?
           <action application="read" data="1 10 ivr/ivr-hello.wav
digits 10000 #"/>
           <action application="phrase" data="spell,${digits}"/>
           <action application="transfer" data="checkdigits digits XML
features"/>
      </condition>

      <condition field="${pincode}" expression="^${some_variable}$">
             < SOME ACTIONS HERE >
      </condition>
 </extension>

On Thu, Oct 22, 2009 at 4:44 AM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Wed, Oct 21, 2009 at 8:07 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>
>> You can also look at using anti-action rather than action after the
>> condition.
>>
>> condition == if
>> action == then
>> anti-action == else
>
> Rupa & Leon,
>
> Nice job of explaining the options. Sometimes we forget about the powerful
> constructs that are available in FreeSWITCH and PCRE. karma++ for both of
> you. :)
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From msc at freeswitch.org  Wed Oct 21 17:02:29 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 21 Oct 2009 17:02:29 -0700
Subject: [Freeswitch-users] NOT in dialplan expression
In-Reply-To: <33c87fa30910211551m40a21daai3405b7c32da6655d@mail.gmail.com>
References: <33c87fa30910210434t268e8026pfd2b5439f26069fd@mail.gmail.com>
	<b39273530910210807j1f8a2fc6l4ff2bf1c154497b9@mail.gmail.com>
	<87f2f3b90910211044t1a02a653ya3bb9e1911099daa@mail.gmail.com>
	<33c87fa30910211551m40a21daai3405b7c32da6655d@mail.gmail.com>
Message-ID: <87f2f3b90910211702x159836f3p6b3d222341856bc6@mail.gmail.com>

On Wed, Oct 21, 2009 at 3:51 PM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> Thanks Guys for your suggestions!  Very much appreciated.
>
> The reason I asked was that I want to check if a variable is set or
> not.  I don't think your suggestions will work because I actually have
> to check to see an undefined variable.  Any way to do this?
>

Not in the dialplan itself. Better off using a script for this.

>
> I want to have the same extension for checking digits read from DTMF.
> The problem is that the digits are not set to the variable until the
> transfer statement, which means I would require 2 extensions.   It
> will make things simpler if I can have the same extension that reads
> the digits and then checks them.
>

You need either to use multiple extensions or use a script. The dialplan was
specifically designed not to be a programming language and the functionality
you seek is best served by using a script or by using transfer to drop the
call through the diaplan again and use another extension.
-MC


>
> For example:
>
>  <extension name="checkdigits">
>      <condition field="destination_number" expression="^checkdigits$"/>
>       <!--The next condition should be true if the variable $digits
> is not set/undefined -->
>        <condition field="${digits}" expression="^[^.+]$" />
> -----<<<< what should be here to check for undefined variable $digits
> ?
>           <action application="read" data="1 10 ivr/ivr-hello.wav
> digits 10000 #"/>
>           <action application="phrase" data="spell,${digits}"/>
>           <action application="transfer" data="checkdigits digits XML
> features"/>
>      </condition>
>
>      <condition field="${pincode}" expression="^${some_variable}$">
>             < SOME ACTIONS HERE >
>      </condition>
>  </extension>
>
>
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From anthony.minessale at gmail.com  Wed Oct 21 17:07:56 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Oct 2009 19:07:56 -0500
Subject: [Freeswitch-users] NOT in dialplan expression
In-Reply-To: <33c87fa30910211551m40a21daai3405b7c32da6655d@mail.gmail.com>
References: <33c87fa30910210434t268e8026pfd2b5439f26069fd@mail.gmail.com>
	<b39273530910210807j1f8a2fc6l4ff2bf1c154497b9@mail.gmail.com>
	<87f2f3b90910211044t1a02a653ya3bb9e1911099daa@mail.gmail.com>
	<33c87fa30910211551m40a21daai3405b7c32da6655d@mail.gmail.com>
Message-ID: <191c3a030910211707m6f6bf2c6q61fa5932f905cc85@mail.gmail.com>

Like rupa said,
use anti-action?

Remember FreeSWITCH XML dialplan is not a programming language it's a
pre-processor routing markup

Se potential use of anti-action below:

 <extension name="checkdigits">
     <condition field="destination_number" expression="^checkdigits$"/>
      <!--The next condition should be true if the variable $digits is not
set/undefined -->
       <condition field="${digits}" expression="^[^.+]$" />
--        <action application="read" data="1 10 ivr/ivr-hello.wav digits
10000 #"/>
          <action application="phrase" data="spell,${digits}"/>
          <action application="transfer" data="checkdigits digits XML
features"/>
          <anti-action application="set" data="not_true=true"/>
     </condition>

     <condition field="${pincode}" expression="^${some_variable}$">
            < SOME ACTIONS HERE >
     </condition>
 </extension>



On Wed, Oct 21, 2009 at 5:51 PM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> Thanks Guys for your suggestions!  Very much appreciated.
>
> The reason I asked was that I want to check if a variable is set or
> not.  I don't think your suggestions will work because I actually have
> to check to see an undefined variable.  Any way to do this?
>
> I want to have the same extension for checking digits read from DTMF.
> The problem is that the digits are not set to the variable until the
> transfer statement, which means I would require 2 extensions.   It
> will make things simpler if I can have the same extension that reads
> the digits and then checks them.
>
> For example:
>
>  <extension name="checkdigits">
>      <condition field="destination_number" expression="^checkdigits$"/>
>       <!--The next condition should be true if the variable $digits
> is not set/undefined -->
>        <condition field="${digits}" expression="^[^.+]$" />
> -----<<<< what should be here to check for undefined variable $digits
> ?
>           <action application="read" data="1 10 ivr/ivr-hello.wav
> digits 10000 #"/>
>           <action application="phrase" data="spell,${digits}"/>
>           <action application="transfer" data="checkdigits digits XML
> features"/>
>      </condition>
>
>      <condition field="${pincode}" expression="^${some_variable}$">
>             < SOME ACTIONS HERE >
>      </condition>
>  </extension>
>
> On Thu, Oct 22, 2009 at 4:44 AM, Michael Collins <msc at freeswitch.org>
> wrote:
> >
> >
> > On Wed, Oct 21, 2009 at 8:07 AM, Rupa Schomaker <rupa at rupa.com> wrote:
> >>
> >> You can also look at using anti-action rather than action after the
> >> condition.
> >>
> >> condition == if
> >> action == then
> >> anti-action == else
> >
> > Rupa & Leon,
> >
> > Nice job of explaining the options. Sometimes we forget about the
> powerful
> > constructs that are available in FreeSWITCH and PCRE. karma++ for both of
> > you. :)
> > -MC
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Wed Oct 21 17:20:41 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Oct 2009 19:20:41 -0500
Subject: [Freeswitch-users] UUID of the newly originated call?
In-Reply-To: <fe67a3340910202306i55a0b9ceve92ab19e000edf11@mail.gmail.com>
References: <fe67a3340910202306i55a0b9ceve92ab19e000edf11@mail.gmail.com>
Message-ID: <191c3a030910211720l26bd0303i2cb11e9adb28037d@mail.gmail.com>

which revision of FS are you using?

On Wed, Oct 21, 2009 at 1:06 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> I've tried with origination_uuid.
>
> First, I tried with SIP and my program executes successfully as what I
> expected. This program initiates a new call when a call comes and let the
> new call to eavesdrop the landed call.
> When, I experimented with PRI(openzap), I'm facing the following error.
> And, I am unable to make any calls(even from CLI). It is reporting the same
> error for the subsequent calls.
> Error:        2009-10-21 11:08:31.661252 [ERR] mod_openzap.c:1760 Session
> for channel 1:1 not found [UUID: 8364eb30-be03-11de-aa07-07e76b7651a9]
>
> Full log is in http://pastebin.freeswitch.org/10780
> My script is here, http://pastebin.freeswitch.org/10781
>
> What is this error for and how to avoid this?
>
> Is there any other way to get the uuid of the originated call except
> explicitly defining(origination_uuid).?!
>
> --
> Regards,
> Nagalenoj H.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
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FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Wed Oct 21 17:31:24 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Oct 2009 19:31:24 -0500
Subject: [Freeswitch-users] CS_REPORTING Channel event state
In-Reply-To: <08C886FF-6D2E-46CD-AC7D-2AF37F18B170@jerris.com>
References: <8ccbff060910200847y5e7ccab6o84d431836970bcb4@mail.gmail.com>
	<08C886FF-6D2E-46CD-AC7D-2AF37F18B170@jerris.com>
Message-ID: <191c3a030910211731y3e4de866qfc8fd0c9edd1db65@mail.gmail.com>

better still,

1) update to trunk with "make current"

2) Try to reproduce, if you can:
 a) open a new shell
 b) make sure you have gdb and gcore installed
 c) run ./support-d/fscore_pb gcore
 d) report the URL




On Tue, Oct 20, 2009 at 8:48 PM, Michael Jerris <mike at jerris.com> wrote:

> REPORTING is the state that it writes to CDR.  If you have calls stuck
> in this state, take one and try to use uuid_kill on it and see if it
> goes away, then get a core off of it and pastebin the thread apply all
> bt (with no other calls up).  What modules are you using for cdr and
> with what configuration?
>
> Mike
>
> On Oct 20, 2009, at 11:47 AM, Dome Charoenyost wrote:
>
> > Dear All
> > What's CS_REPORTING state ?
> > I found many channels not hang up  ans state is CS_REPORTING
> > e264f84a-bd87-11de-9a90-2320c02172de,outbound,2009-10-20
> > 21:50:26,1256050226,sofia/external/xxxxxxxxxxxx at xxx.xxx.xx.
> > 191:7050,CS_REPORTING,FreeSWITCH,xxxxxxxxxxxxxxxxxxxx,xx.xxx.xxx.xxx,
> > 7050,,,XML,public,,,,,
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From ahmedmunir007 at gmail.com  Wed Oct 21 21:08:44 2009
From: ahmedmunir007 at gmail.com (Ahmed Munir)
Date: Thu, 22 Oct 2009 09:08:44 +0500
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179
In-Reply-To: <mailman.1516.1256123625.2495.freeswitch-users@lists.freeswitch.org>
References: <mailman.1516.1256123625.2495.freeswitch-users@lists.freeswitch.org>
Message-ID: <a60fed5e0910212108v4755203y693b8bdec5501f5f@mail.gmail.com>

Hi,

Thanks for reply, it really helped me. One more thing to ask, how can we
make decision against >,<, >=, <= in condition header? Like we use == for
action and != for anti-action.

Kindly highlight it.




>
> ---------- Forwarded message ----------
> From: Ahmed Munir <ahmedmunir007 at gmail.com>
> To: FreeSwitch <freeswitch-users at lists.freeswitch.org>
> Date: Wed, 21 Oct 2009 15:37:15 +0500
> Subject: [Freeswitch-users] Call custom variable in condition
> Hi,
>
> I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I
> want to use it in condition, as I'm listing down the configuration below;
>
> <context name="SIP_incoming">
>    <extension name="call-sip-extensions">
>       <condition field="destination_number" expression="^(\d+)$">
>           <action application="set" data="AUTHENTICATION_STATUS=0"/>
>            <action application="transfer" data="${AUTHENTICATION_STATUS}
> XML Authen_Status"/>
>       </condition>
>    </extension>
> </context>
>
> <context name="Authen_Status">
>      <extension name="exten-auth-status">
>        <condition field="AUTHENTICATION_STATUS" expression="^0$">
>           <action application="answer"/>
>           <action application="playback" data="play.wav"/>
>       </condition>
>     </extension>
>   </context>
>
>
>
>
>  But unfortunately it is not working. Kindly advise me how to do implement
> it(Note: I don't want to call script). And one more thing to ask how can I
> transfer the values within the same context?
>
> --
> Regards,
>
> Ahmed Munir
>
>
>
>
> ---------- Forwarded message ----------
> From: Ghulam Mustafa <mustafa.pk at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Wed, 21 Oct 2009 15:52:24 +0500
> Subject: Re: [Freeswitch-users] Call custom variable in condition
> Ahmed,
>
> you can't use variables set by "set" application within a condition, though
> it doesn't make sense. wondering if there is any logic behind this or it's
> just a simple missing feature. anyone?
>
> -m
>
> Ahmed Munir wrote:
>
>> Hi,
>>
>> I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I
>> want to use it in condition, as I'm listing down the configuration below;
>>
>> <context name="SIP_incoming">
>>   <extension name="call-sip-extensions">
>>      <condition field="destination_number" expression="^(\d+)$">
>>          <action application="set" data="AUTHENTICATION_STATUS=0"/>
>>           <action application="transfer" data="${AUTHENTICATION_STATUS}
>> XML Authen_Status"/>
>>      </condition>
>>   </extension>
>> </context>
>>
>> <context name="Authen_Status">
>>     <extension name="exten-auth-status">
>>       <condition field="AUTHENTICATION_STATUS" expression="^0$">
>>          <action application="answer"/>
>>          <action application="playback" data="play.wav"/>
>>      </condition>
>>    </extension>
>>  </context>
>>
>>
>>
>>
>>  But unfortunately it is not working. Kindly advise me how to do implement
>> it(Note: I don't want to call script). And one more thing to ask how can I
>> transfer the values within the same context?
>>
>> --
>> Regards,
>>
>> Ahmed Munir
>>
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
>
>
> ---------- Forwarded message ----------
> From: Tihomir Culjaga <tculjaga at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Wed, 21 Oct 2009 13:13:13 +0200
> Subject: Re: [Freeswitch-users] Call custom variable in condition
> consider this:
>
>
>
> <context name="SIP_incoming">
>    <extension name="call-sip-extensions">
>       <condition field="destination_number" expression="^(\d+)$">
>           <action application="set" data="AUTHENTICATION_STATUS=0"/>
>            <action application="transfer" data="${AUTHENTICATION_STATUS}
> XML Authen_Status"/>
>       </condition>
>    </extension>
> </context>
>
>
>
> <context name="Authen_Status">
>      <extension name="exten-auth-status">
>        <condition field="${AUTHENTICATION_STATUS}" expression="^0$">
>           <action application="answer"/>
>           <action application="playback" data="play.wav"/>
>       </condition>
>     </extension>
>   </context>
>
>
>
>
>
> here is one of my dialplan. I'm using execute_extension but it is quite the
> same...
>
>
>
>    <extension name="ServiceLookup">
>       <condition field="destination_number" expression="(^300030)(.*)">
>          <action application="lookup_service_destination" data="in
> ${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, in $
> 1, in ${network_addr}:5060, out red_contact, out authResult"/>
>          <action application="log" data="INFO ########################
> ServiceLookup ########################\n"/>
>          <action application="log" data="INFO ########################
> contact = '${red_contact}' ##############\n"/>
>          <action application="log" data="INFO ########################
> CallerNum = '${caller_id_number:6:16}' ##########\n"/>
>          <action application="log" data="INFO ########################
> RADIUS auth = '${authResult}' ##########\n"/>
>
>          <action application="execute_extension" data="doRedirect XML
> public"/>
>         </condition>
>    </extension>
>
>
>    <extension name="doRedirect">
>       <condition field="destination_number" expression="^doRedirect$"/>
>       <condition field="${authResult}" expression="^0$|^60$">
>          <action application="log" data="INFO ########################
> RADIUS auth OK!!!' ##########\n"/>
>          <action application="redirect" data="${red_contact}"/>
>          <anti-action application="log" data="INFO ########################
> RADIUS auth NOK!! ##########\n"/>
>          <anti-action application="respond" data="403 Forbidden"/>
>       </condition>
>
>    </extension>
>
>
>
>
> On Wed, Oct 21, 2009 at 12:37 PM, Ahmed Munir <ahmedmunir007 at gmail.com>wrote:
>
>> Hi,
>>
>> I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I
>> want to use it in condition, as I'm listing down the configuration below;
>>
>> <context name="SIP_incoming">
>>    <extension name="call-sip-extensions">
>>       <condition field="destination_number" expression="^(\d+)$">
>>           <action application="set" data="AUTHENTICATION_STATUS=0"/>
>>            <action application="transfer" data="${AUTHENTICATION_STATUS}
>> XML Authen_Status"/>
>>       </condition>
>>    </extension>
>> </context>
>>
>> <context name="Authen_Status">
>>      <extension name="exten-auth-status">
>>        <condition field="AUTHENTICATION_STATUS" expression="^0$">
>>           <action application="answer"/>
>>           <action application="playback" data="play.wav"/>
>>       </condition>
>>     </extension>
>>   </context>
>>
>>
>>
>>
>>  But unfortunately it is not working. Kindly advise me how to do implement
>> it(Note: I don't want to call script). And one more thing to ask how can I
>> transfer the values within the same context?
>>
>> --
>> Regards,
>>
>> Ahmed Munir
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,

Ahmed Munir
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From mcampbellsmith at gmail.com  Wed Oct 21 21:28:41 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 22 Oct 2009 15:28:41 +1100
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179
In-Reply-To: <a60fed5e0910212108v4755203y693b8bdec5501f5f@mail.gmail.com>
References: <mailman.1516.1256123625.2495.freeswitch-users@lists.freeswitch.org>
	<a60fed5e0910212108v4755203y693b8bdec5501f5f@mail.gmail.com>
Message-ID: <33c87fa30910212128g1c167e6s8b847184921b21d1@mail.gmail.com>

Can't you use the inline statement to set a variable so that it can be
used directly in a condition?

http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions


On Thu, Oct 22, 2009 at 3:08 PM, Ahmed Munir <ahmedmunir007 at gmail.com> wrote:
> Hi,
>
> Thanks for reply, it really helped me. One more thing to ask, how can we
> make decision against >,<, >=, <= in condition header? Like we use == for
> action and != for anti-action.
>
> Kindly highlight it.
>
>
>
>>
>> ---------- Forwarded message ----------
>> From:?Ahmed Munir <ahmedmunir007 at gmail.com>
>> To:?FreeSwitch <freeswitch-users at lists.freeswitch.org>
>> Date:?Wed, 21 Oct 2009 15:37:15 +0500
>> Subject:?[Freeswitch-users] Call custom variable in condition
>> Hi,
>>
>> I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I
>> want to use it in condition, as I'm listing down the configuration below;
>>
>> <context name="SIP_incoming">
>> ?? <extension name="call-sip-extensions">
>> ????? <condition field="destination_number" expression="^(\d+)$">
>> ????????? <action application="set" data="AUTHENTICATION_STATUS=0"/>
>> ?????????? <action application="transfer" data="${AUTHENTICATION_STATUS}
>> XML Authen_Status"/>
>> ????? </condition>
>> ?? </extension>
>> </context>
>>
>> <context name="Authen_Status">
>> ???? <extension name="exten-auth-status">
>> ?????? <condition field="AUTHENTICATION_STATUS" expression="^0$">
>> ????????? <action application="answer"/>
>> ????????? <action application="playback" data="play.wav"/>
>> ????? </condition>
>> ??? </extension>
>> ? </context>
>>
>>
>>
>>
>> ?But unfortunately it is not working. Kindly advise me how to do implement
>> it(Note: I don't want to call script). And one more thing to ask how can I
>> transfer the values within the same context?
>>
>> --
>> Regards,
>>
>> Ahmed Munir
>>
>>
>>
>>
>> ---------- Forwarded message ----------
>> From:?Ghulam Mustafa <mustafa.pk at gmail.com>
>> To:?freeswitch-users at lists.freeswitch.org
>> Date:?Wed, 21 Oct 2009 15:52:24 +0500
>> Subject:?Re: [Freeswitch-users] Call custom variable in condition
>> Ahmed,
>>
>> you can't use variables set by "set" application within a condition,
>> though it doesn't make sense. wondering if there is any logic behind this or
>> it's just a simple missing feature. anyone?
>>
>> -m
>>
>> Ahmed Munir wrote:
>>>
>>> Hi,
>>>
>>> I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I
>>> want to use it in condition, as I'm listing down the configuration below;
>>>
>>> <context name="SIP_incoming">
>>> ? <extension name="call-sip-extensions">
>>> ? ? ?<condition field="destination_number" expression="^(\d+)$">
>>> ? ? ? ? ?<action application="set" data="AUTHENTICATION_STATUS=0"/>
>>> ? ? ? ? ? <action application="transfer" data="${AUTHENTICATION_STATUS}
>>> XML Authen_Status"/>
>>> ? ? ?</condition>
>>> ? </extension>
>>> </context>
>>>
>>> <context name="Authen_Status">
>>> ? ? <extension name="exten-auth-status">
>>> ? ? ? <condition field="AUTHENTICATION_STATUS" expression="^0$">
>>> ? ? ? ? ?<action application="answer"/>
>>> ? ? ? ? ?<action application="playback" data="play.wav"/>
>>> ? ? ?</condition>
>>> ? ?</extension>
>>> ?</context>
>>>
>>>
>>>
>>>
>>> ?But unfortunately it is not working. Kindly advise me how to do
>>> implement it(Note: I don't want to call script). And one more thing to ask
>>> how can I transfer the values within the same context?
>>>
>>> --
>>> Regards,
>>>
>>> Ahmed Munir
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>>
>>
>> ---------- Forwarded message ----------
>> From:?Tihomir Culjaga <tculjaga at gmail.com>
>> To:?freeswitch-users at lists.freeswitch.org
>> Date:?Wed, 21 Oct 2009 13:13:13 +0200
>> Subject:?Re: [Freeswitch-users] Call custom variable in condition
>> consider this:
>>
>>
>>
>> <context name="SIP_incoming">
>> ?? <extension name="call-sip-extensions">
>> ????? <condition field="destination_number" expression="^(\d+)$">
>> ????????? <action application="set" data="AUTHENTICATION_STATUS=0"/>
>> ?????????? <action application="transfer" data="${AUTHENTICATION_STATUS}
>> XML Authen_Status"/>
>> ????? </condition>
>> ?? </extension>
>> </context>
>>
>>
>>
>> <context name="Authen_Status">
>> ???? <extension name="exten-auth-status">
>> ?????? <condition field="${AUTHENTICATION_STATUS}" expression="^0$">
>> ????????? <action application="answer"/>
>> ????????? <action application="playback" data="play.wav"/>
>> ????? </condition>
>> ??? </extension>
>> ? </context>
>>
>>
>>
>>
>>
>> here is one of my dialplan. I'm using execute_extension but it is quite
>> the same...
>>
>>
>>
>> ?? <extension name="ServiceLookup">
>> ????? <condition field="destination_number" expression="(^300030)(.*)">
>> ???????? <action application="lookup_service_destination" data="in
>> ${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, in $
>> 1, in ${network_addr}:5060, out red_contact, out authResult"/>
>> ???????? <action application="log" data="INFO ########################
>> ServiceLookup ########################\n"/>
>> ???????? <action application="log" data="INFO ########################
>> contact = '${red_contact}' ##############\n"/>
>> ???????? <action application="log" data="INFO ########################
>> CallerNum = '${caller_id_number:6:16}' ##########\n"/>
>> ???????? <action application="log" data="INFO ########################
>> RADIUS auth = '${authResult}' ##########\n"/>
>>
>> ???????? <action application="execute_extension" data="doRedirect XML
>> public"/>
>> ??????? </condition>
>> ?? </extension>
>>
>>
>> ?? <extension name="doRedirect">
>> ????? <condition field="destination_number" expression="^doRedirect$"/>
>> ????? <condition field="${authResult}" expression="^0$|^60$">
>> ???????? <action application="log" data="INFO ########################
>> RADIUS auth OK!!!' ##########\n"/>
>> ???????? <action application="redirect" data="${red_contact}"/>
>> ???????? <anti-action application="log" data="INFO
>> ######################## RADIUS auth NOK!! ##########\n"/>
>> ???????? <anti-action application="respond" data="403 Forbidden"/>
>> ????? </condition>
>>
>> ?? </extension>
>>
>>
>>
>>
>> On Wed, Oct 21, 2009 at 12:37 PM, Ahmed Munir <ahmedmunir007 at gmail.com>
>> wrote:
>>>
>>> Hi,
>>>
>>> I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I
>>> want to use it in condition, as I'm listing down the configuration below;
>>>
>>> <context name="SIP_incoming">
>>> ?? <extension name="call-sip-extensions">
>>> ????? <condition field="destination_number" expression="^(\d+)$">
>>> ????????? <action application="set" data="AUTHENTICATION_STATUS=0"/>
>>> ?????????? <action application="transfer" data="${AUTHENTICATION_STATUS}
>>> XML Authen_Status"/>
>>> ????? </condition>
>>> ?? </extension>
>>> </context>
>>>
>>> <context name="Authen_Status">
>>> ???? <extension name="exten-auth-status">
>>> ?????? <condition field="AUTHENTICATION_STATUS" expression="^0$">
>>> ????????? <action application="answer"/>
>>> ????????? <action application="playback" data="play.wav"/>
>>> ????? </condition>
>>> ??? </extension>
>>> ? </context>
>>>
>>>
>>>
>>>
>>> ?But unfortunately it is not working. Kindly advise me how to do
>>> implement it(Note: I don't want to call script). And one more thing to ask
>>> how can I transfer the values within the same context?
>>>
>>> --
>>> Regards,
>>>
>>> Ahmed Munir
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Regards,
>
> Ahmed Munir
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From msc at freeswitch.org  Wed Oct 21 21:36:31 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 21 Oct 2009 21:36:31 -0700
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179
In-Reply-To: <33c87fa30910212128g1c167e6s8b847184921b21d1@mail.gmail.com>
References: <mailman.1516.1256123625.2495.freeswitch-users@lists.freeswitch.org>
	<a60fed5e0910212108v4755203y693b8bdec5501f5f@mail.gmail.com>
	<33c87fa30910212128g1c167e6s8b847184921b21d1@mail.gmail.com>
Message-ID: <87f2f3b90910212136wce5c22dy9fc23a78e8946541@mail.gmail.com>

On Wed, Oct 21, 2009 at 9:28 PM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> Can't you use the inline statement to set a variable so that it can be
> used directly in a condition?
>
> http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
>
>
Yes you can. Just don't abuse it like Tony said. If you find yourself doing
it a lot then it's a sure sign that you're taking the wrong approach.
-MC
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From msc at freeswitch.org  Wed Oct 21 21:41:31 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 21 Oct 2009 21:41:31 -0700
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179
In-Reply-To: <a60fed5e0910212108v4755203y693b8bdec5501f5f@mail.gmail.com>
References: <mailman.1516.1256123625.2495.freeswitch-users@lists.freeswitch.org>
	<a60fed5e0910212108v4755203y693b8bdec5501f5f@mail.gmail.com>
Message-ID: <87f2f3b90910212141w72b8f5a2oce032adb974d71e2@mail.gmail.com>

On Wed, Oct 21, 2009 at 9:08 PM, Ahmed Munir <ahmedmunir007 at gmail.com>wrote:

> Hi,
>
> Thanks for reply, it really helped me. One more thing to ask, how can we
> make decision against >,<, >=, <= in condition header? Like we use == for
> action and != for anti-action.
>
> Kindly highlight it.
>
>
You can only do greater than and less than in the date/time matching. See
the date/time example in the default.xml dialplan file.

You can also use regular expressions if you're in a pinch. For example, if
you need to match numbers >= 1100 and <= 1500 you could just use this regex:

^(1[1234]\d\d|1500)$

The real question, though, is this: what types of values do you need to
match for GT or LT? Date/time? Money? Other? That will determine if you need
to use a script or just the dialplan.
-MC
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From shaheryarkh at googlemail.com  Wed Oct 21 23:57:42 2009
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Thu, 22 Oct 2009 12:57:42 +0600
Subject: [Freeswitch-users] Sofia SIP external profile gateway names
Message-ID: <b16156850910212357s15fe6116tc22b6709f7110a49@mail.gmail.com>

Hi,

I am facing a problem with Sofia SIP external profile. Basically i have 10+
accounts, say a101 - a110, to a single service provider, say xyz.com. For
each account a have created a gateway in external profile's directory i.e.
/usr/local/freeswitch/conf/sip_profiles/external/a101.xml up to a110.xml.

At first the problem was none of the accounts were registering since, FS was
trying to send registration requests to host "a101" instead of xyz.com, but
later when i set xyz.com as proxy and register proxy address, it
successfully registered all accounts.

Now i am facing a similar problem in dialplan, for example if i try to
dialout via gateway a101, call immediately fails with
NORMAL_TEMPORARY_FAILURE. When i trun on sip tracing i don't see any INVITE
message sent to provider xyz.com.

2009-10-22 06:51:55.325393 [NOTICE] switch_channel.c:613 New Channel
sofia/external/00923344224088 at a101 [b7663caf-cd29-4213-9059-ae880d49b0ca]
2009-10-22 06:51:55.516382 [NOTICE] sofia.c:4039 Hangup
sofia/external/00923344224088 at a101 [CS_CONSUME_MEDIA]
[NORMAL_TEMPORARY_FAILURE]
API CALL [originate(sofia/external/00923344224088 at a101 9999)] output:
-ERR NORMAL_TEMPORARY_FAILURE
2009-10-22 06:51:55.527381 [NOTICE] switch_core_session.c:1132 Session 2
(sofia/external/00923344224088 at a101) Ended
2009-10-22 06:51:55.527381 [NOTICE] switch_core_session.c:1134 Close Channel
sofia/external/00923344224088 at a101 [CS_DESTROY]

I think its trying to look up a101 instead of xyz.com to send INVITE. Kindly
help.

Thank you.


-- 
________________________________________________________
|
                     |
| FATAL ERROR                                                           ---
O X |
|_______________________________________________________|
|                        You have moved the mouse.
 |
| Windows must be restarted for the changes to take effect.   |
|                                        <OK>
               |
####################################/


Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From leon at scarlet-internet.nl  Thu Oct 22 01:27:54 2009
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Thu, 22 Oct 2009 10:27:54 +0200
Subject: [Freeswitch-users] Call custom variable in condition
In-Reply-To: <87f2f3b90910211002y7c35c622g4c76472fd8c7bada@mail.gmail.com>
References: <a60fed5e0910210337n45ffefe2wc3e3d9d0b403a568@mail.gmail.com>
	<65d96fc80910210413t1123dbc0xd5740bc0673d638c@mail.gmail.com>
	<191c3a030910210939k7dbf1494gf713df8770356450@mail.gmail.com>
	<87f2f3b90910211002y7c35c622g4c76472fd8c7bada@mail.gmail.com>
Message-ID: <CF72DAFC-61F5-4E8A-B1D9-C0171E823D3E@scarlet-internet.nl>

Hi Michael,

The feature is already documented here:

http://wiki.freeswitch.org/wiki/Dialplan_XML#Clarification
http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions

Perhaps the reason *why* it's the way it is can be expanded a bit ?

regards,

Leon


On Oct 21, 2009, at 7:02 PM, Michael Collins wrote:

>
>
> On Wed, Oct 21, 2009 at 9:39 AM, Anthony Minessale <anthony.minessale at gmail.com 
> > wrote:
> It not only makes sense it's well documented on the wiki page.
> The set line is not happening right when it's encountered, the set  
> line is copied into the channel and executed later after the whole  
> dialplan is parsed.  The dialplan is a pre-processor not a runtime  
> engine.
>
> Here is a new feature in pre-1.0.5 (svn trunk)
>
> Some applications like set can now be executed within the dialplan  
> but you should use it sparingly.
> <action application="set" data="testing=true" inline="true"/>
>
> I'm getting ready to document this feature. For the sake of  
> edification, why is it best to use this sparingly, other than wide- 
> spread use making dialplans all cluttered?
> -MC
>
>
> The inline=true makes it execute inside the dialplan and it's never  
> copied into your resulting extension because it's executed  
> immediately.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From nagalenoj at gmail.com  Thu Oct 22 02:08:55 2009
From: nagalenoj at gmail.com (Nagalenoj)
Date: Thu, 22 Oct 2009 02:08:55 -0700 (PDT)
Subject: [Freeswitch-users] UUID of the newly originated call?
In-Reply-To: <191c3a030910211720l26bd0303i2cb11e9adb28037d@mail.gmail.com>
References: <fe67a3340910202306i55a0b9ceve92ab19e000edf11@mail.gmail.com>
	<191c3a030910211720l26bd0303i2cb11e9adb28037d@mail.gmail.com>
Message-ID: <26006565.post@talk.nabble.com>


I'm using 'FreeSWITCH Version 1.0.trunk (15106)'.


Anthony Minessale-2 wrote:
> 
> which revision of FS are you using?
> 
> On Wed, Oct 21, 2009 at 1:06 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:
> 
>> I've tried with origination_uuid.
>>
>> First, I tried with SIP and my program executes successfully as what I
>> expected. This program initiates a new call when a call comes and let the
>> new call to eavesdrop the landed call.
>> When, I experimented with PRI(openzap), I'm facing the following error.
>> And, I am unable to make any calls(even from CLI). It is reporting the
>> same
>> error for the subsequent calls.
>> Error:        2009-10-21 11:08:31.661252 [ERR] mod_openzap.c:1760 Session
>> for channel 1:1 not found [UUID: 8364eb30-be03-11de-aa07-07e76b7651a9]
>>
>> Full log is in http://pastebin.freeswitch.org/10780
>> My script is here, http://pastebin.freeswitch.org/10781
>>
>> What is this error for and how to avoid this?
>>
>> Is there any other way to get the uuid of the originated call except
>> explicitly defining(origination_uuid).?!
>>
>> --
>> Regards,
>> Nagalenoj H.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/Re%3A-UUID-of-the-newly-originated-call--tp25987024p26006565.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From ahmedmunir007 at gmail.com  Thu Oct 22 02:47:52 2009
From: ahmedmunir007 at gmail.com (Ahmed Munir)
Date: Thu, 22 Oct 2009 14:47:52 +0500
Subject: [Freeswitch-users] Expression in dialplan
Message-ID: <a60fed5e0910220247q54a6a4dfr4b589e76a78d1d52@mail.gmail.com>

Thanks for reply. Actually I want to match values and strings. Like
IP_CALL=="true" or IP_CALL != "true" or Date >= "22/10/09" and vice versa.
Last thing I forgot to mention, how can I implement condition in dial plan
xml when using operation and/or  i.e. IP_CALL == "false" and PST_CALL ==
"false" / IP_CALL == "false" or PST_CALL == "false".

Note: I only want to implement it in dial plan not in any script.


Kindly advise the solution.

---------- Forwarded message ----------
From: Michael Collins <msc at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Date: Wed, 21 Oct 2009 21:36:31 -0700
Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179


On Wed, Oct 21, 2009 at 9:28 PM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> Can't you use the inline statement to set a variable so that it can be
> used directly in a condition?
>
> http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
>
>
Yes you can. Just don't abuse it like Tony said. If you find yourself doing
it a lot then it's a sure sign that you're taking the wrong approach.
-MC



---------- Forwarded message ----------
From: Michael Collins <msc at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Date: Wed, 21 Oct 2009 21:41:31 -0700
Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179


On Wed, Oct 21, 2009 at 9:08 PM, Ahmed Munir <ahmedmunir007 at gmail.com>wrote:

> Hi,
>
> Thanks for reply, it really helped me. One more thing to ask, how can we
> make decision against >,<, >=, <= in condition header? Like we use == for
> action and != for anti-action.
>
> Kindly highlight it.
>
>
You can only do greater than and less than in the date/time matching. See
the date/time example in the default.xml dialplan file.

You can also use regular expressions if you're in a pinch. For example, if
you need to match numbers >= 1100 and <= 1500 you could just use this regex:

^(1[1234]\d\d|1500)$

The real question, though, is this: what types of values do you need to
match for GT or LT? Date/time? Money? Other? That will determine if you need
to use a script or just the dialplan.
-MC
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From rupa at rupa.com  Thu Oct 22 05:51:04 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 22 Oct 2009 07:51:04 -0500
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179
In-Reply-To: <87f2f3b90910212141w72b8f5a2oce032adb974d71e2@mail.gmail.com>
References: <mailman.1516.1256123625.2495.freeswitch-users@lists.freeswitch.org>
	<a60fed5e0910212108v4755203y693b8bdec5501f5f@mail.gmail.com>
	<87f2f3b90910212141w72b8f5a2oce032adb974d71e2@mail.gmail.com>
Message-ID: <b39273530910220551q14d22150ldf8d4bff0b6fd011@mail.gmail.com>

cond would be helpful here?  I updated the wiki on this one just now
with a bit more detail.  It is a api call. so, you'd use it like:

${cond(eval ? trueval : falseval)}

so to get a value of ERR if the var my myvar is > 15 you could:

${cond(${myvar} > 15 ? ERR : OK)}

If both sides of the comparison operator are numeric then it does
numeric comparison otherwise it does lexical string comparison.


On Wed, Oct 21, 2009 at 11:41 PM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Wed, Oct 21, 2009 at 9:08 PM, Ahmed Munir <ahmedmunir007 at gmail.com>
> wrote:
>>
>> Hi,
>>
>> Thanks for reply, it really helped me. One more thing to ask, how can we
>> make decision against >,<, >=, <= in condition header? Like we use == for
>> action and != for anti-action.
>>
>> Kindly highlight it.
>>
>
> You can only do greater than and less than in the date/time matching. See
> the date/time example in the default.xml dialplan file.
>
> You can also use regular expressions if you're in a pinch. For example, if
> you need to match numbers >= 1100 and <= 1500 you could just use this regex:
>
> ^(1[1234]\d\d|1500)$
>
> The real question, though, is this: what types of values do you need to
> match for GT or LT? Date/time? Money? Other? That will determine if you need
> to use a script or just the dialplan.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa



From tculjaga at gmail.com  Thu Oct 22 06:30:34 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 22 Oct 2009 15:30:34 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910201841030.12352@srv.icf.org.ru>
	<65d96fc80910210034nd7e8864hfcc421ec6a6c52f@mail.gmail.com>
	<Pine.LNX.4.64.0910211427520.9175@srv.icf.org.ru>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
Message-ID: <65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>

>
>
> TC>
> TC>I have enabled crash-protection and when i do SIP => H323 call it
> doesn't
> TC>generate coredumps... it is just this thread that is crashing ... pls
> check
> TC>the log downbelow:
>
> core dump in case enabled crash-protection may be unusable, i have a case
> then
> my module crash silently, after this crash-protection is killing sip leg
> and after
> this i get core dump where backtrace show me segfault in libc6, i spent one
> day to
> understand this situation, and then i disable crash-protection i see there
> is actualy
> it crashes. disable crash-protection and show backtrace of crash, i think
> result will
> be different.
>
>
> TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
> TC>======>FSH323Connection::decodeCapability
> TC>
> TC>
> TC>
> TC>Well, I'm not sure if the backtrace is from 1.0.4 ... i will disable
> TC>crass-protection and will send new logs to you.
> TC>
> TC>
> TC>Also, if you like i can give you access to the machine itself...
> TC>
> TC>T.
> TC>
>



Hi, here is the FS log without crash-protection:
http://pastebin.freeswitch.org/10796 and here is the backtrace:
http://pastebin.freeswitch.org/10797



my dialplan looks ok, so i guess it is up to the module.


  <extension name="ENYTHING_ELSE">
    <condition field="destination_number"
expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
      <action application="set"
data="effective_caller_id_number=1001282122"/>
      <action application="set" data="NCX_IP=10.4.4.254"/>
      <action application="set" data="call_timeout=30"/>
      <action application="set" data="hangup_after_bridge=true"/>
      <!--action application="set" data="bypass_media=false"/-->
      <action application="set" data="proxy_media=true"/>

      <!--action application="bridge" data="opal/h323:0$1@${NCX_IP}"/-->
      <action application="bridge" data="h323/0$1@${NCX_IP}"/>
    </condition>
  </extension>



please advice,

T.



> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129-RIPE
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bottleman at icf.org.ru  Thu Oct 22 06:41:01 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Thu, 22 Oct 2009 17:41:01 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910201841030.12352@srv.icf.org.ru>
	<65d96fc80910210034nd7e8864hfcc421ec6a6c52f@mail.gmail.com>
	<Pine.LNX.4.64.0910211427520.9175@srv.icf.org.ru>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>

On 2009-10-22 15:30 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>>
TC>>
TC>> TC>
TC>> TC>I have enabled crash-protection and when i do SIP => H323 call it
TC>> doesn't
TC>> TC>generate coredumps... it is just this thread that is crashing ... pls
TC>> check
TC>> TC>the log downbelow:
TC>>
TC>> core dump in case enabled crash-protection may be unusable, i have a case
TC>> then
TC>> my module crash silently, after this crash-protection is killing sip leg
TC>> and after
TC>> this i get core dump where backtrace show me segfault in libc6, i spent one
TC>> day to
TC>> understand this situation, and then i disable crash-protection i see there
TC>> is actualy
TC>> it crashes. disable crash-protection and show backtrace of crash, i think
TC>> result will
TC>> be different.
TC>>
TC>>
TC>> TC>2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
TC>> TC>======>FSH323Connection::decodeCapability
TC>> TC>
TC>> TC>
TC>> TC>
TC>> TC>Well, I'm not sure if the backtrace is from 1.0.4 ... i will disable
TC>> TC>crass-protection and will send new logs to you.
TC>> TC>
TC>> TC>
TC>> TC>Also, if you like i can give you access to the machine itself...
TC>> TC>
TC>> TC>T.
TC>> TC>
TC>>
TC>
TC>
TC>
TC>Hi, here is the FS log without crash-protection:
TC>http://pastebin.freeswitch.org/10796 and here is the backtrace:
TC>http://pastebin.freeswitch.org/10797

i fix this crash already, please download latest version from same link 
as previous, recompile and try again.

TC>my dialplan looks ok, so i guess it is up to the module.
TC>
TC>
TC>  <extension name="ENYTHING_ELSE">
TC>    <condition field="destination_number"
TC>expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
TC>      <action application="set"
TC>data="effective_caller_id_number=1001282122"/>
TC>      <action application="set" data="NCX_IP=10.4.4.254"/>
TC>      <action application="set" data="call_timeout=30"/>
TC>      <action application="set" data="hangup_after_bridge=true"/>
TC>      <!--action application="set" data="bypass_media=false"/-->
TC>      <action application="set" data="proxy_media=true"/>
TC>
TC>      <!--action application="bridge" data="opal/h323:0$1@${NCX_IP}"/-->
TC>      <action application="bridge" data="h323/0$1@${NCX_IP}"/>
TC>    </condition>
TC>  </extension>
TC>
TC>
TC>
TC>please advice,
TC>
TC>T.
TC>
TC>
TC>
TC>> C ?????????                       With Best Regards
TC>> ???????????? ????.                Georgiewskiy Yuriy
TC>> +7 4872 711666                    +7 4872 711666
TC>> ???? +7 4872 711143               fax +7 4872 711143
TC>> ???????? ??? "?? ?? ??????"       IT Service Ltd
TC>> http://nkoort.ru                  http://nkoort.ru
TC>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
TC>> YG129-RIPE                        YG129-RIPE
TC>>
TC>> _______________________________________________
TC>> FreeSWITCH-users mailing list
TC>> FreeSWITCH-users at lists.freeswitch.org
TC>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> http://www.freeswitch.org
TC>>
TC>>
TC>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From szasz.szabolcs at gmail.com  Thu Oct 22 06:56:29 2009
From: szasz.szabolcs at gmail.com (Szasz Szabolcs)
Date: Thu, 22 Oct 2009 16:56:29 +0300
Subject: [Freeswitch-users] srtp g729 bypass
Message-ID: <4a4d27d30910220656u46e584b3h94c357e6d218462@mail.gmail.com>

Hi

I have set up a freeswitch with TLS and SRTP support. I'm sending  encrypted
calls to Freswitch and the Freeswitch  forwards the calls to an asterisk
unencrypted. I have issue by using G729 in this scenario. My UA supports
g279, asterisk supports g729 transcoding and I understood that freeswitch
supports the g729 in bypass mode. Should it work? What I have to change in
config? So the scenario, I would like to do:
 UA|======TLS+SRTP(G729)=====|FREESWITCH|------------ClearSIP+RTP(G729)------------|Asterisk

Thank you in advance.

Szasz Szabolcs
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From tculjaga at gmail.com  Thu Oct 22 06:59:52 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 22 Oct 2009 15:59:52 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910211427520.9175@srv.icf.org.ru>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
Message-ID: <65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>

>
>
> TC>Hi, here is the FS log without crash-protection:
> TC>http://pastebin.freeswitch.org/10796 and here is the backtrace:
> TC>http://pastebin.freeswitch.org/10797
>
> i fix this crash already, please download latest version from same link
> as previous, recompile and try again.
>
>
outgoing works, I can place an end-to-end call ... except the RTP is realy
delayed ... after approx 30 sec of conversation the audio is delayed more
than 10 seconds.... but i have 2 way audio for outgoing calls:)

Do you need some logs ?


Inbound cals still the same... i suppose you didn't have a chance working on
that as well ...

T.
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From tculjaga at gmail.com  Thu Oct 22 07:04:54 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 22 Oct 2009 16:04:54 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
Message-ID: <65d96fc80910220704t283bfca3w2129ff3575659f97@mail.gmail.com>

On Thu, Oct 22, 2009 at 3:59 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

>
>> TC>Hi, here is the FS log without crash-protection:
>> TC>http://pastebin.freeswitch.org/10796 and here is the backtrace:
>> TC>http://pastebin.freeswitch.org/10797
>>
>> i fix this crash already, please download latest version from same link
>> as previous, recompile and try again.
>>
>>
> outgoing works, I can place an end-to-end call ... except the RTP is realy
> delayed ... after approx 30 sec of conversation the audio is delayed more
> than 10 seconds.... but i have 2 way audio for outgoing calls:)
>
>
one more thing ... it is H323 endpoint => SIP phone audio that is delayed.
SIP phone => H323 endpoint is ok!

T.


> Do you need some logs ?
>
>
> Inbound cals still the same... i suppose you didn't have a chance working
> on that as well ...
>
> T.
>
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From bottleman at icf.org.ru  Thu Oct 22 07:11:45 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Thu, 22 Oct 2009 18:11:45 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910211427520.9175@srv.icf.org.ru>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>

On 2009-10-22 15:59 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>>
TC>>
TC>> TC>Hi, here is the FS log without crash-protection:
TC>> TC>http://pastebin.freeswitch.org/10796 and here is the backtrace:
TC>> TC>http://pastebin.freeswitch.org/10797
TC>>
TC>> i fix this crash already, please download latest version from same link
TC>> as previous, recompile and try again.
TC>>
TC>>
TC>outgoing works, I can place an end-to-end call ... except the RTP is realy
TC>delayed ... after approx 30 sec of conversation the audio is delayed more
TC>than 10 seconds.... but i have 2 way audio for outgoing calls:)
TC>
TC>Do you need some logs ?

try disable medai-proxy, there is issue with rtp now then medai-proxy or transcoding enabled.

TC>Inbound cals still the same... i suppose you didn't have a chance working on
TC>that as well ...

sorry i don't remember what the same, show extension and logs of inbound call. at this time
at this time i have working transit h323<->sip in both direction, ivr is work too, there is some 
issues present but basicaly it work.

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From bottleman at icf.org.ru  Thu Oct 22 07:13:09 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Thu, 22 Oct 2009 18:13:09 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910220704t283bfca3w2129ff3575659f97@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<65d96fc80910220704t283bfca3w2129ff3575659f97@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910221811590.1929@srv.icf.org.ru>

On 2009-10-22 16:04 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>On Thu, Oct 22, 2009 at 3:59 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
TC>
TC>>
TC>>> TC>Hi, here is the FS log without crash-protection:
TC>>> TC>http://pastebin.freeswitch.org/10796 and here is the backtrace:
TC>>> TC>http://pastebin.freeswitch.org/10797
TC>>>
TC>>> i fix this crash already, please download latest version from same link
TC>>> as previous, recompile and try again.
TC>>>
TC>>>
TC>> outgoing works, I can place an end-to-end call ... except the RTP is realy
TC>> delayed ... after approx 30 sec of conversation the audio is delayed more
TC>> than 10 seconds.... but i have 2 way audio for outgoing calls:)
TC>>
TC>>
TC>one more thing ... it is H323 endpoint => SIP phone audio that is delayed.
TC>SIP phone => H323 endpoint is ok!

hm, i have such issue but in reverce direction now.

TC>> Do you need some logs ?
TC>>
TC>>
TC>> Inbound cals still the same... i suppose you didn't have a chance working
TC>> on that as well ...
TC>>
TC>> T.
TC>>
TC>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From anthony.minessale at gmail.com  Thu Oct 22 07:27:54 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 22 Oct 2009 09:27:54 -0500
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910221811590.1929@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<65d96fc80910220704t283bfca3w2129ff3575659f97@mail.gmail.com>
	<Pine.LNX.4.64.0910221811590.1929@srv.icf.org.ru>
Message-ID: <191c3a030910220727v7e62f47dhf172ba8682d5d52a@mail.gmail.com>

crash protection has been completely removed from FreeSWITCH, I certianly
hope you are working on this against SVN trunk?  Also you have been given an
svn area and a jira category for this so you should move all the info from
this thread to jira http://jira.freeswitch.org

It's much easier to collaberate this kind of development when you have the
code in SVN.


2009/10/22 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-22 16:04 +0200, Tihomir Culjaga wrote
> freeswitch-users at lists.fre...:
>
> TC>On Thu, Oct 22, 2009 at 3:59 PM, Tihomir Culjaga <tculjaga at gmail.com>
> wrote:
> TC>
> TC>>
> TC>>> TC>Hi, here is the FS log without crash-protection:
> TC>>> TC>http://pastebin.freeswitch.org/10796 and here is the backtrace:
> TC>>> TC>http://pastebin.freeswitch.org/10797
> TC>>>
> TC>>> i fix this crash already, please download latest version from same
> link
> TC>>> as previous, recompile and try again.
> TC>>>
> TC>>>
> TC>> outgoing works, I can place an end-to-end call ... except the RTP is
> realy
> TC>> delayed ... after approx 30 sec of conversation the audio is delayed
> more
> TC>> than 10 seconds.... but i have 2 way audio for outgoing calls:)
> TC>>
> TC>>
> TC>one more thing ... it is H323 endpoint => SIP phone audio that is
> delayed.
> TC>SIP phone => H323 endpoint is ok!
>
> hm, i have such issue but in reverce direction now.
>
> TC>> Do you need some logs ?
> TC>>
> TC>>
> TC>> Inbound cals still the same... i suppose you didn't have a chance
> working
> TC>> on that as well ...
> TC>>
> TC>> T.
> TC>>
> TC>
>
> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129-RIPE
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From brian at freeswitch.org  Thu Oct 22 07:30:00 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 22 Oct 2009 09:30:00 -0500
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910221811590.1929@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<65d96fc80910220704t283bfca3w2129ff3575659f97@mail.gmail.com>
	<Pine.LNX.4.64.0910221811590.1929@srv.icf.org.ru>
Message-ID: <DE0920EE-584A-4DF9-B8D8-F2C0DF94E8DF@freeswitch.org>

Have you started moving the code into our SVN and using our  
ticketing / issue tracker to help you manage issues?

/b

On Oct 22, 2009, at 9:13 AM, Georgiewskiy Yuriy wrote:

> On 2009-10-22 16:04 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre 
> ...:
>
> TC>On Thu, Oct 22, 2009 at 3:59 PM, Tihomir Culjaga <tculjaga at gmail.com 
> > wrote:
> TC>
> TC>>
> TC>>> TC>Hi, here is the FS log without crash-protection:
> TC>>> TC>http://pastebin.freeswitch.org/10796 and here is the  
> backtrace:
> TC>>> TC>http://pastebin.freeswitch.org/10797
> TC>>>
> TC>>> i fix this crash already, please download latest version from  
> same link
> TC>>> as previous, recompile and try again.
> TC>>>
> TC>>>
> TC>> outgoing works, I can place an end-to-end call ... except the  
> RTP is realy
> TC>> delayed ... after approx 30 sec of conversation the audio is  
> delayed more
> TC>> than 10 seconds.... but i have 2 way audio for outgoing calls:)
> TC>>
> TC>>
> TC>one more thing ... it is H323 endpoint => SIP phone audio that is  
> delayed.
> TC>SIP phone => H323 endpoint is ok!
>
> hm, i have such issue but in reverce direction now.
>
> TC>> Do you need some logs ?
> TC>>
> TC>>
> TC>> Inbound cals still the same... i suppose you didn't have a  
> chance working
> TC>> on that as well ...
> TC>>
> TC>> T.
> TC>>
> TC>
>
> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129- 
> RIPE_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From bottleman at icf.org.ru  Thu Oct 22 07:34:53 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Thu, 22 Oct 2009 18:34:53 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <DE0920EE-584A-4DF9-B8D8-F2C0DF94E8DF@freeswitch.org>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<65d96fc80910220704t283bfca3w2129ff3575659f97@mail.gmail.com>
	<Pine.LNX.4.64.0910221811590.1929@srv.icf.org.ru>
	<DE0920EE-584A-4DF9-B8D8-F2C0DF94E8DF@freeswitch.org>
Message-ID: <Pine.LNX.4.64.0910221833420.6837@srv.icf.org.ru>

On 2009-10-22 09:30 -0500, Brian West wrote freeswitch-users at lists.freeswit...:

hm, you not tell me what account created, and i don't try to do this.

BW>Have you started moving the code into our SVN and using our  
BW>ticketing / issue tracker to help you manage issues?
BW>
BW>/b
BW>
BW>On Oct 22, 2009, at 9:13 AM, Georgiewskiy Yuriy wrote:
BW>
BW>> On 2009-10-22 16:04 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre 
BW>> ...:
BW>>
BW>> TC>On Thu, Oct 22, 2009 at 3:59 PM, Tihomir Culjaga <tculjaga at gmail.com 
BW>> > wrote:
BW>> TC>
BW>> TC>>
BW>> TC>>> TC>Hi, here is the FS log without crash-protection:
BW>> TC>>> TC>http://pastebin.freeswitch.org/10796 and here is the  
BW>> backtrace:
BW>> TC>>> TC>http://pastebin.freeswitch.org/10797
BW>> TC>>>
BW>> TC>>> i fix this crash already, please download latest version from  
BW>> same link
BW>> TC>>> as previous, recompile and try again.
BW>> TC>>>
BW>> TC>>>
BW>> TC>> outgoing works, I can place an end-to-end call ... except the  
BW>> RTP is realy
BW>> TC>> delayed ... after approx 30 sec of conversation the audio is  
BW>> delayed more
BW>> TC>> than 10 seconds.... but i have 2 way audio for outgoing calls:)
BW>> TC>>
BW>> TC>>
BW>> TC>one more thing ... it is H323 endpoint => SIP phone audio that is  
BW>> delayed.
BW>> TC>SIP phone => H323 endpoint is ok!
BW>>
BW>> hm, i have such issue but in reverce direction now.
BW>>
BW>> TC>> Do you need some logs ?
BW>> TC>>
BW>> TC>>
BW>> TC>> Inbound cals still the same... i suppose you didn't have a  
BW>> chance working
BW>> TC>> on that as well ...
BW>> TC>>
BW>> TC>> T.
BW>> TC>>
BW>> TC>
BW>>
BW>> C ?????????                       With Best Regards
BW>> ???????????? ????.                Georgiewskiy Yuriy
BW>> +7 4872 711666                    +7 4872 711666
BW>> ???? +7 4872 711143               fax +7 4872 711143
BW>> ???????? ??? "?? ?? ??????"       IT Service Ltd
BW>> http://nkoort.ru                  http://nkoort.ru
BW>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
BW>> YG129-RIPE                        YG129- 
BW>> RIPE_______________________________________________
BW>> FreeSWITCH-users mailing list
BW>> FreeSWITCH-users at lists.freeswitch.org
BW>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
BW>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
BW>> users
BW>> http://www.freeswitch.org
BW>
BW>
BW>_______________________________________________
BW>FreeSWITCH-users mailing list
BW>FreeSWITCH-users at lists.freeswitch.org
BW>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
BW>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
BW>http://www.freeswitch.org
BW>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From brian at freeswitch.org  Thu Oct 22 07:41:59 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 22 Oct 2009 09:41:59 -0500
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910221833420.6837@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910210439m5b179a7djec8b3e249b2c35b0@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<65d96fc80910220704t283bfca3w2129ff3575659f97@mail.gmail.com>
	<Pine.LNX.4.64.0910221811590.1929@srv.icf.org.ru>
	<DE0920EE-584A-4DF9-B8D8-F2C0DF94E8DF@freeswitch.org>
	<Pine.LNX.4.64.0910221833420.6837@srv.icf.org.ru>
Message-ID: <4EA71BA6-3E47-4A08-9812-2FE6627394E6@freeswitch.org>

AS per the email you and I exchanged we created the account and the  
mod_h323 folder in endpoints....

/b

On Oct 22, 2009, at 9:34 AM, Georgiewskiy Yuriy wrote:

> hm, you not tell me what account created, and i don't try to do this.




From bottleman at icf.org.ru  Thu Oct 22 07:44:42 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Thu, 22 Oct 2009 18:44:42 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <191c3a030910220727v7e62f47dhf172ba8682d5d52a@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<65d96fc80910220704t283bfca3w2129ff3575659f97@mail.gmail.com>
	<Pine.LNX.4.64.0910221811590.1929@srv.icf.org.ru>
	<191c3a030910220727v7e62f47dhf172ba8682d5d52a@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910221842540.8225@srv.icf.org.ru>

On 2009-10-22 09:27 -0500, Anthony Minessale wrote freeswitch-users at lists.f...:

AM>crash protection has been completely removed from FreeSWITCH, I certianly
AM>hope you are working on this against SVN trunk?  

i don't have trunk at this time, my current work is based on 1.0.4 version.

AM>Also you have been given an
AM>svn area and a jira category for this so you should move all the info from
AM>this thread to jira http://jira.freeswitch.org
AM>
AM>It's much easier to collaberate this kind of development when you have the
AM>code in SVN.
AM>
AM>
AM>2009/10/22 Georgiewskiy Yuriy <bottleman at icf.org.ru>
AM>
AM>> On 2009-10-22 16:04 +0200, Tihomir Culjaga wrote
AM>> freeswitch-users at lists.fre...:
AM>>
AM>> TC>On Thu, Oct 22, 2009 at 3:59 PM, Tihomir Culjaga <tculjaga at gmail.com>
AM>> wrote:
AM>> TC>
AM>> TC>>
AM>> TC>>> TC>Hi, here is the FS log without crash-protection:
AM>> TC>>> TC>http://pastebin.freeswitch.org/10796 and here is the backtrace:
AM>> TC>>> TC>http://pastebin.freeswitch.org/10797
AM>> TC>>>
AM>> TC>>> i fix this crash already, please download latest version from same
AM>> link
AM>> TC>>> as previous, recompile and try again.
AM>> TC>>>
AM>> TC>>>
AM>> TC>> outgoing works, I can place an end-to-end call ... except the RTP is
AM>> realy
AM>> TC>> delayed ... after approx 30 sec of conversation the audio is delayed
AM>> more
AM>> TC>> than 10 seconds.... but i have 2 way audio for outgoing calls:)
AM>> TC>>
AM>> TC>>
AM>> TC>one more thing ... it is H323 endpoint => SIP phone audio that is
AM>> delayed.
AM>> TC>SIP phone => H323 endpoint is ok!
AM>>
AM>> hm, i have such issue but in reverce direction now.
AM>>
AM>> TC>> Do you need some logs ?
AM>> TC>>
AM>> TC>>
AM>> TC>> Inbound cals still the same... i suppose you didn't have a chance
AM>> working
AM>> TC>> on that as well ...
AM>> TC>>
AM>> TC>> T.
AM>> TC>>
AM>> TC>
AM>>
AM>> C ?????????                       With Best Regards
AM>> ???????????? ????.                Georgiewskiy Yuriy
AM>> +7 4872 711666                    +7 4872 711666
AM>> ???? +7 4872 711143               fax +7 4872 711143
AM>> ???????? ??? "?? ?? ??????"       IT Service Ltd
AM>> http://nkoort.ru                  http://nkoort.ru
AM>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
AM>> YG129-RIPE                        YG129-RIPE
AM>>
AM>> _______________________________________________
AM>> FreeSWITCH-users mailing list
AM>> FreeSWITCH-users at lists.freeswitch.org
AM>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
AM>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
AM>> http://www.freeswitch.org
AM>>
AM>>
AM>
AM>
AM>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From codecomplete at free.fr  Thu Oct 22 07:48:40 2009
From: codecomplete at free.fr (Fred-145)
Date: Thu, 22 Oct 2009 07:48:40 -0700 (PDT)
Subject: [Freeswitch-users] Porting Freeswitch to ARM?
In-Reply-To: <737802AA-46AB-4327-8F26-E0F7D3E6D921@jerris.com>
References: <25531239.post@talk.nabble.com>
	<737802AA-46AB-4327-8F26-E0F7D3E6D921@jerris.com>
Message-ID: <26011319.post@talk.nabble.com>


Thanks Mike for the link. I'll investigate more whether running FS on
ARM-based devices is a good idea.

For those interested, another thread on the subject:

http://www.nabble.com/Freeswitch-vs.-Asterisk-speed-on-ARM-td25086585.html
-- 
View this message in context: http://www.nabble.com/Porting-Freeswitch-to-ARM--tp25531239p26011319.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From tculjaga at gmail.com  Thu Oct 22 07:53:06 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 22 Oct 2009 16:53:06 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>
Message-ID: <65d96fc80910220753n6d2efa39l470791f75db9a640@mail.gmail.com>

>
> TC>
> TC>Do you need some logs ?
>
> try disable medai-proxy, there is issue with rtp now then medai-proxy or
> transcoding enabled.
>
>
Outbound calls:

disabled rtp proxy and it is still the same issue ... audio delay H323 =>
SIP endpoint.






Inbound calls:

This is the extension i use to register my Avaya SIP phone to FS.


<include>
  <user id="1001">
    <params>
      <param name="password" value="$${default_password}"/>
      <param name="vm-password" value="1001"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="1001"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="Extension 1001"/>
      <variable name="effective_caller_id_number" value="1001"/>
      <variable name="outbound_caller_id_name"
value="$${outbound_caller_name}"/>
      <variable name="outbound_caller_id_number"
value="$${outbound_caller_id}"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>


This is my h323.conf.xml


<configuration name="h323.conf" description="H323 Endpoints">
  <settings>
    <param name="trace-level" value="4"/>
    <param name="context" value="default"/>
    <param name="dialplan" value="XML"/>
    <param name="codec-prefs" value="PCMU,PCMA,GSM,G729,G726"/>
    <param name="gk-address" value=""/>    <!-- empty to disable, "*" to
search LAN -->
    <param name="gk-identifer" value=""/>  <!-- optional name of gk -->
    <param name="gk-interface" value=""/>  <!-- optional listener interface
name -->
  </settings>
  <listeners>
    <listener name="default">
      <param name="h323-ip" value="10.4.62.7"/>
      <param name="h323-port" value="1720"/>
    </listener>
  </listeners>
</configuration>

I'm using default context and an inbound call looks for a registered user in
default context where 1001 user is registered to.



here is the log for an outgoing call:
http://pastebin.freeswitch.org/10799and here is a tshark output:
http://pastebin.freeswitch.org/10800


there are 2 thing that are not working here:


1. no audio at all!
2. hangup from SIP User side doesn't release the H323 leg











two points for your reference in the logs:


1. Here, SIP User disconnected the SIP leg, but nothing was triggered in
mod_h323 ... as the callback function "on_hangup" (in mod_h323.cpp) was
never triggered!

freeswitch at subZero>
freeswitch at subZero>
freeswitch at subZero> recv 371 bytes from udp/[10.4.62.89]:5060 at
14:39:36.714521:
   ------------------------------------------------------------------------
   BYE sip:mod_sofia at 10.4.62.7:5060 SIP/2.0
   From: <sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>
;transport=udp>;tag=-22166b474ae08abf-7_T10.4.62.89
   To: sip:1001282166 at 10.4.62.7 <sip%3A1001282166 at 10.4.62.7>
;tag=Qpc53NZ2cZc1N
   Call-ID: 8aa825c6-39bb-122d-bb89-00110a5be1f0
   CSeq: 127 BYE
   Via: SIP/2.0/UDP
10.4.62.89;branch=z9hG4bK-7e5dc720_442d0f8-2d8bf1174f235bec_B
   Content-Length: 0
   Max-Forwards: 70
   Supported: replaces

   ------------------------------------------------------------------------
2009-10-22 16:39:36.714604 [NOTICE] sofia.c:322 Hangup sofia/internal/
sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [CS_CONSUME_MEDIA]
[NORMAL_CLEARING]
2009-10-22 16:39:36.714604 [DEBUG] switch_channel.c:1683 Send signal
sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [KILL]
2009-10-22 16:39:36.714604 [DEBUG] switch_core_session.c:932 Send signal
sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [BREAK]
send 520 bytes to udp/[10.4.62.89]:5060 at 14:39:36.715258:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP
10.4.62.89;branch=z9hG4bK-7e5dc720_442d0f8-2d8bf1174f235bec_B
   From: <sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>
;transport=udp>;tag=-22166b474ae08abf-7_T10.4.62.89
   To: sip:1001282166 at 10.4.62.7 <sip%3A1001282166 at 10.4.62.7>
;tag=Qpc53NZ2cZc1N
   Call-ID: 8aa825c6-39bb-122d-bb89-00110a5be1f0
   CSeq: 127 BYE
   User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Content-Length: 0

   ------------------------------------------------------------------------
2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:503
(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State
CONSUME_MEDIA going to sleep
2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) Running State
Change CS_HANGUP
2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:434
(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State HANGUP
2009-10-22 16:39:36.721097 [DEBUG] mod_sofia.c:338 Channel sofia/internal/
sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> hanging up, cause:
NORMAL_CLEARING
2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:46
sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> Standard HANGUP,
cause: NORMAL_CLEARING
2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:434
(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State HANGUP
going to sleep
2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:476
(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State Change
CS_HANGUP -> CS_REPORTING
2009-10-22 16:39:36.721097 [DEBUG] switch_core_session.c:932 Send signal
sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [BREAK]
2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) Running State
Change CS_REPORTING
2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:612
(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State REPORTING
2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:53
sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> Standard
REPORTING, cause: NORMAL_CLEARING
2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:612
(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State REPORTING
going to sleep
2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:411
(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State Change
CS_REPORTING -> CS_DESTROY
2009-10-22 16:39:36.721097 [DEBUG] switch_core_session.c:1068 Session 10
(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) Locked, Waiting
on external entities

freeswitch at subZero>





2. switch_ivr_originate.c complains => 2009-10-22 16:39:41.630243 [ERR]
switch_ivr_originate.c:1510 Cannot create outgoing channel of type [user]
cause: [NORMAL_CLEARING]


freeswitch at subZero>
freeswitch at subZero>
freeswitch at subZero> 2009-10-22 16:39:41.630243 [DEBUG] h323pdu.cxx:609
Receiving PDU [ip$10.4.62.7:1720/ip$10.4.62.31:16013] :
  {
    q931pdu = {
      protocolDiscriminator = 8
      callReference = 23
      from = originator
      messageType = ReleaseComplete
      IE: Cause - Normal ca
ll clearing = {
        80 90                                              ..
      }
      IE: User-User = {
        25 80 06 00 08 91 4a 00  04 11 00 11 00 96 40 1f   %.....J....... at .
        4e be 4f 11 de 80 76 ae  99 96 03 ba ff 10 80 01   N.O.
..v.........
        ...
      }
    }
    h225pdu = {
      h323_uu_pdu = {
        h323_message_body = releaseComplete {
          protocolIdentifier = 0.0.8.2250.0.4
          callIdentifier = {
            guid =  16 octets {
              96 40
 1f 4e be 4f 11 de  80 76 ae 99 96 03 ba ff   . at .N.O...v......
            }
          }
        }
        h245Tunneling = true
      }
    }
  }
2009-10-22 16:39:41.630243 [DEBUG] h323.cxx:1292 Handling PDU:
ReleaseComplete callRef=23
2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:745
======>FSH323Connection::OnReceivedReleaseComplete cause = Normal call
clearing value = 16
2009-10-22 16:39:41.630243 [NOTICE] mod_h323.cpp:749 Hangup h323/1001
[CS_EXECUTE] [NORMAL_CLEARING]
2009-10-22 16:39:41.630243 [DEBUG] switch_channel.c:1683 Send signal
h323/1001 [KILL]
2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:840
======>FSH323Connection::kill_channel FSH323Connection
2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:841 Kill 1 on connection
FSH323Connection
2009-10-22 16:39:41.630243 [DEBUG] switch_core_session.c:630 Send signal
h323/1001 [BREAK]
2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:840
======>FSH323Connection::kill_channel FSH323Connection
2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:841 Kill 3 on connection
FSH323Connection
2009-10-22 16:39:41.630243 [DEBUG] switch_ivr_originate.c:2138 Originate
Resulted in Error Cause: 16 [NORMAL_CLEARING]
2009-10-22 16:39:41.630243 [ERR] switch_ivr_originate.c:1510 Cannot create
outgoing channel of type [user] cause: [NORMAL_CLEARING]
2009-10-22 16:39:41.630243 [DEBUG] switch_ivr_originate.c:2138 Originate
Resulted in Error Cause: 16 [NORMAL_CLEARING]
2009-10-22 16:39:41.630243 [INFO] mod_dptools.c:2093 Originate Failed.
Cause: NORMAL_CLEARING
2009-10-22 16:39:41.630243 [DEBUG] mod_dptools.c:2120 Continue on fail
[true]:  Cause: NORMAL_CLEARING
2009-10-22 16:39:41.630243 [DEBUG] switch_core_state_machine.c:491
(h323/1001) State EXECUTE going to sleep
2009-10-22 16:39:41.630243 [DEBUG] switch_core_state_machine.c:398
(h323/1001) Running State Change CS_HANGUP
2009-10-22 16:39:41.630243 [DEBUG] switch_core_state_machine.c:434
(h323/1001) State HANGUP
2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:1454 ======>switch_status_t
on_hangup
2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:1459 ----->
2009-10-22 16:39:41.630243 [DEBUG] h323pdu.cxx:1272 Call End Reason Normal
call clearing
2009-10-22 16:39:41.630243 [DEBUG] h323ep.cxx:2694 Clearing connection ip$
10.4.62.31:16013/23 reason=EndedByRemoteUser
2009-10-22 16:39:41.630243 [DEBUG] h323.cxx:1034 Call end reason for ip$
10.4.62.31:16013/23 set to EndedByRemoteUser
2009-10-22 16:39:41.630243 [DEBUG] h323.cxx:1052 Sending release complete
PDU: callRef=23
2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:764
======>FSH323Connection::OnSendReleaseComplete cause = Normal call clearing
value = 16
2009-10-22 16:39:41.630243 [DEBUG] h323pdu.cxx:609 Sending PDU
[(noaddr)/(noaddr)] :
  command endSessionCommand disconnect <<null>>
2009-10-22 16:39:41.630243 [DEBUG] h323pdu.cxx:609 Sending PDU [ip$
10.4.62.7:1720/ip$10.4.62.31:16013] :
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From tculjaga at gmail.com  Thu Oct 22 08:01:39 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 22 Oct 2009 17:01:39 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910221842540.8225@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<65d96fc80910220704t283bfca3w2129ff3575659f97@mail.gmail.com>
	<Pine.LNX.4.64.0910221811590.1929@srv.icf.org.ru>
	<191c3a030910220727v7e62f47dhf172ba8682d5d52a@mail.gmail.com>
	<Pine.LNX.4.64.0910221842540.8225@srv.icf.org.ru>
Message-ID: <65d96fc80910220801m53996d32h531d1e65aca77f0@mail.gmail.com>

2009/10/22 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-22 09:27 -0500, Anthony Minessale wrote
> freeswitch-users at lists.f...:
>
> AM>crash protection has been completely removed from FreeSWITCH, I
> certianly
> AM>hope you are working on this against SVN trunk?
>
> i don't have trunk at this time, my current work is based on 1.0.4 version.
>
>
Yuriy,

it is better if we move this through a jira ticket, this way it is a mess.
So if you agree, we can open a ticket where we can follow up all issues with
mod_h323.
Also, the same applies to FS trunk... first i wanted to see if i was doing
something wrong when i tried your module. Now, when you fixed outgoing calls
it is time to go on trunk as when we finish this 1.0.4 will be outdated and
obsolete.


so, to continue on this topic i suggest:

1. open a jira ticket
2. move to fs-trunk
3. upload the current src of mod_h323 to the FSSVN


do you agree ?



Tihomir.
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From pippyduck1127 at hotmail.com  Thu Oct 22 08:34:22 2009
From: pippyduck1127 at hotmail.com (Pajongjit Buntaokit)
Date: Thu, 22 Oct 2009 15:34:22 +0000
Subject: [Freeswitch-users] Problem when load CDR Data on the database in
	Real Time
Message-ID: <BAY119-W443376A5BEC1012E7888BDB3BE0@phx.gbl>


Hello Anyone,

Now, I'm attempting to load CDR Data on the database in Real Time.
by following the instruction on this link 
http://wiki.freeswitch.org/wiki/CDR


However, when trying to send create_table.rb to the database,
I'm still struggling with connecting to mySQL database

Note: 

my Operation System is WindowXP 2002.

the ruby version that I used is 1.9.1 p129.

Below is the error message show in the Command Prompt:
------------------------------------------------------------------------- 
C:\cdr>ruby create_table.rb
C:/Ruby19/lib/ruby/gems/1.9.1/gems/sequel-3.5.0/lib/sequel/adapters/mysql.rb:157
:in `query': Mysql::Error: Commands out of sync; you can't run this command now
(Sequel::DatabaseDisconnectError)
        from C:/Ruby19/lib/ruby/gems/1.9.1/gems/sequel-3.5.0/lib/sequel/adapters
/mysql.rb:157:in `_execute'
        from C:/Ruby19/lib/ruby/gems/1.9.1/gems/sequel-3.5.0/lib/sequel/adapters
/mysql.rb:140:in `block in execute'
        from C:/Ruby19/lib/ruby/gems/1.9.1/gems/sequel-3.5.0/lib/sequel/connecti
on_pool.rb:112:in `hold'
        from C:/Ruby19/lib/ruby/gems/1.9.1/gems/sequel-3.5.0/lib/sequel/database
.rb:482:in `synchronize'
        from C:/Ruby19/lib/ruby/gems/1.9.1/gems/sequel-3.5.0/lib/sequel/adapters
/mysql.rb:140:in `execute'
        from C:/Ruby19/lib/ruby/gems/1.9.1/gems/sequel-3.5.0/lib/sequel/database
.rb:313:in `execute_dui'
        from C:/Ruby19/lib/ruby/gems/1.9.1/gems/sequel-3.5.0/lib/sequel/database
.rb:306:in `execute_ddl'
        from C:/Ruby19/lib/ruby/gems/1.9.1/gems/sequel-3.5.0/lib/sequel/database
/schema_methods.rb:188:in `create_table_from_generator'
        from C:/Ruby19/lib/ruby/gems/1.9.1/gems/sequel-3.5.0/lib/sequel/database
/schema_methods.rb:73:in `create_table'
        from create_table.rb:6:in `<main>'
<main>:306: [BUG] Segmentation fault
ruby 1.9.1p129 (2009-05-12 revision 23412) [i386-mswin32]

-- control frame ----------
c:0001 p:0000 s:0002 b:0002 l:0010c4 d:0010c4 TOP    <main>:306
---------------------------
-- Ruby level backtrace information-----------------------------------------

[NOTE]
You may encounter a bug of Ruby interpreter. Bug reports are welcome.
For details: http://www.ruby-lang.org/bugreport.html


This application has requested the Runtime to terminate it in an unusual way.
Please contact the application's support team for more information.

C:\cdr>
------------------------------------------------------------------------- 

Has anyone ever had this problem? Pleaes advise.

PB.






 		 	   		  
_________________________________________________________________
Windows 7: Simplify your PC. Learn more.
http://www.microsoft.com/Windows/windows-7/default.aspx?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_evergreen1:102009
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From freeswitch.noob at gmail.com  Thu Oct 22 08:42:11 2009
From: freeswitch.noob at gmail.com (freeswitch noob)
Date: Thu, 22 Oct 2009 10:42:11 -0500
Subject: [Freeswitch-users] Domain Question
Message-ID: <7f783b530910220842q5c01ddbcj28a6157c30badd52@mail.gmail.com>

I have a noobish question about setting up FS.

I have it installed and running.

I setup a soft client on the machine fs is on and point it to the ip address
of the FS instance and it registers with no issues.

I then setup an entry in my etc/hosts files "mydomain.localhost" and changed
the domain in the soft client.

The registration now fails. with the soft phone giving a 503 error "service
unavailable"

And I am watching the cli and I see no errors come through with the domain
but I see it register with the IP :(

I change it back to the ip address and it works.

Any direction you guys could point me in would be great.
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From kristian.kielhofner at gmail.com  Thu Oct 22 08:44:30 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Thu, 22 Oct 2009 11:44:30 -0400
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
In-Reply-To: <191c3a030910211154gc8412b5gd999fbedffa02b53@mail.gmail.com>
References: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>
	<191c3a030910201434q4c6e65e1h58c5f0a82addc2d3@mail.gmail.com>
	<2d9149cd0910202050i36a7ea82nc7253d87e38f990a@mail.gmail.com>
	<191c3a030910211154gc8412b5gd999fbedffa02b53@mail.gmail.com>
Message-ID: <2d9149cd0910220844v2aa73502j4b88fdbd75a0d833@mail.gmail.com>

An update for Tony, Brian, Mike, and everyone on the list...

I was able to get some phone time with the team yesterday.  Tony
worked on my machine, found the issue, and had it committed within 30
minutes.

I've been testing T.38 all morning between the fax machines in the
office with few issues.

THANKS AGAIN GUYS!

On Wed, Oct 21, 2009 at 2:54 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> can you try trunk and let me know right away,
> if it's still not working i may need ssh access and call you on the phone.
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From helmut.kuper at ewetel.de  Thu Oct 22 08:44:52 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Thu, 22 Oct 2009 17:44:52 +0200
Subject: [Freeswitch-users] Qustion about INFO messages
	after	Connect/Answer
In-Reply-To: <87f2f3b90910211048s208daf1eyfae501091ab75530@mail.gmail.com>
References: <4AD859A4.5000101@ewetel.de>	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>	<4ADC7C09.6070503@ewetel.de>	<191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>	<4ADC85C0.7060908@ewetel.de>
	<4ADC8B6E.7070404@ewetel.de>	<191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>	<87f2f3b90910191435s7528e256r85df5f12fc634f24@mail.gmail.com>	<4ADF483B.4050807@ewetel.de>
	<87f2f3b90910211048s208daf1eyfae501091ab75530@mail.gmail.com>
Message-ID: <4AE07DF4.5040803@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Mike,

here it is:


Dialplan:


    <extension name="Local_Extension">
      <condition field="destination_number" expression="^(10[01][0-9])$">
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
        <action application="set" data="transfer_ringback=$${hold_music}"/>
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="bridge"
data="user/${dialed_extension}@${domain_name}"/>
      </condition>
    </extension>






Debug-Log:


recv 1521 bytes from udp/[85.16.245.206]:1024 at 15:41:02.405834:
   ------------------------------------------------------------------------
   INVITE sip:1000 at 85.16.246.12:5061;user=phone SIP/2.0
   Via: SIP/2.0/UDP 85.16.245.206:1024;branch=z9hG4bK-i48n75d62qts;rport
   From: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=7xpim4o1go
   To: <sip:1000 at 85.16.246.12:5061;user=phone>
   Call-ID: 3c31304b7a80-no9xsnjj0bol
   CSeq: 1 INVITE
   Max-Forwards: 70
   Contact: <sip:1001 at 85.16.245.206:1024;line=eg3wp69a>;reg-id=1
   X-Serialnumber: 0004134002CB
   P-Key-Flags: resolution="31x13", keys="4"
   User-Agent: snom820/8.2.16
   Accept: application/sdp
   Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE
   Allow-Events: talk, hold, refer, call-info
   Supported: timer, 100rel, replaces, from-change
   Session-Expires: 3600;refresher=uas
   Min-SE: 90
   Content-Type: application/sdp
   Content-Length: 732

   v=0
   o=root 411395140 411395140 IN IP4 85.16.245.206
   s=call
   c=IN IP4 85.16.245.206
   t=0 0
   m=audio 49852 RTP/SAVP 0 8 9 99 3 18 4 101
   a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:Um06txYC7vwXdZDohXJ15te5hZ/iWmPd4voRbdby
   a=rtpmap:0 pcmu/8000
   a=rtpmap:8 pcma/8000
   a=rtpmap:9 g722/8000
   a=rtpmap:99 g726-32/8000
   a=rtpmap:3 gsm/8000
   a=rtpmap:18 g729/8000
   a=rtpmap:4 g723/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   a=sendrecv
   m=audio 49852 RTP/AVP 0 8 9 99 3 18 4 101
   a=rtpmap:0 pcmu/8000
   a=rtpmap:8 pcma/8000
   a=rtpmap:9 g722/8000
   a=rtpmap:99 g726-32/8000
   a=rtpmap:3 gsm/8000
   a=rtpmap:18 g729/8000
   a=rtpmap:4 g723/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   a=sendrecv
   ------------------------------------------------------------------------
send 331 bytes to udp/[85.16.245.206]:1024 at 15:41:02.406500:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP
85.16.245.206:1024;branch=z9hG4bK-i48n75d62qts;rport=1024
   From: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=7xpim4o1go
   To: <sip:1000 at 85.16.246.12:5061;user=phone>
   Call-ID: 3c31304b7a80-no9xsnjj0bol
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15180M
   Content-Length: 0

   ------------------------------------------------------------------------
2009-10-22 17:41:02.406509 [DEBUG] sofia.c:4907 IP 85.16.245.206
Approved by acl "clients[]". Access Granted.
2009-10-22 17:41:02.406509 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/1001 at 85.16.246.12:5061 [4d941750-bf21-11de-9c3f-adfc1789590a]
2009-10-22 17:41:02.407509 [DEBUG] sofia.c:3493 Channel
sofia/internal/1001 at 85.16.246.12:5061 entering state [received][100]
2009-10-22 17:41:02.407509 [DEBUG] sofia.c:3500 Remote SDP:
v=0
o=root 411395140 411395140 IN IP4 85.16.245.206
s=call
c=IN IP4 85.16.245.206
t=0 0
m=audio 49852 RTP/SAVP 0 8 9 99 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:Um06txYC7vwXdZDohXJ15te5hZ/iWmPd4voRbdby
a=ptime:20
m=audio 49852 RTP/AVP 0 8 9 99 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

2009-10-22 17:41:02.407509 [DEBUG] sofia.c:3621
(sofia/internal/1001 at 85.16.246.12:5061) State Change CS_NEW -> CS_INIT
2009-10-22 17:41:02.407509 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-22 17:41:02.407509 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_INIT
2009-10-22 17:41:02.407509 [DEBUG] switch_core_state_machine.c:330
(sofia/internal/1001 at 85.16.246.12:5061) State INIT
2009-10-22 17:41:02.407509 [DEBUG] mod_sofia.c:83
sofia/internal/1001 at 85.16.246.12:5061 SOFIA INIT
2009-10-22 17:41:02.407509 [DEBUG] mod_sofia.c:111
(sofia/internal/1001 at 85.16.246.12:5061) State Change CS_INIT -> CS_ROUTING
2009-10-22 17:41:02.407509 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-22 17:41:02.407509 [DEBUG] switch_core_state_machine.c:330
(sofia/internal/1001 at 85.16.246.12:5061) State INIT going to sleep
2009-10-22 17:41:02.407509 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_ROUTING
2009-10-22 17:41:02.407509 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/1001 at 85.16.246.12:5061) State ROUTING
2009-10-22 17:41:02.407509 [DEBUG] mod_sofia.c:130
sofia/internal/1001 at 85.16.246.12:5061 SOFIA ROUTING
2009-10-22 17:41:02.407509 [DEBUG] switch_core_state_machine.c:78
sofia/internal/1001 at 85.16.246.12:5061 Standard ROUTING
2009-10-22 17:41:02.407509 [INFO] mod_dialplan_xml.c:391 Processing 1001
an PBX1->1000 in context default
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->anonymous] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [anonymous]
destination_number(1000) =~ /^\*31([0-9]+)$/ break=never
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 ANTI-Action
set(dialed_extension=${destination_number})
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->is_local] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [is_local]
${ET_is_local}() =~ /(true|false)/ break=on-true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 ANTI-Action
lua(ET_is_local.lua)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 ANTI-Action
transfer(${destination_number} XML default)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->set_domain] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (PASS)
[set_domain] destination_number(1000) =~ /^.*$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
set(domain_name=85.16.246.12)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->302_zero_problem] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[302_zero_problem] ${sip_looped_call}() =~ /true/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->fakeTrisko-Servicenumber as callerid] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[fakeTrisko-Servicenumber as callerid] caller_id_number(1001) =~
/^(49[1-2][0-9])$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->outbound-ssw-ddi] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[outbound-ssw-ddi] ${dialed_extension}() =~ /^0([0-9]+)$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing [default->mail]
continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [mail]
destination_number(1000) =~ /^9997$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing [default->pizza]
continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [pizza]
destination_number(1000) =~ /^2006$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing [default->pbook]
continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [pbook]
destination_number(1000) =~ /^2007$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->shout_1] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [shout_1]
${dialed_extension}() =~ /^9123$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->shout_2] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [shout_2]
${dialed_extension}() =~ /^9124$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->speex_32000] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[speex_32000] ${dialed_extension}() =~ /^9125$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->speex_32000_2] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[speex_32000_2] ${dialed_extension}() =~ /^9126$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->jukebox] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [jukebox]
destination_number(1000) =~ /^9111$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->pickup-extension] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[pickup-extension] ${dialed_extension}() =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->Voicemail-Verwaltung] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[Voicemail-Verwaltung] ${dialed_extension}() =~ /^(1001|9999)$/
break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->Local_Extension] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[Local_Extension] ${ET_is_local}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->parallel call] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [parallel
call] ${dialed_extension}() =~ /^(6000)$/ break=on-false
2009-10-22 17:41:02.409510 [DEBUG] switch_core_state_machine.c:114
(sofia/internal/1001 at 85.16.246.12:5061) State Change CS_ROUTING ->
CS_EXECUTE
2009-10-22 17:41:02.409510 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-22 17:41:02.409510 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/1001 at 85.16.246.12:5061) State ROUTING going to sleep
2009-10-22 17:41:02.409510 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_EXECUTE
2009-10-22 17:41:02.409510 [DEBUG] switch_core_state_machine.c:340
(sofia/internal/1001 at 85.16.246.12:5061) State EXECUTE
2009-10-22 17:41:02.409510 [DEBUG] mod_sofia.c:173
sofia/internal/1001 at 85.16.246.12:5061 SOFIA EXECUTE
2009-10-22 17:41:02.409510 [DEBUG] switch_core_state_machine.c:151
sofia/internal/1001 at 85.16.246.12:5061 Standard EXECUTE
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 set(dialed_extension=1000)
2009-10-22 17:41:02.409510 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET [dialed_extension]=[1000]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 lua(ET_is_local.lua)
2009-10-22 17:41:02.434510 [DEBUG] switch_cpp.cpp:925 destroy/unlink
session from object
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 transfer(1000 XML default)
2009-10-22 17:41:02.434510 [DEBUG] switch_ivr.c:1381
(sofia/internal/1001 at 85.16.246.12:5061) State Change CS_EXECUTE ->
CS_ROUTING
2009-10-22 17:41:02.434510 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-22 17:41:02.434510 [DEBUG] switch_ivr.c:1385
sofia/internal/1001 at 85.16.246.12:5061 receive message [TRANSFER]
2009-10-22 17:41:02.434510 [DEBUG] switch_core_session.c:647 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-22 17:41:02.434510 [NOTICE] switch_ivr.c:1387 Transfer
sofia/internal/1001 at 85.16.246.12:5061 to XML[1000 at default]
2009-10-22 17:41:02.435510 [DEBUG] switch_core_state_machine.c:340
(sofia/internal/1001 at 85.16.246.12:5061) State EXECUTE going to sleep
2009-10-22 17:41:02.435510 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_ROUTING
2009-10-22 17:41:02.435510 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/1001 at 85.16.246.12:5061) State ROUTING
2009-10-22 17:41:02.435510 [DEBUG] mod_sofia.c:130
sofia/internal/1001 at 85.16.246.12:5061 SOFIA ROUTING
2009-10-22 17:41:02.435510 [DEBUG] switch_core_state_machine.c:78
sofia/internal/1001 at 85.16.246.12:5061 Standard ROUTING
2009-10-22 17:41:02.435510 [INFO] mod_dialplan_xml.c:391 Processing 1001
an PBX1->1000 in context default
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->anonymous] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [anonymous]
destination_number(1000) =~ /^\*31([0-9]+)$/ break=never
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 ANTI-Action
set(dialed_extension=${destination_number})
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->is_local] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (PASS) [is_local]
${ET_is_local}(true) =~ /(true|false)/ break=on-true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->set_domain] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (PASS)
[set_domain] destination_number(1000) =~ /^.*$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
set(domain_name=85.16.246.12)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->302_zero_problem] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[302_zero_problem] ${sip_looped_call}() =~ /true/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->fakeTrisko-Servicenumber as callerid] continue=true
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[fakeTrisko-Servicenumber as callerid] caller_id_number(1001) =~
/^(49[1-2][0-9])$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->outbound-ssw-ddi] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[outbound-ssw-ddi] ${dialed_extension}(1000) =~ /^0([0-9]+)$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing [default->mail]
continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [mail]
destination_number(1000) =~ /^9997$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing [default->pizza]
continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [pizza]
destination_number(1000) =~ /^2006$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing [default->pbook]
continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [pbook]
destination_number(1000) =~ /^2007$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->shout_1] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [shout_1]
${dialed_extension}(1000) =~ /^9123$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->shout_2] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [shout_2]
${dialed_extension}(1000) =~ /^9124$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->speex_32000] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[speex_32000] ${dialed_extension}(1000) =~ /^9125$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->speex_32000_2] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[speex_32000_2] ${dialed_extension}(1000) =~ /^9126$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->jukebox] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL) [jukebox]
destination_number(1000) =~ /^9111$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->pickup-extension] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[pickup-extension] ${dialed_extension}(1000) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->Voicemail-Verwaltung] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (FAIL)
[Voicemail-Verwaltung] ${dialed_extension}(1000) =~ /^(1001|9999)$/
break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 parsing
[default->Local_Extension] continue=false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Regex (PASS)
[Local_Extension] ${ET_is_local}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
set(dialed_extension=${destination_number})
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
export(origination_callee_id_name=hubu)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
export(dialed_extension=${destination_number})
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
set(transfer_ringback=local_stream://moh)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
set(hangup_after_bridge=true)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
export(nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name}
var sip_secure_media)})
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
db(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid})
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
set(continue_on_fail=false)
Dialplan: sofia/internal/1001 at 85.16.246.12:5061 Action
bridge(user/${dialed_extension}@85.16.246.12)
2009-10-22 17:41:02.436510 [DEBUG] switch_core_state_machine.c:114
(sofia/internal/1001 at 85.16.246.12:5061) State Change CS_ROUTING ->
CS_EXECUTE
2009-10-22 17:41:02.436510 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-22 17:41:02.436510 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/1001 at 85.16.246.12:5061) State ROUTING going to sleep
2009-10-22 17:41:02.436510 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_EXECUTE
2009-10-22 17:41:02.436510 [DEBUG] switch_core_state_machine.c:340
(sofia/internal/1001 at 85.16.246.12:5061) State EXECUTE
2009-10-22 17:41:02.436510 [DEBUG] mod_sofia.c:173
sofia/internal/1001 at 85.16.246.12:5061 SOFIA EXECUTE
2009-10-22 17:41:02.436510 [DEBUG] switch_core_state_machine.c:151
sofia/internal/1001 at 85.16.246.12:5061 Standard EXECUTE
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 set(dialed_extension=1000)
2009-10-22 17:41:02.436510 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET [dialed_extension]=[1000]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 set(domain_name=85.16.246.12)
2009-10-22 17:41:02.437526 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET [domain_name]=[85.16.246.12]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 set(dialed_extension=1000)
2009-10-22 17:41:02.437526 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET [dialed_extension]=[1000]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061
export(origination_callee_id_name=hubu)
2009-10-22 17:41:02.437526 [DEBUG] mod_dptools.c:835 EXPORT
[origination_callee_id_name]=[hubu]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 export(dialed_extension=1000)
2009-10-22 17:41:02.437526 [DEBUG] mod_dptools.c:835 EXPORT
[dialed_extension]=[1000]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061
set(transfer_ringback=local_stream://moh)
2009-10-22 17:41:02.438511 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET
[transfer_ringback]=[local_stream://moh]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 set(hangup_after_bridge=true)
2009-10-22 17:41:02.438511 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061
export(nolocal:sip_secure_media=)
2009-10-22 17:41:02.461511 [DEBUG] mod_dptools.c:835 EXPORT (REMOTE
ONLY) [sip_secure_media]=[UNDEF]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061
db(insert/85.16.246.12-last_dial//4d941750-bf21-11de-9c3f-adfc1789590a)
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 set(continue_on_fail=false)
2009-10-22 17:41:02.654513 [DEBUG] mod_dptools.c:752
sofia/internal/1001 at 85.16.246.12:5061 SET [continue_on_fail]=[false]
EXECUTE sofia/internal/1001 at 85.16.246.12:5061 bridge(user/1000 at 85.16.246.12)
2009-10-22 17:41:02.679515 [DEBUG] switch_ivr_originate.c:1163 variable
string 0 = [presence_id=1000 at 85.16.246.12]
2009-10-22 17:41:02.679515 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/sip:1000 at 85.16.245.228:1024
[4dbdc384-bf21-11de-9c3f-adfc1789590a]
2009-10-22 17:41:02.679515 [DEBUG] mod_sofia.c:2981
(sofia/internal/sip:1000 at 85.16.245.228:1024) State Change CS_NEW -> CS_INIT
2009-10-22 17:41:02.679515 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-22 17:41:02.679515 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/sip:1000 at 85.16.245.228:1024) Running State Change CS_INIT
2009-10-22 17:41:02.680516 [DEBUG] switch_core_state_machine.c:330
(sofia/internal/sip:1000 at 85.16.245.228:1024) State INIT
2009-10-22 17:41:02.680516 [DEBUG] mod_sofia.c:83
sofia/internal/sip:1000 at 85.16.245.228:1024 SOFIA INIT
send 1310 bytes to udp/[85.16.245.228]:1024 at 15:41:02.681508:
   ------------------------------------------------------------------------
   INVITE sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
   Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKKXH41BcZjB8FS
   Max-Forwards: 68
   From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=HKSNX8D1pKy6K
   To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>
   Call-ID: 251c312d-39c4-122d-3fbf-00144fe6907a
   CSeq: 121994695 INVITE
   Contact: <sip:mod_sofia at 85.16.246.12:5061>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15180M
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
2009-10-22 17:41:02.681515 [DEBUG] mod_sofia.c:111
(sofia/internal/sip:1000 at 85.16.245.228:1024) State Change CS_INIT ->
CS_ROUTING
2009-10-22 17:41:02.681515 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
   Content-Type: application/sdp
   Content-Disposition: session
2009-10-22 17:41:02.681515 [DEBUG] switch_core_state_machine.c:330
(sofia/internal/sip:1000 at 85.16.245.228:1024) State INIT going to sleep
   Content-Length: 325
   X-Serialnumber: 0004134002CB
2009-10-22 17:41:02.681515 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/sip:1000 at 85.16.245.228:1024) Running State Change CS_ROUTING
   P-Key-Flags: resolution="31x13", keys="4"
   X-Actually-Support: UPDATE
   Remote-Party-ID: "1001 an PBX1"
<sip:1001 at 85.16.246.12>;party=calling;screen=yes;privacy=off

   v=0
   o=FreeSWITCH 1256212358 1256212359 IN IP4 85.16.246.12
   s=FreeSWITCH
   c=IN IP4 85.16.246.12
   t=0 0
   m=audio 13704 RTP/AVP 95 103 9 8 101 13
   a=rtpmap:95 CELT/48000
   a=rtpmap:103 SPEEX/32000
   a=rtpmap:9 G722/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:13 CN/8000
   a=ptime:10
   ------------------------------------------------------------------------
2009-10-22 17:41:02.681515 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/sip:1000 at 85.16.245.228:1024) State ROUTING
2009-10-22 17:41:02.681515 [DEBUG] mod_sofia.c:130
sofia/internal/sip:1000 at 85.16.245.228:1024 SOFIA ROUTING
2009-10-22 17:41:02.681515 [DEBUG] switch_ivr_originate.c:66
(sofia/internal/sip:1000 at 85.16.245.228:1024) State Change CS_ROUTING ->
CS_CONSUME_MEDIA
2009-10-22 17:41:02.681515 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-22 17:41:02.681515 [DEBUG] sofia.c:3493 Channel
sofia/internal/sip:1000 at 85.16.245.228:1024 entering state [calling][0]
2009-10-22 17:41:02.681515 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/sip:1000 at 85.16.245.228:1024) State ROUTING going to sleep
2009-10-22 17:41:02.681515 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/sip:1000 at 85.16.245.228:1024) Running State Change
CS_CONSUME_MEDIA
2009-10-22 17:41:02.681515 [DEBUG] switch_core_state_machine.c:352
(sofia/internal/sip:1000 at 85.16.245.228:1024) State CONSUME_MEDIA
2009-10-22 17:41:02.681515 [DEBUG] switch_core_state_machine.c:352
(sofia/internal/sip:1000 at 85.16.245.228:1024) State CONSUME_MEDIA going
to sleep
recv 521 bytes from udp/[85.16.245.228]:1024 at 15:41:02.706318:
   ------------------------------------------------------------------------
   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKKXH41BcZjB8FS
   From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=HKSNX8D1pKy6K
   To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=sdeidvhpxx
   Call-ID: 251c312d-39c4-122d-3fbf-00144fe6907a
   CSeq: 121994695 INVITE
   Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
   Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE
   Allow-Events: talk, hold, refer, call-info
   Content-Length: 0

   ------------------------------------------------------------------------
2009-10-22 17:41:02.706514 [DEBUG] sofia.c:3493 Channel
sofia/internal/sip:1000 at 85.16.245.228:1024 entering state [proceeding][180]
2009-10-22 17:41:02.706514 [NOTICE] sofia.c:3557 Ring-Ready
sofia/internal/sip:1000 at 85.16.245.228:1024!
2009-10-22 17:41:02.706514 [DEBUG] switch_ivr_originate.c:388
sofia/internal/1001 at 85.16.246.12:5061 receive message [RINGING]
2009-10-22 17:41:02.706514 [NOTICE] mod_sofia.c:1525 Ring-Ready
sofia/internal/1001 at 85.16.246.12:5061!
2009-10-22 17:41:02.706514 [DEBUG] switch_core_session.c:647 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-22 17:41:02.706514 [NOTICE] switch_ivr_originate.c:388 Ring
Ready sofia/internal/1001 at 85.16.246.12:5061!
send 737 bytes to udp/[85.16.245.206]:1024 at 15:41:02.706945:
   ------------------------------------------------------------------------
   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP
85.16.245.206:1024;branch=z9hG4bK-i48n75d62qts;rport=1024
   From: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=7xpim4o1go
   To: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=ga0vUDXXSa8Kr
   Call-ID: 3c31304b7a80-no9xsnjj0bol
   CSeq: 1 INVITE
   Contact: <sip:1000 at 85.16.246.12:5061;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15180M
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   X-Actually-Support: UPDATE

   ------------------------------------------------------------------------
2009-10-22 17:41:02.706514 [DEBUG] sofia.c:3493 Channel
sofia/internal/1001 at 85.16.246.12:5061 entering state [early][180]
recv 521 bytes from udp/[85.16.245.228]:1024 at 15:41:03.203506:
   ------------------------------------------------------------------------
   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKKXH41BcZjB8FS
   From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=HKSNX8D1pKy6K
   To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=sdeidvhpxx
   Call-ID: 251c312d-39c4-122d-3fbf-00144fe6907a
   CSeq: 121994695 INVITE
   Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
   Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE
   Allow-Events: talk, hold, refer, call-info
   Content-Length: 0

   ------------------------------------------------------------------------
2009-10-22 17:41:03.203522 [DEBUG] sofia.c:3493 Channel
sofia/internal/sip:1000 at 85.16.245.228:1024 entering state [proceeding][180]
recv 521 bytes from udp/[85.16.245.228]:1024 at 15:41:04.203621:
   ------------------------------------------------------------------------
   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKKXH41BcZjB8FS
   From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=HKSNX8D1pKy6K
   To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=sdeidvhpxx
   Call-ID: 251c312d-39c4-122d-3fbf-00144fe6907a
   CSeq: 121994695 INVITE
   Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
   Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE
   Allow-Events: talk, hold, refer, call-info
   Content-Length: 0

   ------------------------------------------------------------------------
2009-10-22 17:41:04.203541 [DEBUG] sofia.c:3493 Channel
sofia/internal/sip:1000 at 85.16.245.228:1024 entering state [proceeding][180]
recv 888 bytes from udp/[85.16.245.228]:1024 at 15:41:04.773115:
   ------------------------------------------------------------------------
   SIP/2.0 200 Ok
   Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKKXH41BcZjB8FS
   From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=HKSNX8D1pKy6K
   To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=sdeidvhpxx
   Call-ID: 251c312d-39c4-122d-3fbf-00144fe6907a
   CSeq: 121994695 INVITE
   Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
   User-Agent: snom820/LAB_HD_HEAD_11_09_09_11_20
   Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE
   Allow-Events: talk, hold, refer, call-info
   Supported: timer, 100rel, replaces, from-change
   Content-Type: application/sdp
   Content-Length: 242

   v=0
   o=root 45628059 45628060 IN IP4 85.16.245.228
   s=call
   c=IN IP4 85.16.245.228
   t=0 0
   m=audio 53220 RTP/AVP 9 8 101
   a=rtpmap:9 g722/8000
   a=rtpmap:8 pcma/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:10
   a=sendrecv
   ------------------------------------------------------------------------
send 404 bytes to udp/[85.16.245.228]:1024 at 15:41:04.773445:
   ------------------------------------------------------------------------
   ACK sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
   Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKm6aX36v2Fmy2m
   Max-Forwards: 70
   From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=HKSNX8D1pKy6K
   To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=sdeidvhpxx
   Call-ID: 251c312d-39c4-122d-3fbf-00144fe6907a
   CSeq: 121994695 ACK
   Contact: <sip:mod_sofia at 85.16.246.12:5061>
   Content-Length: 0

   ------------------------------------------------------------------------
2009-10-22 17:41:04.773545 [DEBUG] sofia.c:3493 Channel
sofia/internal/sip:1000 at 85.16.245.228:1024 entering state [ready][200]
2009-10-22 17:41:04.773545 [DEBUG] sofia.c:3500 Remote SDP:
v=0
o=root 45628059 45628060 IN IP4 85.16.245.228
s=call
c=IN IP4 85.16.245.228
t=0 0
m=audio 53220 RTP/AVP 9 8 101
a=rtpmap:9 g722/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:10

2009-10-22 17:41:04.773545 [DEBUG] sofia_glue.c:3143 Audio Codec Compare
[g722:9:8000:10]/[CELT:95:48000:10]
2009-10-22 17:41:04.773545 [DEBUG] sofia_glue.c:3143 Audio Codec Compare
[g722:9:8000:10]/[SPEEX:103:32000:20]
2009-10-22 17:41:04.773545 [DEBUG] sofia_glue.c:3143 Audio Codec Compare
[g722:9:8000:10]/[G722:9:8000:20]
2009-10-22 17:41:04.773545 [DEBUG] sofia_glue.c:3143 Audio Codec Compare
[g722:9:8000:10]/[PCMA:8:8000:20]
2009-10-22 17:41:04.773545 [DEBUG] sofia_glue.c:3191 Substituting codec
G722 at 10i@8000h
2009-10-22 17:41:04.773545 [DEBUG] sofia_glue.c:2101 Set Codec
sofia/internal/sip:1000 at 85.16.245.228:1024 G722/8000 10 ms 80 samples
2009-10-22 17:41:04.773545 [DEBUG] sofia_glue.c:3103 Set 2833 dtmf
payload to 101
2009-10-22 17:41:04.773545 [DEBUG] sofia_glue.c:2335 AUDIO RTP
[sofia/internal/sip:1000 at 85.16.245.228:1024] 85.16.246.12 port 13704 ->
85.16.245.228 port 53220 codec: 9 ms: 10
2009-10-22 17:41:04.773545 [DEBUG] switch_rtp.c:1155 Starting timer
[soft] 80 bytes per 10ms
2009-10-22 17:41:04.774778 [DEBUG] switch_channel.c:2105 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-22 17:41:04.774778 [NOTICE] sofia.c:3983 Channel
[sofia/internal/sip:1000 at 85.16.245.228:1024] has been answered
2009-10-22 17:41:04.774778 [DEBUG] switch_channel.c:182
sofia/internal/sip:1000 at 85.16.245.228:1024 receive message [AUDIO_SYNC]
2009-10-22 17:41:04.776545 [DEBUG] switch_ivr_originate.c:2324
sofia/internal/1001 at 85.16.246.12:5061 receive message [ANSWER]
2009-10-22 17:41:04.776545 [DEBUG] sofia_glue.c:3029 Set Remote Key [1
AES_CM_128_HMAC_SHA1_32 inline:Um06txYC7vwXdZDohXJ15te5hZ/iWmPd4voRbdby]
2009-10-22 17:41:04.776545 [DEBUG] sofia_glue.c:2161 Set Local Key [1
AES_CM_128_HMAC_SHA1_32 inline:EKycEU2CPLFUVM4vFRATL21nluVg3RJhmVwJVwA9]
2009-10-22 17:41:04.776545 [DEBUG] sofia_glue.c:3143 Audio Codec Compare
[pcmu:0:8000:20]/[CELT:95:48000:10]
2009-10-22 17:41:04.776545 [DEBUG] sofia_glue.c:3143 Audio Codec Compare
[pcmu:0:8000:20]/[SPEEX:103:32000:20]
2009-10-22 17:41:04.776545 [DEBUG] sofia_glue.c:3143 Audio Codec Compare
[pcmu:0:8000:20]/[G722:9:8000:20]
2009-10-22 17:41:04.776545 [DEBUG] sofia_glue.c:3143 Audio Codec Compare
[pcmu:0:8000:20]/[PCMA:8:8000:20]
2009-10-22 17:41:04.776545 [DEBUG] sofia_glue.c:3143 Audio Codec Compare
[pcma:8:8000:20]/[CELT:95:48000:10]
2009-10-22 17:41:04.776545 [DEBUG] sofia_glue.c:3143 Audio Codec Compare
[pcma:8:8000:20]/[SPEEX:103:32000:20]
2009-10-22 17:41:04.776545 [DEBUG] sofia_glue.c:3143 Audio Codec Compare
[pcma:8:8000:20]/[G722:9:8000:20]
2009-10-22 17:41:04.776545 [DEBUG] sofia_glue.c:3143 Audio Codec Compare
[pcma:8:8000:20]/[PCMA:8:8000:20]
2009-10-22 17:41:04.776545 [DEBUG] sofia_glue.c:2101 Set Codec
sofia/internal/1001 at 85.16.246.12:5061 PCMA/8000 20 ms 160 samples
2009-10-22 17:41:04.776545 [DEBUG] sofia_glue.c:3103 Set 2833 dtmf
payload to 101
2009-10-22 17:41:04.776545 [DEBUG] sofia_glue.c:3077 Our existing sdp is
still good [pcma 85.16.245.206:49852], let's keep it.
2009-10-22 17:41:04.776545 [DEBUG] sofia_glue.c:2335 AUDIO RTP
[sofia/internal/1001 at 85.16.246.12:5061] 85.16.246.12 port 10062 ->
85.16.245.206 port 49852 codec: 8 ms: 20
2009-10-22 17:41:04.776545 [DEBUG] switch_rtp.c:1155 Starting timer
[soft] 160 bytes per 20ms
2009-10-22 17:41:04.778545 [INFO] switch_rtp.c:1019 Activating Secure
RTP SEND
2009-10-22 17:41:04.778545 [INFO] switch_rtp.c:999 Activating Secure RTP
RECV
2009-10-22 17:41:04.778545 [DEBUG] switch_core_sqldb.c:396 Secure Type:
srtp:AES_CM_128_HMAC_SHA1_32
2009-10-22 17:41:04.778545 [DEBUG] switch_core_sqldb.c:396 Secure Type:
srtp:AES_CM_128_HMAC_SHA1_32
2009-10-22 17:41:04.778545 [NOTICE] sofia_glue.c:2770 Pre-Answer
sofia/internal/1001 at 85.16.246.12:5061!
2009-10-22 17:41:04.778545 [DEBUG] mod_sofia.c:543 Local SDP
sofia/internal/1001 at 85.16.246.12:5061:
v=0
o=FreeSWITCH 1256216002 1256216003 IN IP4 85.16.246.12
s=FreeSWITCH
c=IN IP4 85.16.246.12
t=0 0
m=audio 10062 RTP/SAVP 8 101
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:EKycEU2CPLFUVM4vFRATL21nluVg3RJhmVwJVwA9

2009-10-22 17:41:04.778545 [DEBUG] switch_core_session.c:647 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-22 17:41:04.778545 [NOTICE] switch_ivr_originate.c:2324 Channel
[sofia/internal/1001 at 85.16.246.12:5061] has been answered
2009-10-22 17:41:04.778545 [DEBUG] switch_channel.c:182
sofia/internal/1001 at 85.16.246.12:5061 receive message [AUDIO_SYNC]
2009-10-22 17:41:04.778545 [DEBUG] switch_ivr_originate.c:2368 Originate
Resulted in Success: [sofia/internal/sip:1000 at 85.16.245.228:1024]
2009-10-22 17:41:04.778545 [DEBUG] switch_channel.c:182
sofia/internal/sip:1000 at 85.16.245.228:1024 receive message [AUDIO_SYNC]
2009-10-22 17:41:04.778545 [DEBUG] switch_channel.c:182
sofia/internal/1001 at 85.16.246.12:5061 receive message [AUDIO_SYNC]
send 1174 bytes to udp/[85.16.245.206]:1024 at 15:41:04.779182:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP
85.16.245.206:1024;branch=z9hG4bK-i48n75d62qts;rport=1024
   From: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=7xpim4o1go
   To: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=ga0vUDXXSa8Kr
   Call-ID: 3c31304b7a80-no9xsnjj0bol
   CSeq: 1 INVITE
   Contact: <sip:1000 at 85.16.246.12:5061;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15180M
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Session-Expires: 3600;refresher=uas
   Min-SE: 120
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 354
   X-Actually-Support: UPDATE

   v=0
   o=FreeSWITCH 1256216002 1256216003 IN IP4 85.16.246.12
   s=FreeSWITCH
   c=IN IP4 85.16.246.12
   t=0 0
   m=audio 10062 RTP/SAVP 8 101
   a=rtpmap:8 pcma/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20
   a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:EKycEU2CPLFUVM4vFRATL21nluVg3RJhmVwJVwA9
   m=audio 0 RTP/AVP 19
   ------------------------------------------------------------------------
2009-10-22 17:41:04.778545 [DEBUG] sofia.c:3493 Channel
sofia/internal/1001 at 85.16.246.12:5061 entering state [completed][200]
2009-10-22 17:41:04.778545 [DEBUG] sofia.c:407
sofia/internal/sip:1000 at 85.16.245.228:1024 receive message [DISPLAY]
send 674 bytes to udp/[85.16.245.228]:1024 at 15:41:04.779630:
   ------------------------------------------------------------------------
   INFO sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
   Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKNF4N51D6cXmNg
   Max-Forwards: 70
   From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=HKSNX8D1pKy6K
   To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=sdeidvhpxx
2009-10-22 17:41:04.779555 [DEBUG] switch_ivr_originate.c:2368 Originate
Resulted in Success: [sofia/internal/sip:1000 at 85.16.245.228:1024]
   Call-ID: 251c312d-39c4-122d-3fbf-00144fe6907a
   CSeq: 121994696 INFO
   Contact: <sip:mod_sofia at 85.16.246.12:5061>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15180M
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Content-Type: message/sipfrag
   Content-Length: 27
   2009-10-22 17:41:04.779555 [DEBUG] switch_channel.c:182
sofia/internal/sip:1000 at 85.16.245.228:1024 receive message [AUDIO_SYNC]

   From:
   To: "unknown" 1000
   ------------------------------------------------------------------------
2009-10-22 17:41:04.779555 [DEBUG] switch_channel.c:182
sofia/internal/1001 at 85.16.246.12:5061 receive message [AUDIO_SYNC]
2009-10-22 17:41:04.779555 [DEBUG] switch_ivr_bridge.c:975
sofia/internal/sip:1000 at 85.16.245.228:1024 receive message [BRIDGE]
2009-10-22 17:41:04.779555 [DEBUG] switch_core_session.c:647 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-22 17:41:04.779555 [DEBUG] switch_ivr_bridge.c:982
sofia/internal/1001 at 85.16.246.12:5061 receive message [BRIDGE]
2009-10-22 17:41:04.779555 [DEBUG] switch_core_session.c:647 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-22 17:41:04.779555 [DEBUG] switch_ivr_bridge.c:1026
(sofia/internal/sip:1000 at 85.16.245.228:1024) State Change
CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2009-10-22 17:41:04.779555 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-22 17:41:04.779555 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/sip:1000 at 85.16.245.228:1024) Running State Change
CS_EXCHANGE_MEDIA
2009-10-22 17:41:04.780562 [DEBUG] switch_core_state_machine.c:343
(sofia/internal/sip:1000 at 85.16.245.228:1024) State EXCHANGE_MEDIA
2009-10-22 17:41:04.780562 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK
recv 405 bytes from udp/[85.16.245.206]:1024 at 15:41:04.819062:
   ------------------------------------------------------------------------
   ACK sip:1000 at 85.16.246.12:5061;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 85.16.245.206:1024;branch=z9hG4bK-ltlzbi82kr2t;rport
   From: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=7xpim4o1go
   To: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=ga0vUDXXSa8Kr
   Call-ID: 3c31304b7a80-no9xsnjj0bol
   CSeq: 1 ACK
   Max-Forwards: 70
   Contact: <sip:1001 at 85.16.245.206:1024;line=eg3wp69a>;reg-id=1
   Content-Length: 0

   ------------------------------------------------------------------------
2009-10-22 17:41:04.818545 [DEBUG] sofia.c:3493 Channel
sofia/internal/1001 at 85.16.246.12:5061 entering state [ready][200]
2009-10-22 17:41:04.830545 [DEBUG] switch_ivr_bridge.c:122
sofia/internal/sip:1000 at 85.16.245.228:1024 receive message [DISPLAY]
2009-10-22 17:41:04.840545 [DEBUG] switch_ivr_bridge.c:122
sofia/internal/1001 at 85.16.246.12:5061 receive message [DISPLAY]
send 670 bytes to udp/[85.16.245.206]:1024 at 15:41:04.840803:
   ------------------------------------------------------------------------
   INFO sip:1001 at 85.16.245.206:1024;line=eg3wp69a SIP/2.0
   Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKprXe7vy995a8B
   Max-Forwards: 70
   From: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=ga0vUDXXSa8Kr
   To: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=7xpim4o1go
   Call-ID: 3c31304b7a80-no9xsnjj0bol
   CSeq: 121994696 INFO
   Contact: <sip:1000 at 85.16.246.12:5061;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15180M
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Content-Type: message/sipfrag
   Content-Length: 24

   From:
   To: "hubu" 1000
   ------------------------------------------------------------------------
2009-10-22 17:41:04.860546 [DEBUG] switch_core_io.c:652
sofia/internal/sip:1000 at 85.16.245.228:1024 receive message
[TRANSCODING_NECESSARY]
2009-10-22 17:41:04.860546 [DEBUG] switch_core_io.c:891 Engaging Write
Buffer at 320 bytes to accommodate 640->320
2009-10-22 17:41:04.930547 [DEBUG] switch_core_io.c:652
sofia/internal/1001 at 85.16.246.12:5061 receive message
[TRANSCODING_NECESSARY]
2009-10-22 17:41:04.930547 [DEBUG] switch_core_io.c:891 Engaging Write
Buffer at 320 bytes to accommodate 320->320
2009-10-22 17:41:05.000550 [WARNING] mod_sofia.c:808 We were told to use
ptime 10 but what they meant to say was 20
This issue has so far been identified to happen on the following broken
platforms/devices:
Linksys/Sipura aka Cisco
ShoreTel
Sonus/L3
We will try to fix it but some of the devices on this list are so broken
who knows what will happen..
2009-10-22 17:41:05.000550 [DEBUG] sofia_glue.c:2101 Set Codec
sofia/internal/sip:1000 at 85.16.245.228:1024 G722/8000 20 ms 160 samples
2009-10-22 17:41:05.000550 [DEBUG] switch_rtp.c:1064 RE-Starting timer
[soft] 160 bytes per 20000ms
recv 362 bytes from udp/[85.16.245.206]:1024 at 15:41:05.019892:
   ------------------------------------------------------------------------
   SIP/2.0 200 Ok
   Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKprXe7vy995a8B
   From: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=ga0vUDXXSa8Kr
   To: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=7xpim4o1go
   Call-ID: 3c31304b7a80-no9xsnjj0bol
   CSeq: 121994696 INFO
   Contact: <sip:1001 at 85.16.245.206:1024;line=eg3wp69a>;reg-id=1
   Content-Length: 0

   ------------------------------------------------------------------------
recv 372 bytes from udp/[85.16.245.228]:1024 at 15:41:05.177982:
   ------------------------------------------------------------------------
   SIP/2.0 200 Ok
   Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKNF4N51D6cXmNg
   From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=HKSNX8D1pKy6K
   To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=sdeidvhpxx
   Call-ID: 251c312d-39c4-122d-3fbf-00144fe6907a
   CSeq: 121994696 INFO
   Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
   Content-Length: 0

   ------------------------------------------------------------------------
send 679 bytes to udp/[85.16.245.228]:1024 at 15:41:05.178268:
   ------------------------------------------------------------------------
   INFO sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
   Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKQ1p78QFD7e1tQ
   Max-Forwards: 70
   From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=HKSNX8D1pKy6K
   To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=sdeidvhpxx
   Call-ID: 251c312d-39c4-122d-3fbf-00144fe6907a
   CSeq: 121994697 INFO
   Contact: <sip:mod_sofia at 85.16.246.12:5061>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15180M
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Content-Type: message/sipfrag
   Content-Length: 32

   From:
   To: "1001 an PBX1" 1001
   ------------------------------------------------------------------------
recv 372 bytes from udp/[85.16.245.228]:1024 at 15:41:05.271275:
   ------------------------------------------------------------------------
   SIP/2.0 200 Ok
   Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKQ1p78QFD7e1tQ
   From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=HKSNX8D1pKy6K
   To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=sdeidvhpxx
   Call-ID: 251c312d-39c4-122d-3fbf-00144fe6907a
   CSeq: 121994697 INFO
   Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
   Content-Length: 0

   ------------------------------------------------------------------------
recv 589 bytes from udp/[85.16.245.228]:1024 at 15:41:06.336858:
   ------------------------------------------------------------------------
   BYE sip:mod_sofia at 85.16.246.12:5061 SIP/2.0
   Via: SIP/2.0/UDP 85.16.245.228:1024;branch=z9hG4bK-quadh3ddon5i;rport
   From: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=sdeidvhpxx
   To: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=HKSNX8D1pKy6K
   Call-ID: 251c312d-39c4-122d-3fbf-00144fe6907a
   CSeq: 1 BYE
   Max-Forwards: 70
   Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
   User-Agent: snom820/LAB_HD_HEAD_11_09_09_11_20
   RTP-RxStat:
Total_Rx_Pkts=0,Rx_Pkts=66,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0
   RTP-TxStat: Total_Tx_Pkts=67,Tx_Pkts=62,Remote_Tx_Pkts=0
   Content-Length: 0

   ------------------------------------------------------------------------
2009-10-22 17:41:06.336568 [NOTICE] sofia.c:328 Hangup
sofia/internal/sip:1000 at 85.16.245.228:1024 [CS_EXCHANGE_MEDIA]
[NORMAL_CLEARING]
2009-10-22 17:41:06.336568 [DEBUG] switch_channel.c:1896 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [KILL]
2009-10-22 17:41:06.336568 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-22 17:41:06.336568 [DEBUG] switch_core_state_machine.c:449
thread mismatch skipping state handler.
send 510 bytes to udp/[85.16.245.228]:1024 at 15:41:06.337536:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP
85.16.245.228:1024;branch=z9hG4bK-quadh3ddon5i;rport=1024
   From: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=sdeidvhpxx
   To: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=HKSNX8D1pKy6K
   Call-ID: 251c312d-39c4-122d-3fbf-00144fe6907a
   CSeq: 1 BYE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15180M
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Content-Length: 0

   ------------------------------------------------------------------------
2009-10-22 17:41:06.340568 [DEBUG] switch_ivr_bridge.c:453
sofia/internal/sip:1000 at 85.16.245.228:1024 ending bridge by request from
read function
2009-10-22 17:41:06.340568 [DEBUG] switch_ivr_bridge.c:447
sofia/internal/sip:1000 at 85.16.245.228:1024 ending bridge by request from
write function
2009-10-22 17:41:06.340568 [DEBUG] switch_ivr_bridge.c:528 BRIDGE THREAD
DONE [sofia/internal/sip:1000 at 85.16.245.228:1024]
2009-10-22 17:41:06.340568 [DEBUG] switch_ivr_bridge.c:530 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-22 17:41:06.340568 [DEBUG] switch_core_state_machine.c:343
(sofia/internal/sip:1000 at 85.16.245.228:1024) State EXCHANGE_MEDIA going
to sleep
2009-10-22 17:41:06.340568 [DEBUG] switch_ivr_bridge.c:502
sofia/internal/1001 at 85.16.246.12:5061 receive message [UNBRIDGE]
2009-10-22 17:41:06.340568 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/sip:1000 at 85.16.245.228:1024) Running State Change CS_HANGUP
2009-10-22 17:41:06.340568 [DEBUG] switch_core_session.c:647 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-22 17:41:06.340568 [DEBUG] switch_ivr_bridge.c:528 BRIDGE THREAD
DONE [sofia/internal/1001 at 85.16.246.12:5061]
2009-10-22 17:41:06.340568 [DEBUG] switch_ivr_bridge.c:530 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-22 17:41:06.340568 [NOTICE] switch_ivr_bridge.c:1102 Hangup
sofia/internal/1001 at 85.16.246.12:5061 [CS_EXECUTE] [NORMAL_CLEARING]
2009-10-22 17:41:06.340568 [DEBUG] switch_channel.c:1896 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [KILL]
2009-10-22 17:41:06.340568 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-22 17:41:06.340568 [DEBUG] switch_core_state_machine.c:476
(sofia/internal/sip:1000 at 85.16.245.228:1024) State HANGUP
2009-10-22 17:41:06.340568 [DEBUG] mod_sofia.c:344 Channel
sofia/internal/sip:1000 at 85.16.245.228:1024 hanging up, cause:
NORMAL_CLEARING
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:46
sofia/internal/sip:1000 at 85.16.245.228:1024 Standard HANGUP, cause:
NORMAL_CLEARING
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:476
(sofia/internal/sip:1000 at 85.16.245.228:1024) State HANGUP going to sleep
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/sip:1000 at 85.16.245.228:1024) State Change CS_HANGUP ->
CS_REPORTING
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:476
(sofia/internal/1001 at 85.16.246.12:5061) State HANGUP
2009-10-22 17:41:06.341609 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-22 17:41:06.341609 [DEBUG] mod_sofia.c:344 Channel
sofia/internal/1001 at 85.16.246.12:5061 hanging up, cause: NORMAL_CLEARING
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/sip:1000 at 85.16.245.228:1024) Running State Change
CS_REPORTING
2009-10-22 17:41:06.341609 [DEBUG] mod_sofia.c:382 Sending BYE to
sofia/internal/1001 at 85.16.246.12:5061
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:567
(sofia/internal/sip:1000 at 85.16.245.228:1024) State REPORTING
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:53
sofia/internal/sip:1000 at 85.16.245.228:1024 Standard REPORTING, cause:
NORMAL_CLEARING
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:567
(sofia/internal/sip:1000 at 85.16.245.228:1024) State REPORTING going to sleep
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:319
(sofia/internal/sip:1000 at 85.16.245.228:1024) State Change CS_REPORTING
- -> CS_DESTROY
2009-10-22 17:41:06.341609 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/sip:1000 at 85.16.245.228:1024 [BREAK]
2009-10-22 17:41:06.341609 [DEBUG] switch_core_session.c:1114 Session 42
(sofia/internal/sip:1000 at 85.16.245.228:1024) Locked, Waiting on external
entities
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:46
sofia/internal/1001 at 85.16.246.12:5061 Standard HANGUP, cause:
NORMAL_CLEARING
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:476
(sofia/internal/1001 at 85.16.246.12:5061) State HANGUP going to sleep
2009-10-22 17:41:06.341609 [NOTICE] switch_core_session.c:1132 Session
42 (sofia/internal/sip:1000 at 85.16.245.228:1024) Ended
2009-10-22 17:41:06.341609 [NOTICE] switch_core_session.c:1134 Close
Channel sofia/internal/sip:1000 at 85.16.245.228:1024 [CS_DESTROY]
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:413
(sofia/internal/sip:1000 at 85.16.245.228:1024) Running State Change CS_DESTROY
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:424
(sofia/internal/sip:1000 at 85.16.245.228:1024) State DESTROY
2009-10-22 17:41:06.341609 [DEBUG] mod_sofia.c:261
sofia/internal/sip:1000 at 85.16.245.228:1024 SOFIA DESTROY
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:340
(sofia/internal/1001 at 85.16.246.12:5061) State EXECUTE going to sleep
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_HANGUP
2009-10-22 17:41:06.342645 [DEBUG] switch_core_state_machine.c:454
handler already called, skipping state handler.
2009-10-22 17:41:06.341609 [DEBUG] switch_core_state_machine.c:60
sofia/internal/sip:1000 at 85.16.245.228:1024 Standard DESTROY
2009-10-22 17:41:06.342645 [DEBUG] switch_core_state_machine.c:424
(sofia/internal/sip:1000 at 85.16.245.228:1024) State DESTROY going to sleep
send 659 bytes to udp/[85.16.245.206]:1024 at 15:41:06.342919:
   ------------------------------------------------------------------------
   BYE sip:1001 at 85.16.245.206:1024;line=eg3wp69a SIP/2.02009-10-22
17:41:06.342645 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/1001 at 85.16.246.12:5061) State Change CS_HANGUP ->
CS_REPORTING
2009-10-22 17:41:06.342645 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-22 17:41:06.342645 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_REPORTING
2009-10-22 17:41:06.342645 [DEBUG] switch_core_state_machine.c:567
(sofia/internal/1001 at 85.16.246.12:5061) State REPORTING

   Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKrag0aK0g4QQDK
   Max-Forwards: 70
   From: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=ga0vUDXXSa8Kr
   To: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=7xpim4o1go
   Call-ID: 3c31304b7a80-no9xsnjj0bol
   CSeq: 121994697 BYE
   Contact: <sip:1000 at 85.16.246.12:5061;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15180M
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Reason: Q.850;cause=16;text="NORMAL_CLEARING"
   Content-Length: 0

   ------------------------------------------------------------------------
2009-10-22 17:41:06.343570 [DEBUG] switch_core_state_machine.c:53
sofia/internal/1001 at 85.16.246.12:5061 Standard REPORTING, cause:
NORMAL_CLEARING
2009-10-22 17:41:06.343570 [DEBUG] switch_core_state_machine.c:567
(sofia/internal/1001 at 85.16.246.12:5061) State REPORTING going to sleep
2009-10-22 17:41:06.343570 [DEBUG] switch_core_state_machine.c:319
(sofia/internal/1001 at 85.16.246.12:5061) State Change CS_REPORTING ->
CS_DESTROY
2009-10-22 17:41:06.343570 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/1001 at 85.16.246.12:5061 [BREAK]
2009-10-22 17:41:06.343570 [DEBUG] switch_core_session.c:1114 Session 41
(sofia/internal/1001 at 85.16.246.12:5061) Locked, Waiting on external entities
2009-10-22 17:41:06.343570 [NOTICE] switch_core_session.c:1132 Session
41 (sofia/internal/1001 at 85.16.246.12:5061) Ended
2009-10-22 17:41:06.343570 [NOTICE] switch_core_session.c:1134 Close
Channel sofia/internal/1001 at 85.16.246.12:5061 [CS_DESTROY]
2009-10-22 17:41:06.343570 [DEBUG] switch_core_state_machine.c:413
(sofia/internal/1001 at 85.16.246.12:5061) Running State Change CS_DESTROY
2009-10-22 17:41:06.343570 [DEBUG] switch_core_state_machine.c:424
(sofia/internal/1001 at 85.16.246.12:5061) State DESTROY
2009-10-22 17:41:06.343570 [DEBUG] mod_sofia.c:261
sofia/internal/1001 at 85.16.246.12:5061 SOFIA DESTROY
2009-10-22 17:41:06.343570 [DEBUG] switch_core_state_machine.c:60
sofia/internal/1001 at 85.16.246.12:5061 Standard DESTROY
2009-10-22 17:41:06.343570 [DEBUG] switch_core_state_machine.c:424
(sofia/internal/1001 at 85.16.246.12:5061) State DESTROY going to sleep
recv 524 bytes from udp/[85.16.245.206]:1024 at 15:41:06.357848:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKrag0aK0g4QQDK
   From: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=ga0vUDXXSa8Kr
   To: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=7xpim4o1go
   Call-ID: 3c31304b7a80-no9xsnjj0bol
   CSeq: 121994697 BYE
   Contact: <sip:1001 at 85.16.245.206:1024;line=eg3wp69a>;reg-id=1
   User-Agent: snom820/8.2.16
   RTP-RxStat:
Total_Rx_Pkts=61,Rx_Pkts=61,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0
   RTP-TxStat: Total_Tx_Pkts=0,Tx_Pkts=90,Remote_Tx_Pkts=0
   Content-Length: 0

   ------------------------------------------------------------------------









On 21.10.2009 19:48, Michael Collins wrote:
> 
> Okay, you are able to reproduce that "unknown" thing? Can you pastebin a
> fresh debug log w/ SIP trace on, plus and relevant dp changes from the
> default dialplan?
> Thanks,
> MC
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iD8DBQFK4H3z4tZeNddg3dwRAtR3AJ4xUr8Tp4ctmhQn2DIQSvrBSv59QQCeLSsH
5inLJYyajrsYT2+kXcgd22w=
=5B1x
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From tculjaga at gmail.com  Thu Oct 22 08:58:24 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 22 Oct 2009 17:58:24 +0200
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
In-Reply-To: <2d9149cd0910220844v2aa73502j4b88fdbd75a0d833@mail.gmail.com>
References: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>
	<191c3a030910201434q4c6e65e1h58c5f0a82addc2d3@mail.gmail.com>
	<2d9149cd0910202050i36a7ea82nc7253d87e38f990a@mail.gmail.com>
	<191c3a030910211154gc8412b5gd999fbedffa02b53@mail.gmail.com>
	<2d9149cd0910220844v2aa73502j4b88fdbd75a0d833@mail.gmail.com>
Message-ID: <65d96fc80910220858u788c6146sced38e19418e681a@mail.gmail.com>

On Thu, Oct 22, 2009 at 5:44 PM, Kristian Kielhofner <
kristian.kielhofner at gmail.com> wrote:

> An update for Tony, Brian, Mike, and everyone on the list...
>
> I was able to get some phone time with the team yesterday.  Tony
> worked on my machine, found the issue, and had it committed within 30
> minutes.
>
> I've been testing T.38 all morning between the fax machines in the
> office with few issues.
>
>
and what these few issues are? :P



> THANKS AGAIN GUYS!
>
>
-------------- next part --------------
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From brian at freeswitch.org  Thu Oct 22 09:23:27 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 22 Oct 2009 11:23:27 -0500
Subject: [Freeswitch-users] Domain Question
In-Reply-To: <7f783b530910220842q5c01ddbcj28a6157c30badd52@mail.gmail.com>
References: <7f783b530910220842q5c01ddbcj28a6157c30badd52@mail.gmail.com>
Message-ID: <57CAD19F-F048-4491-83AB-A0A07D9A989E@freeswitch.org>

I"m going to guess its because mydomain.localhost doesn't resolve  
outside the machine itself so the softphone never ends up knowing wtf  
to do.

/b

On Oct 22, 2009, at 10:42 AM, freeswitch noob wrote:

> I have a noobish question about setting up FS.
>
> I have it installed and running.
>
> I setup a soft client on the machine fs is on and point it to the ip  
> address of the FS instance and it registers with no issues.
>
> I then setup an entry in my etc/hosts files "mydomain.localhost" and  
> changed the domain in the soft client.
>
> The registration now fails. with the soft phone giving a 503 error  
> "service unavailable"
>
> And I am watching the cli and I see no errors come through with the  
> domain but I see it register with the IP :(
>
> I change it back to the ip address and it works.
>
> Any direction you guys could point me in would be great.




From brian at freeswitch.org  Thu Oct 22 09:25:22 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 22 Oct 2009 11:25:22 -0500
Subject: [Freeswitch-users] Qustion about INFO messages after
	Connect/Answer
In-Reply-To: <4AE07DF4.5040803@ewetel.de>
References: <4AD859A4.5000101@ewetel.de>	<191c3a030910160918n1cb1fe1ao578e63b1b2ac076e@mail.gmail.com>	<4ADC7C09.6070503@ewetel.de>	<191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>	<4ADC85C0.7060908@ewetel.de>
	<4ADC8B6E.7070404@ewetel.de>	<191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>	<87f2f3b90910191435s7528e256r85df5f12fc634f24@mail.gmail.com>	<4ADF483B.4050807@ewetel.de>
	<87f2f3b90910211048s208daf1eyfae501091ab75530@mail.gmail.com>
	<4AE07DF4.5040803@ewetel.de>
Message-ID: <085B77EE-DA3F-4ADD-86DC-E59D1B6553EA@freeswitch.org>

I can't get what exactly you re talking about. Can you clarify ? Also  
please include the packets of interest only not the full trace if its  
not relevant to the bug.

/b

On Oct 22, 2009, at 10:44 AM, Helmut Kuper wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi Mike,
>
> here it is:
>
>
> Dialplan:
>
>
>    <extension name="Local_Extension">
>      <condition field="destination_number" expression="^(10[01][0-9]) 
> $">
>        <action application="set" data="dialed_extension=$1"/>
>        <action application="export" data="dialed_extension=$1"/>
>        <action application="set" data="transfer_ringback=$$ 
> {hold_music}"/>
>        <action application="set" data="hangup_after_bridge=true"/>
>        <action application="bridge"
> data="user/${dialed_extension}@${domain_name}"/>
>      </condition>
>    </extension>
>




From pippyduck1127 at hotmail.com  Thu Oct 22 09:30:35 2009
From: pippyduck1127 at hotmail.com (Pajongjit Buntaokit)
Date: Thu, 22 Oct 2009 16:30:35 +0000
Subject: [Freeswitch-users] Address Rupa: Database for Audio Data
In-Reply-To: <b39273530910130857jb7dec2eidd71e950d3902b25@mail.gmail.com>
References: <BAY119-W52BD912544FAA1DE624B88B3C70@phx.gbl>
Message-ID: <BAY119-W4307982F0031E8BE965E54B3BE0@phx.gbl>


Hi Rupa,

Thanks again for your advice.

I have been searching for the method to record in the freeswitch documentation but I'm still not sure which command and method to make the record for every call automatically.

Which command or method do you use? 
And to make the recording start and stop automatically every time when the calls is started and end,  where should I insert this command? 

Did you use the Mod commands 'uuid_record'?
If so, where to place this commands? 
Please show me some clues?

Thank you very much!

PB
 



> Date: Tue, 13 Oct 2009 09:57:05 -0600
> From: rupa at rupa.comajongjit Buntaokit
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Database for Audio Data
> 
> What I do is record all calls and store the call with the UUID as the
> filename.  Then when the call is hung up a CDR entry is sent to my web
> server.  This CDR contains callerid and other info I might want to
> query by.  The service on the web server inserts appropriate record(s)
> into the database.  The recordings are available to the webserver.
> When one clicks on the "listen" link, the web server serves up the
> recording by UUID in the recording directory.  I have a process that
> periodically removes old recordings from that dir.  I don't purge the
> CDRs, though that is certainly possible.
> 
> On Tue, Oct 13, 2009 at 7:58 AM, Pajongjit Buntaokit
> <pippyduck1127 at hotmail.com> wrote:
> > Hi,
> >
> > Does anyone know whether FreeSWITCH has a function to automatically record
> > every call as an audio file in a server
> > or forward them to be stored in a database with additional parameters such
> > as caller ID, date, starting time and ending time?
> >
> > So that these recorded audio data can be queried and retrieved with the
> > caller ID, date and time.
> >
> > Any suggestion or guidance, please advise.
> >
> > Thank you very much!
> >
> > ________________________________
> > Hotmail: Trusted email with Microsoft?s powerful SPAM protection. Sign up
> > now.
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> 
> 
> -- 
> -Rupa
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
_________________________________________________________________
Windows 7: It works the way you want. Learn more.
http://www.microsoft.com/Windows/windows-7/default.aspx?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_evergreen2:102009
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From anthony.minessale at gmail.com  Thu Oct 22 10:08:00 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 22 Oct 2009 12:08:00 -0500
Subject: [Freeswitch-users] Qustion about INFO messages after
	Connect/Answer
In-Reply-To: <085B77EE-DA3F-4ADD-86DC-E59D1B6553EA@freeswitch.org>
References: <4AD859A4.5000101@ewetel.de>
	<191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>
	<4ADC85C0.7060908@ewetel.de> <4ADC8B6E.7070404@ewetel.de>
	<191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>
	<87f2f3b90910191435s7528e256r85df5f12fc634f24@mail.gmail.com>
	<4ADF483B.4050807@ewetel.de>
	<87f2f3b90910211048s208daf1eyfae501091ab75530@mail.gmail.com>
	<4AE07DF4.5040803@ewetel.de>
	<085B77EE-DA3F-4ADD-86DC-E59D1B6553EA@freeswitch.org>
Message-ID: <191c3a030910221008n60341314hc7a7e6a55a50ef15@mail.gmail.com>

I'm sure that problem is gone in svn trunk.


On Thu, Oct 22, 2009 at 11:25 AM, Brian West <brian at freeswitch.org> wrote:

> I can't get what exactly you re talking about. Can you clarify ? Also
> please include the packets of interest only not the full trace if its
> not relevant to the bug.
>
> /b
>
> On Oct 22, 2009, at 10:44 AM, Helmut Kuper wrote:
>
> > -----BEGIN PGP SIGNED MESSAGE-----
> > Hash: SHA1
> >
> > Hi Mike,
> >
> > here it is:
> >
> >
> > Dialplan:
> >
> >
> >    <extension name="Local_Extension">
> >      <condition field="destination_number" expression="^(10[01][0-9])
> > $">
> >        <action application="set" data="dialed_extension=$1"/>
> >        <action application="export" data="dialed_extension=$1"/>
> >        <action application="set" data="transfer_ringback=$$
> > {hold_music}"/>
> >        <action application="set" data="hangup_after_bridge=true"/>
> >        <action application="bridge"
> > data="user/${dialed_extension}@${domain_name}"/>
> >      </condition>
> >    </extension>
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Thu Oct 22 10:12:53 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 22 Oct 2009 12:12:53 -0500
Subject: [Freeswitch-users] UUID of the newly originated call?
In-Reply-To: <26006565.post@talk.nabble.com>
References: <fe67a3340910202306i55a0b9ceve92ab19e000edf11@mail.gmail.com>
	<191c3a030910211720l26bd0303i2cb11e9adb28037d@mail.gmail.com>
	<26006565.post@talk.nabble.com>
Message-ID: <191c3a030910221012w125a15d4m24f5fbe72c8a859d@mail.gmail.com>

please update to svn trunk with "make current" and try again.


On Thu, Oct 22, 2009 at 4:08 AM, Nagalenoj <nagalenoj at gmail.com> wrote:

>
> I'm using 'FreeSWITCH Version 1.0.trunk (15106)'.
>
>
> Anthony Minessale-2 wrote:
> >
> > which revision of FS are you using?
> >
> > On Wed, Oct 21, 2009 at 1:06 AM, Nagalenoj H. <nagalenoj at gmail.com>
> wrote:
> >
> >> I've tried with origination_uuid.
> >>
> >> First, I tried with SIP and my program executes successfully as what I
> >> expected. This program initiates a new call when a call comes and let
> the
> >> new call to eavesdrop the landed call.
> >> When, I experimented with PRI(openzap), I'm facing the following error.
> >> And, I am unable to make any calls(even from CLI). It is reporting the
> >> same
> >> error for the subsequent calls.
> >> Error:        2009-10-21 11:08:31.661252 [ERR] mod_openzap.c:1760
> Session
> >> for channel 1:1 not found [UUID: 8364eb30-be03-11de-aa07-07e76b7651a9]
> >>
> >> Full log is in http://pastebin.freeswitch.org/10780
> >> My script is here, http://pastebin.freeswitch.org/10781
> >>
> >> What is this error for and how to avoid this?
> >>
> >> Is there any other way to get the uuid of the originated call except
> >> explicitly defining(origination_uuid).?!
> >>
> >> --
> >> Regards,
> >> Nagalenoj H.
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com><
> MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org><
> sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> > pstn:213-799-1400
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> --
> View this message in context:
> http://www.nabble.com/Re%3A-UUID-of-the-newly-originated-call--tp25987024p26006565.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mattdfong at gmail.com  Thu Oct 22 10:42:29 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Fri, 23 Oct 2009 00:42:29 +0700
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
In-Reply-To: <87f2f3b90910191144q25750bayfd7a7448e82e78ff@mail.gmail.com>
References: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
	<177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>
	<4256bf830908041151l55980e3fifb85c75b87535426@mail.gmail.com>
	<3D0B855A-5B2C-4C6F-9306-64B0ECE97091@jerris.com>
	<4256bf830910180725y72539d98gb138934afc18d58b@mail.gmail.com>
	<A5ECDCD9-1A12-4A31-89F0-E6F4C79FE9E9@jerris.com>
	<4256bf830910182148mf7f2bf9vdcccd687e0caa1e7@mail.gmail.com>
	<87f2f3b90910191144q25750bayfd7a7448e82e78ff@mail.gmail.com>
Message-ID: <4256bf830910221042l68ea769cg532bbd339bae0790@mail.gmail.com>

I figured out how to turn debug mode on. For some reason F8 didn't work, so
I just set the default debug mode to DEBUG in switch.conf.xml

http://pastebin.freeswitch.org/10805
<http://pastebin.freeswitch.org/10805>So my program does more or less the
same sequence for an Agent every time when connected to another party. They
uuid_kill the bridged parties uuid, and then are transfered to an extension
that park's the agent (using the transfer_after_bridge variable -- in my
dialplan extn 1998). In some instances the agent is then transfered to
another extension, played a wav file, and then park'ed again

Sometimes, FreeSWITCH unexpectidly hangsup the Agent channel citing an
EXCHANGE_ROUTING_ERROR. It seems (to me to happen at random times)...and
it's more prevelent with some did providers than others. With
didforsale.orgit happens about one ever 5 bridges. With
icall.com it only happens about once every 30 bridges. With IPKall it
happens rarely. I can't figure it out, and it's driving me crazy :)


A GOOD UUID_KILL SEQUENCE:
2009-10-22 17:22:31.741845 [DEBUG] switch_ivr_play_say.c:1432 done playing
file
2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:340
(sofia/external/+14158867717 at 199.173.100.16:5060) State EXECUTE going to
sleep
2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:306
(sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
CS_HIBERNATE
2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:355
(sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE
2009-10-22 17:22:31.741845 [DEBUG] mod_sofia.c:160 sofia/external/+
14158867717 at 199.173.100.16:5060 SOFIA HIBERNATE
2009-10-22 17:22:31.741845 [DEBUG] switch_ivr_bridge.c:650 (sofia/external/+
14158867717 at 199.173.100.16:5060) State Change CS_HIBERNATE -> CS_RESET
2009-10-22 17:22:31.741845 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:355
(sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE going to
sleep
2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:306
(sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
CS_RESET
2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:336
(sofia/external/+14158867717 at 199.173.100.16:5060) State RESET
2009-10-22 17:22:31.741845 [DEBUG] mod_sofia.c:145 sofia/external/+
14158867717 at 199.173.100.16:5060 SOFIA RESET
2009-10-22 17:22:31.741845 [DEBUG] switch_ivr_bridge.c:637 sofia/external/+
14158867717 at 199.173.100.16:5060 CUSTOM RESET
2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:67
sofia/external/+14158867717 at 199.173.100.16:5060 Standard RESET
2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:336
(sofia/external/+14158867717 at 199.173.100.16:5060) State RESET going to sleep
2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:340
(sofia/external/14159927717) State EXECUTE going to sleep
2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:306
(sofia/external/14159927717) Running State Change CS_HIBERNATE
2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:355
(sofia/external/14159927717) State HIBERNATE
2009-10-22 17:22:31.748086 [DEBUG] mod_sofia.c:160
sofia/external/14159927717 SOFIA HIBERNATE
2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:650
(sofia/external/14159927717) State Change CS_HIBERNATE -> CS_RESET
2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/14159927717 [BREAK]
2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:355
(sofia/external/14159927717) State HIBERNATE going to sleep
2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:306
(sofia/external/14159927717) Running State Change CS_RESET
2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:336
(sofia/external/14159927717) State RESET
2009-10-22 17:22:31.748086 [DEBUG] mod_sofia.c:145
sofia/external/14159927717 SOFIA RESET
2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:637
sofia/external/14159927717 CUSTOM RESET
2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:642
(sofia/external/14159927717) State Change CS_RESET -> CS_SOFT_EXECUTE
2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/14159927717 [BREAK]
2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:336
(sofia/external/14159927717) State RESET going to sleep
2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:306
(sofia/external/14159927717) Running State Change CS_SOFT_EXECUTE
2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:346
(sofia/external/14159927717) State SOFT_EXECUTE
2009-10-22 17:22:31.748086 [DEBUG] mod_sofia.c:442 SOFIA TRANSMIT
2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:660
sofia/external/14159927717 CUSTOM SOFT_EXECUTE
2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_originate.c:674
(sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_RESET ->
CS_SOFT_EXECUTE
2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:120 sofia/external/+
14158867717 at 199.173.100.16:5060 receive message [DISPLAY]
2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:120
sofia/external/14159927717 receive message [DISPLAY]
2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:924 (sofia/external/+
14158867717 at 199.173.100.16:5060) State Change CS_SOFT_EXECUTE ->
CS_CONSUME_MEDIA
2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:968 sofia/external/+
14158867717 at 199.173.100.16:5060 receive message [BRIDGE]
2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:630 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:975
sofia/external/14159927717 receive message [BRIDGE]
2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:630 Send signal
sofia/external/14159927717 [BREAK]
2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:1019
(sofia/external/+14158867717 at 199.173.100.16:5060) State Change
CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:22:31.749553 [DEBUG] switch_core_state_machine.c:306
(sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
CS_EXCHANGE_MEDIA
2009-10-22 17:22:31.749553 [DEBUG] switch_core_state_machine.c:343
(sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
2009-10-22 17:22:31.749553 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK
multipler is being overrided to1
2009-10-22 17:22:34.300662 [INFO] switch_cpp.cpp:1116 BG_AG_GEN_CALLS,
Agents total:1 avail:0 OutChan:0 AR:0 +-Calls:0:Q Call NYNoc
multipler is being overrided to1
2009-10-22 17:22:39.303041 [INFO] switch_cpp.cpp:1116 BG_AG_GEN_CALLS,
Agents total:1 avail:0 OutChan:0 AR:0 +-Calls:0:Q Call NYNoc
2009-10-22 17:22:40.626379 [NOTICE] sofia.c:328 Hangup
sofia/external/14159927717 [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
2009-10-22 17:22:40.626379 [DEBUG] switch_channel.c:1810 Send signal
sofia/external/14159927717 [KILL]
2009-10-22 17:22:40.626379 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/14159927717 [BREAK]
active_true=1
--contact NOT in playmsg or conference, update AGENT
2009-10-22 17:22:40.626379 [DEBUG] switch_core_state_machine.c:437 thread
mismatch skipping state handler.
2009-10-22 17:22:40.626379 [INFO] switch_cpp.cpp:1116 EVENTC_HANGUP_CONTACT,
Q_CONTACT fired for agent:1/1 for qcall:1 contact:4801442/4
2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:440
sofia/external/14159927717 ending bridge by request from write function
2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:495 sofia/external/+
14158867717 at 199.173.100.16:5060 receive message [UNBRIDGE]
2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:630 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
DONE [sofia/external/+14158867717 at 199.173.100.16:5060]
2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:523 Send signal
sofia/external/14159927717 [BREAK]
2009-10-22 17:22:40.639760 [DEBUG] switch_ivr.c:1361 (sofia/external/+
14158867717 at 199.173.100.16:5060) State Change CS_EXCHANGE_MEDIA ->
CS_ROUTING
2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:22:40.639760 [DEBUG] switch_ivr.c:1365 sofia/external/+
14158867717 at 199.173.100.16:5060 receive message [TRANSFER]
2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:630 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:22:40.639760 [NOTICE] switch_ivr.c:1367 Transfer
sofia/external/+14158867717 at 199.173.100.16:5060 to xml[1998 at default]
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:343
(sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA going
to sleep
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
(sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
CS_ROUTING
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:333
(sofia/external/+14158867717 at 199.173.100.16:5060) State ROUTING
2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:130 sofia/external/+
14158867717 at 199.173.100.16:5060 SOFIA ROUTING
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:78
sofia/external/+14158867717 at 199.173.100.16:5060 Standard ROUTING
2009-10-22 17:22:40.639760 [INFO] mod_dialplan_xml.c:391 Processing
+14158867717->1998 in context default
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->tod_example] continue=true
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (PASS)
[tod_example] ${strftime(%w)}(4) =~ /^([1-5])$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (PASS)
[tod_example] ${strftime(%H%M)}(1722) =~ /^((09|1[0-7])[0-5][0-9]|1800)$/
break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Action
set(open=true)
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->ivr_demo] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[ivr_demo] destination_number(1998) =~ /^5000$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1975] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1975] destination_number(1998) =~ /^1975$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1971] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1971] destination_number(1998) =~ /^1971$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1972] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1972] destination_number(1998) =~ /^1972$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1920] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1920] destination_number(1998) =~ /^1920$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1981] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1981] destination_number(1998) =~ /^1981$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1982] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1982] destination_number(1998) =~ /^1982$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1481] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1481] destination_number(1998) =~ /^1481$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1989] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1989] destination_number(1998) =~ /^1989$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1990] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1990] destination_number(1998) =~ /^1990$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1991] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1991] destination_number(1998) =~ /^1991$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1992] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1992] destination_number(1998) =~ /^1992$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1993] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1993] destination_number(1998) =~ /^1993$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1994] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1994] destination_number(1998) =~ /^1994$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1995] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1995] destination_number(1998) =~ /^1995$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1996] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1996] destination_number(1998) =~ /^1996$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1997] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
[1997] destination_number(1998) =~ /^1997$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
[default->1998] continue=false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (PASS)
[1998] destination_number(1998) =~ /^1998$/ break=on-false
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Action
set(transfer_after_bridge=1998)
Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Action park()
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:114
(sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_ROUTING ->
CS_EXECUTE
2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:333
(sofia/external/+14158867717 at 199.173.100.16:5060) State ROUTING going to
sleep
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
(sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
CS_EXECUTE
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:340
(sofia/external/+14158867717 at 199.173.100.16:5060) State EXECUTE
2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:173 sofia/external/+
14158867717 at 199.173.100.16:5060 SOFIA EXECUTE
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:151
sofia/external/+14158867717 at 199.173.100.16:5060 Standard EXECUTE
EXECUTE sofia/external/+14158867717 at 199.173.100.16:5060 set(open=true)
2009-10-22 17:22:40.639760 [DEBUG] mod_dptools.c:752 sofia/external/+
14158867717 at 199.173.100.16:5060 SET [open]=[true]
EXECUTE sofia/external/+14158867717 at 199.173.100.16:5060set(transfer_after_bridge=1998)
2009-10-22 17:22:40.639760 [DEBUG] mod_dptools.c:752 sofia/external/+
14158867717 at 199.173.100.16:5060 SET [transfer_after_bridge]=[1998]
EXECUTE sofia/external/+14158867717 at 199.173.100.16:5060 park()
2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:446
sofia/external/14159927717 ending bridge by request from read function
2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
DONE [sofia/external/14159927717]
2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:523 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
subtracting 9 seconds for hh_call: 1095422
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:346
(sofia/external/14159927717) State SOFT_EXECUTE going to sleep
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
(sofia/external/14159927717) Running State Change CS_HANGUP
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:464
(sofia/external/14159927717) State HANGUP
2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:344 Channel
sofia/external/14159927717 hanging up, cause: NORMAL_CLEARING
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:46
sofia/external/14159927717 Standard HANGUP, cause: NORMAL_CLEARING
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:464
(sofia/external/14159927717) State HANGUP going to sleep
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:325
(sofia/external/14159927717) State Change CS_HANGUP -> CS_REPORTING
2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/14159927717 [BREAK]
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
(sofia/external/14159927717) Running State Change CS_REPORTING
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:555
(sofia/external/14159927717) State REPORTING
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:53
sofia/external/14159927717 Standard REPORTING, cause: NORMAL_CLEARING
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:555
(sofia/external/14159927717) State REPORTING going to sleep
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:319
(sofia/external/14159927717) State Change CS_REPORTING -> CS_DESTROY
2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/14159927717 [BREAK]
2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:1069 Session 6
(sofia/external/14159927717) Locked, Waiting on external entities
2009-10-22 17:22:40.639760 [NOTICE] switch_core_session.c:1087 Session 6
(sofia/external/14159927717) Ended
2009-10-22 17:22:40.639760 [NOTICE] switch_core_session.c:1089 Close Channel
sofia/external/14159927717 [CS_DESTROY]
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:401
(sofia/external/14159927717) Running State Change CS_DESTROY
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:412
(sofia/external/14159927717) State DESTROY
2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:261
sofia/external/14159927717 SOFIA DESTROY
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:60
sofia/external/14159927717 Standard DESTROY
2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:412
(sofia/external/14159927717) State DESTROY going to sleep
2009-10-22 17:22:40.639760 [INFO] switch_cpp.cpp:1116 PCHangup
gw:teliax.comhc:NORMAL_CLEARING du:9 cn:sofia/external/14159927717



A BAD UUID_KILL SEQUENCE: (where the agent is hungup, unexpectedly)

--bridged in: 0.005821 seconds
2009-10-22 17:23:26.718930 [INFO] switch_cpp.cpp:1116 AG_CBRIDGE, connected
agent:1/1 contact:4801439/22/1/1
2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_play_say.c:1432 done playing
file
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:340
(sofia/external/+14158867717 at 199.173.100.16:5060) State EXECUTE going to
sleep
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
(sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
CS_HIBERNATE
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
(sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE
2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:160 sofia/external/+
14158867717 at 199.173.100.16:5060 SOFIA HIBERNATE
2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:650 (sofia/external/+
14158867717 at 199.173.100.16:5060) State Change CS_HIBERNATE -> CS_RESET
2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
(sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE going to
sleep
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
(sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
CS_RESET
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
(sofia/external/+14158867717 at 199.173.100.16:5060) State RESET
2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:145 sofia/external/+
14158867717 at 199.173.100.16:5060 SOFIA RESET
2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:637 sofia/external/+
14158867717 at 199.173.100.16:5060 CUSTOM RESET
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:67
sofia/external/+14158867717 at 199.173.100.16:5060 Standard RESET
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
(sofia/external/+14158867717 at 199.173.100.16:5060) State RESET going to sleep
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:340
(sofia/external/14159927717) State EXECUTE going to sleep
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
(sofia/external/14159927717) Running State Change CS_HIBERNATE
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
(sofia/external/14159927717) State HIBERNATE
2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:160
sofia/external/14159927717 SOFIA HIBERNATE
2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:650
(sofia/external/14159927717) State Change CS_HIBERNATE -> CS_RESET
2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/14159927717 [BREAK]
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
(sofia/external/14159927717) State HIBERNATE going to sleep
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
(sofia/external/14159927717) Running State Change CS_RESET
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
(sofia/external/14159927717) State RESET
2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:145
sofia/external/14159927717 SOFIA RESET
2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:637
sofia/external/14159927717 CUSTOM RESET
2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:642
(sofia/external/14159927717) State Change CS_RESET -> CS_SOFT_EXECUTE
2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/14159927717 [BREAK]
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
(sofia/external/14159927717) State RESET going to sleep
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
(sofia/external/14159927717) Running State Change CS_SOFT_EXECUTE
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:346
(sofia/external/14159927717) State SOFT_EXECUTE
2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:442 SOFIA TRANSMIT
2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:660
sofia/external/14159927717 CUSTOM SOFT_EXECUTE
2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_originate.c:674
(sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_RESET ->
CS_SOFT_EXECUTE
2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:120 sofia/external/+
14158867717 at 199.173.100.16:5060 receive message [DISPLAY]
2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:120
sofia/external/14159927717 receive message [DISPLAY]
2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:924 (sofia/external/+
14158867717 at 199.173.100.16:5060) State Change CS_SOFT_EXECUTE ->
CS_CONSUME_MEDIA
2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:968 sofia/external/+
14158867717 at 199.173.100.16:5060 receive message [BRIDGE]
2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:630 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:975
sofia/external/14159927717 receive message [BRIDGE]
2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:630 Send signal
sofia/external/14159927717 [BREAK]
2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:1019
(sofia/external/+14158867717 at 199.173.100.16:5060) State Change
CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
(sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
CS_EXCHANGE_MEDIA
2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:343
(sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK
2009-10-22 17:23:26.740065 [DEBUG] switch_core_io.c:234 sofia/external/+
14158867717 at 199.173.100.16:5060 receive message [TRANSCODING_NECESSARY]
2009-10-22 17:23:27.319618 [DEBUG] switch_core_media_bug.c:393 Removing BUG
from sofia/external/+14158867717 at 199.173.100.16:5060
multipler is being overrided to1
2009-10-22 17:23:29.406414 [INFO] switch_cpp.cpp:1116 BG_AG_GEN_CALLS,
Agents total:1 avail:0 OutChan:0 AR:0 +-Calls:0:Q Call NYNoc
2009-10-22 17:23:31.873352 [NOTICE] mod_commands.c:1523 Hangup
sofia/external/14159927717 [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
2009-10-22 17:23:31.874411 [INFO] switch_cpp.cpp:1116 EVENTC_HANGUP_CONTACT,
Q_CONTACT fired for agent:1/1 for qcall:1 contact:4801439/1
active_true=1
--contact NOT in playmsg or conference, update AGENT
2009-10-22 17:23:31.875475 [DEBUG] switch_channel.c:1810 Send signal
sofia/external/14159927717 [KILL]
2009-10-22 17:23:31.875475 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/14159927717 [BREAK]
2009-10-22 17:23:31.875475 [DEBUG] switch_core_state_machine.c:437 thread
mismatch skipping state handler.
2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:446
sofia/external/14159927717 ending bridge by request from read function
2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
DONE [sofia/external/14159927717]
2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:523 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:495 sofia/external/+
14158867717 at 199.173.100.16:5060 receive message [UNBRIDGE]
2009-10-22 17:23:31.882153 [DEBUG] switch_core_session.c:630 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
DONE [sofia/external/+14158867717 at 199.173.100.16:5060]
2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:523 Send signal
sofia/external/14159927717 [BREAK]
2009-10-22 17:23:31.882153 [NOTICE] switch_ivr.c:1268 Hangup
sofia/external/+14158867717 at 199.173.100.16:5060 [CS_EXCHANGE_MEDIA]
[EXCHANGE_ROUTING_ERROR]
2009-10-22 17:23:31.882153 [DEBUG] switch_channel.c:1810 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [KILL]
2009-10-22 17:23:31.882153 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:23:31.882153 [DEBUG] switch_core_state_machine.c:464
(sofia/external/+14158867717 at 199.173.100.16:5060) State HANGUP
2009-10-22 17:23:31.882153 [DEBUG] mod_sofia.c:344 Channel sofia/external/+
14158867717 at 199.173.100.16:5060 hanging up, cause: EXCHANGE_ROUTING_ERROR
2009-10-22 17:23:31.882153 [DEBUG] mod_sofia.c:382 Sending BYE to
sofia/external/+14158867717 at 199.173.100.16:5060
2009-10-22 17:23:31.882153 [DEBUG] switch_core_state_machine.c:46
sofia/external/+14158867717 at 199.173.100.16:5060 Standard HANGUP, cause:
EXCHANGE_ROUTING_ERROR
2009-10-22 17:23:31.882153 [DEBUG] switch_core_state_machine.c:464
(sofia/external/+14158867717 at 199.173.100.16:5060) State HANGUP going to
sleep
2009-10-22 17:23:31.884357 [INFO] switch_cpp.cpp:1116 HH_HANGUP, agent:1/1
uuid: f8604694-4e44-4c4c-9805-722b3e9d6b68
2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:346
(sofia/external/14159927717) State SOFT_EXECUTE going to sleep
2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:306
(sofia/external/14159927717) Running State Change CS_HANGUP
2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:464
(sofia/external/14159927717) State HANGUP
2009-10-22 17:23:31.885412 [DEBUG] mod_sofia.c:344 Channel
sofia/external/14159927717 hanging up, cause: NORMAL_CLEARING
2009-10-22 17:23:31.885412 [DEBUG] mod_sofia.c:382 Sending BYE to
sofia/external/14159927717
2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:46
sofia/external/14159927717 Standard HANGUP, cause: NORMAL_CLEARING
2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:464
(sofia/external/14159927717) State HANGUP going to sleep
2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:325
(sofia/external/14159927717) State Change CS_HANGUP -> CS_REPORTING
2009-10-22 17:23:31.885412 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/14159927717 [BREAK]
subtracting 5 seconds for hh_call: 1095423
2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:306
(sofia/external/14159927717) Running State Change CS_REPORTING
2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:555
(sofia/external/14159927717) State REPORTING
2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:53
sofia/external/14159927717 Standard REPORTING, cause: NORMAL_CLEARING
2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:555
(sofia/external/14159927717) State REPORTING going to sleep
2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:319
(sofia/external/14159927717) State Change CS_REPORTING -> CS_DESTROY
2009-10-22 17:23:31.885412 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/14159927717 [BREAK]
2009-10-22 17:23:31.885412 [DEBUG] switch_core_session.c:1069 Session 7
(sofia/external/14159927717) Locked, Waiting on external entities
2009-10-22 17:23:31.885412 [NOTICE] switch_core_session.c:1087 Session 7
(sofia/external/14159927717) Ended
2009-10-22 17:23:31.885412 [NOTICE] switch_core_session.c:1089 Close Channel
sofia/external/14159927717 [CS_DESTROY]
2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:401
(sofia/external/14159927717) Running State Change CS_DESTROY
2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:412
(sofia/external/14159927717) State DESTROY
2009-10-22 17:23:31.885412 [DEBUG] mod_sofia.c:261
sofia/external/14159927717 SOFIA DESTROY
2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:60
sofia/external/14159927717 Standard DESTROY
2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:412
(sofia/external/14159927717) State DESTROY going to sleep
2009-10-22 17:23:31.885412 [INFO] switch_cpp.cpp:1116 PCHangup
gw:teliax.comhc:NORMAL_CLEARING du:5 cn:sofia/external/14159927717
2009-10-22 17:23:31.885412 [INFO] switch_cpp.cpp:1116 EVENTC_BILLING_CEIL,
subtracting 5 for hh_call:1095423
2009-10-22 17:23:31.901620 [DEBUG] switch_core_state_machine.c:494 Hangup
Command luarun(hh_hangup.lua f8604694-4e44-4c4c-9805-722b3e9d6b68 1 1):
+OK

2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:343
(sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA going
to sleep
2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:306
(sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
CS_HANGUP
2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:442 handler
already called, skipping state handler.
2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:325
(sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_HANGUP ->
CS_REPORTING
2009-10-22 17:23:31.902676 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:306
(sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
CS_REPORTING
2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:555
(sofia/external/+14158867717 at 199.173.100.16:5060) State REPORTING
2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:53
sofia/external/+14158867717 at 199.173.100.16:5060 Standard REPORTING, cause:
EXCHANGE_ROUTING_ERROR
2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:555
(sofia/external/+14158867717 at 199.173.100.16:5060) State REPORTING going to
sleep
2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:319
(sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_REPORTING
-> CS_DESTROY
2009-10-22 17:23:31.903742 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
2009-10-22 17:23:31.903742 [DEBUG] switch_core_session.c:1069 Session 4
(sofia/external/+14158867717 at 199.173.100.16:5060) Locked, Waiting on
external entities
2009-10-22 17:23:31.903742 [NOTICE] switch_core_session.c:1087 Session 4
(sofia/external/+14158867717 at 199.173.100.16:5060) Ended
2009-10-22 17:23:31.903742 [NOTICE] switch_core_session.c:1089 Close Channel
sofia/external/+14158867717 at 199.173.100.16:5060 [CS_DESTROY]
2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:401
(sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
CS_DESTROY
2009-10-22 17:23:31.904799 [DEBUG] switch_core_state_machine.c:412
(sofia/external/+14158867717 at 199.173.100.16:5060) State DESTROY
2009-10-22 17:23:31.904799 [DEBUG] mod_sofia.c:261 sofia/external/+
14158867717 at 199.173.100.16:5060 SOFIA DESTROY
2009-10-22 17:23:31.904799 [DEBUG] switch_core_state_machine.c:60
sofia/external/+14158867717 at 199.173.100.16:5060 Standard DESTROY
2009-10-22 17:23:31.904799 [DEBUG] switch_core_state_machine.c:412
(sofia/external/+14158867717 at 199.173.100.16:5060) State DESTROY going to
sleep



On Tue, Oct 20, 2009 at 1:44 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Sun, Oct 18, 2009 at 9:48 PM, Matthew Fong <mattdfong at gmail.com> wrote:
>
>> The debug level logs to the console, right? The pastebin, had log level
>> debug, sofia trace on for external and default, and sofia loglevel all 9. Is
>> there another log enable command I'm missing? It seems loglevel all 9
>> outputs enter and exit functions, but at least to my novice eye, it's not
>> too obvious why freeswitch is sending a BYE to my DID provider.
>>
>> Evidently the console loglevel debug did not take because there isn't a
> single DEBUG level line of output. Please try again. Just remember that
> debug level log messages will have A LOT of information so be patient while
> sifting through it. Of course, feel free to pastebin the output and add a
> link to it in this email thread.
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From saeedahmad1981 at gmail.com  Thu Oct 22 10:47:44 2009
From: saeedahmad1981 at gmail.com (Saeed Ahmad)
Date: Thu, 22 Oct 2009 19:47:44 +0200
Subject: [Freeswitch-users] Heartbeat question
In-Reply-To: <86a32abc0910091501y40dd34c4r1bca3eeb2b6fa693@mail.gmail.com>
References: <86a32abc0910081956k91b3600sb8c556e589f5b089@mail.gmail.com> 
	<86a32abc0910082003v3f5d4e6dua8e7eae18fd2f939@mail.gmail.com> 
	<6b65470d0910082036g204d5dcbw26e1e43ef80c7c84@mail.gmail.com> 
	<86a32abc0910082245j36b8b3e9ncaf1cdd358fce319@mail.gmail.com> 
	<191c3a030910090902t38282f72l6b6d1733a758836@mail.gmail.com> 
	<86a32abc0910091427i2395fd80t3bc557186ece3972@mail.gmail.com> 
	<86a32abc0910091430hc151ed8of085fcc3af2f1de4@mail.gmail.com> 
	<86a32abc0910091501y40dd34c4r1bca3eeb2b6fa693@mail.gmail.com>
Message-ID: <c96b95650910221047i545601a4m9e0a645341068d15@mail.gmail.com>

What is heartbeat and what are the uses cases?

Sorry i didn't find much information on wiki.

Thanks.



On Sat, Oct 10, 2009 at 12:01 AM, Diego Viola <diego.viola at gmail.com> wrote:

> Here's my heartbeat script now.
>
> #!/usr/bin/env ruby
>
> require 'rubygems'
> require 'fsr'
> require "fsr/listener/inbound"
>
> def custom_channel_heartbeat_handler(event)
>   puts "Got a SESSION_HEARTBEAT at #{Time.now.strftime('%H:%M:%S')}"
> end
> FSL::Inbound.add_event_hook(:SESSION_HEARTBEAT) {|event|
> custom_channel_heartbeat_handler(event) }
> FSR.start_ies!(FSL::Inbound, :host => "localhost", :port => 8021)
>
> Thanks again.
>
> Diego
>
>
> On Fri, Oct 9, 2009 at 9:30 PM, Diego Viola <diego.viola at gmail.com> wrote:
>
>> Here is on two seconds ;)
>>
>> Got a SESSION_HEARTBEAT at 17:17:13
>> Got a SESSION_HEARTBEAT at 17:17:15
>> Got a SESSION_HEARTBEAT at 17:17:17
>> Got a SESSION_HEARTBEAT at 17:17:19
>> Got a SESSION_HEARTBEAT at 17:17:21
>> Got a SESSION_HEARTBEAT at 17:17:23
>> Got a SESSION_HEARTBEAT at 17:17:25
>> Got a SESSION_HEARTBEAT at 17:17:27
>> Got a SESSION_HEARTBEAT at 17:17:29
>> Got a SESSION_HEARTBEAT at 17:17:31
>> Got a SESSION_HEARTBEAT at 17:17:33
>> Got a SESSION_HEARTBEAT at 17:17:35
>> Got a SESSION_HEARTBEAT at 17:17:37
>> Got a SESSION_HEARTBEAT at 17:17:39
>> Got a SESSION_HEARTBEAT at 17:17:41
>> Got a SESSION_HEARTBEAT at 17:17:43
>> Got a SESSION_HEARTBEAT at 17:17:45
>> Got a SESSION_HEARTBEAT at 17:17:47
>> Got a SESSION_HEARTBEAT at 17:17:49
>> Got a SESSION_HEARTBEAT at 17:17:51
>> Got a SESSION_HEARTBEAT at 17:17:53
>> Got a SESSION_HEARTBEAT at 17:17:55
>> Got a SESSION_HEARTBEAT at 17:17:57
>> Got a SESSION_HEARTBEAT at 17:17:59
>> Got a SESSION_HEARTBEAT at 17:18:01
>> Got a SESSION_HEARTBEAT at 17:18:03
>> Got a SESSION_HEARTBEAT at 17:18:05
>> Got a SESSION_HEARTBEAT at 17:18:07
>> Got a SESSION_HEARTBEAT at 17:18:09
>> Got a SESSION_HEARTBEAT at 17:18:11
>> Got a SESSION_HEARTBEAT at 17:18:13
>> Got a SESSION_HEARTBEAT at 17:18:15
>> Got a SESSION_HEARTBEAT at 17:18:17
>>
>>
>>
>> On Fri, Oct 9, 2009 at 9:27 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>
>>> Thanks Anthony, this solved it. You rock :)
>>>
>>> My program now outputs:
>>>
>>> Got a SESSION_HEARTBEAT at 17:14:59
>>> Got a SESSION_HEARTBEAT at 17:15:00
>>> Got a SESSION_HEARTBEAT at 17:15:02
>>> Got a SESSION_HEARTBEAT at 17:15:03
>>> Got a SESSION_HEARTBEAT at 17:15:04
>>> Got a SESSION_HEARTBEAT at 17:15:05
>>> Got a SESSION_HEARTBEAT at 17:15:06
>>> Got a SESSION_HEARTBEAT at 17:15:07
>>> Got a SESSION_HEARTBEAT at 17:15:08
>>> Got a SESSION_HEARTBEAT at 17:15:09
>>> Got a SESSION_HEARTBEAT at 17:15:10
>>> Got a SESSION_HEARTBEAT at 17:15:11
>>> Got a SESSION_HEARTBEAT at 17:15:12
>>> Got a SESSION_HEARTBEAT at 17:15:13
>>> Got a SESSION_HEARTBEAT at 17:15:14
>>> Got a SESSION_HEARTBEAT at 17:15:15
>>> Got a SESSION_HEARTBEAT at 17:15:16
>>> Got a SESSION_HEARTBEAT at 17:15:17
>>> Got a SESSION_HEARTBEAT at 17:15:18
>>> Got a SESSION_HEARTBEAT at 17:15:19
>>> Got a SESSION_HEARTBEAT at 17:15:20
>>> Got a SESSION_HEARTBEAT at 17:15:21
>>> Got a SESSION_HEARTBEAT at 17:15:22
>>> Got a SESSION_HEARTBEAT at 17:15:23
>>> Got a SESSION_HEARTBEAT at 17:15:24
>>> Got a SESSION_HEARTBEAT at 17:15:25
>>> Got a SESSION_HEARTBEAT at 17:15:26
>>> Got a SESSION_HEARTBEAT at 17:15:27
>>> Got a SESSION_HEARTBEAT at 17:15:28
>>> Got a SESSION_HEARTBEAT at 17:15:29
>>> Got a SESSION_HEARTBEAT at 17:15:30
>>>
>>>
>>>
>>>
>>>
>>> On Fri, Oct 9, 2009 at 4:02 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> Update to trunk and try it with fs_cli it for sure will let you do every
>>>> 1 second
>>>>
>>>> in fs_cli type
>>>>
>>>> /events plain all
>>>>
>>>> if you make that call you will see one every 1 second
>>>>
>>>>
>>>>
>>>> On Fri, Oct 9, 2009 at 12:45 AM, Diego Viola <diego.viola at gmail.com>wrote:
>>>>
>>>>> Nope, I was just wondering why it didn't work at 1 second exactly...
>>>>>
>>>>>
>>>>> On Fri, Oct 9, 2009 at 3:36 AM, William Suffill <
>>>>> william.suffill at gmail.com> wrote:
>>>>>
>>>>>> Why do you need it every second? If you want real time channel counts
>>>>>> you would  be able to track each create/destroy even instead of
>>>>>> relying on the heartbeat summary.
>>>>>>
>>>>>> -- W
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:213-799-1400
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Oct 22 10:54:40 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 22 Oct 2009 10:54:40 -0700
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179
In-Reply-To: <b39273530910220551q14d22150ldf8d4bff0b6fd011@mail.gmail.com>
References: <mailman.1516.1256123625.2495.freeswitch-users@lists.freeswitch.org>
	<a60fed5e0910212108v4755203y693b8bdec5501f5f@mail.gmail.com>
	<87f2f3b90910212141w72b8f5a2oce032adb974d71e2@mail.gmail.com>
	<b39273530910220551q14d22150ldf8d4bff0b6fd011@mail.gmail.com>
Message-ID: <87f2f3b90910221054l8abc3far9a18a66f0959f950@mail.gmail.com>

On Thu, Oct 22, 2009 at 5:51 AM, Rupa Schomaker <rupa at rupa.com> wrote:

> cond would be helpful here?  I updated the wiki on this one just now
> with a bit more detail.  It is a api call. so, you'd use it like:
>
> ${cond(eval ? trueval : falseval)}
>
> so to get a value of ERR if the var my myvar is > 15 you could:
>
> ${cond(${myvar} > 15 ? ERR : OK)}
>
> If both sides of the comparison operator are numeric then it does
> numeric comparison otherwise it does lexical string comparison.
>

Rupa,

Yes, you can do the set/cond API trick but you can only do it in the action
or anti-action tags, not in the condition tags. I'm sure you know that but I
want all those reading this thread to make the connection.
-MC
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From kristian.kielhofner at gmail.com  Thu Oct 22 10:56:15 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Thu, 22 Oct 2009 13:56:15 -0400
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
In-Reply-To: <65d96fc80910220858u788c6146sced38e19418e681a@mail.gmail.com>
References: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>
	<191c3a030910201434q4c6e65e1h58c5f0a82addc2d3@mail.gmail.com>
	<2d9149cd0910202050i36a7ea82nc7253d87e38f990a@mail.gmail.com>
	<191c3a030910211154gc8412b5gd999fbedffa02b53@mail.gmail.com>
	<2d9149cd0910220844v2aa73502j4b88fdbd75a0d833@mail.gmail.com>
	<65d96fc80910220858u788c6146sced38e19418e681a@mail.gmail.com>
Message-ID: <2d9149cd0910221056y3716d8c4s708fbdb6a5563733@mail.gmail.com>

On Thu, Oct 22, 2009 at 11:58 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>
> and what these few issues are? :P
>

  One fax machine here in the office (still testing others) correctly
sends all fax pages.  A minute or so after the fax is marked
"successful" on both sides it hangs up, redials, and resends the last
page...  It never did it while connected to the PSTN but then again my
other fax machine isn't doing it either.  I'm going to test with more
fax machines to see if it's an issue with that specific machine.

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From gkuri at ieee.org  Thu Oct 22 11:16:29 2009
From: gkuri at ieee.org (Gabriel Kuri)
Date: Thu, 22 Oct 2009 11:16:29 -0700
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
In-Reply-To: <2d9149cd0910221056y3716d8c4s708fbdb6a5563733@mail.gmail.com>
References: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>	<191c3a030910201434q4c6e65e1h58c5f0a82addc2d3@mail.gmail.com>	<2d9149cd0910202050i36a7ea82nc7253d87e38f990a@mail.gmail.com>	<191c3a030910211154gc8412b5gd999fbedffa02b53@mail.gmail.com>	<2d9149cd0910220844v2aa73502j4b88fdbd75a0d833@mail.gmail.com>	<65d96fc80910220858u788c6146sced38e19418e681a@mail.gmail.com>
	<2d9149cd0910221056y3716d8c4s708fbdb6a5563733@mail.gmail.com>
Message-ID: <4AE0A17D.3000301@ieee.org>

Out of curiosity, is it a Super G3 (ie v.34) capable FAX? We've had
nothing but intermittent problems with Super G3 FAXes over T.38, unless
v.34 is strictly turned off on the machine.

Gabe


Kristian Kielhofner wrote:
> On Thu, Oct 22, 2009 at 11:58 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>
>   One fax machine here in the office (still testing others) correctly
> sends all fax pages.  A minute or so after the fax is marked
> "successful" on both sides it hangs up, redials, and resends the last
> page...  It never did it while connected to the PSTN but then again my
> other fax machine isn't doing it either.  I'm going to test with more
> fax machines to see if it's an issue with that specific machine.
> 




From msc at freeswitch.org  Thu Oct 22 11:18:41 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 22 Oct 2009 11:18:41 -0700
Subject: [Freeswitch-users] Heartbeat question
In-Reply-To: <c96b95650910221047i545601a4m9e0a645341068d15@mail.gmail.com>
References: <86a32abc0910081956k91b3600sb8c556e589f5b089@mail.gmail.com>
	<86a32abc0910082003v3f5d4e6dua8e7eae18fd2f939@mail.gmail.com>
	<6b65470d0910082036g204d5dcbw26e1e43ef80c7c84@mail.gmail.com>
	<86a32abc0910082245j36b8b3e9ncaf1cdd358fce319@mail.gmail.com>
	<191c3a030910090902t38282f72l6b6d1733a758836@mail.gmail.com>
	<86a32abc0910091427i2395fd80t3bc557186ece3972@mail.gmail.com>
	<86a32abc0910091430hc151ed8of085fcc3af2f1de4@mail.gmail.com>
	<86a32abc0910091501y40dd34c4r1bca3eeb2b6fa693@mail.gmail.com>
	<c96b95650910221047i545601a4m9e0a645341068d15@mail.gmail.com>
Message-ID: <87f2f3b90910221118n5d9fa0b9u92b45227c1d7ec97@mail.gmail.com>

On Thu, Oct 22, 2009 at 10:47 AM, Saeed Ahmad <saeedahmad1981 at gmail.com>wrote:

> What is heartbeat and what are the uses cases?
>
> Sorry i didn't find much information on wiki.
>
> Thanks.
>
> A session heartbeat is just an event that is sent to your script and gives
you updated information about the call in progress. It is *VERY* different
from the system heartbeat, which is sent out to all event socket
subscribers, not just a particular session.

-MC
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From ujjval at simplesignal.com  Thu Oct 22 11:42:49 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Thu, 22 Oct 2009 11:42:49 -0700
Subject: [Freeswitch-users] Connecting to FS CLI...just hangs..
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701E84092@EXMBXCLUS01.citservers.local>

It just hangs....and I CTRL-C out of it.

[root at ss]# ./fs_cli -H 127.0.0.1


^C
[ERROR] libs/esl/fs_cli.c:652 main() Error Connecting [Connection Error]


Freeswitch process is running:

[root at ss bin]# ps -ef|grep free
root      8889 31039  0 12:36 pts/2    00:00:00 ./freeswitch
root      8952 31039  0 12:42 pts/2    00:00:00 grep free


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From tculjaga at gmail.com  Thu Oct 22 12:08:36 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 22 Oct 2009 21:08:36 +0200
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
In-Reply-To: <4AE0A17D.3000301@ieee.org>
References: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>
	<191c3a030910201434q4c6e65e1h58c5f0a82addc2d3@mail.gmail.com>
	<2d9149cd0910202050i36a7ea82nc7253d87e38f990a@mail.gmail.com>
	<191c3a030910211154gc8412b5gd999fbedffa02b53@mail.gmail.com>
	<2d9149cd0910220844v2aa73502j4b88fdbd75a0d833@mail.gmail.com>
	<65d96fc80910220858u788c6146sced38e19418e681a@mail.gmail.com>
	<2d9149cd0910221056y3716d8c4s708fbdb6a5563733@mail.gmail.com>
	<4AE0A17D.3000301@ieee.org>
Message-ID: <65d96fc80910221208h70c86457s4ca5f68289bacad2@mail.gmail.com>

indeed, this looks like a "dialect" problem between your fax machine and
your T.38 device.
Anyhow, T.38  doesn't work well with SG3... I Always have to disable v.34 in
order to have a reliable fax service.

Also, cisco uses to suppress CM so that SG3 timeouts on ANSam the
communication fallbacks to ordinary G3.

Kristian, just for fun, what are you using to send the fax ?

T.

On Thu, Oct 22, 2009 at 8:16 PM, Gabriel Kuri <gkuri at ieee.org> wrote:

> Out of curiosity, is it a Super G3 (ie v.34) capable FAX? We've had
> nothing but intermittent problems with Super G3 FAXes over T.38, unless
> v.34 is strictly turned off on the machine.
>
> Gabe
>
>
> Kristian Kielhofner wrote:
> > On Thu, Oct 22, 2009 at 11:58 AM, Tihomir Culjaga <tculjaga at gmail.com>
> wrote:
> >
> >   One fax machine here in the office (still testing others) correctly
> > sends all fax pages.  A minute or so after the fax is marked
> > "successful" on both sides it hangs up, redials, and resends the last
> > page...  It never did it while connected to the PSTN but then again my
> > other fax machine isn't doing it either.  I'm going to test with more
> > fax machines to see if it's an issue with that specific machine.
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jmesquita at freeswitch.org  Thu Oct 22 12:11:09 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 22 Oct 2009 17:11:09 -0200
Subject: [Freeswitch-users] Connecting to FS CLI...just hangs..
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71701E84092@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84092@EXMBXCLUS01.citservers.local>
Message-ID: <ebbc2d550910221211k18171473p7f4d6f597f6b21d2@mail.gmail.com>

Hangs for how long? Are you sure you are not just waiting on a timeout?

JM

On Thu, Oct 22, 2009 at 4:42 PM, Ujjval Karihaloo
<ujjval at simplesignal.com>wrote:

>  It just hangs?.and I CTRL-C out of it.
>
>
>
> [root at ss]# ./fs_cli -H 127.0.0.1
>
>
>
>
>
> ^C
>
> [ERROR] libs/esl/fs_cli.c:652 main() Error Connecting [Connection Error]
>
>
>
>
>
> Freeswitch process is running:
>
>
>
> [root at ss bin]# ps -ef|grep free
>
> root      8889 31039  0 12:36 pts/2    00:00:00 ./freeswitch
>
> root      8952 31039  0 12:42 pts/2    00:00:00 grep free
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kristian.kielhofner at gmail.com  Thu Oct 22 12:13:58 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Thu, 22 Oct 2009 15:13:58 -0400
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
In-Reply-To: <4AE0A17D.3000301@ieee.org>
References: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>
	<191c3a030910201434q4c6e65e1h58c5f0a82addc2d3@mail.gmail.com>
	<2d9149cd0910202050i36a7ea82nc7253d87e38f990a@mail.gmail.com>
	<191c3a030910211154gc8412b5gd999fbedffa02b53@mail.gmail.com>
	<2d9149cd0910220844v2aa73502j4b88fdbd75a0d833@mail.gmail.com>
	<65d96fc80910220858u788c6146sced38e19418e681a@mail.gmail.com>
	<2d9149cd0910221056y3716d8c4s708fbdb6a5563733@mail.gmail.com>
	<4AE0A17D.3000301@ieee.org>
Message-ID: <2d9149cd0910221213h20661e11pdccafdf407c79b02@mail.gmail.com>

Gabe,

  I don't think any of them are plus the T.38 SDP tells me the bitrate
is 14400, certainly not V.34 speed.

  Are you saying the machine even trying to negotiate V.34 poses a problem?

On Thu, Oct 22, 2009 at 2:16 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
> Out of curiosity, is it a Super G3 (ie v.34) capable FAX? We've had
> nothing but intermittent problems with Super G3 FAXes over T.38, unless
> v.34 is strictly turned off on the machine.
>
> Gabe
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From bottleman at icf.org.ru  Thu Oct 22 12:21:39 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Thu, 22 Oct 2009 23:21:39 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910220753n6d2efa39l470791f75db9a640@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910211617520.32656@srv.icf.org.ru>
	<65d96fc80910210743y2d8dacfdvf55e8acaf476de5f@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>
	<65d96fc80910220753n6d2efa39l470791f75db9a640@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>

On 2009-10-22 16:53 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

finally i fix this rtp bug, check new wersion please.

TC>>
TC>> TC>
TC>> TC>Do you need some logs ?
TC>>
TC>> try disable medai-proxy, there is issue with rtp now then medai-proxy or
TC>> transcoding enabled.
TC>>
TC>>
TC>Outbound calls:
TC>
TC>disabled rtp proxy and it is still the same issue ... audio delay H323 =>
TC>SIP endpoint.
TC>
TC>
TC>
TC>
TC>
TC>
TC>Inbound calls:
TC>
TC>This is the extension i use to register my Avaya SIP phone to FS.
TC>
TC>
TC><include>
TC>  <user id="1001">
TC>    <params>
TC>      <param name="password" value="$${default_password}"/>
TC>      <param name="vm-password" value="1001"/>
TC>    </params>
TC>    <variables>
TC>      <variable name="toll_allow" value="domestic,international,local"/>
TC>      <variable name="accountcode" value="1001"/>
TC>      <variable name="user_context" value="default"/>
TC>      <variable name="effective_caller_id_name" value="Extension 1001"/>
TC>      <variable name="effective_caller_id_number" value="1001"/>
TC>      <variable name="outbound_caller_id_name"
TC>value="$${outbound_caller_name}"/>
TC>      <variable name="outbound_caller_id_number"
TC>value="$${outbound_caller_id}"/>
TC>      <variable name="callgroup" value="techsupport"/>
TC>    </variables>
TC>  </user>
TC></include>
TC>
TC>
TC>This is my h323.conf.xml
TC>
TC>
TC><configuration name="h323.conf" description="H323 Endpoints">
TC>  <settings>
TC>    <param name="trace-level" value="4"/>
TC>    <param name="context" value="default"/>
TC>    <param name="dialplan" value="XML"/>
TC>    <param name="codec-prefs" value="PCMU,PCMA,GSM,G729,G726"/>
TC>    <param name="gk-address" value=""/>    <!-- empty to disable, "*" to
TC>search LAN -->
TC>    <param name="gk-identifer" value=""/>  <!-- optional name of gk -->
TC>    <param name="gk-interface" value=""/>  <!-- optional listener interface
TC>name -->
TC>  </settings>
TC>  <listeners>
TC>    <listener name="default">
TC>      <param name="h323-ip" value="10.4.62.7"/>
TC>      <param name="h323-port" value="1720"/>
TC>    </listener>
TC>  </listeners>
TC></configuration>
TC>
TC>I'm using default context and an inbound call looks for a registered user in
TC>default context where 1001 user is registered to.
TC>
TC>
TC>
TC>here is the log for an outgoing call:
TC>http://pastebin.freeswitch.org/10799and here is a tshark output:
TC>http://pastebin.freeswitch.org/10800
TC>
TC>
TC>there are 2 thing that are not working here:
TC>
TC>
TC>1. no audio at all!
TC>2. hangup from SIP User side doesn't release the H323 leg
TC>
TC>
TC>
TC>
TC>
TC>
TC>
TC>
TC>
TC>
TC>
TC>two points for your reference in the logs:
TC>
TC>
TC>1. Here, SIP User disconnected the SIP leg, but nothing was triggered in
TC>mod_h323 ... as the callback function "on_hangup" (in mod_h323.cpp) was
TC>never triggered!
TC>
TC>freeswitch at subZero>
TC>freeswitch at subZero>
TC>freeswitch at subZero> recv 371 bytes from udp/[10.4.62.89]:5060 at
TC>14:39:36.714521:
TC>   ------------------------------------------------------------------------
TC>   BYE sip:mod_sofia at 10.4.62.7:5060 SIP/2.0
TC>   From: <sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>
TC>;transport=udp>;tag=-22166b474ae08abf-7_T10.4.62.89
TC>   To: sip:1001282166 at 10.4.62.7 <sip%3A1001282166 at 10.4.62.7>
TC>;tag=Qpc53NZ2cZc1N
TC>   Call-ID: 8aa825c6-39bb-122d-bb89-00110a5be1f0
TC>   CSeq: 127 BYE
TC>   Via: SIP/2.0/UDP
TC>10.4.62.89;branch=z9hG4bK-7e5dc720_442d0f8-2d8bf1174f235bec_B
TC>   Content-Length: 0
TC>   Max-Forwards: 70
TC>   Supported: replaces
TC>
TC>   ------------------------------------------------------------------------
TC>2009-10-22 16:39:36.714604 [NOTICE] sofia.c:322 Hangup sofia/internal/
TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [CS_CONSUME_MEDIA]
TC>[NORMAL_CLEARING]
TC>2009-10-22 16:39:36.714604 [DEBUG] switch_channel.c:1683 Send signal
TC>sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [KILL]
TC>2009-10-22 16:39:36.714604 [DEBUG] switch_core_session.c:932 Send signal
TC>sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [BREAK]
TC>send 520 bytes to udp/[10.4.62.89]:5060 at 14:39:36.715258:
TC>   ------------------------------------------------------------------------
TC>   SIP/2.0 200 OK
TC>   Via: SIP/2.0/UDP
TC>10.4.62.89;branch=z9hG4bK-7e5dc720_442d0f8-2d8bf1174f235bec_B
TC>   From: <sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>
TC>;transport=udp>;tag=-22166b474ae08abf-7_T10.4.62.89
TC>   To: sip:1001282166 at 10.4.62.7 <sip%3A1001282166 at 10.4.62.7>
TC>;tag=Qpc53NZ2cZc1N
TC>   Call-ID: 8aa825c6-39bb-122d-bb89-00110a5be1f0
TC>   CSeq: 127 BYE
TC>   User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported
TC>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
TC>NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
TC>   Supported: timer, precondition, path, replaces
TC>   Content-Length: 0
TC>
TC>   ------------------------------------------------------------------------
TC>2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:503
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State
TC>CONSUME_MEDIA going to sleep
TC>2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:398
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) Running State
TC>Change CS_HANGUP
TC>2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:434
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State HANGUP
TC>2009-10-22 16:39:36.721097 [DEBUG] mod_sofia.c:338 Channel sofia/internal/
TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> hanging up, cause:
TC>NORMAL_CLEARING
TC>2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:46
TC>sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> Standard HANGUP,
TC>cause: NORMAL_CLEARING
TC>2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:434
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State HANGUP
TC>going to sleep
TC>2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:476
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State Change
TC>CS_HANGUP -> CS_REPORTING
TC>2009-10-22 16:39:36.721097 [DEBUG] switch_core_session.c:932 Send signal
TC>sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> [BREAK]
TC>2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:398
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) Running State
TC>Change CS_REPORTING
TC>2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:612
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State REPORTING
TC>2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:53
TC>sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> Standard
TC>REPORTING, cause: NORMAL_CLEARING
TC>2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:612
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State REPORTING
TC>going to sleep
TC>2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:411
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) State Change
TC>CS_REPORTING -> CS_DESTROY
TC>2009-10-22 16:39:36.721097 [DEBUG] switch_core_session.c:1068 Session 10
TC>(sofia/internal/sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>) Locked, Waiting
TC>on external entities
TC>
TC>freeswitch at subZero>
TC>
TC>
TC>
TC>
TC>
TC>2. switch_ivr_originate.c complains => 2009-10-22 16:39:41.630243 [ERR]
TC>switch_ivr_originate.c:1510 Cannot create outgoing channel of type [user]
TC>cause: [NORMAL_CLEARING]
TC>
TC>
TC>freeswitch at subZero>
TC>freeswitch at subZero>
TC>freeswitch at subZero> 2009-10-22 16:39:41.630243 [DEBUG] h323pdu.cxx:609
TC>Receiving PDU [ip$10.4.62.7:1720/ip$10.4.62.31:16013] :
TC>  {
TC>    q931pdu = {
TC>      protocolDiscriminator = 8
TC>      callReference = 23
TC>      from = originator
TC>      messageType = ReleaseComplete
TC>      IE: Cause - Normal ca
TC>ll clearing = {
TC>        80 90                                              ..
TC>      }
TC>      IE: User-User = {
TC>        25 80 06 00 08 91 4a 00  04 11 00 11 00 96 40 1f   %.....J....... at .
TC>        4e be 4f 11 de 80 76 ae  99 96 03 ba ff 10 80 01   N.O.
TC>..v.........
TC>        ...
TC>      }
TC>    }
TC>    h225pdu = {
TC>      h323_uu_pdu = {
TC>        h323_message_body = releaseComplete {
TC>          protocolIdentifier = 0.0.8.2250.0.4
TC>          callIdentifier = {
TC>            guid =  16 octets {
TC>              96 40
TC> 1f 4e be 4f 11 de  80 76 ae 99 96 03 ba ff   . at .N.O...v......
TC>            }
TC>          }
TC>        }
TC>        h245Tunneling = true
TC>      }
TC>    }
TC>  }
TC>2009-10-22 16:39:41.630243 [DEBUG] h323.cxx:1292 Handling PDU:
TC>ReleaseComplete callRef=23
TC>2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:745
TC>======>FSH323Connection::OnReceivedReleaseComplete cause = Normal call
TC>clearing value = 16
TC>2009-10-22 16:39:41.630243 [NOTICE] mod_h323.cpp:749 Hangup h323/1001
TC>[CS_EXECUTE] [NORMAL_CLEARING]
TC>2009-10-22 16:39:41.630243 [DEBUG] switch_channel.c:1683 Send signal
TC>h323/1001 [KILL]
TC>2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:840
TC>======>FSH323Connection::kill_channel FSH323Connection
TC>2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:841 Kill 1 on connection
TC>FSH323Connection
TC>2009-10-22 16:39:41.630243 [DEBUG] switch_core_session.c:630 Send signal
TC>h323/1001 [BREAK]
TC>2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:840
TC>======>FSH323Connection::kill_channel FSH323Connection
TC>2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:841 Kill 3 on connection
TC>FSH323Connection
TC>2009-10-22 16:39:41.630243 [DEBUG] switch_ivr_originate.c:2138 Originate
TC>Resulted in Error Cause: 16 [NORMAL_CLEARING]
TC>2009-10-22 16:39:41.630243 [ERR] switch_ivr_originate.c:1510 Cannot create
TC>outgoing channel of type [user] cause: [NORMAL_CLEARING]
TC>2009-10-22 16:39:41.630243 [DEBUG] switch_ivr_originate.c:2138 Originate
TC>Resulted in Error Cause: 16 [NORMAL_CLEARING]
TC>2009-10-22 16:39:41.630243 [INFO] mod_dptools.c:2093 Originate Failed.
TC>Cause: NORMAL_CLEARING
TC>2009-10-22 16:39:41.630243 [DEBUG] mod_dptools.c:2120 Continue on fail
TC>[true]:  Cause: NORMAL_CLEARING
TC>2009-10-22 16:39:41.630243 [DEBUG] switch_core_state_machine.c:491
TC>(h323/1001) State EXECUTE going to sleep
TC>2009-10-22 16:39:41.630243 [DEBUG] switch_core_state_machine.c:398
TC>(h323/1001) Running State Change CS_HANGUP
TC>2009-10-22 16:39:41.630243 [DEBUG] switch_core_state_machine.c:434
TC>(h323/1001) State HANGUP
TC>2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:1454 ======>switch_status_t
TC>on_hangup
TC>2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:1459 ----->
TC>2009-10-22 16:39:41.630243 [DEBUG] h323pdu.cxx:1272 Call End Reason Normal
TC>call clearing
TC>2009-10-22 16:39:41.630243 [DEBUG] h323ep.cxx:2694 Clearing connection ip$
TC>10.4.62.31:16013/23 reason=EndedByRemoteUser
TC>2009-10-22 16:39:41.630243 [DEBUG] h323.cxx:1034 Call end reason for ip$
TC>10.4.62.31:16013/23 set to EndedByRemoteUser
TC>2009-10-22 16:39:41.630243 [DEBUG] h323.cxx:1052 Sending release complete
TC>PDU: callRef=23
TC>2009-10-22 16:39:41.630243 [DEBUG] mod_h323.cpp:764
TC>======>FSH323Connection::OnSendReleaseComplete cause = Normal call clearing
TC>value = 16
TC>2009-10-22 16:39:41.630243 [DEBUG] h323pdu.cxx:609 Sending PDU
TC>[(noaddr)/(noaddr)] :
TC>  command endSessionCommand disconnect <<null>>
TC>2009-10-22 16:39:41.630243 [DEBUG] h323pdu.cxx:609 Sending PDU [ip$
TC>10.4.62.7:1720/ip$10.4.62.31:16013] :
TC>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From anthony.minessale at gmail.com  Thu Oct 22 12:26:44 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 22 Oct 2009 14:26:44 -0500
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
In-Reply-To: <4256bf830910221042l68ea769cg532bbd339bae0790@mail.gmail.com>
References: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
	<177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>
	<4256bf830908041151l55980e3fifb85c75b87535426@mail.gmail.com>
	<3D0B855A-5B2C-4C6F-9306-64B0ECE97091@jerris.com>
	<4256bf830910180725y72539d98gb138934afc18d58b@mail.gmail.com>
	<A5ECDCD9-1A12-4A31-89F0-E6F4C79FE9E9@jerris.com>
	<4256bf830910182148mf7f2bf9vdcccd687e0caa1e7@mail.gmail.com>
	<87f2f3b90910191144q25750bayfd7a7448e82e78ff@mail.gmail.com>
	<4256bf830910221042l68ea769cg532bbd339bae0790@mail.gmail.com>
Message-ID: <191c3a030910221226j167e0adw5f631f759cd7cfa0@mail.gmail.com>

1) please do not flood emails with traces reference pastebins
2) the error is a routing loop being prevented so you are looping the call
into your same box too many times.

You could have figured this out your self by looking at one of the lines of
code where it reported the hangup.

 if (!switch_strlen_zero(forwardvar)) {
        forwardval = atoi(forwardvar) - 1;
    }
    if (forwardval <= 0) {
        switch_channel_hangup(channel, SWITCH_CAUSE_EXCHANGE_ROUTING_ERROR);
        return SWITCH_STATUS_FALSE;
    }


On Thu, Oct 22, 2009 at 12:42 PM, Matthew Fong <mattdfong at gmail.com> wrote:

> I figured out how to turn debug mode on. For some reason F8 didn't work, so
> I just set the default debug mode to DEBUG in switch.conf.xml
>
> http://pastebin.freeswitch.org/10805
> <http://pastebin.freeswitch.org/10805>So my program does more or less the
> same sequence for an Agent every time when connected to another party. They
> uuid_kill the bridged parties uuid, and then are transfered to an extension
> that park's the agent (using the transfer_after_bridge variable -- in my
> dialplan extn 1998). In some instances the agent is then transfered to
> another extension, played a wav file, and then park'ed again
>
> Sometimes, FreeSWITCH unexpectidly hangsup the Agent channel citing an
> EXCHANGE_ROUTING_ERROR. It seems (to me to happen at random times)...and
> it's more prevelent with some did providers than others. With
> didforsale.org it happens about one ever 5 bridges. With icall.com it only
> happens about once every 30 bridges. With IPKall it happens rarely. I can't
> figure it out, and it's driving me crazy :)
>
>
> A GOOD UUID_KILL SEQUENCE:
> 2009-10-22 17:22:31.741845 [DEBUG] switch_ivr_play_say.c:1432 done playing
> file
> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:340
> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXECUTE going to
> sleep
> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
> CS_HIBERNATE
> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:355
> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE
> 2009-10-22 17:22:31.741845 [DEBUG] mod_sofia.c:160 sofia/external/+
> 14158867717 at 199.173.100.16:5060 SOFIA HIBERNATE
> 2009-10-22 17:22:31.741845 [DEBUG] switch_ivr_bridge.c:650
> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
> CS_HIBERNATE -> CS_RESET
> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:355
> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE going to
> sleep
> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
> CS_RESET
> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:336
> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET
> 2009-10-22 17:22:31.741845 [DEBUG] mod_sofia.c:145 sofia/external/+
> 14158867717 at 199.173.100.16:5060 SOFIA RESET
> 2009-10-22 17:22:31.741845 [DEBUG] switch_ivr_bridge.c:637 sofia/external/+
> 14158867717 at 199.173.100.16:5060 CUSTOM RESET
> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:67
> sofia/external/+14158867717 at 199.173.100.16:5060 Standard RESET
> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:336
> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET going to
> sleep
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:340
> (sofia/external/14159927717) State EXECUTE going to sleep
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/14159927717) Running State Change CS_HIBERNATE
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:355
> (sofia/external/14159927717) State HIBERNATE
> 2009-10-22 17:22:31.748086 [DEBUG] mod_sofia.c:160
> sofia/external/14159927717 SOFIA HIBERNATE
> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:650
> (sofia/external/14159927717) State Change CS_HIBERNATE -> CS_RESET
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/14159927717 [BREAK]
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:355
> (sofia/external/14159927717) State HIBERNATE going to sleep
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/14159927717) Running State Change CS_RESET
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:336
> (sofia/external/14159927717) State RESET
> 2009-10-22 17:22:31.748086 [DEBUG] mod_sofia.c:145
> sofia/external/14159927717 SOFIA RESET
> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:637
> sofia/external/14159927717 CUSTOM RESET
> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:642
> (sofia/external/14159927717) State Change CS_RESET -> CS_SOFT_EXECUTE
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/14159927717 [BREAK]
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:336
> (sofia/external/14159927717) State RESET going to sleep
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/14159927717) Running State Change CS_SOFT_EXECUTE
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:346
> (sofia/external/14159927717) State SOFT_EXECUTE
> 2009-10-22 17:22:31.748086 [DEBUG] mod_sofia.c:442 SOFIA TRANSMIT
> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:660
> sofia/external/14159927717 CUSTOM SOFT_EXECUTE
> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_originate.c:674
> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_RESET ->
> CS_SOFT_EXECUTE
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:120 sofia/external/+
> 14158867717 at 199.173.100.16:5060 receive message [DISPLAY]
> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:120
> sofia/external/14159927717 receive message [DISPLAY]
> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:924
> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
> CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:968 sofia/external/+
> 14158867717 at 199.173.100.16:5060 receive message [BRIDGE]
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:630 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:975
> sofia/external/14159927717 receive message [BRIDGE]
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:630 Send signal
> sofia/external/14159927717 [BREAK]
> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:1019
> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:22:31.749553 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
> CS_EXCHANGE_MEDIA
> 2009-10-22 17:22:31.749553 [DEBUG] switch_core_state_machine.c:343
> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
> 2009-10-22 17:22:31.749553 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK
> multipler is being overrided to1
> 2009-10-22 17:22:34.300662 [INFO] switch_cpp.cpp:1116 BG_AG_GEN_CALLS,
> Agents total:1 avail:0 OutChan:0 AR:0 +-Calls:0:Q Call NYNoc
> multipler is being overrided to1
> 2009-10-22 17:22:39.303041 [INFO] switch_cpp.cpp:1116 BG_AG_GEN_CALLS,
> Agents total:1 avail:0 OutChan:0 AR:0 +-Calls:0:Q Call NYNoc
> 2009-10-22 17:22:40.626379 [NOTICE] sofia.c:328 Hangup
> sofia/external/14159927717 [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
> 2009-10-22 17:22:40.626379 [DEBUG] switch_channel.c:1810 Send signal
> sofia/external/14159927717 [KILL]
> 2009-10-22 17:22:40.626379 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/14159927717 [BREAK]
> active_true=1
> --contact NOT in playmsg or conference, update AGENT
> 2009-10-22 17:22:40.626379 [DEBUG] switch_core_state_machine.c:437 thread
> mismatch skipping state handler.
> 2009-10-22 17:22:40.626379 [INFO] switch_cpp.cpp:1116
> EVENTC_HANGUP_CONTACT, Q_CONTACT fired for agent:1/1 for qcall:1
> contact:4801442/4
> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:440
> sofia/external/14159927717 ending bridge by request from write function
> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:495 sofia/external/+
> 14158867717 at 199.173.100.16:5060 receive message [UNBRIDGE]
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:630 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
> DONE [sofia/external/+14158867717 at 199.173.100.16:5060]
> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:523 Send signal
> sofia/external/14159927717 [BREAK]
> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr.c:1361 (sofia/external/+
> 14158867717 at 199.173.100.16:5060) State Change CS_EXCHANGE_MEDIA ->
> CS_ROUTING
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr.c:1365 sofia/external/+
> 14158867717 at 199.173.100.16:5060 receive message [TRANSFER]
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:630 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:22:40.639760 [NOTICE] switch_ivr.c:1367 Transfer
> sofia/external/+14158867717 at 199.173.100.16:5060 to xml[1998 at default]
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:343
> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
> going to sleep
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
> CS_ROUTING
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:333
> (sofia/external/+14158867717 at 199.173.100.16:5060) State ROUTING
> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:130 sofia/external/+
> 14158867717 at 199.173.100.16:5060 SOFIA ROUTING
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:78
> sofia/external/+14158867717 at 199.173.100.16:5060 Standard ROUTING
> 2009-10-22 17:22:40.639760 [INFO] mod_dialplan_xml.c:391 Processing
> +14158867717->1998 in context default
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->tod_example] continue=true
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (PASS)
> [tod_example] ${strftime(%w)}(4) =~ /^([1-5])$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (PASS)
> [tod_example] ${strftime(%H%M)}(1722) =~ /^((09|1[0-7])[0-5][0-9]|1800)$/
> break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Action
> set(open=true)
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->ivr_demo] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [ivr_demo] destination_number(1998) =~ /^5000$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1975] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1975] destination_number(1998) =~ /^1975$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1971] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1971] destination_number(1998) =~ /^1971$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1972] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1972] destination_number(1998) =~ /^1972$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1920] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1920] destination_number(1998) =~ /^1920$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1981] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1981] destination_number(1998) =~ /^1981$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1982] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1982] destination_number(1998) =~ /^1982$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1481] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1481] destination_number(1998) =~ /^1481$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1989] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1989] destination_number(1998) =~ /^1989$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1990] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1990] destination_number(1998) =~ /^1990$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1991] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1991] destination_number(1998) =~ /^1991$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1992] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1992] destination_number(1998) =~ /^1992$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1993] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1993] destination_number(1998) =~ /^1993$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1994] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1994] destination_number(1998) =~ /^1994$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1995] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1995] destination_number(1998) =~ /^1995$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1996] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1996] destination_number(1998) =~ /^1996$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1997] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
> [1997] destination_number(1998) =~ /^1997$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
> [default->1998] continue=false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (PASS)
> [1998] destination_number(1998) =~ /^1998$/ break=on-false
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Action
> set(transfer_after_bridge=1998)
> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Action park()
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:114
> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_ROUTING
> -> CS_EXECUTE
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:333
> (sofia/external/+14158867717 at 199.173.100.16:5060) State ROUTING going to
> sleep
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
> CS_EXECUTE
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:340
> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXECUTE
> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:173 sofia/external/+
> 14158867717 at 199.173.100.16:5060 SOFIA EXECUTE
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:151
> sofia/external/+14158867717 at 199.173.100.16:5060 Standard EXECUTE
> EXECUTE sofia/external/+14158867717 at 199.173.100.16:5060 set(open=true)
> 2009-10-22 17:22:40.639760 [DEBUG] mod_dptools.c:752 sofia/external/+
> 14158867717 at 199.173.100.16:5060 SET [open]=[true]
> EXECUTE sofia/external/+14158867717 at 199.173.100.16:5060set(transfer_after_bridge=1998)
> 2009-10-22 17:22:40.639760 [DEBUG] mod_dptools.c:752 sofia/external/+
> 14158867717 at 199.173.100.16:5060 SET [transfer_after_bridge]=[1998]
> EXECUTE sofia/external/+14158867717 at 199.173.100.16:5060 park()
> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:446
> sofia/external/14159927717 ending bridge by request from read function
> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
> DONE [sofia/external/14159927717]
> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:523 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> subtracting 9 seconds for hh_call: 1095422
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:346
> (sofia/external/14159927717) State SOFT_EXECUTE going to sleep
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/14159927717) Running State Change CS_HANGUP
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:464
> (sofia/external/14159927717) State HANGUP
> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:344 Channel
> sofia/external/14159927717 hanging up, cause: NORMAL_CLEARING
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:46
> sofia/external/14159927717 Standard HANGUP, cause: NORMAL_CLEARING
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:464
> (sofia/external/14159927717) State HANGUP going to sleep
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:325
> (sofia/external/14159927717) State Change CS_HANGUP -> CS_REPORTING
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/14159927717 [BREAK]
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/14159927717) Running State Change CS_REPORTING
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:555
> (sofia/external/14159927717) State REPORTING
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:53
> sofia/external/14159927717 Standard REPORTING, cause: NORMAL_CLEARING
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:555
> (sofia/external/14159927717) State REPORTING going to sleep
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:319
> (sofia/external/14159927717) State Change CS_REPORTING -> CS_DESTROY
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/14159927717 [BREAK]
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:1069 Session 6
> (sofia/external/14159927717) Locked, Waiting on external entities
> 2009-10-22 17:22:40.639760 [NOTICE] switch_core_session.c:1087 Session 6
> (sofia/external/14159927717) Ended
> 2009-10-22 17:22:40.639760 [NOTICE] switch_core_session.c:1089 Close
> Channel sofia/external/14159927717 [CS_DESTROY]
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:401
> (sofia/external/14159927717) Running State Change CS_DESTROY
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:412
> (sofia/external/14159927717) State DESTROY
> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:261
> sofia/external/14159927717 SOFIA DESTROY
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:60
> sofia/external/14159927717 Standard DESTROY
> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:412
> (sofia/external/14159927717) State DESTROY going to sleep
> 2009-10-22 17:22:40.639760 [INFO] switch_cpp.cpp:1116 PCHangup gw:
> teliax.com hc:NORMAL_CLEARING du:9 cn:sofia/external/14159927717
>
>
>
> A BAD UUID_KILL SEQUENCE: (where the agent is hungup, unexpectedly)
>
> --bridged in: 0.005821 seconds
> 2009-10-22 17:23:26.718930 [INFO] switch_cpp.cpp:1116 AG_CBRIDGE, connected
> agent:1/1 contact:4801439/22/1/1
> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_play_say.c:1432 done playing
> file
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:340
> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXECUTE going to
> sleep
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
> CS_HIBERNATE
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE
> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:160 sofia/external/+
> 14158867717 at 199.173.100.16:5060 SOFIA HIBERNATE
> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:650
> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
> CS_HIBERNATE -> CS_RESET
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE going to
> sleep
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
> CS_RESET
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET
> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:145 sofia/external/+
> 14158867717 at 199.173.100.16:5060 SOFIA RESET
> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:637 sofia/external/+
> 14158867717 at 199.173.100.16:5060 CUSTOM RESET
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:67
> sofia/external/+14158867717 at 199.173.100.16:5060 Standard RESET
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET going to
> sleep
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:340
> (sofia/external/14159927717) State EXECUTE going to sleep
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/14159927717) Running State Change CS_HIBERNATE
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
> (sofia/external/14159927717) State HIBERNATE
> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:160
> sofia/external/14159927717 SOFIA HIBERNATE
> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:650
> (sofia/external/14159927717) State Change CS_HIBERNATE -> CS_RESET
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/14159927717 [BREAK]
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
> (sofia/external/14159927717) State HIBERNATE going to sleep
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/14159927717) Running State Change CS_RESET
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
> (sofia/external/14159927717) State RESET
> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:145
> sofia/external/14159927717 SOFIA RESET
> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:637
> sofia/external/14159927717 CUSTOM RESET
> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:642
> (sofia/external/14159927717) State Change CS_RESET -> CS_SOFT_EXECUTE
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/14159927717 [BREAK]
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
> (sofia/external/14159927717) State RESET going to sleep
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/14159927717) Running State Change CS_SOFT_EXECUTE
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:346
> (sofia/external/14159927717) State SOFT_EXECUTE
> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:442 SOFIA TRANSMIT
> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:660
> sofia/external/14159927717 CUSTOM SOFT_EXECUTE
> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_originate.c:674
> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_RESET ->
> CS_SOFT_EXECUTE
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:120 sofia/external/+
> 14158867717 at 199.173.100.16:5060 receive message [DISPLAY]
> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:120
> sofia/external/14159927717 receive message [DISPLAY]
> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:924
> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
> CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:968 sofia/external/+
> 14158867717 at 199.173.100.16:5060 receive message [BRIDGE]
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:630 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:975
> sofia/external/14159927717 receive message [BRIDGE]
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:630 Send signal
> sofia/external/14159927717 [BREAK]
> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:1019
> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
> CS_EXCHANGE_MEDIA
> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:343
> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK
> 2009-10-22 17:23:26.740065 [DEBUG] switch_core_io.c:234 sofia/external/+
> 14158867717 at 199.173.100.16:5060 receive message [TRANSCODING_NECESSARY]
> 2009-10-22 17:23:27.319618 [DEBUG] switch_core_media_bug.c:393 Removing BUG
> from sofia/external/+14158867717 at 199.173.100.16:5060
> multipler is being overrided to1
> 2009-10-22 17:23:29.406414 [INFO] switch_cpp.cpp:1116 BG_AG_GEN_CALLS,
> Agents total:1 avail:0 OutChan:0 AR:0 +-Calls:0:Q Call NYNoc
> 2009-10-22 17:23:31.873352 [NOTICE] mod_commands.c:1523 Hangup
> sofia/external/14159927717 [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
> 2009-10-22 17:23:31.874411 [INFO] switch_cpp.cpp:1116
> EVENTC_HANGUP_CONTACT, Q_CONTACT fired for agent:1/1 for qcall:1
> contact:4801439/1
> active_true=1
> --contact NOT in playmsg or conference, update AGENT
> 2009-10-22 17:23:31.875475 [DEBUG] switch_channel.c:1810 Send signal
> sofia/external/14159927717 [KILL]
> 2009-10-22 17:23:31.875475 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/14159927717 [BREAK]
> 2009-10-22 17:23:31.875475 [DEBUG] switch_core_state_machine.c:437 thread
> mismatch skipping state handler.
> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:446
> sofia/external/14159927717 ending bridge by request from read function
> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
> DONE [sofia/external/14159927717]
> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:523 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:495 sofia/external/+
> 14158867717 at 199.173.100.16:5060 receive message [UNBRIDGE]
> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_session.c:630 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
> DONE [sofia/external/+14158867717 at 199.173.100.16:5060]
> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:523 Send signal
> sofia/external/14159927717 [BREAK]
> 2009-10-22 17:23:31.882153 [NOTICE] switch_ivr.c:1268 Hangup
> sofia/external/+14158867717 at 199.173.100.16:5060 [CS_EXCHANGE_MEDIA]
> [EXCHANGE_ROUTING_ERROR]
> 2009-10-22 17:23:31.882153 [DEBUG] switch_channel.c:1810 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [KILL]
> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_state_machine.c:464
> (sofia/external/+14158867717 at 199.173.100.16:5060) State HANGUP
> 2009-10-22 17:23:31.882153 [DEBUG] mod_sofia.c:344 Channel sofia/external/+
> 14158867717 at 199.173.100.16:5060 hanging up, cause: EXCHANGE_ROUTING_ERROR
> 2009-10-22 17:23:31.882153 [DEBUG] mod_sofia.c:382 Sending BYE to
> sofia/external/+14158867717 at 199.173.100.16:5060
> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_state_machine.c:46
> sofia/external/+14158867717 at 199.173.100.16:5060 Standard HANGUP, cause:
> EXCHANGE_ROUTING_ERROR
> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_state_machine.c:464
> (sofia/external/+14158867717 at 199.173.100.16:5060) State HANGUP going to
> sleep
> 2009-10-22 17:23:31.884357 [INFO] switch_cpp.cpp:1116 HH_HANGUP, agent:1/1
> uuid: f8604694-4e44-4c4c-9805-722b3e9d6b68
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:346
> (sofia/external/14159927717) State SOFT_EXECUTE going to sleep
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/14159927717) Running State Change CS_HANGUP
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:464
> (sofia/external/14159927717) State HANGUP
> 2009-10-22 17:23:31.885412 [DEBUG] mod_sofia.c:344 Channel
> sofia/external/14159927717 hanging up, cause: NORMAL_CLEARING
> 2009-10-22 17:23:31.885412 [DEBUG] mod_sofia.c:382 Sending BYE to
> sofia/external/14159927717
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:46
> sofia/external/14159927717 Standard HANGUP, cause: NORMAL_CLEARING
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:464
> (sofia/external/14159927717) State HANGUP going to sleep
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:325
> (sofia/external/14159927717) State Change CS_HANGUP -> CS_REPORTING
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/14159927717 [BREAK]
> subtracting 5 seconds for hh_call: 1095423
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/14159927717) Running State Change CS_REPORTING
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:555
> (sofia/external/14159927717) State REPORTING
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:53
> sofia/external/14159927717 Standard REPORTING, cause: NORMAL_CLEARING
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:555
> (sofia/external/14159927717) State REPORTING going to sleep
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:319
> (sofia/external/14159927717) State Change CS_REPORTING -> CS_DESTROY
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/14159927717 [BREAK]
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_session.c:1069 Session 7
> (sofia/external/14159927717) Locked, Waiting on external entities
> 2009-10-22 17:23:31.885412 [NOTICE] switch_core_session.c:1087 Session 7
> (sofia/external/14159927717) Ended
> 2009-10-22 17:23:31.885412 [NOTICE] switch_core_session.c:1089 Close
> Channel sofia/external/14159927717 [CS_DESTROY]
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:401
> (sofia/external/14159927717) Running State Change CS_DESTROY
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:412
> (sofia/external/14159927717) State DESTROY
> 2009-10-22 17:23:31.885412 [DEBUG] mod_sofia.c:261
> sofia/external/14159927717 SOFIA DESTROY
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:60
> sofia/external/14159927717 Standard DESTROY
> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:412
> (sofia/external/14159927717) State DESTROY going to sleep
> 2009-10-22 17:23:31.885412 [INFO] switch_cpp.cpp:1116 PCHangup gw:
> teliax.com hc:NORMAL_CLEARING du:5 cn:sofia/external/14159927717
> 2009-10-22 17:23:31.885412 [INFO] switch_cpp.cpp:1116 EVENTC_BILLING_CEIL,
> subtracting 5 for hh_call:1095423
> 2009-10-22 17:23:31.901620 [DEBUG] switch_core_state_machine.c:494 Hangup
> Command luarun(hh_hangup.lua f8604694-4e44-4c4c-9805-722b3e9d6b68 1 1):
> +OK
>
> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:343
> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
> going to sleep
> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
> CS_HANGUP
> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:442 handler
> already called, skipping state handler.
> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:325
> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_HANGUP
> -> CS_REPORTING
> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:306
> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
> CS_REPORTING
> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:555
> (sofia/external/+14158867717 at 199.173.100.16:5060) State REPORTING
> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:53
> sofia/external/+14158867717 at 199.173.100.16:5060 Standard REPORTING, cause:
> EXCHANGE_ROUTING_ERROR
> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:555
> (sofia/external/+14158867717 at 199.173.100.16:5060) State REPORTING going to
> sleep
> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:319
> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
> CS_REPORTING -> CS_DESTROY
> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_session.c:1069 Session 4
> (sofia/external/+14158867717 at 199.173.100.16:5060) Locked, Waiting on
> external entities
> 2009-10-22 17:23:31.903742 [NOTICE] switch_core_session.c:1087 Session 4
> (sofia/external/+14158867717 at 199.173.100.16:5060) Ended
> 2009-10-22 17:23:31.903742 [NOTICE] switch_core_session.c:1089 Close
> Channel sofia/external/+14158867717 at 199.173.100.16:5060 [CS_DESTROY]
> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:401
> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
> CS_DESTROY
> 2009-10-22 17:23:31.904799 [DEBUG] switch_core_state_machine.c:412
> (sofia/external/+14158867717 at 199.173.100.16:5060) State DESTROY
> 2009-10-22 17:23:31.904799 [DEBUG] mod_sofia.c:261 sofia/external/+
> 14158867717 at 199.173.100.16:5060 SOFIA DESTROY
> 2009-10-22 17:23:31.904799 [DEBUG] switch_core_state_machine.c:60
> sofia/external/+14158867717 at 199.173.100.16:5060 Standard DESTROY
> 2009-10-22 17:23:31.904799 [DEBUG] switch_core_state_machine.c:412
> (sofia/external/+14158867717 at 199.173.100.16:5060) State DESTROY going to
> sleep
>
>
>
> On Tue, Oct 20, 2009 at 1:44 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Sun, Oct 18, 2009 at 9:48 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>>
>>> The debug level logs to the console, right? The pastebin, had log level
>>> debug, sofia trace on for external and default, and sofia loglevel all 9. Is
>>> there another log enable command I'm missing? It seems loglevel all 9
>>> outputs enter and exit functions, but at least to my novice eye, it's not
>>> too obvious why freeswitch is sending a BYE to my DID provider.
>>>
>>> Evidently the console loglevel debug did not take because there isn't a
>> single DEBUG level line of output. Please try again. Just remember that
>> debug level log messages will have A LOT of information so be patient while
>> sifting through it. Of course, feel free to pastebin the output and add a
>> link to it in this email thread.
>>
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From gkuri at ieee.org  Thu Oct 22 12:29:07 2009
From: gkuri at ieee.org (Gabriel Kuri)
Date: Thu, 22 Oct 2009 12:29:07 -0700
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
In-Reply-To: <2d9149cd0910221213h20661e11pdccafdf407c79b02@mail.gmail.com>
References: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>	<191c3a030910201434q4c6e65e1h58c5f0a82addc2d3@mail.gmail.com>	<2d9149cd0910202050i36a7ea82nc7253d87e38f990a@mail.gmail.com>	<191c3a030910211154gc8412b5gd999fbedffa02b53@mail.gmail.com>	<2d9149cd0910220844v2aa73502j4b88fdbd75a0d833@mail.gmail.com>	<65d96fc80910220858u788c6146sced38e19418e681a@mail.gmail.com>	<2d9149cd0910221056y3716d8c4s708fbdb6a5563733@mail.gmail.com>	<4AE0A17D.3000301@ieee.org>
	<2d9149cd0910221213h20661e11pdccafdf407c79b02@mail.gmail.com>
Message-ID: <4AE0B283.3070008@ieee.org>

AFAIK T.38 v2 supports a max speed of 14.4 anyhow, so that's the max
speed you'll ever see in the SDP. T.38 v3 supposedly supports v.34
speeds, however no one that I've seen has implemented it yet - not sure
it's even an official standard?

Yes, in my experience, v.34 capable FAXes do not properly negotiate down
to 14.4 speeds, we've always had to disable v.34 (or whatever the option
is called on your particular make/model to force it down to 14.4 or
slower speeds). We use the Cisco/Linksys ATAs - I've heard there's
better ATAs out there that properly negotiate the FAX down to 14.4 on
the T.30 side of the connection, but never had a chance to play with them.

I'd certainly be interested to hear anyone's experience with T.38
reliability and various combinations of FAX machines and make/model of ATAs.

Gabe

Kristian Kielhofner wrote:
> Gabe,
> 
>   I don't think any of them are plus the T.38 SDP tells me the bitrate
> is 14400, certainly not V.34 speed.
> 
>   Are you saying the machine even trying to negotiate V.34 poses a problem?
> 
> On Thu, Oct 22, 2009 at 2:16 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>> Out of curiosity, is it a Super G3 (ie v.34) capable FAX? We've had
>> nothing but intermittent problems with Super G3 FAXes over T.38, unless
>> v.34 is strictly turned off on the machine.
>>
>> Gabe
>>
> 




From rupa at rupa.com  Thu Oct 22 12:31:58 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 22 Oct 2009 14:31:58 -0500
Subject: [Freeswitch-users] Address Rupa: Database for Audio Data
In-Reply-To: <BAY119-W4307982F0031E8BE965E54B3BE0@phx.gbl>
References: <BAY119-W52BD912544FAA1DE624B88B3C70@phx.gbl>
	<b39273530910130857jb7dec2eidd71e950d3902b25@mail.gmail.com>
	<BAY119-W4307982F0031E8BE965E54B3BE0@phx.gbl>
Message-ID: <b39273530910221231u93d3560nbb27036620c46e9b@mail.gmail.com>

I use the dialplan app session_record:

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session

I call that in the appropriate parts before bridging to call.  For
incoming it is just before the bridge to user/username.  For outgoing
it is before I bridge to the outgoing provider.

On Thu, Oct 22, 2009 at 11:30 AM, Pajongjit Buntaokit
<pippyduck1127 at hotmail.com> wrote:
> Hi Rupa,
>
> Thanks again for your advice.
>
> I have been searching for the method to record in the freeswitch
> documentation but I'm still not sure which command and method to make the
> record for every call automatically.
>
> Which command or method do you use?
> And to make the recording start and stop automatically every time when the
> calls is started and end, where should I insert this command?
>
> Did you use the Mod commands 'uuid_record'?
> If so, where to place this commands?
> Please show me some clues?
>
> Thank you very much!
>
> PB
>
>
>
>
>> Date: Tue, 13 Oct 2009 09:57:05 -0600
>> From: rupa at rupa.comajongjit Buntaokit
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Database for Audio Data
>>
>> What I do is record all calls and store the call with the UUID as the
>> filename. Then when the call is hung up a CDR entry is sent to my web
>> server. This CDR contains callerid and other info I might want to
>> query by. The service on the web server inserts appropriate record(s)
>> into the database. The recordings are available to the webserver.
>> When one clicks on the "listen" link, the web server serves up the
>> recording by UUID in the recording directory. I have a process that
>> periodically removes old recordings from that dir. I don't purge the
>> CDRs, though that is certainly possible.
>>
>> On Tue, Oct 13, 2009 at 7:58 AM, Pajongjit Buntaokit
>> <pippyduck1127 at hotmail.com> wrote:
>> > Hi,
>> >
>> > Does anyone know whether FreeSWITCH has a function to automatically
>> > record
>> > every call as an audio file in a server
>> > or forward them to be stored in a database with additional parameters
>> > such
>> > as caller ID, date, starting time and ending time?
>> >
>> > So that these recorded audio data can be queried and retrieved with the
>> > caller ID, date and time.
>> >
>> > Any suggestion or guidance, please advise.
>> >
>> > Thank you very much!
>> >
>> > ________________________________
>> > Hotmail: Trusted email with Microsoft?s powerful SPAM protection. Sign
>> > up
>> > now.
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> ________________________________
> Windows 7: It works the way you want. Learn more.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa



From abovebluesky at gmail.com  Thu Oct 22 12:35:16 2009
From: abovebluesky at gmail.com (Uncle Johny)
Date: Thu, 22 Oct 2009 12:35:16 -0700 (PDT)
Subject: [Freeswitch-users]  3 way conferencing question
Message-ID: <26015903.post@talk.nabble.com>


Hello gents,

I know that it is possible to make a 3-way conferencing using att_xfer and
by pressing 0.
But I'm more interested in the way of doing 3-way conferencing using mod
conference and bind_app. Saying in other words, I want to do the same thing
what 3-way conferencing using att_xfer but with mod conference (eventually
it will be more then 3 people there)

  <extension name="mystuff_enum">
    <condition field="destination_number" expression="^(.*)$">
                <action application="answer"/>
                <action application="bind_meta_app" data="7 ab s
execute_extension::conference XML features"/>
                <action application="set" data="hangup_after_bridge=true"/>
                <action application="bridge"
data="sofia/external/${destination_number}@$${dialoutserver}"/>
    </condition>
  </extension>

in features.xml :
<extension name="conference">
     <condition field="destination_number" expression="^conference$">
        <action application="set" data="continue_on_fail=true"/>
        <action application="read" data="3 10
/usr/local/freeswitch/sounds/en/us/callie/conference/8000/conf-welcome.wav
thirdperson 30000 #"/>
       <!-- I guess some logic here which will put original caller, person
whom we dialed and thirdperson(numbers which we pressed) into a conference
room with unique id -->
     </condition>
</extension>

I will be glad to hear any ideas.

-- 
View this message in context: http://www.nabble.com/3-way-conferencing-question-tp26015903p26015903.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From tculjaga at gmail.com  Thu Oct 22 12:44:02 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 22 Oct 2009 21:44:02 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>
	<65d96fc80910220753n6d2efa39l470791f75db9a640@mail.gmail.com>
	<Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>
Message-ID: <65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>

2009/10/22 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-22 16:53 +0200, Tihomir Culjaga wrote
> freeswitch-users at lists.fre...:
>
> finally i fix this rtp bug, check new wersion please.
>
>
if course i can do that, but tomorrow morning ... i'm not in the office
anymore.
BTW: can we please move the tickets to jira?


it is gonna be easier to track.

Tomorrow i will test on 1.0.4 but please lets move to trunk.

T.
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From ujjval at simplesignal.com  Thu Oct 22 12:45:12 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Thu, 22 Oct 2009 12:45:12 -0700
Subject: [Freeswitch-users] Core Dump  question!
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701E840EB@EXMBXCLUS01.citservers.local>


freeswitch at ss_freeswitch> sofia_gateway_data
Segmentation fault (core dumped)


Just ran the gateway command above w/o any parameters,,, and it core dumped..

I am sure mistakes like that happen...but I not sure if it should core dump  and shutdown.....



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From ujjval at simplesignal.com  Thu Oct 22 12:53:56 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Thu, 22 Oct 2009 12:53:56 -0700
Subject: [Freeswitch-users] Connecting to FS CLI...just hangs..
In-Reply-To: <ebbc2d550910221211k18171473p7f4d6f597f6b21d2@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84092@EXMBXCLUS01.citservers.local>
	<ebbc2d550910221211k18171473p7f4d6f597f6b21d2@mail.gmail.com>
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701E840F9@EXMBXCLUS01.citservers.local>


If I run .freeswitch& ,  get back to the ROOT prompt and then from same window type in fs_cli...it fails...hangs forever

However, if I open another new ssh session and fs_cli from the new session, it works.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jo?o Mesquita
Sent: Thursday, October 22, 2009 1:11 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Connecting to FS CLI...just hangs..

Hangs for how long? Are you sure you are not just waiting on a timeout?

JM
On Thu, Oct 22, 2009 at 4:42 PM, Ujjval Karihaloo <ujjval at simplesignal.com<mailto:ujjval at simplesignal.com>> wrote:
It just hangs....and I CTRL-C out of it.

[root at ss]# ./fs_cli -H 127.0.0.1


^C
[ERROR] libs/esl/fs_cli.c:652 main() Error Connecting [Connection Error]


Freeswitch process is running:

[root at ss bin]# ps -ef|grep free
root      8889 31039  0 12:36 pts/2    00:00:00 ./freeswitch
root      8952 31039  0 12:42 pts/2    00:00:00 grep free



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From msc at freeswitch.org  Thu Oct 22 12:58:26 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 22 Oct 2009 12:58:26 -0700
Subject: [Freeswitch-users] Core Dump question!
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71701E840EB@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701E840EB@EXMBXCLUS01.citservers.local>
Message-ID: <87f2f3b90910221258o6936000dn6eb2e22189c45435@mail.gmail.com>

What SVN rev of FS? What operating system? If you're not on the latest then
do a "make current" and get to the latest SVN and see if you can replicate
the issue.

-MC

On Thu, Oct 22, 2009 at 12:45 PM, Ujjval Karihaloo
<ujjval at simplesignal.com>wrote:

>
>
> freeswitch at ss_freeswitch> sofia_gateway_data
>
> Segmentation fault (core dumped)
>
>
>
>
>
> Just ran the gateway command above w/o any parameters,,, and it core
> dumped..
>
>
>
> I am sure mistakes like that happen?but I not sure if it should core dump
> and shutdown?..
>
>
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From ujjval at simplesignal.com  Thu Oct 22 13:00:21 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Thu, 22 Oct 2009 13:00:21 -0700
Subject: [Freeswitch-users] FS Registration Contact
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701E84100@EXMBXCLUS01.citservers.local>

Hi All,

    I have FS registered to an ITSP. The contact is showing as follows..

Contact: sip:gw+itsp at 1.1.1.1:5080;transport=udp

Itsp is the name of the SIP gatway and its IP is changed to 1.1.1.1

I want the Phone number (FromUser)to show in the contact header in the REGISTER msg going to the ITSP.

How can I do that?


Thx,
Ujjval.
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From bottleman at icf.org.ru  Thu Oct 22 13:07:09 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Fri, 23 Oct 2009 00:07:09 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910211917320.4450@srv.icf.org.ru>
	<65d96fc80910210848p4dc2f8f3k574ef57140af489a@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>
	<65d96fc80910220753n6d2efa39l470791f75db9a640@mail.gmail.com>
	<Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>
	<65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>

On 2009-10-22 21:44 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>2009/10/22 Georgiewskiy Yuriy <bottleman at icf.org.ru>
TC>
TC>> On 2009-10-22 16:53 +0200, Tihomir Culjaga wrote
TC>> freeswitch-users at lists.fre...:
TC>>
TC>> finally i fix this rtp bug, check new wersion please.
TC>>
TC>>
TC>if course i can do that, but tomorrow morning ... i'm not in the office
TC>anymore.
TC>BTW: can we please move the tickets to jira?
TC>
TC>
TC>it is gonna be easier to track.
TC>
TC>Tomorrow i will test on 1.0.4 but please lets move to trunk.

i make it a bit later, to move tickets to jira and source to svn i 
need some time to undertand how this system is works, especially jira.

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From william.suffill at gmail.com  Thu Oct 22 13:13:15 2009
From: william.suffill at gmail.com (William Suffill)
Date: Thu, 22 Oct 2009 16:13:15 -0400
Subject: [Freeswitch-users] Connecting to FS CLI...just hangs..
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71701E840F9@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84092@EXMBXCLUS01.citservers.local>
	<ebbc2d550910221211k18171473p7f4d6f597f6b21d2@mail.gmail.com> 
	<3C04B27FC880044F8FCD735D0D952FF71701E840F9@EXMBXCLUS01.citservers.local>
Message-ID: <6b65470d0910221313i2e87d8f1h2a7b00c98c0089fa@mail.gmail.com>

If it's not windows probably be safer to just do freeswitch -nc  for
no console.  Give it a little to start up then fs_cli should be fine.
netstat -anp can also be used to see that the ports have binded
correctly.

-- W



From tculjaga at gmail.com  Thu Oct 22 13:20:00 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 22 Oct 2009 22:20:00 +0200
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
In-Reply-To: <4AE0B283.3070008@ieee.org>
References: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>
	<191c3a030910201434q4c6e65e1h58c5f0a82addc2d3@mail.gmail.com>
	<2d9149cd0910202050i36a7ea82nc7253d87e38f990a@mail.gmail.com>
	<191c3a030910211154gc8412b5gd999fbedffa02b53@mail.gmail.com>
	<2d9149cd0910220844v2aa73502j4b88fdbd75a0d833@mail.gmail.com>
	<65d96fc80910220858u788c6146sced38e19418e681a@mail.gmail.com>
	<2d9149cd0910221056y3716d8c4s708fbdb6a5563733@mail.gmail.com>
	<4AE0A17D.3000301@ieee.org>
	<2d9149cd0910221213h20661e11pdccafdf407c79b02@mail.gmail.com>
	<4AE0B283.3070008@ieee.org>
Message-ID: <65d96fc80910221320y1512c1f9xb23f734adc6b2972@mail.gmail.com>

well, my experience with T.38 is that we switched it off within our network
and now use passthrough instead :P.

the reason:

   - new fax machines that defaults to v.34 and on some you cannot disable
   SG3 at all... just imagine what pain it is to produce a how to disable V.34
   manual for every damn fax machine
   - a variety of T.30 <> T.38 mapping issues, broken GWs/ATAs that don't do
   a correct encoding/decoding expectantly when it comes to a terminating fax
   server that doesn't send a proper EOP... God save our souls...
   - our network has very low jitter - up to 40 ms , so passthrough works
   perfect
   - bandwidth become cheap :=)
   - and ... don't even remember any fax issues after shutdown T.38

i could go with the list... but i suppose you have better things to do  :P


T.

On Thu, Oct 22, 2009 at 9:29 PM, Gabriel Kuri <gkuri at ieee.org> wrote:

> AFAIK T.38 v2 supports a max speed of 14.4 anyhow, so that's the max
> speed you'll ever see in the SDP. T.38 v3 supposedly supports v.34
> speeds, however no one that I've seen has implemented it yet - not sure
> it's even an official standard?
>
> Yes, in my experience, v.34 capable FAXes do not properly negotiate down
> to 14.4 speeds, we've always had to disable v.34 (or whatever the option
> is called on your particular make/model to force it down to 14.4 or
> slower speeds). We use the Cisco/Linksys ATAs - I've heard there's
> better ATAs out there that properly negotiate the FAX down to 14.4 on
> the T.30 side of the connection, but never had a chance to play with them.
>
> I'd certainly be interested to hear anyone's experience with T.38
> reliability and various combinations of FAX machines and make/model of
> ATAs.
>
> Gabe
>
> Kristian Kielhofner wrote:
> > Gabe,
> >
> >   I don't think any of them are plus the T.38 SDP tells me the bitrate
> > is 14400, certainly not V.34 speed.
> >
> >   Are you saying the machine even trying to negotiate V.34 poses a
> problem?
> >
> > On Thu, Oct 22, 2009 at 2:16 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
> >> Out of curiosity, is it a Super G3 (ie v.34) capable FAX? We've had
> >> nothing but intermittent problems with Super G3 FAXes over T.38, unless
> >> v.34 is strictly turned off on the machine.
> >>
> >> Gabe
> >>
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ujjval at simplesignal.com  Thu Oct 22 13:34:41 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Thu, 22 Oct 2009 13:34:41 -0700
Subject: [Freeswitch-users] FS Registration Contact
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71701E84100@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84100@EXMBXCLUS01.citservers.local>
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701E8412B@EXMBXCLUS01.citservers.local>


Ok I got this one...just put ext-in-contact setting and then define the extension to be same as FromUser in my provider.xml in /conf/sip-profile/external/

  <!--/// extension for inbound calls: *optional* same as username, if blank ///-->
  <param name="extension" value="xxxxxxxxxx"/>
  <!--extra sip params to send in the contact-->
  <param name="extension-in-contact" value="true"/>

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ujjval Karihaloo
Sent: Thursday, October 22, 2009 2:00 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] FS Registration Contact

Hi All,

    I have FS registered to an ITSP. The contact is showing as follows..

Contact: sip:gw+itsp at 1.1.1.1:5080;transport=udp

Itsp is the name of the SIP gatway and its IP is changed to 1.1.1.1

I want the Phone number (FromUser)to show in the contact header in the REGISTER msg going to the ITSP.

How can I do that?


Thx,
Ujjval.
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From peder at networkoblivion.com  Thu Oct 22 13:52:03 2009
From: peder at networkoblivion.com (Peder)
Date: Thu, 22 Oct 2009 15:52:03 -0500
Subject: [Freeswitch-users] Proxy media mode with T.38 re-invite
In-Reply-To: <4AE0B283.3070008@ieee.org>
References: <2d9149cd0910201357r43646050s66d7dd3e64c1658@mail.gmail.com>	<191c3a030910201434q4c6e65e1h58c5f0a82addc2d3@mail.gmail.com>	<2d9149cd0910202050i36a7ea82nc7253d87e38f990a@mail.gmail.com>	<191c3a030910211154gc8412b5gd999fbedffa02b53@mail.gmail.com>	<2d9149cd0910220844v2aa73502j4b88fdbd75a0d833@mail.gmail.com>	<65d96fc80910220858u788c6146sced38e19418e681a@mail.gmail.com>	<2d9149cd0910221056y3716d8c4s708fbdb6a5563733@mail.gmail.com>	<4AE0A17D.3000301@ieee.org>	<2d9149cd0910221213h20661e11pdccafdf407c79b02@mail.gmail.com>
	<4AE0B283.3070008@ieee.org>
Message-ID: <029a01ca5359$823e0da0$86ba28e0$@com>

We have probably 30-40 fax machines running T.38 between Linksys SPA-3102
and Cisco gateways.  We have a pretty good success rate and we have been
doing this for probably 2-3 years.  We have a couple of them going thru an *
1.4 box and it seems to work ok.  One of my projects is to try and get them
all converted to go thru an FS box, I just haven't had time.  Our biggest
issues have always been error correction.  Once we disable that on the fax
machine, it seems to work pretty good.  As a general rule, we have found
that older cheaper machines work better.  New fancy machines are unreliable
and that could certainly be related to v.34 issues, but we haven't bothered
to get that into it to find out.

Peder

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gabriel
Kuri
Sent: Thursday, October 22, 2009 2:29 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Proxy media mode with T.38 re-invite

AFAIK T.38 v2 supports a max speed of 14.4 anyhow, so that's the max
speed you'll ever see in the SDP. T.38 v3 supposedly supports v.34
speeds, however no one that I've seen has implemented it yet - not sure
it's even an official standard?

Yes, in my experience, v.34 capable FAXes do not properly negotiate down
to 14.4 speeds, we've always had to disable v.34 (or whatever the option
is called on your particular make/model to force it down to 14.4 or
slower speeds). We use the Cisco/Linksys ATAs - I've heard there's
better ATAs out there that properly negotiate the FAX down to 14.4 on
the T.30 side of the connection, but never had a chance to play with them.

I'd certainly be interested to hear anyone's experience with T.38
reliability and various combinations of FAX machines and make/model of ATAs.

Gabe

Kristian Kielhofner wrote:
> Gabe,
> 
>   I don't think any of them are plus the T.38 SDP tells me the bitrate
> is 14400, certainly not V.34 speed.
> 
>   Are you saying the machine even trying to negotiate V.34 poses a
problem?
> 
> On Thu, Oct 22, 2009 at 2:16 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>> Out of curiosity, is it a Super G3 (ie v.34) capable FAX? We've had
>> nothing but intermittent problems with Super G3 FAXes over T.38, unless
>> v.34 is strictly turned off on the machine.
>>
>> Gabe
>>
> 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From mike at jerris.com  Thu Oct 22 13:53:11 2009
From: mike at jerris.com (Michael Jerris)
Date: Thu, 22 Oct 2009 16:53:11 -0400
Subject: [Freeswitch-users] Core Dump question!
In-Reply-To: <87f2f3b90910221258o6936000dn6eb2e22189c45435@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701E840EB@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221258o6936000dn6eb2e22189c45435@mail.gmail.com>
Message-ID: <DF11CCC6-5C89-4489-908C-92A0FCC94AE4@jerris.com>

I did test this on trunk and it seems to work right:

freeswitch at default> sofia_gateway_data
-ERR Parameter missing

Mike

On Oct 22, 2009, at 3:58 PM, Michael Collins wrote:

> What SVN rev of FS? What operating system? If you're not on the  
> latest then do a "make current" and get to the latest SVN and see if  
> you can replicate the issue.
>
> -MC
>
> On Thu, Oct 22, 2009 at 12:45 PM, Ujjval Karihaloo <ujjval at simplesignal.com 
> > wrote:
>
> freeswitch at ss_freeswitch> sofia_gateway_data
>
> Segmentation fault (core dumped)
>
>
>
> Just ran the gateway command above w/o any parameters,,, and it core  
> dumped..
>
>
> I am sure mistakes like that happen?but I not sure if it should core  
> dump  and shutdown?..
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From ujjval at simplesignal.com  Thu Oct 22 17:04:39 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Thu, 22 Oct 2009 17:04:39 -0700
Subject: [Freeswitch-users] Core Dump question!
In-Reply-To: <DF11CCC6-5C89-4489-908C-92A0FCC94AE4@jerris.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701E840EB@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221258o6936000dn6eb2e22189c45435@mail.gmail.com>
	<DF11CCC6-5C89-4489-908C-92A0FCC94AE4@jerris.com>
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701E841E8@EXMBXCLUS01.citservers.local>

I do have the core dump, should I open a ticket.
I am running latest Freeswitch 1.0.4 and had done a make current just before it happened.

Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690

SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO  80112
[cid:image001.jpg at 01CA5342.1F8A8880]<http://www.simplesignal.com/>

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Thursday, October 22, 2009 2:53 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Core Dump question!

I did test this on trunk and it seems to work right:

freeswitch at default> sofia_gateway_data
-ERR Parameter missing

Mike

On Oct 22, 2009, at 3:58 PM, Michael Collins wrote:


What SVN rev of FS? What operating system? If you're not on the latest then do a "make current" and get to the latest SVN and see if you can replicate the issue.

-MC
On Thu, Oct 22, 2009 at 12:45 PM, Ujjval Karihaloo <ujjval at simplesignal.com<mailto:ujjval at simplesignal.com>> wrote:

freeswitch at ss_freeswitch> sofia_gateway_data
Segmentation fault (core dumped)


Just ran the gateway command above w/o any parameters,,, and it core dumped..

I am sure mistakes like that happen...but I not sure if it should core dump  and shutdown.....


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From msc at freeswitch.org  Thu Oct 22 17:25:34 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 22 Oct 2009 17:25:34 -0700
Subject: [Freeswitch-users] Core Dump question!
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71701E841E8@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701E840EB@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221258o6936000dn6eb2e22189c45435@mail.gmail.com>
	<DF11CCC6-5C89-4489-908C-92A0FCC94AE4@jerris.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E841E8@EXMBXCLUS01.citservers.local>
Message-ID: <87f2f3b90910221725g6233ec20w49a73864400426d6@mail.gmail.com>

Yes, if this is latest SVN (after a make current) then open a jira.
-MC

On Thu, Oct 22, 2009 at 5:04 PM, Ujjval Karihaloo
<ujjval at simplesignal.com>wrote:

>  I do have the core dump, should I open a ticket.
>
> I am running latest Freeswitch 1.0.4 and had done a make current just
> before it happened.
>
>
>
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From ujjval at simplesignal.com  Thu Oct 22 19:52:12 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Thu, 22 Oct 2009 19:52:12 -0700
Subject: [Freeswitch-users] Core Dump question!
In-Reply-To: <87f2f3b90910221725g6233ec20w49a73864400426d6@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701E840EB@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221258o6936000dn6eb2e22189c45435@mail.gmail.com>
	<DF11CCC6-5C89-4489-908C-92A0FCC94AE4@jerris.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E841E8@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221725g6233ec20w49a73864400426d6@mail.gmail.com>
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701E84200@EXMBXCLUS01.citservers.local>

How do I tell if it's the latest...I downloaded is yesterday..and installed it from freeswitch.org

Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690

SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO  80112
[cid:image001.jpg at 01CA5359.87CB41C0]<http://www.simplesignal.com/>

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Thursday, October 22, 2009 6:26 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Core Dump question!

Yes, if this is latest SVN (after a make current) then open a jira.
-MC
On Thu, Oct 22, 2009 at 5:04 PM, Ujjval Karihaloo <ujjval at simplesignal.com<mailto:ujjval at simplesignal.com>> wrote:
I do have the core dump, should I open a ticket.
I am running latest Freeswitch 1.0.4 and had done a make current just before it happened.


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From ujjval at simplesignal.com  Thu Oct 22 20:01:57 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Thu, 22 Oct 2009 20:01:57 -0700
Subject: [Freeswitch-users] Setting up Conference with Moderator
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local>

Hi

I have the Basic Conferencing working. Here is my Dial Plan.

 I want to be able to setup a Moderator PIN different from other participants, when I add the moderator flag it logs me in directly w/o asking for a PIN..
action application="conference" data="conference.conf at wideband+flags{moderator}+159753<mailto:conference.conf at wideband+flags%7bmoderator%7d+159753>"/>

DialPlan is below for the normal user, and it asks for the PIN with below settings.

       Ujj Inbound from SS - start
    -->
    <extension name="simplesignal">   <!-- your provider or any name you'd like to call it -->
     <condition field="destination_number" expression="2142349127">  <!-- your DID for this gateway-->
        <action application="conference" data="conference.conf at wideband+159753<mailto:conference.conf at wideband+159753>"/>
     </condition>
</extension>

    <!--
       Ujj Inbound from SS - end




And I am using the existing conference.conf.xml file in the auto_loads directory.


Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690

SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO  80112
[cid:image001.jpg at 01CA535A.E43E9870]<http://www.simplesignal.com/>

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From diego.viola at gmail.com  Thu Oct 22 20:04:24 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 23 Oct 2009 03:04:24 +0000
Subject: [Freeswitch-users] Core Dump question!
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71701E84200@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701E840EB@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221258o6936000dn6eb2e22189c45435@mail.gmail.com>
	<DF11CCC6-5C89-4489-908C-92A0FCC94AE4@jerris.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E841E8@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221725g6233ec20w49a73864400426d6@mail.gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E84200@EXMBXCLUS01.citservers.local>
Message-ID: <86a32abc0910222004y46bf08b6jda75b6b8f1979420@mail.gmail.com>

Type "version" on the CLI.

On Fri, Oct 23, 2009 at 2:52 AM, Ujjval Karihaloo
<ujjval at simplesignal.com>wrote:

>  How do I tell if it?s the latest?I downloaded is yesterday..and installed
> it from freeswitch.org
>
>
>
> Ujjval Karihaloo
>
> VP Voice Engineering
>
> IP Phone: +13032428610
>
> E-Fax: +17202391690
>
>
>
> SimpleSignal Inc.
>
> 88 Inverness Circle East
>
> Suite K105
>
> Englewood, CO  80112
>
> [image: bvoip] <http://www.simplesignal.com/>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, October 22, 2009 6:26 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Core Dump question!
>
>
>
> Yes, if this is latest SVN (after a make current) then open a jira.
> -MC
>
> On Thu, Oct 22, 2009 at 5:04 PM, Ujjval Karihaloo <ujjval at simplesignal.com>
> wrote:
>
> I do have the core dump, should I open a ticket.
>
> I am running latest Freeswitch 1.0.4 and had done a make current just
> before it happened.
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mcampbellsmith at gmail.com  Thu Oct 22 20:20:21 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Fri, 23 Oct 2009 14:20:21 +1100
Subject: [Freeswitch-users] Core Dump question!
In-Reply-To: <86a32abc0910222004y46bf08b6jda75b6b8f1979420@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701E840EB@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221258o6936000dn6eb2e22189c45435@mail.gmail.com>
	<DF11CCC6-5C89-4489-908C-92A0FCC94AE4@jerris.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E841E8@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221725g6233ec20w49a73864400426d6@mail.gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E84200@EXMBXCLUS01.citservers.local>
	<86a32abc0910222004y46bf08b6jda75b6b8f1979420@mail.gmail.com>
Message-ID: <33c87fa30910222020s4bd03f5flec7bc7df3ca6876c@mail.gmail.com>

If you downloaded a precomipled version from the web, you do not (probably)
have the latest version.

As Diego said, type 'version' (or press F12) on the cli and compare to the
latest trunk at http://svn.freeswitch.org/svn/freeswitch/trunk/


On Fri, Oct 23, 2009 at 2:04 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Type "version" on the CLI.
>
> On Fri, Oct 23, 2009 at 2:52 AM, Ujjval Karihaloo <ujjval at simplesignal.com
> > wrote:
>
>>  How do I tell if it?s the latest?I downloaded is yesterday..and
>> installed it from freeswitch.org
>>
>>
>>
>> Ujjval Karihaloo
>>
>> VP Voice Engineering
>>
>> IP Phone: +13032428610
>>
>> E-Fax: +17202391690
>>
>>
>>
>> SimpleSignal Inc.
>>
>> 88 Inverness Circle East
>>
>> Suite K105
>>
>> Englewood, CO  80112
>>
>> [image: bvoip] <http://www.simplesignal.com/>
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
>> Collins
>> *Sent:* Thursday, October 22, 2009 6:26 PM
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* Re: [Freeswitch-users] Core Dump question!
>>
>>
>>
>> Yes, if this is latest SVN (after a make current) then open a jira.
>> -MC
>>
>> On Thu, Oct 22, 2009 at 5:04 PM, Ujjval Karihaloo <
>> ujjval at simplesignal.com> wrote:
>>
>> I do have the core dump, should I open a ticket.
>>
>> I am running latest Freeswitch 1.0.4 and had done a make current just
>> before it happened.
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ujjval at simplesignal.com  Thu Oct 22 20:25:24 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Thu, 22 Oct 2009 20:25:24 -0700
Subject: [Freeswitch-users] Core Dump question!
In-Reply-To: <86a32abc0910222004y46bf08b6jda75b6b8f1979420@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701E840EB@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221258o6936000dn6eb2e22189c45435@mail.gmail.com>
	<DF11CCC6-5C89-4489-908C-92A0FCC94AE4@jerris.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E841E8@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221725g6233ec20w49a73864400426d6@mail.gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E84200@EXMBXCLUS01.citservers.local>
	<86a32abc0910222004y46bf08b6jda75b6b8f1979420@mail.gmail.com>
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701E84203@EXMBXCLUS01.citservers.local>

freeswitch at internal> version
FreeSWITCH Version 1.0.4 (exported)

freeswitch at internal>

Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690

SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO  80112
[cid:image001.jpg at 01CA535E.2ACD2880]<http://www.simplesignal.com/>

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Diego Viola
Sent: Thursday, October 22, 2009 9:04 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Core Dump question!

Type "version" on the CLI.
On Fri, Oct 23, 2009 at 2:52 AM, Ujjval Karihaloo <ujjval at simplesignal.com<mailto:ujjval at simplesignal.com>> wrote:
How do I tell if it's the latest...I downloaded is yesterday..and installed it from freeswitch.org<http://freeswitch.org>

Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690

SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO  80112
[cid:image001.jpg at 01CA535E.2ACD2880]<http://www.simplesignal.com/>

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Collins
Sent: Thursday, October 22, 2009 6:26 PM

To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Core Dump question!

Yes, if this is latest SVN (after a make current) then open a jira.
-MC
On Thu, Oct 22, 2009 at 5:04 PM, Ujjval Karihaloo <ujjval at simplesignal.com<mailto:ujjval at simplesignal.com>> wrote:
I do have the core dump, should I open a ticket.
I am running latest Freeswitch 1.0.4 and had done a make current just before it happened.



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From mattdfong at gmail.com  Thu Oct 22 20:19:07 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Fri, 23 Oct 2009 10:19:07 +0700
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
In-Reply-To: <191c3a030910221226j167e0adw5f631f759cd7cfa0@mail.gmail.com>
References: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
	<177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>
	<4256bf830908041151l55980e3fifb85c75b87535426@mail.gmail.com>
	<3D0B855A-5B2C-4C6F-9306-64B0ECE97091@jerris.com>
	<4256bf830910180725y72539d98gb138934afc18d58b@mail.gmail.com>
	<A5ECDCD9-1A12-4A31-89F0-E6F4C79FE9E9@jerris.com>
	<4256bf830910182148mf7f2bf9vdcccd687e0caa1e7@mail.gmail.com>
	<87f2f3b90910191144q25750bayfd7a7448e82e78ff@mail.gmail.com>
	<4256bf830910221042l68ea769cg532bbd339bae0790@mail.gmail.com>
	<191c3a030910221226j167e0adw5f631f759cd7cfa0@mail.gmail.com>
Message-ID: <4256bf830910222019x4bd7929eoadcb757294cc37d2@mail.gmail.com>

Hi Anthm,
Thanks for the response. I'll use pastebin from now on.

How would I go about fixing this so my calls do not get hung-up?

--matt

On Fri, Oct 23, 2009 at 2:26 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> 1) please do not flood emails with traces reference pastebins
> 2) the error is a routing loop being prevented so you are looping the call
> into your same box too many times.
>
> You could have figured this out your self by looking at one of the lines of
> code where it reported the hangup.
>
>  if (!switch_strlen_zero(forwardvar)) {
>         forwardval = atoi(forwardvar) - 1;
>     }
>     if (forwardval <= 0) {
>         switch_channel_hangup(channel,
> SWITCH_CAUSE_EXCHANGE_ROUTING_ERROR);
>         return SWITCH_STATUS_FALSE;
>
>     }
>
>
> On Thu, Oct 22, 2009 at 12:42 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>
>> I figured out how to turn debug mode on. For some reason F8 didn't work,
>> so I just set the default debug mode to DEBUG in switch.conf.xml
>>
>> http://pastebin.freeswitch.org/10805
>> <http://pastebin.freeswitch.org/10805>So my program does more or less the
>> same sequence for an Agent every time when connected to another party. They
>> uuid_kill the bridged parties uuid, and then are transfered to an extension
>> that park's the agent (using the transfer_after_bridge variable -- in my
>> dialplan extn 1998). In some instances the agent is then transfered to
>> another extension, played a wav file, and then park'ed again
>>
>> Sometimes, FreeSWITCH unexpectidly hangsup the Agent channel citing an
>> EXCHANGE_ROUTING_ERROR. It seems (to me to happen at random times)...and
>> it's more prevelent with some did providers than others. With
>> didforsale.org it happens about one ever 5 bridges. With icall.com it
>> only happens about once every 30 bridges. With IPKall it happens rarely. I
>> can't figure it out, and it's driving me crazy :)
>>
>>
>> A GOOD UUID_KILL SEQUENCE:
>> 2009-10-22 17:22:31.741845 [DEBUG] switch_ivr_play_say.c:1432 done playing
>> file
>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:340
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXECUTE going to
>> sleep
>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>> CS_HIBERNATE
>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:355
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE
>> 2009-10-22 17:22:31.741845 [DEBUG] mod_sofia.c:160 sofia/external/+
>> 14158867717 at 199.173.100.16:5060 SOFIA HIBERNATE
>> 2009-10-22 17:22:31.741845 [DEBUG] switch_ivr_bridge.c:650
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>> CS_HIBERNATE -> CS_RESET
>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:355
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE going
>> to sleep
>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>> CS_RESET
>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:336
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET
>> 2009-10-22 17:22:31.741845 [DEBUG] mod_sofia.c:145 sofia/external/+
>> 14158867717 at 199.173.100.16:5060 SOFIA RESET
>> 2009-10-22 17:22:31.741845 [DEBUG] switch_ivr_bridge.c:637
>> sofia/external/+14158867717 at 199.173.100.16:5060 CUSTOM RESET
>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:67
>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard RESET
>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:336
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET going to
>> sleep
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:340
>> (sofia/external/14159927717) State EXECUTE going to sleep
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/14159927717) Running State Change CS_HIBERNATE
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:355
>> (sofia/external/14159927717) State HIBERNATE
>> 2009-10-22 17:22:31.748086 [DEBUG] mod_sofia.c:160
>> sofia/external/14159927717 SOFIA HIBERNATE
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:650
>> (sofia/external/14159927717) State Change CS_HIBERNATE -> CS_RESET
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/14159927717 [BREAK]
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:355
>> (sofia/external/14159927717) State HIBERNATE going to sleep
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/14159927717) Running State Change CS_RESET
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:336
>> (sofia/external/14159927717) State RESET
>> 2009-10-22 17:22:31.748086 [DEBUG] mod_sofia.c:145
>> sofia/external/14159927717 SOFIA RESET
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:637
>> sofia/external/14159927717 CUSTOM RESET
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:642
>> (sofia/external/14159927717) State Change CS_RESET -> CS_SOFT_EXECUTE
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/14159927717 [BREAK]
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:336
>> (sofia/external/14159927717) State RESET going to sleep
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/14159927717) Running State Change CS_SOFT_EXECUTE
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:346
>> (sofia/external/14159927717) State SOFT_EXECUTE
>> 2009-10-22 17:22:31.748086 [DEBUG] mod_sofia.c:442 SOFIA TRANSMIT
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:660
>> sofia/external/14159927717 CUSTOM SOFT_EXECUTE
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_originate.c:674
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_RESET
>> -> CS_SOFT_EXECUTE
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:120
>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message [DISPLAY]
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:120
>> sofia/external/14159927717 receive message [DISPLAY]
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:924
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>> CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:968
>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message [BRIDGE]
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:630 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:975
>> sofia/external/14159927717 receive message [BRIDGE]
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:630 Send signal
>> sofia/external/14159927717 [BREAK]
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:1019
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:22:31.749553 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>> CS_EXCHANGE_MEDIA
>> 2009-10-22 17:22:31.749553 [DEBUG] switch_core_state_machine.c:343
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
>> 2009-10-22 17:22:31.749553 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK
>> multipler is being overrided to1
>> 2009-10-22 17:22:34.300662 [INFO] switch_cpp.cpp:1116 BG_AG_GEN_CALLS,
>> Agents total:1 avail:0 OutChan:0 AR:0 +-Calls:0:Q Call NYNoc
>> multipler is being overrided to1
>> 2009-10-22 17:22:39.303041 [INFO] switch_cpp.cpp:1116 BG_AG_GEN_CALLS,
>> Agents total:1 avail:0 OutChan:0 AR:0 +-Calls:0:Q Call NYNoc
>> 2009-10-22 17:22:40.626379 [NOTICE] sofia.c:328 Hangup
>> sofia/external/14159927717 [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
>> 2009-10-22 17:22:40.626379 [DEBUG] switch_channel.c:1810 Send signal
>> sofia/external/14159927717 [KILL]
>> 2009-10-22 17:22:40.626379 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/14159927717 [BREAK]
>> active_true=1
>> --contact NOT in playmsg or conference, update AGENT
>> 2009-10-22 17:22:40.626379 [DEBUG] switch_core_state_machine.c:437 thread
>> mismatch skipping state handler.
>> 2009-10-22 17:22:40.626379 [INFO] switch_cpp.cpp:1116
>> EVENTC_HANGUP_CONTACT, Q_CONTACT fired for agent:1/1 for qcall:1
>> contact:4801442/4
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:440
>> sofia/external/14159927717 ending bridge by request from write function
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:495
>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message
>> [UNBRIDGE]
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:630 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
>> DONE [sofia/external/+14158867717 at 199.173.100.16:5060]
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:523 Send signal
>> sofia/external/14159927717 [BREAK]
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr.c:1361 (sofia/external/+
>> 14158867717 at 199.173.100.16:5060) State Change CS_EXCHANGE_MEDIA ->
>> CS_ROUTING
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr.c:1365 sofia/external/+
>> 14158867717 at 199.173.100.16:5060 receive message [TRANSFER]
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:630 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:22:40.639760 [NOTICE] switch_ivr.c:1367 Transfer
>> sofia/external/+14158867717 at 199.173.100.16:5060 to xml[1998 at default]
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:343
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
>> going to sleep
>>  2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>> CS_ROUTING
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:333
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State ROUTING
>> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:130 sofia/external/+
>> 14158867717 at 199.173.100.16:5060 SOFIA ROUTING
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:78
>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard ROUTING
>> 2009-10-22 17:22:40.639760 [INFO] mod_dialplan_xml.c:391 Processing
>> +14158867717->1998 in context default
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->tod_example] continue=true
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (PASS)
>> [tod_example] ${strftime(%w)}(4) =~ /^([1-5])$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (PASS)
>> [tod_example] ${strftime(%H%M)}(1722) =~ /^((09|1[0-7])[0-5][0-9]|1800)$/
>> break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Action
>> set(open=true)
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->ivr_demo] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [ivr_demo] destination_number(1998) =~ /^5000$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1975] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1975] destination_number(1998) =~ /^1975$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1971] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1971] destination_number(1998) =~ /^1971$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1972] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1972] destination_number(1998) =~ /^1972$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1920] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1920] destination_number(1998) =~ /^1920$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1981] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1981] destination_number(1998) =~ /^1981$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1982] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1982] destination_number(1998) =~ /^1982$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1481] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1481] destination_number(1998) =~ /^1481$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1989] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1989] destination_number(1998) =~ /^1989$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1990] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1990] destination_number(1998) =~ /^1990$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1991] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1991] destination_number(1998) =~ /^1991$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1992] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1992] destination_number(1998) =~ /^1992$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1993] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1993] destination_number(1998) =~ /^1993$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1994] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1994] destination_number(1998) =~ /^1994$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1995] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1995] destination_number(1998) =~ /^1995$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1996] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1996] destination_number(1998) =~ /^1996$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1997] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>> [1997] destination_number(1998) =~ /^1997$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>> [default->1998] continue=false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (PASS)
>> [1998] destination_number(1998) =~ /^1998$/ break=on-false
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Action
>> set(transfer_after_bridge=1998)
>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Action park()
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:114
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_ROUTING
>> -> CS_EXECUTE
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:333
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State ROUTING going to
>> sleep
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>> CS_EXECUTE
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:340
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXECUTE
>> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:173 sofia/external/+
>> 14158867717 at 199.173.100.16:5060 SOFIA EXECUTE
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:151
>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard EXECUTE
>> EXECUTE sofia/external/+14158867717 at 199.173.100.16:5060 set(open=true)
>> 2009-10-22 17:22:40.639760 [DEBUG] mod_dptools.c:752 sofia/external/+
>> 14158867717 at 199.173.100.16:5060 SET [open]=[true]
>> EXECUTE sofia/external/+14158867717 at 199.173.100.16:5060set(transfer_after_bridge=1998)
>> 2009-10-22 17:22:40.639760 [DEBUG] mod_dptools.c:752 sofia/external/+
>> 14158867717 at 199.173.100.16:5060 SET [transfer_after_bridge]=[1998]
>> EXECUTE sofia/external/+14158867717 at 199.173.100.16:5060 park()
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:446
>> sofia/external/14159927717 ending bridge by request from read function
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
>> DONE [sofia/external/14159927717]
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:523 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> subtracting 9 seconds for hh_call: 1095422
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:346
>> (sofia/external/14159927717) State SOFT_EXECUTE going to sleep
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/14159927717) Running State Change CS_HANGUP
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:464
>> (sofia/external/14159927717) State HANGUP
>> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:344 Channel
>> sofia/external/14159927717 hanging up, cause: NORMAL_CLEARING
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:46
>> sofia/external/14159927717 Standard HANGUP, cause: NORMAL_CLEARING
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:464
>> (sofia/external/14159927717) State HANGUP going to sleep
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:325
>> (sofia/external/14159927717) State Change CS_HANGUP -> CS_REPORTING
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/14159927717 [BREAK]
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/14159927717) Running State Change CS_REPORTING
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:555
>> (sofia/external/14159927717) State REPORTING
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:53
>> sofia/external/14159927717 Standard REPORTING, cause: NORMAL_CLEARING
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:555
>> (sofia/external/14159927717) State REPORTING going to sleep
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:319
>> (sofia/external/14159927717) State Change CS_REPORTING -> CS_DESTROY
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/14159927717 [BREAK]
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:1069 Session 6
>> (sofia/external/14159927717) Locked, Waiting on external entities
>> 2009-10-22 17:22:40.639760 [NOTICE] switch_core_session.c:1087 Session 6
>> (sofia/external/14159927717) Ended
>> 2009-10-22 17:22:40.639760 [NOTICE] switch_core_session.c:1089 Close
>> Channel sofia/external/14159927717 [CS_DESTROY]
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:401
>> (sofia/external/14159927717) Running State Change CS_DESTROY
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:412
>> (sofia/external/14159927717) State DESTROY
>> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:261
>> sofia/external/14159927717 SOFIA DESTROY
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:60
>> sofia/external/14159927717 Standard DESTROY
>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:412
>> (sofia/external/14159927717) State DESTROY going to sleep
>> 2009-10-22 17:22:40.639760 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>> teliax.com hc:NORMAL_CLEARING du:9 cn:sofia/external/14159927717
>>
>>
>>
>> A BAD UUID_KILL SEQUENCE: (where the agent is hungup, unexpectedly)
>>
>> --bridged in: 0.005821 seconds
>> 2009-10-22 17:23:26.718930 [INFO] switch_cpp.cpp:1116 AG_CBRIDGE,
>> connected agent:1/1 contact:4801439/22/1/1
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_play_say.c:1432 done playing
>> file
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:340
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXECUTE going to
>> sleep
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>> CS_HIBERNATE
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE
>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:160 sofia/external/+
>> 14158867717 at 199.173.100.16:5060 SOFIA HIBERNATE
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:650
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>> CS_HIBERNATE -> CS_RESET
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE going
>> to sleep
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>> CS_RESET
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET
>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:145 sofia/external/+
>> 14158867717 at 199.173.100.16:5060 SOFIA RESET
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:637
>> sofia/external/+14158867717 at 199.173.100.16:5060 CUSTOM RESET
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:67
>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard RESET
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET going to
>> sleep
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:340
>> (sofia/external/14159927717) State EXECUTE going to sleep
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/14159927717) Running State Change CS_HIBERNATE
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
>> (sofia/external/14159927717) State HIBERNATE
>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:160
>> sofia/external/14159927717 SOFIA HIBERNATE
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:650
>> (sofia/external/14159927717) State Change CS_HIBERNATE -> CS_RESET
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/14159927717 [BREAK]
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
>> (sofia/external/14159927717) State HIBERNATE going to sleep
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/14159927717) Running State Change CS_RESET
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
>> (sofia/external/14159927717) State RESET
>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:145
>> sofia/external/14159927717 SOFIA RESET
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:637
>> sofia/external/14159927717 CUSTOM RESET
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:642
>> (sofia/external/14159927717) State Change CS_RESET -> CS_SOFT_EXECUTE
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/14159927717 [BREAK]
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
>> (sofia/external/14159927717) State RESET going to sleep
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/14159927717) Running State Change CS_SOFT_EXECUTE
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:346
>> (sofia/external/14159927717) State SOFT_EXECUTE
>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:442 SOFIA TRANSMIT
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:660
>> sofia/external/14159927717 CUSTOM SOFT_EXECUTE
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_originate.c:674
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_RESET
>> -> CS_SOFT_EXECUTE
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:120
>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message [DISPLAY]
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:120
>> sofia/external/14159927717 receive message [DISPLAY]
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:924
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>> CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:968
>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message [BRIDGE]
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:630 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:975
>> sofia/external/14159927717 receive message [BRIDGE]
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:630 Send signal
>> sofia/external/14159927717 [BREAK]
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:1019
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>> CS_EXCHANGE_MEDIA
>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:343
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK
>> 2009-10-22 17:23:26.740065 [DEBUG] switch_core_io.c:234 sofia/external/+
>> 14158867717 at 199.173.100.16:5060 receive message [TRANSCODING_NECESSARY]
>> 2009-10-22 17:23:27.319618 [DEBUG] switch_core_media_bug.c:393 Removing
>> BUG from sofia/external/+14158867717 at 199.173.100.16:5060
>> multipler is being overrided to1
>> 2009-10-22 17:23:29.406414 [INFO] switch_cpp.cpp:1116 BG_AG_GEN_CALLS,
>> Agents total:1 avail:0 OutChan:0 AR:0 +-Calls:0:Q Call NYNoc
>> 2009-10-22 17:23:31.873352 [NOTICE] mod_commands.c:1523 Hangup
>> sofia/external/14159927717 [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
>> 2009-10-22 17:23:31.874411 [INFO] switch_cpp.cpp:1116
>> EVENTC_HANGUP_CONTACT, Q_CONTACT fired for agent:1/1 for qcall:1
>> contact:4801439/1
>> active_true=1
>> --contact NOT in playmsg or conference, update AGENT
>> 2009-10-22 17:23:31.875475 [DEBUG] switch_channel.c:1810 Send signal
>> sofia/external/14159927717 [KILL]
>> 2009-10-22 17:23:31.875475 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/14159927717 [BREAK]
>> 2009-10-22 17:23:31.875475 [DEBUG] switch_core_state_machine.c:437 thread
>> mismatch skipping state handler.
>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:446
>> sofia/external/14159927717 ending bridge by request from read function
>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
>> DONE [sofia/external/14159927717]
>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:523 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:495
>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message
>> [UNBRIDGE]
>> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_session.c:630 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
>> DONE [sofia/external/+14158867717 at 199.173.100.16:5060]
>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:523 Send signal
>> sofia/external/14159927717 [BREAK]
>> 2009-10-22 17:23:31.882153 [NOTICE] switch_ivr.c:1268 Hangup
>> sofia/external/+14158867717 at 199.173.100.16:5060 [CS_EXCHANGE_MEDIA]
>> [EXCHANGE_ROUTING_ERROR]
>>  2009-10-22 17:23:31.882153 [DEBUG] switch_channel.c:1810 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [KILL]
>> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_state_machine.c:464
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HANGUP
>> 2009-10-22 17:23:31.882153 [DEBUG] mod_sofia.c:344 Channel
>> sofia/external/+14158867717 at 199.173.100.16:5060 hanging up, cause:
>> EXCHANGE_ROUTING_ERROR
>> 2009-10-22 17:23:31.882153 [DEBUG] mod_sofia.c:382 Sending BYE to
>> sofia/external/+14158867717 at 199.173.100.16:5060
>> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_state_machine.c:46
>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard HANGUP, cause:
>> EXCHANGE_ROUTING_ERROR
>> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_state_machine.c:464
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HANGUP going to
>> sleep
>> 2009-10-22 17:23:31.884357 [INFO] switch_cpp.cpp:1116 HH_HANGUP, agent:1/1
>> uuid: f8604694-4e44-4c4c-9805-722b3e9d6b68
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:346
>> (sofia/external/14159927717) State SOFT_EXECUTE going to sleep
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/14159927717) Running State Change CS_HANGUP
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:464
>> (sofia/external/14159927717) State HANGUP
>> 2009-10-22 17:23:31.885412 [DEBUG] mod_sofia.c:344 Channel
>> sofia/external/14159927717 hanging up, cause: NORMAL_CLEARING
>> 2009-10-22 17:23:31.885412 [DEBUG] mod_sofia.c:382 Sending BYE to
>> sofia/external/14159927717
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:46
>> sofia/external/14159927717 Standard HANGUP, cause: NORMAL_CLEARING
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:464
>> (sofia/external/14159927717) State HANGUP going to sleep
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:325
>> (sofia/external/14159927717) State Change CS_HANGUP -> CS_REPORTING
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/14159927717 [BREAK]
>> subtracting 5 seconds for hh_call: 1095423
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/14159927717) Running State Change CS_REPORTING
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:555
>> (sofia/external/14159927717) State REPORTING
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:53
>> sofia/external/14159927717 Standard REPORTING, cause: NORMAL_CLEARING
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:555
>> (sofia/external/14159927717) State REPORTING going to sleep
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:319
>> (sofia/external/14159927717) State Change CS_REPORTING -> CS_DESTROY
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/14159927717 [BREAK]
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_session.c:1069 Session 7
>> (sofia/external/14159927717) Locked, Waiting on external entities
>> 2009-10-22 17:23:31.885412 [NOTICE] switch_core_session.c:1087 Session 7
>> (sofia/external/14159927717) Ended
>> 2009-10-22 17:23:31.885412 [NOTICE] switch_core_session.c:1089 Close
>> Channel sofia/external/14159927717 [CS_DESTROY]
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:401
>> (sofia/external/14159927717) Running State Change CS_DESTROY
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:412
>> (sofia/external/14159927717) State DESTROY
>> 2009-10-22 17:23:31.885412 [DEBUG] mod_sofia.c:261
>> sofia/external/14159927717 SOFIA DESTROY
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:60
>> sofia/external/14159927717 Standard DESTROY
>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:412
>> (sofia/external/14159927717) State DESTROY going to sleep
>> 2009-10-22 17:23:31.885412 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>> teliax.com hc:NORMAL_CLEARING du:5 cn:sofia/external/14159927717
>> 2009-10-22 17:23:31.885412 [INFO] switch_cpp.cpp:1116 EVENTC_BILLING_CEIL,
>> subtracting 5 for hh_call:1095423
>> 2009-10-22 17:23:31.901620 [DEBUG] switch_core_state_machine.c:494 Hangup
>> Command luarun(hh_hangup.lua f8604694-4e44-4c4c-9805-722b3e9d6b68 1 1):
>> +OK
>>
>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:343
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
>> going to sleep
>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>> CS_HANGUP
>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:442 handler
>> already called, skipping state handler.
>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:325
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_HANGUP
>> -> CS_REPORTING
>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:306
>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>> CS_REPORTING
>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:555
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State REPORTING
>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:53
>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard REPORTING,
>> cause: EXCHANGE_ROUTING_ERROR
>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:555
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State REPORTING going
>> to sleep
>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:319
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>> CS_REPORTING -> CS_DESTROY
>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_session.c:1069 Session 4
>> (sofia/external/+14158867717 at 199.173.100.16:5060) Locked, Waiting on
>> external entities
>> 2009-10-22 17:23:31.903742 [NOTICE] switch_core_session.c:1087 Session 4
>> (sofia/external/+14158867717 at 199.173.100.16:5060) Ended
>> 2009-10-22 17:23:31.903742 [NOTICE] switch_core_session.c:1089 Close
>> Channel sofia/external/+14158867717 at 199.173.100.16:5060 [CS_DESTROY]
>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:401
>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>> CS_DESTROY
>> 2009-10-22 17:23:31.904799 [DEBUG] switch_core_state_machine.c:412
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State DESTROY
>> 2009-10-22 17:23:31.904799 [DEBUG] mod_sofia.c:261 sofia/external/+
>> 14158867717 at 199.173.100.16:5060 SOFIA DESTROY
>> 2009-10-22 17:23:31.904799 [DEBUG] switch_core_state_machine.c:60
>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard DESTROY
>> 2009-10-22 17:23:31.904799 [DEBUG] switch_core_state_machine.c:412
>> (sofia/external/+14158867717 at 199.173.100.16:5060) State DESTROY going to
>> sleep
>>
>>
>>
>> On Tue, Oct 20, 2009 at 1:44 AM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>>
>>>
>>> On Sun, Oct 18, 2009 at 9:48 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>
>>>> The debug level logs to the console, right? The pastebin, had log level
>>>> debug, sofia trace on for external and default, and sofia loglevel all 9. Is
>>>> there another log enable command I'm missing? It seems loglevel all 9
>>>> outputs enter and exit functions, but at least to my novice eye, it's not
>>>> too obvious why freeswitch is sending a BYE to my DID provider.
>>>>
>>>> Evidently the console loglevel debug did not take because there isn't a
>>> single DEBUG level line of output. Please try again. Just remember that
>>> debug level log messages will have A LOT of information so be patient while
>>> sifting through it. Of course, feel free to pastebin the output and add a
>>> link to it in this email thread.
>>>
>>> -MC
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mcampbellsmith at gmail.com  Thu Oct 22 21:00:49 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Fri, 23 Oct 2009 15:00:49 +1100
Subject: [Freeswitch-users] Core Dump question!
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71701E84203@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701E840EB@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221258o6936000dn6eb2e22189c45435@mail.gmail.com>
	<DF11CCC6-5C89-4489-908C-92A0FCC94AE4@jerris.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E841E8@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221725g6233ec20w49a73864400426d6@mail.gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E84200@EXMBXCLUS01.citservers.local>
	<86a32abc0910222004y46bf08b6jda75b6b8f1979420@mail.gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E84203@EXMBXCLUS01.citservers.local>
Message-ID: <33c87fa30910222100rae92b90uf94d7ae58a5cdc94@mail.gmail.com>

I think the (exported) means you don't have the latest svn, but probably the
officially released build 1.0.4 that can be downloaded from the FS page.

I think you should see something like (the latest trunk is 15203):

freeswitch at internal> version

FreeSWITCH Version 1.0.trunk (15126)

I guess you need to checkout the latest FS trunk

On Fri, Oct 23, 2009 at 2:25 PM, Ujjval Karihaloo
<ujjval at simplesignal.com>wrote:

>  freeswitch at internal> version
>
> FreeSWITCH Version 1.0.4 (exported)
>
>
>
> freeswitch at internal>
>
>
>
> Ujjval Karihaloo
>
> VP Voice Engineering
>
> IP Phone: +13032428610
>
> E-Fax: +17202391690
>
>
>
> SimpleSignal Inc.
>
> 88 Inverness Circle East
>
> Suite K105
>
> Englewood, CO  80112
>
> [image: bvoip] <http://www.simplesignal.com/>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Diego Viola
> *Sent:* Thursday, October 22, 2009 9:04 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Core Dump question!
>
>
>
> Type "version" on the CLI.
>
> On Fri, Oct 23, 2009 at 2:52 AM, Ujjval Karihaloo <ujjval at simplesignal.com>
> wrote:
>
> How do I tell if it?s the latest?I downloaded is yesterday..and installed
> it from freeswitch.org
>
>
>
> Ujjval Karihaloo
>
> VP Voice Engineering
>
> IP Phone: +13032428610
>
> E-Fax: +17202391690
>
>
>
> SimpleSignal Inc.
>
> 88 Inverness Circle East
>
> Suite K105
>
> Englewood, CO  80112
>
> [image: bvoip] <http://www.simplesignal.com/>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, October 22, 2009 6:26 PM
>
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Core Dump question!
>
>
>
> Yes, if this is latest SVN (after a make current) then open a jira.
> -MC
>
> On Thu, Oct 22, 2009 at 5:04 PM, Ujjval Karihaloo <ujjval at simplesignal.com>
> wrote:
>
> I do have the core dump, should I open a ticket.
>
> I am running latest Freeswitch 1.0.4 and had done a make current just
> before it happened.
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From diego.viola at gmail.com  Thu Oct 22 22:15:21 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 23 Oct 2009 05:15:21 +0000
Subject: [Freeswitch-users] Core Dump question!
In-Reply-To: <33c87fa30910222100rae92b90uf94d7ae58a5cdc94@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701E840EB@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221258o6936000dn6eb2e22189c45435@mail.gmail.com>
	<DF11CCC6-5C89-4489-908C-92A0FCC94AE4@jerris.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E841E8@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221725g6233ec20w49a73864400426d6@mail.gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E84200@EXMBXCLUS01.citservers.local>
	<86a32abc0910222004y46bf08b6jda75b6b8f1979420@mail.gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E84203@EXMBXCLUS01.citservers.local>
	<33c87fa30910222100rae92b90uf94d7ae58a5cdc94@mail.gmail.com>
Message-ID: <86a32abc0910222215p6eb79f66lbbef2b245f4d17fd@mail.gmail.com>

He's running 1.0.4, he needs to checkout from SVN trunk and build FS from
scratch.

Diego

On Fri, Oct 23, 2009 at 4:00 AM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> I think the (exported) means you don't have the latest svn, but probably
> the officially released build 1.0.4 that can be downloaded from the FS page.
>
> I think you should see something like (the latest trunk is 15203):
>
> freeswitch at internal> version
>
> FreeSWITCH Version 1.0.trunk (15126)
>
> I guess you need to checkout the latest FS trunk
>
>
> On Fri, Oct 23, 2009 at 2:25 PM, Ujjval Karihaloo <ujjval at simplesignal.com
> > wrote:
>
>>  freeswitch at internal> version
>>
>> FreeSWITCH Version 1.0.4 (exported)
>>
>>
>>
>> freeswitch at internal>
>>
>>
>>
>> Ujjval Karihaloo
>>
>> VP Voice Engineering
>>
>> IP Phone: +13032428610
>>
>> E-Fax: +17202391690
>>
>>
>>
>> SimpleSignal Inc.
>>
>> 88 Inverness Circle East
>>
>> Suite K105
>>
>> Englewood, CO  80112
>>
>> [image: bvoip] <http://www.simplesignal.com/>
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Diego Viola
>> *Sent:* Thursday, October 22, 2009 9:04 PM
>>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* Re: [Freeswitch-users] Core Dump question!
>>
>>
>>
>> Type "version" on the CLI.
>>
>> On Fri, Oct 23, 2009 at 2:52 AM, Ujjval Karihaloo <
>> ujjval at simplesignal.com> wrote:
>>
>> How do I tell if it?s the latest?I downloaded is yesterday..and installed
>> it from freeswitch.org
>>
>>
>>
>> Ujjval Karihaloo
>>
>> VP Voice Engineering
>>
>> IP Phone: +13032428610
>>
>> E-Fax: +17202391690
>>
>>
>>
>> SimpleSignal Inc.
>>
>> 88 Inverness Circle East
>>
>> Suite K105
>>
>> Englewood, CO  80112
>>
>> [image: bvoip] <http://www.simplesignal.com/>
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
>> Collins
>> *Sent:* Thursday, October 22, 2009 6:26 PM
>>
>>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* Re: [Freeswitch-users] Core Dump question!
>>
>>
>>
>> Yes, if this is latest SVN (after a make current) then open a jira.
>> -MC
>>
>> On Thu, Oct 22, 2009 at 5:04 PM, Ujjval Karihaloo <
>> ujjval at simplesignal.com> wrote:
>>
>> I do have the core dump, should I open a ticket.
>>
>> I am running latest Freeswitch 1.0.4 and had done a make current just
>> before it happened.
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From maciej.aniserowicz at gmail.com  Fri Oct 23 00:36:31 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Fri, 23 Oct 2009 00:36:31 -0700 (PDT)
Subject: [Freeswitch-users] "Can not record session. Media not enabled
 on channel."
In-Reply-To: <87f2f3b90910201746h43a31b32m5c163fd98373f915@mail.gmail.com>
References: <1256034659005-3857858.post@n2.nabble.com>
	<87f2f3b90910201746h43a31b32m5c163fd98373f915@mail.gmail.com>
Message-ID: <1256283391890-3877285.post@n2.nabble.com>


The dialplan is very simple:

	<extension name="Recording test">
		<condition field="destination_number" expression="^111111\d*$">
			<action application="sleep" data="3000" />
			<action application="answer"/>
			<action application="say" data="en name_spelled iterated
${destination_number}"/>
			<action application="sleep" data="1000" />
			<action application="playback" data="local_stream://my_music"/>
		</condition>
	</extension>

Before debugging I have another question: I start recording in event handler
for ChannelAnswer event. Is it possible that it's too soon to start
recording? Maybe I should start recording in some other event?

MAniserowicz



mercutioviz wrote:
> 
> What's in the dialplan for this channel? Is bypass-media or proxy-media
> set
> to true? Do a debug trace and post it in pastebin.
> -MC
> 
> On Tue, Oct 20, 2009 at 3:30 AM, Maciej Aniserowicz <
> maciej.aniserowicz at gmail.com> wrote:
> 
>>
>> Hello,
>> I am using the same set of extensions for testing the system during
>> development, they include XLite, Cisco sip phone and several extensions
>> that
>> just play some audio file.
>> Sometimes, very rarely, this message "Can not record session.  Media not
>> enabled on channel." appears on FS console.
>> Like I wrote before, I don't change any codec settings and always use the
>> same set of devices/emulators.
>> What can cause this message?
>>
>> Thanks,
>> Maciej Aniserowicz
>>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3877285.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From tculjaga at gmail.com  Fri Oct 23 01:16:14 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 23 Oct 2009 10:16:14 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>
	<65d96fc80910220753n6d2efa39l470791f75db9a640@mail.gmail.com>
	<Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>
	<65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
	<Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
Message-ID: <65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>

> TC>
> TC>it is gonna be easier to track.
> TC>
> TC>Tomorrow i will test on 1.0.4 but please lets move to trunk.
>
> i make it a bit later, to move tickets to jira and source to svn i
> need some time to undertand how this system is works, especially jira.
>
>
audio issue is better now :)

however i have a few questions:

1. can we control codec framing size via config file setting (e.g. PCMA:20,
PCMU:20)?
2. can we control tunneling via config file setting?
3. can we control mediaWaitForConnect flag within setup message via config
file setting?


Now, when i can test more and place outgoing calls to different equipment, i
found that there is an issue when we get h.225 progress without a fastStart
element.

Here is a tshark:

  5.242526    10.4.62.7 -> 10.4.4.254   H.225.0 CS: setup OpenLogicalChannel

  5.243982   10.4.4.254 -> 10.4.62.7    H.225.0 CS: callProceeding
 10.512617   10.4.4.254 -> 10.4.62.7    H.225.0 CS: progress
 10.983697   10.4.4.254 -> 10.4.62.7    H.225.0 CS: alerting
 20.002796   10.4.4.254 -> 10.4.62.7    H.225.0 CS: connect
 20.002981   10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
terminalCapabilitySet
 20.003210   10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
masterSlaveDetermination
 31.472362   10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: releaseComplete
endSessionCommand


the terminating GW didn't  include a faststart element within a returning
h.225 message we didn't match the capabilities (framing of them) in our
setup (and you are waiting an open LC to start pre_answer).... so now, the
terminator is waiting for the originator to start exchanging TCS/MSD. As
tunneling is true, this should be done using h.225 Facility messages.


your behavior should be like this:

  5.242526    10.4.62.7 -> 10.4.4.254   H.225.0 CS: setup
OpenLogicalChannel  g711A with 30 ms
  5.243982   10.4.4.254 -> 10.4.62.7    H.225.0 CS: callProceeding
 10.512617   10.4.4.254 -> 10.4.62.7    H.225.0 CS: progress

                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS: facility
terminalCapabilitySet
                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS: facility
masterSlaveDetermination

                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
terminalCapabilitySet
                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
masterSlaveDetermination

                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
terminalCapabilitySetAck
                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
masterSlaveDeterminationAck

                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS: facility
terminalCapabilitySetAck
                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS: facility
masterSlaveDeterminationAck

                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS: facility
openlogicalchannel (g711A)
                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
openlogicalchannel (g711A)

                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS: facility
openlogicalchannelAck
                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
openlogicalchannelAck

                       <now you can start pre_answer!>

 10.983697   10.4.4.254 -> 10.4.62.7    H.225.0 CS: alerting
...
...
...
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From tculjaga at gmail.com  Fri Oct 23 01:44:59 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 23 Oct 2009 10:44:59 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>
	<65d96fc80910220753n6d2efa39l470791f75db9a640@mail.gmail.com>
	<Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>
	<65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
	<Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
Message-ID: <65d96fc80910230144r49692f66w56f61085ac42cb44@mail.gmail.com>

btw you are back with an old issue:


static const char modulename[] = "h323";
static const char* h323_formats[] = {
    "G.711-ALaw-64k", "PCMU",
    "G.711-uLaw-64k", "PCMA",
    "GSM-06.10", "GSM",
    "MS-GSM", "msgsm",
    "SpeexNarrow", "speex",
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From red.rain.seven at gmail.com  Fri Oct 23 01:54:45 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Fri, 23 Oct 2009 16:54:45 +0800
Subject: [Freeswitch-users] sched_api doesn't get launched
In-Reply-To: <191c3a030910211309sb727b37g6324e1833f2a1e8c@mail.gmail.com>
References: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
	<3F1747F9-C543-4C6F-BB01-4DD5FA5B7BC5@jerris.com>
	<59ad9ca10910161153h5d531533vd844e49b9ca6d896@mail.gmail.com>
	<87f2f3b90910161237h356b7aa2q6ec12c6ecf78dc34@mail.gmail.com>
	<59ad9ca10910210244q5410f2bdq3b93c25f95673d11@mail.gmail.com>
	<191c3a030910211309sb727b37g6324e1833f2a1e8c@mail.gmail.com>
Message-ID: <59ad9ca10910230154i6d41ddfci8957b152c1884800@mail.gmail.com>

Thanks to c6burns on IRC channel for the tip to use execute_on_answer in
combination with eval, and of course everyone here that pointing me to the
right direction.
I was able to execute sched_api with eval, but not with the combination of
execute_on_answer. The argument just don't get parsed as the eval argument.
Here is the code:

setVariable("execute_on_answer", "eval sched_api(+%ld none uuid_displace
<some uuid> start
/usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux)");


Here is the log:
2009-10-23 16:17:53.633189 [DEBUG] switch_scheduler.c:214 Added task 2
sched_api_function (none) to run at 1256285888
EXECUTE sofia/internal/1688004 at 192.168.1.66 *eval(+OK *Added: 2
)

2009-10-23 16:18:04.52701 [DEBUG] switch_channel.c:1927 sofia/internal/
1688004 at 192.168.1.66 *execute on answer: eval(sched_api(+37 none
uuid_displace 84d74f41-8668-4138-943f-f076e94046ad start
/usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux))*
EXECUTE sofia/internal/1688004 at 192.168.1.66 eval(sched_api(+37 none
uuid_displace 84d74f41-8668-4138-943f-f076e94046ad start
/usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux))


The first part of the log is when I just do execute("eval",
"${sched_api(+%ld none uuid_displace <some uuid> start
/usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux)}");
And I simple get a OK, and then later on the sched_api gets executed.

However, on the excerpt of the second log. which I use the combination with
execute_on_answer like specified in the yellow hight code above. It doesn't
simply return a OK, instead it shows the whole blob of code. and does
nothing later.



On Thu, Oct 22, 2009 at 4:09 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Yes you need an API object as described in other email.
>
> Which line of code from java caused that segfault
> It looks like a simple NULL string issue that we may want to hunt down.
>
>
> On Wed, Oct 21, 2009 at 4:44 AM, Henry Huang <red.rain.seven at gmail.com>wrote:
>
>> I can't seem to find the right thing to use in mod_java to execute api
>> commands, only api_after_bridge
>>
>> 2009-10-21 17:42:46.593094 [NOTICE] mod_sofia.c:1509 Pre-Answer
>> sofia/internal/1688003 at 192.168.1.66!
>> #
>> # A fatal error has been detected by the Java Runtime Environment:
>> #
>> #  SIGSEGV (0xb) at pc=0x004e4480, pid=1927, tid=16116624
>> #
>> # JRE version: 6.0_16-b01
>> # Java VM: Java HotSpot(TM) Client VM (14.2-b01 mixed mode linux-x86 )
>> # Problematic frame:
>> # C  [libc.so.6+0x6f480]  strcpy+0x10
>> #
>> # An error report file with more information is saved as:
>> # /usr/local/freeswitch/bin/hs_err_pid1927.log
>> 2009-10-21 17:42:59.883729 [ERR] switch_core_session.c:1374 Invalid
>> Application sched_api
>> 2009-10-21 17:42:59.883729 [NOTICE] switch_core_session.c:1375 Hangup
>> sofia/internal/1688003 at 192.168.1.66 [CS_EXECUTE]
>> [DESTINATION_OUT_OF_ORDER]
>> #
>> # If you would like to submit a bug report, please visit:
>> #   http://java.sun.com/webapps/bugreport/crash.jsp
>> # The crash happened outside the Java Virtual Machine in native code.
>> # See problematic frame for where to report the bug.
>>
>>
>> On Sat, Oct 17, 2009 at 3:37 AM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>>
>>>
>>> On Fri, Oct 16, 2009 at 11:53 AM, Henry Huang <red.rain.seven at gmail.com>wrote:
>>>
>>>> So how would you trigger it from a script dialplan? The only time it
>>>> seemed to work is when I did setVariable("api_after_bridge", "sched_api blah
>>>> blah blah");
>>>> but then it gets executed after the channel's been teared down. I
>>>> thought api_after_bridge means right after the call gets connected.
>>>>
>>>> I need something to execute an api command right before or right after
>>>> the call gets bridged.
>>>>
>>>> api_after_bridge is a channel variable, so using setVariable works just
>>> fine. If you need to sched_api is an API only. Check these out:
>>> http://wiki.freeswitch.org/wiki/Mod_commands#Misc._Commands
>>>
>>> So you need an API object in order to use it. I don't know the syntax for
>>> creating an api obj in Java but in Lua it goes like this:
>>> api = freeswitch.API();
>>> res = api:execute("sched_api","+300 none my_api my_api_args")
>>>
>>> Remember, if the method you are using isn't found in the dial plan tools
>>> then it isn't a dial plan application. Make sure it's on the list:
>>> http://wiki.freeswitch.org/wiki/Mod_dptools
>>>
>>> On the other hand, API commands are listed here:
>>> http://wiki.freeswitch.org/wiki/Mod_commands
>>>
>>> dptools require a session object, api commands require an api object...
>>>
>>> -MC
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Henry Huang
>> UniC Solution - Communication Unified
>> VoIP & Open Source software Consultant
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From saeedahmad1981 at gmail.com  Fri Oct 23 03:07:40 2009
From: saeedahmad1981 at gmail.com (Saeed Ahmad)
Date: Fri, 23 Oct 2009 12:07:40 +0200
Subject: [Freeswitch-users] Heartbeat question
In-Reply-To: <87f2f3b90910221118n5d9fa0b9u92b45227c1d7ec97@mail.gmail.com>
References: <86a32abc0910081956k91b3600sb8c556e589f5b089@mail.gmail.com> 
	<86a32abc0910082003v3f5d4e6dua8e7eae18fd2f939@mail.gmail.com> 
	<6b65470d0910082036g204d5dcbw26e1e43ef80c7c84@mail.gmail.com> 
	<86a32abc0910082245j36b8b3e9ncaf1cdd358fce319@mail.gmail.com> 
	<191c3a030910090902t38282f72l6b6d1733a758836@mail.gmail.com> 
	<86a32abc0910091427i2395fd80t3bc557186ece3972@mail.gmail.com> 
	<86a32abc0910091430hc151ed8of085fcc3af2f1de4@mail.gmail.com> 
	<86a32abc0910091501y40dd34c4r1bca3eeb2b6fa693@mail.gmail.com> 
	<c96b95650910221047i545601a4m9e0a645341068d15@mail.gmail.com> 
	<87f2f3b90910221118n5d9fa0b9u92b45227c1d7ec97@mail.gmail.com>
Message-ID: <c96b95650910230307p79e71c6o9c46a004ecb72a78@mail.gmail.com>

Yeah the word **hearbeat** was confusing me, now its ok.

So one use case would be to count live calls with much detailed information.


is it *resource*-*intensive** operation? can i do it on wholesale traffic
(500+ calls).*

...any other interesting use cases?

Thanks.

On Thu, Oct 22, 2009 at 8:18 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Thu, Oct 22, 2009 at 10:47 AM, Saeed Ahmad <saeedahmad1981 at gmail.com>wrote:
>
>> What is heartbeat and what are the uses cases?
>>
>> Sorry i didn't find much information on wiki.
>>
>> Thanks.
>>
>> A session heartbeat is just an event that is sent to your script and gives
> you updated information about the call in progress. It is *VERY* different
> from the system heartbeat, which is sent out to all event socket
> subscribers, not just a particular session.
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bottleman at icf.org.ru  Fri Oct 23 04:03:53 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Fri, 23 Oct 2009 15:03:53 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910212118030.19031@srv.icf.org.ru>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>
	<65d96fc80910220753n6d2efa39l470791f75db9a640@mail.gmail.com>
	<Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>
	<65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
	<Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>

On 2009-10-23 10:16 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>> TC>
TC>> TC>it is gonna be easier to track.
TC>> TC>
TC>> TC>Tomorrow i will test on 1.0.4 but please lets move to trunk.
TC>>
TC>> i make it a bit later, to move tickets to jira and source to svn i
TC>> need some time to undertand how this system is works, especially jira.
TC>>
TC>>
TC>audio issue is better now :)
TC>
TC>however i have a few questions:
TC>
TC>1. can we control codec framing size via config file setting (e.g. PCMA:20,
TC>PCMU:20)?

at this time i think no, there is a number issues in codec part now.

TC>2. can we control tunneling via config file setting?

at this time no, i implement it later.

TC>3. can we control mediaWaitForConnect flag within setup message via config
TC>file setting?

what is mediaWaitForConnect flag, may be another trmin in h323?

TC>Now, when i can test more and place outgoing calls to different equipment, i
TC>found that there is an issue when we get h.225 progress without a fastStart
TC>element.
TC>
TC>Here is a tshark:
TC>
TC>  5.242526    10.4.62.7 -> 10.4.4.254   H.225.0 CS: setup OpenLogicalChannel
TC>
TC>  5.243982   10.4.4.254 -> 10.4.62.7    H.225.0 CS: callProceeding
TC> 10.512617   10.4.4.254 -> 10.4.62.7    H.225.0 CS: progress
TC> 10.983697   10.4.4.254 -> 10.4.62.7    H.225.0 CS: alerting
TC> 20.002796   10.4.4.254 -> 10.4.62.7    H.225.0 CS: connect
TC> 20.002981   10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>terminalCapabilitySet
TC> 20.003210   10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>masterSlaveDetermination
TC> 31.472362   10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: releaseComplete
TC>endSessionCommand
TC>
TC>
TC>the terminating GW didn't  include a faststart element within a returning
TC>h.225 message we didn't match the capabilities (framing of them) in our
TC>setup (and you are waiting an open LC to start pre_answer).... so now, the
TC>terminator is waiting for the originator to start exchanging TCS/MSD. As
TC>tunneling is true, this should be done using h.225 Facility messages.
TC>
TC>
TC>your behavior should be like this:
TC>
TC>  5.242526    10.4.62.7 -> 10.4.4.254   H.225.0 CS: setup
TC>OpenLogicalChannel  g711A with 30 ms
TC>  5.243982   10.4.4.254 -> 10.4.62.7    H.225.0 CS: callProceeding
TC> 10.512617   10.4.4.254 -> 10.4.62.7    H.225.0 CS: progress
TC>
TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS: facility
TC>terminalCapabilitySet
TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS: facility
TC>masterSlaveDetermination
TC>
TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>terminalCapabilitySet
TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>masterSlaveDetermination
TC>
TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>terminalCapabilitySetAck
TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>masterSlaveDeterminationAck
TC>
TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS: facility
TC>terminalCapabilitySetAck
TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS: facility
TC>masterSlaveDeterminationAck
TC>
TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS: facility
TC>openlogicalchannel (g711A)
TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>openlogicalchannel (g711A)
TC>
TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS: facility
TC>openlogicalchannelAck
TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>openlogicalchannelAck
TC>
TC>                       <now you can start pre_answer!>
TC>
TC> 10.983697   10.4.4.254 -> 10.4.62.7    H.225.0 CS: alerting
TC>...

may bee, while i in hospital i have a very limited ways for testing, 
especially for inbound calls throuce h323. i find one issues in signaling 
part in h323plus, src/h323.cxx grep "Very Frustrating - S.H." try uncomment 
fast start handling there, may be it help.

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From bottleman at icf.org.ru  Fri Oct 23 04:11:04 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Fri, 23 Oct 2009 15:11:04 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910230144r49692f66w56f61085ac42cb44@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910220630u4a1cae06ra994744d5525e131@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>
	<65d96fc80910220753n6d2efa39l470791f75db9a640@mail.gmail.com>
	<Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>
	<65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
	<Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
	<65d96fc80910230144r49692f66w56f61085ac42cb44@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910231504060.5644@srv.icf.org.ru>

On 2009-10-23 10:44 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>btw you are back with an old issue:
TC>
TC>
TC>static const char modulename[] = "h323";
TC>static const char* h323_formats[] = {
TC>    "G.711-ALaw-64k", "PCMU",
TC>    "G.711-uLaw-64k", "PCMA",
TC>    "GSM-06.10", "GSM",
TC>    "MS-GSM", "msgsm",
TC>    "SpeexNarrow", "speex",

there is a isues with codecs, it is not enoth to put it there to work, 
at this time work only PCMU PCMA GSM and G729.
									    

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From tculjaga at gmail.com  Fri Oct 23 04:22:40 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 23 Oct 2009 13:22:40 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>
	<65d96fc80910220753n6d2efa39l470791f75db9a640@mail.gmail.com>
	<Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>
	<65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
	<Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
	<Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>
Message-ID: <65d96fc80910230422q334506c7v89664f2e61696a8a@mail.gmail.com>

i meant you switched PCMA and PCMU...

T.

2009/10/23 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-23 10:16 +0200, Tihomir Culjaga wrote
> freeswitch-users at lists.fre...:
>
> TC>> TC>
> TC>> TC>it is gonna be easier to track.
> TC>> TC>
> TC>> TC>Tomorrow i will test on 1.0.4 but please lets move to trunk.
> TC>>
> TC>> i make it a bit later, to move tickets to jira and source to svn i
> TC>> need some time to undertand how this system is works, especially jira.
> TC>>
> TC>>
> TC>audio issue is better now :)
> TC>
> TC>however i have a few questions:
> TC>
> TC>1. can we control codec framing size via config file setting (e.g.
> PCMA:20,
> TC>PCMU:20)?
>
> at this time i think no, there is a number issues in codec part now.
>
> TC>2. can we control tunneling via config file setting?
>
> at this time no, i implement it later.
>
> TC>3. can we control mediaWaitForConnect flag within setup message via
> config
> TC>file setting?
>
> what is mediaWaitForConnect flag, may be another trmin in h323?
>
> TC>Now, when i can test more and place outgoing calls to different
> equipment, i
> TC>found that there is an issue when we get h.225 progress without a
> fastStart
> TC>element.
> TC>
> TC>Here is a tshark:
> TC>
> TC>  5.242526    10.4.62.7 -> 10.4.4.254   H.225.0 CS: setup
> OpenLogicalChannel
> TC>
> TC>  5.243982   10.4.4.254 -> 10.4.62.7    H.225.0 CS: callProceeding
> TC> 10.512617   10.4.4.254 -> 10.4.62.7    H.225.0 CS: progress
> TC> 10.983697   10.4.4.254 -> 10.4.62.7    H.225.0 CS: alerting
> TC> 20.002796   10.4.4.254 -> 10.4.62.7    H.225.0 CS: connect
> TC> 20.002981   10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
> TC>terminalCapabilitySet
> TC> 20.003210   10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
> TC>masterSlaveDetermination
> TC> 31.472362   10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS:
> releaseComplete
> TC>endSessionCommand
> TC>
> TC>
> TC>the terminating GW didn't  include a faststart element within a
> returning
> TC>h.225 message we didn't match the capabilities (framing of them) in our
> TC>setup (and you are waiting an open LC to start pre_answer).... so now,
> the
> TC>terminator is waiting for the originator to start exchanging TCS/MSD. As
> TC>tunneling is true, this should be done using h.225 Facility messages.
> TC>
> TC>
> TC>your behavior should be like this:
> TC>
> TC>  5.242526    10.4.62.7 -> 10.4.4.254   H.225.0 CS: setup
> TC>OpenLogicalChannel  g711A with 30 ms
> TC>  5.243982   10.4.4.254 -> 10.4.62.7    H.225.0 CS: callProceeding
> TC> 10.512617   10.4.4.254 -> 10.4.62.7    H.225.0 CS: progress
> TC>
> TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS:
> facility
> TC>terminalCapabilitySet
> TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS:
> facility
> TC>masterSlaveDetermination
> TC>
> TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS:
> facility
> TC>terminalCapabilitySet
> TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS:
> facility
> TC>masterSlaveDetermination
> TC>
> TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS:
> facility
> TC>terminalCapabilitySetAck
> TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS:
> facility
> TC>masterSlaveDeterminationAck
> TC>
> TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS:
> facility
> TC>terminalCapabilitySetAck
> TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS:
> facility
> TC>masterSlaveDeterminationAck
> TC>
> TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS:
> facility
> TC>openlogicalchannel (g711A)
> TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS:
> facility
> TC>openlogicalchannel (g711A)
> TC>
> TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS:
> facility
> TC>openlogicalchannelAck
> TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS:
> facility
> TC>openlogicalchannelAck
> TC>
> TC>                       <now you can start pre_answer!>
> TC>
> TC> 10.983697   10.4.4.254 -> 10.4.62.7    H.225.0 CS: alerting
> TC>...
>
> may bee, while i in hospital i have a very limited ways for testing,
> especially for inbound calls throuce h323. i find one issues in signaling
> part in h323plus, src/h323.cxx grep "Very Frustrating - S.H." try uncomment
> fast start handling there, may be it help.
>
> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129-RIPE
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bottleman at icf.org.ru  Fri Oct 23 04:32:41 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Fri, 23 Oct 2009 15:32:41 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910230422q334506c7v89664f2e61696a8a@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>
	<65d96fc80910220753n6d2efa39l470791f75db9a640@mail.gmail.com>
	<Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>
	<65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
	<Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
	<Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>
	<65d96fc80910230422q334506c7v89664f2e61696a8a@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910231532050.16254@srv.icf.org.ru>

On 2009-10-23 13:22 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

my bad, thx.

TC>i meant you switched PCMA and PCMU...
TC>
TC>T.
TC>
TC>2009/10/23 Georgiewskiy Yuriy <bottleman at icf.org.ru>
TC>
TC>> On 2009-10-23 10:16 +0200, Tihomir Culjaga wrote
TC>> freeswitch-users at lists.fre...:
TC>>
TC>> TC>> TC>
TC>> TC>> TC>it is gonna be easier to track.
TC>> TC>> TC>
TC>> TC>> TC>Tomorrow i will test on 1.0.4 but please lets move to trunk.
TC>> TC>>
TC>> TC>> i make it a bit later, to move tickets to jira and source to svn i
TC>> TC>> need some time to undertand how this system is works, especially jira.
TC>> TC>>
TC>> TC>>
TC>> TC>audio issue is better now :)
TC>> TC>
TC>> TC>however i have a few questions:
TC>> TC>
TC>> TC>1. can we control codec framing size via config file setting (e.g.
TC>> PCMA:20,
TC>> TC>PCMU:20)?
TC>>
TC>> at this time i think no, there is a number issues in codec part now.
TC>>
TC>> TC>2. can we control tunneling via config file setting?
TC>>
TC>> at this time no, i implement it later.
TC>>
TC>> TC>3. can we control mediaWaitForConnect flag within setup message via
TC>> config
TC>> TC>file setting?
TC>>
TC>> what is mediaWaitForConnect flag, may be another trmin in h323?
TC>>
TC>> TC>Now, when i can test more and place outgoing calls to different
TC>> equipment, i
TC>> TC>found that there is an issue when we get h.225 progress without a
TC>> fastStart
TC>> TC>element.
TC>> TC>
TC>> TC>Here is a tshark:
TC>> TC>
TC>> TC>  5.242526    10.4.62.7 -> 10.4.4.254   H.225.0 CS: setup
TC>> OpenLogicalChannel
TC>> TC>
TC>> TC>  5.243982   10.4.4.254 -> 10.4.62.7    H.225.0 CS: callProceeding
TC>> TC> 10.512617   10.4.4.254 -> 10.4.62.7    H.225.0 CS: progress
TC>> TC> 10.983697   10.4.4.254 -> 10.4.62.7    H.225.0 CS: alerting
TC>> TC> 20.002796   10.4.4.254 -> 10.4.62.7    H.225.0 CS: connect
TC>> TC> 20.002981   10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>> TC>terminalCapabilitySet
TC>> TC> 20.003210   10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>> TC>masterSlaveDetermination
TC>> TC> 31.472362   10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS:
TC>> releaseComplete
TC>> TC>endSessionCommand
TC>> TC>
TC>> TC>
TC>> TC>the terminating GW didn't  include a faststart element within a
TC>> returning
TC>> TC>h.225 message we didn't match the capabilities (framing of them) in our
TC>> TC>setup (and you are waiting an open LC to start pre_answer).... so now,
TC>> the
TC>> TC>terminator is waiting for the originator to start exchanging TCS/MSD. As
TC>> TC>tunneling is true, this should be done using h.225 Facility messages.
TC>> TC>
TC>> TC>
TC>> TC>your behavior should be like this:
TC>> TC>
TC>> TC>  5.242526    10.4.62.7 -> 10.4.4.254   H.225.0 CS: setup
TC>> TC>OpenLogicalChannel  g711A with 30 ms
TC>> TC>  5.243982   10.4.4.254 -> 10.4.62.7    H.225.0 CS: callProceeding
TC>> TC> 10.512617   10.4.4.254 -> 10.4.62.7    H.225.0 CS: progress
TC>> TC>
TC>> TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS:
TC>> facility
TC>> TC>terminalCapabilitySet
TC>> TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS:
TC>> facility
TC>> TC>masterSlaveDetermination
TC>> TC>
TC>> TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS:
TC>> facility
TC>> TC>terminalCapabilitySet
TC>> TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS:
TC>> facility
TC>> TC>masterSlaveDetermination
TC>> TC>
TC>> TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS:
TC>> facility
TC>> TC>terminalCapabilitySetAck
TC>> TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS:
TC>> facility
TC>> TC>masterSlaveDeterminationAck
TC>> TC>
TC>> TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS:
TC>> facility
TC>> TC>terminalCapabilitySetAck
TC>> TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS:
TC>> facility
TC>> TC>masterSlaveDeterminationAck
TC>> TC>
TC>> TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS:
TC>> facility
TC>> TC>openlogicalchannel (g711A)
TC>> TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS:
TC>> facility
TC>> TC>openlogicalchannel (g711A)
TC>> TC>
TC>> TC>                     10.4.62.7 -> 10.4.4.254    H.225.0/H.245 CS:
TC>> facility
TC>> TC>openlogicalchannelAck
TC>> TC>                     10.4.4.254 -> 10.4.62.7    H.225.0/H.245 CS:
TC>> facility
TC>> TC>openlogicalchannelAck
TC>> TC>
TC>> TC>                       <now you can start pre_answer!>
TC>> TC>
TC>> TC> 10.983697   10.4.4.254 -> 10.4.62.7    H.225.0 CS: alerting
TC>> TC>...
TC>>
TC>> may bee, while i in hospital i have a very limited ways for testing,
TC>> especially for inbound calls throuce h323. i find one issues in signaling
TC>> part in h323plus, src/h323.cxx grep "Very Frustrating - S.H." try uncomment
TC>> fast start handling there, may be it help.
TC>>
TC>> C ?????????                       With Best Regards
TC>> ???????????? ????.                Georgiewskiy Yuriy
TC>> +7 4872 711666                    +7 4872 711666
TC>> ???? +7 4872 711143               fax +7 4872 711143
TC>> ???????? ??? "?? ?? ??????"       IT Service Ltd
TC>> http://nkoort.ru                  http://nkoort.ru
TC>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
TC>> YG129-RIPE                        YG129-RIPE
TC>>
TC>> _______________________________________________
TC>> FreeSWITCH-users mailing list
TC>> FreeSWITCH-users at lists.freeswitch.org
TC>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>> http://www.freeswitch.org
TC>>
TC>>
TC>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From helmut.kuper at ewetel.de  Fri Oct 23 04:35:54 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Fri, 23 Oct 2009 13:35:54 +0200
Subject: [Freeswitch-users] Qustion about INFO messages
	after	Connect/Answer
In-Reply-To: <191c3a030910221008n60341314hc7a7e6a55a50ef15@mail.gmail.com>
References: <4AD859A4.5000101@ewetel.de>	<191c3a030910190800w5349824dl972866588a9ec59a@mail.gmail.com>	<4ADC85C0.7060908@ewetel.de>
	<4ADC8B6E.7070404@ewetel.de>	<191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>	<87f2f3b90910191435s7528e256r85df5f12fc634f24@mail.gmail.com>	<4ADF483B.4050807@ewetel.de>	<87f2f3b90910211048s208daf1eyfae501091ab75530@mail.gmail.com>	<4AE07DF4.5040803@ewetel.de>	<085B77EE-DA3F-4ADD-86DC-E59D1B6553EA@freeswitch.org>
	<191c3a030910221008n60341314hc7a7e6a55a50ef15@mail.gmail.com>
Message-ID: <4AE1951A.1030906@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Anthony,

thanks! The "unknown" thing is gone as long as you don't use
"originate_callee_id_name"


The "originate_callee_id_name" chvar causes caller's AND callee's
display ("caller name" line)to be updated with the content of that chvar
as soon as you set it.

Let's say you use "hubu" as originate_callee_id_name. As soon as callee
has picked up the phone it's display ("caller name" line) is updated
with "hubu" and right after that with caller's name. The caller's
display ("callee name" line) is updated to "hubu" (as expected).

So it's still not nice to see a name switch in callee's display for
caller's name after picking up.

In the following you see the SIP flows from

a) caller to FS
b) FS to callee


The more interesting one is b) I guess




#######################################################
This is the sip flow from caller to FS, which is OK:



INVITE sip:1000 at 85.16.246.12:5061;user=phone SIP/2.0
23/Oct/2009-13:26:44.730

INVITE sip:1000 at 85.16.246.12:5061;user=phone SIP/2.0
Via: SIP/2.0/UDP 85.16.245.206:1024;branch=z9hG4bK-zdcpisfjx11n;rport
From: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
To: <sip:1000 at 85.16.246.12:5061;user=phone>
Call-ID: 3c32462e758f-y19kuulwdknf
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:1001 at 85.16.245.206:1024;line=eg3wp69a>;reg-id=1
X-Serialnumber: 0004134002CB
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom820/8.2.16
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 734

v=0
o=root 1420196469 1420196469 IN IP4 85.16.245.206
s=call
c=IN IP4 85.16.245.206
t=0 0
m=audio 51286 RTP/SAVP 0 8 9 99 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:CjdPN1m2iLKAyXrQ2pWkQfCMtiISlF94ItTdsDis
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=audio 51286 RTP/AVP 0 8 9 99 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-ev

SIP/2.0 100 Trying
23/Oct/2009-13:26:44.731

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 85.16.245.206:1024;branch=z9hG4bK-zdcpisfjx11n;rport=1024
From: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
To: <sip:1000 at 85.16.246.12:5061;user=phone>
Call-ID: 3c32462e758f-y19kuulwdknf
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
Content-Length: 0


SIP/2.0 180 Ringing
23/Oct/2009-13:26:45.3

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 85.16.245.206:1024;branch=z9hG4bK-zdcpisfjx11n;rport=1024
From: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
To: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
Call-ID: 3c32462e758f-y19kuulwdknf
CSeq: 1 INVITE
Contact: <sip:1000 at 85.16.246.12:5061;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
Content-Length: 0
X-FS-Display-Name: hubu
X-FS-Display-Number:
X-FS-Support: update_display


SIP/2.0 200 OK
23/Oct/2009-13:26:47.367

SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.16.245.206:1024;branch=z9hG4bK-zdcpisfjx11n;rport=1024
From: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
To: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
Call-ID: 3c32462e758f-y19kuulwdknf
CSeq: 1 INVITE
Contact: <sip:1000 at 85.16.246.12:5061;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
Session-Expires: 3600;refresher=uas
Min-SE: 120
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 354
X-FS-Display-Name: hubu
X-FS-Display-Number:
X-FS-Support: update_display

v=0
o=FreeSWITCH 1256285701 1256285702 IN IP4 85.16.246.12
s=FreeSWITCH
c=IN IP4 85.16.246.12
t=0 0
m=audio 11506 RTP/SAVP 8 101
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:/BFm/LUmDBuao0wqg6MS/l4acfzOHoHRFQr3N0R9
m=audio 0 RTP/AVP 19

ACK sip:1000 at 85.16.246.12:5061;transport=udp SIP/2.0
23/Oct/2009-13:26:47.392

ACK sip:1000 at 85.16.246.12:5061;transport=udp SIP/2.0
Via: SIP/2.0/UDP 85.16.245.206:1024;branch=z9hG4bK-d474to81d7gh;rport
From: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
To: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
Call-ID: 3c32462e758f-y19kuulwdknf
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:1001 at 85.16.245.206:1024;line=eg3wp69a>;reg-id=1
Content-Length: 0


INFO sip:1001 at 85.16.245.206:1024;line=eg3wp69a SIP/2.0
23/Oct/2009-13:26:47.416

INFO sip:1001 at 85.16.245.206:1024;line=eg3wp69a SIP/2.0
Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKB244eKZ08Uj3H
Max-Forwards: 70
From: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
To: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
Call-ID: 3c32462e758f-y19kuulwdknf
CSeq: 122030267 INFO
Contact: <sip:1000 at 85.16.246.12:5061;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Type: message/sipfrag
Content-Length: 24

From:
To: "hubu" 1000

SIP/2.0 200 Ok
23/Oct/2009-13:26:47.619

SIP/2.0 200 Ok
Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKB244eKZ08Uj3H
From: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
To: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
Call-ID: 3c32462e758f-y19kuulwdknf
CSeq: 122030267 INFO
Contact: <sip:1001 at 85.16.245.206:1024;line=eg3wp69a>;reg-id=1
Content-Length: 0


BYE sip:1001 at 85.16.245.206:1024;line=eg3wp69a SIP/2.0
23/Oct/2009-13:26:49.18

BYE sip:1001 at 85.16.245.206:1024;line=eg3wp69a SIP/2.0
Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKDmQpj9072DZ8r
Max-Forwards: 70
From: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
To: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
Call-ID: 3c32462e758f-y19kuulwdknf
CSeq: 122030268 BYE
Contact: <sip:1000 at 85.16.246.12:5061;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0


SIP/2.0 200 OK
23/Oct/2009-13:26:49.52

SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKDmQpj9072DZ8r
From: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
To: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
Call-ID: 3c32462e758f-y19kuulwdknf
CSeq: 122030268 BYE
Contact: <sip:1001 at 85.16.245.206:1024;line=eg3wp69a>;reg-id=1
User-Agent: snom820/8.2.16
RTP-RxStat: Total_Rx_Pkts=65,Rx_Pkts=65,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0
RTP-TxStat: Total_Tx_Pkts=0,Tx_Pkts=94,Remote_Tx_Pkts=0
Content-Length: 0





####################################################################

This is the SIP flow from FS to callee.

The first INFO sent from FS to callee is the Problem. It contains "hubu"
as "To:" header. If there is s need for this message I guess the "From"
header should be changed, because from the callee-phone's point of view
the "To:" header is the caller and the "From" header is the callee. In
my opinion this INFO message shouldn't be sent at all, but I don't know
what you intend it to do.


INVITE sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
23/Oct/2009-13:26:44.978

INVITE sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bK86rt91cpH1FBF
Max-Forwards: 68
From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=rp27e1KSe969D
To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 122030266 INVITE
Contact: <sip:mod_sofia at 85.16.246.12:5061>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 325
X-Serialnumber: 0004134002CB
P-Key-Flags: resolution="31x13", keys="4"
X-FS-Support: update_display
Remote-Party-ID: "1001 an PBX1"
<sip:1001 at 85.16.246.12>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1256285400 1256285401 IN IP4 85.16.246.12
s=FreeSWITCH
c=IN IP4 85.16.246.12
t=0 0
m=audio 11804 RTP/AVP 95 103 9 8 101 13
a=rtpmap:95 CELT/48000
a=rtpmap:103 SPEEX/32000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:10

SIP/2.0 180 Ringing
23/Oct/2009-13:26:45.2

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bK86rt91cpH1FBF
From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=rp27e1KSe969D
To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 122030266 INVITE
Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Content-Length: 0


SIP/2.0 180 Ringing
23/Oct/2009-13:26:45.504

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bK86rt91cpH1FBF
From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=rp27e1KSe969D
To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 122030266 INVITE
Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Content-Length: 0


SIP/2.0 180 Ringing
23/Oct/2009-13:26:46.504

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bK86rt91cpH1FBF
From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=rp27e1KSe969D
To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 122030266 INVITE
Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Content-Length: 0


SIP/2.0 200 Ok
23/Oct/2009-13:26:47.361

SIP/2.0 200 Ok
Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bK86rt91cpH1FBF
From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=rp27e1KSe969D
To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 122030266 INVITE
Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
User-Agent: snom820/LAB_HD_HEAD_11_09_09_11_20
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Type: application/sdp
Content-Length: 244

v=0
o=root 353690302 353690303 IN IP4 85.16.245.228
s=call
c=IN IP4 85.16.245.228
t=0 0
m=audio 60878 RTP/AVP 9 8 101
a=rtpmap:9 g722/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:10
a=sendrecv

ACK sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
23/Oct/2009-13:26:47.362

ACK sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bK9FjKBXXSea6Xa
Max-Forwards: 70
From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=rp27e1KSe969D
To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 122030266 ACK
Contact: <sip:mod_sofia at 85.16.246.12:5061>
Content-Length: 0


INFO sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
23/Oct/2009-13:26:47.367

INFO sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKaSBcDreXBKvgp
Max-Forwards: 70
From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=rp27e1KSe969D
To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 122030267 INFO
Contact: <sip:mod_sofia at 85.16.246.12:5061>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Type: message/sipfrag
Content-Length: 24

From:
To: "hubu" 1000

SIP/2.0 200 Ok
23/Oct/2009-13:26:47.766

SIP/2.0 200 Ok
Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKaSBcDreXBKvgp
From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=rp27e1KSe969D
To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 122030267 INFO
Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
Content-Length: 0


INFO sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
23/Oct/2009-13:26:47.766

INFO sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKcByXgeg4548ND
Max-Forwards: 70
From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=rp27e1KSe969D
To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 122030268 INFO
Contact: <sip:mod_sofia at 85.16.246.12:5061>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Type: message/sipfrag
Content-Length: 32

From:
To: "1001 an PBX1" 1001

SIP/2.0 200 Ok
23/Oct/2009-13:26:47.860

SIP/2.0 200 Ok
Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKcByXgeg4548ND
From: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=rp27e1KSe969D
To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 122030268 INFO
Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
Content-Length: 0


BYE sip:mod_sofia at 85.16.246.12:5061 SIP/2.0
23/Oct/2009-13:26:49.3

BYE sip:mod_sofia at 85.16.246.12:5061 SIP/2.0
Via: SIP/2.0/UDP 85.16.245.228:1024;branch=z9hG4bK-susgzpvc38s7;rport
From: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
To: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=rp27e1KSe969D
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 1 BYE
Max-Forwards: 70
Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
User-Agent: snom820/LAB_HD_HEAD_11_09_09_11_20
RTP-RxStat: Total_Rx_Pkts=0,Rx_Pkts=69,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0
RTP-TxStat: Total_Tx_Pkts=70,Tx_Pkts=65,Remote_Tx_Pkts=0
Content-Length: 0


SIP/2.0 200 OK
23/Oct/2009-13:26:49.4

SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.16.245.228:1024;branch=z9hG4bK-susgzpvc38s7;rport=1024
From: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
To: "1001 an PBX1" <sip:1001 at 85.16.246.12>;tag=rp27e1KSe969D
Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
CSeq: 1 BYE
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Length: 0










On 22.10.2009 19:08, Anthony Minessale wrote:
> I'm sure that problem is gone in svn trunk.
> 
> 
> On Thu, Oct 22, 2009 at 11:25 AM, Brian West <brian at freeswitch.org
> <mailto:brian at freeswitch.org>> wrote:
> 
>     I can't get what exactly you re talking about. Can you clarify ? Also
>     please include the packets of interest only not the full trace if its
>     not relevant to the bug.
> 
>     /b
> 
>     On Oct 22, 2009, at 10:44 AM, Helmut Kuper wrote:
> 
>     > -----BEGIN PGP SIGNED MESSAGE-----
>     > Hash: SHA1
>     >
>     > Hi Mike,
>     >
>     > here it is:
>     >
>     >
>     > Dialplan:
>     >
>     >
>     >    <extension name="Local_Extension">
>     >      <condition field="destination_number" expression="^(10[01][0-9])
>     > $">
>     >        <action application="set" data="dialed_extension=$1"/>
>     >        <action application="export" data="dialed_extension=$1"/>
>     >        <action application="set" data="transfer_ringback=$$
>     > {hold_music}"/>
>     >        <action application="set" data="hangup_after_bridge=true"/>
>     >        <action application="bridge"
>     > data="user/${dialed_extension}@${domain_name}"/>
>     >      </condition>
>     >    </extension>
>     >
> 
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

- -- 

Mit freundlichen Gr??en
Helmut Kuper
Gesch?ftseinheit FD - L?sungen f?r Finanzdienstleister
Telefax: (0441) 8000-2799
mailto:helmut.kuper at ewetel.de
___________________________________
EWE TEL GmbH
Cloppenburger Stra?e 310
26133 Oldenburg
EWE TEL GmbH

Handelsregister Amtsgericht Oldenburg HRB 3723
Vorsitzender des Aufsichtsrates: Heiko Harms
Gesch?ftsf?hrung: Hans-Joachim Iken (Vorsitzender),
Ulf Heggenberger, Dr. Norbert Schulz, Dirk Thole
Homepage: http://www.ewetel.de
___________________________________
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=cHej
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From tculjaga at gmail.com  Fri Oct 23 04:52:06 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 23 Oct 2009 13:52:06 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>
	<65d96fc80910220753n6d2efa39l470791f75db9a640@mail.gmail.com>
	<Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>
	<65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
	<Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
	<Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>
Message-ID: <65d96fc80910230452s6b4a179aqa5bc41dc34efc403@mail.gmail.com>

>
> TC>3. can we control mediaWaitForConnect flag within setup message via
> config
> TC>file setting?
>
> what is mediaWaitForConnect flag, may be another trmin in h323?
>
>
>
If the calling endpoint sets the mediaWaitForConnect element to TRUE in the
Setup message, then
the called endpoint shall not send any media until after the Connect message
is sent.
The calling endpoint may begin transmitting media (according to the channels
opened) immediately
upon receiving a Q.931 message containing fastStart. Thus, the called
endpoint must be prepared to
immediately receive media on the channels it accepted in the Q.931 message
containing fastStart.
Note that national requirements may prohibit calling endpoints from
transmitting media prior to
receipt of a Connect message; it is the responsibility of the endpoint to
comply with applicable
requirements.


check "H225_Setup_UUIE & H323SignalPDU::BuildSetup" within src/h323pdu.cxx
(H323plus)



> TC>...
>
> may bee, while i in hospital i have a very limited ways for testing,
> especially for inbound calls throuce h323. i find one issues in signaling
> part in h323plus, src/h323.cxx grep "Very Frustrating - S.H." try uncomment
> fast start handling there, may be it help.
>
>
I'm not sure it is gonna help. This is only for CallProceeding having a
faststart element... What i have is a progress message without a faststart
element but with h245 address... it should go to StartControlChannel but i
think it is stuck since you call pre_answer before it actually opens a LC.


PBoolean H323Connection::OnReceivedProgress(const H323SignalPDU & pdu)
{
  if (pdu.m_h323_uu_pdu.m_h323_message_body.GetTag() !=
H225_H323_UU_PDU_h323_message_body::e_progress)
    return FALSE;
  const H225_Progress_UUIE & progress =
pdu.m_h323_uu_pdu.m_h323_message_body;

  SetRemoteVersions(progress.m_protocolIdentifier);
  SetRemotePartyInfo(pdu);
  SetRemoteApplication(progress.m_destinationInfo);

  // Check for fastStart data and start fast
  if (progress.HasOptionalField(H225_Progress_UUIE::e_fastStart))
    HandleFastStartAcknowledge(progress.m_fastStart);

  // Check that it has the H.245 channel connection info
  if (progress.HasOptionalField(H225_Progress_UUIE::e_h245Address))
    return StartControlChannel(progress.m_h245Address);

  return TRUE;
}



you should handle this and postpone pre_answer until you get an open LC.


T.
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From helmut.kuper at ewetel.de  Fri Oct 23 05:09:50 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Fri, 23 Oct 2009 14:09:50 +0200
Subject: [Freeswitch-users] INFO-Messages: Send_display in pre_answer state
	doesnt work anymore
Message-ID: <4AE19D0E.9050301@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

when I use pre_answer app and right after that the send_display app,
there is no INFO message anymore.

My dialplan condition:

<condition field="${sip_hangup_phrase}" expression="^Do Not Disturb"
break="on-true">
                        <action application="pre_answer"/>
                        <action application="send_display" data="Bitte
nicht stoeren"/>
                        <action application="sleep" data="500"/>
                        <action application="playback"
data="ansagen/22.wav"/>
</condition>

changing "pre_answer" to "answer" works, but I don't want to answer the
call for error messages.

I use latest SVN trunk.

regards
helmut
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CY+yWGyVlyDpDs//WFtCqiA=
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From bottleman at icf.org.ru  Fri Oct 23 05:13:48 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Fri, 23 Oct 2009 16:13:48 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910230452s6b4a179aqa5bc41dc34efc403@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910221738180.17947@srv.icf.org.ru>
	<65d96fc80910220659v3837a174u49fff0eb6bb77cda@mail.gmail.com>
	<Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>
	<65d96fc80910220753n6d2efa39l470791f75db9a640@mail.gmail.com>
	<Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>
	<65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
	<Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
	<Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>
	<65d96fc80910230452s6b4a179aqa5bc41dc34efc403@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910231610220.21492@srv.icf.org.ru>

On 2009-10-23 13:52 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>>
TC>> TC>3. can we control mediaWaitForConnect flag within setup message via
TC>> config
TC>> TC>file setting?
TC>>
TC>> what is mediaWaitForConnect flag, may be another trmin in h323?
TC>>
TC>>
TC>>
TC>If the calling endpoint sets the mediaWaitForConnect element to TRUE in the
TC>Setup message, then
TC>the called endpoint shall not send any media until after the Connect message
TC>is sent.
TC>The calling endpoint may begin transmitting media (according to the channels
TC>opened) immediately
TC>upon receiving a Q.931 message containing fastStart. Thus, the called
TC>endpoint must be prepared to
TC>immediately receive media on the channels it accepted in the Q.931 message
TC>containing fastStart.
TC>Note that national requirements may prohibit calling endpoints from
TC>transmitting media prior to
TC>receipt of a Connect message; it is the responsibility of the endpoint to
TC>comply with applicable
TC>requirements.
TC>
TC>
TC>check "H225_Setup_UUIE & H323SignalPDU::BuildSetup" within src/h323pdu.cxx
TC>(H323plus)

i think it can be implemented later, but, why it may be needed? can you explain some 
situation where it need?

TC>> TC>...
TC>>
TC>> may bee, while i in hospital i have a very limited ways for testing,
TC>> especially for inbound calls throuce h323. i find one issues in signaling
TC>> part in h323plus, src/h323.cxx grep "Very Frustrating - S.H." try uncomment
TC>> fast start handling there, may be it help.
TC>>
TC>>
TC>I'm not sure it is gonna help. This is only for CallProceeding having a
TC>faststart element... What i have is a progress message without a faststart
TC>element but with h245 address... it should go to StartControlChannel but i
TC>think it is stuck since you call pre_answer before it actually opens a LC.
TC>
TC>
TC>PBoolean H323Connection::OnReceivedProgress(const H323SignalPDU & pdu)
TC>{
TC>  if (pdu.m_h323_uu_pdu.m_h323_message_body.GetTag() !=
TC>H225_H323_UU_PDU_h323_message_body::e_progress)
TC>    return FALSE;
TC>  const H225_Progress_UUIE & progress =
TC>pdu.m_h323_uu_pdu.m_h323_message_body;
TC>
TC>  SetRemoteVersions(progress.m_protocolIdentifier);
TC>  SetRemotePartyInfo(pdu);
TC>  SetRemoteApplication(progress.m_destinationInfo);
TC>
TC>  // Check for fastStart data and start fast
TC>  if (progress.HasOptionalField(H225_Progress_UUIE::e_fastStart))
TC>    HandleFastStartAcknowledge(progress.m_fastStart);
TC>
TC>  // Check that it has the H.245 channel connection info
TC>  if (progress.HasOptionalField(H225_Progress_UUIE::e_h245Address))
TC>    return StartControlChannel(progress.m_h245Address);
TC>
TC>  return TRUE;
TC>}
TC>
TC>
TC>
TC>you should handle this and postpone pre_answer until you get an open LC.

pre_answer is not complete at this time, i say it a some kinde of hack, there is
another issues with it ans sofia in case proxy-media true.

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From tculjaga at gmail.com  Fri Oct 23 05:38:40 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 23 Oct 2009 14:38:40 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910231610220.21492@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>
	<65d96fc80910220753n6d2efa39l470791f75db9a640@mail.gmail.com>
	<Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>
	<65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
	<Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
	<Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>
	<65d96fc80910230452s6b4a179aqa5bc41dc34efc403@mail.gmail.com>
	<Pine.LNX.4.64.0910231610220.21492@srv.icf.org.ru>
Message-ID: <65d96fc80910230538w7105be6fve14956ab3b247bf6@mail.gmail.com>

>
> TC>check "H225_Setup_UUIE & H323SignalPDU::BuildSetup" within
> src/h323pdu.cxx
> TC>(H323plus)
>
> i think it can be implemented later, but, why it may be needed? can you
> explain some
> situation where it need?
>
>
> TC>
> TC>you should handle this and postpone pre_answer until you get an open LC.
>
> pre_answer is not complete at this time, i say it a some kinde of hack,
> there is
> another issues with it ans sofia in case proxy-media true.
>
>

bool FSH323Connection::OnReceivedProgress(const H323SignalPDU &pdu)
{
        PTRACE(4, "mod_h323\t======>FSH323Connection::OnReceivedProgress");
        m_txAudioOpened.Wait();
        switch_channel_mark_pre_answered(m_fsChannel);
        return true;
}



so for me the workaround for this was:




bool FSH323Connection::OnReceivedProgress(const H323SignalPDU &pdu)
{
        PTRACE(4, "mod_h323\t======>FSH323Connection::OnReceivedProgress");

        PTRACE(4, "mod_h323\t======>FSH323Connection::OnReceivedProgress -
disabled pre_answer!!!!");

        //m_txAudioOpened.Wait();
        //switch_channel_mark_pre_answered(m_fsChannel);
        return true;
}
-------------- next part --------------
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From bottleman at icf.org.ru  Fri Oct 23 05:53:32 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Fri, 23 Oct 2009 16:53:32 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910230538w7105be6fve14956ab3b247bf6@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910221805410.1929@srv.icf.org.ru>
	<65d96fc80910220753n6d2efa39l470791f75db9a640@mail.gmail.com>
	<Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>
	<65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
	<Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
	<Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>
	<65d96fc80910230452s6b4a179aqa5bc41dc34efc403@mail.gmail.com>
	<Pine.LNX.4.64.0910231610220.21492@srv.icf.org.ru>
	<65d96fc80910230538w7105be6fve14956ab3b247bf6@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910231646210.29582@srv.icf.org.ru>

On 2009-10-23 14:38 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>>
TC>> TC>check "H225_Setup_UUIE & H323SignalPDU::BuildSetup" within
TC>> src/h323pdu.cxx
TC>> TC>(H323plus)
TC>>
TC>> i think it can be implemented later, but, why it may be needed? can you
TC>> explain some
TC>> situation where it need?
TC>>
TC>>
TC>> TC>
TC>> TC>you should handle this and postpone pre_answer until you get an open LC.
TC>>
TC>> pre_answer is not complete at this time, i say it a some kinde of hack,
TC>> there is
TC>> another issues with it ans sofia in case proxy-media true.
TC>>
TC>>
TC>
TC>bool FSH323Connection::OnReceivedProgress(const H323SignalPDU &pdu)
TC>{
TC>        PTRACE(4, "mod_h323\t======>FSH323Connection::OnReceivedProgress");
TC>        m_txAudioOpened.Wait();
TC>        switch_channel_mark_pre_answered(m_fsChannel);
TC>        return true;
TC>}
TC>
TC>
TC>
TC>so for me the workaround for this was:
TC>
TC>
TC>
TC>
TC>bool FSH323Connection::OnReceivedProgress(const H323SignalPDU &pdu)
TC>{
TC>        PTRACE(4, "mod_h323\t======>FSH323Connection::OnReceivedProgress");
TC>
TC>        PTRACE(4, "mod_h323\t======>FSH323Connection::OnReceivedProgress -
TC>disabled pre_answer!!!!");
TC>
TC>        //m_txAudioOpened.Wait();
TC>        //switch_channel_mark_pre_answered(m_fsChannel);
TC>        return true;
TC>}
TC>

in that chase wee are not hear anything going inband if receive progress ind from called h323 endpoint, 
there will bee ringback, for exmaple mobule fone then it out of network. if you dont need
this make this changes until i fix it.

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From Prometheus001 at gmx.net  Fri Oct 23 06:08:40 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 23 Oct 2009 15:08:40 +0200
Subject: [Freeswitch-users] Force registering external gateways though
	OpenSIPS load balancer
Message-ID: <4AE1AAD8.9070908@gmx.net>

Hello,

in a freeswitch cluster (FS1 and FS2) behind an OpenSIPS I want
Freeswitch to register to external gateways through the OpenSIPS load
balancer, in order to later receive incoming calls through the load
balancer.
Is there a way to tell Freeswitch in it's Gateway definition to define
an additional path (e.g. fs_path) for it's registration, so that all
registrations go via the OpenSIPS load balancer?
Current flow
    FS1(register) => external_gateway (1st FS Machine)
    external_gateway(invite) =>FS1
    FS2(register) => external_gateway (2nd FS Machine)
    external_gateway(invite) =>FS2

Desired flow
    FS1(register) => OpenSIPS => external_gateway (1st FS Machine)
    external_gateway(invite) => OpenSIPS => FS1
    FS2(register) => OpenSIPS => external_gateway (2nd FS Machine)
    external_gateway(invite) => OpenSIPS =>FS2
I tried to add "fs_path" to the gateway definition, but had no success.
It still registered directly.

Maybe by adding via route information or modifying contact header?

Best regards
Peter.



From tculjaga at gmail.com  Fri Oct 23 06:14:55 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 23 Oct 2009 15:14:55 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910231646210.29582@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>
	<65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
	<Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
	<Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>
	<65d96fc80910230452s6b4a179aqa5bc41dc34efc403@mail.gmail.com>
	<Pine.LNX.4.64.0910231610220.21492@srv.icf.org.ru>
	<65d96fc80910230538w7105be6fve14956ab3b247bf6@mail.gmail.com>
	<Pine.LNX.4.64.0910231646210.29582@srv.icf.org.ru>
Message-ID: <65d96fc80910230614m2b2504d0r92006b78d267a399@mail.gmail.com>

>
>
> TC>
> TC>bool FSH323Connection::OnReceivedProgress(const H323SignalPDU &pdu)
> TC>{
> TC>        PTRACE(4,
> "mod_h323\t======>FSH323Connection::OnReceivedProgress");
> TC>
> TC>        PTRACE(4, "mod_h323\t======>FSH323Connection::OnReceivedProgress
> -
> TC>disabled pre_answer!!!!");
> TC>
> TC>        //m_txAudioOpened.Wait();
> TC>        //switch_channel_mark_pre_answered(m_fsChannel);
> TC>        return true;
> TC>}
> TC>
>
> in that chase wee are not hear anything going inband if receive progress
> ind from called h323 endpoint,
> there will bee ringback, for exmaple mobule fone then it out of network. if
> you dont need
> this make this changes until i fix it.
>
>

not true, because you have mediaWaitForConnect = false... the terminating
endpoint can send media before H.225 connect  message.... and this actually
works well :P



  7.317880   10.4.62.89 -> 10.4.62.7    SIP/SDP Request: INVITE
sip:00914392122 at singtel, with session description
  7.318319    10.4.62.7 -> 10.4.62.89   SIP Status: 100 Trying
  7.331430    10.4.62.7 -> 10.4.62.89   SIP Status: 407 Proxy Authentication
Required
  7.339420   10.4.62.89 -> 10.4.62.7    SIP Request: ACK
sip:00914392122 at singtel
  7.345078   10.4.62.89 -> 10.4.62.7    SIP/SDP Request: INVITE
sip:00914392122 at singtel, with session description
  7.345378    10.4.62.7 -> 10.4.62.89   SIP Status: 100 Trying
  7.387166    10.4.62.7 -> 10.4.4.254   H.225.0 CS: setup OpenLogicalChannel

  7.388636   10.4.4.254 -> 10.4.62.7    H.225.0 CS: callProceeding
  9.389852   10.4.4.254 -> 10.4.62.7    H.225.0 CS: progress
 10.639897   10.4.4.254 -> 10.4.62.7    H.225.0 CS: alerting
 10.651322    10.4.62.7 -> 10.4.62.89   SIP Status: 180 Ringing
 10.653932    10.4.62.7 -> 10.4.198.113 H.245 terminalCapabilitySet
 10.654565    10.4.62.7 -> 10.4.198.113 H.245 masterSlaveDetermination
 10.659757 10.4.198.113 -> 10.4.62.7    H.245 terminalCapabilitySet
 10.659814 10.4.198.113 -> 10.4.62.7    H.245 masterSlaveDetermination
 10.660161 10.4.198.113 -> 10.4.62.7    H.245 terminalCapabilitySetAck
 10.660238 10.4.198.113 -> 10.4.62.7    H.245 masterSlaveDeterminationAck
 10.666028    10.4.62.7 -> 10.4.198.113 H.245 terminalCapabilitySetAck
 10.670388    10.4.62.7 -> 10.4.198.113 H.245 masterSlaveDeterminationAck
 10.674693 10.4.198.113 -> 10.4.62.7    H.245 openLogicalChannel (g711A)
 10.682410    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 1
#678: OLC found 10.4.62.7/10.4.198.113/129
 10.683902    10.4.62.7 -> 10.4.198.113 H.245 openLogicalChannelAck
 10.687378    10.4.62.7 -> 10.4.198.113 H.245 openLogicalChannel (g711A)
#723: OLC found 10.4.198.113/10.4.62.7/108
 10.691579 10.4.198.113 -> 10.4.62.7    H.245 openLogicalChannelAck
 10.778413  10.4.142.38 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=0, Time=24640
 10.798476  10.4.142.38 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=1, Time=24800
 10.818432  10.4.142.38 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=2, Time=24960

<-------------- snip ------------->

 13.298358  10.4.142.38 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=126, Time=44800
 13.318460  10.4.142.38 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=127, Time=44960
 13.338405  10.4.142.38 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=128, Time=45120
 13.358353  10.4.142.38 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=129, Time=45280
 13.369984   10.4.4.254 -> 10.4.62.7    H.225.0 CS: connect
 13.378381  10.4.142.38 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=130, Time=45440
 13.382330    10.4.62.7 -> 10.4.62.89   SIP/SDP Status: 200 OK, with session
description
 13.388833    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 3
 13.389123    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 3
 13.396419   10.4.62.89 -> 10.4.62.7    SIP Request: ACK
sip:00914392122 at 10.4.62.7:5060;transport=udp
 13.398457  10.4.142.38 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=131, Time=45600
 13.405954   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMU,
SSRC=0xDEF4B36, Seq=27943, Time=991142687
 13.418401  10.4.142.38 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=132, Time=45760
 13.425864   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMU,
SSRC=0xDEF4B36, Seq=27944, Time=991142847
 13.438360  10.4.142.38 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=133, Time=45920
 13.438570    10.4.62.7 -> 10.4.62.89   RTP PT=ITU-T G.711 PCMA,
SSRC=0x172DD4B, Seq=46377, Time=640
 13.446202   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xDEF4B36, Seq=27945, Time=991143007
 13.458320  10.4.142.38 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=134, Time=46080
 13.458467    10.4.62.7 -> 10.4.62.89   RTP PT=ITU-T G.711 PCMA,
SSRC=0x172DD4B, Seq=46378, Time=800
 13.459008    10.4.62.7 -> 10.4.142.38  RTP PT=ITU-T G.711 PCMA,
SSRC=0xB9D8D8, Seq=1379, Time=991143007
 13.466010   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xDEF4B36, Seq=27946, Time=991143167
 13.478408  10.4.142.38 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=135, Time=46240
 13.478470    10.4.62.7 -> 10.4.142.38  RTP PT=ITU-T G.711 PCMA,
SSRC=0xB9D8D8, Seq=1380, Time=991143167
 13.478749    10.4.62.7 -> 10.4.62.89   RTP PT=ITU-T G.711 PCMA,
SSRC=0x172DD4B, Seq=46379, Time=960

 <------------ snip -------------->

 15.098561    10.4.62.7 -> 10.4.62.89   RTP PT=ITU-T G.711 PCMA,
SSRC=0x172DD4B, Seq=46460, Time=13920
 15.099011    10.4.62.7 -> 10.4.142.38  RTP PT=ITU-T G.711 PCMA,
SSRC=0xB9D8D8, Seq=1461, Time=991156127
 15.105847   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xDEF4B36, Seq=28028, Time=991156287
 15.118353  10.4.142.38 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=217, Time=59360
 15.118434    10.4.62.7 -> 10.4.62.89   RTP PT=ITU-T G.711 PCMA,
SSRC=0x172DD4B, Seq=46461, Time=14080
 15.119540    10.4.62.7 -> 10.4.142.38  RTP PT=ITU-T G.711 PCMA,
SSRC=0xB9D8D8, Seq=1462, Time=991156287
 15.122951 10.4.198.113 -> 10.4.62.7    H.245 closeLogicalChannel
 15.122986 10.4.198.113 -> 10.4.62.7    H.245 endSessionCommand
 15.125003    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 3
 15.125257    10.4.62.7 -> 10.4.198.113 H.245 closeLogicalChannelAck
 15.125857   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xDEF4B36, Seq=28029, Time=991156447
 15.127900    10.4.62.7 -> 10.4.198.113 H.245 endSessionCommand
 15.128461    10.4.62.7 -> 10.4.4.254   H.225.0 CS: releaseComplete
 15.138328  10.4.142.38 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=218, Time=59520
 15.139550    10.4.62.7 -> 10.4.62.7    RTP Unknown RTP version 3
 15.142189    10.4.62.7 -> 10.4.62.89   SIP Request: BYE
sip:1001 at 10.4.62.89<sip%3A1001 at 10.4.62.89>
;transport=udp
 15.145990   10.4.62.89 -> 10.4.62.7    RTP PT=ITU-T G.711 PCMA,
SSRC=0xDEF4B36, Seq=28030, Time=991156607
 15.146010    10.4.62.7 -> 10.4.62.89   ICMP Destination unreachable (Port
unreachable)
 15.150016   10.4.62.89 -> 10.4.62.7    SIP Status: 200 OK



 so the real solution is to implement a check for CallProceeding , Progress
and Facility message whether it has a faststart element included. It it is
true than you might start pre_answer.


also, i don't see any handling of Call Proceeding ... what if there is a
fastStart element in CallProceeding message? :)


T.

T.
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From zyyzj2 at gmail.com  Thu Oct 22 23:40:57 2009
From: zyyzj2 at gmail.com (zyyzj)
Date: Fri, 23 Oct 2009 14:40:57 +0800
Subject: [Freeswitch-users] dynamic gateway
Message-ID: <61ee6b990910222340r49ae8fa4oafa1bd4acfecd71c@mail.gmail.com>

hello, all,

    I want bridge call to gateway without register.

    Example:

    <action application="bridge" data="sofia/sipinterface_1/$
1 at 192.168.111.101:5060"/>

    Error is : sofia_reg.c:1568 No Matching gateway found

    My purpose is the gateway is realtime changed from script(etc lua,
php...).

    How can I do that?


-- 
We Create The World!
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From sharad at coraltele.com  Fri Oct 23 04:36:05 2009
From: sharad at coraltele.com (sharad)
Date: Fri, 23 Oct 2009 17:06:05 +0530
Subject: [Freeswitch-users] How to reset the personal greeting
Message-ID: <002001ca53d5$01b5c6b0$0c04a8c0@compaq77db609e>

Hi

I am new to Freeswitch so my question may be a stupid question.

I just want to know how to disable the personal greeting to the default one. One user has recorded his personal greeting now how can he make this default.

I could not find any option for the same.

Plz advice.

Thanks & regards
Sharad garg
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From mattdfong at gmail.com  Fri Oct 23 06:46:53 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Fri, 23 Oct 2009 20:46:53 +0700
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
In-Reply-To: <4256bf830910222019x4bd7929eoadcb757294cc37d2@mail.gmail.com>
References: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
	<4256bf830908041151l55980e3fifb85c75b87535426@mail.gmail.com>
	<3D0B855A-5B2C-4C6F-9306-64B0ECE97091@jerris.com>
	<4256bf830910180725y72539d98gb138934afc18d58b@mail.gmail.com>
	<A5ECDCD9-1A12-4A31-89F0-E6F4C79FE9E9@jerris.com>
	<4256bf830910182148mf7f2bf9vdcccd687e0caa1e7@mail.gmail.com>
	<87f2f3b90910191144q25750bayfd7a7448e82e78ff@mail.gmail.com>
	<4256bf830910221042l68ea769cg532bbd339bae0790@mail.gmail.com>
	<191c3a030910221226j167e0adw5f631f759cd7cfa0@mail.gmail.com>
	<4256bf830910222019x4bd7929eoadcb757294cc37d2@mail.gmail.com>
Message-ID: <4256bf830910230646j506e3629t7223e4fdc04026ce@mail.gmail.com>

Is there a way I can reset the forwardvars for a channel or execute some
action that would reset this value? I'm pretty certain there is no endless
looping going on in my dial plan.
Thanks.

--matt

On Fri, Oct 23, 2009 at 10:19 AM, Matthew Fong <mattdfong at gmail.com> wrote:

> Hi Anthm,
> Thanks for the response. I'll use pastebin from now on.
>
> How would I go about fixing this so my calls do not get hung-up?
>
> --matt
>
>
> On Fri, Oct 23, 2009 at 2:26 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> 1) please do not flood emails with traces reference pastebins
>> 2) the error is a routing loop being prevented so you are looping the call
>> into your same box too many times.
>>
>> You could have figured this out your self by looking at one of the lines
>> of code where it reported the hangup.
>>
>>  if (!switch_strlen_zero(forwardvar)) {
>>         forwardval = atoi(forwardvar) - 1;
>>     }
>>     if (forwardval <= 0) {
>>         switch_channel_hangup(channel,
>> SWITCH_CAUSE_EXCHANGE_ROUTING_ERROR);
>>         return SWITCH_STATUS_FALSE;
>>
>>     }
>>
>>
>> On Thu, Oct 22, 2009 at 12:42 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>>
>>> I figured out how to turn debug mode on. For some reason F8 didn't work,
>>> so I just set the default debug mode to DEBUG in switch.conf.xml
>>>
>>> http://pastebin.freeswitch.org/10805
>>> <http://pastebin.freeswitch.org/10805>So my program does more or less
>>> the same sequence for an Agent every time when connected to another party.
>>> They uuid_kill the bridged parties uuid, and then are transfered to an
>>> extension that park's the agent (using the transfer_after_bridge variable --
>>> in my dialplan extn 1998). In some instances the agent is then transfered to
>>> another extension, played a wav file, and then park'ed again
>>>
>>> Sometimes, FreeSWITCH unexpectidly hangsup the Agent channel citing an
>>> EXCHANGE_ROUTING_ERROR. It seems (to me to happen at random times)...and
>>> it's more prevelent with some did providers than others. With
>>> didforsale.org it happens about one ever 5 bridges. With icall.com it
>>> only happens about once every 30 bridges. With IPKall it happens rarely. I
>>> can't figure it out, and it's driving me crazy :)
>>>
>>>
>>> A GOOD UUID_KILL SEQUENCE:
>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_ivr_play_say.c:1432 done
>>> playing file
>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:340
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXECUTE going to
>>> sleep
>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>> CS_HIBERNATE
>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:355
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE
>>> 2009-10-22 17:22:31.741845 [DEBUG] mod_sofia.c:160 sofia/external/+
>>> 14158867717 at 199.173.100.16:5060 SOFIA HIBERNATE
>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_ivr_bridge.c:650
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>> CS_HIBERNATE -> CS_RESET
>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:355
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE going
>>> to sleep
>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>> CS_RESET
>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:336
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET
>>> 2009-10-22 17:22:31.741845 [DEBUG] mod_sofia.c:145 sofia/external/+
>>> 14158867717 at 199.173.100.16:5060 SOFIA RESET
>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_ivr_bridge.c:637
>>> sofia/external/+14158867717 at 199.173.100.16:5060 CUSTOM RESET
>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:67
>>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard RESET
>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:336
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET going to
>>> sleep
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:340
>>> (sofia/external/14159927717) State EXECUTE going to sleep
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/14159927717) Running State Change CS_HIBERNATE
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:355
>>> (sofia/external/14159927717) State HIBERNATE
>>> 2009-10-22 17:22:31.748086 [DEBUG] mod_sofia.c:160
>>> sofia/external/14159927717 SOFIA HIBERNATE
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:650
>>> (sofia/external/14159927717) State Change CS_HIBERNATE -> CS_RESET
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/14159927717 [BREAK]
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:355
>>> (sofia/external/14159927717) State HIBERNATE going to sleep
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/14159927717) Running State Change CS_RESET
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:336
>>> (sofia/external/14159927717) State RESET
>>> 2009-10-22 17:22:31.748086 [DEBUG] mod_sofia.c:145
>>> sofia/external/14159927717 SOFIA RESET
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:637
>>> sofia/external/14159927717 CUSTOM RESET
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:642
>>> (sofia/external/14159927717) State Change CS_RESET -> CS_SOFT_EXECUTE
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/14159927717 [BREAK]
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:336
>>> (sofia/external/14159927717) State RESET going to sleep
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/14159927717) Running State Change CS_SOFT_EXECUTE
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:346
>>> (sofia/external/14159927717) State SOFT_EXECUTE
>>> 2009-10-22 17:22:31.748086 [DEBUG] mod_sofia.c:442 SOFIA TRANSMIT
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:660
>>> sofia/external/14159927717 CUSTOM SOFT_EXECUTE
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_originate.c:674
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_RESET
>>> -> CS_SOFT_EXECUTE
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:120
>>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message
>>> [DISPLAY]
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:120
>>> sofia/external/14159927717 receive message [DISPLAY]
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:924
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>> CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:968
>>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message [BRIDGE]
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:630 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:975
>>> sofia/external/14159927717 receive message [BRIDGE]
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:630 Send signal
>>> sofia/external/14159927717 [BREAK]
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:1019
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:22:31.749553 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>> CS_EXCHANGE_MEDIA
>>> 2009-10-22 17:22:31.749553 [DEBUG] switch_core_state_machine.c:343
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
>>> 2009-10-22 17:22:31.749553 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK
>>> multipler is being overrided to1
>>> 2009-10-22 17:22:34.300662 [INFO] switch_cpp.cpp:1116 BG_AG_GEN_CALLS,
>>> Agents total:1 avail:0 OutChan:0 AR:0 +-Calls:0:Q Call NYNoc
>>> multipler is being overrided to1
>>> 2009-10-22 17:22:39.303041 [INFO] switch_cpp.cpp:1116 BG_AG_GEN_CALLS,
>>> Agents total:1 avail:0 OutChan:0 AR:0 +-Calls:0:Q Call NYNoc
>>> 2009-10-22 17:22:40.626379 [NOTICE] sofia.c:328 Hangup
>>> sofia/external/14159927717 [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
>>> 2009-10-22 17:22:40.626379 [DEBUG] switch_channel.c:1810 Send signal
>>> sofia/external/14159927717 [KILL]
>>> 2009-10-22 17:22:40.626379 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/14159927717 [BREAK]
>>> active_true=1
>>> --contact NOT in playmsg or conference, update AGENT
>>> 2009-10-22 17:22:40.626379 [DEBUG] switch_core_state_machine.c:437 thread
>>> mismatch skipping state handler.
>>> 2009-10-22 17:22:40.626379 [INFO] switch_cpp.cpp:1116
>>> EVENTC_HANGUP_CONTACT, Q_CONTACT fired for agent:1/1 for qcall:1
>>> contact:4801442/4
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:440
>>> sofia/external/14159927717 ending bridge by request from write function
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:495
>>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message
>>> [UNBRIDGE]
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:630 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
>>> DONE [sofia/external/+14158867717 at 199.173.100.16:5060]
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:523 Send signal
>>> sofia/external/14159927717 [BREAK]
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr.c:1361 (sofia/external/+
>>> 14158867717 at 199.173.100.16:5060) State Change CS_EXCHANGE_MEDIA ->
>>> CS_ROUTING
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr.c:1365 sofia/external/+
>>> 14158867717 at 199.173.100.16:5060 receive message [TRANSFER]
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:630 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:22:40.639760 [NOTICE] switch_ivr.c:1367 Transfer
>>> sofia/external/+14158867717 at 199.173.100.16:5060 to xml[1998 at default]
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:343
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
>>> going to sleep
>>>  2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>> CS_ROUTING
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:333
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State ROUTING
>>> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:130 sofia/external/+
>>> 14158867717 at 199.173.100.16:5060 SOFIA ROUTING
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:78
>>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard ROUTING
>>> 2009-10-22 17:22:40.639760 [INFO] mod_dialplan_xml.c:391 Processing
>>> +14158867717->1998 in context default
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->tod_example] continue=true
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (PASS)
>>> [tod_example] ${strftime(%w)}(4) =~ /^([1-5])$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (PASS)
>>> [tod_example] ${strftime(%H%M)}(1722) =~ /^((09|1[0-7])[0-5][0-9]|1800)$/
>>> break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Action
>>> set(open=true)
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->ivr_demo] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [ivr_demo] destination_number(1998) =~ /^5000$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1975] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1975] destination_number(1998) =~ /^1975$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1971] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1971] destination_number(1998) =~ /^1971$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1972] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1972] destination_number(1998) =~ /^1972$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1920] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1920] destination_number(1998) =~ /^1920$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1981] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1981] destination_number(1998) =~ /^1981$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1982] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1982] destination_number(1998) =~ /^1982$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1481] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1481] destination_number(1998) =~ /^1481$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1989] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1989] destination_number(1998) =~ /^1989$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1990] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1990] destination_number(1998) =~ /^1990$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1991] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1991] destination_number(1998) =~ /^1991$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1992] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1992] destination_number(1998) =~ /^1992$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1993] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1993] destination_number(1998) =~ /^1993$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1994] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1994] destination_number(1998) =~ /^1994$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1995] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1995] destination_number(1998) =~ /^1995$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1996] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1996] destination_number(1998) =~ /^1996$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1997] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>> [1997] destination_number(1998) =~ /^1997$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>> [default->1998] continue=false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (PASS)
>>> [1998] destination_number(1998) =~ /^1998$/ break=on-false
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Action
>>> set(transfer_after_bridge=1998)
>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Action park()
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:114
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>> CS_ROUTING -> CS_EXECUTE
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:333
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State ROUTING going to
>>> sleep
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>> CS_EXECUTE
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:340
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXECUTE
>>> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:173 sofia/external/+
>>> 14158867717 at 199.173.100.16:5060 SOFIA EXECUTE
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:151
>>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard EXECUTE
>>> EXECUTE sofia/external/+14158867717 at 199.173.100.16:5060 set(open=true)
>>> 2009-10-22 17:22:40.639760 [DEBUG] mod_dptools.c:752 sofia/external/+
>>> 14158867717 at 199.173.100.16:5060 SET [open]=[true]
>>> EXECUTE sofia/external/+14158867717 at 199.173.100.16:5060set(transfer_after_bridge=1998)
>>> 2009-10-22 17:22:40.639760 [DEBUG] mod_dptools.c:752 sofia/external/+
>>> 14158867717 at 199.173.100.16:5060 SET [transfer_after_bridge]=[1998]
>>> EXECUTE sofia/external/+14158867717 at 199.173.100.16:5060 park()
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:446
>>> sofia/external/14159927717 ending bridge by request from read function
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
>>> DONE [sofia/external/14159927717]
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:523 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> subtracting 9 seconds for hh_call: 1095422
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:346
>>> (sofia/external/14159927717) State SOFT_EXECUTE going to sleep
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/14159927717) Running State Change CS_HANGUP
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:464
>>> (sofia/external/14159927717) State HANGUP
>>> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:344 Channel
>>> sofia/external/14159927717 hanging up, cause: NORMAL_CLEARING
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:46
>>> sofia/external/14159927717 Standard HANGUP, cause: NORMAL_CLEARING
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:464
>>> (sofia/external/14159927717) State HANGUP going to sleep
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:325
>>> (sofia/external/14159927717) State Change CS_HANGUP -> CS_REPORTING
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/14159927717 [BREAK]
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/14159927717) Running State Change CS_REPORTING
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:555
>>> (sofia/external/14159927717) State REPORTING
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:53
>>> sofia/external/14159927717 Standard REPORTING, cause: NORMAL_CLEARING
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:555
>>> (sofia/external/14159927717) State REPORTING going to sleep
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:319
>>> (sofia/external/14159927717) State Change CS_REPORTING -> CS_DESTROY
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/14159927717 [BREAK]
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:1069 Session 6
>>> (sofia/external/14159927717) Locked, Waiting on external entities
>>> 2009-10-22 17:22:40.639760 [NOTICE] switch_core_session.c:1087 Session 6
>>> (sofia/external/14159927717) Ended
>>> 2009-10-22 17:22:40.639760 [NOTICE] switch_core_session.c:1089 Close
>>> Channel sofia/external/14159927717 [CS_DESTROY]
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:401
>>> (sofia/external/14159927717) Running State Change CS_DESTROY
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:412
>>> (sofia/external/14159927717) State DESTROY
>>> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:261
>>> sofia/external/14159927717 SOFIA DESTROY
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:60
>>> sofia/external/14159927717 Standard DESTROY
>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:412
>>> (sofia/external/14159927717) State DESTROY going to sleep
>>> 2009-10-22 17:22:40.639760 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>>> teliax.com hc:NORMAL_CLEARING du:9 cn:sofia/external/14159927717
>>>
>>>
>>>
>>> A BAD UUID_KILL SEQUENCE: (where the agent is hungup, unexpectedly)
>>>
>>> --bridged in: 0.005821 seconds
>>> 2009-10-22 17:23:26.718930 [INFO] switch_cpp.cpp:1116 AG_CBRIDGE,
>>> connected agent:1/1 contact:4801439/22/1/1
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_play_say.c:1432 done
>>> playing file
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:340
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXECUTE going to
>>> sleep
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>> CS_HIBERNATE
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE
>>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:160 sofia/external/+
>>> 14158867717 at 199.173.100.16:5060 SOFIA HIBERNATE
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:650
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>> CS_HIBERNATE -> CS_RESET
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE going
>>> to sleep
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>> CS_RESET
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET
>>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:145 sofia/external/+
>>> 14158867717 at 199.173.100.16:5060 SOFIA RESET
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:637
>>> sofia/external/+14158867717 at 199.173.100.16:5060 CUSTOM RESET
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:67
>>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard RESET
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET going to
>>> sleep
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:340
>>> (sofia/external/14159927717) State EXECUTE going to sleep
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/14159927717) Running State Change CS_HIBERNATE
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
>>> (sofia/external/14159927717) State HIBERNATE
>>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:160
>>> sofia/external/14159927717 SOFIA HIBERNATE
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:650
>>> (sofia/external/14159927717) State Change CS_HIBERNATE -> CS_RESET
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/14159927717 [BREAK]
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
>>> (sofia/external/14159927717) State HIBERNATE going to sleep
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/14159927717) Running State Change CS_RESET
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
>>> (sofia/external/14159927717) State RESET
>>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:145
>>> sofia/external/14159927717 SOFIA RESET
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:637
>>> sofia/external/14159927717 CUSTOM RESET
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:642
>>> (sofia/external/14159927717) State Change CS_RESET -> CS_SOFT_EXECUTE
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/14159927717 [BREAK]
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
>>> (sofia/external/14159927717) State RESET going to sleep
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/14159927717) Running State Change CS_SOFT_EXECUTE
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:346
>>> (sofia/external/14159927717) State SOFT_EXECUTE
>>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:442 SOFIA TRANSMIT
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:660
>>> sofia/external/14159927717 CUSTOM SOFT_EXECUTE
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_originate.c:674
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_RESET
>>> -> CS_SOFT_EXECUTE
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:120
>>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message
>>> [DISPLAY]
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:120
>>> sofia/external/14159927717 receive message [DISPLAY]
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:924
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>> CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:968
>>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message [BRIDGE]
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:630 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:975
>>> sofia/external/14159927717 receive message [BRIDGE]
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:630 Send signal
>>> sofia/external/14159927717 [BREAK]
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:1019
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>> CS_EXCHANGE_MEDIA
>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:343
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
>>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK
>>> 2009-10-22 17:23:26.740065 [DEBUG] switch_core_io.c:234 sofia/external/+
>>> 14158867717 at 199.173.100.16:5060 receive message [TRANSCODING_NECESSARY]
>>> 2009-10-22 17:23:27.319618 [DEBUG] switch_core_media_bug.c:393 Removing
>>> BUG from sofia/external/+14158867717 at 199.173.100.16:5060
>>> multipler is being overrided to1
>>> 2009-10-22 17:23:29.406414 [INFO] switch_cpp.cpp:1116 BG_AG_GEN_CALLS,
>>> Agents total:1 avail:0 OutChan:0 AR:0 +-Calls:0:Q Call NYNoc
>>> 2009-10-22 17:23:31.873352 [NOTICE] mod_commands.c:1523 Hangup
>>> sofia/external/14159927717 [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
>>> 2009-10-22 17:23:31.874411 [INFO] switch_cpp.cpp:1116
>>> EVENTC_HANGUP_CONTACT, Q_CONTACT fired for agent:1/1 for qcall:1
>>> contact:4801439/1
>>> active_true=1
>>> --contact NOT in playmsg or conference, update AGENT
>>> 2009-10-22 17:23:31.875475 [DEBUG] switch_channel.c:1810 Send signal
>>> sofia/external/14159927717 [KILL]
>>> 2009-10-22 17:23:31.875475 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/14159927717 [BREAK]
>>> 2009-10-22 17:23:31.875475 [DEBUG] switch_core_state_machine.c:437 thread
>>> mismatch skipping state handler.
>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:446
>>> sofia/external/14159927717 ending bridge by request from read function
>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
>>> DONE [sofia/external/14159927717]
>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:523 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:495
>>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message
>>> [UNBRIDGE]
>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_session.c:630 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
>>> DONE [sofia/external/+14158867717 at 199.173.100.16:5060]
>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:523 Send signal
>>> sofia/external/14159927717 [BREAK]
>>> 2009-10-22 17:23:31.882153 [NOTICE] switch_ivr.c:1268 Hangup
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [CS_EXCHANGE_MEDIA]
>>> [EXCHANGE_ROUTING_ERROR]
>>>  2009-10-22 17:23:31.882153 [DEBUG] switch_channel.c:1810 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [KILL]
>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_state_machine.c:464
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HANGUP
>>> 2009-10-22 17:23:31.882153 [DEBUG] mod_sofia.c:344 Channel
>>> sofia/external/+14158867717 at 199.173.100.16:5060 hanging up, cause:
>>> EXCHANGE_ROUTING_ERROR
>>> 2009-10-22 17:23:31.882153 [DEBUG] mod_sofia.c:382 Sending BYE to
>>> sofia/external/+14158867717 at 199.173.100.16:5060
>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_state_machine.c:46
>>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard HANGUP, cause:
>>> EXCHANGE_ROUTING_ERROR
>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_state_machine.c:464
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HANGUP going to
>>> sleep
>>> 2009-10-22 17:23:31.884357 [INFO] switch_cpp.cpp:1116 HH_HANGUP,
>>> agent:1/1 uuid: f8604694-4e44-4c4c-9805-722b3e9d6b68
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:346
>>> (sofia/external/14159927717) State SOFT_EXECUTE going to sleep
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/14159927717) Running State Change CS_HANGUP
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:464
>>> (sofia/external/14159927717) State HANGUP
>>> 2009-10-22 17:23:31.885412 [DEBUG] mod_sofia.c:344 Channel
>>> sofia/external/14159927717 hanging up, cause: NORMAL_CLEARING
>>> 2009-10-22 17:23:31.885412 [DEBUG] mod_sofia.c:382 Sending BYE to
>>> sofia/external/14159927717
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:46
>>> sofia/external/14159927717 Standard HANGUP, cause: NORMAL_CLEARING
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:464
>>> (sofia/external/14159927717) State HANGUP going to sleep
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:325
>>> (sofia/external/14159927717) State Change CS_HANGUP -> CS_REPORTING
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/14159927717 [BREAK]
>>> subtracting 5 seconds for hh_call: 1095423
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/14159927717) Running State Change CS_REPORTING
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:555
>>> (sofia/external/14159927717) State REPORTING
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:53
>>> sofia/external/14159927717 Standard REPORTING, cause: NORMAL_CLEARING
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:555
>>> (sofia/external/14159927717) State REPORTING going to sleep
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:319
>>> (sofia/external/14159927717) State Change CS_REPORTING -> CS_DESTROY
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/14159927717 [BREAK]
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_session.c:1069 Session 7
>>> (sofia/external/14159927717) Locked, Waiting on external entities
>>> 2009-10-22 17:23:31.885412 [NOTICE] switch_core_session.c:1087 Session 7
>>> (sofia/external/14159927717) Ended
>>> 2009-10-22 17:23:31.885412 [NOTICE] switch_core_session.c:1089 Close
>>> Channel sofia/external/14159927717 [CS_DESTROY]
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:401
>>> (sofia/external/14159927717) Running State Change CS_DESTROY
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:412
>>> (sofia/external/14159927717) State DESTROY
>>> 2009-10-22 17:23:31.885412 [DEBUG] mod_sofia.c:261
>>> sofia/external/14159927717 SOFIA DESTROY
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:60
>>> sofia/external/14159927717 Standard DESTROY
>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:412
>>> (sofia/external/14159927717) State DESTROY going to sleep
>>> 2009-10-22 17:23:31.885412 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>>> teliax.com hc:NORMAL_CLEARING du:5 cn:sofia/external/14159927717
>>> 2009-10-22 17:23:31.885412 [INFO] switch_cpp.cpp:1116
>>> EVENTC_BILLING_CEIL, subtracting 5 for hh_call:1095423
>>> 2009-10-22 17:23:31.901620 [DEBUG] switch_core_state_machine.c:494 Hangup
>>> Command luarun(hh_hangup.lua f8604694-4e44-4c4c-9805-722b3e9d6b68 1 1):
>>> +OK
>>>
>>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:343
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
>>> going to sleep
>>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>> CS_HANGUP
>>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:442
>>> handler already called, skipping state handler.
>>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:325
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_HANGUP
>>> -> CS_REPORTING
>>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:306
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>> CS_REPORTING
>>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:555
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State REPORTING
>>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:53
>>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard REPORTING,
>>> cause: EXCHANGE_ROUTING_ERROR
>>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:555
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State REPORTING going
>>> to sleep
>>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:319
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>> CS_REPORTING -> CS_DESTROY
>>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_session.c:1069 Session 4
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Locked, Waiting on
>>> external entities
>>> 2009-10-22 17:23:31.903742 [NOTICE] switch_core_session.c:1087 Session 4
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Ended
>>> 2009-10-22 17:23:31.903742 [NOTICE] switch_core_session.c:1089 Close
>>> Channel sofia/external/+14158867717 at 199.173.100.16:5060 [CS_DESTROY]
>>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:401
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>> CS_DESTROY
>>> 2009-10-22 17:23:31.904799 [DEBUG] switch_core_state_machine.c:412
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State DESTROY
>>> 2009-10-22 17:23:31.904799 [DEBUG] mod_sofia.c:261 sofia/external/+
>>> 14158867717 at 199.173.100.16:5060 SOFIA DESTROY
>>> 2009-10-22 17:23:31.904799 [DEBUG] switch_core_state_machine.c:60
>>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard DESTROY
>>> 2009-10-22 17:23:31.904799 [DEBUG] switch_core_state_machine.c:412
>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State DESTROY going to
>>> sleep
>>>
>>>
>>>
>>> On Tue, Oct 20, 2009 at 1:44 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>>
>>>>
>>>> On Sun, Oct 18, 2009 at 9:48 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>>
>>>>> The debug level logs to the console, right? The pastebin, had log level
>>>>> debug, sofia trace on for external and default, and sofia loglevel all 9. Is
>>>>> there another log enable command I'm missing? It seems loglevel all 9
>>>>> outputs enter and exit functions, but at least to my novice eye, it's not
>>>>> too obvious why freeswitch is sending a BYE to my DID provider.
>>>>>
>>>>> Evidently the console loglevel debug did not take because there isn't a
>>>> single DEBUG level line of output. Please try again. Just remember that
>>>> debug level log messages will have A LOT of information so be patient while
>>>> sifting through it. Of course, feel free to pastebin the output and add a
>>>> link to it in this email thread.
>>>>
>>>> -MC
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From anthony.minessale at gmail.com  Fri Oct 23 07:33:01 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 23 Oct 2009 09:33:01 -0500
Subject: [Freeswitch-users] INFO-Messages: Send_display in pre_answer
	state doesnt work anymore
In-Reply-To: <4AE19D0E.9050301@ewetel.de>
References: <4AE19D0E.9050301@ewetel.de>
Message-ID: <191c3a030910230733i4fce1c55v892056f549f67ed0@mail.gmail.com>

There was a concern with letting this happen before answer.
I changed it in last revision, try it out.
The problem is that the way polycom does it, it violates the RFC.

Also know you can set sip_callee_id_name and sip_callee_id_number before you
send a ring_ready pre_answer or answer to send the display update in that
packet instead of sending a separate one.


On Fri, Oct 23, 2009 at 7:09 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> when I use pre_answer app and right after that the send_display app,
> there is no INFO message anymore.
>
> My dialplan condition:
>
> <condition field="${sip_hangup_phrase}" expression="^Do Not Disturb"
> break="on-true">
>                        <action application="pre_answer"/>
>                        <action application="send_display" data="Bitte
> nicht stoeren"/>
>                        <action application="sleep" data="500"/>
>                        <action application="playback"
> data="ansagen/22.wav"/>
> </condition>
>
> changing "pre_answer" to "answer" works, but I don't want to answer the
> call for error messages.
>
> I use latest SVN trunk.
>
> regards
> helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
>
> iD8DBQFK4Z0O4tZeNddg3dwRAkB/AKCLN9luR0KGunHVGso2t3wJIo4dqQCdGdjp
> CY+yWGyVlyDpDs//WFtCqiA=
> =eG4W
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Fri Oct 23 07:36:31 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 23 Oct 2009 09:36:31 -0500
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
In-Reply-To: <4256bf830910230646j506e3629t7223e4fdc04026ce@mail.gmail.com>
References: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
	<3D0B855A-5B2C-4C6F-9306-64B0ECE97091@jerris.com>
	<4256bf830910180725y72539d98gb138934afc18d58b@mail.gmail.com>
	<A5ECDCD9-1A12-4A31-89F0-E6F4C79FE9E9@jerris.com>
	<4256bf830910182148mf7f2bf9vdcccd687e0caa1e7@mail.gmail.com>
	<87f2f3b90910191144q25750bayfd7a7448e82e78ff@mail.gmail.com>
	<4256bf830910221042l68ea769cg532bbd339bae0790@mail.gmail.com>
	<191c3a030910221226j167e0adw5f631f759cd7cfa0@mail.gmail.com>
	<4256bf830910222019x4bd7929eoadcb757294cc37d2@mail.gmail.com>
	<4256bf830910230646j506e3629t7223e4fdc04026ce@mail.gmail.com>
Message-ID: <191c3a030910230736o7a4201e8s4addd39d9e1c83ae@mail.gmail.com>

you can set the variable max_forwards to a high number.
It does not magically set itself so you must be looping through many
transfers at least.


On Fri, Oct 23, 2009 at 8:46 AM, Matthew Fong <mattdfong at gmail.com> wrote:

> Is there a way I can reset the forwardvars for a channel or execute some
> action that would reset this value? I'm pretty certain there is no endless
> looping going on in my dial plan.
> Thanks.
>
> --matt
>
>
> On Fri, Oct 23, 2009 at 10:19 AM, Matthew Fong <mattdfong at gmail.com>wrote:
>
>> Hi Anthm,
>> Thanks for the response. I'll use pastebin from now on.
>>
>> How would I go about fixing this so my calls do not get hung-up?
>>
>> --matt
>>
>>
>> On Fri, Oct 23, 2009 at 2:26 AM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> 1) please do not flood emails with traces reference pastebins
>>> 2) the error is a routing loop being prevented so you are looping the
>>> call into your same box too many times.
>>>
>>> You could have figured this out your self by looking at one of the lines
>>> of code where it reported the hangup.
>>>
>>>  if (!switch_strlen_zero(forwardvar)) {
>>>         forwardval = atoi(forwardvar) - 1;
>>>     }
>>>     if (forwardval <= 0) {
>>>         switch_channel_hangup(channel,
>>> SWITCH_CAUSE_EXCHANGE_ROUTING_ERROR);
>>>         return SWITCH_STATUS_FALSE;
>>>
>>>     }
>>>
>>>
>>> On Thu, Oct 22, 2009 at 12:42 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>
>>>> I figured out how to turn debug mode on. For some reason F8 didn't work,
>>>> so I just set the default debug mode to DEBUG in switch.conf.xml
>>>>
>>>> http://pastebin.freeswitch.org/10805
>>>> <http://pastebin.freeswitch.org/10805>So my program does more or less
>>>> the same sequence for an Agent every time when connected to another party.
>>>> They uuid_kill the bridged parties uuid, and then are transfered to an
>>>> extension that park's the agent (using the transfer_after_bridge variable --
>>>> in my dialplan extn 1998). In some instances the agent is then transfered to
>>>> another extension, played a wav file, and then park'ed again
>>>>
>>>> Sometimes, FreeSWITCH unexpectidly hangsup the Agent channel citing an
>>>> EXCHANGE_ROUTING_ERROR. It seems (to me to happen at random times)...and
>>>> it's more prevelent with some did providers than others. With
>>>> didforsale.org it happens about one ever 5 bridges. With icall.com it
>>>> only happens about once every 30 bridges. With IPKall it happens rarely. I
>>>> can't figure it out, and it's driving me crazy :)
>>>>
>>>>
>>>> A GOOD UUID_KILL SEQUENCE:
>>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_ivr_play_say.c:1432 done
>>>> playing file
>>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:340
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXECUTE going
>>>> to sleep
>>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>>> CS_HIBERNATE
>>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:355
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE
>>>> 2009-10-22 17:22:31.741845 [DEBUG] mod_sofia.c:160 sofia/external/+
>>>> 14158867717 at 199.173.100.16:5060 SOFIA HIBERNATE
>>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_ivr_bridge.c:650
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>>> CS_HIBERNATE -> CS_RESET
>>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:355
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE going
>>>> to sleep
>>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>>> CS_RESET
>>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:336
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET
>>>> 2009-10-22 17:22:31.741845 [DEBUG] mod_sofia.c:145 sofia/external/+
>>>> 14158867717 at 199.173.100.16:5060 SOFIA RESET
>>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_ivr_bridge.c:637
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 CUSTOM RESET
>>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:67
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard RESET
>>>> 2009-10-22 17:22:31.741845 [DEBUG] switch_core_state_machine.c:336
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET going to
>>>> sleep
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:340
>>>> (sofia/external/14159927717) State EXECUTE going to sleep
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/14159927717) Running State Change CS_HIBERNATE
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:355
>>>> (sofia/external/14159927717) State HIBERNATE
>>>> 2009-10-22 17:22:31.748086 [DEBUG] mod_sofia.c:160
>>>> sofia/external/14159927717 SOFIA HIBERNATE
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:650
>>>> (sofia/external/14159927717) State Change CS_HIBERNATE -> CS_RESET
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/14159927717 [BREAK]
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:355
>>>> (sofia/external/14159927717) State HIBERNATE going to sleep
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/14159927717) Running State Change CS_RESET
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:336
>>>> (sofia/external/14159927717) State RESET
>>>> 2009-10-22 17:22:31.748086 [DEBUG] mod_sofia.c:145
>>>> sofia/external/14159927717 SOFIA RESET
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:637
>>>> sofia/external/14159927717 CUSTOM RESET
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:642
>>>> (sofia/external/14159927717) State Change CS_RESET -> CS_SOFT_EXECUTE
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/14159927717 [BREAK]
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:336
>>>> (sofia/external/14159927717) State RESET going to sleep
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/14159927717) Running State Change CS_SOFT_EXECUTE
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_state_machine.c:346
>>>> (sofia/external/14159927717) State SOFT_EXECUTE
>>>> 2009-10-22 17:22:31.748086 [DEBUG] mod_sofia.c:442 SOFIA TRANSMIT
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:660
>>>> sofia/external/14159927717 CUSTOM SOFT_EXECUTE
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_originate.c:674
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_RESET
>>>> -> CS_SOFT_EXECUTE
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:120
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message
>>>> [DISPLAY]
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:120
>>>> sofia/external/14159927717 receive message [DISPLAY]
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:924
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>>> CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:968
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message
>>>> [BRIDGE]
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:630 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:975
>>>> sofia/external/14159927717 receive message [BRIDGE]
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:630 Send signal
>>>> sofia/external/14159927717 [BREAK]
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_ivr_bridge.c:1019
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>>> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
>>>> 2009-10-22 17:22:31.748086 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:22:31.749553 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>>> CS_EXCHANGE_MEDIA
>>>> 2009-10-22 17:22:31.749553 [DEBUG] switch_core_state_machine.c:343
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
>>>> 2009-10-22 17:22:31.749553 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK
>>>> multipler is being overrided to1
>>>> 2009-10-22 17:22:34.300662 [INFO] switch_cpp.cpp:1116 BG_AG_GEN_CALLS,
>>>> Agents total:1 avail:0 OutChan:0 AR:0 +-Calls:0:Q Call NYNoc
>>>> multipler is being overrided to1
>>>> 2009-10-22 17:22:39.303041 [INFO] switch_cpp.cpp:1116 BG_AG_GEN_CALLS,
>>>> Agents total:1 avail:0 OutChan:0 AR:0 +-Calls:0:Q Call NYNoc
>>>> 2009-10-22 17:22:40.626379 [NOTICE] sofia.c:328 Hangup
>>>> sofia/external/14159927717 [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
>>>> 2009-10-22 17:22:40.626379 [DEBUG] switch_channel.c:1810 Send signal
>>>> sofia/external/14159927717 [KILL]
>>>> 2009-10-22 17:22:40.626379 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/14159927717 [BREAK]
>>>> active_true=1
>>>> --contact NOT in playmsg or conference, update AGENT
>>>> 2009-10-22 17:22:40.626379 [DEBUG] switch_core_state_machine.c:437
>>>> thread mismatch skipping state handler.
>>>> 2009-10-22 17:22:40.626379 [INFO] switch_cpp.cpp:1116
>>>> EVENTC_HANGUP_CONTACT, Q_CONTACT fired for agent:1/1 for qcall:1
>>>> contact:4801442/4
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:440
>>>> sofia/external/14159927717 ending bridge by request from write function
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:495
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message
>>>> [UNBRIDGE]
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:630 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
>>>> DONE [sofia/external/+14158867717 at 199.173.100.16:5060]
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:523 Send signal
>>>> sofia/external/14159927717 [BREAK]
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr.c:1361 (sofia/external/+
>>>> 14158867717 at 199.173.100.16:5060) State Change CS_EXCHANGE_MEDIA ->
>>>> CS_ROUTING
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr.c:1365 sofia/external/+
>>>> 14158867717 at 199.173.100.16:5060 receive message [TRANSFER]
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:630 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:22:40.639760 [NOTICE] switch_ivr.c:1367 Transfer
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 to xml[1998 at default]
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:343
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
>>>> going to sleep
>>>>  2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>>> CS_ROUTING
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:333
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State ROUTING
>>>> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:130 sofia/external/+
>>>> 14158867717 at 199.173.100.16:5060 SOFIA ROUTING
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:78
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard ROUTING
>>>> 2009-10-22 17:22:40.639760 [INFO] mod_dialplan_xml.c:391 Processing
>>>> +14158867717->1998 in context default
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->tod_example] continue=true
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (PASS)
>>>> [tod_example] ${strftime(%w)}(4) =~ /^([1-5])$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (PASS)
>>>> [tod_example] ${strftime(%H%M)}(1722) =~ /^((09|1[0-7])[0-5][0-9]|1800)$/
>>>> break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Action
>>>> set(open=true)
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->ivr_demo] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [ivr_demo] destination_number(1998) =~ /^5000$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1975] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1975] destination_number(1998) =~ /^1975$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1971] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1971] destination_number(1998) =~ /^1971$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1972] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1972] destination_number(1998) =~ /^1972$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1920] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1920] destination_number(1998) =~ /^1920$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1981] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1981] destination_number(1998) =~ /^1981$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1982] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1982] destination_number(1998) =~ /^1982$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1481] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1481] destination_number(1998) =~ /^1481$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1989] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1989] destination_number(1998) =~ /^1989$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1990] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1990] destination_number(1998) =~ /^1990$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1991] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1991] destination_number(1998) =~ /^1991$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1992] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1992] destination_number(1998) =~ /^1992$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1993] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1993] destination_number(1998) =~ /^1993$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1994] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1994] destination_number(1998) =~ /^1994$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1995] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1995] destination_number(1998) =~ /^1995$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1996] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1996] destination_number(1998) =~ /^1996$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1997] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (FAIL)
>>>> [1997] destination_number(1998) =~ /^1997$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 parsing
>>>> [default->1998] continue=false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Regex (PASS)
>>>> [1998] destination_number(1998) =~ /^1998$/ break=on-false
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Action
>>>> set(transfer_after_bridge=1998)
>>>> Dialplan: sofia/external/+14158867717 at 199.173.100.16:5060 Action park()
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:114
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>>> CS_ROUTING -> CS_EXECUTE
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:333
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State ROUTING going
>>>> to sleep
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>>> CS_EXECUTE
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:340
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXECUTE
>>>> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:173 sofia/external/+
>>>> 14158867717 at 199.173.100.16:5060 SOFIA EXECUTE
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:151
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard EXECUTE
>>>> EXECUTE sofia/external/+14158867717 at 199.173.100.16:5060 set(open=true)
>>>> 2009-10-22 17:22:40.639760 [DEBUG] mod_dptools.c:752 sofia/external/+
>>>> 14158867717 at 199.173.100.16:5060 SET [open]=[true]
>>>> EXECUTE sofia/external/+14158867717 at 199.173.100.16:5060set(transfer_after_bridge=1998)
>>>> 2009-10-22 17:22:40.639760 [DEBUG] mod_dptools.c:752 sofia/external/+
>>>> 14158867717 at 199.173.100.16:5060 SET [transfer_after_bridge]=[1998]
>>>> EXECUTE sofia/external/+14158867717 at 199.173.100.16:5060 park()
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:446
>>>> sofia/external/14159927717 ending bridge by request from read function
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
>>>> DONE [sofia/external/14159927717]
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_ivr_bridge.c:523 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> subtracting 9 seconds for hh_call: 1095422
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:346
>>>> (sofia/external/14159927717) State SOFT_EXECUTE going to sleep
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/14159927717) Running State Change CS_HANGUP
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:464
>>>> (sofia/external/14159927717) State HANGUP
>>>> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:344 Channel
>>>> sofia/external/14159927717 hanging up, cause: NORMAL_CLEARING
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:46
>>>> sofia/external/14159927717 Standard HANGUP, cause: NORMAL_CLEARING
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:464
>>>> (sofia/external/14159927717) State HANGUP going to sleep
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:325
>>>> (sofia/external/14159927717) State Change CS_HANGUP -> CS_REPORTING
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/14159927717 [BREAK]
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/14159927717) Running State Change CS_REPORTING
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:555
>>>> (sofia/external/14159927717) State REPORTING
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:53
>>>> sofia/external/14159927717 Standard REPORTING, cause: NORMAL_CLEARING
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:555
>>>> (sofia/external/14159927717) State REPORTING going to sleep
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:319
>>>> (sofia/external/14159927717) State Change CS_REPORTING -> CS_DESTROY
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/14159927717 [BREAK]
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_session.c:1069 Session 6
>>>> (sofia/external/14159927717) Locked, Waiting on external entities
>>>> 2009-10-22 17:22:40.639760 [NOTICE] switch_core_session.c:1087 Session 6
>>>> (sofia/external/14159927717) Ended
>>>> 2009-10-22 17:22:40.639760 [NOTICE] switch_core_session.c:1089 Close
>>>> Channel sofia/external/14159927717 [CS_DESTROY]
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:401
>>>> (sofia/external/14159927717) Running State Change CS_DESTROY
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:412
>>>> (sofia/external/14159927717) State DESTROY
>>>> 2009-10-22 17:22:40.639760 [DEBUG] mod_sofia.c:261
>>>> sofia/external/14159927717 SOFIA DESTROY
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:60
>>>> sofia/external/14159927717 Standard DESTROY
>>>> 2009-10-22 17:22:40.639760 [DEBUG] switch_core_state_machine.c:412
>>>> (sofia/external/14159927717) State DESTROY going to sleep
>>>> 2009-10-22 17:22:40.639760 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>>>> teliax.com hc:NORMAL_CLEARING du:9 cn:sofia/external/14159927717
>>>>
>>>>
>>>>
>>>> A BAD UUID_KILL SEQUENCE: (where the agent is hungup, unexpectedly)
>>>>
>>>> --bridged in: 0.005821 seconds
>>>> 2009-10-22 17:23:26.718930 [INFO] switch_cpp.cpp:1116 AG_CBRIDGE,
>>>> connected agent:1/1 contact:4801439/22/1/1
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_play_say.c:1432 done
>>>> playing file
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:340
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXECUTE going
>>>> to sleep
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>>> CS_HIBERNATE
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE
>>>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:160 sofia/external/+
>>>> 14158867717 at 199.173.100.16:5060 SOFIA HIBERNATE
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:650
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>>> CS_HIBERNATE -> CS_RESET
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HIBERNATE going
>>>> to sleep
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>>> CS_RESET
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET
>>>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:145 sofia/external/+
>>>> 14158867717 at 199.173.100.16:5060 SOFIA RESET
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:637
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 CUSTOM RESET
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:67
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard RESET
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State RESET going to
>>>> sleep
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:340
>>>> (sofia/external/14159927717) State EXECUTE going to sleep
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/14159927717) Running State Change CS_HIBERNATE
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
>>>> (sofia/external/14159927717) State HIBERNATE
>>>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:160
>>>> sofia/external/14159927717 SOFIA HIBERNATE
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:650
>>>> (sofia/external/14159927717) State Change CS_HIBERNATE -> CS_RESET
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/14159927717 [BREAK]
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:355
>>>> (sofia/external/14159927717) State HIBERNATE going to sleep
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/14159927717) Running State Change CS_RESET
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
>>>> (sofia/external/14159927717) State RESET
>>>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:145
>>>> sofia/external/14159927717 SOFIA RESET
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:637
>>>> sofia/external/14159927717 CUSTOM RESET
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:642
>>>> (sofia/external/14159927717) State Change CS_RESET -> CS_SOFT_EXECUTE
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/14159927717 [BREAK]
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:336
>>>> (sofia/external/14159927717) State RESET going to sleep
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/14159927717) Running State Change CS_SOFT_EXECUTE
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:346
>>>> (sofia/external/14159927717) State SOFT_EXECUTE
>>>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:442 SOFIA TRANSMIT
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:660
>>>> sofia/external/14159927717 CUSTOM SOFT_EXECUTE
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_originate.c:674
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change CS_RESET
>>>> -> CS_SOFT_EXECUTE
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:120
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message
>>>> [DISPLAY]
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:120
>>>> sofia/external/14159927717 receive message [DISPLAY]
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:924
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>>> CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:968
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message
>>>> [BRIDGE]
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:630 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:975
>>>> sofia/external/14159927717 receive message [BRIDGE]
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:630 Send signal
>>>> sofia/external/14159927717 [BREAK]
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_ivr_bridge.c:1019
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>>> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>>> CS_EXCHANGE_MEDIA
>>>> 2009-10-22 17:23:26.721553 [DEBUG] switch_core_state_machine.c:343
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
>>>> 2009-10-22 17:23:26.721553 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK
>>>> 2009-10-22 17:23:26.740065 [DEBUG] switch_core_io.c:234 sofia/external/+
>>>> 14158867717 at 199.173.100.16:5060 receive message [TRANSCODING_NECESSARY]
>>>> 2009-10-22 17:23:27.319618 [DEBUG] switch_core_media_bug.c:393 Removing
>>>> BUG from sofia/external/+14158867717 at 199.173.100.16:5060
>>>> multipler is being overrided to1
>>>> 2009-10-22 17:23:29.406414 [INFO] switch_cpp.cpp:1116 BG_AG_GEN_CALLS,
>>>> Agents total:1 avail:0 OutChan:0 AR:0 +-Calls:0:Q Call NYNoc
>>>> 2009-10-22 17:23:31.873352 [NOTICE] mod_commands.c:1523 Hangup
>>>> sofia/external/14159927717 [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
>>>> 2009-10-22 17:23:31.874411 [INFO] switch_cpp.cpp:1116
>>>> EVENTC_HANGUP_CONTACT, Q_CONTACT fired for agent:1/1 for qcall:1
>>>> contact:4801439/1
>>>> active_true=1
>>>> --contact NOT in playmsg or conference, update AGENT
>>>> 2009-10-22 17:23:31.875475 [DEBUG] switch_channel.c:1810 Send signal
>>>> sofia/external/14159927717 [KILL]
>>>> 2009-10-22 17:23:31.875475 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/14159927717 [BREAK]
>>>> 2009-10-22 17:23:31.875475 [DEBUG] switch_core_state_machine.c:437
>>>> thread mismatch skipping state handler.
>>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:446
>>>> sofia/external/14159927717 ending bridge by request from read function
>>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
>>>> DONE [sofia/external/14159927717]
>>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:523 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:495
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 receive message
>>>> [UNBRIDGE]
>>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_session.c:630 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:521 BRIDGE THREAD
>>>> DONE [sofia/external/+14158867717 at 199.173.100.16:5060]
>>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_ivr_bridge.c:523 Send signal
>>>> sofia/external/14159927717 [BREAK]
>>>> 2009-10-22 17:23:31.882153 [NOTICE] switch_ivr.c:1268 Hangup
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [CS_EXCHANGE_MEDIA]
>>>> [EXCHANGE_ROUTING_ERROR]
>>>>  2009-10-22 17:23:31.882153 [DEBUG] switch_channel.c:1810 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [KILL]
>>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_state_machine.c:464
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HANGUP
>>>> 2009-10-22 17:23:31.882153 [DEBUG] mod_sofia.c:344 Channel
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 hanging up, cause:
>>>> EXCHANGE_ROUTING_ERROR
>>>> 2009-10-22 17:23:31.882153 [DEBUG] mod_sofia.c:382 Sending BYE to
>>>> sofia/external/+14158867717 at 199.173.100.16:5060
>>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_state_machine.c:46
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard HANGUP, cause:
>>>> EXCHANGE_ROUTING_ERROR
>>>> 2009-10-22 17:23:31.882153 [DEBUG] switch_core_state_machine.c:464
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State HANGUP going to
>>>> sleep
>>>> 2009-10-22 17:23:31.884357 [INFO] switch_cpp.cpp:1116 HH_HANGUP,
>>>> agent:1/1 uuid: f8604694-4e44-4c4c-9805-722b3e9d6b68
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:346
>>>> (sofia/external/14159927717) State SOFT_EXECUTE going to sleep
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/14159927717) Running State Change CS_HANGUP
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:464
>>>> (sofia/external/14159927717) State HANGUP
>>>> 2009-10-22 17:23:31.885412 [DEBUG] mod_sofia.c:344 Channel
>>>> sofia/external/14159927717 hanging up, cause: NORMAL_CLEARING
>>>> 2009-10-22 17:23:31.885412 [DEBUG] mod_sofia.c:382 Sending BYE to
>>>> sofia/external/14159927717
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:46
>>>> sofia/external/14159927717 Standard HANGUP, cause: NORMAL_CLEARING
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:464
>>>> (sofia/external/14159927717) State HANGUP going to sleep
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:325
>>>> (sofia/external/14159927717) State Change CS_HANGUP -> CS_REPORTING
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/14159927717 [BREAK]
>>>> subtracting 5 seconds for hh_call: 1095423
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/14159927717) Running State Change CS_REPORTING
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:555
>>>> (sofia/external/14159927717) State REPORTING
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:53
>>>> sofia/external/14159927717 Standard REPORTING, cause: NORMAL_CLEARING
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:555
>>>> (sofia/external/14159927717) State REPORTING going to sleep
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:319
>>>> (sofia/external/14159927717) State Change CS_REPORTING -> CS_DESTROY
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/14159927717 [BREAK]
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_session.c:1069 Session 7
>>>> (sofia/external/14159927717) Locked, Waiting on external entities
>>>> 2009-10-22 17:23:31.885412 [NOTICE] switch_core_session.c:1087 Session 7
>>>> (sofia/external/14159927717) Ended
>>>> 2009-10-22 17:23:31.885412 [NOTICE] switch_core_session.c:1089 Close
>>>> Channel sofia/external/14159927717 [CS_DESTROY]
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:401
>>>> (sofia/external/14159927717) Running State Change CS_DESTROY
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:412
>>>> (sofia/external/14159927717) State DESTROY
>>>> 2009-10-22 17:23:31.885412 [DEBUG] mod_sofia.c:261
>>>> sofia/external/14159927717 SOFIA DESTROY
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:60
>>>> sofia/external/14159927717 Standard DESTROY
>>>> 2009-10-22 17:23:31.885412 [DEBUG] switch_core_state_machine.c:412
>>>> (sofia/external/14159927717) State DESTROY going to sleep
>>>> 2009-10-22 17:23:31.885412 [INFO] switch_cpp.cpp:1116 PCHangup gw:
>>>> teliax.com hc:NORMAL_CLEARING du:5 cn:sofia/external/14159927717
>>>> 2009-10-22 17:23:31.885412 [INFO] switch_cpp.cpp:1116
>>>> EVENTC_BILLING_CEIL, subtracting 5 for hh_call:1095423
>>>> 2009-10-22 17:23:31.901620 [DEBUG] switch_core_state_machine.c:494
>>>> Hangup Command luarun(hh_hangup.lua f8604694-4e44-4c4c-9805-722b3e9d6b68 1
>>>> 1):
>>>> +OK
>>>>
>>>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:343
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State EXCHANGE_MEDIA
>>>> going to sleep
>>>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>>> CS_HANGUP
>>>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:442
>>>> handler already called, skipping state handler.
>>>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:325
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>>> CS_HANGUP -> CS_REPORTING
>>>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:23:31.902676 [DEBUG] switch_core_state_machine.c:306
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>>> CS_REPORTING
>>>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:555
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State REPORTING
>>>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:53
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard REPORTING,
>>>> cause: EXCHANGE_ROUTING_ERROR
>>>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:555
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State REPORTING going
>>>> to sleep
>>>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:319
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State Change
>>>> CS_REPORTING -> CS_DESTROY
>>>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_session.c:932 Send signal
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 [BREAK]
>>>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_session.c:1069 Session 4
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Locked, Waiting on
>>>> external entities
>>>> 2009-10-22 17:23:31.903742 [NOTICE] switch_core_session.c:1087 Session 4
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Ended
>>>> 2009-10-22 17:23:31.903742 [NOTICE] switch_core_session.c:1089 Close
>>>> Channel sofia/external/+14158867717 at 199.173.100.16:5060 [CS_DESTROY]
>>>> 2009-10-22 17:23:31.903742 [DEBUG] switch_core_state_machine.c:401
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) Running State Change
>>>> CS_DESTROY
>>>> 2009-10-22 17:23:31.904799 [DEBUG] switch_core_state_machine.c:412
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State DESTROY
>>>> 2009-10-22 17:23:31.904799 [DEBUG] mod_sofia.c:261 sofia/external/+
>>>> 14158867717 at 199.173.100.16:5060 SOFIA DESTROY
>>>> 2009-10-22 17:23:31.904799 [DEBUG] switch_core_state_machine.c:60
>>>> sofia/external/+14158867717 at 199.173.100.16:5060 Standard DESTROY
>>>> 2009-10-22 17:23:31.904799 [DEBUG] switch_core_state_machine.c:412
>>>> (sofia/external/+14158867717 at 199.173.100.16:5060) State DESTROY going
>>>> to sleep
>>>>
>>>>
>>>>
>>>> On Tue, Oct 20, 2009 at 1:44 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>>
>>>>>
>>>>> On Sun, Oct 18, 2009 at 9:48 PM, Matthew Fong <mattdfong at gmail.com>wrote:
>>>>>
>>>>>> The debug level logs to the console, right? The pastebin, had log
>>>>>> level debug, sofia trace on for external and default, and sofia loglevel all
>>>>>> 9. Is there another log enable command I'm missing? It seems loglevel all 9
>>>>>> outputs enter and exit functions, but at least to my novice eye, it's not
>>>>>> too obvious why freeswitch is sending a BYE to my DID provider.
>>>>>>
>>>>>> Evidently the console loglevel debug did not take because there isn't
>>>>> a single DEBUG level line of output. Please try again. Just remember that
>>>>> debug level log messages will have A LOT of information so be patient while
>>>>> sifting through it. Of course, feel free to pastebin the output and add a
>>>>> link to it in this email thread.
>>>>>
>>>>> -MC
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:213-799-1400
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From tculjaga at gmail.com  Fri Oct 23 07:57:30 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 23 Oct 2009 16:57:30 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <65d96fc80910230614m2b2504d0r92006b78d267a399@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
	<Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
	<Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>
	<65d96fc80910230452s6b4a179aqa5bc41dc34efc403@mail.gmail.com>
	<Pine.LNX.4.64.0910231610220.21492@srv.icf.org.ru>
	<65d96fc80910230538w7105be6fve14956ab3b247bf6@mail.gmail.com>
	<Pine.LNX.4.64.0910231646210.29582@srv.icf.org.ru>
	<65d96fc80910230614m2b2504d0r92006b78d267a399@mail.gmail.com>
Message-ID: <65d96fc80910230757h4595f396o2dc1a1bb781d3a8a@mail.gmail.com>

a solution to H323 endpoint => FS => SIP user no audio issue

is to disable a wait for tx Audio ... for  case
SWITCH_MESSAGE_INDICATE_ANSWER:{

//m_txAudioOpened.Wait();


                case SWITCH_MESSAGE_INDICATE_ANSWER:{

                        switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_CONSOLE, "ANSWER: we got Answer event\n");

                        if (switch_channel_test_flag(channel, CF_OUTBOUND))
{

                                switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_CONSOLE, "ANSWER: we got Answer event - CF_OUTBOUND
\n");
                                return SWITCH_STATUS_FALSE;
                        }
                        AnsweringCall(H323Connection::AnswerCallNow);

                        switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_CONSOLE, "ANSWER: suppose the call is Answered Now\n");
                        PTRACE(4, "mod_h323\tMedia started on connection "
<< *this);

                        // test
                        //switch_channel_mark_answered(m_fsChannel);

                        m_rxAudioOpened.Wait();
                        switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_CONSOLE, "ANSWER: wait for m_rxAudioOpened\n");
                        //m_txAudioOpened.Wait();
                        switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_CONSOLE, "ANSWER: we disable wait for m_txAudioOpened\n");

                        switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_CONSOLE, "ANSWER: were waiting for rx/tx AudioOpen\n");

                        if (!switch_channel_test_flag(m_fsChannel,
CF_EARLY_MEDIA)) {

                                switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_CONSOLE, "ANSWER: we have early media\n");

                                PTRACE(4,
"mod_h323\t-------------------->switch_channel_mark_answered(m_fsChannel) "
<< *this);
                                switch_channel_mark_answered(m_fsChannel);
                                switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_CONSOLE, "ANSWER: answered in early Media\n");
                        }
                        break;
                }


Now, I'm able to both originate and terminate cals with 2-way audio...
the signaling looks correct...



outgoing:

1369.425046    10.4.62.7 -> 10.4.62.89   SIP/SDP Request: INVITE
sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>;transport=udp, with session
description
1369.426255    10.4.62.7 -> 10.4.62.31   H.225.0 CS: alerting
1369.435950   10.4.62.89 -> 10.4.62.7    SIP Status: 100 Trying
1369.449065   10.4.62.89 -> 10.4.62.7    SIP Status: 180 Ringing
1369.605109    10.4.62.7 -> 10.4.62.31   H.225.0 CS: progress
OpenLogicalChannel
1369.609788   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
terminalCapabilitySet
1369.610489   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
masterSlaveDetermination
1369.619071    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
terminalCapabilitySet
1369.620349    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
terminalCapabilitySetAck
1369.623215   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
terminalCapabilitySetAck
1369.625591    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
masterSlaveDeterminationAck
1369.628174   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
masterSlaveDeterminationAck
1370.966958   10.4.62.89 -> 10.4.62.7    SIP/SDP Status: 200 OK, with
session description
1370.967431    10.4.62.7 -> 10.4.62.89   SIP Request: ACK
sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>;transport=udp
1370.975172    10.4.62.7 -> 10.4.62.31   H.225.0 CS: connect
1372.354383   10.4.62.89 -> 10.4.62.7    SIP Request: BYE
sip:mod_sofia at 10.4.62.7:5060
1372.355147    10.4.62.7 -> 10.4.62.89   SIP Status: 200 OK
1372.392904    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: releaseComplete
endSessionCommand
1372.397302   10.4.62.31 -> 10.4.62.7    H.225.0 CS: releaseComplete


incoming:


1502.817154   10.4.62.31 -> 10.4.62.7    H.225.0 CS: setup
OpenLogicalChannel
1502.833732    10.4.62.7 -> 10.4.62.31   H.225.0 CS: callProceeding
1502.850909    10.4.62.7 -> 10.4.62.89   SIP/SDP Request: INVITE
sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>;transport=udp, with session
description
1502.851758    10.4.62.7 -> 10.4.62.31   H.225.0 CS: alerting
1502.861828   10.4.62.89 -> 10.4.62.7    SIP Status: 100 Trying
1502.875127   10.4.62.89 -> 10.4.62.7    SIP Status: 180 Ringing
1503.033258    10.4.62.7 -> 10.4.62.31   H.225.0 CS: progress
OpenLogicalChannel
1503.037908   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
terminalCapabilitySet
1503.038608   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
masterSlaveDetermination
1503.050154    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
terminalCapabilitySet
1503.051381    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
terminalCapabilitySetAck
1503.054297   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
terminalCapabilitySetAck
1503.054917    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
masterSlaveDeterminationAck
1503.057933   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
masterSlaveDeterminationAck
1505.485493   10.4.62.89 -> 10.4.62.7    SIP/SDP Status: 200 OK, with
session description
1505.486018    10.4.62.7 -> 10.4.62.89   SIP Request: ACK
sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>;transport=udp
1505.493611    10.4.62.7 -> 10.4.62.31   H.225.0 CS: connect
1509.565959   10.4.62.89 -> 10.4.62.7    SIP Request: BYE
sip:mod_sofia at 10.4.62.7:5060
1509.566722    10.4.62.7 -> 10.4.62.89   SIP Status: 200 OK
1509.577435    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: releaseComplete
endSessionCommand
1509.582066   10.4.62.31 -> 10.4.62.7    H.225.0 CS: releaseComplete



... i still need to check the CDRs as well but here we are :)
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From sicfslist at gmail.com  Fri Oct 23 07:58:10 2009
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Fri, 23 Oct 2009 09:58:10 -0500
Subject: [Freeswitch-users] dynamic gateway
In-Reply-To: <61ee6b990910222340r49ae8fa4oafa1bd4acfecd71c@mail.gmail.com>
References: <61ee6b990910222340r49ae8fa4oafa1bd4acfecd71c@mail.gmail.com>
Message-ID: <35b355e90910230758u53a76a2fo875ac2d6182e63e3@mail.gmail.com>

Hello:

Your issue is that this portion of the above string "sipinterface_1" has to
be a valid sip profile.  To do this:
  -- determine the appropriate sip profile you want to use (usually
external)
  -- send the traffic to whatever ip you want via this portion of your
string $1 at 192.168.111.101:5060

You can't dynamically change the /sipinterface_1/ parameter.

SDR
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From anthony.minessale at gmail.com  Fri Oct 23 08:02:07 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 23 Oct 2009 10:02:07 -0500
Subject: [Freeswitch-users] sched_api doesn't get launched
In-Reply-To: <59ad9ca10910230154i6d41ddfci8957b152c1884800@mail.gmail.com>
References: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
	<3F1747F9-C543-4C6F-BB01-4DD5FA5B7BC5@jerris.com>
	<59ad9ca10910161153h5d531533vd844e49b9ca6d896@mail.gmail.com>
	<87f2f3b90910161237h356b7aa2q6ec12c6ecf78dc34@mail.gmail.com>
	<59ad9ca10910210244q5410f2bdq3b93c25f95673d11@mail.gmail.com>
	<191c3a030910211309sb727b37g6324e1833f2a1e8c@mail.gmail.com>
	<59ad9ca10910230154i6d41ddfci8957b152c1884800@mail.gmail.com>
Message-ID: <191c3a030910230802m1cb847agfdaf6e436296e748@mail.gmail.com>

it's probably related to escaping the data.
I was sick of watching you suffer so i added api_on_answer variable to
trunk.


On Fri, Oct 23, 2009 at 3:54 AM, Henry Huang <red.rain.seven at gmail.com>wrote:

> Thanks to c6burns on IRC channel for the tip to use execute_on_answer in
> combination with eval, and of course everyone here that pointing me to the
> right direction.
> I was able to execute sched_api with eval, but not with the combination of
> execute_on_answer. The argument just don't get parsed as the eval argument.
> Here is the code:
>
> setVariable("execute_on_answer", "eval sched_api(+%ld none uuid_displace
> <some uuid> start
> /usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux)");
>
>
> Here is the log:
> 2009-10-23 16:17:53.633189 [DEBUG] switch_scheduler.c:214 Added task 2
> sched_api_function (none) to run at 1256285888
> EXECUTE sofia/internal/1688004 at 192.168.1.66 *eval(+OK *Added: 2
> )
>
> 2009-10-23 16:18:04.52701 [DEBUG] switch_channel.c:1927 sofia/internal/
> 1688004 at 192.168.1.66 *execute on answer: eval(sched_api(+37 none
> uuid_displace 84d74f41-8668-4138-943f-f076e94046ad start
> /usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux))*
> EXECUTE sofia/internal/1688004 at 192.168.1.66 eval(sched_api(+37 none
> uuid_displace 84d74f41-8668-4138-943f-f076e94046ad start
> /usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux))
>
>
> The first part of the log is when I just do execute("eval",
> "${sched_api(+%ld none uuid_displace <some uuid> start
> /usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux)}");
> And I simple get a OK, and then later on the sched_api gets executed.
>
> However, on the excerpt of the second log. which I use the combination with
> execute_on_answer like specified in the yellow hight code above. It doesn't
> simply return a OK, instead it shows the whole blob of code. and does
> nothing later.
>
>
>
> On Thu, Oct 22, 2009 at 4:09 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Yes you need an API object as described in other email.
>>
>> Which line of code from java caused that segfault
>> It looks like a simple NULL string issue that we may want to hunt down.
>>
>>
>> On Wed, Oct 21, 2009 at 4:44 AM, Henry Huang <red.rain.seven at gmail.com>wrote:
>>
>>> I can't seem to find the right thing to use in mod_java to execute api
>>> commands, only api_after_bridge
>>>
>>> 2009-10-21 17:42:46.593094 [NOTICE] mod_sofia.c:1509 Pre-Answer
>>> sofia/internal/1688003 at 192.168.1.66!
>>> #
>>> # A fatal error has been detected by the Java Runtime Environment:
>>> #
>>> #  SIGSEGV (0xb) at pc=0x004e4480, pid=1927, tid=16116624
>>> #
>>> # JRE version: 6.0_16-b01
>>> # Java VM: Java HotSpot(TM) Client VM (14.2-b01 mixed mode linux-x86 )
>>> # Problematic frame:
>>> # C  [libc.so.6+0x6f480]  strcpy+0x10
>>> #
>>> # An error report file with more information is saved as:
>>> # /usr/local/freeswitch/bin/hs_err_pid1927.log
>>> 2009-10-21 17:42:59.883729 [ERR] switch_core_session.c:1374 Invalid
>>> Application sched_api
>>> 2009-10-21 17:42:59.883729 [NOTICE] switch_core_session.c:1375 Hangup
>>> sofia/internal/1688003 at 192.168.1.66 [CS_EXECUTE]
>>> [DESTINATION_OUT_OF_ORDER]
>>> #
>>> # If you would like to submit a bug report, please visit:
>>> #   http://java.sun.com/webapps/bugreport/crash.jsp
>>> # The crash happened outside the Java Virtual Machine in native code.
>>> # See problematic frame for where to report the bug.
>>>
>>>
>>> On Sat, Oct 17, 2009 at 3:37 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>>
>>>>
>>>> On Fri, Oct 16, 2009 at 11:53 AM, Henry Huang <red.rain.seven at gmail.com
>>>> > wrote:
>>>>
>>>>> So how would you trigger it from a script dialplan? The only time it
>>>>> seemed to work is when I did setVariable("api_after_bridge", "sched_api blah
>>>>> blah blah");
>>>>> but then it gets executed after the channel's been teared down. I
>>>>> thought api_after_bridge means right after the call gets connected.
>>>>>
>>>>> I need something to execute an api command right before or right after
>>>>> the call gets bridged.
>>>>>
>>>>> api_after_bridge is a channel variable, so using setVariable works just
>>>> fine. If you need to sched_api is an API only. Check these out:
>>>> http://wiki.freeswitch.org/wiki/Mod_commands#Misc._Commands
>>>>
>>>> So you need an API object in order to use it. I don't know the syntax
>>>> for creating an api obj in Java but in Lua it goes like this:
>>>> api = freeswitch.API();
>>>> res = api:execute("sched_api","+300 none my_api my_api_args")
>>>>
>>>> Remember, if the method you are using isn't found in the dial plan tools
>>>> then it isn't a dial plan application. Make sure it's on the list:
>>>> http://wiki.freeswitch.org/wiki/Mod_dptools
>>>>
>>>> On the other hand, API commands are listed here:
>>>> http://wiki.freeswitch.org/wiki/Mod_commands
>>>>
>>>> dptools require a session object, api commands require an api object...
>>>>
>>>> -MC
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Henry Huang
>>> UniC Solution - Communication Unified
>>> VoIP & Open Source software Consultant
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Henry Huang
> UniC Solution - Communication Unified
> VoIP & Open Source software Consultant
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From saeedahmad1981 at gmail.com  Fri Oct 23 08:08:38 2009
From: saeedahmad1981 at gmail.com (Saeed Ahmad)
Date: Fri, 23 Oct 2009 17:08:38 +0200
Subject: [Freeswitch-users] dynamic gateway
In-Reply-To: <35b355e90910230758u53a76a2fo875ac2d6182e63e3@mail.gmail.com>
References: <61ee6b990910222340r49ae8fa4oafa1bd4acfecd71c@mail.gmail.com> 
	<35b355e90910230758u53a76a2fo875ac2d6182e63e3@mail.gmail.com>
Message-ID: <c96b95650910230808l7b532529r106431d3f858b1ae@mail.gmail.com>

what you get if you run

sofia status profile sipinterface_1
- SAT

On Fri, Oct 23, 2009 at 4:58 PM, Shelby Ramsey <sicfslist at gmail.com> wrote:

> sipinterface_1
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From bottleman at icf.org.ru  Fri Oct 23 08:17:47 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Fri, 23 Oct 2009 19:17:47 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910230757h4595f396o2dc1a1bb781d3a8a@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
	<Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
	<Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>
	<65d96fc80910230452s6b4a179aqa5bc41dc34efc403@mail.gmail.com>
	<Pine.LNX.4.64.0910231610220.21492@srv.icf.org.ru>
	<65d96fc80910230538w7105be6fve14956ab3b247bf6@mail.gmail.com>
	<Pine.LNX.4.64.0910231646210.29582@srv.icf.org.ru>
	<65d96fc80910230614m2b2504d0r92006b78d267a399@mail.gmail.com>
	<65d96fc80910230757h4595f396o2dc1a1bb781d3a8a@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910231903510.17804@srv.icf.org.ru>

On 2009-10-23 16:57 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

i have question to developers about one proce in fs

src/switch_ivr_originate.c

static switch_status_t originate_on_consume_media_transmit(switch_core_session_t *session)
{
    switch_channel_t *channel = switch_core_session_get_channel(session);

		if (!switch_channel_test_flag(channel, CF_PROXY_MODE)) {
		while (switch_channel_get_state(channel) == CS_CONSUME_MEDIA && !switch_channel_test_flag(chann
			if (!switch_channel_media_ready(channel)) {
			    switch_yield(10000);
			} else {
				switch_ivr_sleep(session, 10, SWITCH_FALSE, NULL);
			}
		}
	    }

	switch_channel_clear_state_handler(channel, &originate_state_handlers);

	return SWITCH_STATUS_FALSE;
}

what exacly it do? 

call scheme like this sip->fs->h323->gk->h323(on same fs)->fs(same too) and there i have no audio issues.
if bridge connect while it sleep i have audio, if it not sleep while bridge connect i have no audio.

TC>a solution to H323 endpoint => FS => SIP user no audio issue
TC>
TC>is to disable a wait for tx Audio ... for  case
TC>SWITCH_MESSAGE_INDICATE_ANSWER:{
TC>
TC>//m_txAudioOpened.Wait();
TC>
TC>
TC>                case SWITCH_MESSAGE_INDICATE_ANSWER:{
TC>
TC>                        switch_log_printf(SWITCH_CHANNEL_LOG,
TC>SWITCH_LOG_CONSOLE, "ANSWER: we got Answer event\n");
TC>
TC>                        if (switch_channel_test_flag(channel, CF_OUTBOUND))
TC>{
TC>
TC>                                switch_log_printf(SWITCH_CHANNEL_LOG,
TC>SWITCH_LOG_CONSOLE, "ANSWER: we got Answer event - CF_OUTBOUND
TC>\n");
TC>                                return SWITCH_STATUS_FALSE;
TC>                        }
TC>                        AnsweringCall(H323Connection::AnswerCallNow);
TC>
TC>                        switch_log_printf(SWITCH_CHANNEL_LOG,
TC>SWITCH_LOG_CONSOLE, "ANSWER: suppose the call is Answered Now\n");
TC>                        PTRACE(4, "mod_h323\tMedia started on connection "
TC><< *this);
TC>
TC>                        // test
TC>                        //switch_channel_mark_answered(m_fsChannel);
TC>
TC>                        m_rxAudioOpened.Wait();
TC>                        switch_log_printf(SWITCH_CHANNEL_LOG,
TC>SWITCH_LOG_CONSOLE, "ANSWER: wait for m_rxAudioOpened\n");
TC>                        //m_txAudioOpened.Wait();
TC>                        switch_log_printf(SWITCH_CHANNEL_LOG,
TC>SWITCH_LOG_CONSOLE, "ANSWER: we disable wait for m_txAudioOpened\n");
TC>
TC>                        switch_log_printf(SWITCH_CHANNEL_LOG,
TC>SWITCH_LOG_CONSOLE, "ANSWER: were waiting for rx/tx AudioOpen\n");
TC>
TC>                        if (!switch_channel_test_flag(m_fsChannel,
TC>CF_EARLY_MEDIA)) {
TC>
TC>                                switch_log_printf(SWITCH_CHANNEL_LOG,
TC>SWITCH_LOG_CONSOLE, "ANSWER: we have early media\n");
TC>
TC>                                PTRACE(4,
TC>"mod_h323\t-------------------->switch_channel_mark_answered(m_fsChannel) "
TC><< *this);
TC>                                switch_channel_mark_answered(m_fsChannel);
TC>                                switch_log_printf(SWITCH_CHANNEL_LOG,
TC>SWITCH_LOG_CONSOLE, "ANSWER: answered in early Media\n");
TC>                        }
TC>                        break;
TC>                }
TC>
TC>
TC>Now, I'm able to both originate and terminate cals with 2-way audio...
TC>the signaling looks correct...
TC>
TC>
TC>
TC>outgoing:
TC>
TC>1369.425046    10.4.62.7 -> 10.4.62.89   SIP/SDP Request: INVITE
TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>;transport=udp, with session
TC>description
TC>1369.426255    10.4.62.7 -> 10.4.62.31   H.225.0 CS: alerting
TC>1369.435950   10.4.62.89 -> 10.4.62.7    SIP Status: 100 Trying
TC>1369.449065   10.4.62.89 -> 10.4.62.7    SIP Status: 180 Ringing
TC>1369.605109    10.4.62.7 -> 10.4.62.31   H.225.0 CS: progress
TC>OpenLogicalChannel
TC>1369.609788   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>terminalCapabilitySet
TC>1369.610489   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>masterSlaveDetermination
TC>1369.619071    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
TC>terminalCapabilitySet
TC>1369.620349    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
TC>terminalCapabilitySetAck
TC>1369.623215   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>terminalCapabilitySetAck
TC>1369.625591    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
TC>masterSlaveDeterminationAck
TC>1369.628174   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>masterSlaveDeterminationAck
TC>1370.966958   10.4.62.89 -> 10.4.62.7    SIP/SDP Status: 200 OK, with
TC>session description
TC>1370.967431    10.4.62.7 -> 10.4.62.89   SIP Request: ACK
TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>;transport=udp
TC>1370.975172    10.4.62.7 -> 10.4.62.31   H.225.0 CS: connect
TC>1372.354383   10.4.62.89 -> 10.4.62.7    SIP Request: BYE
TC>sip:mod_sofia at 10.4.62.7:5060
TC>1372.355147    10.4.62.7 -> 10.4.62.89   SIP Status: 200 OK
TC>1372.392904    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: releaseComplete
TC>endSessionCommand
TC>1372.397302   10.4.62.31 -> 10.4.62.7    H.225.0 CS: releaseComplete
TC>
TC>
TC>incoming:
TC>
TC>
TC>1502.817154   10.4.62.31 -> 10.4.62.7    H.225.0 CS: setup
TC>OpenLogicalChannel
TC>1502.833732    10.4.62.7 -> 10.4.62.31   H.225.0 CS: callProceeding
TC>1502.850909    10.4.62.7 -> 10.4.62.89   SIP/SDP Request: INVITE
TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>;transport=udp, with session
TC>description
TC>1502.851758    10.4.62.7 -> 10.4.62.31   H.225.0 CS: alerting
TC>1502.861828   10.4.62.89 -> 10.4.62.7    SIP Status: 100 Trying
TC>1502.875127   10.4.62.89 -> 10.4.62.7    SIP Status: 180 Ringing
TC>1503.033258    10.4.62.7 -> 10.4.62.31   H.225.0 CS: progress
TC>OpenLogicalChannel
TC>1503.037908   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>terminalCapabilitySet
TC>1503.038608   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>masterSlaveDetermination
TC>1503.050154    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
TC>terminalCapabilitySet
TC>1503.051381    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
TC>terminalCapabilitySetAck
TC>1503.054297   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>terminalCapabilitySetAck
TC>1503.054917    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
TC>masterSlaveDeterminationAck
TC>1503.057933   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
TC>masterSlaveDeterminationAck
TC>1505.485493   10.4.62.89 -> 10.4.62.7    SIP/SDP Status: 200 OK, with
TC>session description
TC>1505.486018    10.4.62.7 -> 10.4.62.89   SIP Request: ACK
TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89>;transport=udp
TC>1505.493611    10.4.62.7 -> 10.4.62.31   H.225.0 CS: connect
TC>1509.565959   10.4.62.89 -> 10.4.62.7    SIP Request: BYE
TC>sip:mod_sofia at 10.4.62.7:5060
TC>1509.566722    10.4.62.7 -> 10.4.62.89   SIP Status: 200 OK
TC>1509.577435    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: releaseComplete
TC>endSessionCommand
TC>1509.582066   10.4.62.31 -> 10.4.62.7    H.225.0 CS: releaseComplete
TC>
TC>
TC>
TC>... i still need to check the CDRs as well but here we are :)
TC>

can you send a diff? in you call scheme call from h323 endpoint to fs is not have RAS?, 
because i don't have no audio issues in transit from h323 to sip, but my calls a going
thorough GK and fs is regitered on them, my call scheme is h323ep-RAS->GK-RAS->fs.

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From anthony.minessale at gmail.com  Fri Oct 23 08:37:03 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 23 Oct 2009 10:37:03 -0500
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910231903510.17804@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
	<Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>
	<65d96fc80910230452s6b4a179aqa5bc41dc34efc403@mail.gmail.com>
	<Pine.LNX.4.64.0910231610220.21492@srv.icf.org.ru>
	<65d96fc80910230538w7105be6fve14956ab3b247bf6@mail.gmail.com>
	<Pine.LNX.4.64.0910231646210.29582@srv.icf.org.ru>
	<65d96fc80910230614m2b2504d0r92006b78d267a399@mail.gmail.com>
	<65d96fc80910230757h4595f396o2dc1a1bb781d3a8a@mail.gmail.com>
	<Pine.LNX.4.64.0910231903510.17804@srv.icf.org.ru>
Message-ID: <191c3a030910230837x78ea6093i387617a1adfdf958@mail.gmail.com>

if you were on trunk that line of code would be gone.
you really can't do development on 1.0.4 its 6 months old and it will cause
you more trouble than you think when you eventually upgrade if you do not do
it soon.


2009/10/23 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-23 16:57 +0200, Tihomir Culjaga wrote
> freeswitch-users at lists.fre...:
>
> i have question to developers about one proce in fs
>
> src/switch_ivr_originate.c
>
> static switch_status_t
> originate_on_consume_media_transmit(switch_core_session_t *session)
> {
>    switch_channel_t *channel = switch_core_session_get_channel(session);
>
>                if (!switch_channel_test_flag(channel, CF_PROXY_MODE)) {
>                while (switch_channel_get_state(channel) == CS_CONSUME_MEDIA
> && !switch_channel_test_flag(chann
>                        if (!switch_channel_media_ready(channel)) {
>                            switch_yield(10000);
>                        } else {
>                                switch_ivr_sleep(session, 10, SWITCH_FALSE,
> NULL);
>                        }
>                }
>            }
>
>        switch_channel_clear_state_handler(channel,
> &originate_state_handlers);
>
>        return SWITCH_STATUS_FALSE;
> }
>
> what exacly it do?
>
> call scheme like this sip->fs->h323->gk->h323(on same fs)->fs(same too) and
> there i have no audio issues.
> if bridge connect while it sleep i have audio, if it not sleep while bridge
> connect i have no audio.
>
> TC>a solution to H323 endpoint => FS => SIP user no audio issue
> TC>
> TC>is to disable a wait for tx Audio ... for  case
> TC>SWITCH_MESSAGE_INDICATE_ANSWER:{
> TC>
> TC>//m_txAudioOpened.Wait();
> TC>
> TC>
> TC>                case SWITCH_MESSAGE_INDICATE_ANSWER:{
> TC>
> TC>                        switch_log_printf(SWITCH_CHANNEL_LOG,
> TC>SWITCH_LOG_CONSOLE, "ANSWER: we got Answer event\n");
> TC>
> TC>                        if (switch_channel_test_flag(channel,
> CF_OUTBOUND))
> TC>{
> TC>
> TC>                                switch_log_printf(SWITCH_CHANNEL_LOG,
> TC>SWITCH_LOG_CONSOLE, "ANSWER: we got Answer event - CF_OUTBOUND
> TC>\n");
> TC>                                return SWITCH_STATUS_FALSE;
> TC>                        }
> TC>                        AnsweringCall(H323Connection::AnswerCallNow);
> TC>
> TC>                        switch_log_printf(SWITCH_CHANNEL_LOG,
> TC>SWITCH_LOG_CONSOLE, "ANSWER: suppose the call is Answered Now\n");
> TC>                        PTRACE(4, "mod_h323\tMedia started on connection
> "
> TC><< *this);
> TC>
> TC>                        // test
> TC>                        //switch_channel_mark_answered(m_fsChannel);
> TC>
> TC>                        m_rxAudioOpened.Wait();
> TC>                        switch_log_printf(SWITCH_CHANNEL_LOG,
> TC>SWITCH_LOG_CONSOLE, "ANSWER: wait for m_rxAudioOpened\n");
> TC>                        //m_txAudioOpened.Wait();
> TC>                        switch_log_printf(SWITCH_CHANNEL_LOG,
> TC>SWITCH_LOG_CONSOLE, "ANSWER: we disable wait for m_txAudioOpened\n");
> TC>
> TC>                        switch_log_printf(SWITCH_CHANNEL_LOG,
> TC>SWITCH_LOG_CONSOLE, "ANSWER: were waiting for rx/tx AudioOpen\n");
> TC>
> TC>                        if (!switch_channel_test_flag(m_fsChannel,
> TC>CF_EARLY_MEDIA)) {
> TC>
> TC>                                switch_log_printf(SWITCH_CHANNEL_LOG,
> TC>SWITCH_LOG_CONSOLE, "ANSWER: we have early media\n");
> TC>
> TC>                                PTRACE(4,
> TC>"mod_h323\t-------------------->switch_channel_mark_answered(m_fsChannel)
> "
> TC><< *this);
> TC>
>  switch_channel_mark_answered(m_fsChannel);
> TC>                                switch_log_printf(SWITCH_CHANNEL_LOG,
> TC>SWITCH_LOG_CONSOLE, "ANSWER: answered in early Media\n");
> TC>                        }
> TC>                        break;
> TC>                }
> TC>
> TC>
> TC>Now, I'm able to both originate and terminate cals with 2-way audio...
> TC>the signaling looks correct...
> TC>
> TC>
> TC>
> TC>outgoing:
> TC>
> TC>1369.425046    10.4.62.7 -> 10.4.62.89   SIP/SDP Request: INVITE
> TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> <sip%3A1001 at 10.4.62.89<sip%253A1001 at 10.4.62.89>>;transport=udp,
> with session
> TC>description
> TC>1369.426255    10.4.62.7 -> 10.4.62.31   H.225.0 CS: alerting
> TC>1369.435950   10.4.62.89 -> 10.4.62.7    SIP Status: 100 Trying
> TC>1369.449065   10.4.62.89 -> 10.4.62.7    SIP Status: 180 Ringing
> TC>1369.605109    10.4.62.7 -> 10.4.62.31   H.225.0 CS: progress
> TC>OpenLogicalChannel
> TC>1369.609788   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
> TC>terminalCapabilitySet
> TC>1369.610489   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
> TC>masterSlaveDetermination
> TC>1369.619071    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
> TC>terminalCapabilitySet
> TC>1369.620349    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
> TC>terminalCapabilitySetAck
> TC>1369.623215   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
> TC>terminalCapabilitySetAck
> TC>1369.625591    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
> TC>masterSlaveDeterminationAck
> TC>1369.628174   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
> TC>masterSlaveDeterminationAck
> TC>1370.966958   10.4.62.89 -> 10.4.62.7    SIP/SDP Status: 200 OK, with
> TC>session description
> TC>1370.967431    10.4.62.7 -> 10.4.62.89   SIP Request: ACK
> TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> <sip%3A1001 at 10.4.62.89<sip%253A1001 at 10.4.62.89>
> >;transport=udp
> TC>1370.975172    10.4.62.7 -> 10.4.62.31   H.225.0 CS: connect
> TC>1372.354383   10.4.62.89 -> 10.4.62.7    SIP Request: BYE
> TC>sip:mod_sofia at 10.4.62.7:5060
> TC>1372.355147    10.4.62.7 -> 10.4.62.89   SIP Status: 200 OK
> TC>1372.392904    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS:
> releaseComplete
> TC>endSessionCommand
> TC>1372.397302   10.4.62.31 -> 10.4.62.7    H.225.0 CS: releaseComplete
> TC>
> TC>
> TC>incoming:
> TC>
> TC>
> TC>1502.817154   10.4.62.31 -> 10.4.62.7    H.225.0 CS: setup
> TC>OpenLogicalChannel
> TC>1502.833732    10.4.62.7 -> 10.4.62.31   H.225.0 CS: callProceeding
> TC>1502.850909    10.4.62.7 -> 10.4.62.89   SIP/SDP Request: INVITE
> TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> <sip%3A1001 at 10.4.62.89<sip%253A1001 at 10.4.62.89>>;transport=udp,
> with session
> TC>description
> TC>1502.851758    10.4.62.7 -> 10.4.62.31   H.225.0 CS: alerting
> TC>1502.861828   10.4.62.89 -> 10.4.62.7    SIP Status: 100 Trying
> TC>1502.875127   10.4.62.89 -> 10.4.62.7    SIP Status: 180 Ringing
> TC>1503.033258    10.4.62.7 -> 10.4.62.31   H.225.0 CS: progress
> TC>OpenLogicalChannel
> TC>1503.037908   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
> TC>terminalCapabilitySet
> TC>1503.038608   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
> TC>masterSlaveDetermination
> TC>1503.050154    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
> TC>terminalCapabilitySet
> TC>1503.051381    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
> TC>terminalCapabilitySetAck
> TC>1503.054297   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
> TC>terminalCapabilitySetAck
> TC>1503.054917    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
> TC>masterSlaveDeterminationAck
> TC>1503.057933   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
> TC>masterSlaveDeterminationAck
> TC>1505.485493   10.4.62.89 -> 10.4.62.7    SIP/SDP Status: 200 OK, with
> TC>session description
> TC>1505.486018    10.4.62.7 -> 10.4.62.89   SIP Request: ACK
> TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> <sip%3A1001 at 10.4.62.89<sip%253A1001 at 10.4.62.89>
> >;transport=udp
> TC>1505.493611    10.4.62.7 -> 10.4.62.31   H.225.0 CS: connect
> TC>1509.565959   10.4.62.89 -> 10.4.62.7    SIP Request: BYE
> TC>sip:mod_sofia at 10.4.62.7:5060
> TC>1509.566722    10.4.62.7 -> 10.4.62.89   SIP Status: 200 OK
> TC>1509.577435    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS:
> releaseComplete
> TC>endSessionCommand
> TC>1509.582066   10.4.62.31 -> 10.4.62.7    H.225.0 CS: releaseComplete
> TC>
> TC>
> TC>
> TC>... i still need to check the CDRs as well but here we are :)
> TC>
>
> can you send a diff? in you call scheme call from h323 endpoint to fs is
> not have RAS?,
> because i don't have no audio issues in transit from h323 to sip, but my
> calls a going
> thorough GK and fs is regitered on them, my call scheme is
> h323ep-RAS->GK-RAS->fs.
>
> C ?????????                       With Best Regards
> ???????????? ????.                Georgiewskiy Yuriy
> +7 4872 711666                    +7 4872 711666
> ???? +7 4872 711143               fax +7 4872 711143
> ???????? ??? "?? ?? ??????"       IT Service Ltd
> http://nkoort.ru                  http://nkoort.ru
> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
> YG129-RIPE                        YG129-RIPE
>
> _______________________________________________
> FreeSWITCH-users mailing list
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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-- 
Anthony Minessale II

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From bottleman at icf.org.ru  Fri Oct 23 08:46:58 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Fri, 23 Oct 2009 19:46:58 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <191c3a030910230837x78ea6093i387617a1adfdf958@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
	<Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>
	<65d96fc80910230452s6b4a179aqa5bc41dc34efc403@mail.gmail.com>
	<Pine.LNX.4.64.0910231610220.21492@srv.icf.org.ru>
	<65d96fc80910230538w7105be6fve14956ab3b247bf6@mail.gmail.com>
	<Pine.LNX.4.64.0910231646210.29582@srv.icf.org.ru>
	<65d96fc80910230614m2b2504d0r92006b78d267a399@mail.gmail.com>
	<65d96fc80910230757h4595f396o2dc1a1bb781d3a8a@mail.gmail.com>
	<Pine.LNX.4.64.0910231903510.17804@srv.icf.org.ru>
	<191c3a030910230837x78ea6093i387617a1adfdf958@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910231943250.22146@srv.icf.org.ru>

On 2009-10-23 10:37 -0500, Anthony Minessale wrote freeswitch-users at lists.f...:

i have no way to install trunk at this time, i will go out of hospital about one week later, after 
this i will can try it on trunk.

AM>if you were on trunk that line of code would be gone.
AM>you really can't do development on 1.0.4 its 6 months old and it will cause
AM>you more trouble than you think when you eventually upgrade if you do not do
AM>it soon.
AM>
AM>
AM>2009/10/23 Georgiewskiy Yuriy <bottleman at icf.org.ru>
AM>
AM>> On 2009-10-23 16:57 +0200, Tihomir Culjaga wrote
AM>> freeswitch-users at lists.fre...:
AM>>
AM>> i have question to developers about one proce in fs
AM>>
AM>> src/switch_ivr_originate.c
AM>>
AM>> static switch_status_t
AM>> originate_on_consume_media_transmit(switch_core_session_t *session)
AM>> {
AM>>    switch_channel_t *channel = switch_core_session_get_channel(session);
AM>>
AM>>                if (!switch_channel_test_flag(channel, CF_PROXY_MODE)) {
AM>>                while (switch_channel_get_state(channel) == CS_CONSUME_MEDIA
AM>> && !switch_channel_test_flag(chann
AM>>                        if (!switch_channel_media_ready(channel)) {
AM>>                            switch_yield(10000);
AM>>                        } else {
AM>>                                switch_ivr_sleep(session, 10, SWITCH_FALSE,
AM>> NULL);
AM>>                        }
AM>>                }
AM>>            }
AM>>
AM>>        switch_channel_clear_state_handler(channel,
AM>> &originate_state_handlers);
AM>>
AM>>        return SWITCH_STATUS_FALSE;
AM>> }
AM>>
AM>> what exacly it do?
AM>>
AM>> call scheme like this sip->fs->h323->gk->h323(on same fs)->fs(same too) and
AM>> there i have no audio issues.
AM>> if bridge connect while it sleep i have audio, if it not sleep while bridge
AM>> connect i have no audio.
AM>>
AM>> TC>a solution to H323 endpoint => FS => SIP user no audio issue
AM>> TC>
AM>> TC>is to disable a wait for tx Audio ... for  case
AM>> TC>SWITCH_MESSAGE_INDICATE_ANSWER:{
AM>> TC>
AM>> TC>//m_txAudioOpened.Wait();
AM>> TC>
AM>> TC>
AM>> TC>                case SWITCH_MESSAGE_INDICATE_ANSWER:{
AM>> TC>
AM>> TC>                        switch_log_printf(SWITCH_CHANNEL_LOG,
AM>> TC>SWITCH_LOG_CONSOLE, "ANSWER: we got Answer event\n");
AM>> TC>
AM>> TC>                        if (switch_channel_test_flag(channel,
AM>> CF_OUTBOUND))
AM>> TC>{
AM>> TC>
AM>> TC>                                switch_log_printf(SWITCH_CHANNEL_LOG,
AM>> TC>SWITCH_LOG_CONSOLE, "ANSWER: we got Answer event - CF_OUTBOUND
AM>> TC>\n");
AM>> TC>                                return SWITCH_STATUS_FALSE;
AM>> TC>                        }
AM>> TC>                        AnsweringCall(H323Connection::AnswerCallNow);
AM>> TC>
AM>> TC>                        switch_log_printf(SWITCH_CHANNEL_LOG,
AM>> TC>SWITCH_LOG_CONSOLE, "ANSWER: suppose the call is Answered Now\n");
AM>> TC>                        PTRACE(4, "mod_h323\tMedia started on connection
AM>> "
AM>> TC><< *this);
AM>> TC>
AM>> TC>                        // test
AM>> TC>                        //switch_channel_mark_answered(m_fsChannel);
AM>> TC>
AM>> TC>                        m_rxAudioOpened.Wait();
AM>> TC>                        switch_log_printf(SWITCH_CHANNEL_LOG,
AM>> TC>SWITCH_LOG_CONSOLE, "ANSWER: wait for m_rxAudioOpened\n");
AM>> TC>                        //m_txAudioOpened.Wait();
AM>> TC>                        switch_log_printf(SWITCH_CHANNEL_LOG,
AM>> TC>SWITCH_LOG_CONSOLE, "ANSWER: we disable wait for m_txAudioOpened\n");
AM>> TC>
AM>> TC>                        switch_log_printf(SWITCH_CHANNEL_LOG,
AM>> TC>SWITCH_LOG_CONSOLE, "ANSWER: were waiting for rx/tx AudioOpen\n");
AM>> TC>
AM>> TC>                        if (!switch_channel_test_flag(m_fsChannel,
AM>> TC>CF_EARLY_MEDIA)) {
AM>> TC>
AM>> TC>                                switch_log_printf(SWITCH_CHANNEL_LOG,
AM>> TC>SWITCH_LOG_CONSOLE, "ANSWER: we have early media\n");
AM>> TC>
AM>> TC>                                PTRACE(4,
AM>> TC>"mod_h323\t-------------------->switch_channel_mark_answered(m_fsChannel)
AM>> "
AM>> TC><< *this);
AM>> TC>
AM>>  switch_channel_mark_answered(m_fsChannel);
AM>> TC>                                switch_log_printf(SWITCH_CHANNEL_LOG,
AM>> TC>SWITCH_LOG_CONSOLE, "ANSWER: answered in early Media\n");
AM>> TC>                        }
AM>> TC>                        break;
AM>> TC>                }
AM>> TC>
AM>> TC>
AM>> TC>Now, I'm able to both originate and terminate cals with 2-way audio...
AM>> TC>the signaling looks correct...
AM>> TC>
AM>> TC>
AM>> TC>
AM>> TC>outgoing:
AM>> TC>
AM>> TC>1369.425046    10.4.62.7 -> 10.4.62.89   SIP/SDP Request: INVITE
AM>> TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> <sip%3A1001 at 10.4.62.89<sip%253A1001 at 10.4.62.89>>;transport=udp,
AM>> with session
AM>> TC>description
AM>> TC>1369.426255    10.4.62.7 -> 10.4.62.31   H.225.0 CS: alerting
AM>> TC>1369.435950   10.4.62.89 -> 10.4.62.7    SIP Status: 100 Trying
AM>> TC>1369.449065   10.4.62.89 -> 10.4.62.7    SIP Status: 180 Ringing
AM>> TC>1369.605109    10.4.62.7 -> 10.4.62.31   H.225.0 CS: progress
AM>> TC>OpenLogicalChannel
AM>> TC>1369.609788   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
AM>> TC>terminalCapabilitySet
AM>> TC>1369.610489   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
AM>> TC>masterSlaveDetermination
AM>> TC>1369.619071    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
AM>> TC>terminalCapabilitySet
AM>> TC>1369.620349    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
AM>> TC>terminalCapabilitySetAck
AM>> TC>1369.623215   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
AM>> TC>terminalCapabilitySetAck
AM>> TC>1369.625591    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
AM>> TC>masterSlaveDeterminationAck
AM>> TC>1369.628174   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
AM>> TC>masterSlaveDeterminationAck
AM>> TC>1370.966958   10.4.62.89 -> 10.4.62.7    SIP/SDP Status: 200 OK, with
AM>> TC>session description
AM>> TC>1370.967431    10.4.62.7 -> 10.4.62.89   SIP Request: ACK
AM>> TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> <sip%3A1001 at 10.4.62.89<sip%253A1001 at 10.4.62.89>
AM>> >;transport=udp
AM>> TC>1370.975172    10.4.62.7 -> 10.4.62.31   H.225.0 CS: connect
AM>> TC>1372.354383   10.4.62.89 -> 10.4.62.7    SIP Request: BYE
AM>> TC>sip:mod_sofia at 10.4.62.7:5060
AM>> TC>1372.355147    10.4.62.7 -> 10.4.62.89   SIP Status: 200 OK
AM>> TC>1372.392904    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS:
AM>> releaseComplete
AM>> TC>endSessionCommand
AM>> TC>1372.397302   10.4.62.31 -> 10.4.62.7    H.225.0 CS: releaseComplete
AM>> TC>
AM>> TC>
AM>> TC>incoming:
AM>> TC>
AM>> TC>
AM>> TC>1502.817154   10.4.62.31 -> 10.4.62.7    H.225.0 CS: setup
AM>> TC>OpenLogicalChannel
AM>> TC>1502.833732    10.4.62.7 -> 10.4.62.31   H.225.0 CS: callProceeding
AM>> TC>1502.850909    10.4.62.7 -> 10.4.62.89   SIP/SDP Request: INVITE
AM>> TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> <sip%3A1001 at 10.4.62.89<sip%253A1001 at 10.4.62.89>>;transport=udp,
AM>> with session
AM>> TC>description
AM>> TC>1502.851758    10.4.62.7 -> 10.4.62.31   H.225.0 CS: alerting
AM>> TC>1502.861828   10.4.62.89 -> 10.4.62.7    SIP Status: 100 Trying
AM>> TC>1502.875127   10.4.62.89 -> 10.4.62.7    SIP Status: 180 Ringing
AM>> TC>1503.033258    10.4.62.7 -> 10.4.62.31   H.225.0 CS: progress
AM>> TC>OpenLogicalChannel
AM>> TC>1503.037908   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
AM>> TC>terminalCapabilitySet
AM>> TC>1503.038608   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
AM>> TC>masterSlaveDetermination
AM>> TC>1503.050154    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
AM>> TC>terminalCapabilitySet
AM>> TC>1503.051381    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
AM>> TC>terminalCapabilitySetAck
AM>> TC>1503.054297   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
AM>> TC>terminalCapabilitySetAck
AM>> TC>1503.054917    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS: empty
AM>> TC>masterSlaveDeterminationAck
AM>> TC>1503.057933   10.4.62.31 -> 10.4.62.7    H.225.0/H.245 CS: facility
AM>> TC>masterSlaveDeterminationAck
AM>> TC>1505.485493   10.4.62.89 -> 10.4.62.7    SIP/SDP Status: 200 OK, with
AM>> TC>session description
AM>> TC>1505.486018    10.4.62.7 -> 10.4.62.89   SIP Request: ACK
AM>> TC>sip:1001 at 10.4.62.89 <sip%3A1001 at 10.4.62.89> <sip%3A1001 at 10.4.62.89<sip%253A1001 at 10.4.62.89>
AM>> >;transport=udp
AM>> TC>1505.493611    10.4.62.7 -> 10.4.62.31   H.225.0 CS: connect
AM>> TC>1509.565959   10.4.62.89 -> 10.4.62.7    SIP Request: BYE
AM>> TC>sip:mod_sofia at 10.4.62.7:5060
AM>> TC>1509.566722    10.4.62.7 -> 10.4.62.89   SIP Status: 200 OK
AM>> TC>1509.577435    10.4.62.7 -> 10.4.62.31   H.225.0/H.245 CS:
AM>> releaseComplete
AM>> TC>endSessionCommand
AM>> TC>1509.582066   10.4.62.31 -> 10.4.62.7    H.225.0 CS: releaseComplete
AM>> TC>
AM>> TC>
AM>> TC>
AM>> TC>... i still need to check the CDRs as well but here we are :)
AM>> TC>
AM>>
AM>> can you send a diff? in you call scheme call from h323 endpoint to fs is
AM>> not have RAS?,
AM>> because i don't have no audio issues in transit from h323 to sip, but my
AM>> calls a going
AM>> thorough GK and fs is regitered on them, my call scheme is
AM>> h323ep-RAS->GK-RAS->fs.
AM>>
AM>> C ?????????                       With Best Regards
AM>> ???????????? ????.                Georgiewskiy Yuriy
AM>> +7 4872 711666                    +7 4872 711666
AM>> ???? +7 4872 711143               fax +7 4872 711143
AM>> ???????? ??? "?? ?? ??????"       IT Service Ltd
AM>> http://nkoort.ru                  http://nkoort.ru
AM>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
AM>> YG129-RIPE                        YG129-RIPE
AM>>
AM>> _______________________________________________
AM>> FreeSWITCH-users mailing list
AM>> FreeSWITCH-users at lists.freeswitch.org
AM>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
AM>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
AM>> http://www.freeswitch.org
AM>>
AM>>
AM>
AM>
AM>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From anthony.minessale at gmail.com  Fri Oct 23 08:51:57 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 23 Oct 2009 10:51:57 -0500
Subject: [Freeswitch-users] Qustion about INFO messages after
	Connect/Answer
In-Reply-To: <4AE1951A.1030906@ewetel.de>
References: <4AD859A4.5000101@ewetel.de> <4ADC8B6E.7070404@ewetel.de>
	<191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>
	<87f2f3b90910191435s7528e256r85df5f12fc634f24@mail.gmail.com>
	<4ADF483B.4050807@ewetel.de>
	<87f2f3b90910211048s208daf1eyfae501091ab75530@mail.gmail.com>
	<4AE07DF4.5040803@ewetel.de>
	<085B77EE-DA3F-4ADD-86DC-E59D1B6553EA@freeswitch.org>
	<191c3a030910221008n60341314hc7a7e6a55a50ef15@mail.gmail.com>
	<4AE1951A.1030906@ewetel.de>
Message-ID: <191c3a030910230851i6e0b081ajaf6c294a7f9cd3a8@mail.gmail.com>

should be even better in 15210


On Fri, Oct 23, 2009 at 6:35 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi Anthony,
>
> thanks! The "unknown" thing is gone as long as you don't use
> "originate_callee_id_name"
>
>
> The "originate_callee_id_name" chvar causes caller's AND callee's
> display ("caller name" line)to be updated with the content of that chvar
> as soon as you set it.
>
> Let's say you use "hubu" as originate_callee_id_name. As soon as callee
> has picked up the phone it's display ("caller name" line) is updated
> with "hubu" and right after that with caller's name. The caller's
> display ("callee name" line) is updated to "hubu" (as expected).
>
> So it's still not nice to see a name switch in callee's display for
> caller's name after picking up.
>
> In the following you see the SIP flows from
>
> a) caller to FS
> b) FS to callee
>
>
> The more interesting one is b) I guess
>
>
>
>
> #######################################################
> This is the sip flow from caller to FS, which is OK:
>
>
>
> INVITE sip:1000 at 85.16.246.12:5061;user=phone SIP/2.0
> 23/Oct/2009-13:26:44.730
>
> INVITE sip:1000 at 85.16.246.12:5061;user=phone SIP/2.0
> Via: SIP/2.0/UDP 85.16.245.206:1024;branch=z9hG4bK-zdcpisfjx11n;rport
> From: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
> To: <sip:1000 at 85.16.246.12:5061;user=phone>
> Call-ID: 3c32462e758f-y19kuulwdknf
> CSeq: 1 INVITE
> Max-Forwards: 70
> Contact: <sip:1001 at 85.16.245.206:1024;line=eg3wp69a>;reg-id=1
> X-Serialnumber: 0004134002CB
> P-Key-Flags: resolution="31x13", keys="4"
> User-Agent: snom820/8.2.16
> Accept: application/sdp
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
> PRACK, MESSAGE, INFO, UPDATE
> Allow-Events: talk, hold, refer, call-info
> Supported: timer, 100rel, replaces, from-change
> Session-Expires: 3600;refresher=uas
> Min-SE: 90
> Content-Type: application/sdp
> Content-Length: 734
>
> v=0
> o=root 1420196469 1420196469 IN IP4 85.16.245.206
> s=call
> c=IN IP4 85.16.245.206
> t=0 0
> m=audio 51286 RTP/SAVP 0 8 9 99 3 18 4 101
> a=crypto:1 AES_CM_128_HMAC_SHA1_32
> inline:CjdPN1m2iLKAyXrQ2pWkQfCMtiISlF94ItTdsDis
> a=rtpmap:0 pcmu/8000
> a=rtpmap:8 pcma/8000
> a=rtpmap:9 g722/8000
> a=rtpmap:99 g726-32/8000
> a=rtpmap:3 gsm/8000
> a=rtpmap:18 g729/8000
> a=rtpmap:4 g723/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
> m=audio 51286 RTP/AVP 0 8 9 99 3 18 4 101
> a=rtpmap:0 pcmu/8000
> a=rtpmap:8 pcma/8000
> a=rtpmap:9 g722/8000
> a=rtpmap:99 g726-32/8000
> a=rtpmap:3 gsm/8000
> a=rtpmap:18 g729/8000
> a=rtpmap:4 g723/8000
> a=rtpmap:101 telephone-ev
>
> SIP/2.0 100 Trying
> 23/Oct/2009-13:26:44.731
>
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 85.16.245.206:1024;branch=z9hG4bK-zdcpisfjx11n;rport=1024
> From: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
> To: <sip:1000 at 85.16.246.12:5061;user=phone>
> Call-ID: 3c32462e758f-y19kuulwdknf
> CSeq: 1 INVITE
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
> Content-Length: 0
>
>
> SIP/2.0 180 Ringing
> 23/Oct/2009-13:26:45.3
>
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 85.16.245.206:1024;branch=z9hG4bK-zdcpisfjx11n;rport=1024
> From: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
> To: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
> Call-ID: 3c32462e758f-y19kuulwdknf
> CSeq: 1 INVITE
> Contact: <sip:1000 at 85.16.246.12:5061;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
> Accept: application/sdp
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
> Content-Length: 0
> X-FS-Display-Name: hubu
> X-FS-Display-Number:
> X-FS-Support: update_display
>
>
> SIP/2.0 200 OK
> 23/Oct/2009-13:26:47.367
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 85.16.245.206:1024;branch=z9hG4bK-zdcpisfjx11n;rport=1024
> From: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
> To: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
> Call-ID: 3c32462e758f-y19kuulwdknf
> CSeq: 1 INVITE
> Contact: <sip:1000 at 85.16.246.12:5061;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
> Session-Expires: 3600;refresher=uas
> Min-SE: 120
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 354
> X-FS-Display-Name: hubu
> X-FS-Display-Number:
> X-FS-Support: update_display
>
> v=0
> o=FreeSWITCH 1256285701 1256285702 IN IP4 85.16.246.12
> s=FreeSWITCH
> c=IN IP4 85.16.246.12
> t=0 0
> m=audio 11506 RTP/SAVP 8 101
> a=rtpmap:8 pcma/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=crypto:1 AES_CM_128_HMAC_SHA1_32
> inline:/BFm/LUmDBuao0wqg6MS/l4acfzOHoHRFQr3N0R9
> m=audio 0 RTP/AVP 19
>
> ACK sip:1000 at 85.16.246.12:5061;transport=udp SIP/2.0
> 23/Oct/2009-13:26:47.392
>
> ACK sip:1000 at 85.16.246.12:5061;transport=udp SIP/2.0
> Via: SIP/2.0/UDP 85.16.245.206:1024;branch=z9hG4bK-d474to81d7gh;rport
> From: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
> To: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
> Call-ID: 3c32462e758f-y19kuulwdknf
> CSeq: 1 ACK
> Max-Forwards: 70
> Contact: <sip:1001 at 85.16.245.206:1024;line=eg3wp69a>;reg-id=1
> Content-Length: 0
>
>
> INFO sip:1001 at 85.16.245.206:1024;line=eg3wp69a SIP/2.0
> 23/Oct/2009-13:26:47.416
>
> INFO sip:1001 at 85.16.245.206:1024;line=eg3wp69a SIP/2.0
> Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKB244eKZ08Uj3H
> Max-Forwards: 70
> From: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
> To: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
> Call-ID: 3c32462e758f-y19kuulwdknf
> CSeq: 122030267 INFO
> Contact: <sip:1000 at 85.16.246.12:5061;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Content-Type: message/sipfrag
> Content-Length: 24
>
> From:
> To: "hubu" 1000
>
> SIP/2.0 200 Ok
> 23/Oct/2009-13:26:47.619
>
> SIP/2.0 200 Ok
> Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKB244eKZ08Uj3H
> From: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
> To: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
> Call-ID: 3c32462e758f-y19kuulwdknf
> CSeq: 122030267 INFO
> Contact: <sip:1001 at 85.16.245.206:1024;line=eg3wp69a>;reg-id=1
> Content-Length: 0
>
>
> BYE sip:1001 at 85.16.245.206:1024;line=eg3wp69a SIP/2.0
> 23/Oct/2009-13:26:49.18
>
> BYE sip:1001 at 85.16.245.206:1024;line=eg3wp69a SIP/2.0
> Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKDmQpj9072DZ8r
> Max-Forwards: 70
> From: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
> To: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
> Call-ID: 3c32462e758f-y19kuulwdknf
> CSeq: 122030268 BYE
> Contact: <sip:1000 at 85.16.246.12:5061;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Reason: Q.850;cause=16;text="NORMAL_CLEARING"
> Content-Length: 0
>
>
> SIP/2.0 200 OK
> 23/Oct/2009-13:26:49.52
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKDmQpj9072DZ8r
> From: <sip:1000 at 85.16.246.12:5061;user=phone>;tag=QD9eD62NH0gQj
> To: "1001 an PBX1" <sip:1001 at 85.16.246.12:5061>;tag=724ehd063s
> Call-ID: 3c32462e758f-y19kuulwdknf
> CSeq: 122030268 BYE
> Contact: <sip:1001 at 85.16.245.206:1024;line=eg3wp69a>;reg-id=1
> User-Agent: snom820/8.2.16
> RTP-RxStat:
> Total_Rx_Pkts=65,Rx_Pkts=65,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0
> RTP-TxStat: Total_Tx_Pkts=0,Tx_Pkts=94,Remote_Tx_Pkts=0
> Content-Length: 0
>
>
>
>
>
> ####################################################################
>
> This is the SIP flow from FS to callee.
>
> The first INFO sent from FS to callee is the Problem. It contains "hubu"
> as "To:" header. If there is s need for this message I guess the "From"
> header should be changed, because from the callee-phone's point of view
> the "To:" header is the caller and the "From" header is the callee. In
> my opinion this INFO message shouldn't be sent at all, but I don't know
> what you intend it to do.
>
>
> INVITE sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
> 23/Oct/2009-13:26:44.978
>
> INVITE sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
> Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bK86rt91cpH1FBF
> Max-Forwards: 68
> From: "1001 an PBX1" <sip:1001 at 85.16.246.12 <sip%3A1001 at 85.16.246.12>
> >;tag=rp27e1KSe969D
> To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>
> Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
> CSeq: 122030266 INVITE
> Contact: <sip:mod_sofia at 85.16.246.12:5061>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 325
> X-Serialnumber: 0004134002CB
> P-Key-Flags: resolution="31x13", keys="4"
> X-FS-Support: update_display
> Remote-Party-ID: "1001 an PBX1"
> <sip:1001 at 85.16.246.12 <sip%3A1001 at 85.16.246.12>
> >;party=calling;screen=yes;privacy=off
>
> v=0
> o=FreeSWITCH 1256285400 1256285401 IN IP4 85.16.246.12
> s=FreeSWITCH
> c=IN IP4 85.16.246.12
> t=0 0
> m=audio 11804 RTP/AVP 95 103 9 8 101 13
> a=rtpmap:95 CELT/48000
> a=rtpmap:103 SPEEX/32000
> a=rtpmap:9 G722/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:10
>
> SIP/2.0 180 Ringing
> 23/Oct/2009-13:26:45.2
>
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bK86rt91cpH1FBF
> From: "1001 an PBX1" <sip:1001 at 85.16.246.12 <sip%3A1001 at 85.16.246.12>
> >;tag=rp27e1KSe969D
> To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
> Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
> CSeq: 122030266 INVITE
> Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
> PRACK, MESSAGE, INFO, UPDATE
> Allow-Events: talk, hold, refer, call-info
> Content-Length: 0
>
>
> SIP/2.0 180 Ringing
> 23/Oct/2009-13:26:45.504
>
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bK86rt91cpH1FBF
> From: "1001 an PBX1" <sip:1001 at 85.16.246.12 <sip%3A1001 at 85.16.246.12>
> >;tag=rp27e1KSe969D
> To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
> Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
> CSeq: 122030266 INVITE
> Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
> PRACK, MESSAGE, INFO, UPDATE
> Allow-Events: talk, hold, refer, call-info
> Content-Length: 0
>
>
> SIP/2.0 180 Ringing
> 23/Oct/2009-13:26:46.504
>
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bK86rt91cpH1FBF
> From: "1001 an PBX1" <sip:1001 at 85.16.246.12 <sip%3A1001 at 85.16.246.12>
> >;tag=rp27e1KSe969D
> To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
> Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
> CSeq: 122030266 INVITE
> Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
> PRACK, MESSAGE, INFO, UPDATE
> Allow-Events: talk, hold, refer, call-info
> Content-Length: 0
>
>
> SIP/2.0 200 Ok
> 23/Oct/2009-13:26:47.361
>
> SIP/2.0 200 Ok
> Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bK86rt91cpH1FBF
> From: "1001 an PBX1" <sip:1001 at 85.16.246.12 <sip%3A1001 at 85.16.246.12>
> >;tag=rp27e1KSe969D
> To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
> Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
> CSeq: 122030266 INVITE
> Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
> User-Agent: snom820/LAB_HD_HEAD_11_09_09_11_20
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
> PRACK, MESSAGE, INFO, UPDATE
> Allow-Events: talk, hold, refer, call-info
> Supported: timer, 100rel, replaces, from-change
> Content-Type: application/sdp
> Content-Length: 244
>
> v=0
> o=root 353690302 353690303 IN IP4 85.16.245.228
> s=call
> c=IN IP4 85.16.245.228
> t=0 0
> m=audio 60878 RTP/AVP 9 8 101
> a=rtpmap:9 g722/8000
> a=rtpmap:8 pcma/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:10
> a=sendrecv
>
> ACK sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
> 23/Oct/2009-13:26:47.362
>
> ACK sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
> Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bK9FjKBXXSea6Xa
> Max-Forwards: 70
> From: "1001 an PBX1" <sip:1001 at 85.16.246.12 <sip%3A1001 at 85.16.246.12>
> >;tag=rp27e1KSe969D
> To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
> Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
> CSeq: 122030266 ACK
> Contact: <sip:mod_sofia at 85.16.246.12:5061>
> Content-Length: 0
>
>
> INFO sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
> 23/Oct/2009-13:26:47.367
>
> INFO sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
> Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKaSBcDreXBKvgp
> Max-Forwards: 70
> From: "1001 an PBX1" <sip:1001 at 85.16.246.12 <sip%3A1001 at 85.16.246.12>
> >;tag=rp27e1KSe969D
> To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
> Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
> CSeq: 122030267 INFO
> Contact: <sip:mod_sofia at 85.16.246.12:5061>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Content-Type: message/sipfrag
> Content-Length: 24
>
> From:
> To: "hubu" 1000
>
> SIP/2.0 200 Ok
> 23/Oct/2009-13:26:47.766
>
> SIP/2.0 200 Ok
> Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKaSBcDreXBKvgp
> From: "1001 an PBX1" <sip:1001 at 85.16.246.12 <sip%3A1001 at 85.16.246.12>
> >;tag=rp27e1KSe969D
> To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
> Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
> CSeq: 122030267 INFO
> Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
> Content-Length: 0
>
>
> INFO sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
> 23/Oct/2009-13:26:47.766
>
> INFO sip:1000 at 85.16.245.228:1024;line=5jkc7koy SIP/2.0
> Via: SIP/2.0/UDP 85.16.246.12:5061;rport;branch=z9hG4bKcByXgeg4548ND
> Max-Forwards: 70
> From: "1001 an PBX1" <sip:1001 at 85.16.246.12 <sip%3A1001 at 85.16.246.12>
> >;tag=rp27e1KSe969D
> To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
> Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
> CSeq: 122030268 INFO
> Contact: <sip:mod_sofia at 85.16.246.12:5061>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Content-Type: message/sipfrag
> Content-Length: 32
>
> From:
> To: "1001 an PBX1" 1001
>
> SIP/2.0 200 Ok
> 23/Oct/2009-13:26:47.860
>
> SIP/2.0 200 Ok
> Via: SIP/2.0/UDP 85.16.246.12:5061;rport=5061;branch=z9hG4bKcByXgeg4548ND
> From: "1001 an PBX1" <sip:1001 at 85.16.246.12 <sip%3A1001 at 85.16.246.12>
> >;tag=rp27e1KSe969D
> To: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
> Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
> CSeq: 122030268 INFO
> Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
> Content-Length: 0
>
>
> BYE sip:mod_sofia at 85.16.246.12:5061 SIP/2.0
> 23/Oct/2009-13:26:49.3
>
> BYE sip:mod_sofia at 85.16.246.12:5061 SIP/2.0
> Via: SIP/2.0/UDP 85.16.245.228:1024;branch=z9hG4bK-susgzpvc38s7;rport
> From: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
> To: "1001 an PBX1" <sip:1001 at 85.16.246.12 <sip%3A1001 at 85.16.246.12>
> >;tag=rp27e1KSe969D
> Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
> CSeq: 1 BYE
> Max-Forwards: 70
> Contact: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;reg-id=1
> User-Agent: snom820/LAB_HD_HEAD_11_09_09_11_20
> RTP-RxStat: Total_Rx_Pkts=0,Rx_Pkts=69,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0
> RTP-TxStat: Total_Tx_Pkts=70,Tx_Pkts=65,Remote_Tx_Pkts=0
> Content-Length: 0
>
>
> SIP/2.0 200 OK
> 23/Oct/2009-13:26:49.4
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 85.16.245.228:1024;branch=z9hG4bK-susgzpvc38s7;rport=1024
> From: <sip:1000 at 85.16.245.228:1024;line=5jkc7koy>;tag=a7mygnrad4
> To: "1001 an PBX1" <sip:1001 at 85.16.246.12 <sip%3A1001 at 85.16.246.12>
> >;tag=rp27e1KSe969D
> Call-ID: c93948e5-3a69-122d-94bc-00144fe6907a
> CSeq: 1 BYE
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15203M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Content-Length: 0
>
>
>
>
>
>
>
>
>
>
> On 22.10.2009 19:08, Anthony Minessale wrote:
> > I'm sure that problem is gone in svn trunk.
> >
> >
> > On Thu, Oct 22, 2009 at 11:25 AM, Brian West <brian at freeswitch.org
> > <mailto:brian at freeswitch.org>> wrote:
> >
> >     I can't get what exactly you re talking about. Can you clarify ? Also
> >     please include the packets of interest only not the full trace if its
> >     not relevant to the bug.
> >
> >     /b
> >
> >     On Oct 22, 2009, at 10:44 AM, Helmut Kuper wrote:
> >
> >     > -----BEGIN PGP SIGNED MESSAGE-----
> >     > Hash: SHA1
> >     >
> >     > Hi Mike,
> >     >
> >     > here it is:
> >     >
> >     >
> >     > Dialplan:
> >     >
> >     >
> >     >    <extension name="Local_Extension">
> >     >      <condition field="destination_number"
> expression="^(10[01][0-9])
> >     > $">
> >     >        <action application="set" data="dialed_extension=$1"/>
> >     >        <action application="export" data="dialed_extension=$1"/>
> >     >        <action application="set" data="transfer_ringback=$$
> >     > {hold_music}"/>
> >     >        <action application="set" data="hangup_after_bridge=true"/>
> >     >        <action application="bridge"
> >     > data="user/${dialed_extension}@${domain_name}"/>
> >     >      </condition>
> >     >    </extension>
> >     >
> >
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> > iax:guest at conference.freeswitch.org/888
> > <http://iax:guest at conference.freeswitch.org/888>
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> > pstn:213-799-1400
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> - --
>
> Mit freundlichen Gr??en
> Helmut Kuper
> Gesch?ftseinheit FD - L?sungen f?r Finanzdienstleister
> Telefax: (0441) 8000-2799
> mailto:helmut.kuper at ewetel.de
> ___________________________________
> EWE TEL GmbH
> Cloppenburger Stra?e 310
> 26133 Oldenburg
> EWE TEL GmbH
>
> Handelsregister Amtsgericht Oldenburg HRB 3723
> Vorsitzender des Aufsichtsrates: Heiko Harms
> Gesch?ftsf?hrung: Hans-Joachim Iken (Vorsitzender),
> Ulf Heggenberger, Dr. Norbert Schulz, Dirk Thole
> Homepage: http://www.ewetel.de
> ___________________________________
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
>
> iD8DBQFK4ZUZ4tZeNddg3dwRAr+lAJ9b3geDcwT0bkz++YG9Sjq62AtHBQCfWALu
> jkUQIoNN6CYJFp10ebA5MIU=
> =cHej
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Fri Oct 23 09:00:48 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 23 Oct 2009 09:00:48 -0700
Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Starting
	Shortly, Please Start 	Calling In
Message-ID: <87f2f3b90910230900l2201894bjc7a57c8093617b8@mail.gmail.com>

FYI,

The weekly call will begin soon. The agenda is here:
http://bit.ly/O0oGB

Note that the wiki is giving me problems with editing. I will let you know
when Raymond and I get it squared away.
-MC
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From tculjaga at gmail.com  Fri Oct 23 09:26:40 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 23 Oct 2009 18:26:40 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910231943250.22146@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910230452s6b4a179aqa5bc41dc34efc403@mail.gmail.com>
	<Pine.LNX.4.64.0910231610220.21492@srv.icf.org.ru>
	<65d96fc80910230538w7105be6fve14956ab3b247bf6@mail.gmail.com>
	<Pine.LNX.4.64.0910231646210.29582@srv.icf.org.ru>
	<65d96fc80910230614m2b2504d0r92006b78d267a399@mail.gmail.com>
	<65d96fc80910230757h4595f396o2dc1a1bb781d3a8a@mail.gmail.com>
	<Pine.LNX.4.64.0910231903510.17804@srv.icf.org.ru>
	<191c3a030910230837x78ea6093i387617a1adfdf958@mail.gmail.com>
	<Pine.LNX.4.64.0910231943250.22146@srv.icf.org.ru>
Message-ID: <65d96fc80910230926y6c3e7ee3i9bb2182da1ba8757@mail.gmail.com>

2009/10/23 Georgiewskiy Yuriy <bottleman at icf.org.ru>

> On 2009-10-23 10:37 -0500, Anthony Minessale wrote
> freeswitch-users at lists.f...:
>
> i have no way to install trunk at this time, i will go out of hospital
> about one week later, after
> this i will can try it on trunk.
>
> AM>if you were on trunk that line of code would be gone.
> AM>you really can't do development on 1.0.4 its 6 months old and it will
> cause
> AM>you more trouble than you think when you eventually upgrade if you do
> not do
>

uh... you are still in there ... damn sorry to hear that.
I know what the feeling is... i was in the same position a couple of months
ago... anyhow at least i had plenty of time to do my private research.. hope
it is the case with you as well.

so, if you have that time i can offer you my development server with trunk
installed and all the hands on.


let me know...

BTW: it really doesn't have sense to develop on 1.0.4 ... the proof of
concept was done. I'm able to place calls in both directions so, lets move
to trunk now.

T.
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From ujjval at simplesignal.com  Fri Oct 23 09:54:48 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Fri, 23 Oct 2009 09:54:48 -0700
Subject: [Freeswitch-users] Setting up Conference with Moderator
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local>
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local>

Any ideas on this one. Look slike only way rite now is to have a different Dest Phone number for a moderator.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ujjval Karihaloo
Sent: Thursday, October 22, 2009 9:02 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Setting up Conference with Moderator

Hi

I have the Basic Conferencing working. Here is my Dial Plan.

 I want to be able to setup a Moderator PIN different from other participants, when I add the moderator flag it logs me in directly w/o asking for a PIN..
action application="conference" data="conference.conf at wideband+flags{moderator}+159753<mailto:conference.conf at wideband+flags%7bmoderator%7d+159753>"/>

DialPlan is below for the normal user, and it asks for the PIN with below settings.

       Ujj Inbound from SS - start
    -->
    <extension name="simplesignal">   <!-- your provider or any name you'd like to call it -->
     <condition field="destination_number" expression="2142349127">  <!-- your DID for this gateway-->
        <action application="conference" data="conference.conf at wideband+159753<mailto:conference.conf at wideband+159753>"/>
     </condition>
</extension>

    <!--
       Ujj Inbound from SS - end




And I am using the existing conference.conf.xml file in the auto_loads directory.


Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690

SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO  80112
[cid:image001.jpg at 01CA53CF.3D2B8B40]<http://www.simplesignal.com/>

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From saeedahmad1981 at gmail.com  Fri Oct 23 10:16:46 2009
From: saeedahmad1981 at gmail.com (Saeed Ahmad)
Date: Fri, 23 Oct 2009 19:16:46 +0200
Subject: [Freeswitch-users] CPU Usage Question
In-Reply-To: <4add4a7a.1408c00a.4283.ffffc11f@mx.google.com>
References: <4add4a7a.1408c00a.4283.ffffc11f@mx.google.com>
Message-ID: <c96b95650910231016me63190fr4641ed7737637a85@mail.gmail.com>

I think we can't create two profiles per single IP. So what is the
workaround to solve this issue?

SAT

On Tue, Oct 20, 2009 at 7:28 AM, brian at freeswitch.org
<brian at freeswitch.org>wrote:

> It's because Sofia used one thread per profile so depending on load you
> will see one CPU used more.
>
> /b
>
> Sent from my HTC on the Now Network from Sprint!
>
> ----- Reply message -----
> From: "DJB" <djbinter at yahoo.com>
> Date: Mon, Oct 19, 2009 9:13 PM
> Subject: [Freeswitch-users] CPU Usage Question
> To: "FREESWITCH-USERS MAILING LIST" <freeswitch-users at lists.freeswitch.org
> >
>
>
> We have about 1500 concurrent calls on the server.  And, the CPU usage is
> shown below:
>
> 02:06:46 AM  CPU   %user   %nice    %sys %iowait    %irq   %soft  %steal
> %idle    intr/s
> 02:06:46 AM  all    0.06    0.00    0.03    0.44    0.00    0.00    0.00
> 99.47   1028.25
> 02:06:46 AM    0    0.08    0.03    0.03    0.01    0.00    0.01    0.00
> 99.84   1007.07
> 02:06:46 AM    1    0.06    0.00    0.02    0.01    0.00    0.00    0.00
> 99.91      0.00
> 02:06:46 AM    2    0.07    0.00    0.08    3.49    0.02    0.00    0.00
> 96.34     21.18
> 02:06:46 AM    3    0.05    0.00    0.01    0.01    0.00    0.00    0.00
> 99.94      0.00
> 02:06:46 AM    4    0.04    0.00    0.02    0.00    0.00    0.00    0.00
> 99.93      0.00
> 02:06:46 AM    5    0.05    0.00    0.02    0.00    0.00    0.00    0.00
> 99.92      0.00
> 02:06:46 AM    6    0.05    0.00    0.02    0.02    0.00    0.00    0.00
> 99.91      0.00
> 02:06:46 AM    7    0.05    0.00    0.02    0.01    0.00    0.00    0.00
> 99.93      0.00
>
>
> 1) I am wondering whether it is normal that every time when i checked, only
> CPU 2 always have the highest %iowait (in this case 3.49%).
>
> 2) Shoudn't it spread out and balance them to utilize all CPUs evenly?
>
> Thank you.
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rob4manhere at gmail.com  Fri Oct 23 10:30:23 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Fri, 23 Oct 2009 12:30:23 -0500
Subject: [Freeswitch-users] Setting up Conference with Moderator
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local>
	<3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local>
Message-ID: <E97CD3D0-52FF-44D1-BB89-23816045CE21@gmail.com>

The first caller isn't challenged for the pin (don't really know why--  
maybe somebody else can elaborate on how the pin is designed to be  
used).  So to work around it, I validate the conference pin and  
moderator pin independently via an IVR (or dynamically with a script  
and odbc call) then put them in the conference with the right profile  
and flags.

Here is a simple example (I created these wavs so they aren't in your  
default setup but you can get the idea):

in ivr.conf:

     <menu name="conference_menu"
           greet-long="welcome_please_enter_conference_pin.wav"
           greet-short="check_and_try_again.wav"
	  invalid-sound="passcode_invalid.wav"
	  exit-sound="voicemail/vm-goodbye.wav"
           timeout="10000"
           inter-digit-timeout="5000"
           max-failures="3"
           max-timeouts="3"
           digit-len="7">
       <entry action="menu-sub" digits="123456"  
param="conference_123456_moderator_menu" /> <!-- conference moderator  
menu -->
     </menu>

     <menu name="conference_123456_moderator_menu"
           greet- 
long 
= 
"conference_confirmed_enter_moderator_pin_or_1_to_join_as_participant 
.wav"
           greet-short="check_moderator_pin_or_1_to_join.wav"
           invalid-sound="invalid_moderator_pin.wav"
           exit-sound="voicemail/vm-goodbye.wav"
           timeout="10000"
           inter-digit-timeout="5000"
           max-failures="3"
           max-timeouts="3"
           digit-len="5">
       <entry action="menu-exec-app" digits="1234" param="conference  
123456 at default+flags{moderator}" />
       <entry action="menu-exec-app" digits="1" param="conference  
123456 at default+flags{}" />
      </menu>


You also need to add the "wait-mod" conference-flag to the conference  
profile if you want to allow participants to join but not let them  
talk until the moderator arrives.

In conference.conf.xml add to the profile you're using (default in  
this example):

       <param name="conference-flags" value="wait-mod" />


 From your public dial plan call the IVR or script:
   <action application="ivr" data="conference_menu" />


Now you should be able to call, enter a conference number of 123456  
and sit listening to music until the moderator joins (using pin 1234)  
and starts the call.  Maybe I completely misunderstood the built-in  
pin system but since the first person isn't challenged I had to do it  
this way.

Good luck!

Rob


On Oct 23, 2009, at 11:54 AM, Ujjval Karihaloo wrote:

> Any ideas on this one. Look slike only way rite now is to have a  
> different Dest Phone number for a moderator.
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Ujjval Karihaloo
> Sent: Thursday, October 22, 2009 9:02 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Setting up Conference with Moderator
>
> Hi
>
> I have the Basic Conferencing working. Here is my Dial Plan.
>
>  I want to be able to setup a Moderator PIN different from other  
> participants, when I add the moderator flag it logs me in directly w/ 
> o asking for a PIN..
> action application="conference" data="conference.conf at wideband 
> +flags{moderator}+159753"/>
>
> DialPlan is below for the normal user, and it asks for the PIN with  
> below settings.
>
>        Ujj Inbound from SS - start
>     -->
>     <extension name="simplesignal">   <!-- your provider or any name  
> you'd like to call it -->
>      <condition field="destination_number" expression="2142349127">   
> <!-- your DID for this gateway-->
>         <action application="conference"  
> data="conference.conf at wideband+159753"/>
>      </condition>
> </extension>
>
>     <!--
>        Ujj Inbound from SS ? end
>
>
>
>
> And I am using the existing conference.conf.xml file in the  
> auto_loads directory.
>
>
> Ujjval Karihaloo
> VP Voice Engineering
> IP Phone: +13032428610
> E-Fax: +17202391690
>
> SimpleSignal Inc.
> 88 Inverness Circle East
> Suite K105
> Englewood, CO  80112
> <image001.jpg>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jerry.richards at teotech.com  Fri Oct 23 10:31:06 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 23 Oct 2009 10:31:06 -0700
Subject: [Freeswitch-users] Inbound DTMF Not Recognized By IVR
Message-ID: <573AFFBA055B4ECD83A910C12BE67129@greyhawk.tonecommander.com>


I installed FS on a machine with a Sangoma A101D (PRI) card and if I make an
inbound call to the FS IVR, it does not recognize DTMF digits from the PSTN
phone.  If I call IVR from an internal phone, then it does recognize the
DTMF digits.  I have mostly default configurations for everything.

Best Regards,
JErry




From brian at freeswitch.org  Fri Oct 23 10:43:01 2009
From: brian at freeswitch.org (Brian West)
Date: Fri, 23 Oct 2009 12:43:01 -0500
Subject: [Freeswitch-users] Setting up Conference with Moderator
In-Reply-To: <E97CD3D0-52FF-44D1-BB89-23816045CE21@gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local>
	<3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local>
	<E97CD3D0-52FF-44D1-BB89-23816045CE21@gmail.com>
Message-ID: <71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>

Well first off you're not defining a pine here...

confname at profilename+flags{mute|deaf|waste|moderator}+[conference pin  
number]

That might be why its not asking for a pin.

/b

On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:

>       <entry action="menu-exec-app" digits="1" param="conference  
> 123456 at default+flags{}" />




From chris at cloudtel.com  Fri Oct 23 10:47:43 2009
From: chris at cloudtel.com (Chris Burns)
Date: Fri, 23 Oct 2009 13:47:43 -0400
Subject: [Freeswitch-users] sched_api doesn't get launched
In-Reply-To: <191c3a030910230802m1cb847agfdaf6e436296e748@mail.gmail.com>
References: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
	<59ad9ca10910230154i6d41ddfci8957b152c1884800@mail.gmail.com>
	<191c3a030910230802m1cb847agfdaf6e436296e748@mail.gmail.com>
Message-ID: <200910231347.43883.chris@cloudtel.com>

poor bbhenry :)

Added in r15207 please test and update docu if necessary:
http://wiki.freeswitch.org/wiki/Variable_api_on_answer

On October 23, 2009 11:02:07 am Anthony Minessale wrote:
> it's probably related to escaping the data.
> I was sick of watching you suffer so i added api_on_answer variable to
> trunk.
>
> On Fri, Oct 23, 2009 at 3:54 AM, Henry Huang 
<red.rain.seven at gmail.com>wrote:
> > Thanks to c6burns on IRC channel for the tip to use execute_on_answer in
> > combination with eval, and of course everyone here that pointing me to
> > the right direction.
> > I was able to execute sched_api with eval, but not with the combination
> > of execute_on_answer. The argument just don't get parsed as the eval
> > argument. Here is the code:
> >
> > setVariable("execute_on_answer", "eval sched_api(+%ld none uuid_displace
> > <some uuid> start
> > /usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux)");
> >
> >
> > Here is the log:
> > 2009-10-23 16:17:53.633189 [DEBUG] switch_scheduler.c:214 Added task 2
> > sched_api_function (none) to run at 1256285888
> > EXECUTE sofia/internal/1688004 at 192.168.1.66 *eval(+OK *Added: 2
> > )
> >
> > 2009-10-23 16:18:04.52701 [DEBUG] switch_channel.c:1927 sofia/internal/
> > 1688004 at 192.168.1.66 *execute on answer: eval(sched_api(+37 none
> > uuid_displace 84d74f41-8668-4138-943f-f076e94046ad start
> > /usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux))*
> > EXECUTE sofia/internal/1688004 at 192.168.1.66 eval(sched_api(+37 none
> > uuid_displace 84d74f41-8668-4138-943f-f076e94046ad start
> > /usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux))
> >
> >
> > The first part of the log is when I just do execute("eval",
> > "${sched_api(+%ld none uuid_displace <some uuid> start
> > /usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux)}");
> > And I simple get a OK, and then later on the sched_api gets executed.
> >
> > However, on the excerpt of the second log. which I use the combination
> > with execute_on_answer like specified in the yellow hight code above. It
> > doesn't simply return a OK, instead it shows the whole blob of code. and
> > does nothing later.
> >
> >
> >
> > On Thu, Oct 22, 2009 at 4:09 AM, Anthony Minessale <
> >
> > anthony.minessale at gmail.com> wrote:
> >> Yes you need an API object as described in other email.
> >>
> >> Which line of code from java caused that segfault
> >> It looks like a simple NULL string issue that we may want to hunt down.
> >>
> >> On Wed, Oct 21, 2009 at 4:44 AM, Henry Huang 
<red.rain.seven at gmail.com>wrote:
> >>> I can't seem to find the right thing to use in mod_java to execute api
> >>> commands, only api_after_bridge
> >>>
> >>> 2009-10-21 17:42:46.593094 [NOTICE] mod_sofia.c:1509 Pre-Answer
> >>> sofia/internal/1688003 at 192.168.1.66!
> >>> #
> >>> # A fatal error has been detected by the Java Runtime Environment:
> >>> #
> >>> #  SIGSEGV (0xb) at pc=0x004e4480, pid=1927, tid=16116624
> >>> #
> >>> # JRE version: 6.0_16-b01
> >>> # Java VM: Java HotSpot(TM) Client VM (14.2-b01 mixed mode linux-x86 )
> >>> # Problematic frame:
> >>> # C  [libc.so.6+0x6f480]  strcpy+0x10
> >>> #
> >>> # An error report file with more information is saved as:
> >>> # /usr/local/freeswitch/bin/hs_err_pid1927.log
> >>> 2009-10-21 17:42:59.883729 [ERR] switch_core_session.c:1374 Invalid
> >>> Application sched_api
> >>> 2009-10-21 17:42:59.883729 [NOTICE] switch_core_session.c:1375 Hangup
> >>> sofia/internal/1688003 at 192.168.1.66 [CS_EXECUTE]
> >>> [DESTINATION_OUT_OF_ORDER]
> >>> #
> >>> # If you would like to submit a bug report, please visit:
> >>> #   http://java.sun.com/webapps/bugreport/crash.jsp
> >>> # The crash happened outside the Java Virtual Machine in native code.
> >>> # See problematic frame for where to report the bug.
> >>>
> >>> On Sat, Oct 17, 2009 at 3:37 AM, Michael Collins 
<msc at freeswitch.org>wrote:
> >>>> On Fri, Oct 16, 2009 at 11:53 AM, Henry Huang
> >>>> <red.rain.seven at gmail.com
> >>>>
> >>>> > wrote:
> >>>>>
> >>>>> So how would you trigger it from a script dialplan? The only time it
> >>>>> seemed to work is when I did setVariable("api_after_bridge",
> >>>>> "sched_api blah blah blah");
> >>>>> but then it gets executed after the channel's been teared down. I
> >>>>> thought api_after_bridge means right after the call gets connected.
> >>>>>
> >>>>> I need something to execute an api command right before or right
> >>>>> after the call gets bridged.
> >>>>>
> >>>>> api_after_bridge is a channel variable, so using setVariable works
> >>>>> just
> >>>>
> >>>> fine. If you need to sched_api is an API only. Check these out:
> >>>> http://wiki.freeswitch.org/wiki/Mod_commands#Misc._Commands
> >>>>
> >>>> So you need an API object in order to use it. I don't know the syntax
> >>>> for creating an api obj in Java but in Lua it goes like this:
> >>>> api = freeswitch.API();
> >>>> res = api:execute("sched_api","+300 none my_api my_api_args")
> >>>>
> >>>> Remember, if the method you are using isn't found in the dial plan
> >>>> tools then it isn't a dial plan application. Make sure it's on the
> >>>> list: http://wiki.freeswitch.org/wiki/Mod_dptools
> >>>>
> >>>> On the other hand, API commands are listed here:
> >>>> http://wiki.freeswitch.org/wiki/Mod_commands
> >>>>
> >>>> dptools require a session object, api commands require an api
> >>>> object...
> >>>>
> >>>> -MC
> >>>>
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>
> >>> --
> >>> Henry Huang
> >>> UniC Solution - Communication Unified
> >>> VoIP & Open Source software Consultant
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> >>>s http://www.freeswitch.org
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessa
> >>le at gmail.com> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> iax:guest at conference.freeswitch.org/888
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at co
> >>nference.freeswitch.org> pstn:213-799-1400
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > --
> > Henry Huang
> > UniC Solution - Communication Unified
> > VoIP & Open Source software Consultant
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org





From rob4manhere at gmail.com  Fri Oct 23 11:00:12 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Fri, 23 Oct 2009 13:00:12 -0500
Subject: [Freeswitch-users] Setting up Conference with Moderator
In-Reply-To: <71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local>
	<3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local>
	<E97CD3D0-52FF-44D1-BB89-23816045CE21@gmail.com>
	<71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>
Message-ID: <89C5467E-A7BC-4DA6-94C2-51FAF5793E68@gmail.com>

Right- this was an example without using the built-in pin but instead  
using IVR.


On Oct 23, 2009, at 12:43 PM, Brian West wrote:

> Well first off you're not defining a pine here...
>
> confname at profilename+flags{mute|deaf|waste|moderator}+[conference pin
> number]
>
> That might be why its not asking for a pin.
>
> /b
>
> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:
>
>>      <entry action="menu-exec-app" digits="1" param="conference
>> 123456 at default+flags{}" />
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From kristian.kielhofner at gmail.com  Fri Oct 23 11:14:07 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 23 Oct 2009 14:14:07 -0400
Subject: [Freeswitch-users] Issues with SIP+TCP?
Message-ID: <2d9149cd0910231114p7dd3c0cfo10ad30003929e4f8@mail.gmail.com>

Hello everyone,

  I'm having some issues with SIP and TCP.  I've used it before with
success but I'm seeing some strange behavior...

 Level 7 debugs with siptrace on both profiles.  UDP invite from
softphone comes in on port 5062, it's supposed to bridge to
10.70.0.62.  When configured to use UDP FS sends an INVITE (nothing
currently answers) while TCP doesn't send anything (confirmed with
siptrace and packet sniffer).  I confirmed this behavior with a
gateway configured for TCP and appending ;transport=tcp to a bridge
line.

  This is trunk rev 15211 on an Intel Mac running Snow Leopard.  I've
also confirmed this behavior on an Intel Linux machine running Ubuntu
(not sure of version ATM).

TCP:

http://pastebin.freeswitch.org/10825

UDP:

http://pastebin.freeswitch.org/10826

dialplan (UDP):

    <extension name="smhpbx">
      <condition field="destination_number" expression="^(7887)$">
        <action application="set" data="call_timeout=60"/>
        <action application="set" data="effective_caller_id_name=Voalte Test"/>
        <action application="set"
data="effective_caller_id_number=19412848354"/>
        <action application="bridge" data="sofia/avaya/7887 at 10.70.0.62"/>
      </condition>
    </extension>

dialplan (TCP):

    <extension name="smhpbx">
      <condition field="destination_number" expression="^(7887)$">
        <action application="set" data="call_timeout=60"/>
        <action application="set" data="effective_caller_id_name=Voalte Test"/>
        <action application="set"
data="effective_caller_id_number=19412848354"/>
        <action application="bridge"
data="sofia/avaya/7887 at 10.70.0.62;transport=tcp"/>
      </condition>
    </extension>

  Any thoughts?

Thanks!

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From bottleman at icf.org.ru  Fri Oct 23 11:17:05 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Fri, 23 Oct 2009 22:17:05 +0400 (MSD)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910230926y6c3e7ee3i9bb2182da1ba8757@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910230452s6b4a179aqa5bc41dc34efc403@mail.gmail.com>
	<Pine.LNX.4.64.0910231610220.21492@srv.icf.org.ru>
	<65d96fc80910230538w7105be6fve14956ab3b247bf6@mail.gmail.com>
	<Pine.LNX.4.64.0910231646210.29582@srv.icf.org.ru>
	<65d96fc80910230614m2b2504d0r92006b78d267a399@mail.gmail.com>
	<65d96fc80910230757h4595f396o2dc1a1bb781d3a8a@mail.gmail.com>
	<Pine.LNX.4.64.0910231903510.17804@srv.icf.org.ru>
	<191c3a030910230837x78ea6093i387617a1adfdf958@mail.gmail.com>
	<Pine.LNX.4.64.0910231943250.22146@srv.icf.org.ru>
	<65d96fc80910230926y6c3e7ee3i9bb2182da1ba8757@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910232208030.29940@srv.icf.org.ru>

On 2009-10-23 18:26 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:

TC>2009/10/23 Georgiewskiy Yuriy <bottleman at icf.org.ru>
TC>
TC>> On 2009-10-23 10:37 -0500, Anthony Minessale wrote
TC>> freeswitch-users at lists.f...:
TC>>
TC>> i have no way to install trunk at this time, i will go out of hospital
TC>> about one week later, after
TC>> this i will can try it on trunk.
TC>>
TC>> AM>if you were on trunk that line of code would be gone.
TC>> AM>you really can't do development on 1.0.4 its 6 months old and it will
TC>> cause
TC>> AM>you more trouble than you think when you eventually upgrade if you do
TC>> not do
TC>>
TC>
TC>uh... you are still in there ... damn sorry to hear that.
TC>I know what the feeling is... i was in the same position a couple of months
TC>ago... anyhow at least i had plenty of time to do my private research.. hope
TC>it is the case with you as well.
TC>
TC>so, if you have that time i can offer you my development server with trunk
TC>installed and all the hands on.
TC>
TC>
TC>let me know...
TC>
TC>BTW: it really doesn't have sense to develop on 1.0.4 ... the proof of
TC>concept was done. I'm able to place calls in both directions so, lets move
TC>to trunk now.

i have my own voip infrastructure on my work, and it's better for me to use it for tests, 
and use my personal on work too, i have problem with various hardware and call making, 
there is i have dialup connection and all what i can - make one call whit only g729 codec, 
or receive one call, no more, it's is a bandwich limitation. from hardware i have only ip 
phone artdio ipf2000 and one addpac gateway. also ssh sesssion is very slow.

P.S. people from russian community report what current version of module work fine on fs 
trunk version.

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From rob4manhere at gmail.com  Fri Oct 23 11:22:53 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Fri, 23 Oct 2009 13:22:53 -0500
Subject: [Freeswitch-users] Setting up Conference with Moderator
In-Reply-To: <71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local>
	<3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local>
	<E97CD3D0-52FF-44D1-BB89-23816045CE21@gmail.com>
	<71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>
Message-ID: <114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com>

I just re-tested with the pin in my dial plan:

   <action application="conference" data="conference 123456 at default 
+flags{}+1234" />

And it doesn't challenge me for the pin.  I just drop right in.  I  
figured this is how it was intended, since the wiki says the pin is  
set initially and only challenged in later attempts [by future callers]:

"The first time a conference name (confname) is used, it will be  
created on demand, and the pin will be set to what ever is specified  
at that time: the pin in the data string if specified, or if not, the  
"pin" setting in the conference profile, and if that is also  
unspecified, then there is no pin protection. Any later attempt to  
join the conference must specify the same pin number, if one existed  
when it was created. "


The wiki also says that the wait-mod might be  "used in conjunction  
with an IVR where the moderators are authenticated with an extra pass- 
code", which is what I did.  I guess that's why I didn't understand  
the point of the +pin.

I'm sure there's a scenario where its used and useful, the wiki just  
doesn't explain it.

Rob

On Oct 23, 2009, at 12:43 PM, Brian West wrote:

> Well first off you're not defining a pine here...
>
> confname at profilename+flags{mute|deaf|waste|moderator}+[conference pin
> number]
>
> That might be why its not asking for a pin.
>
> /b
>
> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:
>
>>      <entry action="menu-exec-app" digits="1" param="conference
>> 123456 at default+flags{}" />
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From msc at freeswitch.org  Fri Oct 23 12:00:28 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 23 Oct 2009 12:00:28 -0700
Subject: [Freeswitch-users] "Can not record session. Media not enabled
	on channel."
In-Reply-To: <1256283391890-3877285.post@n2.nabble.com>
References: <1256034659005-3857858.post@n2.nabble.com>
	<87f2f3b90910201746h43a31b32m5c163fd98373f915@mail.gmail.com>
	<1256283391890-3877285.post@n2.nabble.com>
Message-ID: <87f2f3b90910231200v71e72661id4cc9f2b7fba5610@mail.gmail.com>

On Fri, Oct 23, 2009 at 12:36 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> The dialplan is very simple:
>
>        <extension name="Recording test">
>                <condition field="destination_number"
> expression="^111111\d*$">
>                        <action application="sleep" data="3000" />
>                        <action application="answer"/>
>                        <action application="say" data="en name_spelled
> iterated
> ${destination_number}"/>
>                        <action application="sleep" data="1000" />
>                        <action application="playback"
> data="local_stream://my_music"/>
>                </condition>
>        </extension>
>
> Before debugging I have another question: I start recording in event
> handler
> for ChannelAnswer event. Is it possible that it's too soon to start
> recording? Maybe I should start recording in some other event?
>

That would be an odd scenario but maybe. It would be best if you could catch
it in the act so that we could see exactly what is happening. The other
thing you could do is deliberately start recording on the channel prior to
answering and see if you always get the error. In other words, try to make
it fail under a certain set of circumstances to see if your theory is
correct.
-MC
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From msc at freeswitch.org  Fri Oct 23 12:01:44 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 23 Oct 2009 12:01:44 -0700
Subject: [Freeswitch-users] CPU Usage Question
In-Reply-To: <c96b95650910231016me63190fr4641ed7737637a85@mail.gmail.com>
References: <4add4a7a.1408c00a.4283.ffffc11f@mx.google.com>
	<c96b95650910231016me63190fr4641ed7737637a85@mail.gmail.com>
Message-ID: <87f2f3b90910231201x11644dfal60c4a8556404abe@mail.gmail.com>

On Fri, Oct 23, 2009 at 10:16 AM, Saeed Ahmad <saeedahmad1981 at gmail.com>wrote:

> I think we can't create two profiles per single IP. So what is the
> workaround to solve this issue?
>
> SAT
>
>
Two profiles can share the same IP address, just not the same IP and port.
The example internal and external profiles both use the same IP address but
they use different ports (5060 and 5080).
-MC
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From freeswitch.noob at gmail.com  Fri Oct 23 12:34:09 2009
From: freeswitch.noob at gmail.com (freeswitch noob)
Date: Fri, 23 Oct 2009 14:34:09 -0500
Subject: [Freeswitch-users] Hostname
Message-ID: <7f783b530910231234u5d0e2fedxea4884d9bcddc9d3@mail.gmail.com>

I have mod xml_curl installed and I am getting the following passed to my
script.

[hostname] => myhost.local
[section] => dialplan


I also have multiple versions of FS running on the same box.  Is there a way
to have each FS instance on my box have a unique hostname ?

Thanks in advance.
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From kristian.kielhofner at gmail.com  Fri Oct 23 12:46:11 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 23 Oct 2009 15:46:11 -0400
Subject: [Freeswitch-users] Hostname
In-Reply-To: <7f783b530910231234u5d0e2fedxea4884d9bcddc9d3@mail.gmail.com>
References: <7f783b530910231234u5d0e2fedxea4884d9bcddc9d3@mail.gmail.com>
Message-ID: <2d9149cd0910231246t58d2bbd3k3466d9c0fca18000@mail.gmail.com>

Can't you use different contexts or something else to tell them apart?

On Fri, Oct 23, 2009 at 3:34 PM, freeswitch noob
<freeswitch.noob at gmail.com> wrote:
> I have mod xml_curl installed and I am getting the following passed to my
> script.
>
> [hostname] => myhost.local
> [section] => dialplan
>
>
> I also have multiple versions of FS running on the same box.? Is there a way
> to have each FS instance on my box have a unique hostname ?
>
> Thanks in advance.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From chris at cloudtel.com  Fri Oct 23 12:58:09 2009
From: chris at cloudtel.com (Chris Burns)
Date: Fri, 23 Oct 2009 15:58:09 -0400
Subject: [Freeswitch-users] Hostname
In-Reply-To: <2d9149cd0910231246t58d2bbd3k3466d9c0fca18000@mail.gmail.com>
References: <7f783b530910231234u5d0e2fedxea4884d9bcddc9d3@mail.gmail.com>
	<2d9149cd0910231246t58d2bbd3k3466d9c0fca18000@mail.gmail.com>
Message-ID: <200910231558.09699.chris@cloudtel.com>

one real quick way would be put different GET var in each server's binding

On October 23, 2009 03:46:11 pm Kristian Kielhofner wrote:
> Can't you use different contexts or something else to tell them apart?
>
> On Fri, Oct 23, 2009 at 3:34 PM, freeswitch noob
>
> <freeswitch.noob at gmail.com> wrote:
> > I have mod xml_curl installed and I am getting the following passed to my
> > script.
> >
> > [hostname] => myhost.local
> > [section] => dialplan
> >
> >
> > I also have multiple versions of FS running on the same box.? Is there a
> > way to have each FS instance on my box have a unique hostname ?
> >
> > Thanks in advance.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org





From freeswitch.noob at gmail.com  Fri Oct 23 13:42:48 2009
From: freeswitch.noob at gmail.com (freeswitch noob)
Date: Fri, 23 Oct 2009 15:42:48 -0500
Subject: [Freeswitch-users] Hostname
In-Reply-To: <200910231558.09699.chris@cloudtel.com>
References: <7f783b530910231234u5d0e2fedxea4884d9bcddc9d3@mail.gmail.com>
	<2d9149cd0910231246t58d2bbd3k3466d9c0fca18000@mail.gmail.com>
	<200910231558.09699.chris@cloudtel.com>
Message-ID: <7f783b530910231342t5393763aueff6bbc679b3c55@mail.gmail.com>

Yeah, I was just trying to make it easier on myself.  I have scripts from a
friend that parse xml_curl requests based on the hostname, I was hoping to
not have to re-write them to read something else from the post that FS makes
from xml_curl.  But from what it sounds like I will have to.


On Fri, Oct 23, 2009 at 2:58 PM, Chris Burns <chris at cloudtel.com> wrote:

> one real quick way would be put different GET var in each server's binding
>
> On October 23, 2009 03:46:11 pm Kristian Kielhofner wrote:
> > Can't you use different contexts or something else to tell them apart?
> >
> > On Fri, Oct 23, 2009 at 3:34 PM, freeswitch noob
> >
> > <freeswitch.noob at gmail.com> wrote:
> > > I have mod xml_curl installed and I am getting the following passed to
> my
> > > script.
> > >
> > > [hostname] => myhost.local
> > > [section] => dialplan
> > >
> > >
> > > I also have multiple versions of FS running on the same box.  Is there
> a
> > > way to have each FS instance on my box have a unique hostname ?
> > >
> > > Thanks in advance.
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From freeswitch-users-list at metik.com  Fri Oct 23 14:37:35 2009
From: freeswitch-users-list at metik.com (Metik)
Date: Fri, 23 Oct 2009 17:37:35 -0400
Subject: [Freeswitch-users] Hostname
In-Reply-To: <7f783b530910231342t5393763aueff6bbc679b3c55@mail.gmail.com>
References: <7f783b530910231234u5d0e2fedxea4884d9bcddc9d3@mail.gmail.com><2d9149cd0910231246t58d2bbd3k3466d9c0fca18000@mail.gmail.com><200910231558.09699.chris@cloudtel.com>
	<7f783b530910231342t5393763aueff6bbc679b3c55@mail.gmail.com>
Message-ID: <65286EEC1128407F82A019CF0EFB9C38@chaos>

Why not simply overwrite the value of the variable used throughout the script... 

-- xml_curl.conf --
...
      <param name="gateway-url" value="http://localhost/index.php?xhostname=myhost" bindings="dialplan"/>
...

-- index.php --

<?

$_REQUEST['hostname'] = "$xhostname";

...
  ----- Original Message ----- 
  From: freeswitch noob 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, October 23, 2009 4:42 PM
  Subject: Re: [Freeswitch-users] Hostname


  Yeah, I was just trying to make it easier on myself.  I have scripts from a friend that parse xml_curl requests based on the hostname, I was hoping to not have to re-write them to read something else from the post that FS makes from xml_curl.  But from what it sounds like I will have to.



  On Fri, Oct 23, 2009 at 2:58 PM, Chris Burns <chris at cloudtel.com> wrote:

    one real quick way would be put different GET var in each server's binding


    On October 23, 2009 03:46:11 pm Kristian Kielhofner wrote:
    > Can't you use different contexts or something else to tell them apart?
    >
    > On Fri, Oct 23, 2009 at 3:34 PM, freeswitch noob
    >
    > <freeswitch.noob at gmail.com> wrote:
    > > I have mod xml_curl installed and I am getting the following passed to my
    > > script.
    > >
    > > [hostname] => myhost.local
    > > [section] => dialplan
    > >
    > >
    > > I also have multiple versions of FS running on the same box.  Is there a
    > > way to have each FS instance on my box have a unique hostname ?
    > >
    > > Thanks in advance.
    > >
    > > _______________________________________________
    > > FreeSWITCH-users mailing list
    > > FreeSWITCH-users at lists.freeswitch.org
    > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    > > http://www.freeswitch.org



    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





------------------------------------------------------------------------------


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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From freeswitch-users-list at metik.com  Fri Oct 23 14:48:05 2009
From: freeswitch-users-list at metik.com (Metik)
Date: Fri, 23 Oct 2009 17:48:05 -0400
Subject: [Freeswitch-users] Hostname
Message-ID: <AEDF2544E2D24EC5ACEB0361357C16EE@chaos>

Please note that this would essentially be taking Chris' suggestion a little further but the effort involved would be minimal.  
  ----- Original Message ----- 
  From: Metik 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, October 23, 2009 5:37 PM
  Subject: Re: [Freeswitch-users] Hostname


  Why not simply overwrite the value of the variable used throughout the script... 

  -- xml_curl.conf --
  ...
        <param name="gateway-url" value="http://localhost/index.php?xhostname=myhost" bindings="dialplan"/>
  ...

  -- index.php --

  <?

  $_REQUEST['hostname'] = "$xhostname";

  ...
    ----- Original Message ----- 
    From: freeswitch noob 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Friday, October 23, 2009 4:42 PM
    Subject: Re: [Freeswitch-users] Hostname


    Yeah, I was just trying to make it easier on myself.  I have scripts from a friend that parse xml_curl requests based on the hostname, I was hoping to not have to re-write them to read something else from the post that FS makes from xml_curl.  But from what it sounds like I will have to.



    On Fri, Oct 23, 2009 at 2:58 PM, Chris Burns <chris at cloudtel.com> wrote:

      one real quick way would be put different GET var in each server's binding


      On October 23, 2009 03:46:11 pm Kristian Kielhofner wrote:
      > Can't you use different contexts or something else to tell them apart?
      >
      > On Fri, Oct 23, 2009 at 3:34 PM, freeswitch noob
      >
      > <freeswitch.noob at gmail.com> wrote:
      > > I have mod xml_curl installed and I am getting the following passed to my
      > > script.
      > >
      > > [hostname] => myhost.local
      > > [section] => dialplan
      > >
      > >
      > > I also have multiple versions of FS running on the same box.  Is there a
      > > way to have each FS instance on my box have a unique hostname ?
      > >
      > > Thanks in advance.
      > >
      > > _______________________________________________
      > > FreeSWITCH-users mailing list
      > > FreeSWITCH-users at lists.freeswitch.org
      > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      > > http://www.freeswitch.org



      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org





----------------------------------------------------------------------------


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    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From freeswitch-users-list at metik.com  Fri Oct 23 15:18:41 2009
From: freeswitch-users-list at metik.com (Metik)
Date: Fri, 23 Oct 2009 18:18:41 -0400
Subject: [Freeswitch-users] Hostname
In-Reply-To: <AEDF2544E2D24EC5ACEB0361357C16EE@chaos>
References: <AEDF2544E2D24EC5ACEB0361357C16EE@chaos>
Message-ID: <A52BE10FDCCE4267966353EA5C5F02D7@chaos>

Forget it--this will not work because FS uses a POST (vs GET). Most likely it would attempt to actually POST to a file named "index.php?xhostname=myhost".

However, if there was some way to add an arbitrary POST variable to the HTTP transaction, it would work. I am sure the FS developers have already thought of this and its a case of RTFM.

-metik
  ----- Original Message ----- 
  From: Metik 
  To: Metik ; freeswitch-users at lists.freeswitch.org 
  Sent: Friday, October 23, 2009 5:48 PM
  Subject: Re: [Freeswitch-users] Hostname


  Please note that this would essentially be taking Chris' suggestion a little further but the effort involved would be minimal.  
    ----- Original Message ----- 
    From: Metik 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Friday, October 23, 2009 5:37 PM
    Subject: Re: [Freeswitch-users] Hostname


    Why not simply overwrite the value of the variable used throughout the script... 

    -- xml_curl.conf --
    ...
          <param name="gateway-url" value="http://localhost/index.php?xhostname=myhost" bindings="dialplan"/>
    ...

    -- index.php --

    <?

    $_REQUEST['hostname'] = "$xhostname";

    ...
      ----- Original Message ----- 
      From: freeswitch noob 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Friday, October 23, 2009 4:42 PM
      Subject: Re: [Freeswitch-users] Hostname


      Yeah, I was just trying to make it easier on myself.  I have scripts from a friend that parse xml_curl requests based on the hostname, I was hoping to not have to re-write them to read something else from the post that FS makes from xml_curl.  But from what it sounds like I will have to.



      On Fri, Oct 23, 2009 at 2:58 PM, Chris Burns <chris at cloudtel.com> wrote:

        one real quick way would be put different GET var in each server's binding


        On October 23, 2009 03:46:11 pm Kristian Kielhofner wrote:
        > Can't you use different contexts or something else to tell them apart?
        >
        > On Fri, Oct 23, 2009 at 3:34 PM, freeswitch noob
        >
        > <freeswitch.noob at gmail.com> wrote:
        > > I have mod xml_curl installed and I am getting the following passed to my
        > > script.
        > >
        > > [hostname] => myhost.local
        > > [section] => dialplan
        > >
        > >
        > > I also have multiple versions of FS running on the same box.  Is there a
        > > way to have each FS instance on my box have a unique hostname ?
        > >
        > > Thanks in advance.
        > >
        > > _______________________________________________
        > > FreeSWITCH-users mailing list
        > > FreeSWITCH-users at lists.freeswitch.org
        > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        > > http://www.freeswitch.org



        _______________________________________________
        FreeSWITCH-users mailing list
        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org





--------------------------------------------------------------------------


      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From chris at cloudtel.com  Fri Oct 23 16:23:03 2009
From: chris at cloudtel.com (Chris Burns)
Date: Fri, 23 Oct 2009 19:23:03 -0400
Subject: [Freeswitch-users] Hostname
In-Reply-To: <A52BE10FDCCE4267966353EA5C5F02D7@chaos>
References: <AEDF2544E2D24EC5ACEB0361357C16EE@chaos>
	<A52BE10FDCCE4267966353EA5C5F02D7@chaos>
Message-ID: <200910231923.03648.chris@cloudtel.com>

Fake the hostname in the query string:
http://test/dir.php?hostname=mmmpotatoes

Add this to the top of your buddy's script (if its even PHP):
$hostname = $_POST['hostname'] = $_REQUEST['hostname'] = $_GET['hostname'];


On October 23, 2009 06:18:41 pm Metik wrote:
> Forget it--this will not work because FS uses a POST (vs GET). Most likely
> it would attempt to actually POST to a file named
> "index.php?xhostname=myhost".
>
> However, if there was some way to add an arbitrary POST variable to the
> HTTP transaction, it would work. I am sure the FS developers have already
> thought of this and its a case of RTFM.
>
> -metik
>   ----- Original Message -----
>   From: Metik
>   To: Metik ; freeswitch-users at lists.freeswitch.org
>   Sent: Friday, October 23, 2009 5:48 PM
>   Subject: Re: [Freeswitch-users] Hostname
>
>
>   Please note that this would essentially be taking Chris' suggestion a
> little further but the effort involved would be minimal. ----- Original
> Message -----
>     From: Metik
>     To: freeswitch-users at lists.freeswitch.org
>     Sent: Friday, October 23, 2009 5:37 PM
>     Subject: Re: [Freeswitch-users] Hostname
>
>
>     Why not simply overwrite the value of the variable used throughout the
> script...
>
>     -- xml_curl.conf --
>     ...
>           <param name="gateway-url"
> value="http://localhost/index.php?xhostname=myhost" bindings="dialplan"/>
> ...
>
>     -- index.php --
>
>     <?
>
>     $_REQUEST['hostname'] = "$xhostname";
>
>     ...
>       ----- Original Message -----
>       From: freeswitch noob
>       To: freeswitch-users at lists.freeswitch.org
>       Sent: Friday, October 23, 2009 4:42 PM
>       Subject: Re: [Freeswitch-users] Hostname
>
>
>       Yeah, I was just trying to make it easier on myself.  I have scripts
> from a friend that parse xml_curl requests based on the hostname, I was
> hoping to not have to re-write them to read something else from the post
> that FS makes from xml_curl.  But from what it sounds like I will have to.
>
>
>
>       On Fri, Oct 23, 2009 at 2:58 PM, Chris Burns <chris at cloudtel.com>
> wrote:
>
>         one real quick way would be put different GET var in each server's
> binding
>
>         On October 23, 2009 03:46:11 pm Kristian Kielhofner wrote:
>         > Can't you use different contexts or something else to tell them
>         > apart?
>         >
>         > On Fri, Oct 23, 2009 at 3:34 PM, freeswitch noob
>         >
>         > <freeswitch.noob at gmail.com> wrote:
>         > > I have mod xml_curl installed and I am getting the following
>         > > passed to my script.
>         > >
>         > > [hostname] => myhost.local
>         > > [section] => dialplan
>         > >
>         > >
>         > > I also have multiple versions of FS running on the same box. 
>         > > Is there a way to have each FS instance on my box have a unique
>         > > hostname ?
>         > >
>         > > Thanks in advance.
>         > >
>         > > _______________________________________________
>         > > FreeSWITCH-users mailing list
>         > > FreeSWITCH-users at lists.freeswitch.org
>         > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswi
>         > >tch-users http://www.freeswitch.org
>
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>        
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
> --------------------------------------------------------------------------
>
>
>       _______________________________________________
>       FreeSWITCH-users mailing list
>       FreeSWITCH-users at lists.freeswitch.org
>       http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>      
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From nagalenoj at gmail.com  Fri Oct 23 23:11:36 2009
From: nagalenoj at gmail.com (Nagalenoj)
Date: Fri, 23 Oct 2009 23:11:36 -0700 (PDT)
Subject: [Freeswitch-users] UUID of the newly originated call?
In-Reply-To: <191c3a030910221012w125a15d4m24f5fbe72c8a859d@mail.gmail.com>
References: <fe67a3340910202306i55a0b9ceve92ab19e000edf11@mail.gmail.com>
	<191c3a030910211720l26bd0303i2cb11e9adb28037d@mail.gmail.com>
	<26006565.post@talk.nabble.com>
	<191c3a030910221012w125a15d4m24f5fbe72c8a859d@mail.gmail.com>
Message-ID: <26036438.post@talk.nabble.com>


It is working fine in the current version(15203) of SVN. Thanks.


Anthony Minessale-2 wrote:
> 
> please update to svn trunk with "make current" and try again.
> 
> 
> On Thu, Oct 22, 2009 at 4:08 AM, Nagalenoj <nagalenoj at gmail.com> wrote:
> 
>>
>> I'm using 'FreeSWITCH Version 1.0.trunk (15106)'.
>>
>>
>> Anthony Minessale-2 wrote:
>> >
>> > which revision of FS are you using?
>> >
>> > On Wed, Oct 21, 2009 at 1:06 AM, Nagalenoj H. <nagalenoj at gmail.com>
>> wrote:
>> >
>> >> I've tried with origination_uuid.
>> >>
>> >> First, I tried with SIP and my program executes successfully as what I
>> >> expected. This program initiates a new call when a call comes and let
>> the
>> >> new call to eavesdrop the landed call.
>> >> When, I experimented with PRI(openzap), I'm facing the following
>> error.
>> >> And, I am unable to make any calls(even from CLI). It is reporting the
>> >> same
>> >> error for the subsequent calls.
>> >> Error:        2009-10-21 11:08:31.661252 [ERR] mod_openzap.c:1760
>> Session
>> >> for channel 1:1 not found [UUID: 8364eb30-be03-11de-aa07-07e76b7651a9]
>> >>
>> >> Full log is in http://pastebin.freeswitch.org/10780
>> >> My script is here, http://pastebin.freeswitch.org/10781
>> >>
>> >> What is this error for and how to avoid this?
>> >>
>> >> Is there any other way to get the uuid of the originated call except
>> >> explicitly defining(origination_uuid).?!
>> >>
>> >> --
>> >> Regards,
>> >> Nagalenoj H.
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> <MSN%3Aanthony_minessale at hotmail.com><
>> MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
>> >
>> >
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
>> >
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> <sip%3A888 at conference.freeswitch.org><
>> sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
>> >
>> > iax:guest at conference.freeswitch.org/888
>> >
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
>> >
>> > pstn:213-799-1400
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> --
>> View this message in context:
>> http://www.nabble.com/Re%3A-UUID-of-the-newly-originated-call--tp25987024p26006565.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/Re%3A-UUID-of-the-newly-originated-call--tp25987024p26036438.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From tculjaga at gmail.com  Sat Oct 24 01:04:33 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Sat, 24 Oct 2009 10:04:33 +0200
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910232208030.29940@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910230538w7105be6fve14956ab3b247bf6@mail.gmail.com>
	<Pine.LNX.4.64.0910231646210.29582@srv.icf.org.ru>
	<65d96fc80910230614m2b2504d0r92006b78d267a399@mail.gmail.com>
	<65d96fc80910230757h4595f396o2dc1a1bb781d3a8a@mail.gmail.com>
	<Pine.LNX.4.64.0910231903510.17804@srv.icf.org.ru>
	<191c3a030910230837x78ea6093i387617a1adfdf958@mail.gmail.com>
	<Pine.LNX.4.64.0910231943250.22146@srv.icf.org.ru>
	<65d96fc80910230926y6c3e7ee3i9bb2182da1ba8757@mail.gmail.com>
	<Pine.LNX.4.64.0910232208030.29940@srv.icf.org.ru>
Message-ID: <65d96fc80910240104m458849c0xf7a17d3560496cbf@mail.gmail.com>

> TC>BTW: it really doesn't have sense to develop on 1.0.4 ... the proof of
> TC>concept was done. I'm able to place calls in both directions so, lets
> move
> TC>to trunk now.
>
> i have my own voip infrastructure on my work, and it's better for me to use
> it for tests,
> and use my personal on work too, i have problem with various hardware and
> call making,
> there is i have dialup connection and all what i can - make one call whit
> only g729 codec,
> or receive one call, no more, it's is a bandwich limitation. from hardware
> i have only ip
> phone artdio ipf2000 and one addpac gateway. also ssh sesssion is very
> slow.
>
>
I see... it is not the ideal development environment :)... Indeed, a dialup
is not a good thing... i guess your g729 is compressed to the maximum :)



> P.S. people from russian community report what current version of module
> work fine on fs
> trunk version.
>
>
that's strange that they report it working as "m_txAudioOpened" is never
gonna be ready in pre_answer :P... i had to comment it to make it working.

anyhow, i moved everything to trunk and will do some tests on Monday.

T
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From raffaele.p.guidi at gmail.com  Sat Oct 24 01:42:35 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Sat, 24 Oct 2009 10:42:35 +0200
Subject: [Freeswitch-users] Brazilians (Off-Topic)
In-Reply-To: <7A73AEA0-9138-4E02-897A-85159405A4EE@jerris.com>
References: <SNT102-W13E52679EA9B9D8606FDFDC1C40@phx.gbl>
	<86a32abc0910171513n35d7d0echff940a16685ae1a4@mail.gmail.com>
	<86a32abc0910171521n403094bfvf27aa7c2a1c1f798@mail.gmail.com>
	<86a32abc0910171535l606eb9c6id54f7b3b5ae1b670@mail.gmail.com>
	<86a32abc0910171545l17f3f539ldba432db550c3eaa@mail.gmail.com>
	<86a32abc0910171601u6ab06e42xcf303301c85c57b1@mail.gmail.com>
	<86a32abc0910171605y73ecb75cqfb13923ad19adc56@mail.gmail.com>
	<86a32abc0910171711je1e57bduf2bb672472365ce1@mail.gmail.com>
	<00E26046-328F-440A-81F4-671F197F114C@gmail.com>
	<7A73AEA0-9138-4E02-897A-85159405A4EE@jerris.com>
Message-ID: <b0a8792f0910240142j1807a9d9i5f8f9bf46b2bf090@mail.gmail.com>

+1 for Confluence

On Tue, Oct 20, 2009 at 00:55, Michael Jerris <mike at jerris.com> wrote:

> we do have a license for this, people didn't seem to like it last time
> we looked at it, I can't recall why.
>
> On Oct 19, 2009, at 4:24 PM, Roberto Martins wrote:
>
> > what about http://www.atlassian.com/software/confluence/ they give
> > free licenses to open source project, and FS is using JIRA.
> >
> > roberto
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From odermann at googlemail.com  Sat Oct 24 05:13:33 2009
From: odermann at googlemail.com (Dennis)
Date: Sat, 24 Oct 2009 14:13:33 +0200
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
	<191c3a030910131101p418e0961t373db68285248ff6@mail.gmail.com>
	<538F4CF43B294D0EACABA281E1913B4C@chaos>
	<65d96fc80910131224m73f35b87g355be818ad3c0caf@mail.gmail.com>
	<48B637E0506F4A809E4C0CAF39BD08BB@chaos>
	<65d96fc80910132348t202905fbub57cc4c814eb4e21@mail.gmail.com>
	<5e414ed0910140731w1c7ebedr150e69cda8073155@mail.gmail.com>
	<191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com>
	<5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com>
	<191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com>
Message-ID: <5e414ed0910240513q316905ai5cf8c2ef63b52f60@mail.gmail.com>

ok, as written, i come back after some tests with fs and a thomson cirpack.

it did not work - at least in our tests.

we are using socket outbound and when a call comes in, it starts the
socket of fs. the number may be 123456. fs sends the respond 484 and
our carrier receives this information. but fs ends the call with
hangup_cause = invalid_number_format.
the carrier has one more digit for the phone number and sends 1234567
and the above mentioned behavior repeats.

the behavior we want and expected is, that the call stays in the
socket after response 484, so that the carrier can send the 1234567
into the same socket.
the management, when fs should send response 484 and when fs should be
answered would be programmed by us.
it also important, that fs keeps the call in the socket, so we can
tell fs, to answer the call after x seconds anyway.

any ideas, what we could do?



From djbinter at yahoo.com  Sat Oct 24 10:35:14 2009
From: djbinter at yahoo.com (DJB)
Date: Sat, 24 Oct 2009 10:35:14 -0700 (PDT)
Subject: [Freeswitch-users] Core Dump question!
In-Reply-To: <86a32abc0910222215p6eb79f66lbbef2b245f4d17fd@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701E840EB@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221258o6936000dn6eb2e22189c45435@mail.gmail.com>
	<DF11CCC6-5C89-4489-908C-92A0FCC94AE4@jerris.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E841E8@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221725g6233ec20w49a73864400426d6@mail.gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E84200@EXMBXCLUS01.citservers.local>
	<86a32abc0910222004y46bf08b6jda75b6b8f1979420@mail.gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E84203@EXMBXCLUS01.citservers.local>
	<33c87fa30910222100rae92b90uf94d7ae58a5cdc94@mail.gmail.com>
	<86a32abc0910222215p6eb79f66lbbef2b245f4d17fd@mail.gmail.com>
Message-ID: <227986.79244.qm@web37507.mail.mud.yahoo.com>

I wonder if he run "make current" from his 1.0.4 source file, it should update to the current trunk, wasn't it?

Dorn.



________________________________
From: Diego Viola <diego.viola at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Thu, October 22, 2009 10:15:21 PM
Subject: Re: [Freeswitch-users] Core Dump question!

He's running 1.0.4, he needs to checkout from SVN trunk and build FS from scratch.

Diego


On Fri, Oct 23, 2009 at 4:00 AM, Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:

I think the (exported) means you don't have the latest svn, but probably the officially released build 1.0.4 that can be downloaded from the FS page.
>
>I think you should see something like (the latest trunk is 15203):
>
>freeswitch at internal> version
>
>FreeSWITCH Version 1.0.trunk (15126)
>
>I guess you need to checkout the latest FS trunk
>
>
>
>On Fri, Oct 23, 2009 at 2:25 PM, Ujjval Karihaloo <ujjval at simplesignal.com> wrote:
>
>>>
>>
>>
>>
>>
>>
>>
>>
>>
>>>>
>>>>
>>freeswitch at internal> version
>>FreeSWITCH Version 1.0.4 (exported)
>> 
>>freeswitch at internal>
>>>>
>> 
>>Ujjval Karihaloo
>>VP Voice Engineering
>>IP Phone: +13032428610
>>E-Fax: +17202391690
>> 
>>SimpleSignal Inc.
>>88 Inverness Circle East
>>Suite K105
>>Englewood, CO  80112
>> 
>>>>
>>From:freeswitch-users-bounces at lists.freeswitch.org >>[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Diego
>>Viola
>>Sent: Thursday, October 22, 2009 9:04 PM
>>
>>To: freeswitch-users at lists.freeswitch.org
>>Subject: Re: [Freeswitch-users] Core Dump question!
>>>>
>> 
>>Type "version" on the
>>CLI.
>>>>
>>On Fri, Oct 23, 2009 at 2:52 AM, Ujjval Karihaloo <ujjval at simplesignal.com> wrote:
>>>>
>>>>
>>How do I tell if it?s the latest?I
>>downloaded is yesterday..and installed it from freeswitch.org
>>>>
>> 
>>Ujjval Karihaloo
>>VP Voice Engineering
>>IP Phone: +13032428610
>>E-Fax: +17202391690
>> 
>>SimpleSignal Inc.
>>88 Inverness Circle East
>>Suite K105
>>Englewood, CO  80112
>> 
>>>>
>>From:freeswitch-users-bounces at lists.freeswitch.org >>[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf
>>Of Michael Collins
>>Sent: Thursday, October 22, 2009 6:26 PM
>>>>
>>
>>To: freeswitch-users at lists.freeswitch.org
>>Subject: Re: [Freeswitch-users] Core Dump question!
>> 
>>Yes, if
>>this is latest SVN (after a make current) then open a jira.
>>>>-MC
>>>>
>>>>
>>>>
>>On
>>Thu, Oct 22, 2009 at 5:04 PM, Ujjval Karihaloo <ujjval at simplesignal.com>
>>wrote:
>>>>
>>>>
>>I do have the core dump, should I open a
>>ticket.
>>I am running latest Freeswitch 1.0.4 and
>>had done a make current just before it happened.
>> 
>> 
>>
>>>>_______________________________________________
>>>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>> 
>>_______________________________________________
>>>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>
>_______________________________________________
>>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>



      
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From rob4manhere at gmail.com  Sat Oct 24 10:53:23 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Sat, 24 Oct 2009 12:53:23 -0500
Subject: [Freeswitch-users] Core Dump question!
In-Reply-To: <227986.79244.qm@web37507.mail.mud.yahoo.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701E840EB@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221258o6936000dn6eb2e22189c45435@mail.gmail.com>
	<DF11CCC6-5C89-4489-908C-92A0FCC94AE4@jerris.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E841E8@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221725g6233ec20w49a73864400426d6@mail.gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E84200@EXMBXCLUS01.citservers.local>
	<86a32abc0910222004y46bf08b6jda75b6b8f1979420@mail.gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E84203@EXMBXCLUS01.citservers.local>
	<33c87fa30910222100rae92b90uf94d7ae58a5cdc94@mail.gmail.com>
	<86a32abc0910222215p6eb79f66lbbef2b245f4d17fd@mail.gmail.com>
	<227986.79244.qm@web37507.mail.mud.yahoo.com>
Message-ID: <DCF3C3F0-99EC-4E50-8979-771F0083396F@gmail.com>

No- "make current" won't work because it will try to do an "svn  
update", which won't be applicable since its not a checked out working  
copy of the svn repository.  He'll have to checkout the trunk with:
svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch
from, http://wiki.freeswitch.org/wiki/Installation_Guide

Rob

On Oct 24, 2009, at 12:35 PM, DJB wrote:

> I wonder if he run "make current" from his 1.0.4 source file, it  
> should update to the current trunk, wasn't it?
>
> Dorn.
>
> From: Diego Viola <diego.viola at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Thu, October 22, 2009 10:15:21 PM
> Subject: Re: [Freeswitch-users] Core Dump question!
>
> He's running 1.0.4, he needs to checkout from SVN trunk and build FS  
> from scratch.
>
> Diego
>
> On Fri, Oct 23, 2009 at 4:00 AM, Mark Campbell-Smith <mcampbellsmith at gmail.com 
> > wrote:
> I think the (exported) means you don't have the latest svn, but  
> probably the officially released build 1.0.4 that can be downloaded  
> from the FS page.
>
> I think you should see something like (the latest trunk is 15203):
>
> freeswitch at internal> version
>
> FreeSWITCH Version 1.0.trunk (15126)
>
> I guess you need to checkout the latest FS trunk
>
>
> On Fri, Oct 23, 2009 at 2:25 PM, Ujjval Karihaloo <ujjval at simplesignal.com 
> > wrote:
> freeswitch at internal> version
>
> FreeSWITCH Version 1.0.4 (exported)
>
>
> freeswitch at internal>
>
>
> Ujjval Karihaloo
>
> VP Voice Engineering
>
> IP Phone: +13032428610
>
> E-Fax: +17202391690
>
>
> SimpleSignal Inc.
>
> 88 Inverness Circle East
>
> Suite K105
>
> Englewood, CO  80112
>
> <image001.jpg>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Diego Viola
> Sent: Thursday, October 22, 2009 9:04 PM
>
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Core Dump question!
>
> Type "version" on the CLI.
>
> On Fri, Oct 23, 2009 at 2:52 AM, Ujjval Karihaloo <ujjval at simplesignal.com 
> > wrote:
>
> How do I tell if it?s the latest?I downloaded is yesterday..and  
> installed it from freeswitch.org
>
>
> Ujjval Karihaloo
>
> VP Voice Engineering
>
> IP Phone: +13032428610
>
> E-Fax: +17202391690
>
>
> SimpleSignal Inc.
>
> 88 Inverness Circle East
>
> Suite K105
>
> Englewood, CO  80112
>
> <image001.jpg>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Collins
> Sent: Thursday, October 22, 2009 6:26 PM
>
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Core Dump question!
>
>
> Yes, if this is latest SVN (after a make current) then open a jira.
> -MC
>
> On Thu, Oct 22, 2009 at 5:04 PM, Ujjval Karihaloo <ujjval at simplesignal.com 
> > wrote:
>
> I do have the core dump, should I open a ticket.
>
> I am running latest Freeswitch 1.0.4 and had done a make current  
> just before it happened.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From diego.viola at gmail.com  Sat Oct 24 17:28:19 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 25 Oct 2009 00:28:19 +0000
Subject: [Freeswitch-users] Core Dump question!
In-Reply-To: <DCF3C3F0-99EC-4E50-8979-771F0083396F@gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701E840EB@EXMBXCLUS01.citservers.local>
	<3C04B27FC880044F8FCD735D0D952FF71701E841E8@EXMBXCLUS01.citservers.local>
	<87f2f3b90910221725g6233ec20w49a73864400426d6@mail.gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E84200@EXMBXCLUS01.citservers.local>
	<86a32abc0910222004y46bf08b6jda75b6b8f1979420@mail.gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701E84203@EXMBXCLUS01.citservers.local>
	<33c87fa30910222100rae92b90uf94d7ae58a5cdc94@mail.gmail.com>
	<86a32abc0910222215p6eb79f66lbbef2b245f4d17fd@mail.gmail.com>
	<227986.79244.qm@web37507.mail.mud.yahoo.com>
	<DCF3C3F0-99EC-4E50-8979-771F0083396F@gmail.com>
Message-ID: <86a32abc0910241728i23de0605h61d732fba6aeecd8@mail.gmail.com>

What Rob said.

On Sat, Oct 24, 2009 at 5:53 PM, Rob Forman <rob4manhere at gmail.com> wrote:

> No- "make current" won't work because it will try to do an "svn
> update", which won't be applicable since its not a checked out working
> copy of the svn repository.  He'll have to checkout the trunk with:
> svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch
> from, http://wiki.freeswitch.org/wiki/Installation_Guide
>
> Rob
>
> On Oct 24, 2009, at 12:35 PM, DJB wrote:
>
> > I wonder if he run "make current" from his 1.0.4 source file, it
> > should update to the current trunk, wasn't it?
> >
> > Dorn.
> >
> > From: Diego Viola <diego.viola at gmail.com>
> > To: freeswitch-users at lists.freeswitch.org
> > Sent: Thu, October 22, 2009 10:15:21 PM
> > Subject: Re: [Freeswitch-users] Core Dump question!
> >
> > He's running 1.0.4, he needs to checkout from SVN trunk and build FS
> > from scratch.
> >
> > Diego
> >
> > On Fri, Oct 23, 2009 at 4:00 AM, Mark Campbell-Smith <
> mcampbellsmith at gmail.com
> > > wrote:
> > I think the (exported) means you don't have the latest svn, but
> > probably the officially released build 1.0.4 that can be downloaded
> > from the FS page.
> >
> > I think you should see something like (the latest trunk is 15203):
> >
> > freeswitch at internal> version
> >
> > FreeSWITCH Version 1.0.trunk (15126)
> >
> > I guess you need to checkout the latest FS trunk
> >
> >
> > On Fri, Oct 23, 2009 at 2:25 PM, Ujjval Karihaloo <
> ujjval at simplesignal.com
> > > wrote:
> > freeswitch at internal> version
> >
> > FreeSWITCH Version 1.0.4 (exported)
> >
> >
> > freeswitch at internal>
> >
> >
> > Ujjval Karihaloo
> >
> > VP Voice Engineering
> >
> > IP Phone: +13032428610
> >
> > E-Fax: +17202391690
> >
> >
> > SimpleSignal Inc.
> >
> > 88 Inverness Circle East
> >
> > Suite K105
> >
> > Englewood, CO  80112
> >
> > <image001.jpg>
> >
> >
> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org
> > ] On Behalf Of Diego Viola
> > Sent: Thursday, October 22, 2009 9:04 PM
> >
> >
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] Core Dump question!
> >
> > Type "version" on the CLI.
> >
> > On Fri, Oct 23, 2009 at 2:52 AM, Ujjval Karihaloo <
> ujjval at simplesignal.com
> > > wrote:
> >
> > How do I tell if it?s the latest?I downloaded is yesterday..and
> > installed it from freeswitch.org
> >
> >
> > Ujjval Karihaloo
> >
> > VP Voice Engineering
> >
> > IP Phone: +13032428610
> >
> > E-Fax: +17202391690
> >
> >
> > SimpleSignal Inc.
> >
> > 88 Inverness Circle East
> >
> > Suite K105
> >
> > Englewood, CO  80112
> >
> > <image001.jpg>
> >
> >
> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org
> > ] On Behalf Of Michael Collins
> > Sent: Thursday, October 22, 2009 6:26 PM
> >
> >
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] Core Dump question!
> >
> >
> > Yes, if this is latest SVN (after a make current) then open a jira.
> > -MC
> >
> > On Thu, Oct 22, 2009 at 5:04 PM, Ujjval Karihaloo <
> ujjval at simplesignal.com
> > > wrote:
> >
> > I do have the core dump, should I open a ticket.
> >
> > I am running latest Freeswitch 1.0.4 and had done a make current
> > just before it happened.
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From djbinter at yahoo.com  Sat Oct 24 22:55:38 2009
From: djbinter at yahoo.com (DJB)
Date: Sat, 24 Oct 2009 22:55:38 -0700 (PDT)
Subject: [Freeswitch-users] SIP UPDATE Method
Message-ID: <528227.52311.qm@web37508.mail.mud.yahoo.com>

I am wondering why after I update to trunk-15225, the Allow: UPDATE method is no longer there.

User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15225
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE

Am I missing something here?

Thank you.


      



From juanbackson at gmail.com  Sun Oct 25 05:23:37 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Sun, 25 Oct 2009 20:23:37 +0800
Subject: [Freeswitch-users] sip chat question
Message-ID: <27c25bc40910250523g23d4bde6k30d6a88d75391785@mail.gmail.com>

Hi,

I am not able to send chat message from the console and the error I am
getting is a strange "Invalid Profile".

Does anyone know what is wrong?


freeswitch at server.com> chat sip|454047001854840 at 192.168.1.87|
454047001854840 at 192.168.1.87|testing
API CALL [chat(sip|454047001854840 at 192.168.1.87|454047001854840 at 192.168.1.87|testing)]
output:
Error! Message Not Sent
freeswitch at server.com> 2009-10-25 10:51:01.793313 [ERR] sofia_presence.c:93
Chat proto [dp]
from [454047001854840 at 192.168.1.87]
to [454047001854840 at 192.168.1.87]
testing
Invalid Profile 192.168.1.87

freeswitch at server.com>
freeswitch at server.com> sofia status
API CALL [sofia(status)] output:
                     Name          Type
Data     State
=================================================================================================
                 external       profile    sip:mod_sofia at 192.168.1.87:5080
RUNNING (0)
                      internal       profile
sip:mod_sofia at 192.168.1.87:5060     RUNNING (0)
=================================================================================================
2 profiles 0 aliases
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From pippyduck1127 at hotmail.com  Sun Oct 25 11:17:32 2009
From: pippyduck1127 at hotmail.com (Pajongjit Buntaokit)
Date: Sun, 25 Oct 2009 18:17:32 +0000
Subject: [Freeswitch-users] Address Rupa: Database for Audio Data
In-Reply-To: <b39273530910221231u93d3560nbb27036620c46e9b@mail.gmail.com>
References: <BAY119-W52BD912544FAA1DE624B88B3C70@phx.gbl>
	<b39273530910130857jb7dec2eidd71e950d3902b25@mail.gmail.com>
Message-ID: <BAY119-W472B92F30085695612C877B3BB0@phx.gbl>


Hi Rupa,

Thank you very much. It works now. 

----
PB

> Date: Thu, 22 Oct 2009 14:31:58 -0500
> From: rupa at rupa.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Address Rupa: Database for Audio Data
> 
> I use the dialplan app session_record:
> 
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session
> 
> I call that in the appropriate parts before bridging to call.  For
> incoming it is just before the bridge to user/username.  For outgoing
> it is before I bridge to the outgoing provider.
> 
> On Thu, Oct 22, 2009 at 11:30 AM, Pajongjit Buntaokit
> <pippyduck1127 at hotmail.com> wrote:
> > Hi Rupa,
> >
> > Thanks again for your advice.
> >
> > I have been searching for the method to record in the freeswitch
> > documentation but I'm still not sure which command and method to make the
> > record for every call automatically.
> >
> > Which command or method do you use?
> > And to make the recording start and stop automatically every time when the
> > calls is started and end, where should I insert this command?
> >
> > Did you use the Mod commands 'uuid_record'?
> > If so, where to place this commands?
> > Please show me some clues?
> >
> > Thank you very much!
> >
> > PB
> >
> >
> >
> >
> >> Date: Tue, 13 Oct 2009 09:57:05 -0600
> >> From: rupa at rupa.comajongjit Buntaokit
> >> To: freeswitch-users at lists.freeswitch.org
> >> Subject: Re: [Freeswitch-users] Database for Audio Data
> >>
> >> What I do is record all calls and store the call with the UUID as the
> >> filename. Then when the call is hung up a CDR entry is sent to my web
> >> server. This CDR contains callerid and other info I might want to
> >> query by. The service on the web server inserts appropriate record(s)
> >> into the database. The recordings are available to the webserver.
> >> When one clicks on the "listen" link, the web server serves up the
> >> recording by UUID in the recording directory. I have a process that
> >> periodically removes old recordings from that dir. I don't purge the
> >> CDRs, though that is certainly possible.
> >>
> >> On Tue, Oct 13, 2009 at 7:58 AM, Pajongjit Buntaokit
> >> <pippyduck1127 at hotmail.com> wrote:
> >> > Hi,
> >> >
> >> > Does anyone know whether FreeSWITCH has a function to automatically
> >> > record
> >> > every call as an audio file in a server
> >> > or forward them to be stored in a database with additional parameters
> >> > such
> >> > as caller ID, date, starting time and ending time?
> >> >
> >> > So that these recorded audio data can be queried and retrieved with the
> >> > caller ID, date and time.
> >> >
> >> > Any suggestion or guidance, please advise.
> >> >
> >> > Thank you very much!
> >> >
> >> > ________________________________
> >> > Hotmail: Trusted email with Microsoft?s powerful SPAM protection. Sign
> >> > up
> >> > now.
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> -Rupa
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > ________________________________
> > Windows 7: It works the way you want. Learn more.
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> 
> 
> -- 
> -Rupa
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
_________________________________________________________________
Windows 7: Simplify your PC. Learn more.
http://www.microsoft.com/Windows/windows-7/default.aspx?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_evergreen1:102009
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From ujjval at simplesignal.com  Sun Oct 25 13:15:51 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Sun, 25 Oct 2009 13:15:51 -0700
Subject: [Freeswitch-users] Setting up Conference with Moderator
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local>
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701B6DCDD@EXMBXCLUS01.citservers.local>

One more try for help for COnferencing using moderator,

Anyone successfully accomplished this? Help appreciated!
________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ujjval Karihaloo [ujjval at simplesignal.com]
Sent: Thursday, October 22, 2009 9:01 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Setting up Conference with Moderator

Hi

I have the Basic Conferencing working. Here is my Dial Plan.

 I want to be able to setup a Moderator PIN different from other participants, when I add the moderator flag it logs me in directly w/o asking for a PIN..
action application="conference" data="conference.conf at wideband+flags{moderator}+159753<mailto:conference.conf at wideband+flags%7bmoderator%7d+159753>"/>

DialPlan is below for the normal user, and it asks for the PIN with below settings.

       Ujj Inbound from SS - start
    -->
    <extension name="simplesignal">   <!-- your provider or any name you'd like to call it -->
     <condition field="destination_number" expression="xxxxxxxxxx">  <!-- your DID for this gateway-->
        <action application="conference" data="conference.conf at wideband+159753<mailto:conference.conf at wideband+159753>"/>
     </condition>
</extension>

    <!--
       Ujj Inbound from SS ? end




And I am using the existing conference.conf.xml file in the auto_loads directory.


Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690

SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO  80112
[cid:image001.jpg at 01CA535A.E43E9870]<http://www.simplesignal.com/>

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From red.rain.seven at gmail.com  Sun Oct 25 19:58:57 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Mon, 26 Oct 2009 10:58:57 +0800
Subject: [Freeswitch-users] sched_api doesn't get launched
In-Reply-To: <200910231347.43883.chris@cloudtel.com>
References: <59ad9ca10910130314g1bd9d995o1b05ed33ffbf385e@mail.gmail.com>
	<59ad9ca10910230154i6d41ddfci8957b152c1884800@mail.gmail.com>
	<191c3a030910230802m1cb847agfdaf6e436296e748@mail.gmail.com>
	<200910231347.43883.chris@cloudtel.com>
Message-ID: <59ad9ca10910251958k314d1145rbb816d508358e52f@mail.gmail.com>

Anthony:

You are da man. Thank you. It's very kind of you to release me from endless
trial :P

On Sat, Oct 24, 2009 at 1:47 AM, Chris Burns <chris at cloudtel.com> wrote:

> poor bbhenry :)
>
> Added in r15207 please test and update docu if necessary:
> http://wiki.freeswitch.org/wiki/Variable_api_on_answer
>
> On October 23, 2009 11:02:07 am Anthony Minessale wrote:
> > it's probably related to escaping the data.
> > I was sick of watching you suffer so i added api_on_answer variable to
> > trunk.
> >
> > On Fri, Oct 23, 2009 at 3:54 AM, Henry Huang
> <red.rain.seven at gmail.com>wrote:
> > > Thanks to c6burns on IRC channel for the tip to use execute_on_answer
> in
> > > combination with eval, and of course everyone here that pointing me to
> > > the right direction.
> > > I was able to execute sched_api with eval, but not with the combination
> > > of execute_on_answer. The argument just don't get parsed as the eval
> > > argument. Here is the code:
> > >
> > > setVariable("execute_on_answer", "eval sched_api(+%ld none
> uuid_displace
> > > <some uuid> start
> > > /usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20
> mux)");
> > >
> > >
> > > Here is the log:
> > > 2009-10-23 16:17:53.633189 [DEBUG] switch_scheduler.c:214 Added task 2
> > > sched_api_function (none) to run at 1256285888
> > > EXECUTE sofia/internal/1688004 at 192.168.1.66 *eval(+OK *Added: 2
> > > )
> > >
> > > 2009-10-23 16:18:04.52701 [DEBUG] switch_channel.c:1927 sofia/internal/
> > > 1688004 at 192.168.1.66 *execute on answer: eval(sched_api(+37 none
> > > uuid_displace 84d74f41-8668-4138-943f-f076e94046ad start
> > > /usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux))*
> > > EXECUTE sofia/internal/1688004 at 192.168.1.66 eval(sched_api(+37 none
> > > uuid_displace 84d74f41-8668-4138-943f-f076e94046ad start
> > > /usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux))
> > >
> > >
> > > The first part of the log is when I just do execute("eval",
> > > "${sched_api(+%ld none uuid_displace <some uuid> start
> > > /usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20
> mux)}");
> > > And I simple get a OK, and then later on the sched_api gets executed.
> > >
> > > However, on the excerpt of the second log. which I use the combination
> > > with execute_on_answer like specified in the yellow hight code above.
> It
> > > doesn't simply return a OK, instead it shows the whole blob of code.
> and
> > > does nothing later.
> > >
> > >
> > >
> > > On Thu, Oct 22, 2009 at 4:09 AM, Anthony Minessale <
> > >
> > > anthony.minessale at gmail.com> wrote:
> > >> Yes you need an API object as described in other email.
> > >>
> > >> Which line of code from java caused that segfault
> > >> It looks like a simple NULL string issue that we may want to hunt
> down.
> > >>
> > >> On Wed, Oct 21, 2009 at 4:44 AM, Henry Huang
> <red.rain.seven at gmail.com>wrote:
> > >>> I can't seem to find the right thing to use in mod_java to execute
> api
> > >>> commands, only api_after_bridge
> > >>>
> > >>> 2009-10-21 17:42:46.593094 [NOTICE] mod_sofia.c:1509 Pre-Answer
> > >>> sofia/internal/1688003 at 192.168.1.66!
> > >>> #
> > >>> # A fatal error has been detected by the Java Runtime Environment:
> > >>> #
> > >>> #  SIGSEGV (0xb) at pc=0x004e4480, pid=1927, tid=16116624
> > >>> #
> > >>> # JRE version: 6.0_16-b01
> > >>> # Java VM: Java HotSpot(TM) Client VM (14.2-b01 mixed mode linux-x86
> )
> > >>> # Problematic frame:
> > >>> # C  [libc.so.6+0x6f480]  strcpy+0x10
> > >>> #
> > >>> # An error report file with more information is saved as:
> > >>> # /usr/local/freeswitch/bin/hs_err_pid1927.log
> > >>> 2009-10-21 17:42:59.883729 [ERR] switch_core_session.c:1374 Invalid
> > >>> Application sched_api
> > >>> 2009-10-21 17:42:59.883729 [NOTICE] switch_core_session.c:1375 Hangup
> > >>> sofia/internal/1688003 at 192.168.1.66 [CS_EXECUTE]
> > >>> [DESTINATION_OUT_OF_ORDER]
> > >>> #
> > >>> # If you would like to submit a bug report, please visit:
> > >>> #   http://java.sun.com/webapps/bugreport/crash.jsp
> > >>> # The crash happened outside the Java Virtual Machine in native code.
> > >>> # See problematic frame for where to report the bug.
> > >>>
> > >>> On Sat, Oct 17, 2009 at 3:37 AM, Michael Collins
> <msc at freeswitch.org>wrote:
> > >>>> On Fri, Oct 16, 2009 at 11:53 AM, Henry Huang
> > >>>> <red.rain.seven at gmail.com
> > >>>>
> > >>>> > wrote:
> > >>>>>
> > >>>>> So how would you trigger it from a script dialplan? The only time
> it
> > >>>>> seemed to work is when I did setVariable("api_after_bridge",
> > >>>>> "sched_api blah blah blah");
> > >>>>> but then it gets executed after the channel's been teared down. I
> > >>>>> thought api_after_bridge means right after the call gets connected.
> > >>>>>
> > >>>>> I need something to execute an api command right before or right
> > >>>>> after the call gets bridged.
> > >>>>>
> > >>>>> api_after_bridge is a channel variable, so using setVariable works
> > >>>>> just
> > >>>>
> > >>>> fine. If you need to sched_api is an API only. Check these out:
> > >>>> http://wiki.freeswitch.org/wiki/Mod_commands#Misc._Commands
> > >>>>
> > >>>> So you need an API object in order to use it. I don't know the
> syntax
> > >>>> for creating an api obj in Java but in Lua it goes like this:
> > >>>> api = freeswitch.API();
> > >>>> res = api:execute("sched_api","+300 none my_api my_api_args")
> > >>>>
> > >>>> Remember, if the method you are using isn't found in the dial plan
> > >>>> tools then it isn't a dial plan application. Make sure it's on the
> > >>>> list: http://wiki.freeswitch.org/wiki/Mod_dptools
> > >>>>
> > >>>> On the other hand, API commands are listed here:
> > >>>> http://wiki.freeswitch.org/wiki/Mod_commands
> > >>>>
> > >>>> dptools require a session object, api commands require an api
> > >>>> object...
> > >>>>
> > >>>> -MC
> > >>>>
> > >>>>
> > >>>> _______________________________________________
> > >>>> FreeSWITCH-users mailing list
> > >>>> FreeSWITCH-users at lists.freeswitch.org
> > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >>>> UNSUBSCRIBE:
> > >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >>>> http://www.freeswitch.org
> > >>>
> > >>> --
> > >>> Henry Huang
> > >>> UniC Solution - Communication Unified
> > >>> VoIP & Open Source software Consultant
> > >>>
> > >>> _______________________________________________
> > >>> FreeSWITCH-users mailing list
> > >>> FreeSWITCH-users at lists.freeswitch.org
> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-user
> > >>>s http://www.freeswitch.org
> > >>
> > >> --
> > >> Anthony Minessale II
> > >>
> > >> FreeSWITCH http://www.freeswitch.org/
> > >> ClueCon http://www.cluecon.com/
> > >> Twitter: http://twitter.com/FreeSWITCH_wire
> > >>
> > >> AIM: anthm
> > >> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com><
> MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> <PAYPAL%3Aanthony.minessa
> > >>le at gmail.com> IRC: irc.freenode.net #freeswitch
> > >>
> > >> FreeSWITCH Developer Conference
> > >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org><
> sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> > >> iax:guest at conference.freeswitch.org/888
> > >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> <googletalk%3Aconf%2B888 at co
> > >>nference.freeswitch.org> pstn:213-799-1400
> > >>
> > >> _______________________________________________
> > >> FreeSWITCH-users mailing list
> > >> FreeSWITCH-users at lists.freeswitch.org
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >> http://www.freeswitch.org
> > >
> > > --
> > > Henry Huang
> > > UniC Solution - Communication Unified
> > > VoIP & Open Source software Consultant
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From ujjval at simplesignal.com  Sun Oct 25 22:05:02 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Sun, 25 Oct 2009 22:05:02 -0700
Subject: [Freeswitch-users] Downloaded tar vs latest SVN - 200 OK has more
	headers
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE2@EXMBXCLUS01.citservers.local>

Hi,

 I used the downlaoded TAR ball and my calls worked, however, when upgrading to the SVN release...my SBC is rejecting the 200 OK (when the FS answers the call - using Conferencing app)..

Here are teh bad and good 200 OK,,,,I see a lot of additional headers startin gwith X:FS , can I remove them?

Bad one
-----------------
   SIP/2.0 200 OK
   v: SIP/2.0/UDP x.x.x.210:5060;branch=z9hG4bK1662379c08490b63b0b9be6978d93ebe
   f: "SIMPLE SIGNAL  "<sip:2222220208 at x.x.x.20:5080;user=phone>;tag=945d1734da594444aac9d071edc798a7
   To: "trunkgroup user" <sip:1111114847 at simplesignal.net;ssig=33402ab16f12e359b1f871b80bf7ff1a>;tag=gX1Z34952QBpS
   i: 674babc2dbf2a90da97bd68642865eae-23f978 at x.x.x.210<mailto:674babc2dbf2a90da97bd68642865eae-23f978 at x.x.x.210>
   CSeq: 3240 INVITE
   Contact: <sip:1111114847 at x.x.x.20:5080;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15225
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 249
   X-FS-Display-Name: +11111114847
   X-FS-Display-Number: +11111114847
   X-FS-Support: update_display
   P-Asserted-Identity: "+11111114847" <+11111114847>
   v=0
   o=FreeSWITCH 1256498762 1256498763 IN IP4 x.x.x.20
   s=FreeSWITCH
   c=IN IP4 x.x.x.20
   t=0 0
   m=audio 29406 RTP/AVP 0 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20
------------------------------
Good One
-------------------------------
   SIP/2.0 200 OK
   v: SIP/2.0/UDP x.x.x.210:5060;branch=z9hG4bKffebde0efa06a6aa7772bae5ca085627
   f: "SIMPLE SIGNAL  "<sip:2222220208 at x.x.x.20:5080;user=phone>;tag=895fd35d4979e6e5b0cdcc795b8c8df2
   To: "trunkgroup user" <sip:1111114847 at simplesignal.net;ssig=33402ab16f12e359b1f871b80bf7ff1a>;tag=KBXKrBQU99c6D
   i: 28f0d47f1163e0d478247e6729762d98-23fedb at x.x.x.210<mailto:28f0d47f1163e0d478247e6729762d98-23fedb at x.x.x.210>
   CSeq: 3278 INVITE
   Contact: <sip:1111114847 at x.x.x.20:5080;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 249
   v=0
   o=FreeSWITCH 1256512215 1256512216 IN IP4 x.x.x.20
   s=FreeSWITCH
   c=IN IP4 x.x.x.20
   t=0 0
   m=audio 17376 RTP/AVP 0 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20

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From velu.technical at gmail.com  Mon Oct 26 00:20:05 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Mon, 26 Oct 2009 12:50:05 +0530
Subject: [Freeswitch-users] meaning of created_time channel variable.
Message-ID: <1452e2980910260020m624a6a1bk2c3a5c36cb1cb5cf@mail.gmail.com>

Dear All,
     What is the value of created_time channel variable? Is this epoch
seconds?

Thanks & Regards,
Velusamy.
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From maciej.aniserowicz at gmail.com  Mon Oct 26 00:21:17 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Mon, 26 Oct 2009 00:21:17 -0700 (PDT)
Subject: [Freeswitch-users] "Can not record session. Media not enabled
 on channel."
In-Reply-To: <87f2f3b90910231200v71e72661id4cc9f2b7fba5610@mail.gmail.com>
References: <1256034659005-3857858.post@n2.nabble.com>
	<87f2f3b90910201746h43a31b32m5c163fd98373f915@mail.gmail.com>
	<1256283391890-3877285.post@n2.nabble.com>
	<87f2f3b90910231200v71e72661id4cc9f2b7fba5610@mail.gmail.com>
Message-ID: <1256541677550-3890610.post@n2.nabble.com>


Yes, I can confirm - this exact error occurs each time when I start recording
before the call is answered (just after sending ORIGINATE command) - but I
think that's completely understandable that media is not ready for an
unanswered call.
But... is there any other event that guarantees media to be ready?




mercutioviz wrote:
> 
> On Fri, Oct 23, 2009 at 12:36 AM, Maciej Aniserowicz <
> maciej.aniserowicz at gmail.com> wrote:
> 
>>
>> The dialplan is very simple:
>>
>>        <extension name="Recording test">
>>                <condition field="destination_number"
>> expression="^111111\d*$">
>>                        <action application="sleep" data="3000" />
>>                        <action application="answer"/>
>>                        <action application="say" data="en name_spelled
>> iterated
>> ${destination_number}"/>
>>                        <action application="sleep" data="1000" />
>>                        <action application="playback"
>> data="local_stream://my_music"/>
>>                </condition>
>>        </extension>
>>
>> Before debugging I have another question: I start recording in event
>> handler
>> for ChannelAnswer event. Is it possible that it's too soon to start
>> recording? Maybe I should start recording in some other event?
>>
> 
> That would be an odd scenario but maybe. It would be best if you could
> catch
> it in the act so that we could see exactly what is happening. The other
> thing you could do is deliberately start recording on the channel prior to
> answering and see if you always get the error. In other words, try to make
> it fail under a certain set of circumstances to see if your theory is
> correct.
> -MC
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3890610.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From tculjaga at gmail.com  Mon Oct 26 03:16:20 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 26 Oct 2009 11:16:20 +0100
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <65d96fc80910240104m458849c0xf7a17d3560496cbf@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910231646210.29582@srv.icf.org.ru>
	<65d96fc80910230614m2b2504d0r92006b78d267a399@mail.gmail.com>
	<65d96fc80910230757h4595f396o2dc1a1bb781d3a8a@mail.gmail.com>
	<Pine.LNX.4.64.0910231903510.17804@srv.icf.org.ru>
	<191c3a030910230837x78ea6093i387617a1adfdf958@mail.gmail.com>
	<Pine.LNX.4.64.0910231943250.22146@srv.icf.org.ru>
	<65d96fc80910230926y6c3e7ee3i9bb2182da1ba8757@mail.gmail.com>
	<Pine.LNX.4.64.0910232208030.29940@srv.icf.org.ru>
	<65d96fc80910240104m458849c0xf7a17d3560496cbf@mail.gmail.com>
Message-ID: <65d96fc80910260316k3f040488wa029504bf972c56e@mail.gmail.com>

>
>
>
>> P.S. people from russian community report what current version of module
>> work fine on fs
>> trunk version.
>>
>>
> that's strange that they report it working as "m_txAudioOpened" is never
> gonna be ready in pre_answer :P... i had to comment it to make it working.
>
> anyhow, i moved everything to trunk and will do some tests on Monday.
>
> T
>


Hello Yuriy, I tried the trunk (FreeSWITCH Version 1.0.trunk (15216M)) and
i'm getting some nice coredumps...


FS crashes when placing outgoing calls.

    coredump on outbound call: FS log and backtrace
http://pastebin.freeswitch.org/10834


FS crashes on incoming calls.

    coredump on inbound call: FS log and backtrace
http://pastebin.freeswitch.org/10835


FS crashes when i try to load mod_h323. I need 3-4 attempts to load it
without a crash.

     coredump on mod_h323 load: FS log and backtrace:
http://pastebin.freeswitch.org/10833


FS crashes on shutdown procedure if mod_h323 was loaded previously.

    coredump on mod_h323 load: FS log and backtrace:
http://pastebin.freeswitch.org/10836



It is quite bad :)
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From helmut.kuper at ewetel.de  Mon Oct 26 04:00:14 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Mon, 26 Oct 2009 12:00:14 +0100
Subject: [Freeswitch-users] Qustion about INFO messages
	after	Connect/Answer
In-Reply-To: <191c3a030910230851i6e0b081ajaf6c294a7f9cd3a8@mail.gmail.com>
References: <4AD859A4.5000101@ewetel.de>
	<4ADC8B6E.7070404@ewetel.de>	<191c3a030910191307h31ef05c2wa271d54582126632@mail.gmail.com>	<87f2f3b90910191435s7528e256r85df5f12fc634f24@mail.gmail.com>	<4ADF483B.4050807@ewetel.de>	<87f2f3b90910211048s208daf1eyfae501091ab75530@mail.gmail.com>	<4AE07DF4.5040803@ewetel.de>	<085B77EE-DA3F-4ADD-86DC-E59D1B6553EA@freeswitch.org>	<191c3a030910221008n60341314hc7a7e6a55a50ef15@mail.gmail.com>	<4AE1951A.1030906@ewetel.de>
	<191c3a030910230851i6e0b081ajaf6c294a7f9cd3a8@mail.gmail.com>
Message-ID: <4AE5813E.2030506@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Anthony,

hm, no not really. There is no change in this behaviour in "FreeSWITCH
Version 1.0.trunk (15225M)"


There is still a caller name change in callee's display.

I'm not sure who is wrong here. Either FS or Snom ...


regards
Helmut


On 23.10.2009 17:51, Anthony Minessale wrote:
> should be even better in 15210
> 
> 
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u9RajDfF0rNdzkaUOVjULmk=
=697e
-----END PGP SIGNATURE-----



From dome at tel.co.th  Mon Oct 26 04:24:28 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Mon, 26 Oct 2009 18:24:28 +0700
Subject: [Freeswitch-users] mod_nibblebill and memory problem
Message-ID: <8ccbff060910260424y7fb9f6eakbb45900838e6fd6d@mail.gmail.com>

Dear All,
             I'm running mod_nibblebill for my prepaid solution. I
still have problem with memory. I have 4 GB RAM and runing debian
squeeze 64 bit and 200 calls concurrent
         Last time nibblebill running with 1 min heartbeat. when i
check memory by htop FS user memory 2% anf growth to 60-89% in 8-9 hr.
and then FS crash.
         Now i change hearbeat to 0 it's mean nibble update balance
when end of call. but everything are same FS start from 2% and growth
to 20% in 2 days.
        When i unload nibblebill FS running fine.
        My question is when concurrent calls drop to 1-2 calls why FS
(I think nibblebill) still use memory ? something wrong in nibblebill
?

BG
Dome C.



From freeswitch.noob at gmail.com  Mon Oct 26 06:16:53 2009
From: freeswitch.noob at gmail.com (freeswitch noob)
Date: Mon, 26 Oct 2009 08:16:53 -0500
Subject: [Freeswitch-users] Hostname
In-Reply-To: <65286EEC1128407F82A019CF0EFB9C38@chaos>
References: <7f783b530910231234u5d0e2fedxea4884d9bcddc9d3@mail.gmail.com>
	<2d9149cd0910231246t58d2bbd3k3466d9c0fca18000@mail.gmail.com>
	<200910231558.09699.chris@cloudtel.com>
	<7f783b530910231342t5393763aueff6bbc679b3c55@mail.gmail.com>
	<65286EEC1128407F82A019CF0EFB9C38@chaos>
Message-ID: <7f783b530910260616u35189aafleba089d1e41f17c@mail.gmail.com>

Thank you for the get idea,  It works perfectly.


On Fri, Oct 23, 2009 at 4:37 PM, Metik <freeswitch-users-list at metik.com>wrote:

>  Why not simply overwrite the value of the variable used throughout the
> script...
>
> -- xml_curl.conf --
> ...
>       <param name="gateway-url" value="
> http://localhost/index.php?xhostname=myhost" bindings="dialplan"/>
> ...
>
> -- index.php --
>
> <?
>
> $_REQUEST['hostname'] = "$xhostname";
>
> ...
>
> ----- Original Message -----
> *From:* freeswitch noob <freeswitch.noob at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Friday, October 23, 2009 4:42 PM
> *Subject:* Re: [Freeswitch-users] Hostname
>
> Yeah, I was just trying to make it easier on myself.  I have scripts from a
> friend that parse xml_curl requests based on the hostname, I was hoping to
> not have to re-write them to read something else from the post that FS makes
> from xml_curl.  But from what it sounds like I will have to.
>
>
> On Fri, Oct 23, 2009 at 2:58 PM, Chris Burns <chris at cloudtel.com> wrote:
>
>> one real quick way would be put different GET var in each server's binding
>>
>> On October 23, 2009 03:46:11 pm Kristian Kielhofner wrote:
>> > Can't you use different contexts or something else to tell them apart?
>> >
>> > On Fri, Oct 23, 2009 at 3:34 PM, freeswitch noob
>> >
>> > <freeswitch.noob at gmail.com> wrote:
>> > > I have mod xml_curl installed and I am getting the following passed to
>> my
>> > > script.
>> > >
>> > > [hostname] => myhost.local
>> > > [section] => dialplan
>> > >
>> > >
>> > > I also have multiple versions of FS running on the same box.  Is there
>> a
>> > > way to have each FS instance on my box have a unique hostname ?
>> > >
>> > > Thanks in advance.
>> > >
>> > > _______________________________________________
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From helmut.kuper at ewetel.de  Mon Oct 26 07:24:14 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Mon, 26 Oct 2009 15:24:14 +0100
Subject: [Freeswitch-users] INFO-Messages: Send_display in pre_answer
 state doesnt work anymore
In-Reply-To: <191c3a030910230733i4fce1c55v892056f549f67ed0@mail.gmail.com>
References: <4AE19D0E.9050301@ewetel.de>
	<191c3a030910230733i4fce1c55v892056f549f67ed0@mail.gmail.com>
Message-ID: <4AE5B10E.7050800@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Anthony,

thx, it works now again. But the only way get it to work is using
send_display after pre_answer. "(sip|originate)_callee_id_name" doesn't
work with playback.

regards
Helmut


On 23.10.2009 16:33, Anthony Minessale wrote:
> There was a concern with letting this happen before answer.
> I changed it in last revision, try it out.
> The problem is that the way polycom does it, it violates the RFC.
> 
> Also know you can set sip_callee_id_name and sip_callee_id_number before
> you send a ring_ready pre_answer or answer to send the display update in
> that packet instead of sending a separate one.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)

iD8DBQFK5bEO4tZeNddg3dwRAkycAKCg61dUy1jkmzauQUDliqYw81aA5ACgmeim
hayGIQk5KlRN8edZGLom1mQ=
=a5QJ
-----END PGP SIGNATURE-----



From brian at freeswitch.org  Mon Oct 26 07:35:07 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 26 Oct 2009 09:35:07 -0500
Subject: [Freeswitch-users] Downloaded tar vs latest SVN - 200 OK has
	more headers
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE2@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE2@EXMBXCLUS01.citservers.local>
Message-ID: <6444A50B-E5F2-4CB5-8AF9-C3D8EBE67BA4@freeswitch.org>

No you can't remove them... And they are 100% valid so your SBC is in  
the wrong.

/b

On Oct 26, 2009, at 12:05 AM, Ujjval Karihaloo wrote:

> Hi,
>
>  I used the downlaoded TAR ball and my calls worked, however, when  
> upgrading to the SVN release...my SBC is rejecting the 200 OK (when  
> the FS answers the call - using Conferencing app)..
>
> Here are teh bad and good 200 OK,,,,I see a lot of additional  
> headers startin gwith X:FS , can I remove them?
>

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From anthony.minessale at gmail.com  Mon Oct 26 07:41:55 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Oct 2009 09:41:55 -0500
Subject: [Freeswitch-users] Qustion about INFO messages after
	Connect/Answer
In-Reply-To: <4AE5813E.2030506@ewetel.de>
References: <4AD859A4.5000101@ewetel.de>
	<87f2f3b90910191435s7528e256r85df5f12fc634f24@mail.gmail.com>
	<4ADF483B.4050807@ewetel.de>
	<87f2f3b90910211048s208daf1eyfae501091ab75530@mail.gmail.com>
	<4AE07DF4.5040803@ewetel.de>
	<085B77EE-DA3F-4ADD-86DC-E59D1B6553EA@freeswitch.org>
	<191c3a030910221008n60341314hc7a7e6a55a50ef15@mail.gmail.com>
	<4AE1951A.1030906@ewetel.de>
	<191c3a030910230851i6e0b081ajaf6c294a7f9cd3a8@mail.gmail.com>
	<4AE5813E.2030506@ewetel.de>
Message-ID: <191c3a030910260741i32578e38qf504a496a4fba475@mail.gmail.com>

Could you maybe consolidate all of your problems into 1 thread.  I am
getting dizzy.  You have 2 on the same subject and you say it works on one
and does not on the other.

Last week we tested all of this with latest trunk and there is no longer any
problems of any sort with the display related stuff.


On Mon, Oct 26, 2009 at 6:00 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi Anthony,
>
> hm, no not really. There is no change in this behaviour in "FreeSWITCH
> Version 1.0.trunk (15225M)"
>
>
> There is still a caller name change in callee's display.
>
> I'm not sure who is wrong here. Either FS or Snom ...
>
>
> regards
> Helmut
>
>
> On 23.10.2009 17:51, Anthony Minessale wrote:
> > should be even better in 15210
> >
> >
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
>
> iD8DBQFK5YE94tZeNddg3dwRAigrAJ0chk/kiDQo2Z6yFbjpJovmPor46ACfValq
> u9RajDfF0rNdzkaUOVjULmk=
> =697e
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From lei.tlfly at gmail.com  Sun Oct 25 06:58:37 2009
From: lei.tlfly at gmail.com (Lei Tang)
Date: Sun, 25 Oct 2009 21:58:37 +0800
Subject: [Freeswitch-users] How to load user account from databse ?
Message-ID: <50c41b4e0910250658n74f92d01v56f1ab82087ad809@mail.gmail.com>

Hi All:
   I'm a newbie to FS.  I'm using  FS as a sbc and have about 20000 user
account . Does somebody can tell me how to  make FS load use account
information from a database such as mssql or mysql?  Could you give me a
sample configuration file?
   Thanks a lots.
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From kristian.kielhofner at gmail.com  Mon Oct 26 07:56:32 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Mon, 26 Oct 2009 10:56:32 -0400
Subject: [Freeswitch-users] Resend: Issues with SIP + TCP?
Message-ID: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>

I originally sent this last Friday but I've been unable to confirm it
ever made it to the list.

Hello everyone,

 I'm having some issues with SIP and TCP.  I've used it before with
success but I'm seeing some strange behavior...

 Level 7 debugs with siptrace on both profiles.  UDP invite from
softphone comes in on port 5062, it's supposed to bridge to
10.70.0.62.  When configured to use UDP FS sends an INVITE (nothing
currently answers) while TCP doesn't send anything (confirmed with
siptrace and packet sniffer).  I confirmed this behavior with a
gateway configured for TCP and appending ;transport=tcp to a bridge
line.

 This is trunk rev 15211 on an Intel Mac running Snow Leopard.  I've
also confirmed this behavior on an Intel Linux machine running Ubuntu
(not sure of version ATM).

TCP:

http://pastebin.freeswitch.org/10825

UDP:

http://pastebin.freeswitch.org/10826

dialplan (UDP):

   <extension name="smhpbx">
     <condition field="destination_number" expression="^(7887)$">
       <action application="set" data="call_timeout=60"/>
       <action application="set" data="effective_caller_id_name=Voalte Test"/>
       <action application="set"
data="effective_caller_id_number=19412848354"/>
       <action application="bridge" data="sofia/avaya/7887 at 10.70.0.62"/>
     </condition>
   </extension>

dialplan (TCP):

   <extension name="smhpbx">
     <condition field="destination_number" expression="^(7887)$">
       <action application="set" data="call_timeout=60"/>
       <action application="set" data="effective_caller_id_name=Voalte Test"/>
       <action application="set"
data="effective_caller_id_number=19412848354"/>
       <action application="bridge"
data="sofia/avaya/7887 at 10.70.0.62;transport=tcp"/>
     </condition>
   </extension>

 Any thoughts?

Thanks!

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From freeswitch.noob at gmail.com  Mon Oct 26 07:59:37 2009
From: freeswitch.noob at gmail.com (freeswitch noob)
Date: Mon, 26 Oct 2009 09:59:37 -0500
Subject: [Freeswitch-users] How to load user account from databse ?
In-Reply-To: <50c41b4e0910250658n74f92d01v56f1ab82087ad809@mail.gmail.com>
References: <50c41b4e0910250658n74f92d01v56f1ab82087ad809@mail.gmail.com>
Message-ID: <7f783b530910260759p6864e582y8a5d6a1db349e22c@mail.gmail.com>

Lei,

I am still learning myself, but I think I can help enough and others can
chime in where I am wrong.

If you have FS up and running you will need to install xml_curl (
http://wiki.freeswitch.org/wiki/Mod_xml_curl).



Then you will need to edit the config for it in your
conf/autoload_configs/xml_curl_conf.xml and add the following to the file.

<binding name="externalDir">
      <param name="gateway-url" value="http://my.webserver.com/index.php"
bindings="directory"/>
    </binding>


Then for every registered user call, a request will be posted to that script
location.

The posts vary but look something like:

[hostname] =>
test.local
[section] =>
directory

[tag_name] =>
domain

[key_name] =>
name

[key_value] =>
company1.test.local
[action] =>
sip_auth

[sip_profile] =>
internal
[sip_user_agent] => eyeBeam release 1100v stamp
47073
[sip_auth_username] =>
100
[sip_auth_realm] =>
company1.test.local
[sip_auth_nonce] =>
37bd639a-d181-4df3-b53d-ab4172ca3be9
[sip_auth_uri] =>
sip:company1.test.local;transport=udp
[sip_contact_user] =>
100
[sip_contact_host] =>
10.43.43.2
[sip_to_user] =>
100
[sip_to_host] =>
company1.test.local
[sip_from_user] =>
100
[sip_from_host] =>
company1.test.local
[sip_request_host] =>
company1.test.local
[sip_auth_qop] =>
auth
[sip_auth_cnonce] =>
207aae18a2af959346f87a2a3c2c7f8a
[sip_auth_nc] =>
00000001
[sip_auth_response] =>
770a1519789ba7606b3dcc6c4b7a99c5
[sip_auth_method] =>
REGISTER
[key] =>
id

[user] =>
100

[domain] =>
company1.test.local

[ip] => 10.43.43.2


Then your script can handle the connection to the DB and the verification of
the user information.  My example uses PHP but any server side language that
can be posted to can be used in the above scenario.


On Sun, Oct 25, 2009 at 8:58 AM, Lei Tang <lei.tlfly at gmail.com> wrote:

> Hi All:
>    I'm a newbie to FS.  I'm using  FS as a sbc and have about 20000 user
> account . Does somebody can tell me how to  make FS load use account
> information from a database such as mssql or mysql?  Could you give me a
> sample configuration file?
>    Thanks a lots.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Mon Oct 26 08:08:59 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Oct 2009 10:08:59 -0500
Subject: [Freeswitch-users] SIP UPDATE Method
In-Reply-To: <528227.52311.qm@web37508.mail.mud.yahoo.com>
References: <528227.52311.qm@web37508.mail.mud.yahoo.com>
Message-ID: <191c3a030910260808y45952bffu7118057caba4ea4@mail.gmail.com>

It was never there before and it caused extreme havoc once we added it so we
took it away again.


On Sun, Oct 25, 2009 at 12:55 AM, DJB <djbinter at yahoo.com> wrote:

> I am wondering why after I update to trunk-15225, the Allow: UPDATE method
> is no longer there.
>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15225
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER,
> NOTIFY, PUBLISH, SUBSCRIBE
>
> Am I missing something here?
>
> Thank you.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From kristian.kielhofner at gmail.com  Mon Oct 26 08:15:54 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Mon, 26 Oct 2009 11:15:54 -0400
Subject: [Freeswitch-users] Downloaded tar vs latest SVN - 200 OK has
	more headers
In-Reply-To: <6444A50B-E5F2-4CB5-8AF9-C3D8EBE67BA4@freeswitch.org>
References: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE2@EXMBXCLUS01.citservers.local>
	<6444A50B-E5F2-4CB5-8AF9-C3D8EBE67BA4@freeswitch.org>
Message-ID: <2d9149cd0910260815s5dc39ec2n869b0bdbfb0333ac@mail.gmail.com>

Brian,

  This bothers me a bit.

  Of course they are "valid".  Anything prefixed with X- should be
ignored by the remote end unless they are specifically looking for it.

  However:

1)  We all know that just because the spec says they are valid doesn't
make it so with every vendor.
2)  As long as these headers are being attached people are going to
keep coming to the list with issues because of them.
3)  Attaching (seemingly arbitrary) headers like this breaks "interop
agreements" that we (the users) have with carriers and other vendors.
Every header we add is another reason why $CARRIER will refuse to
support me because "my INVITE doesn't match what I had during interop
testing".
4)  Why do we /need/ these headers in the first place?  What is using
them?  If we don't really, really, really need them they shouldn't be
there.
5)  Depending on the reason for them being there, this should still be
configurable.  I just don't see why not.  Obviously we could always
just patch the code (Sonus log statements :) but that probably
shouldn't be necessary here.

  I'm not trying to be difficult; I'm just looking for a better
explanation and real discussion.

On Mon, Oct 26, 2009 at 10:35 AM, Brian West <brian at freeswitch.org> wrote:
> No you can't remove them... And they are 100% valid so your SBC is in the
> wrong.
> /b
> On Oct 26, 2009, at 12:05 AM, Ujjval Karihaloo wrote:
>
> Hi,
>
> ?I used the downlaoded TAR ball and my calls worked, however, when upgrading
> to the SVN release...my SBC is rejecting the 200 OK (when the FS answers the
> call - using?Conferencing app)..
>
> Here are teh bad and good 200 OK,,,,I see a lot of additional headers
> startin gwith X:FS , can I remove them?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From shiyanov at gmail.com  Mon Oct 26 08:20:48 2009
From: shiyanov at gmail.com (Artem Shiyanov)
Date: Mon, 26 Oct 2009 18:20:48 +0300
Subject: [Freeswitch-users] Mod_socket: custom caller sip domain in
	originate command
Message-ID: <d211f54a0910260820v5372c65exeba8a3d2bfca5da4@mail.gmail.com>

Hi there!

Please, suggest how to specify custom caller sip domain (logical) in
originate command.
I've been trying several alternatives but no one worked:
1) specify full sip address in
origination_caller_id_number=1000 at uat.pbx.mblagov.starpoundtech.net - FS
adds its IP address so the
result From address would be 1000 at uat.pbx.mblagov.starpoundtech.net@
172.26.200.250
2) specify sip_from_uri varuable like follows:
bgapi originate {origination_caller_id_number=1000,
sip_from_uri=1000 at uat.pbx.mblagov.starpoundtech.net,originate_timeout=10}[origination_uuid=bf8e9778-d410-4504-988d-2d405303183c]sofia/internal/1002%172.26.200.250
&park()
FS doesn't even dial the called endpoint.

Thanks,
Artem
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From ujjval at simplesignal.com  Mon Oct 26 08:21:17 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Mon, 26 Oct 2009 08:21:17 -0700
Subject: [Freeswitch-users] Setting up Conference with Moderator
In-Reply-To: <114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local>
	<3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local>
	<E97CD3D0-52FF-44D1-BB89-23816045CE21@gmail.com>
	<71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>,
	<114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com>
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local>

Thx a lot Rob, reading the wiki your way or using IVR seems correct..
===============
The wiki also says that the wait-mod might be  "used in conjunction 
with an IVR where the moderators are authenticated with an extra pass-
code", which is what I did.  I guess that's why I didn't understand 
the point of the +pin.
======================

I will try it out.

Again thx a lot for your help. Will keep everyone posted.

________________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Forman [rob4manhere at gmail.com]
Sent: Friday, October 23, 2009 12:22 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator

I just re-tested with the pin in my dial plan:

   <action application="conference" data="conference 123456 at default
+flags{}+1234" />

And it doesn't challenge me for the pin.  I just drop right in.  I
figured this is how it was intended, since the wiki says the pin is
set initially and only challenged in later attempts [by future callers]:

"The first time a conference name (confname) is used, it will be
created on demand, and the pin will be set to what ever is specified
at that time: the pin in the data string if specified, or if not, the
"pin" setting in the conference profile, and if that is also
unspecified, then there is no pin protection. Any later attempt to
join the conference must specify the same pin number, if one existed
when it was created. "


The wiki also says that the wait-mod might be  "used in conjunction
with an IVR where the moderators are authenticated with an extra pass-
code", which is what I did.  I guess that's why I didn't understand
the point of the +pin.

I'm sure there's a scenario where its used and useful, the wiki just
doesn't explain it.

Rob

On Oct 23, 2009, at 12:43 PM, Brian West wrote:

> Well first off you're not defining a pine here...
>
> confname at profilename+flags{mute|deaf|waste|moderator}+[conference pin
> number]
>
> That might be why its not asking for a pin.
>
> /b
>
> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:
>
>>      <entry action="menu-exec-app" digits="1" param="conference
>> 123456 at default+flags{}" />
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From anthony.minessale at gmail.com  Mon Oct 26 08:25:09 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Oct 2009 10:25:09 -0500
Subject: [Freeswitch-users] Resend: Issues with SIP + TCP?
In-Reply-To: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>
References: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>
Message-ID: <191c3a030910260825w25b1891g523d5e30d72ee94f@mail.gmail.com>

i cant seem to reproduce it.

originate sofia/internal/1235 at conference.freeswitch.org;transport=tcp 9998

I get a working call and trace.

Could you possibly have a dns error?  I know it's an ip but it may still
fail if it has no dns.

try

sofia loglevel all 9

and look for other errors.




On Mon, Oct 26, 2009 at 9:56 AM, Kristian Kielhofner <
kristian.kielhofner at gmail.com> wrote:

> I originally sent this last Friday but I've been unable to confirm it
> ever made it to the list.
>
> Hello everyone,
>
>  I'm having some issues with SIP and TCP.  I've used it before with
> success but I'm seeing some strange behavior...
>
>  Level 7 debugs with siptrace on both profiles.  UDP invite from
> softphone comes in on port 5062, it's supposed to bridge to
> 10.70.0.62.  When configured to use UDP FS sends an INVITE (nothing
> currently answers) while TCP doesn't send anything (confirmed with
> siptrace and packet sniffer).  I confirmed this behavior with a
> gateway configured for TCP and appending ;transport=tcp to a bridge
> line.
>
>  This is trunk rev 15211 on an Intel Mac running Snow Leopard.  I've
> also confirmed this behavior on an Intel Linux machine running Ubuntu
> (not sure of version ATM).
>
> TCP:
>
> http://pastebin.freeswitch.org/10825
>
> UDP:
>
> http://pastebin.freeswitch.org/10826
>
> dialplan (UDP):
>
>   <extension name="smhpbx">
>     <condition field="destination_number" expression="^(7887)$">
>       <action application="set" data="call_timeout=60"/>
>       <action application="set" data="effective_caller_id_name=Voalte
> Test"/>
>       <action application="set"
> data="effective_caller_id_number=19412848354"/>
>       <action application="bridge" data="sofia/avaya/7887 at 10.70.0.62"/>
>     </condition>
>   </extension>
>
> dialplan (TCP):
>
>   <extension name="smhpbx">
>     <condition field="destination_number" expression="^(7887)$">
>       <action application="set" data="call_timeout=60"/>
>       <action application="set" data="effective_caller_id_name=Voalte
> Test"/>
>       <action application="set"
> data="effective_caller_id_number=19412848354"/>
>       <action application="bridge"
> data="sofia/avaya/7887 at 10.70.0.62;transport=tcp"/>
>     </condition>
>   </extension>
>
>  Any thoughts?
>
> Thanks!
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mayamatakeshi at gmail.com  Mon Oct 26 08:32:51 2009
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Tue, 27 Oct 2009 00:32:51 +0900
Subject: [Freeswitch-users] Mod_socket: custom caller sip domain in
	originate command
In-Reply-To: <d211f54a0910260820v5372c65exeba8a3d2bfca5da4@mail.gmail.com>
References: <d211f54a0910260820v5372c65exeba8a3d2bfca5da4@mail.gmail.com>
Message-ID: <15b9404e0910260832o3a297df5y1dc2093000decf17@mail.gmail.com>

On Tue, Oct 27, 2009 at 12:20 AM, Artem Shiyanov <shiyanov at gmail.com> wrote:

> Hi there!
>
> Please, suggest how to specify custom caller sip domain (logical) in
> originate command.
> I've been trying several alternatives but no one worked:
> 1) specify full sip address in
> origination_caller_id_number=1000 at uat.pbx.mblagov.starpoundtech.net - FS
> adds its IP address so the
> result From address would be 1000 at uat.pbx.mblagov.starpoundtech.net@
> 172.26.200.250
> 2) specify sip_from_uri varuable like follows:
> bgapi originate {origination_caller_id_number=1000,
> sip_from_uri=1000 at uat.pbx.mblagov.starpoundtech.net,originate_timeout=10}[origination_uuid=bf8e9778-d410-4504-988d-2d405303183c]sofia/internal/1002%172.26.200.250
> &park()
> FS doesn't even dial the called endpoint.
>

I use
sip_invite_domain
on bridge. I suppose it works with originate too.
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From delianspam at gmail.com  Mon Oct 26 08:35:30 2009
From: delianspam at gmail.com (Delian Tashev)
Date: Mon, 26 Oct 2009 17:35:30 +0200
Subject: [Freeswitch-users] Reroute a Call Based on the Disconnect Cause
References: <mailman.3342.1253283103.2787.freeswitch-users@lists.freeswitch.org>
Message-ID: <12a101ca5651$f47839e0$dd68ada0$@com>

Hello VoIP Geeks!

I am looking for a way to reroute calls on specific disconnect reasons. My
application is in Python. I will provide my solves so far, which for some
reason I cannot use. Why I need rerouting? Well, I am using the LCR module
which provides multiple routes per destination, like this:
routes=
[lcr_carrier=fs-rocks.com,lcr_rate=0.15000]sofia/default/1174 at 194.135.34.200
:5062|
[lcr_carrier=fs-rocks.com,lcr_rate=0.16000]sofia/default/1174 at 194.135.34.200
:5064

Solve 1: Python solve
Split the routes string to multiple routes. Bridge the call. Check if the
call failed and bridge again.

ivr_reroute_on=(\

    "CALL_REJECTED",\
    "CRASH",\
    "INVALID_NUMBER_FORMAT",\
    "NORMAL_TEMPORARY_FAILURE",\
    "NO_ROUTE_DESTINATION",\
    "NO_USER_RESPONSE",\

    "NUMBER_CHANGED",\

    "RECOVERY_ON_TIMER_EXPIRE",\

    "SERVICE_NOT_IMPLEMENTED",\

)       
...
session.execute("bridge", route)
hangup_cause = str(session.getVariable("originate_disposition"))
if hangup_cause in ivr_reroute_on:
 #bridge again
 ...
else:
 ...
 #do not bridge again

This is tested and works. However my manager does not likes it. I am not
commenting on the side effects of this code (i.e. if you have some
timers/schedules/scheduled events etc).

Solve 2: Freeswitch Source Code solve
Modify the source code of "switch_ivr_originate.c":

                                                if (to &&
!oglobals.continue_on_timeout) {
                                                                goto
outer_for;
                                                }
                                                after line 2523:

                                                +if
(switch_channel_cause2str(*cause) == SWITCH_CAUSE_CALL_REJECTED ) {
                                                +             goto
outer_for;
                                                +}

(In the best case I will add a new channel variable to Freeswitch to control
all the disconnect reasons from here) However my manager again does not
likes this. He says that I am not good enough in C/C++ to use this solution.

Solve 3: continue_on_fail
I have tried using the channel variable "continue_on_fail" which I believe
works only in the dialplan with no luck. I was unable to get it running both
in the dialplan and in Python. By the way I saw that "continue_on_fail" is
indeed implemented elsewhere, while "continue_on_timeout" is exactly in
"switch_ivr_originate.c" as expected.

Solve 4:
So can you please people recommend me any other solution? Any function or
variable to do the trick?

Thank you for your time and help!

Delian Tashev




From anthony.minessale at gmail.com  Mon Oct 26 08:41:11 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Oct 2009 10:41:11 -0500
Subject: [Freeswitch-users] Downloaded tar vs latest SVN - 200 OK has
	more headers
In-Reply-To: <2d9149cd0910260815s5dc39ec2n869b0bdbfb0333ac@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE2@EXMBXCLUS01.citservers.local>
	<6444A50B-E5F2-4CB5-8AF9-C3D8EBE67BA4@freeswitch.org>
	<2d9149cd0910260815s5dc39ec2n869b0bdbfb0333ac@mail.gmail.com>
Message-ID: <191c3a030910260841i6f0f0390mc5bfeab6e5ec7ab6@mail.gmail.com>

The headers are used to pass the callee-id info back to the other side so
you have the id of who you called.
The standards have failed us in this case as everything does it differently
to the point that there is no standard thus we have invented our own way to
carry this across from one FreeSWITCH box to another, but of course we can
never make anybody happy. =/


On Mon, Oct 26, 2009 at 10:15 AM, Kristian Kielhofner <
kristian.kielhofner at gmail.com> wrote:

> Brian,
>
>  This bothers me a bit.
>
>  Of course they are "valid".  Anything prefixed with X- should be
> ignored by the remote end unless they are specifically looking for it.
>
>  However:
>
> 1)  We all know that just because the spec says they are valid doesn't
> make it so with every vendor.
> 2)  As long as these headers are being attached people are going to
> keep coming to the list with issues because of them.
> 3)  Attaching (seemingly arbitrary) headers like this breaks "interop
> agreements" that we (the users) have with carriers and other vendors.
> Every header we add is another reason why $CARRIER will refuse to
> support me because "my INVITE doesn't match what I had during interop
> testing".
> 4)  Why do we /need/ these headers in the first place?  What is using
> them?  If we don't really, really, really need them they shouldn't be
> there.
> 5)  Depending on the reason for them being there, this should still be
> configurable.  I just don't see why not.  Obviously we could always
> just patch the code (Sonus log statements :) but that probably
> shouldn't be necessary here.
>
>  I'm not trying to be difficult; I'm just looking for a better
> explanation and real discussion.
>
> On Mon, Oct 26, 2009 at 10:35 AM, Brian West <brian at freeswitch.org> wrote:
> > No you can't remove them... And they are 100% valid so your SBC is in the
> > wrong.
> > /b
> > On Oct 26, 2009, at 12:05 AM, Ujjval Karihaloo wrote:
> >
> > Hi,
> >
> >  I used the downlaoded TAR ball and my calls worked, however, when
> upgrading
> > to the SVN release...my SBC is rejecting the 200 OK (when the FS answers
> the
> > call - using Conferencing app)..
> >
> > Here are teh bad and good 200 OK,,,,I see a lot of additional headers
> > startin gwith X:FS , can I remove them?
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From ujjval at simplesignal.com  Mon Oct 26 08:39:53 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Mon, 26 Oct 2009 08:39:53 -0700
Subject: [Freeswitch-users] Estimating Call Capacity
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE7@EXMBXCLUS01.citservers.local>

With the following spec for CPU and Memory can someone help me guesstimating how many simultaneous calls and Calls/sec a FS server can handle - Used as a Conferencing Server.

cat /proc/cpuinfo
processor       : 0
vendor_id       : GenuineIntel
cpu family      : 15
model           : 4
model name      : Intel(R) Pentium(R) 4 CPU 2.80GHz
stepping        : 1
cpu MHz         : 2800.386
cache size      : 1024 KB
physical id     : 0
siblings        : 2
core id         : 0
cpu cores       : 1
fdiv_bug        : no
hlt_bug         : no
f00f_bug        : no
coma_bug        : no
fpu             : yes
fpu_exception   : yes
cpuid level     : 5
wp              : yes
flags           : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm constant_tsc pni monitor ds_cpl cid cx16 xtpr
bogomips        : 5604.12

processor       : 1
vendor_id       : GenuineIntel
cpu family      : 15
model           : 4
model name      : Intel(R) Pentium(R) 4 CPU 2.80GHz
stepping        : 1
cpu MHz         : 2800.386
cache size      : 1024 KB
physical id     : 0
siblings        : 2
core id         : 0
cpu cores       : 1
fdiv_bug        : no
hlt_bug         : no
f00f_bug        : no
coma_bug        : no
fpu             : yes
fpu_exception   : yes
cpuid level     : 5
wp              : yes
flags           : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm constant_tsc pni monitor ds_cpl cid cx16 xtpr
bogomips        : 5600.22


# cat /proc/meminfo
MemTotal:      1035080 kB
MemFree:        517972 kB
Buffers:         85812 kB
Cached:         271264 kB
SwapCached:          0 kB
Active:         224292 kB
Inactive:       223008 kB
HighTotal:      130816 kB
HighFree:        29484 kB
LowTotal:       904264 kB
LowFree:        488488 kB
SwapTotal:     2031608 kB
SwapFree:      2031520 kB
Dirty:              80 kB
Writeback:           0 kB
AnonPages:       90172 kB
Mapped:          39880 kB
Slab:            60060 kB
PageTables:       3232 kB
NFS_Unstable:        0 kB
Bounce:              0 kB
CommitLimit:   2549148 kB
Committed_AS:   345780 kB
VmallocTotal:   114680 kB
VmallocUsed:      3584 kB
VmallocChunk:   110888 kB
HugePages_Total:     0
HugePages_Free:      0
HugePages_Rsvd:      0
Hugepagesize:     4096 kB


From kristian.kielhofner at gmail.com  Mon Oct 26 08:47:20 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Mon, 26 Oct 2009 11:47:20 -0400
Subject: [Freeswitch-users] Resend: Issues with SIP + TCP?
In-Reply-To: <191c3a030910260825w25b1891g523d5e30d72ee94f@mail.gmail.com>
References: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>
	<191c3a030910260825w25b1891g523d5e30d72ee94f@mail.gmail.com>
Message-ID: <2d9149cd0910260847i718c420ek99d8d291aa37e143@mail.gmail.com>

Tony,

  It seemed strange to me too (I'm using TCP in other places).

  I'll take another look at this with your suggestions for debugging.

  Thanks!

On Mon, Oct 26, 2009 at 11:25 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> i cant seem to reproduce it.
>
> originate sofia/internal/1235 at conference.freeswitch.org;transport=tcp 9998
>
> I get a working call and trace.
>
> Could you possibly have a dns error?? I know it's an ip but it may still
> fail if it has no dns.
>
> try
>
> sofia loglevel all 9
>
> and look for other errors.
>
>
>
>
> On Mon, Oct 26, 2009 at 9:56 AM, Kristian Kielhofner
> <kristian.kielhofner at gmail.com> wrote:
>>
>> I originally sent this last Friday but I've been unable to confirm it
>> ever made it to the list.
>>
>> Hello everyone,
>>
>> ?I'm having some issues with SIP and TCP. ?I've used it before with
>> success but I'm seeing some strange behavior...
>>
>> ?Level 7 debugs with siptrace on both profiles. ?UDP invite from
>> softphone comes in on port 5062, it's supposed to bridge to
>> 10.70.0.62. ?When configured to use UDP FS sends an INVITE (nothing
>> currently answers) while TCP doesn't send anything (confirmed with
>> siptrace and packet sniffer). ?I confirmed this behavior with a
>> gateway configured for TCP and appending ;transport=tcp to a bridge
>> line.
>>
>> ?This is trunk rev 15211 on an Intel Mac running Snow Leopard. ?I've
>> also confirmed this behavior on an Intel Linux machine running Ubuntu
>> (not sure of version ATM).
>>
>> TCP:
>>
>> http://pastebin.freeswitch.org/10825
>>
>> UDP:
>>
>> http://pastebin.freeswitch.org/10826
>>
>> dialplan (UDP):
>>
>> ? <extension name="smhpbx">
>> ? ? <condition field="destination_number" expression="^(7887)$">
>> ? ? ? <action application="set" data="call_timeout=60"/>
>> ? ? ? <action application="set" data="effective_caller_id_name=Voalte
>> Test"/>
>> ? ? ? <action application="set"
>> data="effective_caller_id_number=19412848354"/>
>> ? ? ? <action application="bridge" data="sofia/avaya/7887 at 10.70.0.62"/>
>> ? ? </condition>
>> ? </extension>
>>
>> dialplan (TCP):
>>
>> ? <extension name="smhpbx">
>> ? ? <condition field="destination_number" expression="^(7887)$">
>> ? ? ? <action application="set" data="call_timeout=60"/>
>> ? ? ? <action application="set" data="effective_caller_id_name=Voalte
>> Test"/>
>> ? ? ? <action application="set"
>> data="effective_caller_id_number=19412848354"/>
>> ? ? ? <action application="bridge"
>> data="sofia/avaya/7887 at 10.70.0.62;transport=tcp"/>
>> ? ? </condition>
>> ? </extension>
>>
>> ?Any thoughts?
>>
>> Thanks!
>>
>> --
>> Kristian Kielhofner
>> http://www.astlinux.org
>> http://blog.krisk.org
>> http://www.star2star.com
>> http://www.submityoursip.com
>> http://www.voalte.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From helmut.kuper at ewetel.de  Mon Oct 26 08:51:15 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Mon, 26 Oct 2009 16:51:15 +0100
Subject: [Freeswitch-users] Qustion about INFO messages
	after	Connect/Answer
In-Reply-To: <191c3a030910260741i32578e38qf504a496a4fba475@mail.gmail.com>
References: <4AD859A4.5000101@ewetel.de>	<87f2f3b90910191435s7528e256r85df5f12fc634f24@mail.gmail.com>	<4ADF483B.4050807@ewetel.de>	<87f2f3b90910211048s208daf1eyfae501091ab75530@mail.gmail.com>	<4AE07DF4.5040803@ewetel.de>	<085B77EE-DA3F-4ADD-86DC-E59D1B6553EA@freeswitch.org>	<191c3a030910221008n60341314hc7a7e6a55a50ef15@mail.gmail.com>	<4AE1951A.1030906@ewetel.de>	<191c3a030910230851i6e0b081ajaf6c294a7f9cd3a8@mail.gmail.com>	<4AE5813E.2030506@ewetel.de>
	<191c3a030910260741i32578e38qf504a496a4fba475@mail.gmail.com>
Message-ID: <4AE5C573.6020601@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Anthony,

sorry for making you dizzy ... bun in fact in my point of view I have
two different problems.

1.
One concerns the way using send_display in pre_answer mode. Simply to
send error texts to caller's display. This works again with latest trunk.


2.
The other one (this thread) concerns the (in my eyes newly introduced)
two INFO messages which FS sends to callee after callee picked up his
phone. The first INFO switches callee's display to a name set in
"originaton_callee_id_name" (a) and immediately after that the second
INFO switches it back to callee's real name (b). You can see this in
display only if (a) and (b) are not the same.

I used Snom 370 phones with FW 8.2.16 as caller and callee. I did an
internal call (sip to sip).


Maybe this is the same code problem, but on my level they are two
different problems, so sorry for confusing you. I hope this clears
things up.


On 26.10.2009 15:41, Anthony Minessale wrote:
> Could you maybe consolidate all of your problems into 1 thread.  I am
> getting dizzy.  You have 2 on the same subject and you say it works on
> one and does not on the other.
> 
> Last week we tested all of this with latest trunk and there is no longer
> any problems of any sort with the display related stuff.

-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)

iD8DBQFK5cVz4tZeNddg3dwRAt2+AJsEQuRTEhE74+XvciTXOC0+kr0tbwCfRzUf
fgVIZwAV/IthjWwvXzRO3TA=
=n7Sr
-----END PGP SIGNATURE-----



From shiyanov at gmail.com  Mon Oct 26 08:56:57 2009
From: shiyanov at gmail.com (Artem Shiyanov)
Date: Mon, 26 Oct 2009 18:56:57 +0300
Subject: [Freeswitch-users] Mod_socket: custom caller sip domain in
	originate command
In-Reply-To: <15b9404e0910260832o3a297df5y1dc2093000decf17@mail.gmail.com>
References: <d211f54a0910260820v5372c65exeba8a3d2bfca5da4@mail.gmail.com>
	<15b9404e0910260832o3a297df5y1dc2093000decf17@mail.gmail.com>
Message-ID: <d211f54a0910260856l53e59888i87704b6ba2ef5310@mail.gmail.com>

Tested- it works!
Thanks a lot!!



On Mon, Oct 26, 2009 at 6:32 PM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:

>
>
> On Tue, Oct 27, 2009 at 12:20 AM, Artem Shiyanov <shiyanov at gmail.com>wrote:
>
>> Hi there!
>>
>> Please, suggest how to specify custom caller sip domain (logical) in
>> originate command.
>> I've been trying several alternatives but no one worked:
>> 1) specify full sip address in
>> origination_caller_id_number=1000 at uat.pbx.mblagov.starpoundtech.net - FS
>> adds its IP address so the
>> result From address would be 1000 at uat.pbx.mblagov.starpoundtech.net@
>> 172.26.200.250
>> 2) specify sip_from_uri varuable like follows:
>> bgapi originate {origination_caller_id_number=1000,
>> sip_from_uri=1000 at uat.pbx.mblagov.starpoundtech.net,originate_timeout=10}[origination_uuid=bf8e9778-d410-4504-988d-2d405303183c]sofia/internal/1002%172.26.200.250
>> &park()
>> FS doesn't even dial the called endpoint.
>>
>
> I use
> sip_invite_domain
> on bridge. I suppose it works with originate too.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kristian.kielhofner at gmail.com  Mon Oct 26 08:59:41 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Mon, 26 Oct 2009 11:59:41 -0400
Subject: [Freeswitch-users] Downloaded tar vs latest SVN - 200 OK has
	more headers
In-Reply-To: <191c3a030910260841i6f0f0390mc5bfeab6e5ec7ab6@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE2@EXMBXCLUS01.citservers.local>
	<6444A50B-E5F2-4CB5-8AF9-C3D8EBE67BA4@freeswitch.org>
	<2d9149cd0910260815s5dc39ec2n869b0bdbfb0333ac@mail.gmail.com>
	<191c3a030910260841i6f0f0390mc5bfeab6e5ec7ab6@mail.gmail.com>
Message-ID: <2d9149cd0910260859v21b70bcnd949ff24170858b0@mail.gmail.com>

Anthony,

  So if I'm understanding you correctly, if you are always using
FreeSWITCH as an edge to other systems you should be able to safely
disable these headers?

On Mon, Oct 26, 2009 at 11:41 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> The headers are used to pass the callee-id info back to the other side so
> you have the id of who you called.
> The standards have failed us in this case as everything does it differently
> to the point that there is no standard thus we have invented our own way to
> carry this across from one FreeSWITCH box to another, but of course we can
> never make anybody happy. =/
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From anthony.minessale at gmail.com  Mon Oct 26 09:04:38 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Oct 2009 11:04:38 -0500
Subject: [Freeswitch-users] Qustion about INFO messages after
	Connect/Answer
In-Reply-To: <4AE5C573.6020601@ewetel.de>
References: <4AD859A4.5000101@ewetel.de>
	<87f2f3b90910211048s208daf1eyfae501091ab75530@mail.gmail.com>
	<4AE07DF4.5040803@ewetel.de>
	<085B77EE-DA3F-4ADD-86DC-E59D1B6553EA@freeswitch.org>
	<191c3a030910221008n60341314hc7a7e6a55a50ef15@mail.gmail.com>
	<4AE1951A.1030906@ewetel.de>
	<191c3a030910230851i6e0b081ajaf6c294a7f9cd3a8@mail.gmail.com>
	<4AE5813E.2030506@ewetel.de>
	<191c3a030910260741i32578e38qf504a496a4fba475@mail.gmail.com>
	<4AE5C573.6020601@ewetel.de>
Message-ID: <191c3a030910260904j275b752fi1135338c2d29ed2b@mail.gmail.com>

depending on your dialplan every time you bridge to a channel it changes the
display to match who you are talking to.  if you tried to set it with the
variable then you call someone that is going to cause this.  Take away the
display app and/or any special variables and let it naturally work.


On Mon, Oct 26, 2009 at 10:51 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi Anthony,
>
> sorry for making you dizzy ... bun in fact in my point of view I have
> two different problems.
>
> 1.
> One concerns the way using send_display in pre_answer mode. Simply to
> send error texts to caller's display. This works again with latest trunk.
>
>
> 2.
> The other one (this thread) concerns the (in my eyes newly introduced)
> two INFO messages which FS sends to callee after callee picked up his
> phone. The first INFO switches callee's display to a name set in
> "originaton_callee_id_name" (a) and immediately after that the second
> INFO switches it back to callee's real name (b). You can see this in
> display only if (a) and (b) are not the same.
>
> I used Snom 370 phones with FW 8.2.16 as caller and callee. I did an
> internal call (sip to sip).
>
>
> Maybe this is the same code problem, but on my level they are two
> different problems, so sorry for confusing you. I hope this clears
> things up.
>
>
> On 26.10.2009 15:41, Anthony Minessale wrote:
> > Could you maybe consolidate all of your problems into 1 thread.  I am
> > getting dizzy.  You have 2 on the same subject and you say it works on
> > one and does not on the other.
> >
> > Last week we tested all of this with latest trunk and there is no longer
> > any problems of any sort with the display related stuff.
>
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
>
> iD8DBQFK5cVz4tZeNddg3dwRAt2+AJsEQuRTEhE74+XvciTXOC0+kr0tbwCfRzUf
> fgVIZwAV/IthjWwvXzRO3TA=
> =n7Sr
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From tculjaga at gmail.com  Mon Oct 26 09:05:15 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 26 Oct 2009 17:05:15 +0100
Subject: [Freeswitch-users] Downloaded tar vs latest SVN - 200 OK has
	more headers
In-Reply-To: <191c3a030910260841i6f0f0390mc5bfeab6e5ec7ab6@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE2@EXMBXCLUS01.citservers.local>
	<6444A50B-E5F2-4CB5-8AF9-C3D8EBE67BA4@freeswitch.org>
	<2d9149cd0910260815s5dc39ec2n869b0bdbfb0333ac@mail.gmail.com>
	<191c3a030910260841i6f0f0390mc5bfeab6e5ec7ab6@mail.gmail.com>
Message-ID: <65d96fc80910260905r3953ba23p346271cdbafe25a@mail.gmail.com>

On Mon, Oct 26, 2009 at 4:41 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> The headers are used to pass the callee-id info back to the other side so
> you have the id of who you called.
> The standards have failed us in this case as everything does it differently
> to the point that there is no standard thus we have invented our own way to
> carry this across from one FreeSWITCH box to another, but of course we can
> never make anybody happy. =/
>
>
I agree with you, X headers should be ignored by the equipment normally.
Anyhow Kristian has a point here; there will be a lot of complains because
of broken SIP stack on many vendor equipments

So, can you consider some customizable a config option for such headers?

T.
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From anthony.minessale at gmail.com  Mon Oct 26 09:16:58 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Oct 2009 11:16:58 -0500
Subject: [Freeswitch-users] Downloaded tar vs latest SVN - 200 OK has
	more headers
In-Reply-To: <65d96fc80910260905r3953ba23p346271cdbafe25a@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE2@EXMBXCLUS01.citservers.local>
	<6444A50B-E5F2-4CB5-8AF9-C3D8EBE67BA4@freeswitch.org>
	<2d9149cd0910260815s5dc39ec2n869b0bdbfb0333ac@mail.gmail.com>
	<191c3a030910260841i6f0f0390mc5bfeab6e5ec7ab6@mail.gmail.com>
	<65d96fc80910260905r3953ba23p346271cdbafe25a@mail.gmail.com>
Message-ID: <191c3a030910260916x47884d07t9abc9b779afbde68@mail.gmail.com>

Thus perpetuating the wild-west of sip where you can't do anything according
to spec because you have to worry about stupid things not keeping up.
Sounds like the education system where I live too.....

I'll see what I can do.  It's always the other end that ppl pay for that
drive the free stuff to change its code.


On Mon, Oct 26, 2009 at 11:05 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:

>
>
> On Mon, Oct 26, 2009 at 4:41 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> The headers are used to pass the callee-id info back to the other side so
>> you have the id of who you called.
>> The standards have failed us in this case as everything does it
>> differently to the point that there is no standard thus we have invented our
>> own way to carry this across from one FreeSWITCH box to another, but of
>> course we can never make anybody happy. =/
>>
>>
> I agree with you, X headers should be ignored by the equipment normally.
> Anyhow Kristian has a point here; there will be a lot of complains because
> of broken SIP stack on many vendor equipments
>
> So, can you consider some customizable a config option for such headers?
>
> T.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From peter.olsson at visionutveckling.se  Mon Oct 26 09:56:01 2009
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 26 Oct 2009 17:56:01 +0100
Subject: [Freeswitch-users] Resend: Issues with SIP + TCP?
In-Reply-To: <2d9149cd0910260847i718c420ek99d8d291aa37e143@mail.gmail.com>
References: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>
	<191c3a030910260825w25b1891g523d5e30d72ee94f@mail.gmail.com>
	<2d9149cd0910260847i718c420ek99d8d291aa37e143@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C550895310C@cooper>

I'm also having problems with this. When running FS compiled about 10 days ago it works fine (don't remember exact revision), but when using latest SVN it doesn't work anymore. I seems like it's trying to use UDP when it should use TCP.

My setup is this: Avaya Communication Manager PBX -> Talks TLS to Avaya SIP SES Server -> Talks TCP to FreeSwitch.

The replies from FS seems to be sent using UDP instead of TCP, and when I keep the config and revert to the 10 day old version it starts working again, so there is definately something wrong.

I'll try to do some more testing, and get back with some SIP-traces as well.

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Kristian Kielhofner
Skickat: den 26 oktober 2009 16:47
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Resend: Issues with SIP + TCP?

Tony,

  It seemed strange to me too (I'm using TCP in other places).

  I'll take another look at this with your suggestions for debugging.

  Thanks!

On Mon, Oct 26, 2009 at 11:25 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> i cant seem to reproduce it.
>
> originate sofia/internal/1235 at conference.freeswitch.org;transport=tcp 9998
>
> I get a working call and trace.
>
> Could you possibly have a dns error?? I know it's an ip but it may still
> fail if it has no dns.
>
> try
>
> sofia loglevel all 9
>
> and look for other errors.
>
>
>
>
> On Mon, Oct 26, 2009 at 9:56 AM, Kristian Kielhofner
> <kristian.kielhofner at gmail.com> wrote:
>>
>> I originally sent this last Friday but I've been unable to confirm it
>> ever made it to the list.
>>
>> Hello everyone,
>>
>> ?I'm having some issues with SIP and TCP. ?I've used it before with
>> success but I'm seeing some strange behavior...
>>
>> ?Level 7 debugs with siptrace on both profiles. ?UDP invite from
>> softphone comes in on port 5062, it's supposed to bridge to
>> 10.70.0.62. ?When configured to use UDP FS sends an INVITE (nothing
>> currently answers) while TCP doesn't send anything (confirmed with
>> siptrace and packet sniffer). ?I confirmed this behavior with a
>> gateway configured for TCP and appending ;transport=tcp to a bridge
>> line.
>>
>> ?This is trunk rev 15211 on an Intel Mac running Snow Leopard. ?I've
>> also confirmed this behavior on an Intel Linux machine running Ubuntu
>> (not sure of version ATM).
>>
>> TCP:
>>
>> http://pastebin.freeswitch.org/10825
>>
>> UDP:
>>
>> http://pastebin.freeswitch.org/10826
>>
>> dialplan (UDP):
>>
>> ? <extension name="smhpbx">
>> ? ? <condition field="destination_number" expression="^(7887)$">
>> ? ? ? <action application="set" data="call_timeout=60"/>
>> ? ? ? <action application="set" data="effective_caller_id_name=Voalte
>> Test"/>
>> ? ? ? <action application="set"
>> data="effective_caller_id_number=19412848354"/>
>> ? ? ? <action application="bridge" data="sofia/avaya/7887 at 10.70.0.62"/>
>> ? ? </condition>
>> ? </extension>
>>
>> dialplan (TCP):
>>
>> ? <extension name="smhpbx">
>> ? ? <condition field="destination_number" expression="^(7887)$">
>> ? ? ? <action application="set" data="call_timeout=60"/>
>> ? ? ? <action application="set" data="effective_caller_id_name=Voalte
>> Test"/>
>> ? ? ? <action application="set"
>> data="effective_caller_id_number=19412848354"/>
>> ? ? ? <action application="bridge"
>> data="sofia/avaya/7887 at 10.70.0.62;transport=tcp"/>
>> ? ? </condition>
>> ? </extension>
>>
>> ?Any thoughts?
>>
>> Thanks!
>>
>> --
>> Kristian Kielhofner
>> http://www.astlinux.org
>> http://blog.krisk.org
>> http://www.star2star.com
>> http://www.submityoursip.com
>> http://www.voalte.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4ae5c6c832935743011996!




From peter.olsson at visionutveckling.se  Mon Oct 26 10:06:47 2009
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 26 Oct 2009 18:06:47 +0100
Subject: [Freeswitch-users] Resend: Issues with SIP + TCP?
In-Reply-To: <2d9149cd0910260847i718c420ek99d8d291aa37e143@mail.gmail.com>
References: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>
	<191c3a030910260825w25b1891g523d5e30d72ee94f@mail.gmail.com>
	<2d9149cd0910260847i718c420ek99d8d291aa37e143@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5508953113@cooper>

Hmm... I remembered incorrectly about my setup :) The Avaya PBX talks TLS to the Avaya SES Server, and then UDP to FS, not TCP - sorry, my bad!

However, something that has changed the last 10 days seems to affect my setup so it doesn't work anymore. I'll do some more SIP tracing, and get back when I know more about it.

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Kristian Kielhofner
Skickat: den 26 oktober 2009 16:47
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Resend: Issues with SIP + TCP?

Tony,

  It seemed strange to me too (I'm using TCP in other places).

  I'll take another look at this with your suggestions for debugging.

  Thanks!

On Mon, Oct 26, 2009 at 11:25 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> i cant seem to reproduce it.
>
> originate sofia/internal/1235 at conference.freeswitch.org;transport=tcp 9998
>
> I get a working call and trace.
>
> Could you possibly have a dns error?? I know it's an ip but it may still
> fail if it has no dns.
>
> try
>
> sofia loglevel all 9
>
> and look for other errors.
>
>
>
>
> On Mon, Oct 26, 2009 at 9:56 AM, Kristian Kielhofner
> <kristian.kielhofner at gmail.com> wrote:
>>
>> I originally sent this last Friday but I've been unable to confirm it
>> ever made it to the list.
>>
>> Hello everyone,
>>
>> ?I'm having some issues with SIP and TCP. ?I've used it before with
>> success but I'm seeing some strange behavior...
>>
>> ?Level 7 debugs with siptrace on both profiles. ?UDP invite from
>> softphone comes in on port 5062, it's supposed to bridge to
>> 10.70.0.62. ?When configured to use UDP FS sends an INVITE (nothing
>> currently answers) while TCP doesn't send anything (confirmed with
>> siptrace and packet sniffer). ?I confirmed this behavior with a
>> gateway configured for TCP and appending ;transport=tcp to a bridge
>> line.
>>
>> ?This is trunk rev 15211 on an Intel Mac running Snow Leopard. ?I've
>> also confirmed this behavior on an Intel Linux machine running Ubuntu
>> (not sure of version ATM).
>>
>> TCP:
>>
>> http://pastebin.freeswitch.org/10825
>>
>> UDP:
>>
>> http://pastebin.freeswitch.org/10826
>>
>> dialplan (UDP):
>>
>> ? <extension name="smhpbx">
>> ? ? <condition field="destination_number" expression="^(7887)$">
>> ? ? ? <action application="set" data="call_timeout=60"/>
>> ? ? ? <action application="set" data="effective_caller_id_name=Voalte
>> Test"/>
>> ? ? ? <action application="set"
>> data="effective_caller_id_number=19412848354"/>
>> ? ? ? <action application="bridge" data="sofia/avaya/7887 at 10.70.0.62"/>
>> ? ? </condition>
>> ? </extension>
>>
>> dialplan (TCP):
>>
>> ? <extension name="smhpbx">
>> ? ? <condition field="destination_number" expression="^(7887)$">
>> ? ? ? <action application="set" data="call_timeout=60"/>
>> ? ? ? <action application="set" data="effective_caller_id_name=Voalte
>> Test"/>
>> ? ? ? <action application="set"
>> data="effective_caller_id_number=19412848354"/>
>> ? ? ? <action application="bridge"
>> data="sofia/avaya/7887 at 10.70.0.62;transport=tcp"/>
>> ? ? </condition>
>> ? </extension>
>>
>> ?Any thoughts?
>>
>> Thanks!
>>
>> --
>> Kristian Kielhofner
>> http://www.astlinux.org
>> http://blog.krisk.org
>> http://www.star2star.com
>> http://www.submityoursip.com
>> http://www.voalte.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4ae5c6c832935743011996!




From brian at freeswitch.org  Mon Oct 26 10:18:48 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 26 Oct 2009 12:18:48 -0500
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE7@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE7@EXMBXCLUS01.citservers.local>
Message-ID: <C49F867E-9993-42EB-9E48-BA442996A5C2@freeswitch.org>

You'll have to do your own load testing.  Nobody can really tell you  
exactly how many you'll get.

/b

On Oct 26, 2009, at 10:39 AM, Ujjval Karihaloo wrote:

> With the following spec for CPU and Memory can someone help me  
> guesstimating how many simultaneous calls and Calls/sec a FS server  
> can handle - Used as a Conferencing Server.




From djbinter at yahoo.com  Mon Oct 26 10:19:50 2009
From: djbinter at yahoo.com (DJB)
Date: Mon, 26 Oct 2009 10:19:50 -0700 (PDT)
Subject: [Freeswitch-users] SIP UPDATE Method
In-Reply-To: <191c3a030910260808y45952bffu7118057caba4ea4@mail.gmail.com>
References: <528227.52311.qm@web37508.mail.mud.yahoo.com>
	<191c3a030910260808y45952bffu7118057caba4ea4@mail.gmail.com>
Message-ID: <33829.77997.qm@web37506.mail.mud.yahoo.com>

I wonder whether you will consider to put it back on the next version 1.0.5 since 1.0.4 has it?  

Regards,
Dorn B.




________________________________
From: Anthony Minessale <anthony.minessale at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Mon, October 26, 2009 8:08:59 AM
Subject: Re: [Freeswitch-users] SIP UPDATE Method

It was never there before and it caused extreme havoc once we added it so we took it away again.



On Sun, Oct 25, 2009 at 12:55 AM, DJB <djbinter at yahoo.com> wrote:

I am wondering why after I update to trunk-15225, the Allow: UPDATE method is no longer there.
>
>>User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15225
>>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>
>>Am I missing something here?
>
>>Thank you.
>
>
>
>
>>_______________________________________________
>>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
pstn:213-799-1400



      
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From brian at freeswitch.org  Mon Oct 26 10:22:42 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 26 Oct 2009 12:22:42 -0500
Subject: [Freeswitch-users] Resend: Issues with SIP + TCP?
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5508953113@cooper>
References: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>
	<191c3a030910260825w25b1891g523d5e30d72ee94f@mail.gmail.com>
	<2d9149cd0910260847i718c420ek99d8d291aa37e143@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953113@cooper>
Message-ID: <DB49781A-1F3A-40CC-B357-6DDE08059FED@freeswitch.org>

Finding the exact rev that broke it would be helpful.

/b

On Oct 26, 2009, at 12:06 PM, Peter Olsson wrote:

> Hmm... I remembered incorrectly about my setup :) The Avaya PBX  
> talks TLS to the Avaya SES Server, and then UDP to FS, not TCP -  
> sorry, my bad!
>
> However, something that has changed the last 10 days seems to affect  
> my setup so it doesn't work anymore. I'll do some more SIP tracing,  
> and get back when I know more about it.
>
> /Peter




From larclap at yahoo.com  Mon Oct 26 10:37:23 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Mon, 26 Oct 2009 10:37:23 -0700
Subject: [Freeswitch-users] Setup advice on small LAN
Message-ID: <005501ca5662$fa594aa0$ef0bdfe0$@com>

I am currently running FreeSwitch very successfully, thanks to the help from
many on this list. I am new to Linux so it was a challenge.

 

FS runs on a small LAN with about 5 other computers. The connection to the
internet is DSL with 3M down and 768kb down via Covad. The ITSP is
Flowroute.

 

If one of the computers does a big download, it messes with FS in two ways.
If a connection is made, the voices are broken up, intermittent and
difficult to understand. If the download is long enough, the connection to
Flowroute is no longer usable due to registration failure.

 

Somehow I need to isolate the FS box from the rest of the LAN, or give its
traffic precedence. Covad's suggestion was to place the FS box in the DMZ.
If I have to, I'll get another DSL line and isolate it that way.

 

Is there a write-up anywhere that might help me with this problem, or
lacking that, can anyone offer advice?

 

581 2009-10-26 07:15:58.307011 [NOTICE] sofia_reg.c:333 Registering
flowroute

582 2009-10-26 07:15:59.275622 [DEBUG] sofia.c:707 nua_i_outbound: unknown
event 8: 101 NAT detected

583 2009-10-26 07:15:59.470887 [DEBUG] sofia_reg.c:1414 Changing expire time
to 2243 by request of proxy sip:sip.flowroute.com

 ...

1087 2009-10-26 07:16:00.623158 [DEBUG] mod_event_socket.c:2302 Socket up
listening on 127.0.0.1:8021

1088 2009-10-26 07:16:01.382171 [DEBUG] sofia.c:707 nua_i_outbound: unknown
event 8: 101 NAT detected

1090 2009-10-26 07:26:34.918724 [DEBUG] sofia_reg.c:1414 Changing expire
time to 1607 by request of proxy sip:sip.flowroute.c

1091 2009-10-26 07:44:56.715662 [DEBUG] sofia_reg.c:1414 Changing expire
time to 505 by request of proxy sip:sip.flowroute.co

1092 2009-10-26 07:49:52.278202 [DEBUG] sofia_reg.c:1414 Changing expire
time to 210 by request of proxy sip:sip.flowroute.co

1093 2009-10-26 07:52:30.468045 [DEBUG] sofia_reg.c:1414 Changing expire
time to 52 by request of proxy sip:sip.flowroute.com

1094 2009-10-26 07:52:49.234006 [DEBUG] sofia_reg.c:1414 Changing expire
time to 33 by request of proxy sip:sip.flowroute.com

1095 2009-10-26 07:53:00.605161 [DEBUG] sofia_reg.c:1414 Changing expire
time to 22 by request of proxy sip:sip.flowroute.com

1096 2009-10-26 07:53:12.379214 [DEBUG] sofia_reg.c:1414 Changing expire
time to 10 by request of proxy sip:sip.flowroute.com

1097 2009-10-26 07:53:19.627029 [DEBUG] sofia_reg.c:1414 Changing expire
time to 3 by request of proxy sip:sip.flowroute.com

1098 2009-10-26 07:53:20.106923 [NOTICE] sofia_reg.c:333 Registering
flowroute

1099 2009-10-26 07:53:20.454870 [DEBUG] sofia_reg.c:1414 Changing expire
time to 2 by request of proxy sip:sip.flowroute.com

1100 2009-10-26 07:53:20.705833 [NOTICE] sofia_reg.c:333 Registering
flowroute

1101 2009-10-26 07:53:20.952781 [DEBUG] sofia_reg.c:1414 Changing expire
time to 1 by request of proxy sip:sip.flowroute.com

1102 2009-10-26 07:53:21.205740 [NOTICE] sofia_reg.c:333 Registering
flowroute

1103 2009-10-26 07:53:21.705656 [NOTICE] sofia_reg.c:333 Registering
flowroute

1104 2009-10-26 07:53:22.002609 [DEBUG] sofia_reg.c:1414 Changing expire
time to 60 by request of proxy sip:sip.flowroute.com

1105 2009-10-26 09:06:17.554726 [ERR] sofia_reg.c:1425 flowroute
Registration Failed with status Operation has no matching challenge  [904].
failure #1

 

Thanks Lars

 

 

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From peter.olsson at visionutveckling.se  Mon Oct 26 10:39:01 2009
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 26 Oct 2009 18:39:01 +0100
Subject: [Freeswitch-users] Resend: Issues with SIP + TCP?
In-Reply-To: <DB49781A-1F3A-40CC-B357-6DDE08059FED@freeswitch.org>
References: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>
	<191c3a030910260825w25b1891g523d5e30d72ee94f@mail.gmail.com>
	<2d9149cd0910260847i718c420ek99d8d291aa37e143@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953113@cooper>
	<DB49781A-1F3A-40CC-B357-6DDE08059FED@freeswitch.org>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5508953116@cooper>

Yes, I know :) However, now I think this is related to the new headers introduced, it's probably not a TCP issue.

Everything seems to work just fine until the 200 OK is sent, the Avaya PBX doesn't seem to accept that reply anymore.

The only differences I've found between a working revision, and a non-working is this;

In the working 200-OK transaction method UPDATE is listed in the Allow-header, and there is a header called "X-Actually-Support: UPDATE".

In the non-working one I don't have these, and instead I have these headers;
   X-FS-Display-Name: 9099
   X-FS-Display-Number: 9099
   X-FS-Support: update_display
   P-Asserted-Identity: "9099" <9099>

So I guess this could be related to the other thread going on right now "Downloaded tar vs latest SVN - 200 OK has more headers", and not a TCP issue. It's probably the Avaya doing something wrong (not the first time), but still it seems these changes affect more systems than mine.

I'm just using it as a lab setup for now, so if anyone want me to test something, I can do it immediately.

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West
Skickat: den 26 oktober 2009 18:23
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Resend: Issues with SIP + TCP?

Finding the exact rev that broke it would be helpful.

/b

On Oct 26, 2009, at 12:06 PM, Peter Olsson wrote:

> Hmm... I remembered incorrectly about my setup :) The Avaya PBX  
> talks TLS to the Avaya SES Server, and then UDP to FS, not TCP -  
> sorry, my bad!
>
> However, something that has changed the last 10 days seems to affect  
> my setup so it doesn't work anymore. I'll do some more SIP tracing,  
> and get back when I know more about it.
>
> /Peter


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4ae5dcba32932131620271!




From brian at freeswitch.org  Mon Oct 26 10:46:24 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 26 Oct 2009 12:46:24 -0500
Subject: [Freeswitch-users] Resend: Issues with SIP + TCP?
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5508953116@cooper>
References: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>
	<191c3a030910260825w25b1891g523d5e30d72ee94f@mail.gmail.com>
	<2d9149cd0910260847i718c420ek99d8d291aa37e143@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953113@cooper>
	<DB49781A-1F3A-40CC-B357-6DDE08059FED@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953116@cooper>
Message-ID: <E5059624-512D-4FF8-8D7D-974FDAD89F04@freeswitch.org>

Bet your hardware just barfs on those like others have... I mean  
really I HATE SIP. This is stupid.

/b

On Oct 26, 2009, at 12:39 PM, Peter Olsson wrote:

> In the non-working one I don't have these, and instead I have these  
> headers;
>   X-FS-Display-Name: 9099
>   X-FS-Display-Number: 9099
>   X-FS-Support: update_display
>   P-Asserted-Identity: "9099" <9099>




From brian at freeswitch.org  Mon Oct 26 10:46:46 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 26 Oct 2009 12:46:46 -0500
Subject: [Freeswitch-users] Setup advice on small LAN
In-Reply-To: <005501ca5662$fa594aa0$ef0bdfe0$@com>
References: <005501ca5662$fa594aa0$ef0bdfe0$@com>
Message-ID: <5CB08703-5F97-451C-93C6-E41AF61A0027@freeswitch.org>

Get a dedicated DSL line.  They aren't that expensive... I have four  
of them at my house!

/b

On Oct 26, 2009, at 12:37 PM, Lars Zeb wrote:

>
> Is there a write-up anywhere that might help me with this problem,  
> or lacking that, can anyone offer advice?

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From rob4manhere at gmail.com  Mon Oct 26 10:52:44 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Mon, 26 Oct 2009 12:52:44 -0500
Subject: [Freeswitch-users] Setup advice on small LAN
In-Reply-To: <005501ca5662$fa594aa0$ef0bdfe0$@com>
References: <005501ca5662$fa594aa0$ef0bdfe0$@com>
Message-ID: <A11DEDF1-BDC0-4E84-93CE-79F00E2E18BE@gmail.com>

Get a dedicated DSL.  That'll work better than any sort of traffic  
prioritization or shaping (I've tried).

Depending on your average channel use and codec, you could probably go  
with the smallest package and be fine.

Rob

On Oct 26, 2009, at 12:37 PM, Lars Zeb wrote:

> I am currently running FreeSwitch very successfully, thanks to the  
> help from many on this list. I am new to Linux so it was a challenge.
>
> FS runs on a small LAN with about 5 other computers. The connection  
> to the internet is DSL with 3M down and 768kb down via Covad. The  
> ITSP is Flowroute.
>
> If one of the computers does a big download, it messes with FS in  
> two ways. If a connection is made, the voices are broken up,  
> intermittent and difficult to understand. If the download is long  
> enough, the connection to Flowroute is no longer usable due to  
> registration failure.
>
> Somehow I need to isolate the FS box from the rest of the LAN, or  
> give its traffic precedence. Covad?s suggestion was to place the FS  
> box in the DMZ. If I have to, I?ll get another DSL line and isolate  
> it that way.
>
> Is there a write-up anywhere that might help me with this problem,  
> or lacking that, can anyone offer advice?
>
> 581 2009-10-26 07:15:58.307011 [NOTICE] sofia_reg.c:333 Registering  
> flowroute
> 582 2009-10-26 07:15:59.275622 [DEBUG] sofia.c:707 nua_i_outbound:  
> unknown event 8: 101 NAT detected
> 583 2009-10-26 07:15:59.470887 [DEBUG] sofia_reg.c:1414 Changing  
> expire time to 2243 by request of proxy sip:sip.flowroute.com
>  ...
> 1087 2009-10-26 07:16:00.623158 [DEBUG] mod_event_socket.c:2302  
> Socket up listening on 127.0.0.1:8021
> 1088 2009-10-26 07:16:01.382171 [DEBUG] sofia.c:707 nua_i_outbound:  
> unknown event 8: 101 NAT detected
> 1090 2009-10-26 07:26:34.918724 [DEBUG] sofia_reg.c:1414 Changing  
> expire time to 1607 by request of proxy sip:sip.flowroute.c
> 1091 2009-10-26 07:44:56.715662 [DEBUG] sofia_reg.c:1414 Changing  
> expire time to 505 by request of proxy sip:sip.flowroute.co
> 1092 2009-10-26 07:49:52.278202 [DEBUG] sofia_reg.c:1414 Changing  
> expire time to 210 by request of proxy sip:sip.flowroute.co
> 1093 2009-10-26 07:52:30.468045 [DEBUG] sofia_reg.c:1414 Changing  
> expire time to 52 by request of proxy sip:sip.flowroute.com
> 1094 2009-10-26 07:52:49.234006 [DEBUG] sofia_reg.c:1414 Changing  
> expire time to 33 by request of proxy sip:sip.flowroute.com
> 1095 2009-10-26 07:53:00.605161 [DEBUG] sofia_reg.c:1414 Changing  
> expire time to 22 by request of proxy sip:sip.flowroute.com
> 1096 2009-10-26 07:53:12.379214 [DEBUG] sofia_reg.c:1414 Changing  
> expire time to 10 by request of proxy sip:sip.flowroute.com
> 1097 2009-10-26 07:53:19.627029 [DEBUG] sofia_reg.c:1414 Changing  
> expire time to 3 by request of proxy sip:sip.flowroute.com
> 1098 2009-10-26 07:53:20.106923 [NOTICE] sofia_reg.c:333 Registering  
> flowroute
> 1099 2009-10-26 07:53:20.454870 [DEBUG] sofia_reg.c:1414 Changing  
> expire time to 2 by request of proxy sip:sip.flowroute.com
> 1100 2009-10-26 07:53:20.705833 [NOTICE] sofia_reg.c:333 Registering  
> flowroute
> 1101 2009-10-26 07:53:20.952781 [DEBUG] sofia_reg.c:1414 Changing  
> expire time to 1 by request of proxy sip:sip.flowroute.com
> 1102 2009-10-26 07:53:21.205740 [NOTICE] sofia_reg.c:333 Registering  
> flowroute
> 1103 2009-10-26 07:53:21.705656 [NOTICE] sofia_reg.c:333 Registering  
> flowroute
> 1104 2009-10-26 07:53:22.002609 [DEBUG] sofia_reg.c:1414 Changing  
> expire time to 60 by request of proxy sip:sip.flowroute.com
> 1105 2009-10-26 09:06:17.554726 [ERR] sofia_reg.c:1425 flowroute  
> Registration Failed with status Operation has no matching challenge   
> [904]. failure #1
>
> Thanks Lars
>
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus  
> signature database 4545 (20091026) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From william.suffill at gmail.com  Mon Oct 26 10:57:25 2009
From: william.suffill at gmail.com (William Suffill)
Date: Mon, 26 Oct 2009 13:57:25 -0400
Subject: [Freeswitch-users] Setup advice on small LAN
In-Reply-To: <5CB08703-5F97-451C-93C6-E41AF61A0027@freeswitch.org>
References: <005501ca5662$fa594aa0$ef0bdfe0$@com>
	<5CB08703-5F97-451C-93C6-E41AF61A0027@freeswitch.org>
Message-ID: <6b65470d0910261057k21655368w13b958d31345be3d@mail.gmail.com>

4 sounds a bit overkill. Hope you got a good deal. But yes isolating
it from general internet traffic is ideal. There are other methods to
reach that goal but they get rather complex once you realize all the
things that can effect calls. =/

-- W



From shouldbeq931 at googlemail.com  Mon Oct 26 10:56:50 2009
From: shouldbeq931 at googlemail.com (shouldbe q931)
Date: Mon, 26 Oct 2009 17:56:50 +0000
Subject: [Freeswitch-users] Setup advice on small LAN
In-Reply-To: <5CB08703-5F97-451C-93C6-E41AF61A0027@freeswitch.org>
References: <005501ca5662$fa594aa0$ef0bdfe0$@com>
	<5CB08703-5F97-451C-93C6-E41AF61A0027@freeswitch.org>
Message-ID: <649eaa470910261056j7ead284fq3290d5b3ea08fc7b@mail.gmail.com>

I agree, it might be a "dirty" solution, but its so much easier than
trying to get QoS running on a DSL line, or learning how to run a
traffic classifier...

On Mon, Oct 26, 2009 at 5:46 PM, Brian West <brian at freeswitch.org> wrote:
> Get a dedicated DSL line. ?They aren't that expensive... I have four of them
> at my house!
> /b
> On Oct 26, 2009, at 12:37 PM, Lars Zeb wrote:
>
>
> Is there a write-up anywhere that might help me with this problem, or
> lacking that, can anyone offer advice?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From anthony.minessale at gmail.com  Mon Oct 26 11:24:36 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Oct 2009 13:24:36 -0500
Subject: [Freeswitch-users] Resend: Issues with SIP + TCP?
In-Reply-To: <E5059624-512D-4FF8-8D7D-974FDAD89F04@freeswitch.org>
References: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>
	<191c3a030910260825w25b1891g523d5e30d72ee94f@mail.gmail.com>
	<2d9149cd0910260847i718c420ek99d8d291aa37e143@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953113@cooper>
	<DB49781A-1F3A-40CC-B357-6DDE08059FED@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953116@cooper>
	<E5059624-512D-4FF8-8D7D-974FDAD89F04@freeswitch.org>
Message-ID: <191c3a030910261124g188d6bdene89c54cbe65a0648@mail.gmail.com>

try r15230

add the profile param

<param name="pass-callee-id" value="false"/>


On Mon, Oct 26, 2009 at 12:46 PM, Brian West <brian at freeswitch.org> wrote:

> Bet your hardware just barfs on those like others have... I mean
> really I HATE SIP. This is stupid.
>
> /b
>
> On Oct 26, 2009, at 12:39 PM, Peter Olsson wrote:
>
> > In the non-working one I don't have these, and instead I have these
> > headers;
> >   X-FS-Display-Name: 9099
> >   X-FS-Display-Number: 9099
> >   X-FS-Support: update_display
> >   P-Asserted-Identity: "9099" <9099>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Mon Oct 26 11:26:48 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Oct 2009 13:26:48 -0500
Subject: [Freeswitch-users] Downloaded tar vs latest SVN - 200 OK has
	more headers
In-Reply-To: <191c3a030910260916x47884d07t9abc9b779afbde68@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE2@EXMBXCLUS01.citservers.local>
	<6444A50B-E5F2-4CB5-8AF9-C3D8EBE67BA4@freeswitch.org>
	<2d9149cd0910260815s5dc39ec2n869b0bdbfb0333ac@mail.gmail.com>
	<191c3a030910260841i6f0f0390mc5bfeab6e5ec7ab6@mail.gmail.com>
	<65d96fc80910260905r3953ba23p346271cdbafe25a@mail.gmail.com>
	<191c3a030910260916x47884d07t9abc9b779afbde68@mail.gmail.com>
Message-ID: <191c3a030910261126l7cf12dd1n369acb79f4fc35a4@mail.gmail.com>

This is ridiculous but here it is


try r15230

add the profile param

<param name="pass-callee-id" value="false"/>



On Mon, Oct 26, 2009 at 11:16 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Thus perpetuating the wild-west of sip where you can't do anything
> according to spec because you have to worry about stupid things not keeping
> up.  Sounds like the education system where I live too.....
>
> I'll see what I can do.  It's always the other end that ppl pay for that
> drive the free stuff to change its code.
>
>
> On Mon, Oct 26, 2009 at 11:05 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>>
>>
>> On Mon, Oct 26, 2009 at 4:41 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> The headers are used to pass the callee-id info back to the other side so
>>> you have the id of who you called.
>>> The standards have failed us in this case as everything does it
>>> differently to the point that there is no standard thus we have invented our
>>> own way to carry this across from one FreeSWITCH box to another, but of
>>> course we can never make anybody happy. =/
>>>
>>>
>> I agree with you, X headers should be ignored by the equipment normally.
>> Anyhow Kristian has a point here; there will be a lot of complains because
>> of broken SIP stack on many vendor equipments
>>
>> So, can you consider some customizable a config option for such headers?
>>
>> T.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From peter.olsson at visionutveckling.se  Mon Oct 26 11:27:24 2009
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 26 Oct 2009 19:27:24 +0100
Subject: [Freeswitch-users] Resend: Issues with SIP + TCP?
In-Reply-To: <E5059624-512D-4FF8-8D7D-974FDAD89F04@freeswitch.org>
References: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>
	<191c3a030910260825w25b1891g523d5e30d72ee94f@mail.gmail.com>
	<2d9149cd0910260847i718c420ek99d8d291aa37e143@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953113@cooper>
	<DB49781A-1F3A-40CC-B357-6DDE08059FED@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953116@cooper>
	<E5059624-512D-4FF8-8D7D-974FDAD89F04@freeswitch.org>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C550895311E@cooper>

I understand your frustration :) We deal with SIP integration with about 10 different PBX vendors today, And it's always something that doesn't work as it should. Right now I don't have anything more connected to FS though.

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West
Skickat: den 26 oktober 2009 18:46
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Resend: Issues with SIP + TCP?

Bet your hardware just barfs on those like others have... I mean  
really I HATE SIP. This is stupid.

/b

On Oct 26, 2009, at 12:39 PM, Peter Olsson wrote:

> In the non-working one I don't have these, and instead I have these  
> headers;
>   X-FS-Display-Name: 9099
>   X-FS-Display-Number: 9099
>   X-FS-Support: update_display
>   P-Asserted-Identity: "9099" <9099>


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4ae5e23832938073513968!




From mike at jerris.com  Mon Oct 26 11:29:13 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 26 Oct 2009 14:29:13 -0400
Subject: [Freeswitch-users] Downloaded tar vs latest SVN - 200 OK has
	more headers
In-Reply-To: <191c3a030910260916x47884d07t9abc9b779afbde68@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE2@EXMBXCLUS01.citservers.local>
	<6444A50B-E5F2-4CB5-8AF9-C3D8EBE67BA4@freeswitch.org>
	<2d9149cd0910260815s5dc39ec2n869b0bdbfb0333ac@mail.gmail.com>
	<191c3a030910260841i6f0f0390mc5bfeab6e5ec7ab6@mail.gmail.com>
	<65d96fc80910260905r3953ba23p346271cdbafe25a@mail.gmail.com>
	<191c3a030910260916x47884d07t9abc9b779afbde68@mail.gmail.com>
Message-ID: <5EA77040-5701-4AC8-99A3-DC8AB3FBBDBB@jerris.com>

New sofia profile param as follows:

     <!-- set this param to false if your gateway for some reason  
hates X- headers that is is supposed to ignore-->
     <!--<param name="pass-callee-id" value="false"/>-->


On Oct 26, 2009, at 12:16 PM, Anthony Minessale wrote:

> Thus perpetuating the wild-west of sip where you can't do anything  
> according to spec because you have to worry about stupid things not  
> keeping up.  Sounds like the education system where I live too.....
>
> I'll see what I can do.  It's always the other end that ppl pay for  
> that drive the free stuff to change its code.
>
>
> On Mon, Oct 26, 2009 at 11:05 AM, Tihomir Culjaga  
> <tculjaga at gmail.com> wrote:
>
>
> On Mon, Oct 26, 2009 at 4:41 PM, Anthony Minessale <anthony.minessale at gmail.com 
> > wrote:
> The headers are used to pass the callee-id info back to the other  
> side so you have the id of who you called.
> The standards have failed us in this case as everything does it  
> differently to the point that there is no standard thus we have  
> invented our own way to carry this across from one FreeSWITCH box to  
> another, but of course we can never make anybody happy. =/
>
>
> I agree with you, X headers should be ignored by the equipment  
> normally. Anyhow Kristian has a point here; there will be a lot of  
> complains because of broken SIP stack on many vendor equipments
>
> So, can you consider some customizable a config option for such  
> headers?
>
> T.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From brian at freeswitch.org  Mon Oct 26 11:40:27 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 26 Oct 2009 13:40:27 -0500
Subject: [Freeswitch-users] Resend: Issues with SIP + TCP?
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C550895311E@cooper>
References: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>
	<191c3a030910260825w25b1891g523d5e30d72ee94f@mail.gmail.com>
	<2d9149cd0910260847i718c420ek99d8d291aa37e143@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953113@cooper>
	<DB49781A-1F3A-40CC-B357-6DDE08059FED@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953116@cooper>
	<E5059624-512D-4FF8-8D7D-974FDAD89F04@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C550895311E@cooper>
Message-ID: <2C225D94-CD78-4683-A886-F767566033BF@freeswitch.org>

At some point we'll have to NO NO NO fix your broken crap.  :P  The  
reason we have sip so broken now is NOBODY CAN SAY NO!... Say it with  
me... "NO!"

/b

On Oct 26, 2009, at 1:27 PM, Peter Olsson wrote:

> I understand your frustration :) We deal with SIP integration with  
> about 10 different PBX vendors today, And it's always something that  
> doesn't work as it should. Right now I don't have anything more  
> connected to FS though.
>
> /Peter




From anthony.minessale at gmail.com  Mon Oct 26 11:45:31 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Oct 2009 13:45:31 -0500
Subject: [Freeswitch-users] SIP UPDATE Method
In-Reply-To: <33829.77997.qm@web37506.mail.mud.yahoo.com>
References: <528227.52311.qm@web37508.mail.mud.yahoo.com>
	<191c3a030910260808y45952bffu7118057caba4ea4@mail.gmail.com>
	<33829.77997.qm@web37506.mail.mud.yahoo.com>
Message-ID: <191c3a030910261145q46128a92w75d73158a7f73572@mail.gmail.com>

what exactly are you expecting to use it for?
We never really supported it anyway.


On Mon, Oct 26, 2009 at 12:19 PM, DJB <djbinter at yahoo.com> wrote:

> I wonder whether you will consider to put it back on the next version 1.0.5
> since 1.0.4 has it?
>
> Regards,
> Dorn B.
>
>
> ------------------------------
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Mon, October 26, 2009 8:08:59 AM
> *Subject:* Re: [Freeswitch-users] SIP UPDATE Method
>
> It was never there before and it caused extreme havoc once we added it so
> we took it away again.
>
>
> On Sun, Oct 25, 2009 at 12:55 AM, DJB <djbinter at yahoo.com> wrote:
>
>> I am wondering why after I update to trunk-15225, the Allow: UPDATE method
>> is no longer there.
>>
>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15225
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER,
>> NOTIFY, PUBLISH, SUBSCRIBE
>>
>> Am I missing something here?
>>
>> Thank you.
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From dwhite at olp.net  Mon Oct 26 11:48:04 2009
From: dwhite at olp.net (Dan White)
Date: Mon, 26 Oct 2009 13:48:04 -0500
Subject: [Freeswitch-users] Setup advice on small LAN
In-Reply-To: <005501ca5662$fa594aa0$ef0bdfe0$@com>
References: <005501ca5662$fa594aa0$ef0bdfe0$@com>
Message-ID: <20091026184804.GA5031@dan.olp.net>

On 26/10/09?10:37?-0700, Lars Zeb wrote:
>If one of the computers does a big download, it messes with FS in two ways.
>If a connection is made, the voices are broken up, intermittent and
>difficult to understand. If the download is long enough, the connection to
>Flowroute is no longer usable due to registration failure.

QoS is a two way street. You could spend a lot of time getting your egress
traffic properly prioritized, but if your ISP does not do the same
prioritization on your ingress traffic (toward you), you'll still have
problems during downloads.

If your friendly neighborhood ISP will work with you on prioritization,
that's another matter.

-- 
Dan White



From freeswitch-users-list at metik.com  Mon Oct 26 12:25:31 2009
From: freeswitch-users-list at metik.com (Metik)
Date: Mon, 26 Oct 2009 15:25:31 -0400
Subject: [Freeswitch-users] Setup advice on small LAN
In-Reply-To: <005501ca5662$fa594aa0$ef0bdfe0$@com>
References: <005501ca5662$fa594aa0$ef0bdfe0$@com>
Message-ID: <A52116049AB24FAFB9A2E1B9A0B069D4@chaos>

Lars,

If your ISP is a COVAD partner, ask them to reprovision the line for VOA (voice optimized access).  They will provide you with two seperate VLANs (one for best effort data and the other for real-time/voice traffic).  If they are unable to do so or do not understand your request, feel free to email my off list and I can help you. The cost difference should be minimal unless you move to "second line" (a/k/a "naked", "unbundled", or "dedicated") DSL.

-metik
  ----- Original Message ----- 
  From: Lars Zeb 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, October 26, 2009 1:37 PM
  Subject: [Freeswitch-users] Setup advice on small LAN


  I am currently running FreeSwitch very successfully, thanks to the help from many on this list. I am new to Linux so it was a challenge.

   

  FS runs on a small LAN with about 5 other computers. The connection to the internet is DSL with 3M down and 768kb down via Covad. The ITSP is Flowroute.

   

  If one of the computers does a big download, it messes with FS in two ways. If a connection is made, the voices are broken up, intermittent and difficult to understand. If the download is long enough, the connection to Flowroute is no longer usable due to registration failure.

   

  Somehow I need to isolate the FS box from the rest of the LAN, or give its traffic precedence. Covad's suggestion was to place the FS box in the DMZ. If I have to, I'll get another DSL line and isolate it that way.

   

  Is there a write-up anywhere that might help me with this problem, or lacking that, can anyone offer advice?

   

  581 2009-10-26 07:15:58.307011 [NOTICE] sofia_reg.c:333 Registering flowroute

  582 2009-10-26 07:15:59.275622 [DEBUG] sofia.c:707 nua_i_outbound: unknown event 8: 101 NAT detected

  583 2009-10-26 07:15:59.470887 [DEBUG] sofia_reg.c:1414 Changing expire time to 2243 by request of proxy sip:sip.flowroute.com

   ...

  1087 2009-10-26 07:16:00.623158 [DEBUG] mod_event_socket.c:2302 Socket up listening on 127.0.0.1:8021

  1088 2009-10-26 07:16:01.382171 [DEBUG] sofia.c:707 nua_i_outbound: unknown event 8: 101 NAT detected

  1090 2009-10-26 07:26:34.918724 [DEBUG] sofia_reg.c:1414 Changing expire time to 1607 by request of proxy sip:sip.flowroute.c

  1091 2009-10-26 07:44:56.715662 [DEBUG] sofia_reg.c:1414 Changing expire time to 505 by request of proxy sip:sip.flowroute.co

  1092 2009-10-26 07:49:52.278202 [DEBUG] sofia_reg.c:1414 Changing expire time to 210 by request of proxy sip:sip.flowroute.co

  1093 2009-10-26 07:52:30.468045 [DEBUG] sofia_reg.c:1414 Changing expire time to 52 by request of proxy sip:sip.flowroute.com

  1094 2009-10-26 07:52:49.234006 [DEBUG] sofia_reg.c:1414 Changing expire time to 33 by request of proxy sip:sip.flowroute.com

  1095 2009-10-26 07:53:00.605161 [DEBUG] sofia_reg.c:1414 Changing expire time to 22 by request of proxy sip:sip.flowroute.com

  1096 2009-10-26 07:53:12.379214 [DEBUG] sofia_reg.c:1414 Changing expire time to 10 by request of proxy sip:sip.flowroute.com

  1097 2009-10-26 07:53:19.627029 [DEBUG] sofia_reg.c:1414 Changing expire time to 3 by request of proxy sip:sip.flowroute.com

  1098 2009-10-26 07:53:20.106923 [NOTICE] sofia_reg.c:333 Registering flowroute

  1099 2009-10-26 07:53:20.454870 [DEBUG] sofia_reg.c:1414 Changing expire time to 2 by request of proxy sip:sip.flowroute.com

  1100 2009-10-26 07:53:20.705833 [NOTICE] sofia_reg.c:333 Registering flowroute

  1101 2009-10-26 07:53:20.952781 [DEBUG] sofia_reg.c:1414 Changing expire time to 1 by request of proxy sip:sip.flowroute.com

  1102 2009-10-26 07:53:21.205740 [NOTICE] sofia_reg.c:333 Registering flowroute

  1103 2009-10-26 07:53:21.705656 [NOTICE] sofia_reg.c:333 Registering flowroute

  1104 2009-10-26 07:53:22.002609 [DEBUG] sofia_reg.c:1414 Changing expire time to 60 by request of proxy sip:sip.flowroute.com

  1105 2009-10-26 09:06:17.554726 [ERR] sofia_reg.c:1425 flowroute Registration Failed with status Operation has no matching challenge  [904]. failure #1

   

  Thanks Lars

   

   



  __________ Information from ESET NOD32 Antivirus, version of virus signature database 4545 (20091026) __________

  The message was checked by ESET NOD32 Antivirus.

  http://www.eset.com



------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From djbinter at yahoo.com  Mon Oct 26 12:29:30 2009
From: djbinter at yahoo.com (DJB)
Date: Mon, 26 Oct 2009 12:29:30 -0700 (PDT)
Subject: [Freeswitch-users] SIP UPDATE Method
In-Reply-To: <191c3a030910261145q46128a92w75d73158a7f73572@mail.gmail.com>
References: <528227.52311.qm@web37508.mail.mud.yahoo.com>
	<191c3a030910260808y45952bffu7118057caba4ea4@mail.gmail.com>
	<33829.77997.qm@web37506.mail.mud.yahoo.com>
	<191c3a030910261145q46128a92w75d73158a7f73572@mail.gmail.com>
Message-ID: <514887.94094.qm@web37502.mail.mud.yahoo.com>

We were trying to see whether we can adjust call duration on Session timers.  It was a question from the application developers.  I am not sure what they are trying to do exactly.  

Thank you.



________________________________
From: Anthony Minessale <anthony.minessale at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Mon, October 26, 2009 11:45:31 AM
Subject: Re: [Freeswitch-users] SIP UPDATE Method

what exactly are you expecting to use it for?
We never really supported it anyway.



On Mon, Oct 26, 2009 at 12:19 PM, DJB <djbinter at yahoo.com> wrote:

I wonder whether you will consider to put it back on the next version 1.0.5 since 1.0.4 has it?  
>
>
>Regards,
>Dorn B.
>
>
>
>
>
________________________________
From: Anthony Minessale <anthony.minessale at gmail.com>
>To: freeswitch-users at lists.freeswitch.org
>Sent: Mon, October 26, 2009 8:08:59 AM
>Subject: Re: [Freeswitch-users] SIP UPDATE Method
>
>
>It was never there before and it
> caused extreme havoc once we added it so we took it away again.
>
>
>
>On Sun, Oct 25, 2009 at 12:55 AM, DJB <djbinter at yahoo.com> wrote:
>
>I am wondering why after I update to trunk-15225, the Allow: UPDATE method is no longer there.
>>
>>>>User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15225
>>>>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>
>>>>Am I missing something here?
>>
>>>>Thank you.
>>
>>
>>
>>
>>>>_______________________________________________
>>>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>
>
>-- 
>Anthony Minessale II
>
>FreeSWITCH http://www.freeswitch.org/
>ClueCon http://www.cluecon.com/
>>
>Twitter: http://twitter.com/FreeSWITCH_wire
>
>AIM: anthm
>MSN:anthony_minessale at hotmail.com
>>GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>IRC: irc.freenode.net #freeswitch
>
>FreeSWITCH Developer Conference
>sip:888 at conference.freeswitch.org
>iax:guest at conference.freeswitch.org/888
>googletalk:conf+888 at conference.freeswitch.org
>pstn:213-799-1400
>
>
>_______________________________________________
>>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
pstn:213-799-1400



      
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From anthony.minessale at gmail.com  Mon Oct 26 12:47:08 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Oct 2009 14:47:08 -0500
Subject: [Freeswitch-users] SIP UPDATE Method
In-Reply-To: <514887.94094.qm@web37502.mail.mud.yahoo.com>
References: <528227.52311.qm@web37508.mail.mud.yahoo.com>
	<191c3a030910260808y45952bffu7118057caba4ea4@mail.gmail.com>
	<33829.77997.qm@web37506.mail.mud.yahoo.com>
	<191c3a030910261145q46128a92w75d73158a7f73572@mail.gmail.com>
	<514887.94094.qm@web37502.mail.mud.yahoo.com>
Message-ID: <191c3a030910261247n12f1917al8110cf2636f41e20@mail.gmail.com>

I think session timers will use invite if there is no update.

the session-timeout profile param should control that but you have to double
the number you actually want because it sends the new invite at the halfway
point.



On Mon, Oct 26, 2009 at 2:29 PM, DJB <djbinter at yahoo.com> wrote:

> We were trying to see whether we can adjust call duration on Session
> timers.  It was a question from the application developers.  I am not sure
> what they are trying to do exactly.
>
> Thank you.
>
> ------------------------------
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Mon, October 26, 2009 11:45:31 AM
>
> *Subject:* Re: [Freeswitch-users] SIP UPDATE Method
>
> what exactly are you expecting to use it for?
> We never really supported it anyway.
>
>
> On Mon, Oct 26, 2009 at 12:19 PM, DJB <djbinter at yahoo.com> wrote:
>
>> I wonder whether you will consider to put it back on the next version
>> 1.0.5 since 1.0.4 has it?
>>
>> Regards,
>> Dorn B.
>>
>>
>> ------------------------------
>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Sent:* Mon, October 26, 2009 8:08:59 AM
>> *Subject:* Re: [Freeswitch-users] SIP UPDATE Method
>>
>> It was never there before and it caused extreme havoc once we added it so
>> we took it away again.
>>
>>
>> On Sun, Oct 25, 2009 at 12:55 AM, DJB <djbinter at yahoo.com> wrote:
>>
>>> I am wondering why after I update to trunk-15225, the Allow: UPDATE
>>> method is no longer there.
>>>
>>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15225
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER,
>>> NOTIFY, PUBLISH, SUBSCRIBE
>>>
>>> Am I missing something here?
>>>
>>> Thank you.
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
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From anthony.minessale at gmail.com  Mon Oct 26 12:50:04 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Oct 2009 14:50:04 -0500
Subject: [Freeswitch-users] SIP UPDATE Method
In-Reply-To: <191c3a030910261247n12f1917al8110cf2636f41e20@mail.gmail.com>
References: <528227.52311.qm@web37508.mail.mud.yahoo.com>
	<191c3a030910260808y45952bffu7118057caba4ea4@mail.gmail.com>
	<33829.77997.qm@web37506.mail.mud.yahoo.com>
	<191c3a030910261145q46128a92w75d73158a7f73572@mail.gmail.com>
	<514887.94094.qm@web37502.mail.mud.yahoo.com>
	<191c3a030910261247n12f1917al8110cf2636f41e20@mail.gmail.com>
Message-ID: <191c3a030910261250q3b933692jd028f2067ae33252@mail.gmail.com>

but i think the minimum value you can set is 120

On Mon, Oct 26, 2009 at 2:47 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> I think session timers will use invite if there is no update.
>
> the session-timeout profile param should control that but you have to
> double the number you actually want because it sends the new invite at the
> halfway point.
>
>
>
>
> On Mon, Oct 26, 2009 at 2:29 PM, DJB <djbinter at yahoo.com> wrote:
>
>> We were trying to see whether we can adjust call duration on Session
>> timers.  It was a question from the application developers.  I am not sure
>> what they are trying to do exactly.
>>
>> Thank you.
>>
>> ------------------------------
>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Sent:* Mon, October 26, 2009 11:45:31 AM
>>
>> *Subject:* Re: [Freeswitch-users] SIP UPDATE Method
>>
>> what exactly are you expecting to use it for?
>> We never really supported it anyway.
>>
>>
>> On Mon, Oct 26, 2009 at 12:19 PM, DJB <djbinter at yahoo.com> wrote:
>>
>>> I wonder whether you will consider to put it back on the next version
>>> 1.0.5 since 1.0.4 has it?
>>>
>>> Regards,
>>> Dorn B.
>>>
>>>
>>> ------------------------------
>>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>>> *To:* freeswitch-users at lists.freeswitch.org
>>> *Sent:* Mon, October 26, 2009 8:08:59 AM
>>> *Subject:* Re: [Freeswitch-users] SIP UPDATE Method
>>>
>>> It was never there before and it caused extreme havoc once we added it so
>>> we took it away again.
>>>
>>>
>>> On Sun, Oct 25, 2009 at 12:55 AM, DJB <djbinter at yahoo.com> wrote:
>>>
>>>> I am wondering why after I update to trunk-15225, the Allow: UPDATE
>>>> method is no longer there.
>>>>
>>>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15225
>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
>>>> REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>>
>>>> Am I missing something here?
>>>>
>>>> Thank you.
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:213-799-1400
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
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From ujjval at simplesignal.com  Mon Oct 26 12:50:50 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Mon, 26 Oct 2009 12:50:50 -0700
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <C49F867E-9993-42EB-9E48-BA442996A5C2@freeswitch.org>
References: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE7@EXMBXCLUS01.citservers.local>,
	<C49F867E-9993-42EB-9E48-BA442996A5C2@freeswitch.org>
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71701B6DCEE@EXMBXCLUS01.citservers.local>

Are there any benchmarking test results available publicly?
________________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West [brian at freeswitch.org]
Sent: Monday, October 26, 2009 11:18 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Estimating Call Capacity

You'll have to do your own load testing.  Nobody can really tell you
exactly how many you'll get.

/b

On Oct 26, 2009, at 10:39 AM, Ujjval Karihaloo wrote:

> With the following spec for CPU and Memory can someone help me
> guesstimating how many simultaneous calls and Calls/sec a FS server
> can handle - Used as a Conferencing Server.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From anthony.minessale at gmail.com  Mon Oct 26 12:59:01 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Oct 2009 14:59:01 -0500
Subject: [Freeswitch-users] SIP UPDATE Method
In-Reply-To: <191c3a030910261250q3b933692jd028f2067ae33252@mail.gmail.com>
References: <528227.52311.qm@web37508.mail.mud.yahoo.com>
	<191c3a030910260808y45952bffu7118057caba4ea4@mail.gmail.com>
	<33829.77997.qm@web37506.mail.mud.yahoo.com>
	<191c3a030910261145q46128a92w75d73158a7f73572@mail.gmail.com>
	<514887.94094.qm@web37502.mail.mud.yahoo.com>
	<191c3a030910261247n12f1917al8110cf2636f41e20@mail.gmail.com>
	<191c3a030910261250q3b933692jd028f2067ae33252@mail.gmail.com>
Message-ID: <191c3a030910261259p3b1fcfecvbbaed9af0af4d206@mail.gmail.com>

in r15233 i put it back to the way it originally was but I may have to
remove that if it causes more problems.
We tried to handle update for display updating which was not working but the
handler for it was still in place which may have broken some automatic
behavior regarding update so I added it back to how it was originally to
determine that.


On Mon, Oct 26, 2009 at 2:50 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> but i think the minimum value you can set is 120
>
>
> On Mon, Oct 26, 2009 at 2:47 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> I think session timers will use invite if there is no update.
>>
>> the session-timeout profile param should control that but you have to
>> double the number you actually want because it sends the new invite at the
>> halfway point.
>>
>>
>>
>>
>> On Mon, Oct 26, 2009 at 2:29 PM, DJB <djbinter at yahoo.com> wrote:
>>
>>> We were trying to see whether we can adjust call duration on Session
>>> timers.  It was a question from the application developers.  I am not sure
>>> what they are trying to do exactly.
>>>
>>> Thank you.
>>>
>>> ------------------------------
>>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>>> *To:* freeswitch-users at lists.freeswitch.org
>>> *Sent:* Mon, October 26, 2009 11:45:31 AM
>>>
>>> *Subject:* Re: [Freeswitch-users] SIP UPDATE Method
>>>
>>> what exactly are you expecting to use it for?
>>> We never really supported it anyway.
>>>
>>>
>>> On Mon, Oct 26, 2009 at 12:19 PM, DJB <djbinter at yahoo.com> wrote:
>>>
>>>> I wonder whether you will consider to put it back on the next version
>>>> 1.0.5 since 1.0.4 has it?
>>>>
>>>> Regards,
>>>> Dorn B.
>>>>
>>>>
>>>> ------------------------------
>>>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>>>> *To:* freeswitch-users at lists.freeswitch.org
>>>> *Sent:* Mon, October 26, 2009 8:08:59 AM
>>>> *Subject:* Re: [Freeswitch-users] SIP UPDATE Method
>>>>
>>>> It was never there before and it caused extreme havoc once we added it
>>>> so we took it away again.
>>>>
>>>>
>>>> On Sun, Oct 25, 2009 at 12:55 AM, DJB <djbinter at yahoo.com> wrote:
>>>>
>>>>> I am wondering why after I update to trunk-15225, the Allow: UPDATE
>>>>> method is no longer there.
>>>>>
>>>>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15225
>>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER,
>>>>> REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>>>
>>>>> Am I missing something here?
>>>>>
>>>>> Thank you.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:213-799-1400
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:213-799-1400
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Mon Oct 26 13:03:59 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 26 Oct 2009 13:03:59 -0700
Subject: [Freeswitch-users] Resend: Issues with SIP + TCP?
In-Reply-To: <2C225D94-CD78-4683-A886-F767566033BF@freeswitch.org>
References: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>
	<191c3a030910260825w25b1891g523d5e30d72ee94f@mail.gmail.com>
	<2d9149cd0910260847i718c420ek99d8d291aa37e143@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953113@cooper>
	<DB49781A-1F3A-40CC-B357-6DDE08059FED@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953116@cooper>
	<E5059624-512D-4FF8-8D7D-974FDAD89F04@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C550895311E@cooper>
	<2C225D94-CD78-4683-A886-F767566033BF@freeswitch.org>
Message-ID: <87f2f3b90910261303i71d0c4afx3b154a2035d6fdd0@mail.gmail.com>

On Mon, Oct 26, 2009 at 11:40 AM, Brian West <brian at freeswitch.org> wrote:

> At some point we'll have to NO NO NO fix your broken crap.  :P  The
> reason we have sip so broken now is NOBODY CAN SAY NO!... Say it with
> me... "NO!"
>
> /b
>
> I was wondering... does anyone make a "SIP certification" program kinda
like a pen-tester except to find all the ways your SIP setup is broken? Just
curious.
-MC
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From brian at freeswitch.org  Mon Oct 26 13:13:00 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 26 Oct 2009 15:13:00 -0500
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71701B6DCEE@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE7@EXMBXCLUS01.citservers.local>,
	<C49F867E-9993-42EB-9E48-BA442996A5C2@freeswitch.org>
	<3C04B27FC880044F8FCD735D0D952FF71701B6DCEE@EXMBXCLUS01.citservers.local>
Message-ID: <2726B936-348C-40E2-B8F0-D3EB6E732C48@freeswitch.org>

I highly doubt it... You can wait for someone to post their results  
but in the end you'll have to do your own load testing because not  
everyone's numbers will jive with your use case.  Which is the reason  
the project never posts or endorses a set call count.

/b

On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote:

> Are there any benchmarking test results available publicly?
> ________________________________________




From brian at freeswitch.org  Mon Oct 26 13:18:37 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 26 Oct 2009 15:18:37 -0500
Subject: [Freeswitch-users] Resend: Issues with SIP + TCP?
In-Reply-To: <87f2f3b90910261303i71d0c4afx3b154a2035d6fdd0@mail.gmail.com>
References: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>
	<191c3a030910260825w25b1891g523d5e30d72ee94f@mail.gmail.com>
	<2d9149cd0910260847i718c420ek99d8d291aa37e143@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953113@cooper>
	<DB49781A-1F3A-40CC-B357-6DDE08059FED@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953116@cooper>
	<E5059624-512D-4FF8-8D7D-974FDAD89F04@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C550895311E@cooper>
	<2C225D94-CD78-4683-A886-F767566033BF@freeswitch.org>
	<87f2f3b90910261303i71d0c4afx3b154a2035d6fdd0@mail.gmail.com>
Message-ID: <214AD29A-7DEB-4D42-BFF7-7FDDB0901125@freeswitch.org>

You have SIPit, which was the SIP Backoff till Pillsbury got their  
panties in a wad.

/b

On Oct 26, 2009, at 3:03 PM, Michael Collins wrote:

> I was wondering... does anyone make a "SIP certification" program  
> kinda like a pen-tester except to find all the ways your SIP setup  
> is broken? Just curious.
> -MC




From msc at freeswitch.org  Mon Oct 26 13:20:00 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 26 Oct 2009 13:20:00 -0700
Subject: [Freeswitch-users] How to load user account from databse ?
In-Reply-To: <50c41b4e0910250658n74f92d01v56f1ab82087ad809@mail.gmail.com>
References: <50c41b4e0910250658n74f92d01v56f1ab82087ad809@mail.gmail.com>
Message-ID: <87f2f3b90910261320t7cce76eam167da2dcf0dbc38d@mail.gmail.com>

On Sun, Oct 25, 2009 at 6:58 AM, Lei Tang <lei.tlfly at gmail.com> wrote:

> Hi All:
>    I'm a newbie to FS.  I'm using  FS as a sbc and have about 20000 user
> account . Does somebody can tell me how to  make FS load use account
> information from a database such as mssql or mysql?  Could you give me a
> sample configuration file?
>    Thanks a lots.
>
>
Lei,

The feature that you want is mod_xml_curl - it allows you to pull config
information from a web server, and that web server will do the db lookup.
For more information check out these resources:

Wiki docs: http://wiki.freeswitch.org/wiki/Mod_xml_curl
Example:
http://fisheye.freeswitch.org/browse/FreeSWITCH/contrib/trixter/xml-curl

We don't have a soup-to-nuts how-to on mod_xml_curl because it can be done
in so many different ways. Someone could write a book on implementing
xml_curl techniques. Your best bet is to figure out how you want to store
your user information, then decide what's the best way to pull that
information from the database, then setup a web server to handle the
request/return process. Between the docs and the examples you should be able
to get up and running.

-MC
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From vinuth.madinur at gmail.com  Mon Oct 26 13:28:54 2009
From: vinuth.madinur at gmail.com (Vinuth Madinur)
Date: Tue, 27 Oct 2009 01:58:54 +0530
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <2726B936-348C-40E2-B8F0-D3EB6E732C48@freeswitch.org>
References: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE7@EXMBXCLUS01.citservers.local>
	<C49F867E-9993-42EB-9E48-BA442996A5C2@freeswitch.org>
	<3C04B27FC880044F8FCD735D0D952FF71701B6DCEE@EXMBXCLUS01.citservers.local>
	<2726B936-348C-40E2-B8F0-D3EB6E732C48@freeswitch.org>
Message-ID: <910309030910261328g4e2f51balaf73c033892c6147@mail.gmail.com>

Here are a few benchmarks that I had stumbled upon.

http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2

<http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2>
Thanks,
Vinuth.


On Tue, Oct 27, 2009 at 1:43 AM, Brian West <brian at freeswitch.org> wrote:

> I highly doubt it... You can wait for someone to post their results
> but in the end you'll have to do your own load testing because not
> everyone's numbers will jive with your use case.  Which is the reason
> the project never posts or endorses a set call count.
>
> /b
>
> On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote:
>
> > Are there any benchmarking test results available publicly?
> > ________________________________________
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kristian.kielhofner at gmail.com  Mon Oct 26 13:33:07 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Mon, 26 Oct 2009 16:33:07 -0400
Subject: [Freeswitch-users] Downloaded tar vs latest SVN - 200 OK has
	more headers
In-Reply-To: <191c3a030910261126l7cf12dd1n369acb79f4fc35a4@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE2@EXMBXCLUS01.citservers.local>
	<6444A50B-E5F2-4CB5-8AF9-C3D8EBE67BA4@freeswitch.org>
	<2d9149cd0910260815s5dc39ec2n869b0bdbfb0333ac@mail.gmail.com>
	<191c3a030910260841i6f0f0390mc5bfeab6e5ec7ab6@mail.gmail.com>
	<65d96fc80910260905r3953ba23p346271cdbafe25a@mail.gmail.com>
	<191c3a030910260916x47884d07t9abc9b779afbde68@mail.gmail.com>
	<191c3a030910261126l7cf12dd1n369acb79f4fc35a4@mail.gmail.com>
Message-ID: <2d9149cd0910261333w6129a272lfca3e6495aa9305e@mail.gmail.com>

It is ridiculous but thank you very much!

On Mon, Oct 26, 2009 at 2:26 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> This is ridiculous but here it is
>
>
> try r15230
>
> add the profile param
>
> <param name="pass-callee-id" value="false"/>
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From woof at iwoof.org  Mon Oct 26 13:47:54 2009
From: woof at iwoof.org (Andy Spitzer)
Date: Mon, 26 Oct 2009 16:47:54 -0400
Subject: [Freeswitch-users] Resend: Issues with SIP + TCP?
In-Reply-To: <87f2f3b90910261303i71d0c4afx3b154a2035d6fdd0@mail.gmail.com>
References: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>
	<191c3a030910260825w25b1891g523d5e30d72ee94f@mail.gmail.com>
	<2d9149cd0910260847i718c420ek99d8d291aa37e143@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953113@cooper>
	<DB49781A-1F3A-40CC-B357-6DDE08059FED@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953116@cooper>
	<E5059624-512D-4FF8-8D7D-974FDAD89F04@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C550895311E@cooper>
	<2C225D94-CD78-4683-A886-F767566033BF@freeswitch.org>
	<87f2f3b90910261303i71d0c4afx3b154a2035d6fdd0@mail.gmail.com>
Message-ID: <op.u2fdd2srg0dksq@woof3.fuzzy>

Woof!

On Mon, 26 Oct 2009 16:03:59 -0400, Michael Collins <msc at freeswitch.org>  
wrote:
>> I was wondering... does anyone make a "SIP certification" program kinda
> like a pen-tester except to find all the ways your SIP setup is broken?  
> Just curious.

Here is a start:
    http://interop.sipxecs.org/

It's best for testing phone implementations, but it can handle other UA's  
as well.

--Woof!



From gmaruzz at celliax.org  Mon Oct 26 13:55:45 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 26 Oct 2009 21:55:45 +0100
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <910309030910261328g4e2f51balaf73c033892c6147@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE7@EXMBXCLUS01.citservers.local>
	<C49F867E-9993-42EB-9E48-BA442996A5C2@freeswitch.org>
	<3C04B27FC880044F8FCD735D0D952FF71701B6DCEE@EXMBXCLUS01.citservers.local>
	<2726B936-348C-40E2-B8F0-D3EB6E732C48@freeswitch.org>
	<910309030910261328g4e2f51balaf73c033892c6147@mail.gmail.com>
Message-ID: <7b197bef0910261355v22840195s8cc71bd28ebd57ca@mail.gmail.com>

On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur
<vinuth.madinur at gmail.com> wrote:
> Here are a few benchmarks that I had stumbled upon.
> http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2

Please remember NO benchmarks are endorsed by the FS community or
developers, because there are just too many variables, and a simple
figure is just useful for marketing hype, not for real dimensioning.

You MUST do your own benchmarking, so you get an idea about how to
dimension for your own use case and hardware.


> Thanks,
> Vinuth.
>
> On Tue, Oct 27, 2009 at 1:43 AM, Brian West <brian at freeswitch.org> wrote:
>>
>> I highly doubt it... You can wait for someone to post their results
>> but in the end you'll have to do your own load testing because not
>> everyone's numbers will jive with your use case. ?Which is the reason
>> the project never posts or endorses a set call count.
>>
>> /b
>>
>> On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote:
>>
>> > Are there any benchmarking test results available publicly?
>> > ________________________________________
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From msc at freeswitch.org  Mon Oct 26 14:22:51 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 26 Oct 2009 14:22:51 -0700
Subject: [Freeswitch-users] "Can not record session. Media not enabled
	on channel."
In-Reply-To: <1256541677550-3890610.post@n2.nabble.com>
References: <1256034659005-3857858.post@n2.nabble.com>
	<87f2f3b90910201746h43a31b32m5c163fd98373f915@mail.gmail.com>
	<1256283391890-3877285.post@n2.nabble.com>
	<87f2f3b90910231200v71e72661id4cc9f2b7fba5610@mail.gmail.com>
	<1256541677550-3890610.post@n2.nabble.com>
Message-ID: <87f2f3b90910261422n6952fa26ya6a2a66452365146@mail.gmail.com>

On Mon, Oct 26, 2009 at 12:21 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> Yes, I can confirm - this exact error occurs each time when I start
> recording
> before the call is answered (just after sending ORIGINATE command) - but I
> think that's completely understandable that media is not ready for an
> unanswered call.
> But... is there any other event that guarantees media to be ready?
>
> Update to latest SVN and try again.
-MC
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From sjmudd at pobox.com  Mon Oct 26 14:23:42 2009
From: sjmudd at pobox.com (Simon J Mudd)
Date: Mon, 26 Oct 2009 22:23:42 +0100
Subject: [Freeswitch-users] problems registering on www.freeswitch.org
Message-ID: <20091026212342.GA28484@mad06.wl0.org>

Hi,

I tried to register on freeswitch.org using username sjmudd. It said the
username already existed so I requested having my password resent. Nothing
recieved.  I've also tried to register a new user simon.mudd with the
same results, no mail received.

I also see the following when trying to register which looks worrying:
http://merlin.wl0.org/20091026/e46494be187a53b3c27740aaff360910abd4facb.png

I'm pretty sure that I'm not filtering out the emails but can't be sure.
Could whoever maintains the web page contact me off list to help me
determine where the problem is?

Thanks,

Simon Mudd



From msc at freeswitch.org  Mon Oct 26 14:24:34 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 26 Oct 2009 14:24:34 -0700
Subject: [Freeswitch-users] Inbound DTMF Not Recognized By IVR
In-Reply-To: <573AFFBA055B4ECD83A910C12BE67129@greyhawk.tonecommander.com>
References: <573AFFBA055B4ECD83A910C12BE67129@greyhawk.tonecommander.com>
Message-ID: <87f2f3b90910261424s52b37a81ge79dc148fa9c181@mail.gmail.com>

On Fri, Oct 23, 2009 at 10:31 AM, Jerry Richards <jerry.richards at teotech.com
> wrote:

>
> I installed FS on a machine with a Sangoma A101D (PRI) card and if I make
> an
> inbound call to the FS IVR, it does not recognize DTMF digits from the PSTN
> phone.  If I call IVR from an internal phone, then it does recognize the
> DTMF digits.  I have mostly default configurations for everything.
>
> Best Regards,
> JErry
>
> I'm assuming that the DTMFs are coming in-band? If so make sure that you
issue the start_dtmf dialplan app.
-MC

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf
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From msc at freeswitch.org  Mon Oct 26 14:27:43 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 26 Oct 2009 14:27:43 -0700
Subject: [Freeswitch-users] problems registering on www.freeswitch.org
In-Reply-To: <20091026212342.GA28484@mad06.wl0.org>
References: <20091026212342.GA28484@mad06.wl0.org>
Message-ID: <87f2f3b90910261427w11a4f747r897894ea69e5fcd1@mail.gmail.com>

On Mon, Oct 26, 2009 at 2:23 PM, Simon J Mudd <sjmudd at pobox.com> wrote:

> Hi,
>
> I tried to register on freeswitch.org using username sjmudd. It said the
> username already existed so I requested having my password resent. Nothing
> recieved.  I've also tried to register a new user simon.mudd with the
> same results, no mail received.
>
> I also see the following when trying to register which looks worrying:
> http://merlin.wl0.org/20091026/e46494be187a53b3c27740aaff360910abd4facb.png
>
> I'm pretty sure that I'm not filtering out the emails but can't be sure.
> Could whoever maintains the web page contact me off list to help me
> determine where the problem is?
>
> Thanks,
>
> Simon Mudd
>
Thanks for letting us know. I'll check it out.
-MC
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From tculjaga at gmail.com  Mon Oct 26 14:28:44 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 26 Oct 2009 22:28:44 +0100
Subject: [Freeswitch-users] Downloaded tar vs latest SVN - 200 OK has
	more headers
In-Reply-To: <191c3a030910261126l7cf12dd1n369acb79f4fc35a4@mail.gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE2@EXMBXCLUS01.citservers.local>
	<6444A50B-E5F2-4CB5-8AF9-C3D8EBE67BA4@freeswitch.org>
	<2d9149cd0910260815s5dc39ec2n869b0bdbfb0333ac@mail.gmail.com>
	<191c3a030910260841i6f0f0390mc5bfeab6e5ec7ab6@mail.gmail.com>
	<65d96fc80910260905r3953ba23p346271cdbafe25a@mail.gmail.com>
	<191c3a030910260916x47884d07t9abc9b779afbde68@mail.gmail.com>
	<191c3a030910261126l7cf12dd1n369acb79f4fc35a4@mail.gmail.com>
Message-ID: <65d96fc80910261428v764eba7w5c58f5f99714c5f1@mail.gmail.com>

On Mon, Oct 26, 2009 at 7:26 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> This is ridiculous but here it is
>
>
> try r15230
>
> add the profile param
>
> <param name="pass-callee-id" value="false"/>
>
>
sorry for that but, this will save you a lot of e-mail explaining why calls
are not going through...

thanks man!

T.
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From tculjaga at gmail.com  Mon Oct 26 14:34:19 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 26 Oct 2009 22:34:19 +0100
Subject: [Freeswitch-users] Setup advice on small LAN
In-Reply-To: <A52116049AB24FAFB9A2E1B9A0B069D4@chaos>
References: <005501ca5662$fa594aa0$ef0bdfe0$@com>
	<A52116049AB24FAFB9A2E1B9A0B069D4@chaos>
Message-ID: <65d96fc80910261434n2938e812pfd62b3144d5a0559@mail.gmail.com>

>
>
> If one of the computers does a big download, it messes with FS in two ways.
> If a connection is made, the voices are broken up, intermittent and
> difficult to understand. If the download is long enough, the connection to
> Flowroute is no longer usable due to registration failure.
>
>

In any case, regardless if you are using a dedicated or mixed dsl line you
should flag your voice traffic properly.

signaling AF41, RTP EF... your voice traffic must never be flagged as pure
date when sending it through open internet!



T.
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From costa.zikalala at gmail.com  Mon Oct 26 14:39:31 2009
From: costa.zikalala at gmail.com (Costa Zikalala)
Date: Mon, 26 Oct 2009 23:39:31 +0200
Subject: [Freeswitch-users] How to load user account from databse ?
In-Reply-To: <87f2f3b90910261320t7cce76eam167da2dcf0dbc38d@mail.gmail.com>
References: <50c41b4e0910250658n74f92d01v56f1ab82087ad809@mail.gmail.com> 
	<87f2f3b90910261320t7cce76eam167da2dcf0dbc38d@mail.gmail.com>
Message-ID: <59daa2cd0910261439v414129edwbf8c98564c50cc13@mail.gmail.com>

Thanks for your response Michael,

Both the resources you've referred to don't explicitly say much about
databases. Could you elaborate just a bit on how on information would
retrieved from a database (MySQL) and presented to mod_xml_curl.

Thanks again.



2009/10/26 Michael Collins <msc at freeswitch.org>

>
>
> On Sun, Oct 25, 2009 at 6:58 AM, Lei Tang <lei.tlfly at gmail.com> wrote:
>
>> Hi All:
>>    I'm a newbie to FS.  I'm using  FS as a sbc and have about 20000 user
>> account . Does somebody can tell me how to  make FS load use account
>> information from a database such as mssql or mysql?  Could you give me a
>> sample configuration file?
>>    Thanks a lots.
>>
>>
> Lei,
>
> The feature that you want is mod_xml_curl - it allows you to pull config
> information from a web server, and that web server will do the db lookup.
> For more information check out these resources:
>
> Wiki docs: http://wiki.freeswitch.org/wiki/Mod_xml_curl
> Example:
> http://fisheye.freeswitch.org/browse/FreeSWITCH/contrib/trixter/xml-curl
>
> We don't have a soup-to-nuts how-to on mod_xml_curl because it can be done
> in so many different ways. Someone could write a book on implementing
> xml_curl techniques. Your best bet is to figure out how you want to store
> your user information, then decide what's the best way to pull that
> information from the database, then setup a web server to handle the
> request/return process. Between the docs and the examples you should be able
> to get up and running.
>
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From freeswitch-users-list at metik.com  Mon Oct 26 15:41:54 2009
From: freeswitch-users-list at metik.com (Metik)
Date: Mon, 26 Oct 2009 18:41:54 -0400
Subject: [Freeswitch-users] Setup advice on small LAN
In-Reply-To: <65d96fc80910261434n2938e812pfd62b3144d5a0559@mail.gmail.com>
References: <005501ca5662$fa594aa0$ef0bdfe0$@com><A52116049AB24FAFB9A2E1B9A0B069D4@chaos>
	<65d96fc80910261434n2938e812pfd62b3144d5a0559@mail.gmail.com>
Message-ID: <360485E934C6406FB0DE6B4ACCA2D021@chaos>

Unfortunately, most North American providers ignore (and in most cases reclassify) it before it reaches their border routers and it will be treated as best effort. Typically, the problems are introduced within the first mile and its simply a matter of getting the packet safely pass the edge unless your ISP is grossly oversubscribed (in terms of servicable traffic limited by their particular hardware and bandwidth).

If it is too costly to move to "second line" or "dedicated" DSL, he should be able to improve audio quality by acquiring a broadband router that has minimal QoS capabilities and adequate CPU (since the majority of them use software based queuing and packet fragmentation).  The only caveat is that the degree of success can vary between firmware versions.  Some of the consumer (gaming) or small business (VPN) grade routers work well (Linksys, DLINK, etc.).

-metik  


  From: Tihomir Culjaga 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, October 26, 2009 5:34 PM
  Subject: Re: [Freeswitch-users] Setup advice on small LAN




      If one of the computers does a big download, it messes with FS in two ways. If a connection is made, the voices are broken up, intermittent and difficult to understand. If the download is long enough, the connection to Flowroute is no longer usable due to registration failure.



  In any case, regardless if you are using a dedicated or mixed dsl line you should flag your voice traffic properly.

  signaling AF41, RTP EF... your voice traffic must never be flagged as pure date when sending it through open internet!



  T.



------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From egable+freeswitch at gmail.com  Mon Oct 26 15:01:29 2009
From: egable+freeswitch at gmail.com (Eliot Gable)
Date: Mon, 26 Oct 2009 18:01:29 -0400
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <bf6923ed0910261446x6a184ecdnee2e73e904406246@mail.gmail.com>
References: <bf6923ed0910261446x6a184ecdnee2e73e904406246@mail.gmail.com>
Message-ID: <bf6923ed0910261501k1ff8c892rfc198c31e30ab286@mail.gmail.com>

Although, FYI, I just benchmarked mod_xml_curl on a separate web app
server from FS with FS on a Dell R710 with their current best
processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and 32
GB memory. The web app server is less than half the power of the R710.
I maxed the web app server at 300 calls per second (both setting up
and tearing down) and the R710 running FS was 65% idle. No audio was
being proxied through FS, though. If I were running the web app server
on an equivalent R710, they probably would have been on-par with each
other in performance. Extrapolating, I expect that in such a case I
should be able to get at least 650 CPS out of FS, though for
production I would probably limit it to 400 CPS or less so I leave
room for miscellaneous tasks. I maxed out the R710 at over 16,000
simultaneous calls (again, no audio proxying) but the only reason I
couldn't do more was because I hit some sort of thread creation limit
in Linux. There was about 17 GB of memory used for this many calls.
This should give you some ballpark idea of what you can accomplish
with FS.

At some point, I will track down and resolve the thread creation
issue, at which time I believe call limits will be limited either by a
complex combination of available memory, the speed of the processor,
the cost of thread context switching, calls per second setup rate, and
call duration.

--
Eliot Gable

> -----Original Message-----
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli
>
> Sent: Monday, October 26, 2009 4:56 PM
>
> To: freeswitch-users at lists.freeswitch.org
>
> Subject: Re: [Freeswitch-users] Estimating Call Capacity
>
>
>
> On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur
>
> <vinuth.madinur at gmail.com> wrote:
>
> > Here are a few benchmarks that I had stumbled upon.
>
> > http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2
>
>
>
> Please remember NO benchmarks are endorsed by the FS community or
>
> developers, because there are just too many variables, and a simple
>
> figure is just useful for marketing hype, not for real dimensioning.
>
>
>
> You MUST do your own benchmarking, so you get an idea about how to
>
> dimension for your own use case and hardware.
>
>
>
>
>
> > Thanks,
>
> > Vinuth.
>
> >
>
> > On Tue, Oct 27, 2009 at 1:43 AM, Brian West <brian at freeswitch.org> wrote:
>
> >>
>
> >> I highly doubt it... You can wait for someone to post their results
>
> >> but in the end you'll have to do your own load testing because not
>
> >> everyone's numbers will jive with your use case.? Which is the reason
>
> >> the project never posts or endorses a set call count.
>
> >>
>
> >> /b
>
> >>
>
> >> On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote:
>
> >>
>
> >> > Are there any benchmarking test results available publicly?
>
> >> > ________________________________________
>
> >>
>
> >>
>
> >> _______________________________________________
>
> >> FreeSWITCH-users mailing list
>
> >> FreeSWITCH-users at lists.freeswitch.org
>
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> >> http://www.freeswitch.org
>
> >
>
> >
>
> > _______________________________________________
>
> > FreeSWITCH-users mailing list
>
> > FreeSWITCH-users at lists.freeswitch.org
>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> > http://www.freeswitch.org
>
> >
>
> >
>
>
>
>
>
>
>
> --
>
> Sincerely,
>
>
>
> Giovanni Maruzzelli
>
> Cell : +39-347-2665618
>
>
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org



From anthony.minessale at gmail.com  Mon Oct 26 16:03:37 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Oct 2009 18:03:37 -0500
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <bf6923ed0910261501k1ff8c892rfc198c31e30ab286@mail.gmail.com>
References: <bf6923ed0910261446x6a184ecdnee2e73e904406246@mail.gmail.com>
	<bf6923ed0910261501k1ff8c892rfc198c31e30ab286@mail.gmail.com>
Message-ID: <191c3a030910261603t6b539101p10ba7327ae9584e2@mail.gmail.com>

i wonder if we can at least get a taco-bell steak burrito for that if we
can't win the s-prize


On Mon, Oct 26, 2009 at 5:01 PM, Eliot Gable
<egable+freeswitch at gmail.com<egable%2Bfreeswitch at gmail.com>
> wrote:

> Although, FYI, I just benchmarked mod_xml_curl on a separate web app
> server from FS with FS on a Dell R710 with their current best
> processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and 32
> GB memory. The web app server is less than half the power of the R710.
> I maxed the web app server at 300 calls per second (both setting up
> and tearing down) and the R710 running FS was 65% idle. No audio was
> being proxied through FS, though. If I were running the web app server
> on an equivalent R710, they probably would have been on-par with each
> other in performance. Extrapolating, I expect that in such a case I
> should be able to get at least 650 CPS out of FS, though for
> production I would probably limit it to 400 CPS or less so I leave
> room for miscellaneous tasks. I maxed out the R710 at over 16,000
> simultaneous calls (again, no audio proxying) but the only reason I
> couldn't do more was because I hit some sort of thread creation limit
> in Linux. There was about 17 GB of memory used for this many calls.
> This should give you some ballpark idea of what you can accomplish
> with FS.
>
> At some point, I will track down and resolve the thread creation
> issue, at which time I believe call limits will be limited either by a
> complex combination of available memory, the speed of the processor,
> the cost of thread context switching, calls per second setup rate, and
> call duration.
>
> --
> Eliot Gable
>
> > -----Original Message-----
> >
> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Giovanni
> Maruzzelli
> >
> > Sent: Monday, October 26, 2009 4:56 PM
> >
> > To: freeswitch-users at lists.freeswitch.org
> >
> > Subject: Re: [Freeswitch-users] Estimating Call Capacity
> >
> >
> >
> > On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur
> >
> > <vinuth.madinur at gmail.com> wrote:
> >
> > > Here are a few benchmarks that I had stumbled upon.
> >
> > >
> http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2<http://wiki.voiceworks.pl/display/%7Epawel/FreeSwitch+performance+on+SUN+x2200+M2>
> >
> >
> >
> > Please remember NO benchmarks are endorsed by the FS community or
> >
> > developers, because there are just too many variables, and a simple
> >
> > figure is just useful for marketing hype, not for real dimensioning.
> >
> >
> >
> > You MUST do your own benchmarking, so you get an idea about how to
> >
> > dimension for your own use case and hardware.
> >
> >
> >
> >
> >
> > > Thanks,
> >
> > > Vinuth.
> >
> > >
> >
> > > On Tue, Oct 27, 2009 at 1:43 AM, Brian West <brian at freeswitch.org>
> wrote:
> >
> > >>
> >
> > >> I highly doubt it... You can wait for someone to post their results
> >
> > >> but in the end you'll have to do your own load testing because not
> >
> > >> everyone's numbers will jive with your use case.  Which is the reason
> >
> > >> the project never posts or endorses a set call count.
> >
> > >>
> >
> > >> /b
> >
> > >>
> >
> > >> On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote:
> >
> > >>
> >
> > >> > Are there any benchmarking test results available publicly?
> >
> > >> > ________________________________________
> >
> > >>
> >
> > >>
> >
> > >> _______________________________________________
> >
> > >> FreeSWITCH-users mailing list
> >
> > >> FreeSWITCH-users at lists.freeswitch.org
> >
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > >> http://www.freeswitch.org
> >
> > >
> >
> > >
> >
> > > _______________________________________________
> >
> > > FreeSWITCH-users mailing list
> >
> > > FreeSWITCH-users at lists.freeswitch.org
> >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > > http://www.freeswitch.org
> >
> > >
> >
> > >
> >
> >
> >
> >
> >
> >
> >
> > --
> >
> > Sincerely,
> >
> >
> >
> > Giovanni Maruzzelli
> >
> > Cell : +39-347-2665618
> >
> >
> >
> > _______________________________________________
> >
> > FreeSWITCH-users mailing list
> >
> > FreeSWITCH-users at lists.freeswitch.org
> >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Mon Oct 26 17:04:57 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 26 Oct 2009 17:04:57 -0700
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <191c3a030910261603t6b539101p10ba7327ae9584e2@mail.gmail.com>
References: <bf6923ed0910261446x6a184ecdnee2e73e904406246@mail.gmail.com>
	<bf6923ed0910261501k1ff8c892rfc198c31e30ab286@mail.gmail.com>
	<191c3a030910261603t6b539101p10ba7327ae9584e2@mail.gmail.com>
Message-ID: <87f2f3b90910261704n2708c573wc3ecbf80571b39f6@mail.gmail.com>

On Mon, Oct 26, 2009 at 4:03 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> i wonder if we can at least get a taco-bell steak burrito for that if we
> can't win the s-prize
>
>
>
Or at least a chalupa.
-MC
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From jmesquita at freeswitch.org  Mon Oct 26 17:08:47 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 26 Oct 2009 22:08:47 -0200
Subject: [Freeswitch-users] mod_nibblebill and memory problem
In-Reply-To: <8ccbff060910260424y7fb9f6eakbb45900838e6fd6d@mail.gmail.com>
References: <8ccbff060910260424y7fb9f6eakbb45900838e6fd6d@mail.gmail.com>
Message-ID: <ebbc2d550910261708rad012f0rddb8401e744a309f@mail.gmail.com>

Why don't you get us more information for debugging? We could use some vg
output, maybe?

JM

On Mon, Oct 26, 2009 at 9:24 AM, Dome Charoenyost <dome at tel.co.th> wrote:

> Dear All,
>             I'm running mod_nibblebill for my prepaid solution. I
> still have problem with memory. I have 4 GB RAM and runing debian
> squeeze 64 bit and 200 calls concurrent
>         Last time nibblebill running with 1 min heartbeat. when i
> check memory by htop FS user memory 2% anf growth to 60-89% in 8-9 hr.
> and then FS crash.
>         Now i change hearbeat to 0 it's mean nibble update balance
> when end of call. but everything are same FS start from 2% and growth
> to 20% in 2 days.
>        When i unload nibblebill FS running fine.
>        My question is when concurrent calls drop to 1-2 calls why FS
> (I think nibblebill) still use memory ? something wrong in nibblebill
> ?
>
> BG
> Dome C.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Mon Oct 26 17:38:04 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 26 Oct 2009 19:38:04 -0500
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <87f2f3b90910261704n2708c573wc3ecbf80571b39f6@mail.gmail.com>
References: <bf6923ed0910261446x6a184ecdnee2e73e904406246@mail.gmail.com>
	<bf6923ed0910261501k1ff8c892rfc198c31e30ab286@mail.gmail.com>
	<191c3a030910261603t6b539101p10ba7327ae9584e2@mail.gmail.com>
	<87f2f3b90910261704n2708c573wc3ecbf80571b39f6@mail.gmail.com>
Message-ID: <11489FD5-12F9-4EF3-BA2E-11ED25E02902@freeswitch.org>

yo quiero taco bell

/b

On Oct 26, 2009, at 7:04 PM, Michael Collins wrote:

> Or at least a chalupa.
> -MC




From larclap at yahoo.com  Mon Oct 26 19:29:40 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Mon, 26 Oct 2009 19:29:40 -0700
Subject: [Freeswitch-users] Problem with inbound call answered but no sound
Message-ID: <016e01ca56ad$56362380$02a26a80$@com>

I have tried to update (make current) twice since 15183. All inbound calls
are picked up but the caller hears nothing but a couple of clicks. The most
recent version I've tried is 15241.

 

Any ideas on what may be causing this? 

 

http://pastebin.freeswitch.org/10843

 

Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686
i386 GNU/Linux

 

Thanks Lars

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From brian at freeswitch.org  Mon Oct 26 19:45:42 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 26 Oct 2009 21:45:42 -0500
Subject: [Freeswitch-users] Problem with inbound call answered but no
	sound
In-Reply-To: <016e01ca56ad$56362380$02a26a80$@com>
References: <016e01ca56ad$56362380$02a26a80$@com>
Message-ID: <168575F8-6507-4AD2-B448-2D6FDA0152B6@freeswitch.org>

you're behind nat and you didn't set the ext-rtp-ip or ext-sip-ip  
correctly?

/b

On Oct 26, 2009, at 9:29 PM, Lars Zeb wrote:

> I have tried to update (make current) twice since 15183. All inbound  
> calls are picked up but the caller hears nothing but a couple of  
> clicks. The most recent version I?ve tried is 15241.
>
> Any ideas on what may be causing this?
>
> http://pastebin.freeswitch.org/10843
>
> Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686  
> i686 i386 GNU/Linux
>
> Thanks Lars

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From egable+freeswitch at gmail.com  Mon Oct 26 20:02:15 2009
From: egable+freeswitch at gmail.com (Eliot Gable)
Date: Mon, 26 Oct 2009 23:02:15 -0400
Subject: [Freeswitch-users] Problem with inbound call answered but no
	sound
In-Reply-To: <bf6923ed0910262001p69074ebk699ce7af802e0c9@mail.gmail.com>
References: <016e01ca56ad$56362380$02a26a80$@com>
	<bf6923ed0910262001p69074ebk699ce7af802e0c9@mail.gmail.com>
Message-ID: <bf6923ed0910262002s5afa3c44mc20780ac468f217c@mail.gmail.com>

FYI, it generally makes debugging easier if you do this:

sofia profile external siptrace on
sofia profile internal siptrace on

That way you can see the actual signaling and it is usually more clear
what is going on. In most cases, you will probably be able to figure
it out yourself just looking at the signaling.


 On Mon, Oct 26, 2009 at 10:29 PM, Lars Zeb <larclap at yahoo.com> wrote:
> I have tried to update (make current) twice since 15183. All inbound calls
> are picked up but the caller hears nothing but a couple of clicks. The most
> recent version I?ve tried is 15241.
>
>
>
> Any ideas on what may be causing this?
>
>
>
> http://pastebin.freeswitch.org/10843
>
>
>
> Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686
> i386 GNU/Linux
>
>
>
> Thanks Lars
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero



From larclap at yahoo.com  Mon Oct 26 20:03:53 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Mon, 26 Oct 2009 20:03:53 -0700
Subject: [Freeswitch-users] Problem with inbound call answered but
	no	sound
In-Reply-To: <168575F8-6507-4AD2-B448-2D6FDA0152B6@freeswitch.org>
References: <016e01ca56ad$56362380$02a26a80$@com>
	<168575F8-6507-4AD2-B448-2D6FDA0152B6@freeswitch.org>
Message-ID: <018801ca56b2$1e63e0f0$5b2ba2d0$@com>

I haven't changed anything since v15183, where it worked OK.

 

In conf/sip_profiles/external:

 

     <param name="ext-rtp-ip" value="auto-nat"/>

     <param name="ext-sip-ip" value="auto-nat"/>

 

And

 

 

Name                    external

Domain Name             N/A

DBName                  sofia_reg_external

Pres Hosts

Dialplan                XML

Context                 public

Challenge Realm         auto_to

RTP-IP                  192.168.10.29

Ext-RTP-IP              192.168.10.29

SIP-IP                  192.168.10.29

Ext-SIP-IP              192.168.10.29

URL                     sip:mod_sofia at 192.168.10.29:5090

BIND-URL                sip:mod_sofia at 192.168.10.29:5090

HOLD-MUSIC              local_stream://moh

OUTBOUND-PROXY          N/A

CODECS                  PCMU,PCMA,GSM

TEL-EVENT               101

DTMF-MODE               rfc2833

CNG                     13

SESSION-TO              0

MAX-DIALOG              0

NOMEDIA                 false

LATE-NEG                false

PROXY-MEDIA             false

AGGRESSIVENAT           false

STUN-ENABLED            true

STUN-AUTO-DISABLE       false

CALLS-IN                1

FAILED-CALLS-IN         1

CALLS-OUT               0

FAILED-CALLS-OUT        0

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Monday, October 26, 2009 7:46 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Problem with inbound call answered but no
sound

 

you're behind nat and you didn't set the ext-rtp-ip or ext-sip-ip correctly?

 

/b

 

On Oct 26, 2009, at 9:29 PM, Lars Zeb wrote:





I have tried to update (make current) twice since 15183. All inbound calls
are picked up but the caller hears nothing but a couple of clicks. The most
recent version I've tried is 15241.

 

Any ideas on what may be causing this?

 

http://pastebin.freeswitch.org/10843

 

Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686
i386 GNU/Linux

 

Thanks Lars

 

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From red.rain.seven at gmail.com  Mon Oct 26 20:26:55 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Tue, 27 Oct 2009 11:26:55 +0800
Subject: [Freeswitch-users] How to load user account from databse ?
In-Reply-To: <59daa2cd0910261439v414129edwbf8c98564c50cc13@mail.gmail.com>
References: <50c41b4e0910250658n74f92d01v56f1ab82087ad809@mail.gmail.com>
	<87f2f3b90910261320t7cce76eam167da2dcf0dbc38d@mail.gmail.com>
	<59daa2cd0910261439v414129edwbf8c98564c50cc13@mail.gmail.com>
Message-ID: <59ad9ca10910262026y193f1c70q1d1b9c6b82e2ee8@mail.gmail.com>

We have used xml_odbc to get user data from database.

One question though, xml_odbc will load all the user info in database to
memory just like it is loading from plain xml config files when FS starts. I
was wondering if mod_xml_curl is doing this differently by only requesting
user information on demand (real time)? and not loading all the user
information from the beginning of FS start.



On Tue, Oct 27, 2009 at 5:39 AM, Costa Zikalala <costa.zikalala at gmail.com>wrote:

> Thanks for your response Michael,
>
> Both the resources you've referred to don't explicitly say much about
> databases. Could you elaborate just a bit on how on information would
> retrieved from a database (MySQL) and presented to mod_xml_curl.
>
> Thanks again.
>
>
>
> 2009/10/26 Michael Collins <msc at freeswitch.org>
>
>>
>>
>> On Sun, Oct 25, 2009 at 6:58 AM, Lei Tang <lei.tlfly at gmail.com> wrote:
>>
>>> Hi All:
>>>    I'm a newbie to FS.  I'm using  FS as a sbc and have about 20000 user
>>> account . Does somebody can tell me how to  make FS load use account
>>> information from a database such as mssql or mysql?  Could you give me a
>>> sample configuration file?
>>>    Thanks a lots.
>>>
>>>
>> Lei,
>>
>> The feature that you want is mod_xml_curl - it allows you to pull config
>> information from a web server, and that web server will do the db lookup.
>> For more information check out these resources:
>>
>> Wiki docs: http://wiki.freeswitch.org/wiki/Mod_xml_curl
>> Example:
>> http://fisheye.freeswitch.org/browse/FreeSWITCH/contrib/trixter/xml-curl
>>
>> We don't have a soup-to-nuts how-to on mod_xml_curl because it can be done
>> in so many different ways. Someone could write a book on implementing
>> xml_curl techniques. Your best bet is to figure out how you want to store
>> your user information, then decide what's the best way to pull that
>> information from the database, then setup a web server to handle the
>> request/return process. Between the docs and the examples you should be able
>> to get up and running.
>>
>> -MC
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From red.rain.seven at gmail.com  Mon Oct 26 20:34:33 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Tue, 27 Oct 2009 11:34:33 +0800
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <11489FD5-12F9-4EF3-BA2E-11ED25E02902@freeswitch.org>
References: <bf6923ed0910261446x6a184ecdnee2e73e904406246@mail.gmail.com>
	<bf6923ed0910261501k1ff8c892rfc198c31e30ab286@mail.gmail.com>
	<191c3a030910261603t6b539101p10ba7327ae9584e2@mail.gmail.com>
	<87f2f3b90910261704n2708c573wc3ecbf80571b39f6@mail.gmail.com>
	<11489FD5-12F9-4EF3-BA2E-11ED25E02902@freeswitch.org>
Message-ID: <59ad9ca10910262034v654f84bev5690057ce7dbeb98@mail.gmail.com>

ah..... do you have to mention about taco bell that now I have moved away
from US.
~~~drooling~~~

On Tue, Oct 27, 2009 at 8:38 AM, Brian West <brian at freeswitch.org> wrote:

> yo quiero taco bell
>
> /b
>
> On Oct 26, 2009, at 7:04 PM, Michael Collins wrote:
>
> > Or at least a chalupa.
> > -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From lakindia89 at gmail.com  Mon Oct 26 22:43:05 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Tue, 27 Oct 2009 11:13:05 +0530
Subject: [Freeswitch-users] Playing Background music as well as a file
Message-ID: <7d79b3930910262243h4d7bbbb9j83717b6f4d2d766a@mail.gmail.com>

Hi all,
    I've done experimenting with the uuid_displace and mux. What mux does is
playing a file when the conversation is also happening. But I've a different
requirement.
I need to play a background music to a UUID, that will get played
continuously and also I need to play some other voice message to that uuid.

Is it possible in freeswitch? If so please guide me on how to do that!
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From lei.tlfly at gmail.com  Mon Oct 26 23:41:17 2009
From: lei.tlfly at gmail.com (Lei Tang)
Date: Tue, 27 Oct 2009 14:41:17 +0800
Subject: [Freeswitch-users] How to load user account from databse ?
In-Reply-To: <59ad9ca10910262026y193f1c70q1d1b9c6b82e2ee8@mail.gmail.com>
References: <50c41b4e0910250658n74f92d01v56f1ab82087ad809@mail.gmail.com>
	<87f2f3b90910261320t7cce76eam167da2dcf0dbc38d@mail.gmail.com>
	<59daa2cd0910261439v414129edwbf8c98564c50cc13@mail.gmail.com>
	<59ad9ca10910262026y193f1c70q1d1b9c6b82e2ee8@mail.gmail.com>
Message-ID: <50c41b4e0910262341n5c4ae984ra2b96d47c20db6a1@mail.gmail.com>

Hi  noob and Michael, thanks for your answers,  I'll try to use
mod_xml_curl.
Hi  Henry,  http://wiki.freeswitch.org/wiki/Mod_xml_curl has mentioned, FS
will post a request to webserver when it get a registration request. you can
refer to the doc for more detail.




Lei.Tang
lei.tlfly at gmail.com
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From ivan at myrvold.org  Mon Oct 26 23:46:40 2009
From: ivan at myrvold.org (Ivan C Myrvold)
Date: Tue, 27 Oct 2009 07:46:40 +0100
Subject: [Freeswitch-users] SIP provider with extern rtp server
Message-ID: <A2D2E918-B494-4179-96BA-CA3A7717846C@myrvold.org>

I have used a SIP provider for more than a year. A few days ago, he  
said he was moving to a new server, and asked me to reconfigure. I  
did, and everything seemed to work fine, until I did an outgoing call  
to an external telephone. I found out I had no audio, in neither  
direction. Incoming calls was working fine.

My provider said that the rtp is not going through the sip server, as  
it did earlier, but now through several other IP's.

Do I have to do some special configuration to handle that?

Ivan



From dome at tel.co.th  Tue Oct 27 00:07:14 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Tue, 27 Oct 2009 14:07:14 +0700
Subject: [Freeswitch-users] Playing Background music as well as a file
In-Reply-To: <7d79b3930910262243h4d7bbbb9j83717b6f4d2d766a@mail.gmail.com>
References: <7d79b3930910262243h4d7bbbb9j83717b6f4d2d766a@mail.gmail.com>
Message-ID: <8ccbff060910270007x48b9ecdahf919a4f0e266f4e5@mail.gmail.com>

2009/10/27 lakshmanan ganapathy <lakindia89 at gmail.com>:
> Hi all,
> ??? I've done experimenting with the uuid_displace and mux. What mux does is
> playing a file when the conversation is also happening. But I've a different
> requirement.

Is posible to use uuid_displace in dialplan ?  i want to do music
background base on called id

> I need to play a background music to a UUID, that will get played
> continuously and also I need to play some other voice message to that uuid.
>
> Is it possible in freeswitch? If so please guide me on how to do that!
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From red.rain.seven at gmail.com  Tue Oct 27 00:51:01 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Tue, 27 Oct 2009 15:51:01 +0800
Subject: [Freeswitch-users] How to load user account from databse ?
In-Reply-To: <50c41b4e0910262341n5c4ae984ra2b96d47c20db6a1@mail.gmail.com>
References: <50c41b4e0910250658n74f92d01v56f1ab82087ad809@mail.gmail.com>
	<87f2f3b90910261320t7cce76eam167da2dcf0dbc38d@mail.gmail.com>
	<59daa2cd0910261439v414129edwbf8c98564c50cc13@mail.gmail.com>
	<59ad9ca10910262026y193f1c70q1d1b9c6b82e2ee8@mail.gmail.com>
	<50c41b4e0910262341n5c4ae984ra2b96d47c20db6a1@mail.gmail.com>
Message-ID: <59ad9ca10910270051n256fbb2g684dc7e920b4999e@mail.gmail.com>

Sorry, guys, I need to fix my earlier statement about xml_odbc. I just
talked to my developer and xml_odbc IS loading registered user on real time,
not from the beginning of the FS start.

So Lei Tang, you can use xml_odbc as well

On Tue, Oct 27, 2009 at 2:41 PM, Lei Tang <lei.tlfly at gmail.com> wrote:

> Hi  noob and Michael, thanks for your answers,  I'll try to use
> mod_xml_curl.
> Hi  Henry,  http://wiki.freeswitch.org/wiki/Mod_xml_curl has mentioned, FS
> will post a request to webserver when it get a registration request. you can
> refer to the doc for more detail.
>
>
>
>
>
> Lei.Tang
> lei.tlfly at gmail.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From peter.olsson at visionutveckling.se  Tue Oct 27 01:20:29 2009
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 27 Oct 2009 09:20:29 +0100
Subject: [Freeswitch-users] Resend: Issues with SIP + TCP?
In-Reply-To: <2C225D94-CD78-4683-A886-F767566033BF@freeswitch.org>
References: <2d9149cd0910260756h65f9201ah2aceb8f488d834ff@mail.gmail.com>
	<191c3a030910260825w25b1891g523d5e30d72ee94f@mail.gmail.com>
	<2d9149cd0910260847i718c420ek99d8d291aa37e143@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953113@cooper>
	<DB49781A-1F3A-40CC-B357-6DDE08059FED@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C5508953116@cooper>
	<E5059624-512D-4FF8-8D7D-974FDAD89F04@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C550895311E@cooper>
	<2C225D94-CD78-4683-A886-F767566033BF@freeswitch.org>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C55089531CB@cooper>

Thanks Brian, it works now. I'll try to learn to say NO next time :)

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West
Skickat: den 26 oktober 2009 19:40
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Resend: Issues with SIP + TCP?

At some point we'll have to NO NO NO fix your broken crap.  :P  The  
reason we have sip so broken now is NOBODY CAN SAY NO!... Say it with  
me... "NO!"

/b

On Oct 26, 2009, at 1:27 PM, Peter Olsson wrote:

> I understand your frustration :) We deal with SIP integration with  
> about 10 different PBX vendors today, And it's always something that  
> doesn't work as it should. Right now I don't have anything more  
> connected to FS though.
>
> /Peter


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4ae5ef4a32936548940180!




From lei.tlfly at gmail.com  Tue Oct 27 01:49:32 2009
From: lei.tlfly at gmail.com (Lei Tang)
Date: Tue, 27 Oct 2009 16:49:32 +0800
Subject: [Freeswitch-users] how to config FS with two net interface?
Message-ID: <50c41b4e0910270149k2eaf8ee8v78ef24776947f270@mail.gmail.com>

Hi all, I run FS on a machine with two net interface, each interface has a
ip addr, one of the them connect to public network(has ip addr A), the
other  connect to a private network(has ip addr B), FS server as a SIP
server for public through A, all outbound call will bridge to a softswitch
in private network through B. here is my sofia config file and diaplan
config:

sofia internal.xml
....
<param name="rtp-ip" value="A"/>
<param name="sip-ip" value="A"/>
 ....

sofia external.xml
....
<param name="rtp-ip" value="B"/>
<param name="sip-ip" value="B"/>
....

dialplan
......
<extension name="OUTBOUND">
    <condition field="destination_number" expression="^(\d+)$">
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION"/>
        <action application="set"
data="effective_caller_id_number=xxxxxxx"/>  <!--here change the caller
number -->
        <action application="bridge"
data="sofia/external/${destination_number}@xxxxx"/>
      </condition>
    </extension>
.....

then call seq is
sipAgent --> [internal -->(bridge)-->external] -->softswith
                          FREESWITCH

the question is, when sipAgent make a outbound call, FS can't recevie the
caller's up audio stream, I traced the SIP packets, found that FS has return
addr B in SDP when ack the invite request from sipAgent, the ack packet is
===============
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
xxxxx:12208;branch=z9hG4bK-d8754z-dc750d57652c7c51-1---d8754z-;rport=12208
From: "1000" <sip:xxxx at A>;tag=cb4d3c4e
To: "65960581" <sip:xxxx at A>;tag=DtvSc0QX01yKN
Call-ID: ZTI2NmIwZGZiYzlhOGNkNTdiYWUzMzkzZTMwYzgxZjI.
CSeq: 2 INVITE
Contact: <sip:xxxxxx at B:5060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 245

v=0
o=FreeSWITCH 1256598185 1256598186 IN IP4 B   ;>>>>wrong this is the ip addr
of the adapter connect to the private network
s=FreeSWITCH
c=IN IP4 B ;>>>>wrong this is the ip addr of the adapter connect to the
private network
t=0 0
m=audio 31066 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
================
I think FS should return A in SDP, not the external binding addr (B), does
somebody known how to solve this problem?

-- 
Lei.Tang
lei.tlfly at gmail.com
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From srinivas.ksvreddy at gmail.com  Tue Oct 27 02:32:48 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 27 Oct 2009 15:02:48 +0530
Subject: [Freeswitch-users] reload from managed
Message-ID: <f8af5740910270232k689e8856sbc0dc8b668d9eacf@mail.gmail.com>

Hi,

how can i call reload from managed code? and i called
Csharp_switch_xml_open_root, this is working first time, then onwards it is
not working.
is this the correct function to call reload?

-- 
Srinivasula Reddy K
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From srinivas.ksvreddy at gmail.com  Tue Oct 27 05:48:32 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 27 Oct 2009 18:18:32 +0530
Subject: [Freeswitch-users] switch_xml_open_root
Message-ID: <f8af5740910270548n3fe7883fp18f126afd421d89e@mail.gmail.com>

**

Hi,

when i am calling  switch_xml_open_root(1,err) . i am getting this warning
message.
HEAP[FreeSwitch.exe]: Invalid Address specified to RtlFreeHeap( 00160000,
100E9FD0 ). can any know. please help me.

Thanks
Srinivasula Reddy K
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From lakindia89 at gmail.com  Tue Oct 27 06:05:49 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Tue, 27 Oct 2009 18:35:49 +0530
Subject: [Freeswitch-users] Playing Background music as well as a file
In-Reply-To: <8ccbff060910270007x48b9ecdahf919a4f0e266f4e5@mail.gmail.com>
References: <7d79b3930910262243h4d7bbbb9j83717b6f4d2d766a@mail.gmail.com>
	<8ccbff060910270007x48b9ecdahf919a4f0e266f4e5@mail.gmail.com>
Message-ID: <7d79b3930910270605s675ac0fcn3e03e08f5f9d0584@mail.gmail.com>

I tried the uuid_displace by using the Event Socket Outbound.
Not tried within dialplan, and I'm not sure whether that could be done in
dialplan itself.


On Tue, Oct 27, 2009 at 12:37 PM, Dome Charoenyost <dome at tel.co.th> wrote:

> 2009/10/27 lakshmanan ganapathy <lakindia89 at gmail.com>:
> > Hi all,
> >     I've done experimenting with the uuid_displace and mux. What mux does
> is
> > playing a file when the conversation is also happening. But I've a
> different
> > requirement.
>
> Is posible to use uuid_displace in dialplan ?  i want to do music
> background base on called id
>
> > I need to play a background music to a UUID, that will get played
> > continuously and also I need to play some other voice message to that
> uuid.
> >
> > Is it possible in freeswitch? If so please guide me on how to do that!
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From egable+freeswitch at gmail.com  Tue Oct 27 06:57:23 2009
From: egable+freeswitch at gmail.com (Eliot Gable)
Date: Tue, 27 Oct 2009 09:57:23 -0400
Subject: [Freeswitch-users] how to config FS with two net interface?
In-Reply-To: <50c41b4e0910270149k2eaf8ee8v78ef24776947f270@mail.gmail.com>
References: <50c41b4e0910270149k2eaf8ee8v78ef24776947f270@mail.gmail.com>
Message-ID: <bf6923ed0910270657n22fb6115n6314398006612e25@mail.gmail.com>

Try setting ext-rtp-ip and ext-sip-ip on both profiles.

On Tue, Oct 27, 2009 at 4:49 AM, Lei Tang <lei.tlfly at gmail.com> wrote:
> Hi all, I run FS on a machine with two net interface, each interface has a
> ip addr, one of the them connect to public network(has ip addr A), the
> other? connect to a private network(has ip addr B), FS server as a SIP
> server for public through A, all outbound call will bridge to a softswitch
> in private network through B. here is my sofia config file and diaplan
> config:
>
> sofia internal.xml
> ....
> <param name="rtp-ip" value="A"/>
> <param name="sip-ip" value="A"/>
> ?....
>
> sofia external.xml
> ....
> <param name="rtp-ip" value="B"/>
> <param name="sip-ip" value="B"/>
> ....
>
> dialplan
> ......
> <extension name="OUTBOUND">
> ??? <condition field="destination_number" expression="^(\d+)$">
> ??????? <action application="set" data="hangup_after_bridge=true"/>
> ??????? <action application="set"
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION"/>
> ??????? <action application="set"
> data="effective_caller_id_number=xxxxxxx"/>? <!--here change the caller
> number -->
> ??? ??? <action application="bridge"
> data="sofia/external/${destination_number}@xxxxx"/>
> ????? </condition>
> ??? </extension>
> .....
>
> then call seq is
> sipAgent --> [internal -->(bridge)-->external] -->softswith
> ????????????????????????? FREESWITCH
>
> the question is, when sipAgent make a outbound call, FS can't recevie the
> caller's up audio stream, I traced the SIP packets, found that FS has return
> addr B in SDP when ack the invite request from sipAgent, the ack packet is
> ===============
> SIP/2.0 183 Session Progress
> Via: SIP/2.0/UDP
> xxxxx:12208;branch=z9hG4bK-d8754z-dc750d57652c7c51-1---d8754z-;rport=12208
> From: "1000" <sip:xxxx at A>;tag=cb4d3c4e
> To: "65960581" <sip:xxxx at A>;tag=DtvSc0QX01yKN
> Call-ID: ZTI2NmIwZGZiYzlhOGNkNTdiYWUzMzkzZTMwYzgxZjI.
> CSeq: 2 INVITE
> Contact: <sip:xxxxxx at B:5060;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460
> Accept: application/sdp
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
> REFER, UPDATE, REGISTER, INFO, PUBLISH
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 245
>
> v=0
> o=FreeSWITCH 1256598185 1256598186 IN IP4 B?? ;>>>>wrong this is the ip addr
> of the adapter connect to the private network
> s=FreeSWITCH
> c=IN IP4 B ;>>>>wrong this is the ip addr of the adapter connect to the
> private network
> t=0 0
> m=audio 31066 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> ================
> I think FS should return A in SDP, not the external binding addr (B), does
> somebody known how to solve this problem?
>
> --
> Lei.Tang
> lei.tlfly at gmail.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero



From egable+freeswitch at gmail.com  Tue Oct 27 06:58:45 2009
From: egable+freeswitch at gmail.com (Eliot Gable)
Date: Tue, 27 Oct 2009 09:58:45 -0400
Subject: [Freeswitch-users] SIP provider with extern rtp server
In-Reply-To: <A2D2E918-B494-4179-96BA-CA3A7717846C@myrvold.org>
References: <A2D2E918-B494-4179-96BA-CA3A7717846C@myrvold.org>
Message-ID: <bf6923ed0910270658t61bb6687ne0b7d3ec5090e2ff@mail.gmail.com>

Make sure you let their media IPs through your firewall. Also, if you
are behind a NAT, check you have things passing to the correct
internal address.

On Tue, Oct 27, 2009 at 2:46 AM, Ivan C Myrvold <ivan at myrvold.org> wrote:
> I have used a SIP provider for more than a year. A few days ago, he
> said he was moving to a new server, and asked me to reconfigure. I
> did, and everything seemed to work fine, until I did an outgoing call
> to an external telephone. I found out I had no audio, in neither
> direction. Incoming calls was working fine.
>
> My provider said that the rtp is not going through the sip server, as
> it did earlier, but now through several other IP's.
>
> Do I have to do some special configuration to handle that?
>
> Ivan
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero



From maciej.aniserowicz at gmail.com  Tue Oct 27 07:55:34 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Tue, 27 Oct 2009 07:55:34 -0700 (PDT)
Subject: [Freeswitch-users] "Can not record session. Media not enabled
 on channel."
In-Reply-To: <87f2f3b90910261422n6952fa26ya6a2a66452365146@mail.gmail.com>
References: <1256034659005-3857858.post@n2.nabble.com>
	<87f2f3b90910201746h43a31b32m5c163fd98373f915@mail.gmail.com>
	<1256283391890-3877285.post@n2.nabble.com>
	<87f2f3b90910231200v71e72661id4cc9f2b7fba5610@mail.gmail.com>
	<1256541677550-3890610.post@n2.nabble.com>
	<87f2f3b90910261422n6952fa26ya6a2a66452365146@mail.gmail.com>
Message-ID: <1256655334215-3899478.post@n2.nabble.com>


Sorry, trunk does not compile on win7, here are the details:


rev.15247

---------------------------
Microsoft Visual C++ Debug Library
---------------------------
Debug Assertion Failed!

Program: ...ev\Projects\External\FreeSWITCH\Original\Debug\FreeSwitch.exe
File: f:\dd\vctools\crt_bld\self_x86\crt\src\dbgheap.c
Line: 1317

Expression: _CrtIsValidHeapPointer(pUserData)

For information on how your program can cause an assertion
failure, see the Visual C++ documentation on asserts.

(Press Retry to debug the application)
---------------------------
Abort   Retry   Ignore   
---------------------------

VS Call stack:

  ntdll.dll!77ccfadc()  
  [Frames below may be incorrect and/or missing, no symbols loaded for ntdll.dll] 
  ntdll.dll!77c9272c()  
  ntdll.dll!77c5e1ef()  
  msvcr90d.dll!_free_dbg_nolock(void * pUserData=0x00664b88, int nBlockUse=1)  Line 1317 + 0x9 bytes C++
  msvcr90d.dll!_free_dbg(void * pUserData=0x00664b88, int nBlockUse=1)  Line 1258 + 0xd bytes C++
  msvcr90d.dll!free(void * pUserData=0x00664b88)  Line 49 + 0xb bytes C++
> FreeSwitch.dll!switch_core_mime_add_type(const char * type=0x003bcd1c, const char * ext=0x003bcd37)  Line 748 + 0xc bytes C
  FreeSwitch.dll!load_mime_types()  Line 791 C
  FreeSwitch.dll!switch_core_init(unsigned int flags=65, switch_bool_t console=SWITCH_TRUE, const char * * err=0x003bf68c)  Line 1244 C
  FreeSwitch.dll!switch_core_init_and_modload(unsigned int flags=65, switch_bool_t console=SWITCH_TRUE, const char * * err=0x003bf68c)  Line 1454 + 0x11 bytes C
  FreeSwitch.exe!main(int argc=1, char * * argv=0x02144c40)  Line 764 + 0x23 bytes C
  FreeSwitch.exe!__tmainCRTStartup()  Line 586 + 0x19 bytes C
  FreeSwitch.exe!mainCRTStartup()  Line 403 C
  kernel32.dll!77713677()  
  ntdll.dll!77c39d72()  
  ntdll.dll!77c39d45()  


Error occurs in :

SWITCH_DECLARE(switch_status_t) switch_core_mime_add_type(const char *type, const char *ext)
{
 const char *check;
 switch_status_t status = SWITCH_STATUS_FALSE;

 switch_assert(type);
 switch_assert(ext);

 check = (const char *) switch_core_hash_find(runtime.mime_types, ext);

 if (!check) {
  char *ptype = switch_core_permanent_strdup(type);
  char *ext_list = strdup(ext);
  int argc = 0;
  char *argv[20] = { 0 };
  int x;

  switch_assert(ext_list);

  if ((argc = switch_separate_string(ext_list, ' ', argv, (sizeof(argv) / sizeof(argv[0]))))) {

   for (x = 0; x < argc; x++) {
    if (argv[x] && ptype) {
     switch_core_hash_insert(runtime.mime_types, argv[x], ptype);
    }
   }

   status = SWITCH_STATUS_SUCCESS;
  }

  free(ext_list);  // <--- HERE
 }

 return status;
}


  ----- Original Message ----- 
  From: mercutioviz [via freeswitch-users] 
  To: Maciej Aniserowicz 
  Sent: Monday, October 26, 2009 10:32 PM
  Subject: Re: [Freeswitch-users] "Can not record session. Media not enabled on channel."





  On Mon, Oct 26, 2009 at 12:21 AM, Maciej Aniserowicz <[hidden email]> wrote:


    Yes, I can confirm - this exact error occurs each time when I start recording
    before the call is answered (just after sending ORIGINATE command) - but I
    think that's completely understandable that media is not ready for an
    unanswered call.
    But... is there any other event that guarantees media to be ready?



  Update to latest SVN and try again.
  -MC



  _______________________________________________ 
  FreeSWITCH-users mailing list 
  [hidden email] 
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org



------------------------------------------------------------------------------

  View message @ http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3895104.html 
  To unsubscribe from Re: "Can not record session. Media not enabled on channel.", click here. 


-- 
View this message in context: http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3899478.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
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From rupa at rupa.com  Tue Oct 27 08:18:21 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 27 Oct 2009 10:18:21 -0500
Subject: [Freeswitch-users] reload from managed
In-Reply-To: <f8af5740910270232k689e8856sbc0dc8b668d9eacf@mail.gmail.com>
References: <f8af5740910270232k689e8856sbc0dc8b668d9eacf@mail.gmail.com>
Message-ID: <b39273530910270818j1e7f8c2eqdce44c3abdc72cc7@mail.gmail.com>

Is it necessary for you to call the reload function directly?  Why not
make an api call using switch_api_execute and the command "reloadxml".


On Tue, Oct 27, 2009 at 4:32 AM, srinivasula reddy
<srinivas.ksvreddy at gmail.com> wrote:
> Hi,
>
> how can i call reload from managed code? and i called
> Csharp_switch_xml_open_root, this is working first time, then onwards it is
> not working.
> is this the correct function to call reload?
>
> --
> Srinivasula Reddy K
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa



From anthony.minessale at gmail.com  Tue Oct 27 08:18:22 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 27 Oct 2009 10:18:22 -0500
Subject: [Freeswitch-users] "Can not record session. Media not enabled
	on channel."
In-Reply-To: <1256655334215-3899478.post@n2.nabble.com>
References: <1256034659005-3857858.post@n2.nabble.com>
	<87f2f3b90910201746h43a31b32m5c163fd98373f915@mail.gmail.com>
	<1256283391890-3877285.post@n2.nabble.com>
	<87f2f3b90910231200v71e72661id4cc9f2b7fba5610@mail.gmail.com>
	<1256541677550-3890610.post@n2.nabble.com>
	<87f2f3b90910261422n6952fa26ya6a2a66452365146@mail.gmail.com>
	<1256655334215-3899478.post@n2.nabble.com>
Message-ID: <191c3a030910270818y79f436d9h8f6311c4671502c2@mail.gmail.com>

won't compile or won't run?
maybe you should try rebuilding it.


On Tue, Oct 27, 2009 at 9:55 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

> Sorry, trunk does not compile on win7, here are the details:
>
>
> rev.15247
>
> ---------------------------
> Microsoft Visual C++ Debug Library
> ---------------------------
> Debug Assertion Failed!
>
> Program: ...ev\Projects\External\FreeSWITCH\Original\Debug\FreeSwitch.exe
> File: f:\dd\vctools\crt_bld\self_x86\crt\src\dbgheap.c
> Line: 1317
>
> Expression: _CrtIsValidHeapPointer(pUserData)
>
> For information on how your program can cause an assertion
> failure, see the Visual C++ documentation on asserts.
>
> (Press Retry to debug the application)
> ---------------------------
> Abort   Retry   Ignore
> ---------------------------
>
> VS Call stack:
>
>   ntdll.dll!77ccfadc()
>   [Frames below may be incorrect and/or missing, no symbols loaded for
> ntdll.dll]
>   ntdll.dll!77c9272c()
>   ntdll.dll!77c5e1ef()
>   msvcr90d.dll!_free_dbg_nolock(void * pUserData=0x00664b88, int
> nBlockUse=1)  Line 1317 + 0x9 bytes C++
>   msvcr90d.dll!_free_dbg(void * pUserData=0x00664b88, int nBlockUse=1)
> Line 1258 + 0xd bytes C++
>   msvcr90d.dll!free(void * pUserData=0x00664b88)  Line 49 + 0xb bytes C++
> > FreeSwitch.dll!switch_core_mime_add_type(const char * type=0x003bcd1c,
> const char * ext=0x003bcd37)  Line 748 + 0xc bytes C
>   FreeSwitch.dll!load_mime_types()  Line 791 C
>   FreeSwitch.dll!switch_core_init(unsigned int flags=65, switch_bool_t
> console=SWITCH_TRUE, const char * * err=0x003bf68c)  Line 1244 C
>   FreeSwitch.dll!switch_core_init_and_modload(unsigned int flags=65,
> switch_bool_t console=SWITCH_TRUE, const char * * err=0x003bf68c)  Line 1454
> + 0x11 bytes C
>   FreeSwitch.exe!main(int argc=1, char * * argv=0x02144c40)  Line 764 +
> 0x23 bytes C
>   FreeSwitch.exe!__tmainCRTStartup()  Line 586 + 0x19 bytes C
>   FreeSwitch.exe!mainCRTStartup()  Line 403 C
>   kernel32.dll!77713677()
>   ntdll.dll!77c39d72()
>   ntdll.dll!77c39d45()
>
>
> Error occurs in :
>
> SWITCH_DECLARE(switch_status_t) switch_core_mime_add_type(const char *type,
> const char *ext)
> {
>  const char *check;
>  switch_status_t status = SWITCH_STATUS_FALSE;
>
>  switch_assert(type);
>  switch_assert(ext);
>
>  check = (const char *) switch_core_hash_find(runtime.mime_types, ext);
>
>  if (!check) {
>   char *ptype = switch_core_permanent_strdup(type);
>   char *ext_list = strdup(ext);
>   int argc = 0;
>   char *argv[20] = { 0 };
>   int x;
>
>   switch_assert(ext_list);
>
>   if ((argc = switch_separate_string(ext_list, ' ', argv, (sizeof(argv) /
> sizeof(argv[0]))))) {
>
>    for (x = 0; x < argc; x++) {
>     if (argv[x] && ptype) {
>      switch_core_hash_insert(runtime.mime_types, argv[x], ptype);
>     }
>    }
>
>    status = SWITCH_STATUS_SUCCESS;
>   }
>
>   free(ext_list);  // <--- HERE
>  }
>
>  return status;
> }
>
>
>
> ----- Original Message -----
> *From:* [hidden email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3899478&i=0>
> *To:* [hidden email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3899478&i=1>
> *Sent:* Monday, October 26, 2009 10:32 PM
> *Subject:* Re: [Freeswitch-users] "Can not record session. Media not
> enabled on channel."
>
>
>
> On Mon, Oct 26, 2009 at 12:21 AM, Maciej Aniserowicz <[hidden email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3895104&i=0>
> > wrote:
>
>>
>> Yes, I can confirm - this exact error occurs each time when I start
>> recording
>> before the call is answered (just after sending ORIGINATE command) - but I
>> think that's completely understandable that media is not ready for an
>> unanswered call.
>> But... is there any other event that guarantees media to be ready?
>>
>> Update to latest SVN and try again.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> [hidden email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3895104&i=1>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> ------------------------------
> View this message in context: Re: [Freeswitch-users] "Can not record
> session. Media not enabled on channel."<http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3899478.html>
>
> Sent from the freeswitch-users mailing list archive<http://n2.nabble.com/freeswitch-users-f2379917.html>at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From brian at freeswitch.org  Tue Oct 27 08:26:27 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 27 Oct 2009 10:26:27 -0500
Subject: [Freeswitch-users] Problem with inbound call answered but no
	sound
In-Reply-To: <018801ca56b2$1e63e0f0$5b2ba2d0$@com>
References: <016e01ca56ad$56362380$02a26a80$@com>
	<168575F8-6507-4AD2-B448-2D6FDA0152B6@freeswitch.org>
	<018801ca56b2$1e63e0f0$5b2ba2d0$@com>
Message-ID: <BACDC899-CE4F-42E1-B2A0-C0C0E31F8787@freeswitch.org>

The issue is you're not behind nat-pmp or upnp so it can't figure out  
what your public IP is... you'll have to fill that in manually or  
enable those options on your router if possible.

/b

On Oct 26, 2009, at 10:03 PM, Lars Zeb wrote:

> I haven?t changed anything since v15183, where it worked OK.
>
> In conf/sip_profiles/external:
>
>      <param name="ext-rtp-ip" value="auto-nat"/>
>      <param name="ext-sip-ip" value="auto-nat"/>
>
> And
>

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From ivan at myrvold.org  Tue Oct 27 08:52:10 2009
From: ivan at myrvold.org (Ivan C Myrvold)
Date: Tue, 27 Oct 2009 16:52:10 +0100
Subject: [Freeswitch-users] SIP provider with extern rtp server
In-Reply-To: <bf6923ed0910270658t61bb6687ne0b7d3ec5090e2ff@mail.gmail.com>
References: <A2D2E918-B494-4179-96BA-CA3A7717846C@myrvold.org>
	<bf6923ed0910270658t61bb6687ne0b7d3ec5090e2ff@mail.gmail.com>
Message-ID: <CB8008BD-4066-4B5B-9043-214CFE938CD8@myrvold.org>

The server is on a public IP, so there is no nat issue here.

I can also see the rtp messages on wireshark starting just after the  
183 Session Progress message on the server, but just in one direction,  
coming in to the server.
So it looks like Freeswitch is stopping the rtp.
Is this because the rtp originates from another ip than the  sip  
provider ip?

Ivan

Den 27. okt. 2009 kl. 14:58 skrev Eliot Gable:

> Make sure you let their media IPs through your firewall. Also, if you
> are behind a NAT, check you have things passing to the correct
> internal address.
>
> On Tue, Oct 27, 2009 at 2:46 AM, Ivan C Myrvold <ivan at myrvold.org>  
> wrote:
>> I have used a SIP provider for more than a year. A few days ago, he
>> said he was moving to a new server, and asked me to reconfigure. I
>> did, and everything seemed to work fine, until I did an outgoing call
>> to an external telephone. I found out I had no audio, in neither
>> direction. Incoming calls was working fine.
>>
>> My provider said that the rtp is not going through the sip server, as
>> it did earlier, but now through several other IP's.
>>
>> Do I have to do some special configuration to handle that?
>>
>> Ivan
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>
>
>
> -- 
> Eliot Gable
>
> "We do not inherit the Earth from our ancestors: we borrow it from our
> children." ~David Brower
>
> "I decided the words were too conservative for me. We're not borrowing
> from our children, we're stealing from them--and it's not even
> considered to be a crime." ~David Brower
>
> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
> live; not live to eat.) ~Marcus Tullius Cicero
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>




From larclap at yahoo.com  Tue Oct 27 08:58:25 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Tue, 27 Oct 2009 08:58:25 -0700
Subject: [Freeswitch-users] Problem with inbound call answered but
	no	sound
In-Reply-To: <BACDC899-CE4F-42E1-B2A0-C0C0E31F8787@freeswitch.org>
References: <016e01ca56ad$56362380$02a26a80$@com>	<168575F8-6507-4AD2-B448-2D6FDA0152B6@freeswitch.org>	<018801ca56b2$1e63e0f0$5b2ba2d0$@com>
	<BACDC899-CE4F-42E1-B2A0-C0C0E31F8787@freeswitch.org>
Message-ID: <00d601ca571e$519f7300$f4de5900$@com>

Thanks for the reply, Brian.

 

Did something in FS change between v15183 and v15225 to make this occur? I
ask because this same configuration worked OK in the earlier version.

 

Lars

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Tuesday, October 27, 2009 8:26 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Problem with inbound call answered but no
sound

 

The issue is you're not behind nat-pmp or upnp so it can't figure out what
your public IP is... you'll have to fill that in manually or enable those
options on your router if possible.

 

/b

 

On Oct 26, 2009, at 10:03 PM, Lars Zeb wrote:





I haven't changed anything since v15183, where it worked OK.

 

In conf/sip_profiles/external:

 

     <param name="ext-rtp-ip" value="auto-nat"/>

     <param name="ext-sip-ip" value="auto-nat"/>

 

And

 

 

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From brian at freeswitch.org  Tue Oct 27 09:20:40 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 27 Oct 2009 11:20:40 -0500
Subject: [Freeswitch-users] Problem with inbound call answered but no
	sound
In-Reply-To: <00d601ca571e$519f7300$f4de5900$@com>
References: <016e01ca56ad$56362380$02a26a80$@com>	<168575F8-6507-4AD2-B448-2D6FDA0152B6@freeswitch.org>	<018801ca56b2$1e63e0f0$5b2ba2d0$@com>
	<BACDC899-CE4F-42E1-B2A0-C0C0E31F8787@freeswitch.org>
	<00d601ca571e$519f7300$f4de5900$@com>
Message-ID: <6D6D0E8C-84D0-4160-8457-3B6045DFDBC6@freeswitch.org>

Let me review that and see I can't off hand think of anything that  
would cause that.
/b

On Oct 27, 2009, at 10:58 AM, Lars Zeb wrote:

> Thanks for the reply, Brian.
>
> Did something in FS change between v15183 and v15225 to make this  
> occur? I ask because this same configuration worked OK in the  
> earlier version.
>
> Lars

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From egable+freeswitch at gmail.com  Tue Oct 27 10:34:29 2009
From: egable+freeswitch at gmail.com (Eliot Gable)
Date: Tue, 27 Oct 2009 13:34:29 -0400
Subject: [Freeswitch-users] SIP provider with extern rtp server
In-Reply-To: <CB8008BD-4066-4B5B-9043-214CFE938CD8@myrvold.org>
References: <A2D2E918-B494-4179-96BA-CA3A7717846C@myrvold.org>
	<bf6923ed0910270658t61bb6687ne0b7d3ec5090e2ff@mail.gmail.com>
	<CB8008BD-4066-4B5B-9043-214CFE938CD8@myrvold.org>
Message-ID: <bf6923ed0910271034w6c7e0cc5t685a5377764755e6@mail.gmail.com>

No, the IP address the media originates from does not need to be tied
to the SIP IP address. Can you send a Wireshark capture taken on the
FreeSWITCH server of both call legs? Or, if you can, pastebin a debug
log from FreeSWITCH console with sofia loglevel set to 9 and siptrace
on for any Sofia SIP profiles involved.

On Tue, Oct 27, 2009 at 11:52 AM, Ivan C Myrvold <ivan at myrvold.org> wrote:
> The server is on a public IP, so there is no nat issue here.
>
> I can also see the rtp messages on wireshark starting just after the
> 183 Session Progress message on the server, but just in one direction,
> coming in to the server.
> So it looks like Freeswitch is stopping the rtp.
> Is this because the rtp originates from another ip than the ?sip
> provider ip?
>
> Ivan
>
> Den 27. okt. 2009 kl. 14:58 skrev Eliot Gable:
>
>> Make sure you let their media IPs through your firewall. Also, if you
>> are behind a NAT, check you have things passing to the correct
>> internal address.
>>
>> On Tue, Oct 27, 2009 at 2:46 AM, Ivan C Myrvold <ivan at myrvold.org>
>> wrote:
>>> I have used a SIP provider for more than a year. A few days ago, he
>>> said he was moving to a new server, and asked me to reconfigure. I
>>> did, and everything seemed to work fine, until I did an outgoing call
>>> to an external telephone. I found out I had no audio, in neither
>>> direction. Incoming calls was working fine.
>>>
>>> My provider said that the rtp is not going through the sip server, as
>>> it did earlier, but now through several other IP's.
>>>
>>> Do I have to do some special configuration to handle that?
>>>
>>> Ivan
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Eliot Gable
>>
>> "We do not inherit the Earth from our ancestors: we borrow it from our
>> children." ~David Brower
>>
>> "I decided the words were too conservative for me. We're not borrowing
>> from our children, we're stealing from them--and it's not even
>> considered to be a crime." ~David Brower
>>
>> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
>> live; not live to eat.) ~Marcus Tullius Cicero
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero



From larclap at yahoo.com  Tue Oct 27 10:33:18 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Tue, 27 Oct 2009 10:33:18 -0700
Subject: [Freeswitch-users] Forcing endpoint registration
Message-ID: <010201ca572b$984e02a0$c8ea07e0$@com>

Is the following the correct command to force registration of an endpoint
below which is not showing up in 'sofia status profile internal' currently,
but when it does it looks like the following?

 

Call-ID:        mjkg8c3zlqwjzfd.hrt5

User:           1019 at 192.168.10.29

Contact:        "user" <sip:1019 at 192.168.10.103;line=3852>

Agent:          snom-m3-SIP/02.02 (MAC=0004132A31C7; HW=1)

Status:         Registered(UDP)(unknown) EXP(2009-10-27 10:46:07)

Host:           fs

IP:             192.168.10.103

Port:           5060

Auth-User:      1019

Auth-Realm:     192.168.10.29

MWI-Account:    1019 at 192.168.10.29

 

sofia profile internal  flush_inbound_reg  <mailto:1019 at 192.168.10.29>
1019 at 192.168.10.29

 

Thanks Lars

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From msc at freeswitch.org  Tue Oct 27 10:56:09 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 27 Oct 2009 10:56:09 -0700
Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Agenda - Oct
	30th
Message-ID: <87f2f3b90910271056v672b60ebybc34cb040760dfc2@mail.gmail.com>

Greetings all!

I just wanted to say thank you to those who have been contributing to the
documentation and cleanup efforts. Please keep up the good work. FYI, we
still have lots of janitorial stuff to do. Just keep checking the latest
conf call agenda for updates. This week's agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_30

We are always looking for people to help out with documentation and other
little projects so please feel free to ask me if you would like to assist.
Also, we are working on having members of the community give brief
presentations during the weekly conference calls. We want people to join the
conference and share with the group information about how they're using
FreeSWITCH in production, or they can give little tutorials on how to use
various parts of FS. We are also working on getting a web-cast enabled so
that we can have a little video to go with the audio. If you have something
you'd like to discuss with group please let me know.

Thanks for being such a great community!
-Michael
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From brian at freeswitch.org  Tue Oct 27 11:06:09 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 27 Oct 2009 13:06:09 -0500
Subject: [Freeswitch-users] Forcing endpoint registration
In-Reply-To: <010201ca572b$984e02a0$c8ea07e0$@com>
References: <010201ca572b$984e02a0$c8ea07e0$@com>
Message-ID: <D87B950A-EB02-4CA1-A6E7-E8570BAE49C4@freeswitch.org>

 From FreeSWITCH you can't force and endpoint to register... thats the  
endpoints job... what you're doing is just flushing the reg out of  
FreeSWITCH so now it can't find the endpoint...  Try adding a reboot  
on there maybe i'll reboot the device and cause it to register again...

/b

On Oct 27, 2009, at 12:33 PM, Lars Zeb wrote:

> Is the following the correct command to force registration of an  
> endpoint below which is not showing up in ?sofia status profile  
> internal? currently, but when it does it looks like the following?
>
> Call-ID:        mjkg8c3zlqwjzfd.hrt5
> User:           1019 at 192.168.10.29
> Contact:        "user" <sip:1019 at 192.168.10.103;line=3852>
> Agent:          snom-m3-SIP/02.02 (MAC=0004132A31C7; HW=1)
> Status:         Registered(UDP)(unknown) EXP(2009-10-27 10:46:07)
> Host:           fs
> IP:             192.168.10.103
> Port:           5060
> Auth-User:      1019
> Auth-Realm:     192.168.10.29
> MWI-Account:    1019 at 192.168.10.29
>
> sofia profile internal  flush_inbound_reg 1019 at 192.168.10.29
>
> Thanks Lars

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From jerry.richards at teotech.com  Tue Oct 27 12:17:07 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 27 Oct 2009 12:17:07 -0700
Subject: [Freeswitch-users] FS Training
Message-ID: <5B56DAA56F1F43EC973C226E8CD9BCFC@greyhawk.tonecommander.com>


Did the voting booth close?  I was unable to vote.  I'm not sure what link
to click and I have had some strange issues with my FS account today.

I would be interested in paid training.  Do you have plans for offering a
training session at your locale?  Or would you travel onsite to provide
training?

Best Regards,
Jerry




From cliff at develix.com  Tue Oct 27 12:51:50 2009
From: cliff at develix.com (Cliff Wells)
Date: Tue, 27 Oct 2009 12:51:50 -0700
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <bf6923ed0910261501k1ff8c892rfc198c31e30ab286@mail.gmail.com>
References: <bf6923ed0910261446x6a184ecdnee2e73e904406246@mail.gmail.com>
	<bf6923ed0910261501k1ff8c892rfc198c31e30ab286@mail.gmail.com>
Message-ID: <1256673110.9746.29.camel@portable-evil>

A little off-topic, but since call-capacity is the subject, what are
people using to analyze their CDR's to discover this?   I'm handling
about 30k calls per day but have only a bandwidth-based guesstimate of
the peak number of concurrent calls I'm handling.

If there's an open source solution, I'd appreciate a pointer.

Regards,
Cliff

On Mon, 2009-10-26 at 18:01 -0400, Eliot Gable wrote:
> Although, FYI, I just benchmarked mod_xml_curl on a separate web app
> server from FS with FS on a Dell R710 with their current best
> processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and 32
> GB memory. The web app server is less than half the power of the R710.
> I maxed the web app server at 300 calls per second (both setting up
> and tearing down) and the R710 running FS was 65% idle. No audio was
> being proxied through FS, though. If I were running the web app server
> on an equivalent R710, they probably would have been on-par with each
> other in performance. Extrapolating, I expect that in such a case I
> should be able to get at least 650 CPS out of FS, though for
> production I would probably limit it to 400 CPS or less so I leave
> room for miscellaneous tasks. I maxed out the R710 at over 16,000
> simultaneous calls (again, no audio proxying) but the only reason I
> couldn't do more was because I hit some sort of thread creation limit
> in Linux. There was about 17 GB of memory used for this many calls.
> This should give you some ballpark idea of what you can accomplish
> with FS.
> 
> At some point, I will track down and resolve the thread creation
> issue, at which time I believe call limits will be limited either by a
> complex combination of available memory, the speed of the processor,
> the cost of thread context switching, calls per second setup rate, and
> call duration.
> 
> --
> Eliot Gable
> 
> > -----Original Message-----
> >
> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli
> >
> > Sent: Monday, October 26, 2009 4:56 PM
> >
> > To: freeswitch-users at lists.freeswitch.org
> >
> > Subject: Re: [Freeswitch-users] Estimating Call Capacity
> >
> >
> >
> > On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur
> >
> > <vinuth.madinur at gmail.com> wrote:
> >
> > > Here are a few benchmarks that I had stumbled upon.
> >
> > > http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2
> >
> >
> >
> > Please remember NO benchmarks are endorsed by the FS community or
> >
> > developers, because there are just too many variables, and a simple
> >
> > figure is just useful for marketing hype, not for real dimensioning.
> >
> >
> >
> > You MUST do your own benchmarking, so you get an idea about how to
> >
> > dimension for your own use case and hardware.
> >
> >
> >
> >
> >
> > > Thanks,
> >
> > > Vinuth.
> >
> > >
> >
> > > On Tue, Oct 27, 2009 at 1:43 AM, Brian West <brian at freeswitch.org> wrote:
> >
> > >>
> >
> > >> I highly doubt it... You can wait for someone to post their results
> >
> > >> but in the end you'll have to do your own load testing because not
> >
> > >> everyone's numbers will jive with your use case.  Which is the reason
> >
> > >> the project never posts or endorses a set call count.
> >
> > >>
> >
> > >> /b
> >
> > >>
> >
> > >> On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote:
> >
> > >>
> >
> > >> > Are there any benchmarking test results available publicly?
> >
> > >> > ________________________________________
> >
> > >>
> >
> > >>
> >
> > >> _______________________________________________
> >
> > >> FreeSWITCH-users mailing list
> >
> > >> FreeSWITCH-users at lists.freeswitch.org
> >
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > >> http://www.freeswitch.org
> >
> > >
> >
> > >
> >
> > > _______________________________________________
> >
> > > FreeSWITCH-users mailing list
> >
> > > FreeSWITCH-users at lists.freeswitch.org
> >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > > http://www.freeswitch.org
> >
> > >
> >
> > >
> >
> >
> >
> >
> >
> >
> >
> > --
> >
> > Sincerely,
> >
> >
> >
> > Giovanni Maruzzelli
> >
> > Cell : +39-347-2665618
> >
> >
> >
> > _______________________________________________
> >
> > FreeSWITCH-users mailing list
> >
> > FreeSWITCH-users at lists.freeswitch.org
> >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 
http://www.google.com/search?q=vonage+sucks




From damin at nacs.net  Tue Oct 27 13:06:32 2009
From: damin at nacs.net (Gregory Boehnlein)
Date: Tue, 27 Oct 2009 16:06:32 -0400
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <1256673110.9746.29.camel@portable-evil>
References: <bf6923ed0910261446x6a184ecdnee2e73e904406246@mail.gmail.com>	<bf6923ed0910261501k1ff8c892rfc198c31e30ab286@mail.gmail.com>
	<1256673110.9746.29.camel@portable-evil>
Message-ID: <000f01ca5740$fd9d7a40$f8d86ec0$@net>

I'm fond of Vqmanager from ManagEngine. It is a passive SIP monitor. I.E.
you mirror the ports that your FS or Asterisk boxes, and VQmanager sniffs
the mirrors, tracking all sorts of good data.

You can install it on a Centos box, and get a free trial.

http://www.manageengine.com/products/vqmanager/index.html

What is really cool is that it actually monitors the RTP/RTCP as well as all
of the SIP headers and archives the calls, so you can look at calls from
several days ago and see EXACTLY what happened on them. I have used this
extensively to pinpoint bad Level 3 and X/O media gateways.. Much better
than trying to sniff packets in real-time and MAYBE catch a problem..

I've also used it to find/fix several SIP issues w/ odd endpoints.. Very
easy to see..

> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> users-bounces at lists.freeswitch.org] On Behalf Of Cliff Wells
> Sent: Tuesday, October 27, 2009 3:52 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Estimating Call Capacity
> 
> A little off-topic, but since call-capacity is the subject, what are
> people using to analyze their CDR's to discover this?   I'm handling
> about 30k calls per day but have only a bandwidth-based guesstimate of
> the peak number of concurrent calls I'm handling.
> 
> If there's an open source solution, I'd appreciate a pointer.
> 
> Regards,
> Cliff
> 
> On Mon, 2009-10-26 at 18:01 -0400, Eliot Gable wrote:
> > Although, FYI, I just benchmarked mod_xml_curl on a separate web app
> > server from FS with FS on a Dell R710 with their current best
> > processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and
> 32
> > GB memory. The web app server is less than half the power of the
> R710.
> > I maxed the web app server at 300 calls per second (both setting up
> > and tearing down) and the R710 running FS was 65% idle. No audio was
> > being proxied through FS, though. If I were running the web app
> server
> > on an equivalent R710, they probably would have been on-par with each
> > other in performance. Extrapolating, I expect that in such a case I
> > should be able to get at least 650 CPS out of FS, though for
> > production I would probably limit it to 400 CPS or less so I leave
> > room for miscellaneous tasks. I maxed out the R710 at over 16,000
> > simultaneous calls (again, no audio proxying) but the only reason I
> > couldn't do more was because I hit some sort of thread creation limit
> > in Linux. There was about 17 GB of memory used for this many calls.
> > This should give you some ballpark idea of what you can accomplish
> > with FS.
> >
> > At some point, I will track down and resolve the thread creation
> > issue, at which time I believe call limits will be limited either by
> a
> > complex combination of available memory, the speed of the processor,
> > the cost of thread context switching, calls per second setup rate,
> and
> > call duration.
> >
> > --
> > Eliot Gable
> >
> > > -----Original Message-----
> > >
> > > From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Giovanni Maruzzelli
> > >
> > > Sent: Monday, October 26, 2009 4:56 PM
> > >
> > > To: freeswitch-users at lists.freeswitch.org
> > >
> > > Subject: Re: [Freeswitch-users] Estimating Call Capacity
> > >
> > >
> > >
> > > On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur
> > >
> > > <vinuth.madinur at gmail.com> wrote:
> > >
> > > > Here are a few benchmarks that I had stumbled upon.
> > >
> > > >
> http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+
> x2200+M2
> > >
> > >
> > >
> > > Please remember NO benchmarks are endorsed by the FS community or
> > >
> > > developers, because there are just too many variables, and a simple
> > >
> > > figure is just useful for marketing hype, not for real
> dimensioning.
> > >
> > >
> > >
> > > You MUST do your own benchmarking, so you get an idea about how to
> > >
> > > dimension for your own use case and hardware.
> > >
> > >
> > >
> > >
> > >
> > > > Thanks,
> > >
> > > > Vinuth.
> > >
> > > >
> > >
> > > > On Tue, Oct 27, 2009 at 1:43 AM, Brian West
> <brian at freeswitch.org> wrote:
> > >
> > > >>
> > >
> > > >> I highly doubt it... You can wait for someone to post their
> results
> > >
> > > >> but in the end you'll have to do your own load testing because
> not
> > >
> > > >> everyone's numbers will jive with your use case.  Which is the
> reason
> > >
> > > >> the project never posts or endorses a set call count.
> > >
> > > >>
> > >
> > > >> /b
> > >
> > > >>
> > >
> > > >> On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote:
> > >
> > > >>
> > >
> > > >> > Are there any benchmarking test results available publicly?
> > >
> > > >> > ________________________________________
> > >
> > > >>
> > >
> > > >>
> > >
> > > >> _______________________________________________
> > >
> > > >> FreeSWITCH-users mailing list
> > >
> > > >> FreeSWITCH-users at lists.freeswitch.org
> > >
> > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > > >>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> users
> > >
> > > >> http://www.freeswitch.org
> > >
> > > >
> > >
> > > >
> > >
> > > > _______________________________________________
> > >
> > > > FreeSWITCH-users mailing list
> > >
> > > > FreeSWITCH-users at lists.freeswitch.org
> > >
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > > >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> users
> > >
> > > > http://www.freeswitch.org
> > >
> > > >
> > >
> > > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > > --
> > >
> > > Sincerely,
> > >
> > >
> > >
> > > Giovanni Maruzzelli
> > >
> > > Cell : +39-347-2665618
> > >
> > >
> > >
> > > _______________________________________________
> > >
> > > FreeSWITCH-users mailing list
> > >
> > > FreeSWITCH-users at lists.freeswitch.org
> > >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> users
> > >
> > > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> users
> > http://www.freeswitch.org
> --
> http://www.google.com/search?q=vonage+sucks
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> users
> http://www.freeswitch.org
> 
> --
> This message has been scanned for viruses and
> dangerous content by N2Net Mailshield, and is
> believed to be clean.




From sicfslist at gmail.com  Tue Oct 27 13:19:15 2009
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Tue, 27 Oct 2009 15:19:15 -0500
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <1256673110.9746.29.camel@portable-evil>
References: <bf6923ed0910261446x6a184ecdnee2e73e904406246@mail.gmail.com>	<bf6923ed0910261501k1ff8c892rfc198c31e30ab286@mail.gmail.com>
	<1256673110.9746.29.camel@portable-evil>
Message-ID: <4AE755C3.6060307@gmail.com>

Cliff,

Try using xml_rpc ... status or show channels will give you what you need.

SDR

Cliff Wells wrote:
> A little off-topic, but since call-capacity is the subject, what are
> people using to analyze their CDR's to discover this?   I'm handling
> about 30k calls per day but have only a bandwidth-based guesstimate of
> the peak number of concurrent calls I'm handling.
>
> If there's an open source solution, I'd appreciate a pointer.
>
> Regards,
> Cliff
>
> On Mon, 2009-10-26 at 18:01 -0400, Eliot Gable wrote:
>   
>> Although, FYI, I just benchmarked mod_xml_curl on a separate web app
>> server from FS with FS on a Dell R710 with their current best
>> processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and 32
>> GB memory. The web app server is less than half the power of the R710.
>> I maxed the web app server at 300 calls per second (both setting up
>> and tearing down) and the R710 running FS was 65% idle. No audio was
>> being proxied through FS, though. If I were running the web app server
>> on an equivalent R710, they probably would have been on-par with each
>> other in performance. Extrapolating, I expect that in such a case I
>> should be able to get at least 650 CPS out of FS, though for
>> production I would probably limit it to 400 CPS or less so I leave
>> room for miscellaneous tasks. I maxed out the R710 at over 16,000
>> simultaneous calls (again, no audio proxying) but the only reason I
>> couldn't do more was because I hit some sort of thread creation limit
>> in Linux. There was about 17 GB of memory used for this many calls.
>> This should give you some ballpark idea of what you can accomplish
>> with FS.
>>
>> At some point, I will track down and resolve the thread creation
>> issue, at which time I believe call limits will be limited either by a
>> complex combination of available memory, the speed of the processor,
>> the cost of thread context switching, calls per second setup rate, and
>> call duration.
>>
>> --
>> Eliot Gable
>>
>>     
>>> -----Original Message-----
>>>
>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli
>>>
>>> Sent: Monday, October 26, 2009 4:56 PM
>>>
>>> To: freeswitch-users at lists.freeswitch.org
>>>
>>> Subject: Re: [Freeswitch-users] Estimating Call Capacity
>>>
>>>
>>>
>>> On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur
>>>
>>> <vinuth.madinur at gmail.com> wrote:
>>>
>>>       
>>>> Here are a few benchmarks that I had stumbled upon.
>>>>         
>>>> http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2
>>>>         
>>>
>>> Please remember NO benchmarks are endorsed by the FS community or
>>>
>>> developers, because there are just too many variables, and a simple
>>>
>>> figure is just useful for marketing hype, not for real dimensioning.
>>>
>>>
>>>
>>> You MUST do your own benchmarking, so you get an idea about how to
>>>
>>> dimension for your own use case and hardware.
>>>
>>>
>>>
>>>
>>>
>>>       
>>>> Thanks,
>>>>         
>>>> Vinuth.
>>>>         
>>>> On Tue, Oct 27, 2009 at 1:43 AM, Brian West <brian at freeswitch.org> wrote:
>>>>         
>>>>> I highly doubt it... You can wait for someone to post their results
>>>>>           
>>>>> but in the end you'll have to do your own load testing because not
>>>>>           
>>>>> everyone's numbers will jive with your use case.  Which is the reason
>>>>>           
>>>>> the project never posts or endorses a set call count.
>>>>>           
>>>>> /b
>>>>>           
>>>>> On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote:
>>>>>           
>>>>>> Are there any benchmarking test results available publicly?
>>>>>>             
>>>>>> ________________________________________
>>>>>>             
>>>>> _______________________________________________
>>>>>           
>>>>> FreeSWITCH-users mailing list
>>>>>           
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>           
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>           
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>           
>>>>> http://www.freeswitch.org
>>>>>           
>>>> _______________________________________________
>>>>         
>>>> FreeSWITCH-users mailing list
>>>>         
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>         
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>         
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>         
>>>> http://www.freeswitch.org
>>>>         
>>>
>>>
>>>
>>>
>>>
>>> --
>>>
>>> Sincerely,
>>>
>>>
>>>
>>> Giovanni Maruzzelli
>>>
>>> Cell : +39-347-2665618
>>>
>>>
>>>
>>> _______________________________________________
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> http://www.freeswitch.org
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     




From msc at freeswitch.org  Tue Oct 27 13:39:30 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 27 Oct 2009 13:39:30 -0700
Subject: [Freeswitch-users] FS Training
In-Reply-To: <5B56DAA56F1F43EC973C226E8CD9BCFC@greyhawk.tonecommander.com>
References: <5B56DAA56F1F43EC973C226E8CD9BCFC@greyhawk.tonecommander.com>
Message-ID: <87f2f3b90910271339p5153fb00r59a53fb753b6cdbc@mail.gmail.com>

On Tue, Oct 27, 2009 at 12:17 PM, Jerry Richards <jerry.richards at teotech.com
> wrote:

>
> Did the voting booth close?  I was unable to vote.  I'm not sure what link
> to click and I have had some strange issues with my FS account today.
>
> I would be interested in paid training.  Do you have plans for offering a
> training session at your locale?  Or would you travel onsite to provide
> training?
>
> Jerry,

Yes the poll closed last Thursday. I'm glad to hear that you are interested
in paid training. We are still evaluating all the options. I will be
contacting people off list to discuss specifics. This is still in the
incubation stage, so no ETA on anything just yet.

-MC
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From mike at jerris.com  Tue Oct 27 14:31:26 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 27 Oct 2009 17:31:26 -0400
Subject: [Freeswitch-users] switch_xml_open_root
In-Reply-To: <f8af5740910270548n3fe7883fp18f126afd421d89e@mail.gmail.com>
References: <f8af5740910270548n3fe7883fp18f126afd421d89e@mail.gmail.com>
Message-ID: <86D18B9D-CE41-44A2-9CE1-B0EE00BE141C@jerris.com>

You probably should not be calling that function, what are you trying  
to do?

Mike

On Oct 27, 2009, at 8:48 AM, srinivasula reddy wrote:

>
>
> Hi,
>
> when i am calling  switch_xml_open_root(1,err) . i am getting this  
> warning message.
> HEAP[FreeSwitch.exe]: Invalid Address specified to RtlFreeHeap 
> ( 00160000, 100E9FD0 ). can any know. please help me.

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From msc at freeswitch.org  Tue Oct 27 16:34:43 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 27 Oct 2009 16:34:43 -0700
Subject: [Freeswitch-users] switch_xml_open_root
In-Reply-To: <86D18B9D-CE41-44A2-9CE1-B0EE00BE141C@jerris.com>
References: <f8af5740910270548n3fe7883fp18f126afd421d89e@mail.gmail.com>
	<86D18B9D-CE41-44A2-9CE1-B0EE00BE141C@jerris.com>
Message-ID: <87f2f3b90910271634i639ee11bta144542ff72a4718@mail.gmail.com>

On Tue, Oct 27, 2009 at 2:31 PM, Michael Jerris <mike at jerris.com> wrote:

> You probably should not be calling that function, what are you trying to
> do?
>
> Mike
>
> On Oct 27, 2009, at 8:48 AM, srinivasula reddy wrote:
>

You might also want to move this conversation to the -dev list because it's
a little intense for the -users list.
-MC
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From bottleman at icf.org.ru  Tue Oct 27 17:16:57 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Wed, 28 Oct 2009 03:16:57 +0300 (MSK)
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
 connect
In-Reply-To: <65d96fc80910230614m2b2504d0r92006b78d267a399@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910222321080.12215@srv.icf.org.ru>
	<65d96fc80910221244i115b6c4dl18ab89dc4f41ccf1@mail.gmail.com>
	<Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
	<Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>
	<65d96fc80910230452s6b4a179aqa5bc41dc34efc403@mail.gmail.com>
	<Pine.LNX.4.64.0910231610220.21492@srv.icf.org.ru>
	<65d96fc80910230538w7105be6fve14956ab3b247bf6@mail.gmail.com>
	<Pine.LNX.4.64.0910231646210.29582@srv.icf.org.ru>
	<65d96fc80910230614m2b2504d0r92006b78d267a399@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910280313001.31425@srv.icf.org.ru>

On 2009-10-23 15:14 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:


TC>
TC> so the real solution is to implement a check for CallProceeding , Progress
TC>and Facility message whether it has a faststart element included. It it is
TC>true than you might start pre_answer.
TC>
TC>
TC>also, i don't see any handling of Call Proceeding ... what if there is a
TC>fastStart element in CallProceeding message? :)

Handling of fastStart in CallProceeding is commented out in h323plus library,
this is exploration from h323plus developers about this:


Yes that should be mera.

The problem is that Callproceeding does not always come from the remote it
may be generated by the gatekeeper. MERA where sending fast start elements
in the Call proceeding and connect. The call proceeding where not valid and
causing the media to fail. Normally (although valid) EP's do not set Fast
Start in Call proceeding so the code was disabled to resolve the MERA issue.

if you wont read "bugs" file in mod_h323, there is explaned how to enable it.

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From lei.tlfly at gmail.com  Tue Oct 27 18:14:54 2009
From: lei.tlfly at gmail.com (Lei Tang)
Date: Wed, 28 Oct 2009 09:14:54 +0800
Subject: [Freeswitch-users] how to config FS with two net interface?
In-Reply-To: <bf6923ed0910270657n22fb6115n6314398006612e25@mail.gmail.com>
References: <50c41b4e0910270149k2eaf8ee8v78ef24776947f270@mail.gmail.com>
	<bf6923ed0910270657n22fb6115n6314398006612e25@mail.gmail.com>
Message-ID: <50c41b4e0910271814h162d7878lac592351d0eefc7a@mail.gmail.com>

Thanks Eliot, It works.

2009/10/27 Eliot Gable
<egable+freeswitch at gmail.com<egable%2Bfreeswitch at gmail.com>
>

> Try setting ext-rtp-ip and ext-sip-ip on both profiles.
>
> On Tue, Oct 27, 2009 at 4:49 AM, Lei Tang <lei.tlfly at gmail.com> wrote:
> > Hi all, I run FS on a machine with two net interface, each interface has
> a
> > ip addr, one of the them connect to public network(has ip addr A), the
> > other  connect to a private network(has ip addr B), FS server as a SIP
> > server for public through A, all outbound call will bridge to a
> softswitch
> > in private network through B. here is my sofia config file and diaplan
> > config:
> >
> > sofia internal.xml
> > ....
> > <param name="rtp-ip" value="A"/>
> > <param name="sip-ip" value="A"/>
> >  ....
> >
> > sofia external.xml
> > ....
> > <param name="rtp-ip" value="B"/>
> > <param name="sip-ip" value="B"/>
> > ....
> >
> > dialplan
> > ......
> > <extension name="OUTBOUND">
> >     <condition field="destination_number" expression="^(\d+)$">
> >         <action application="set" data="hangup_after_bridge=true"/>
> >         <action application="set"
> >
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION"/>
> >         <action application="set"
> > data="effective_caller_id_number=xxxxxxx"/>  <!--here change the caller
> > number -->
> >         <action application="bridge"
> > data="sofia/external/${destination_number}@xxxxx"/>
> >       </condition>
> >     </extension>
> > .....
> >
> > then call seq is
> > sipAgent --> [internal -->(bridge)-->external] -->softswith
> >                           FREESWITCH
> >
> > the question is, when sipAgent make a outbound call, FS can't recevie the
> > caller's up audio stream, I traced the SIP packets, found that FS has
> return
> > addr B in SDP when ack the invite request from sipAgent, the ack packet
> is
> > ===============
> > SIP/2.0 183 Session Progress
> > Via: SIP/2.0/UDP
> >
> xxxxx:12208;branch=z9hG4bK-d8754z-dc750d57652c7c51-1---d8754z-;rport=12208
> > From: "1000" <sip:xxxx at A>;tag=cb4d3c4e
> > To: "65960581" <sip:xxxx at A>;tag=DtvSc0QX01yKN
> > Call-ID: ZTI2NmIwZGZiYzlhOGNkNTdiYWUzMzkzZTMwYzgxZjI.
> > CSeq: 2 INVITE
> > Contact: <sip:xxxxxx at B:5060;transport=udp>
> > User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460
> > Accept: application/sdp
> > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY,
> > REFER, UPDATE, REGISTER, INFO, PUBLISH
> > Supported: timer, precondition, path, replaces
> > Allow-Events: talk, presence, dialog, call-info, sla,
> > include-session-description, presence.winfo, message-summary, refer
> > Content-Type: application/sdp
> > Content-Disposition: session
> > Content-Length: 245
> >
> > v=0
> > o=FreeSWITCH 1256598185 1256598186 IN IP4 B   ;>>>>wrong this is the ip
> addr
> > of the adapter connect to the private network
> > s=FreeSWITCH
> > c=IN IP4 B ;>>>>wrong this is the ip addr of the adapter connect to the
> > private network
> > t=0 0
> > m=audio 31066 RTP/AVP 0 101
> > a=rtpmap:0 PCMU/8000
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-16
> > a=silenceSupp:off - - - -
> > a=ptime:20
> > ================
> > I think FS should return A in SDP, not the external binding addr (B),
> does
> > somebody known how to solve this problem?
> >
> > --
> > Lei.Tang
> > lei.tlfly at gmail.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Eliot Gable
>
> "We do not inherit the Earth from our ancestors: we borrow it from our
> children." ~David Brower
>
> "I decided the words were too conservative for me. We're not borrowing
> from our children, we're stealing from them--and it's not even
> considered to be a crime." ~David Brower
>
> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
> live; not live to eat.) ~Marcus Tullius Cicero
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Lei.Tang
lei.tlfly at gmail.com
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From brian at freeswitch.org  Tue Oct 27 18:25:00 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 27 Oct 2009 20:25:00 -0500
Subject: [Freeswitch-users] how to config FS with two net interface?
In-Reply-To: <50c41b4e0910271814h162d7878lac592351d0eefc7a@mail.gmail.com>
References: <50c41b4e0910270149k2eaf8ee8v78ef24776947f270@mail.gmail.com>
	<bf6923ed0910270657n22fb6115n6314398006612e25@mail.gmail.com>
	<50c41b4e0910271814h162d7878lac592351d0eefc7a@mail.gmail.com>
Message-ID: <B1381E00-7962-4498-8ACC-3369F310A6E6@freeswitch.org>

you should have just needed to set the rtp-ip and the sip-ip on both  
profiles to their values and it would have worked fine... their would  
have been no need to set the ext-*-ip equiv.

/b

On Oct 27, 2009, at 8:14 PM, Lei Tang wrote:

> Thanks Eliot, It works.




From lei.tlfly at gmail.com  Tue Oct 27 18:34:56 2009
From: lei.tlfly at gmail.com (Lei Tang)
Date: Wed, 28 Oct 2009 09:34:56 +0800
Subject: [Freeswitch-users] how to config FS with two net interface?
In-Reply-To: <B1381E00-7962-4498-8ACC-3369F310A6E6@freeswitch.org>
References: <50c41b4e0910270149k2eaf8ee8v78ef24776947f270@mail.gmail.com>
	<bf6923ed0910270657n22fb6115n6314398006612e25@mail.gmail.com>
	<50c41b4e0910271814h162d7878lac592351d0eefc7a@mail.gmail.com>
	<B1381E00-7962-4498-8ACC-3369F310A6E6@freeswitch.org>
Message-ID: <50c41b4e0910271834w1e0bf668yaf03f2b6179266ef@mail.gmail.com>

Hi Brian, It doesn't work if I only set rtp-ip and sip-ip. when  I set the
ext-rtp-ip, it works fine.

2009/10/28 Brian West <brian at freeswitch.org>

> you should have just needed to set the rtp-ip and the sip-ip on both
> profiles to their values and it would have worked fine... their would
> have been no need to set the ext-*-ip equiv.
>
> /b
>
> On Oct 27, 2009, at 8:14 PM, Lei Tang wrote:
>
> > Thanks Eliot, It works.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Lei.Tang
lei.tlfly at gmail.com
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From brian at freeswitch.org  Tue Oct 27 19:07:00 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 27 Oct 2009 21:07:00 -0500
Subject: [Freeswitch-users] how to config FS with two net interface?
In-Reply-To: <50c41b4e0910271834w1e0bf668yaf03f2b6179266ef@mail.gmail.com>
References: <50c41b4e0910270149k2eaf8ee8v78ef24776947f270@mail.gmail.com>
	<bf6923ed0910270657n22fb6115n6314398006612e25@mail.gmail.com>
	<50c41b4e0910271814h162d7878lac592351d0eefc7a@mail.gmail.com>
	<B1381E00-7962-4498-8ACC-3369F310A6E6@freeswitch.org>
	<50c41b4e0910271834w1e0bf668yaf03f2b6179266ef@mail.gmail.com>
Message-ID: <DE3384C4-FB19-47E7-A784-CB0D8F58D376@freeswitch.org>

Ok then something is broken badly... and that makes NO sense.  Because  
it works on my box.

/b

On Oct 27, 2009, at 8:34 PM, Lei Tang wrote:

> Hi Brian, It doesn't work if I only set rtp-ip and sip-ip. when  I  
> set the ext-rtp-ip, it works fine.
>
> 2009/10/28 Brian West <brian at freeswitch.org>
> you should have just needed to set the rtp-ip and the sip-ip on both
> profiles to their values and it would have worked fine... their would
> have been no need to set the ext-*-ip equiv.
>
> /b
>
> On Oct 27, 2009, at 8:14 PM, Lei Tang wrote:
>
> > Thanks Eliot, It works.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
> -- 
> Lei.Tang
> lei.tlfly at gmail.com
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From srinivas.ksvreddy at gmail.com  Tue Oct 27 21:37:05 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Wed, 28 Oct 2009 10:07:05 +0530
Subject: [Freeswitch-users] switch_xml_open_root
In-Reply-To: <86D18B9D-CE41-44A2-9CE1-B0EE00BE141C@jerris.com>
References: <f8af5740910270548n3fe7883fp18f126afd421d89e@mail.gmail.com>
	<86D18B9D-CE41-44A2-9CE1-B0EE00BE141C@jerris.com>
Message-ID: <f8af5740910272137t22fa559arc4b562a232634627@mail.gmail.com>

Hi mike,

thank for your reply.
i am trying to call that function from swig.cs.   its working fine first
time with the warning information, then onwards it is not working.
is there other way can i call reload function from freeswitch.managed code.
any help

srinivas



On Wed, Oct 28, 2009 at 3:01 AM, Michael Jerris <mike at jerris.com> wrote:

> You probably should not be calling that function, what are you trying to
> do?
>
> Mike
>
> On Oct 27, 2009, at 8:48 AM, srinivasula reddy wrote:
>
> *
> *
> Hi,
>
> when i am calling  switch_xml_open_root(1,err) . i am getting this warning
> message.
> HEAP[FreeSwitch.exe]: Invalid Address specified to RtlFreeHeap( 00160000,
> 100E9FD0 ). can any know. please help me.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Srinivasula Reddy K
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From lon at kickasspixels.com  Tue Oct 27 21:51:40 2009
From: lon at kickasspixels.com (Lon Baker)
Date: Tue, 27 Oct 2009 21:51:40 -0700
Subject: [Freeswitch-users] Retrieve conference member state using cli?
Message-ID: <5d3e0dc60910272151o13f7fad9k35c4f9f61cb1b287@mail.gmail.com>

Just like you can call conference list for conferences, is there a way to
retrieve the profile state of a conference member using the cli or xml/rpc?

Lon
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From mike at jerris.com  Tue Oct 27 22:21:32 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 28 Oct 2009 01:21:32 -0400
Subject: [Freeswitch-users] switch_xml_open_root
In-Reply-To: <f8af5740910272137t22fa559arc4b562a232634627@mail.gmail.com>
References: <f8af5740910270548n3fe7883fp18f126afd421d89e@mail.gmail.com>
	<86D18B9D-CE41-44A2-9CE1-B0EE00BE141C@jerris.com>
	<f8af5740910272137t22fa559arc4b562a232634627@mail.gmail.com>
Message-ID: <2B91DB5F-A48C-4ECC-A13F-FABF3CAED1BB@jerris.com>

just api execute reloadxml

Mike

On Oct 28, 2009, at 12:37 AM, srinivasula reddy wrote:

> Hi mike,
>
> thank for your reply.
> i am trying to call that function from swig.cs.   its working fine  
> first time with the warning information, then onwards it is not  
> working.
> is there other way can i call reload function from  
> freeswitch.managed code. any help
>



From srinivas.ksvreddy at gmail.com  Tue Oct 27 22:34:49 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Wed, 28 Oct 2009 11:04:49 +0530
Subject: [Freeswitch-users] reload from managed
In-Reply-To: <b39273530910270818j1e7f8c2eqdce44c3abdc72cc7@mail.gmail.com>
References: <f8af5740910270232k689e8856sbc0dc8b668d9eacf@mail.gmail.com>
	<b39273530910270818j1e7f8c2eqdce44c3abdc72cc7@mail.gmail.com>
Message-ID: <f8af5740910272234i5e7a7f94vff1d6a00a24352f3@mail.gmail.com>

i tried switch_api_execute its working fine, thanks rupa

On Tue, Oct 27, 2009 at 8:48 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> Is it necessary for you to call the reload function directly?  Why not
> make an api call using switch_api_execute and the command "reloadxml".
>
>
> On Tue, Oct 27, 2009 at 4:32 AM, srinivasula reddy
> <srinivas.ksvreddy at gmail.com> wrote:
> > Hi,
> >
> > how can i call reload from managed code? and i called
> > Csharp_switch_xml_open_root, this is working first time, then onwards it
> is
> > not working.
> > is this the correct function to call reload?
> >
> > --
> > Srinivasula Reddy K
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Srinivasula Reddy K
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From velu.technical at gmail.com  Wed Oct 28 01:29:38 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Wed, 28 Oct 2009 13:59:38 +0530
Subject: [Freeswitch-users] How to stop the playback files
Message-ID: <1452e2980910280129wab717eam4ae9da2554ab2e1@mail.gmail.com>

Dear All,
     I have played the list of voice files in playback like the following by
using ESL perl module,

      $conn->execute("set","playback_delimiter=!");
      $conn->execute("set","playback_sleep_val=100");
 $conn->playback($sound_path."ivr/ivr-welcome_to_freeswitch.wav!ivr/ivr-this_ivr_will_let_you_test_features.wav!ivr/ivr-you_may_exit_by_hanging_up.wav");

 In a loop I am checking the DTMF event, if that event comes I should stop
the above palyback. How can I do it?

Regards,
Velusamy.
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From mayamatakeshi at gmail.com  Wed Oct 28 01:33:18 2009
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Wed, 28 Oct 2009 17:33:18 +0900
Subject: [Freeswitch-users] Channel variables not being set when FS calls
	user
Message-ID: <15b9404e0910280133x27563aa4jc578a2092450c702@mail.gmail.com>

Hello,
in this page
http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide
we can read:

=============================
Variables:
Any variables defined in the domain or user will be defined as channel
variables when there is a call to user or when there is an inbound calls
from that user.
=============================

I can see the channel variables are set when there is an incoming call from
the user, but not when FS sends a call to the user.
Can someone confirm what is the expected behavior?

br,
takeshi
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From brian at freeswitch.org  Wed Oct 28 01:55:13 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 28 Oct 2009 03:55:13 -0500
Subject: [Freeswitch-users] Retrieve conference member state using cli?
In-Reply-To: <5d3e0dc60910272151o13f7fad9k35c4f9f61cb1b287@mail.gmail.com>
References: <5d3e0dc60910272151o13f7fad9k35c4f9f61cb1b287@mail.gmail.com>
Message-ID: <41933E56-8B6A-4D54-9989-D69BA6566515@freeswitch.org>

Type "conference" at the cli and the help will be displayed... same  
can be called over XML RPC.

/b

On Oct 27, 2009, at 11:51 PM, Lon Baker wrote:

> Just like you can call conference list for conferences, is there a  
> way to retrieve the profile state of a conference member using the  
> cli or xml/rpc?
>
> Lon




From brian at freeswitch.org  Wed Oct 28 01:56:43 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 28 Oct 2009 03:56:43 -0500
Subject: [Freeswitch-users] How to stop the playback files
In-Reply-To: <1452e2980910280129wab717eam4ae9da2554ab2e1@mail.gmail.com>
References: <1452e2980910280129wab717eam4ae9da2554ab2e1@mail.gmail.com>
Message-ID: <98BA0749-A312-4565-80A7-49B0BA48C1F0@freeswitch.org>

http://wiki.freeswitch.org/wiki/Channel_Variables#playback_terminators

Are you using sync or async socket?

/b

On Oct 28, 2009, at 3:29 AM, velusamy velu wrote:

> Dear All,
>      I have played the list of voice files in playback like the  
> following by using ESL perl module,
>
>       $conn->execute("set","playback_delimiter=!");
>       $conn->execute("set","playback_sleep_val=100");
>  $conn->playback($sound_path."ivr/ivr-welcome_to_freeswitch.wav!ivr/ 
> ivr-this_ivr_will_let_you_test_features.wav!ivr/ivr- 
> you_may_exit_by_hanging_up.wav");
>
>  In a loop I am checking the DTMF event, if that event comes I  
> should stop the above palyback. How can I do it?
>
> Regards,
> Velusamy.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From brian at freeswitch.org  Wed Oct 28 01:57:39 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 28 Oct 2009 03:57:39 -0500
Subject: [Freeswitch-users] Channel variables not being set when FS
	calls user
In-Reply-To: <15b9404e0910280133x27563aa4jc578a2092450c702@mail.gmail.com>
References: <15b9404e0910280133x27563aa4jc578a2092450c702@mail.gmail.com>
Message-ID: <D3006963-53C3-45E2-B67A-5B06B4AC4EBB@freeswitch.org>

Yes it is the expected behavior... if you wish to set them on an  
inbound call TO the user you'll use the set_user api to do so.  Its  
kinda like sudo in FreeSWITCH it'll load up all the variables for that  
user.

/b

On Oct 28, 2009, at 3:33 AM, mayamatakeshi wrote:

> I can see the channel variables are set when there is an incoming  
> call from the user, but not when FS sends a call to the user.
> Can someone confirm what is the expected behavior?




From srinivas.ksvreddy at gmail.com  Wed Oct 28 02:07:27 2009
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Wed, 28 Oct 2009 14:37:27 +0530
Subject: [Freeswitch-users] db
Message-ID: <f8af5740910280207o4fdcaed2i7403d3f4bdff2fb2@mail.gmail.com>

Hi,

can i use sqlserver instead of sqllite. and can two freeswitch servers can
share same database(sqllite or sqlserver). any help would be great.

Thanks
Srinivasula Reddy K
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From mayamatakeshi at gmail.com  Wed Oct 28 02:23:22 2009
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Wed, 28 Oct 2009 18:23:22 +0900
Subject: [Freeswitch-users] Channel variables not being set when FS
	calls user
In-Reply-To: <D3006963-53C3-45E2-B67A-5B06B4AC4EBB@freeswitch.org>
References: <15b9404e0910280133x27563aa4jc578a2092450c702@mail.gmail.com>
	<D3006963-53C3-45E2-B67A-5B06B4AC4EBB@freeswitch.org>
Message-ID: <15b9404e0910280223s59183346me4783584b32148f8@mail.gmail.com>

Got it. Thanks.

On Wed, Oct 28, 2009 at 5:57 PM, Brian West <brian at freeswitch.org> wrote:

> Yes it is the expected behavior... if you wish to set them on an
> inbound call TO the user you'll use the set_user api to do so.  Its
> kinda like sudo in FreeSWITCH it'll load up all the variables for that
> user.
>
> /b
>
> On Oct 28, 2009, at 3:33 AM, mayamatakeshi wrote:
>
> > I can see the channel variables are set when there is an incoming
> > call from the user, but not when FS sends a call to the user.
> > Can someone confirm what is the expected behavior?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tculjaga at gmail.com  Wed Oct 28 02:24:25 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 28 Oct 2009 10:24:25 +0100
Subject: [Freeswitch-users] Fwd: mod_opal - call charged before H.225
	connect
In-Reply-To: <Pine.LNX.4.64.0910280313001.31425@srv.icf.org.ru>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<Pine.LNX.4.64.0910222352180.16915@srv.icf.org.ru>
	<65d96fc80910230116k3519418fjb49f277b7268a075@mail.gmail.com>
	<Pine.LNX.4.64.0910231446490.5644@srv.icf.org.ru>
	<65d96fc80910230452s6b4a179aqa5bc41dc34efc403@mail.gmail.com>
	<Pine.LNX.4.64.0910231610220.21492@srv.icf.org.ru>
	<65d96fc80910230538w7105be6fve14956ab3b247bf6@mail.gmail.com>
	<Pine.LNX.4.64.0910231646210.29582@srv.icf.org.ru>
	<65d96fc80910230614m2b2504d0r92006b78d267a399@mail.gmail.com>
	<Pine.LNX.4.64.0910280313001.31425@srv.icf.org.ru>
Message-ID: <65d96fc80910280224w55dd0acasfaac3ffe603e4bd4@mail.gmail.com>

>
>
>
> Handling of fastStart in CallProceeding is commented out in h323plus
> library,
> this is exploration from h323plus developers about this:
>
>
> Yes that should be mera.
>
> The problem is that Callproceeding does not always come from the remote it
> may be generated by the gatekeeper.


this is a feature .. called force_callproceeding. It means MERA will send a
provisional CallProceeding in order not to timeout on calls that don't
respond with that message on time. If this message contains a faststart
element it is certanly a bug and it has to be reported to them.


> MERA where sending fast start elements
> in the Call proceeding and connect. The call proceeding where not valid and
> causing the media to fail.


well if there is a correct faststart element within a connect message (or
alerting or facility or progress), the originator should adjust the media
resources accordingly. Here what could went wrong is just the media before
the next faststart element in the row.


> Normally (although valid) EP's do not set Fast
> Start in Call proceeding so the code was disabled to resolve the MERA
> issue.
>
>
well, this is unlikely as fast start element can be included in call
proceeding message. The developer's task is to determine whether a call
proceeding message is to be trusted or not.
Also, provisional call proceeding messages don't have faststart element
included! There are equipment (Cisco PGW / HSI) that are sending call
proceeding with faststart element and h245Controll (OLC + TCS/MSD) that has
to be treated correctly. Unfortunately, just disabling handling of
callproceeding faststart element is not a real option...



> if you wont read "bugs" file in mod_h323, there is explaned how to enable
> it.
>


of course i can enable it during build time but this will not solve interop
issues later we can encounter...




Do you maybe have some sniffs/traces of the wrong call proceeding message ?



...anyhow this is the expected behaviour when a GK/Proxy sends a provisional
Call Proceding to the terminator and later it receives the real Call
Proceeding carring faststart and h245Controll element within.

Entities in the signalling path shall also use the Facility message or the
Progress message to convey
any new information (such as Q.931 information elements, CallProceeding-UUIE
fields, tunnelled
non-H.323 protocols, and encapsulated H.245 messages) received in a Call
Proceeding message to
the other endpoint if the entity has already sent a Call Proceeding message.
This will allow the
entity, for example, to transmit the fastStart element to facilitate proper
establishment of a Fast
Connect call and/or a Progress Indicator to indicate the presence of in-band
tones and
announcements. When using the Facility message to carrying such information
extracted from the
Call Proceeding message, the reason in the Facility should be set to
forwardedElements.



in other words:

ORIG                          GK                                TERM
-------------------------------------------------------------------------------------------------------------
Setup OLC                     =>
Call proceeding (prov)        <=
                              Setup OLC                         =>
                              Call proceeding (OLC+TCS/MSD)     <=
Facility (OLC+TCS/MSD)        <=


<----------- normal call establishment scenario follows ----------->
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From velu.technical at gmail.com  Wed Oct 28 02:29:45 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Wed, 28 Oct 2009 14:59:45 +0530
Subject: [Freeswitch-users] How to stop the playback files
In-Reply-To: <98BA0749-A312-4565-80A7-49B0BA48C1F0@freeswitch.org>
References: <1452e2980910280129wab717eam4ae9da2554ab2e1@mail.gmail.com>
	<98BA0749-A312-4565-80A7-49B0BA48C1F0@freeswitch.org>
Message-ID: <1452e2980910280229h7d30bf84lb5143dc46eea95@mail.gmail.com>

I am using async socket. Shall I set the regular expression for
palyback_terminators?

On Wed, Oct 28, 2009 at 2:26 PM, Brian West <brian at freeswitch.org> wrote:

> http://wiki.freeswitch.org/wiki/Channel_Variables#playback_terminators
>
> Are you using sync or async socket?
>
> /b
>
> On Oct 28, 2009, at 3:29 AM, velusamy velu wrote:
>
> > Dear All,
> >      I have played the list of voice files in playback like the
> > following by using ESL perl module,
> >
> >       $conn->execute("set","playback_delimiter=!");
> >       $conn->execute("set","playback_sleep_val=100");
> >  $conn->playback($sound_path."ivr/ivr-welcome_to_freeswitch.wav!ivr/
> > ivr-this_ivr_will_let_you_test_features.wav!ivr/ivr-
> > you_may_exit_by_hanging_up.wav");
> >
> >  In a loop I am checking the DTMF event, if that event comes I
> > should stop the above palyback. How can I do it?
> >
> > Regards,
> > Velusamy.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Wed Oct 28 02:53:27 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 28 Oct 2009 04:53:27 -0500
Subject: [Freeswitch-users] db
In-Reply-To: <f8af5740910280207o4fdcaed2i7403d3f4bdff2fb2@mail.gmail.com>
References: <f8af5740910280207o4fdcaed2i7403d3f4bdff2fb2@mail.gmail.com>
Message-ID: <F760E917-11D5-428B-A1F0-31E9F3351C82@freeswitch.org>

http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core

Please don't cross post if possible.

/b

On Oct 28, 2009, at 4:07 AM, srinivasula reddy wrote:

> Hi,
>
> can i use sqlserver instead of sqllite. and can two freeswitch  
> servers can share same database(sqllite or sqlserver). any help  
> would be great.
>
> Thanks
> Srinivasula Reddy K
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From delianspam at gmail.com  Wed Oct 28 03:54:39 2009
From: delianspam at gmail.com (Delian Tashev)
Date: Wed, 28 Oct 2009 12:54:39 +0200
Subject: [Freeswitch-users] How to stop the playback files
In-Reply-To: <004801ca57b6$bd14c5c0$373e5140$@net>
References: <1452e2980910280129wab717eam4ae9da2554ab2e1@mail.gmail.com>	<98BA0749-A312-4565-80A7-49B0BA48C1F0@freeswitch.org>
	<004801ca57b6$bd14c5c0$373e5140$@net>
Message-ID: <005301ca57bd$0cce1bb0$266a5310$@com>

Hello Velusamy!

You may also be interested in using the method:
http://wiki.freeswitch.org/wiki/Mod_lua#session:playAndGetDigits
It play a file, gets DTMFs and can be terminated if the maximum number of
digits is reached (i.e. 1 digit).

Delian Tashev

-----Original Message-----
From: Brian West [mailto:brian at freeswitch.org] 
Sent: Wednesday, October 28, 2009 10:57
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] How to stop the playback files

http://wiki.freeswitch.org/wiki/Channel_Variables#playback_terminators

Are you using sync or async socket?

/b

On Oct 28, 2009, at 3:29 AM, velusamy velu wrote:

> Dear All,
>      I have played the list of voice files in playback like the  
> following by using ESL perl module,
>
>       $conn->execute("set","playback_delimiter=!");
>       $conn->execute("set","playback_sleep_val=100");
>  $conn->playback($sound_path."ivr/ivr-welcome_to_freeswitch.wav!ivr/ 
> ivr-this_ivr_will_let_you_test_features.wav!ivr/ivr- 
> you_may_exit_by_hanging_up.wav");
>
>  In a loop I am checking the DTMF event, if that event comes I  
> should stop the above palyback. How can I do it?
>
> Regards,
> Velusamy.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org









From brian at freeswitch.org  Wed Oct 28 07:13:29 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 28 Oct 2009 09:13:29 -0500
Subject: [Freeswitch-users] Problem with inbound call answered but no
	sound
In-Reply-To: <00d601ca571e$519f7300$f4de5900$@com>
References: <016e01ca56ad$56362380$02a26a80$@com>	<168575F8-6507-4AD2-B448-2D6FDA0152B6@freeswitch.org>	<018801ca56b2$1e63e0f0$5b2ba2d0$@com>
	<BACDC899-CE4F-42E1-B2A0-C0C0E31F8787@freeswitch.org>
	<00d601ca571e$519f7300$f4de5900$@com>
Message-ID: <8D1A994D-2DCD-4016-9BEF-D6FBD3FBAA1E@freeswitch.org>

What kind of router are you behind?

/b

On Oct 27, 2009, at 10:58 AM, Lars Zeb wrote:

> Thanks for the reply, Brian.
>
> Did something in FS change between v15183 and v15225 to make this  
> occur? I ask because this same configuration worked OK in the  
> earlier version.
>
> Lars
>

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From ivan at myrvold.org  Wed Oct 28 07:37:30 2009
From: ivan at myrvold.org (Ivan C Myrvold)
Date: Wed, 28 Oct 2009 15:37:30 +0100
Subject: [Freeswitch-users] SIP provider with extern rtp server
In-Reply-To: <bf6923ed0910271034w6c7e0cc5t685a5377764755e6@mail.gmail.com>
References: <A2D2E918-B494-4179-96BA-CA3A7717846C@myrvold.org>
	<bf6923ed0910270658t61bb6687ne0b7d3ec5090e2ff@mail.gmail.com>
	<CB8008BD-4066-4B5B-9043-214CFE938CD8@myrvold.org>
	<bf6923ed0910271034w6c7e0cc5t685a5377764755e6@mail.gmail.com>
Message-ID: <FA2C693B-FCE1-4BDD-BCE5-302AC699CEDB@myrvold.org>

Here is a debug log from a call from an internal phone out to an  
external (my iPhone with nbr 91316356):
http://pastebin.freeswitch.org/108578

Ivan

Den 27. okt. 2009 kl. 18:34 skrev Eliot Gable:

> No, the IP address the media originates from does not need to be tied
> to the SIP IP address. Can you send a Wireshark capture taken on the
> FreeSWITCH server of both call legs? Or, if you can, pastebin a debug
> log from FreeSWITCH console with sofia loglevel set to 9 and siptrace
> on for any Sofia SIP profiles involved.
>
> On Tue, Oct 27, 2009 at 11:52 AM, Ivan C Myrvold <ivan at myrvold.org>  
> wrote:
>> The server is on a public IP, so there is no nat issue here.
>>
>> I can also see the rtp messages on wireshark starting just after the
>> 183 Session Progress message on the server, but just in one  
>> direction,
>> coming in to the server.
>> So it looks like Freeswitch is stopping the rtp.
>> Is this because the rtp originates from another ip than the  sip
>> provider ip?
>>
>> Ivan
>>
>> Den 27. okt. 2009 kl. 14:58 skrev Eliot Gable:
>>
>>> Make sure you let their media IPs through your firewall. Also, if  
>>> you
>>> are behind a NAT, check you have things passing to the correct
>>> internal address.
>>>
>>> On Tue, Oct 27, 2009 at 2:46 AM, Ivan C Myrvold <ivan at myrvold.org>
>>> wrote:
>>>> I have used a SIP provider for more than a year. A few days ago, he
>>>> said he was moving to a new server, and asked me to reconfigure. I
>>>> did, and everything seemed to work fine, until I did an outgoing  
>>>> call
>>>> to an external telephone. I found out I had no audio, in neither
>>>> direction. Incoming calls was working fine.
>>>>
>>>> My provider said that the rtp is not going through the sip  
>>>> server, as
>>>> it did earlier, but now through several other IP's.
>>>>
>>>> Do I have to do some special configuration to handle that?
>>>>
>>>> Ivan
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>> users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Eliot Gable
>>>
>>> "We do not inherit the Earth from our ancestors: we borrow it from  
>>> our
>>> children." ~David Brower
>>>
>>> "I decided the words were too conservative for me. We're not  
>>> borrowing
>>> from our children, we're stealing from them--and it's not even
>>> considered to be a crime." ~David Brower
>>>
>>> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
>>> live; not live to eat.) ~Marcus Tullius Cicero
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>
>
>
> -- 
> Eliot Gable
>
> "We do not inherit the Earth from our ancestors: we borrow it from our
> children." ~David Brower
>
> "I decided the words were too conservative for me. We're not borrowing
> from our children, we're stealing from them--and it's not even
> considered to be a crime." ~David Brower
>
> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
> live; not live to eat.) ~Marcus Tullius Cicero
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>




From brian at freeswitch.org  Wed Oct 28 07:52:08 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 28 Oct 2009 09:52:08 -0500
Subject: [Freeswitch-users] [Freeswitch-dev] Build Issue
In-Reply-To: <3181A30B8C35AB4AA8577B78DDF4613805FCA7DE@nickel.mettonigroup.com>
References: <3181A30B8C35AB4AA8577B78DDF4613805FCA7DE@nickel.mettonigroup.com>
Message-ID: <8B9A5FFA-6940-4A43-983D-B8C8E954EAFE@freeswitch.org>

Clean and build again... the issue was files.freeswitch.org was having  
a little drama so I suspect it didn't download everything properly?

/b

On Oct 28, 2009, at 9:40 AM, Suneel Papineni wrote:

> Hi,
>
> Just now I have downloaded SVN version of FreeSwitch from http://svn.freeswitch.org/svn/freeswitch/trunk/ 
> .
>
> Tried to build the solution (Freeswitch.2008.sln ) using, but it  
> thrown many fatal errors(C1083)  and some warnings. Please find  
> below the list of errors and warnings.
>
> Could anyone let me know where I made mistake and source to get  
> these missing files.
>
> Thanks & Regards
> Suneel
>
>
> Warning               1              warning C4018: '<' : signed/ 
> unsigned mismatch                d:\Office\Projects\Freeswitch\libs 
> \pcre\pcre_ucp_searchfuncs.c           158         libpcre
> Warning               2              warning C4018: '<=' : signed/ 
> unsigned mismatch                d:\Office\Projects\Freeswitch\libs 
> \pcre\pcre_ucp_searchfuncs.c           163         libpcre
> Warning               3              Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> libteletone
> Error      4              fatal error C1083: Cannot open source  
> file: '..\..\pthreads-w32-2-7-0-release\pthread.c': No such file or  
> directory             c1            pthread
> Warning               5              Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_spidermonkey
> Error      6              fatal error C1083: Cannot open source  
> file: '..\..\mpg123\src\libmpg123\dct64.c': No such file or  
> directory                c1            libmpg123
> Error      7              fatal error C1083: Cannot open source  
> file: '..\..\mpg123\src\libmpg123\decode.c': No such file or  
> directory             c1            libmpg123
> Error      8              fatal error C1083: Cannot open source  
> file: '..\..\mpg123\src\libmpg123\decode_2to1.c': No such file or  
> directory             c1            libmpg123
> Error      9              fatal error C1083: Cannot open source  
> file: '..\..\mpg123\src\libmpg123\decode_4to1.c': No such file or  
> directory             c1            libmpg123
> Error      10           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\decode_ntom.c': No such file or  
> directory             c1            libmpg123
> Error      11           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\equalizer.c': No such file or  
> directory             c1            libmpg123
> Error      12           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\format.c': No such file or  
> directory                c1            libmpg123
> Error      13           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\frame.c': No such file or  
> directory                c1            libmpg123
> Error      14           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\icy.c': No such file or  
> directory                c1            libmpg123
> Error      15           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\icy2utf8.c': No such file or  
> directory             c1            libmpg123
> Error      16           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\id3.c': No such file or  
> directory                c1            libmpg123
> Error      17           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\layer1.c': No such file or  
> directory                c1            libmpg123
> Error      18           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\layer2.c': No such file or  
> directory                c1            libmpg123
> Error      19           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\layer3.c': No such file or  
> directory                c1            libmpg123
> Error      20           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\libmpg123.c': No such file or  
> directory             c1            libmpg123
> Error      21           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\optimize.c': No such file or  
> directory             c1            libmpg123
> Error      22           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\parse.c': No such file or  
> directory                c1            libmpg123
> Error      23           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\readers.c': No such file or  
> directory             c1            libmpg123
> Error      24           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\stringbuf.c': No such file or  
> directory             c1            libmpg123
> Error      25           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\tabinit.c': No such file or  
> directory                c1            libmpg123
> Error      26           fatal error C1083: Cannot open source file:  
> '..\..\mpg123\src\libmpg123\compat.c': No such file or  
> directory             c1            libmpg123
> Error      27           fatal error C1083: Cannot open source file:  
> '..\..\libshout-2.2.2\src\httpp\httpp.c': No such file or  
> directory                c1            libshout
> Error      28           fatal error C1083: Cannot open source file:  
> '..\..\libshout-2.2.2\src\mp3.c': No such file or directory        
> c1                libshout
> Error      29           fatal error C1083: Cannot open source file:  
> '..\..\libshout-2.2.2\src\ogg.c': No such file or directory          
> c1                libshout
> Error      30           fatal error C1083: Cannot open source file:  
> '..\..\libshout-2.2.2\src\net\resolver.c': No such file or  
> directory             c1            libshout
> Error      31           fatal error C1083: Cannot open source file:  
> '..\..\libshout-2.2.2\src\shout.c': No such file or directory      
> c1                libshout
> Error      32           fatal error C1083: Cannot open source file:  
> '..\..\libshout-2.2.2\src\net\sock.c': No such file or  
> directory                c1            libshout
> Error      33           fatal error C1083: Cannot open source file:  
> '..\..\libshout-2.2.2\src\thread\thread.c': No such file or  
> directory             c1            libshout
> Error      34           fatal error C1083: Cannot open source file:  
> '..\..\libshout-2.2.2\src\timing\timing.c': No such file or  
> directory             c1            libshout
> Error      35           fatal error C1083: Cannot open source file:  
> '..\..\libshout-2.2.2\src\util.c': No such file or directory          
> c1                libshout
> Error      36           fatal error C1083: Cannot open source file:  
> '..\..\libshout-2.2.2\src\avl\avl.c': No such file or directory   
> c1                libshout
> Error      37           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\bitstream.c': No such file or  
> directory             c1            libmp3lame
> Error      38           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\version.c': No such file or  
> directory             c1            libmp3lame
> Error      39           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\VbrTag.c': No such file or  
> directory                c1            libmp3lame
> Error      40           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\vbrquantize.c': No such file or  
> directory             c1            libmp3lame
> Error      41           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\util.c': No such file or  
> directory                c1            libmp3lame
> Error      42           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\takehiro.c': No such file or  
> directory             c1            libmp3lame
> Error      43           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\set_get.c': No such file or  
> directory             c1            libmp3lame
> Error      44           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\reservoir.c': No such file or  
> directory             c1            libmp3lame
> Error      45           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\quantize_pvt.c': No such file or  
> directory             c1            libmp3lame
> Error      46           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\quantize.c': No such file or  
> directory             c1            libmp3lame
> Error      47           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\psymodel.c': No such file or  
> directory             c1            libmp3lame
> Error      48           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\presets.c': No such file or  
> directory             c1            libmp3lame
> Error      49           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\newmdct.c': No such file or  
> directory             c1            libmp3lame
> Error      50           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\mpglib_interface.c': No such file or  
> directory        c1            libmp3lame
> Error      51           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\lame.c': No such file or  
> directory                c1            libmp3lame
> Error      52           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\id3tag.c': No such file or  
> directory                c1            libmp3lame
> Error      53           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\gain_analysis.c': No such file or  
> directory             c1            libmp3lame
> Error      54           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\fft.c': No such file or  
> directory                c1            libmp3lame
> Error      55           fatal error C1083: Cannot open source file:  
> '..\..\lame-3.97\libmp3lame\encoder.c': No such file or  
> directory             c1            libmp3lame
> Error      56           fatal error C1083: Cannot open include file:  
> 'pthread.h': No such file or directory                d:\Office 
> \Projects\Freeswitch\libs\sofia-sip\libsofia-sip-ua\su 
> \su_alloc_lock.c              40                libsofia_sip_ua_static
> Error      57           fatal error C1083: Cannot open include file:  
> 'pthread.h': No such file or directory                d:\office 
> \projects\freeswitch\libs\sofia-sip\libsofia-sip-ua\su 
> \su_port.h          547         libsofia_sip_ua_static
> Error      58           fatal error C1083: Cannot open include file:  
> 'pthread.h': No such file or directory                d:\Office 
> \Projects\Freeswitch\libs\sofia-sip\libsofia-sip-ua\su 
> \su_log.c            46           libsofia_sip_ua_static
> Error      59           fatal error C1083: Cannot open include file:  
> 'pthread.h': No such file or directory                d:\office 
> \projects\freeswitch\libs\sofia-sip\libsofia-sip-ua\su 
> \su_port.h          547         libsofia_sip_ua_static
> Error      60           fatal error C1083: Cannot open include file:  
> 'pthread.h': No such file or directory                d:\office 
> \projects\freeswitch\libs\sofia-sip\libsofia-sip-ua\su 
> \su_port.h          547         libsofia_sip_ua_static
> Error      61           fatal error C1083: Cannot open include file:  
> 'pthread.h': No such file or directory                d:\Office 
> \Projects\Freeswitch\libs\sofia-sip\libsofia-sip-ua\su 
> \su_root.c          46           libsofia_sip_ua_static
> Error      62           fatal error C1083: Cannot open include file:  
> 'pthread.h': No such file or directory                d:\office 
> \projects\freeswitch\libs\sofia-sip\libsofia-sip-ua\su 
> \su_port.h          547         libsofia_sip_ua_static
> Error      63           fatal error C1083: Cannot open include file:  
> 'pthread.h': No such file or directory                d:\office 
> \projects\freeswitch\libs\sofia-sip\libsofia-sip-ua\su 
> \su_port.h          547         libsofia_sip_ua_static
> Error      64           fatal error C1083: Cannot open include file:  
> 'pthread.h': No such file or directory                d:\Office 
> \Projects\Freeswitch\libs\sofia-sip\libsofia-sip-ua\su 
> \su_uniqueid.c                85                libsofia_sip_ua_static
> Error      65           fatal error C1083: Cannot open include file:  
> 'pthread.h': No such file or directory                d:\office 
> \projects\freeswitch\libs\sofia-sip\libsofia-sip-ua\su 
> \su_port.h          547         libsofia_sip_ua_static
> Error      66           error BK1506 : cannot open file '.\Debug 
> \su_alloc_lock.sbr': No such file or directory      
> BSCMAKE                libsofia_sip_ua_static
> Warning               67           Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             esl
> Error      68           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\audio\au_none.c': No such file or  
> directory                c1            flite
> Error      69           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\audio\au_streaming.c': No such file or  
> directory             c1            flite
> Error      70           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\audio\au_wince.c': No such file or  
> directory             c1            flite
> Error      71           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\audio\audio.c': No such file or  
> directory                c1            flite
> Error      72           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\audio\auserver.c': No such file or  
> directory                c1            flite
> Error      73           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmulex\cmu_lex.c': No such file or  
> directory             c1            flite
> Error      74           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmulex\cmu_lex_data.c': No such file or  
> directory             c1            flite
> Error      75           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmulex\cmu_lex_entries.c': No such file or  
> directory        c1            flite
> Error      76           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmulex\cmu_lts_model.c': No such file or  
> directory        c1            flite
> Error      77           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmulex\cmu_lts_rules.c': No such file or  
> directory             c1            flite
> Error      78           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmulex\cmu_postlex.c': No such file or  
> directory             c1            flite
> Error      79           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_time_awb\cmu_time_awb_cart.c': No such  
> file or directory               c1            flite
> Error      80           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_time_awb\cmu_time_awb_clunits.c': No  
> such file or directory              c1            flite
> Error      81           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_time_awb\cmu_time_awb_lex_entry.c': No  
> such file or directory        c1            flite
> Error      82           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_time_awb\cmu_time_awb_lpc.c': No such  
> file or directory               c1            flite
> Error      83           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_time_awb\cmu_time_awb_mcep.c': No such  
> file or directory               c1            flite
> Error      84           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_kal\cmu_us_kal.c': No such file or  
> directory        c1            flite
> Error      85           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_kal\cmu_us_kal_diphone.c': No such  
> file or directory      c1            flite
> Error      86           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_kal\cmu_us_kal_lpc.c': No such file  
> or directory                c1            flite
> Error      87           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_kal\cmu_us_kal_res.c': No such file  
> or directory                c1            flite
> Error      88           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_kal\cmu_us_kal_residx.c': No such  
> file or directory      c1            flite
> Error      89           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\utils\cst_alloc.c': No such file or  
> directory                c1            flite
> Error      90           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\utils\cst_args.c': No such file or  
> directory                c1            flite
> Error      91           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\stats\cst_cart.c': No such file or  
> directory                c1            flite
> Error      92           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\cg\cst_cg.c': No such file or directory     
> c1                flite
> Error      93           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\wavesynth\cst_clunits.c': No such file or  
> directory             c1            flite
> Error      94           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\wavesynth\cst_diphone.c': No such file or  
> directory             c1            flite
> Error      95           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\utils\cst_endian.c': No such file or  
> directory             c1            flite
> Error      96           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\utils\cst_error.c': No such file or  
> directory                c1            flite
> Error      97           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\utils\cst_features.c': No such file or  
> directory             c1            flite
> Error      98           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\hrg\cst_ffeature.c': No such file or  
> directory             c1            flite
> Error      99           fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\synth\cst_ffeatures.c': No such file or  
> directory             c1            flite
> Error      100         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\utils\cst_file_stdio.c': No such file or  
> directory             c1            flite
> Error      101         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\hrg\cst_item.c': No such file or  
> directory                c1            flite
> Error      102         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\lexicon\cst_lexicon.c': No such file or  
> directory             c1            flite
> Error      103         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\speech\cst_lpcres.c': No such file or  
> directory             c1            flite
> Error      104         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\lexicon\cst_lts.c': No such file or  
> directory                c1            flite
> Error      105         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\lexicon\cst_lts_rewrites.c': No such file or  
> directory             c1            flite
> Error      106         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\cg\cst_mlpg.c': No such file or  
> directory                c1            flite
> Error      107         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\cg\cst_mlsa.c': No such file or  
> directory                c1            flite
> Error      108         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\utils\cst_mmap_win32.c': No such file or  
> directory             c1            flite
> Error      109         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\synth\cst_phoneset.c': No such file or  
> directory             c1            flite
> Error      110         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\wavesynth\cst_reflpc.c': No such file or  
> directory             c1            flite
> Error      111         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\regex\cst_regex.c': No such file or  
> directory             c1            flite
> Error      112         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\hrg\cst_rel_io.c': No such file or  
> directory                c1            flite
> Error      113         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\hrg\cst_relation.c': No such file or  
> directory             c1            flite
> Error      114         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\wavesynth\cst_sigpr.c': No such file or  
> directory             c1            flite
> Error      115         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\utils\cst_socket.c': No such file or  
> directory             c1            flite
> Error      116         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\stats\cst_ss.c': No such file or  
> directory                c1            flite
> Error      117         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\utils\cst_string.c': No such file or  
> directory                c1            flite
> Error      118         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\wavesynth\cst_sts.c': No such file or  
> directory             c1            flite
> Error      119         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\synth\cst_synth.c': No such file or  
> directory             c1            flite
> Error      120         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\utils\cst_tokenstream.c': No such file or  
> directory             c1            flite
> Error      121         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\speech\cst_track.c': No such file or  
> directory             c1            flite
> Error      122         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\speech\cst_track_io.c': No such file or  
> directory             c1            flite
> Error      123         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\wavesynth\cst_units.c': No such file or  
> directory             c1            flite
> Error      124         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\synth\cst_utt_utils.c': No such file or  
> directory             c1            flite
> Error      125         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\hrg\cst_utterance.c': No such file or  
> directory             c1            flite
> Error      126         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\utils\cst_val.c': No such file or  
> directory                c1            flite
> Error      127         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\utils\cst_val_const.c': No such file or  
> directory             c1            flite
> Error      128         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\utils\cst_val_user.c': No such file or  
> directory             c1            flite
> Error      129         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\cg\cst_vc.c': No such file or directory     
> c1                flite
> Error      130         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\stats\cst_viterbi.c': No such file or  
> directory             c1            flite
> Error      131         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\synth\cst_voice.c': No such file or  
> directory             c1            flite
> Error      132         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\speech\cst_wave.c': No such file or  
> directory             c1            flite
> Error      133         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\speech\cst_wave_io.c': No such file or  
> directory             c1            flite
> Error      134         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\speech\cst_wave_utils.c': No such file or  
> directory             c1            flite
> Error      135         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\synth\flite.c': No such file or  
> directory                c1            flite
> Error      136         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\speech\rateconv.c': No such file or  
> directory             c1            flite
> Error      137         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\regex\regexp.c': No such file or  
> directory                c1            flite
> Error      138         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\regex\regsub.c': No such file or  
> directory                c1            flite
> Error      139         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\usenglish\us_aswd.c': No such file or  
> directory             c1            flite
> Error      140         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\usenglish\us_dur_stats.c': No such file or  
> directory             c1            flite
> Error      141         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\usenglish\us_durz_cart.c': No such file or  
> directory             c1            flite
> Error      142         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\usenglish\us_expand.c': No such file or  
> directory             c1            flite
> Error      143         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\usenglish\us_f0_model.c': No such file or  
> directory             c1            flite
> Error      144         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\usenglish\us_f0lr.c': No such file or  
> directory             c1            flite
> Error      145         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\usenglish\us_ffeatures.c': No such file or  
> directory             c1            flite
> Error      146         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\usenglish\us_gpos.c': No such file or  
> directory             c1            flite
> Error      147         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\usenglish\us_int_accent_cart.c': No such  
> file or directory                c1            flite
> Error      148         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\usenglish\us_int_tone_cart.c': No such file  
> or directory                c1            flite
> Error      149         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\usenglish\us_nums_cart.c': No such file or  
> directory        c1            flite
> Error      150         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\usenglish\us_phoneset.c': No such file or  
> directory             c1            flite
> Error      151         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\usenglish\us_phrasing_cart.c': No such file  
> or directory                c1            flite
> Error      152         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\usenglish\us_text.c': No such file or  
> directory             c1            flite
> Error      153         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\usenglish\usenglish.c': No such file or  
> directory             c1            flite
> Error      154         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_slt\cmu_us_slt.c': No such file or  
> directory             c1            flite
> Error      155         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_slt\cmu_us_slt_cg.c': No such file  
> or directory                c1            flite
> Error      156         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_slt\cmu_us_slt_cg_durmodel.c': No  
> such file or directory               c1            flite
> Error      157         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_slt\cmu_us_slt_cg_f0_trees.c': No  
> such file or directory               c1            flite
> Error      158         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_slt\cmu_us_slt_cg_mcep_trees.c': No  
> such file or directory              c1            flite
> Error      159         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_slt\cmu_us_slt_cg_params.c': No such  
> file or directory      c1            flite
> Error      160         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_slt\cmu_us_slt_cg_phonestate.c': No  
> such file or directory               c1            flite
> Error      161         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_awb\cmu_us_awb.c': No such file or  
> directory                c1            flite
> Error      162         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_awb\cmu_us_awb_cg.c': No such file  
> or directory      c1            flite
> Error      163         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_awb\cmu_us_awb_cg_durmodel.c': No  
> such file or directory           c1            flite
> Error      164         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_awb\cmu_us_awb_cg_f0_trees.c': No  
> such file or directory              c1            flite
> Error      165         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_awb\cmu_us_awb_cg_mcep_trees.c': No  
> such file or directory       c1            flite
> Error      166         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_awb\cmu_us_awb_cg_params.c': No such  
> file or directory                c1            flite
> Error      167         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_awb\cmu_us_awb_cg_phonestate.c': No  
> such file or directory       c1            flite
> Error      168         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_rms\cmu_us_rms.c': No such file or  
> directory        c1            flite
> Error      169         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_rms\cmu_us_rms_cg.c': No such file  
> or directory                c1            flite
> Error      170         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_rms\cmu_us_rms_cg_durmodel.c': No  
> such file or directory             c1            flite
> Error      171         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_rms\cmu_us_rms_cg_f0_trees.c': No  
> such file or directory                c1            flite
> Error      172         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_rms\cmu_us_rms_cg_mcep_trees.c': No  
> such file or directory         c1            flite
> Error      173         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_rms\cmu_us_rms_cg_params.c': No such  
> file or directory               c1            flite
> Error      174         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\lang\cmu_us_rms\cmu_us_rms_cg_phonestate.c': No  
> such file or directory         c1            flite
> Error      175         fatal error C1083: Cannot open source file:  
> '..\..\flite-1.3.99\src\audio\au_command.c': No such file or  
> directory             c1            flite
> Warning               176         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_fax
> Warning               177         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_lcr
> Warning               178         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_pocketsphinx
> Warning               179         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_spidermonkey_curl
> Warning               180         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_fsv
> Warning               181         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_voipcodecs
> Warning               182         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_tone_stream
> Warning               183         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_cdr_csv
> Warning               184         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_logfile
> Warning               185         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_dialplan_asterisk
> Warning               186         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_expr
> Warning               187         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_fifo
> Warning               188         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_say_nl
> Warning               189         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_say_it
> Warning               190         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_say_fr
> Warning               191         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_say_es
> Warning               192         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_say_de
> Warning               193         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_voicemail
> Warning               194         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_spidermonkey_socket
> Warning               195         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_local_stream
> Warning               196         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_esf
> Warning               197         warning C4244: 'function' :  
> conversion from 'int' to 'uint8_t', possible loss of  
> data                d:\Office\Projects\Freeswitch\src\mod 
> \applications\mod_esf\mod_esf.c         139         mod_esf
> Warning               198         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_h26x
> Warning               199         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_amr
> Warning               200         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_xml_cdr
> Warning               201         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_say_en
> Warning               202         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_xml_curl
> Warning               203         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_spidermonkey_odbc
> Warning               204         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_enum
> Warning               205         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_easyroute
> Warning               206         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_spidermonkey_teletone
> Warning               207         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_spidermonkey_core_db
> Warning               208         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_native_file
> Warning               209         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_g723_1
> Warning               210         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             fs_cli
> Warning               211         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_siren
> Warning               212         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_sofia
> Warning               213         warning C4018: '>' : signed/ 
> unsigned mismatch                d:\Office\Projects\Freeswitch\src 
> \mod\endpoints\mod_sofia\sofia.c 2145       mod_sofia
> Warning               214         warning C4018: '<' : signed/ 
> unsigned mismatch                d:\Office\Projects\Freeswitch\src 
> \mod\endpoints\mod_sofia\sofia.c 2145       mod_sofia
> Warning               215         warning C4018: '>' : signed/ 
> unsigned mismatch                d:\Office\Projects\Freeswitch\src 
> \mod\endpoints\mod_sofia\sofia.c 2832       mod_sofia
> Warning               216         warning C4018: '<' : signed/ 
> unsigned mismatch                d:\Office\Projects\Freeswitch\src 
> \mod\endpoints\mod_sofia\sofia.c 2832       mod_sofia
> Warning               217         warning C4702: unreachable  
> code                d:\office\projects\freeswitch\src\mod\endpoints 
> \mod_sofia\sofia.c  4005       mod_sofia
> Error      218         fatal error LNK1181: cannot open input file  
> '..\..\..\..\libs\win32\sofia\debug 
> \libsofia_sip_ua_static.lib'                mod_sofia           
> mod_sofia
> Error      219         fatal error C1083: Cannot open include file:  
> 'os.h': No such file or directory                d:\Office\Projects 
> \Freeswitch\libs\win32\libshout\shout\shout.h        26            
> mod_shout
> Warning               220         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_snom
> Warning               221         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_say_zh
> Warning               222         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_managed
> Warning               223         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_vmd
> Warning               224         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_file_string
> Warning               225         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_nibblebill
> Warning               226         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_say_ru
> Warning               227         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_limit
> Warning               228         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_event_socket
> Warning               229         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_dptools
> Warning               230         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_conference
> Warning               231         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_rss
> Warning               232         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_xml_rpc
> Warning               233         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_console
> Warning               234         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_commands
> Warning               235         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_dingaling
> Warning               236         warning C4018: '<' : signed/ 
> unsigned mismatch                d:\Office\Projects\Freeswitch\src 
> \mod\endpoints\mod_dingaling\mod_dingaling.c   2326        
> mod_dingaling
> Warning               237         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_ilbc
> Warning               238         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_loopback
> Warning               239         warning C4189: 'code' : local  
> variable is initialized but not referenced                d:\Office 
> \Projects\Freeswitch\src\mod\languages\mod_lua\mod_lua.cpp         
> 393         mod_lua
> Warning               240         warning C4101: 'error' :  
> unreferenced local variable                d:\Office\Projects 
> \Freeswitch\src\mod\languages\mod_lua\mod_lua.cpp        395          
> mod_lua
> Warning               241         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_event_multicast
> Warning               242         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_zeroconf
> Warning               243         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_dialplan_directory
> Warning               244         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_ldap
> Warning               245         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_dialplan_xml
> Warning               246         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_speex
> Warning               247         warning LNK4075: ignoring '/ 
> EDITANDCONTINUE' due to '/INCREMENTAL:NO'  
> specification                libspeexdsp.lib                mod_speex
> Warning               248         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_PortAudio
> Warning               249         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_iax
> Warning               250         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_sndfile
> Warning               251         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl             mod_g729
> Warning               252         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> mod_valet_parking
> Warning               253         Command line warning D9040 :  
> ignoring option '/analyze'; Code Analysis warnings are not available  
> in this edition of the compiler               cl              
> FreeSwitchConsole
> *************************************************************************
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From maciej.aniserowicz at gmail.com  Wed Oct 28 09:21:43 2009
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Wed, 28 Oct 2009 09:21:43 -0700 (PDT)
Subject: [Freeswitch-users] "Can not record session. Media not enabled
 on channel."
In-Reply-To: <191c3a030910270818y79f436d9h8f6311c4671502c2@mail.gmail.com>
References: <1256034659005-3857858.post@n2.nabble.com>
	<87f2f3b90910201746h43a31b32m5c163fd98373f915@mail.gmail.com>
	<1256283391890-3877285.post@n2.nabble.com>
	<87f2f3b90910231200v71e72661id4cc9f2b7fba5610@mail.gmail.com>
	<1256541677550-3890610.post@n2.nabble.com>
	<87f2f3b90910261422n6952fa26ya6a2a66452365146@mail.gmail.com>
	<1256655334215-3899478.post@n2.nabble.com>
	<191c3a030910270818y79f436d9h8f6311c4671502c2@mail.gmail.com>
Message-ID: <1256746903296-3906568.post@n2.nabble.com>


Correct - compiled but did not run. Works fine now.

I'll see if the error shows up again and let you know if it does.
Thanks,
MA



Anthony Minessale wrote:
> 
> won't compile or won't run?
> maybe you should try rebuilding it.
> 
> 
> On Tue, Oct 27, 2009 at 9:55 AM, Maciej Aniserowicz <
> maciej.aniserowicz at gmail.com> wrote:
> 
>> Sorry, trunk does not compile on win7, here are the details:
>>
>>
>> rev.15247
>>
>> ---------------------------
>> Microsoft Visual C++ Debug Library
>> ---------------------------
>> Debug Assertion Failed!
>>
>> 
>>
>>
>>
>> ----- Original Message -----
>> *From:* [hidden
>> email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3899478&i=0>
>> *To:* [hidden
>> email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3899478&i=1>
>> *Sent:* Monday, October 26, 2009 10:32 PM
>> *Subject:* Re: [Freeswitch-users] "Can not record session. Media not
>> enabled on channel."
>>
>>
>>
>> On Mon, Oct 26, 2009 at 12:21 AM, Maciej Aniserowicz <[hidden
>> email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3895104&i=0>
>> > wrote:
>>
>>>
>>> Yes, I can confirm - this exact error occurs each time when I start
>>> recording
>>> before the call is answered (just after sending ORIGINATE command) - but
>>> I
>>> think that's completely understandable that media is not ready for an
>>> unanswered call.
>>> But... is there any other event that guarantees media to be ready?
>>>
>>> Update to latest SVN and try again.
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> [hidden
>> email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3895104&i=1>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> ------------------------------
>> View this message in context: Re: [Freeswitch-users] "Can not record
>> session. Media not enabled on
>> channel."<http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3899478.html>
>>
>> Sent from the freeswitch-users mailing list
>> archive<http://n2.nabble.com/freeswitch-users-f2379917.html>at
>> Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3906568.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From mayamatakeshi at gmail.com  Wed Oct 28 09:35:35 2009
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 29 Oct 2009 01:35:35 +0900
Subject: [Freeswitch-users] No RTP destination auto-correction
Message-ID: <15b9404e0910280935u16f08ff6nd8527d8c0b9f32a8@mail.gmail.com>

I'm seeing a strange issue with eyebeam behind NAT.
If I make a call from eyebeam to FS, I can see FS receives packets from
eyebeam's nat address (confirmed with tcpdump and mod_event_socket DTMF) but
it doesn't change the RTP destination to the source of those packets and
sends the packets to the private address advertised in the SDP.
However, if I do hold/unhold and send some digits, FS logs "Auto Changing
port from X to Y" and starts to send the packets to the correct address.
This always work after a hold/unhold + digits but never happen just after
the first INVITE transaction.
Also, I don't see the problem if eyebeam receives a call from FS.
Has anyone seen this?

br,
takeshi
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From mattdfong at gmail.com  Wed Oct 28 09:37:56 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Wed, 28 Oct 2009 23:37:56 +0700
Subject: [Freeswitch-users] Determining Frame Size
Message-ID: <4256bf830910280937n1f946518w423597fa6584c26f@mail.gmail.com>

I noticed that the wait_for_silence's <silence_hits> <listen_hits> use
number of frames as base values (rather than seconds or milliseconds). I'd
like to wait for 5 seconds of consistent silence, but I'm wondering how I
can determine the sample rate (and thus frames per second) for an existing
call. Is there a variable I can read? Thanks.

--matt
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From anthony.minessale at gmail.com  Wed Oct 28 09:43:31 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 28 Oct 2009 11:43:31 -0500
Subject: [Freeswitch-users] No RTP destination auto-correction
In-Reply-To: <15b9404e0910280935u16f08ff6nd8527d8c0b9f32a8@mail.gmail.com>
References: <15b9404e0910280935u16f08ff6nd8527d8c0b9f32a8@mail.gmail.com>
Message-ID: <191c3a030910280943h2998c349kd22843247106490@mail.gmail.com>

are you >= r15256 because an issue with this was fixed in that revision.



On Wed, Oct 28, 2009 at 11:35 AM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:

> I'm seeing a strange issue with eyebeam behind NAT.
> If I make a call from eyebeam to FS, I can see FS receives packets from
> eyebeam's nat address (confirmed with tcpdump and mod_event_socket DTMF) but
> it doesn't change the RTP destination to the source of those packets and
> sends the packets to the private address advertised in the SDP.
> However, if I do hold/unhold and send some digits, FS logs "Auto Changing
> port from X to Y" and starts to send the packets to the correct address.
> This always work after a hold/unhold + digits but never happen just after
> the first INVITE transaction.
> Also, I don't see the problem if eyebeam receives a call from FS.
> Has anyone seen this?
>
> br,
> takeshi
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From brian at freeswitch.org  Wed Oct 28 09:51:54 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 28 Oct 2009 11:51:54 -0500
Subject: [Freeswitch-users] No RTP destination auto-correction
In-Reply-To: <15b9404e0910280935u16f08ff6nd8527d8c0b9f32a8@mail.gmail.com>
References: <15b9404e0910280935u16f08ff6nd8527d8c0b9f32a8@mail.gmail.com>
Message-ID: <A08ECEE6-46AF-497A-81CC-D65CB6BC0D95@freeswitch.org>

have you updated to the latest SVN?

/b

On Oct 28, 2009, at 11:35 AM, mayamatakeshi wrote:

> I'm seeing a strange issue with eyebeam behind NAT.
> If I make a call from eyebeam to FS, I can see FS receives packets  
> from eyebeam's nat address (confirmed with tcpdump and  
> mod_event_socket DTMF) but it doesn't change the RTP destination to  
> the source of those packets and sends the packets to the private  
> address advertised in the SDP.
> However, if I do hold/unhold and send some digits, FS logs "Auto  
> Changing port from X to Y" and starts to send the packets to the  
> correct address. This always work after a hold/unhold + digits but  
> never happen just after the first INVITE transaction.
> Also, I don't see the problem if eyebeam receives a call from FS.
> Has anyone seen this?
>
> br,
> takeshi
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From djbinter at yahoo.com  Wed Oct 28 09:56:07 2009
From: djbinter at yahoo.com (DJB)
Date: Wed, 28 Oct 2009 09:56:07 -0700 (PDT)
Subject: [Freeswitch-users] SIP UPDATE Method
In-Reply-To: <191c3a030910261259p3b1fcfecvbbaed9af0af4d206@mail.gmail.com>
References: <528227.52311.qm@web37508.mail.mud.yahoo.com>
	<191c3a030910260808y45952bffu7118057caba4ea4@mail.gmail.com>
	<33829.77997.qm@web37506.mail.mud.yahoo.com>
	<191c3a030910261145q46128a92w75d73158a7f73572@mail.gmail.com>
	<514887.94094.qm@web37502.mail.mud.yahoo.com>
	<191c3a030910261247n12f1917al8110cf2636f41e20@mail.gmail.com>
	<191c3a030910261250q3b933692jd028f2067ae33252@mail.gmail.com>
	<191c3a030910261259p3b1fcfecvbbaed9af0af4d206@mail.gmail.com>
Message-ID: <938120.75712.qm@web37508.mail.mud.yahoo.com>

Thank you very much, Anthony.  I really appreciate your help and excellent work of you and your team.

Regards,
Dorn B.



________________________________
From: Anthony Minessale <anthony.minessale at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Mon, October 26, 2009 12:59:01 PM
Subject: Re: [Freeswitch-users] SIP UPDATE Method

in r15233 i put it back to the way it originally was but I may have to remove that if it causes more problems.
We tried to handle update for display updating which was not working but the handler for it was still in place which may have broken some automatic behavior regarding update so I added it back to how it was originally to determine that.



On Mon, Oct 26, 2009 at 2:50 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

>but i think the minimum value you can set is 120
>
>
>
>On Mon, Oct 26, 2009 at 2:47 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>
>I think session timers will use invite if there is no update.
>>
>>the session-timeout profile param should control that but you have to double the number you actually want because it sends the new invite at the halfway point.
>>>>
>>
>>
>>
>>
>>
>>On Mon, Oct 26, 2009 at 2:29 PM, DJB <djbinter at yahoo.com> wrote:
>>
>>>>>
>>>
>>>We were trying to see whether we can adjust call duration on Session timers.  It was a question from the application developers.  I am not sure what they are trying to do exactly.  
>>>
>>>
>>>Thank you.
>>>
>>>
>>>>>>
>>>
________________________________
From: Anthony Minessale <anthony.minessale at gmail.com>
>>>To: freeswitch-users at lists.freeswitch.org
>>>Sent: Mon, October 26, 2009 11:45:31 AM
>>>>>>
>>>
>>>
>>>Subject: Re: [Freeswitch-users] SIP UPDATE Method
>>>
>>>
>>>what exactly are you expecting to use it for?
>>>>>>We never really supported it anyway.
>>>
>>>
>>>
>>>On Mon, Oct 26, 2009 at 12:19 PM, DJB <djbinter at yahoo.com> wrote:
>>>
>>>I wonder whether you will consider to put it back on the next version 1.0.5 since 1.0.4 has it?  
>>>>
>>>>
>>>>Regards,
>>>>Dorn B.
>>>>
>>>>
>>>>
>>>>
>>>>>>>>
>>>>
>>>>
________________________________
From: Anthony Minessale <anthony.minessale at gmail.com>
>>>>To: freeswitch-users at lists.freeswitch.org
>>>>Sent: Mon, October 26, 2009 8:08:59 AM
>>>>Subject: Re: [Freeswitch-users] SIP UPDATE Method
>>>>
>>>>
>>>>It was never there before and it
>>>> caused extreme havoc once we added it so we took it away again.
>>>>
>>>>
>>>>
>>>>On Sun, Oct 25, 2009 at 12:55 AM, DJB <djbinter at yahoo.com> wrote:
>>>>
>>>>I am wondering why after I update to trunk-15225, the Allow: UPDATE method is no longer there.
>>>>>
>>>>>>>>>>User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15225
>>>>>>>>>>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>>>
>>>>>>>>>>Am I missing something here?
>>>>>
>>>>>>>>>>Thank you.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>>>>>>_______________________________________________
>>>>>>>>>>FreeSWITCH-users mailing list
>>>>>FreeSWITCH-users at lists.freeswitch.org
>>>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>-- 
>>>>Anthony Minessale II
>>>>
>>>>FreeSWITCH http://www.freeswitch.org/
>>>>ClueCon http://www.cluecon.com/
>>>>>>>>
>>>>
>>>>
>>>>
>>>>Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>>AIM: anthm
>>>>MSN:anthony_minessale at hotmail.com
>>>>>>>>
>>>>
>>>>
>>>>GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>>>>>IRC: irc.freenode.net #freeswitch
>>>>
>>>>FreeSWITCH Developer Conference
>>>>sip:888 at conference.freeswitch.org
>>>>iax:guest at conference.freeswitch.org/888
>>>>googletalk:conf+888 at conference.freeswitch.org
>>>>pstn:213-799-1400
>>>>
>>>>
>>>>_______________________________________________
>>>>>>>>FreeSWITCH-users mailing list
>>>>FreeSWITCH-users at lists.freeswitch.org
>>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>>-- 
>>>Anthony Minessale II
>>>
>>>FreeSWITCH http://www.freeswitch.org/
>>>ClueCon http://www.cluecon.com/
>>>>>>
>>>
>>>
>>>Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>>AIM: anthm
>>>MSN:anthony_minessale at hotmail.com
>>>>>>
>>>
>>>GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>>>IRC: irc.freenode.net #freeswitch
>>>
>>>FreeSWITCH Developer Conference
>>>sip:888 at conference.freeswitch.org
>>>iax:guest at conference.freeswitch.org/888
>>>googletalk:conf+888 at conference.freeswitch.org
>>>pstn:213-799-1400
>>>
>>>
>>>_______________________________________________
>>>>>>FreeSWITCH-users mailing list
>>>FreeSWITCH-users at lists.freeswitch.org
>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>http://www.freeswitch.org
>>>
>>>
>>
>>
>>-- 
>>Anthony Minessale II
>>
>>FreeSWITCH http://www.freeswitch.org/
>>ClueCon http://www.cluecon.com/
>>>>
>>
>>Twitter: http://twitter.com/FreeSWITCH_wire
>>
>>AIM: anthm
>>MSN:anthony_minessale at hotmail.com
>>>>
>>GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>IRC: irc.freenode.net #freeswitch
>>
>>FreeSWITCH Developer Conference
>>sip:888 at conference.freeswitch.org
>>iax:guest at conference.freeswitch.org/888
>>googletalk:conf+888 at conference.freeswitch.org
>>pstn:213-799-1400
>>
>
>
>-- 
>Anthony Minessale II
>
>FreeSWITCH http://www.freeswitch.org/
>ClueCon http://www.cluecon.com/
>>
>Twitter: http://twitter.com/FreeSWITCH_wire
>
>AIM: anthm
>MSN:anthony_minessale at hotmail.com
>>GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>IRC: irc.freenode.net #freeswitch
>
>FreeSWITCH Developer Conference
>sip:888 at conference.freeswitch.org
>iax:guest at conference.freeswitch.org/888
>googletalk:conf+888 at conference.freeswitch.org
>pstn:213-799-1400
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
pstn:213-799-1400



      
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From msc at freeswitch.org  Wed Oct 28 10:00:18 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 28 Oct 2009 10:00:18 -0700
Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.5pre3 is available
	for testing!
Message-ID: <87f2f3b90910281000r4c2021b8u7232320614737e30@mail.gmail.com>

Hello FreeSWITCHers,

The latest FreeSWITCH version is now available for download on the
files<http://files.freeswitch.org/freeswitch-1.0.5pre3.tar.gz>site.
The announcement story is in the main
FreeSWITCH <http://www.freeswitch.org/node/212> site. Please download and
test, and then test some more. We need your feedback. The sooner we get your
feedback, the more quickly we can roll the official 1.0.5.

Thanks!
-Michael
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From jakub at pawlinski.pl  Wed Oct 28 09:46:21 2009
From: jakub at pawlinski.pl (=?ISO-8859-2?Q?Jakub_Pawli=F1ski?=)
Date: Wed, 28 Oct 2009 17:46:21 +0100
Subject: [Freeswitch-users] Problem with hangin bri
Message-ID: <69a9ce230910280946h5eae8c58m72f9b9492f08a329@mail.gmail.com>

Hi,
I have some problems with bri status. I have 3 chanel isdn modem, and zaptel
compatible quad bri card.  I can invoke calls from my voip phone to cell
phone, and vice versa, but when i make inbound and outbound connection in
nearly same time something goes wrong with chanells and after few calls all
of them has hangup status.

There is log about that in attachement, see "is already in use waiting for
it to become available." phrase. Time of this event is about 14:43:35.
Unload and Load open_zap module helped, but its not an solution because of
lost connections.
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From jakub at pawlinski.pl  Wed Oct 28 09:49:22 2009
From: jakub at pawlinski.pl (=?ISO-8859-2?Q?Jakub_Pawli=F1ski?=)
Date: Wed, 28 Oct 2009 17:49:22 +0100
Subject: [Freeswitch-users] Problem with hangin bri
In-Reply-To: <69a9ce230910280946h5eae8c58m72f9b9492f08a329@mail.gmail.com>
References: <69a9ce230910280946h5eae8c58m72f9b9492f08a329@mail.gmail.com>
Message-ID: <69a9ce230910280949j6b48d367q6bbbe4f6e445938@mail.gmail.com>

Hi,
I have some problems with bri status. I have 3 chanel isdn modem, and zaptel
compatible quad bri card.  I can invoke calls from my voip phone to cell
phone, and vice versa, but when i make inbound and outbound connection in
nearly same time something goes wrong with chanells and after few calls all
of them has hangup status.

There is log about that in attachement, see "is already in use waiting for
it to become available." phrase. Time of this event is about 14:43:35.
Unload and Load open_zap module helped, but its not an solution because of
lost connections.
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From mayamatakeshi at gmail.com  Wed Oct 28 10:23:51 2009
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 29 Oct 2009 02:23:51 +0900
Subject: [Freeswitch-users] No RTP destination auto-correction
In-Reply-To: <A08ECEE6-46AF-497A-81CC-D65CB6BC0D95@freeswitch.org>
References: <15b9404e0910280935u16f08ff6nd8527d8c0b9f32a8@mail.gmail.com>
	<A08ECEE6-46AF-497A-81CC-D65CB6BC0D95@freeswitch.org>
Message-ID: <15b9404e0910281023v6a84bba0s1ea00621f7eb51ed@mail.gmail.com>

I'm really sorry, guys.
I was just some revisions behind and forgot to make current and test before
posting.
It's fine now.
Thank you.

On Thu, Oct 29, 2009 at 1:51 AM, Brian West <brian at freeswitch.org> wrote:

> have you updated to the latest SVN?
>
> /b
>
> On Oct 28, 2009, at 11:35 AM, mayamatakeshi wrote:
>
> > I'm seeing a strange issue with eyebeam behind NAT.
> > If I make a call from eyebeam to FS, I can see FS receives packets
> > from eyebeam's nat address (confirmed with tcpdump and
> > mod_event_socket DTMF) but it doesn't change the RTP destination to
> > the source of those packets and sends the packets to the private
> > address advertised in the SDP.
> > However, if I do hold/unhold and send some digits, FS logs "Auto
> > Changing port from X to Y" and starts to send the packets to the
> > correct address. This always work after a hold/unhold + digits but
> > never happen just after the first INVITE transaction.
> > Also, I don't see the problem if eyebeam receives a call from FS.
> > Has anyone seen this?
> >
> > br,
> > takeshi
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From davis.erwin at gmail.com  Wed Oct 28 10:39:07 2009
From: davis.erwin at gmail.com (Erwin Davis)
Date: Wed, 28 Oct 2009 13:39:07 -0400
Subject: [Freeswitch-users] error in loading spidermonkey
Message-ID: <a38a91d50910281039pdf28a4fh536f44d238b9a954@mail.gmail.com>

Hi, I got an error in loading mod_spidermonkey. my fs in ver 1.04 runs on
fedora 8 VM. Any clue? thanks,

[CRIT] switch_loadable_module.c:871 Error Loading module
/usr/local/freeswitch/mod/mod_spidermonkey.so
**/usr/lib/libnss3.so: undefined symbol: PR_UnloadLibrary**
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From william.suffill at gmail.com  Wed Oct 28 10:48:04 2009
From: william.suffill at gmail.com (William Suffill)
Date: Wed, 28 Oct 2009 13:48:04 -0400
Subject: [Freeswitch-users] error in loading spidermonkey
In-Reply-To: <a38a91d50910281039pdf28a4fh536f44d238b9a954@mail.gmail.com>
References: <a38a91d50910281039pdf28a4fh536f44d238b9a954@mail.gmail.com>
Message-ID: <6b65470d0910281048l6bd3d6exfa8504149fd78a31@mail.gmail.com>

Per http://jira.freeswitch.org/browse/MDXMLINT-23

Try to configure it with --without-libcurl.


-- W



From msc at freeswitch.org  Wed Oct 28 11:12:53 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 28 Oct 2009 11:12:53 -0700
Subject: [Freeswitch-users] SIP provider with extern rtp server
In-Reply-To: <FA2C693B-FCE1-4BDD-BCE5-302AC699CEDB@myrvold.org>
References: <A2D2E918-B494-4179-96BA-CA3A7717846C@myrvold.org>
	<bf6923ed0910270658t61bb6687ne0b7d3ec5090e2ff@mail.gmail.com>
	<CB8008BD-4066-4B5B-9043-214CFE938CD8@myrvold.org>
	<bf6923ed0910271034w6c7e0cc5t685a5377764755e6@mail.gmail.com>
	<FA2C693B-FCE1-4BDD-BCE5-302AC699CEDB@myrvold.org>
Message-ID: <87f2f3b90910281112g6e72d22elcfd653991ecd50cc@mail.gmail.com>

On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold <ivan at myrvold.org> wrote:

> Here is a debug log from a call from an internal phone out to an
> external (my iPhone with nbr 91316356):
> http://pastebin.freeswitch.org/108578
>
> Ivan
>
> Uh... you wanna try that PB number again?
-MC
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From msc at freeswitch.org  Wed Oct 28 11:17:54 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 28 Oct 2009 11:17:54 -0700
Subject: [Freeswitch-users] Problem with hangin bri
In-Reply-To: <69a9ce230910280946h5eae8c58m72f9b9492f08a329@mail.gmail.com>
References: <69a9ce230910280946h5eae8c58m72f9b9492f08a329@mail.gmail.com>
Message-ID: <87f2f3b90910281117i38eabfaalbf412adcc7d608fe@mail.gmail.com>

Okay, obligatory questions:
Which version of FreeSWITCH are you running?
Which PRI library are you using?
Which BRI card are you using?

-MC

On Wed, Oct 28, 2009 at 9:46 AM, Jakub Pawli?ski <jakub at pawlinski.pl> wrote:

> Hi,
> I have some problems with bri status. I have 3 chanel isdn modem, and
> zaptel compatible quad bri card.  I can invoke calls from my voip phone to
> cell phone, and vice versa, but when i make inbound and outbound connection
> in nearly same time something goes wrong with chanells and after few calls
> all of them has hangup status.
>
> There is log about that in attachement, see "is already in use waiting for
> it to become available." phrase. Time of this event is about 14:43:35.
> Unload and Load open_zap module helped, but its not an solution because of
> lost connections.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Oct 28 11:23:54 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 28 Oct 2009 11:23:54 -0700
Subject: [Freeswitch-users] meaning of created_time channel variable.
In-Reply-To: <1452e2980910260020m624a6a1bk2c3a5c36cb1cb5cf@mail.gmail.com>
References: <1452e2980910260020m624a6a1bk2c3a5c36cb1cb5cf@mail.gmail.com>
Message-ID: <87f2f3b90910281123j6b5b49f6xbc369d9a1bb77e5a@mail.gmail.com>

On Mon, Oct 26, 2009 at 12:20 AM, velusamy velu <velu.technical at gmail.com>wrote:

> Dear All,
>      What is the value of created_time channel variable? Is this epoch
> seconds?
>

It's epoch microseconds. Divide by 1,000,000 to get epoch seconds...
-MC
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From larclap at yahoo.com  Wed Oct 28 11:42:01 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Wed, 28 Oct 2009 11:42:01 -0700
Subject: [Freeswitch-users] Problem with inbound call answered but
	no	sound
In-Reply-To: <8D1A994D-2DCD-4016-9BEF-D6FBD3FBAA1E@freeswitch.org>
References: <016e01ca56ad$56362380$02a26a80$@com>	<168575F8-6507-4AD2-B448-2D6FDA0152B6@freeswitch.org>	<018801ca56b2$1e63e0f0$5b2ba2d0$@com>	<BACDC899-CE4F-42E1-B2A0-C0C0E31F8787@freeswitch.org>	<00d601ca571e$519f7300$f4de5900$@com>
	<8D1A994D-2DCD-4016-9BEF-D6FBD3FBAA1E@freeswitch.org>
Message-ID: <004201ca57fe$56f92a40$04eb7ec0$@com>

BroadXent ADSL 8120 ? Netscreen 5XP

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Wednesday, October 28, 2009 7:13 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Problem with inbound call answered but no
sound

 

What kind of router are you behind?

 

/b

 

On Oct 27, 2009, at 10:58 AM, Lars Zeb wrote:





Thanks for the reply, Brian.

 

Did something in FS change between v15183 and v15225 to make this occur? I
ask because this same configuration worked OK in the earlier version.

 

Lars

 

 

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From mike at jerris.com  Wed Oct 28 11:51:48 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 28 Oct 2009 14:51:48 -0400
Subject: [Freeswitch-users] error in loading spidermonkey
In-Reply-To: <a38a91d50910281039pdf28a4fh536f44d238b9a954@mail.gmail.com>
References: <a38a91d50910281039pdf28a4fh536f44d238b9a954@mail.gmail.com>
Message-ID: <8F404160-9658-4D67-A074-53022C2F5560@jerris.com>

The libcurl is broken on your distro.  You can fix this by configuring  
with --without-libcurl which will use our working in tree copy instead  
of the broken one from your distro.

Mike

On Oct 28, 2009, at 1:39 PM, Erwin Davis wrote:

> Hi, I got an error in loading mod_spidermonkey. my fs in ver 1.04  
> runs on fedora 8 VM. Any clue? thanks,
>
> [CRIT] switch_loadable_module.c:871 Error Loading module /usr/local/ 
> freeswitch/mod/mod_spidermonkey.so
> **/usr/lib/libnss3.so: undefined symbol: PR_UnloadLibrary**
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From mi+thun at aldan.algebra.com  Wed Oct 28 10:30:39 2009
From: mi+thun at aldan.algebra.com (Mikhail T.)
Date: Wed, 28 Oct 2009 13:30:39 -0400
Subject: [Freeswitch-users] FreeSWITCH on FreeBSD
Message-ID: <4AE87FBF.1020701@aldan.algebra.com>

Hello!

I'm researching a phone system for an organization with 8-12 members.
The place has an Internet connection via Verizon FIOS. My first choice
of the operating system is FreeBSD (preferably -- on amd64 platform).
FreeBSD's port (net/freeswitch) seems nice and currently installs
version 1.0.4.3. Although fairly proficient with Unix and FreeBSD in
particular, I'm very new to the field of computer telephony, so my
questions are most likely to be rather naive. I apologize...

    * What hardware is known to work well on FreeBSD?
    * What else -- other than a computer with a phone card -- will I need?
    * Do end-user phones have to be regular analog phones (all
      plugged-in to the computer), or IP-phones, such as made by Avaya
      (connecting via an Ethernet switch)?
    * How is the computer connecting to the world? Do I plug-in the
      incoming analog phone line(s) somewhere, or is it going to go
      through the Internet?
    * If the connection to the world is over the Internet, don't we need
      an account with someone else, or is Verizon known to work with
      FreeSWITCH-based PBX-systems directly?
    * Should I not simply buy a used Avaya-system like this:

http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=110450339950&_trksid=p2759.l1259
      What will I gain and what will I lose (other than personal
      satisfaction, of course) by going that route instead of building
      my own box?

Thanks a lot! Yours,

    -mi





From Suneel.Papineni at mettoni.com  Wed Oct 28 10:42:08 2009
From: Suneel.Papineni at mettoni.com (Suneel Papineni)
Date: Wed, 28 Oct 2009 17:42:08 -0000
Subject: [Freeswitch-users] Network cable disconnection
Message-ID: <3181A30B8C35AB4AA8577B78DDF4613805FCA8F7@nickel.mettonigroup.com>

Hi,

 

I have configured FreeSwitch and played it by making calls. Every thing
went fine. Then I tried to see what happens if network cable is
unplugged. Freeswitch shown an error saying ping failed to the gateway
and IP address is changed to local system IP. After sometime network
cable plugged back in and in another 30 seconds freeswitch has shown it
got the IP of gateway and changed back from local IP to gateway IP.
After that when I tried to make calls, calls are not going through. Even
there is no log displayed in Freeswitch. Please find below the logs...

 

freeswitch at QA360BASExp> 2009-10-28 16:46:13.156250 [WARNING]
sofia.c:2814 Ping failed ihub-trunk

2009-10-28 16:46:13.234375 [INFO] mod_sofia.c:3160 IP change detected
[192.168.12.10]->[192.168.20.138] []->[]

2009-10-28 16:46:13.265625 [NOTICE] sofia_glue.c:3444 Reload XML
[Success]

2009-10-28 16:46:13.265625 [INFO] switch_time.c:661 Timezone reloaded 0
definitions

2009-10-28 16:46:14.156250 [NOTICE] sofia.c:921 Waiting for worker
thread

2009-10-28 16:46:14.156250 [NOTICE] sofia_glue.c:3505 deleted gateway
ihub-trunk

2009-10-28 16:46:14.156250 [NOTICE] sofia_reg.c:2122 Added gateway
'ihub-trunk' to profile 'TrunkExternal'

2009-10-28 16:46:14.156250 [NOTICE] sofia.c:2767 Started Profile
TrunkExternal [sofia_reg_TrunkExternal]

2009-10-28 16:46:14.156250 [ERR] sofia.c:805 Error Creating SIP UA for
profile: TrunkExternal

2009-10-28 16:46:14.156250 [NOTICE] sofia_glue.c:3505 deleted gateway
ihub-trunk

2009-10-28 16:46:53.234375 [INFO] mod_sofia.c:3160 IP change detected
[192.168.20.138]->[192.168.12.10] []->[]

2009-10-28 16:46:53.281250 [NOTICE] sofia_glue.c:3444 Reload XML
[Success]

2009-10-2816:46:53.281250 [INFO] switch_time.c:661 Timezone reloaded 0
definitions

 

--After this there were no messages

 

Calls went through if I reload freeswitch. 

 

Another surprise thing is I checked for the active ports in the system
before and after cable unplugged. Port 5060 disappeared after cable is
plugged out. This doesn't display even after cable is plugged back in. 

 

Thanks & Regards

Suneel

 

 


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From marty at makolink.com  Wed Oct 28 11:17:14 2009
From: marty at makolink.com (Martin Hickman)
Date: Wed, 28 Oct 2009 18:17:14 +0000
Subject: [Freeswitch-users] Multi-tenancy context manipulation
Message-ID: <1256753834.7175.21.camel@kombu>

Hello, it's my first post, and I'm quite new to Freeswitch...

I'm looking for a bit of direction / advice on Multi-tenant set-ups. I
have configured Freeswitch as per the fine Multi-tenant guide on the
Wiki, but this leaves some aspects such as transfers and IVR still in
the default context.

For example, transfering a call that is passed to features.xml, or
connecting to an extension thorough demo_ivr will fail if the call is
left in the default context.

Would you recommend a separate features.xml for each tenants context? If
the features provided by features.xml and the ivr are the same for all
tenants, could you advise on the best way to implement one features.xml
for all tenants?

Kind regards,
Marty Hickman




From jerry.richards at teotech.com  Wed Oct 28 13:37:25 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 28 Oct 2009 13:37:25 -0700
Subject: [Freeswitch-users] IVR Intro Clipped
Message-ID: <93CC79CC678F4CAEAC236BEEFBA955C9@greyhawk.tonecommander.com>


I notice that when I call IVR from the PSTN, the "Welcome to Freeswitch..."
introduction is clipped at the beginning, so it sounds like "come to
Freeswitch".  If I call 5000 internally, then I always hear the full
introduction.  What can I do to resolve this?

My XML config looks like:

<extension name="ivr_demo">
   <condition field="destination_number" expression="5000">
      <action application="answer"/>
      <action application="start_dtmf"/>
      <action application="ivr" data="demo_ivr"/>
   </condition>
</extension>

Best Regards,
Jerry




From ivan at myrvold.org  Wed Oct 28 13:51:15 2009
From: ivan at myrvold.org (Ivan C Myrvold)
Date: Wed, 28 Oct 2009 21:51:15 +0100
Subject: [Freeswitch-users] SIP provider with extern rtp server
In-Reply-To: <87f2f3b90910281112g6e72d22elcfd653991ecd50cc@mail.gmail.com>
References: <A2D2E918-B494-4179-96BA-CA3A7717846C@myrvold.org>
	<bf6923ed0910270658t61bb6687ne0b7d3ec5090e2ff@mail.gmail.com>
	<CB8008BD-4066-4B5B-9043-214CFE938CD8@myrvold.org>
	<bf6923ed0910271034w6c7e0cc5t685a5377764755e6@mail.gmail.com>
	<FA2C693B-FCE1-4BDD-BCE5-302AC699CEDB@myrvold.org>
	<87f2f3b90910281112g6e72d22elcfd653991ecd50cc@mail.gmail.com>
Message-ID: <8AC09649-2585-4BE7-A959-A7AC41650789@myrvold.org>

Oh, what happened to it?

Anyway, here is a new pb:

http://pastebin.freeswitch.org/10867

Ivan
Den 28. okt. 2009 kl. 19:12 skrev Michael Collins:

>
>
> On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold <ivan at myrvold.org>  
> wrote:
> Here is a debug log from a call from an internal phone out to an
> external (my iPhone with nbr 91316356):
> http://pastebin.freeswitch.org/108578
>
> Ivan
>
> Uh... you wanna try that PB number again?
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From brian at freeswitch.org  Wed Oct 28 13:51:16 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 28 Oct 2009 15:51:16 -0500
Subject: [Freeswitch-users] IVR Intro Clipped
In-Reply-To: <93CC79CC678F4CAEAC236BEEFBA955C9@greyhawk.tonecommander.com>
References: <93CC79CC678F4CAEAC236BEEFBA955C9@greyhawk.tonecommander.com>
Message-ID: <3A585780-F132-4D4B-AD97-66904A9C3446@freeswitch.org>

Sleep 1000 ms... we usually bring up media too fast before the other  
end is ready.

/b

On Oct 28, 2009, at 3:37 PM, Jerry Richards wrote:

>
> I notice that when I call IVR from the PSTN, the "Welcome to  
> Freeswitch..."
> introduction is clipped at the beginning, so it sounds like "come to
> Freeswitch".  If I call 5000 internally, then I always hear the full
> introduction.  What can I do to resolve this?
>
> My XML config looks like:
>
> <extension name="ivr_demo">
>   <condition field="destination_number" expression="5000">
>      <action application="answer"/>
>      <action application="start_dtmf"/>
>      <action application="ivr" data="demo_ivr"/>
>   </condition>
> </extension>
>
> Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From mariusz_kolo at wp.pl  Wed Oct 28 14:12:22 2009
From: mariusz_kolo at wp.pl (=?UTF-8?B?TWFyaXVzeiBLb8WCb2R6aWVqY3p5aw==?=)
Date: Wed, 28 Oct 2009 22:12:22 +0100
Subject: [Freeswitch-users] Problem with hangin bri
In-Reply-To: <87f2f3b90910281117i38eabfaalbf412adcc7d608fe@mail.gmail.com>
References: <69a9ce230910280946h5eae8c58m72f9b9492f08a329@mail.gmail.com>
	<87f2f3b90910281117i38eabfaalbf412adcc7d608fe@mail.gmail.com>
Message-ID: <4AE8B3B6.9000106@wp.pl>

Hi

I'm also working on this project, so i can answer your questions

Which version of FreeSWITCH are you running?

FreeSWITCH Version 1.0.trunk (15246)

Which PRI library are you using?
openzap Native stack

openzap.conf

[span zt BRI1]
trunk_type => bri
b-channel => 1-2
d-channel=> 3

openzap.conf

<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    <param name="debug" value="0"/>
  </settings>

   <pri_spans>
    <span id="1">
     <param name="mode" value="user"/>
     <param name="dialect" value="q931"/>
     <param name="dialplan" value="XML"/>
     <param name="context" value="default"/>
    </span>
  </pri_spans>
</configuration>

Which BRI card are you using?

Producer: http://www.phoniceq.com/
card model: http://quadbri.phoniceq.com/

Card instalation process (instruction from producer)

1) download bristuff staff from

http://junghanns.net/downloads/bristuff-0.4.0-RC3h.tar.gz
or
http://junghanns.net/downloads/bristuff-0.3.0-PRE-1y-z.tar.gz

unpack it and go to bristuff-*

2) download patcher from
http://quadbri.phoniceq.com/driver/bristuff/qozap-bristuff-0.3.0-PRE-1y-j-enableLEDS.patch

patch it using 

patch -p0 < qozap-bristuff-0.3.0-PRE-1y-j-enableLEDS.patch

3) you can check card using zttest (result should be 99.x)

Producer has said, that we are first client, it wants to use this card in freeswitch 

we are using 1 port (S/T interface). Our NT is "NT1 plus 2b1q" 


Thanks

Michael Collins pisze:
> Okay, obligatory questions:
> Which version of FreeSWITCH are you running?
> Which PRI library are you using?
> Which BRI card are you using?
>
> -MC
>
> On Wed, Oct 28, 2009 at 9:46 AM, Jakub Pawli?ski <jakub at pawlinski.pl 
> <mailto:jakub at pawlinski.pl>> wrote:
>
>     Hi,
>     I have some problems with bri status. I have 3 chanel isdn modem,
>     and zaptel compatible quad bri card.  I can invoke calls from my
>     voip phone to cell phone, and vice versa, but when i make inbound
>     and outbound connection in nearly same time something goes wrong
>     with chanells and after few calls all of them has hangup status.
>
>     There is log about that in attachement, see "is already in use
>     waiting for it to become available." phrase. Time of this event is
>     about 14:43:35. Unload and Load open_zap module helped, but its
>     not an solution because of lost connections.
>
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


-- 
Mariusz Ko?odziejczyk

Advanced Developing Architecture S.C.

tel.   : +48 609 381 316
e-mail : mariusz_kolo at wp.pl




From jerry.richards at teotech.com  Wed Oct 28 14:36:17 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 28 Oct 2009 14:36:17 -0700
Subject: [Freeswitch-users] IVR Intro Clipped
In-Reply-To: <3A585780-F132-4D4B-AD97-66904A9C3446@freeswitch.org>
References: <93CC79CC678F4CAEAC236BEEFBA955C9@greyhawk.tonecommander.com>
	<3A585780-F132-4D4B-AD97-66904A9C3446@freeswitch.org>
Message-ID: <5B061E041F8E4D8B81464B910D32F47E@greyhawk.tonecommander.com>


I modified my dialplan as shown, but the clipping persists.  Should the
sleep be placed somewhere else?

<extension name="ivr_demo">
  <condition field="destination_number" expression="5000">
     <action application="sleep" data="1000"\>
     <action application="answer"/>
     <action application="start_dtmf"/>
     <action application="ivr" data="demo_ivr"/>
  </condition>
</extension>

Best Regards,
Jerry
 

-----Original Message-----
From: Brian West [mailto:brian at freeswitch.org] 
Sent: Wednesday, October 28, 2009 1:51 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] IVR Intro Clipped

Sleep 1000 ms... we usually bring up media too fast before the other end is
ready.

/b

On Oct 28, 2009, at 3:37 PM, Jerry Richards wrote:

>
> I notice that when I call IVR from the PSTN, the "Welcome to 
> Freeswitch..."
> introduction is clipped at the beginning, so it sounds like "come to 
> Freeswitch".  If I call 5000 internally, then I always hear the full 
> introduction.  What can I do to resolve this?
>
> My XML config looks like:
>
> <extension name="ivr_demo">
>   <condition field="destination_number" expression="5000">
>      <action application="answer"/>
>      <action application="start_dtmf"/>
>      <action application="ivr" data="demo_ivr"/>
>   </condition>
> </extension>
>
> Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> users
> http://www.freeswitch.org







From frank at carmickle.com  Wed Oct 28 14:45:31 2009
From: frank at carmickle.com (Frank Carmickle)
Date: Wed, 28 Oct 2009 17:45:31 -0400
Subject: [Freeswitch-users] IVR Intro Clipped
In-Reply-To: <5B061E041F8E4D8B81464B910D32F47E@greyhawk.tonecommander.com>
References: <93CC79CC678F4CAEAC236BEEFBA955C9@greyhawk.tonecommander.com>
	<3A585780-F132-4D4B-AD97-66904A9C3446@freeswitch.org>
	<5B061E041F8E4D8B81464B910D32F47E@greyhawk.tonecommander.com>
Message-ID: <20091028214531.GM13895@base.carmickle.com>

Hello

Yes.  Answer first.

--FC


On Wed, Oct 28, Jerry Richards wrote:
> 
> I modified my dialplan as shown, but the clipping persists.  Should the
> sleep be placed somewhere else?
> 
> <extension name="ivr_demo">
>   <condition field="destination_number" expression="5000">
>      <action application="sleep" data="1000"\>
>      <action application="answer"/>
>      <action application="start_dtmf"/>
>      <action application="ivr" data="demo_ivr"/>
>   </condition>
> </extension>
> 
> Best Regards,
> Jerry
>  
> 
> -----Original Message-----
> From: Brian West [mailto:brian at freeswitch.org] 
> Sent: Wednesday, October 28, 2009 1:51 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] IVR Intro Clipped
> 
> Sleep 1000 ms... we usually bring up media too fast before the other end is
> ready.
> 
> /b
> 
> On Oct 28, 2009, at 3:37 PM, Jerry Richards wrote:
> 
> >
> > I notice that when I call IVR from the PSTN, the "Welcome to 
> > Freeswitch..."
> > introduction is clipped at the beginning, so it sounds like "come to 
> > Freeswitch".  If I call 5000 internally, then I always hear the full 
> > introduction.  What can I do to resolve this?
> >
> > My XML config looks like:
> >
> > <extension name="ivr_demo">
> >   <condition field="destination_number" expression="5000">
> >      <action application="answer"/>
> >      <action application="start_dtmf"/>
> >      <action application="ivr" data="demo_ivr"/>
> >   </condition>
> > </extension>
> >
> > Best Regards,
> > Jerry
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
> 
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From sicfslist at gmail.com  Wed Oct 28 14:47:53 2009
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Wed, 28 Oct 2009 16:47:53 -0500
Subject: [Freeswitch-users] IVR Intro Clipped
In-Reply-To: <5B061E041F8E4D8B81464B910D32F47E@greyhawk.tonecommander.com>
References: <93CC79CC678F4CAEAC236BEEFBA955C9@greyhawk.tonecommander.com>	<3A585780-F132-4D4B-AD97-66904A9C3446@freeswitch.org>
	<5B061E041F8E4D8B81464B910D32F47E@greyhawk.tonecommander.com>
Message-ID: <4AE8BC09.1060602@gmail.com>

Jerry,

Put the sleep after the answer.  That should fix it.

Shelby

Jerry Richards wrote:
> I modified my dialplan as shown, but the clipping persists.  Should the
> sleep be placed somewhere else?
>
> <extension name="ivr_demo">
>   <condition field="destination_number" expression="5000">
>      <action application="sleep" data="1000"\>
>      <action application="answer"/>
>      <action application="start_dtmf"/>
>      <action application="ivr" data="demo_ivr"/>
>   </condition>
> </extension>
>
> Best Regards,
> Jerry
>  
>
> -----Original Message-----
> From: Brian West [mailto:brian at freeswitch.org] 
> Sent: Wednesday, October 28, 2009 1:51 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] IVR Intro Clipped
>
> Sleep 1000 ms... we usually bring up media too fast before the other end is
> ready.
>
> /b
>
> On Oct 28, 2009, at 3:37 PM, Jerry Richards wrote:
>
>   
>> I notice that when I call IVR from the PSTN, the "Welcome to 
>> Freeswitch..."
>> introduction is clipped at the beginning, so it sounds like "come to 
>> Freeswitch".  If I call 5000 internally, then I always hear the full 
>> introduction.  What can I do to resolve this?
>>
>> My XML config looks like:
>>
>> <extension name="ivr_demo">
>>   <condition field="destination_number" expression="5000">
>>      <action application="answer"/>
>>      <action application="start_dtmf"/>
>>      <action application="ivr" data="demo_ivr"/>
>>   </condition>
>> </extension>
>>
>> Best Regards,
>> Jerry
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>>     
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   




From cliff at develix.com  Wed Oct 28 14:55:17 2009
From: cliff at develix.com (Cliff Wells)
Date: Wed, 28 Oct 2009 14:55:17 -0700
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <000f01ca5740$fd9d7a40$f8d86ec0$@net>
References: <bf6923ed0910261446x6a184ecdnee2e73e904406246@mail.gmail.com>
	<bf6923ed0910261501k1ff8c892rfc198c31e30ab286@mail.gmail.com>
	<1256673110.9746.29.camel@portable-evil>
	<000f01ca5740$fd9d7a40$f8d86ec0$@net>
Message-ID: <1256766917.28102.29.camel@portable-evil>

That's really nice looking software.   Thanks for the pointer.

Cliff


On Tue, 2009-10-27 at 16:06 -0400, Gregory Boehnlein wrote:
> I'm fond of Vqmanager from ManagEngine. It is a passive SIP monitor. I.E.
> you mirror the ports that your FS or Asterisk boxes, and VQmanager sniffs
> the mirrors, tracking all sorts of good data.
> 
> You can install it on a Centos box, and get a free trial.
> 
> http://www.manageengine.com/products/vqmanager/index.html
> 
> What is really cool is that it actually monitors the RTP/RTCP as well as all
> of the SIP headers and archives the calls, so you can look at calls from
> several days ago and see EXACTLY what happened on them. I have used this
> extensively to pinpoint bad Level 3 and X/O media gateways.. Much better
> than trying to sniff packets in real-time and MAYBE catch a problem..
> 
> I've also used it to find/fix several SIP issues w/ odd endpoints.. Very
> easy to see..
> 
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> > users-bounces at lists.freeswitch.org] On Behalf Of Cliff Wells
> > Sent: Tuesday, October 27, 2009 3:52 PM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] Estimating Call Capacity
> > 
> > A little off-topic, but since call-capacity is the subject, what are
> > people using to analyze their CDR's to discover this?   I'm handling
> > about 30k calls per day but have only a bandwidth-based guesstimate of
> > the peak number of concurrent calls I'm handling.
> > 
> > If there's an open source solution, I'd appreciate a pointer.
> > 
> > Regards,
> > Cliff
> > 
> > On Mon, 2009-10-26 at 18:01 -0400, Eliot Gable wrote:
> > > Although, FYI, I just benchmarked mod_xml_curl on a separate web app
> > > server from FS with FS on a Dell R710 with their current best
> > > processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and
> > 32
> > > GB memory. The web app server is less than half the power of the
> > R710.
> > > I maxed the web app server at 300 calls per second (both setting up
> > > and tearing down) and the R710 running FS was 65% idle. No audio was
> > > being proxied through FS, though. If I were running the web app
> > server
> > > on an equivalent R710, they probably would have been on-par with each
> > > other in performance. Extrapolating, I expect that in such a case I
> > > should be able to get at least 650 CPS out of FS, though for
> > > production I would probably limit it to 400 CPS or less so I leave
> > > room for miscellaneous tasks. I maxed out the R710 at over 16,000
> > > simultaneous calls (again, no audio proxying) but the only reason I
> > > couldn't do more was because I hit some sort of thread creation limit
> > > in Linux. There was about 17 GB of memory used for this many calls.
> > > This should give you some ballpark idea of what you can accomplish
> > > with FS.
> > >
> > > At some point, I will track down and resolve the thread creation
> > > issue, at which time I believe call limits will be limited either by
> > a
> > > complex combination of available memory, the speed of the processor,
> > > the cost of thread context switching, calls per second setup rate,
> > and
> > > call duration.
> > >
> > > --
> > > Eliot Gable
> > >
> > > > -----Original Message-----
> > > >
> > > > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> > Giovanni Maruzzelli
> > > >
> > > > Sent: Monday, October 26, 2009 4:56 PM
> > > >
> > > > To: freeswitch-users at lists.freeswitch.org
> > > >
> > > > Subject: Re: [Freeswitch-users] Estimating Call Capacity
> > > >
> > > >
> > > >
> > > > On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur
> > > >
> > > > <vinuth.madinur at gmail.com> wrote:
> > > >
> > > > > Here are a few benchmarks that I had stumbled upon.
> > > >
> > > > >
> > http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+
> > x2200+M2
> > > >
> > > >
> > > >
> > > > Please remember NO benchmarks are endorsed by the FS community or
> > > >
> > > > developers, because there are just too many variables, and a simple
> > > >
> > > > figure is just useful for marketing hype, not for real
> > dimensioning.
> > > >
> > > >
> > > >
> > > > You MUST do your own benchmarking, so you get an idea about how to
> > > >
> > > > dimension for your own use case and hardware.
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > > Thanks,
> > > >
> > > > > Vinuth.
> > > >
> > > > >
> > > >
> > > > > On Tue, Oct 27, 2009 at 1:43 AM, Brian West
> > <brian at freeswitch.org> wrote:
> > > >
> > > > >>
> > > >
> > > > >> I highly doubt it... You can wait for someone to post their
> > results
> > > >
> > > > >> but in the end you'll have to do your own load testing because
> > not
> > > >
> > > > >> everyone's numbers will jive with your use case.  Which is the
> > reason
> > > >
> > > > >> the project never posts or endorses a set call count.
> > > >
> > > > >>
> > > >
> > > > >> /b
> > > >
> > > > >>
> > > >
> > > > >> On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote:
> > > >
> > > > >>
> > > >
> > > > >> > Are there any benchmarking test results available publicly?
> > > >
> > > > >> > ________________________________________
> > > >
> > > > >>
> > > >
> > > > >>
> > > >
> > > > >> _______________________________________________
> > > >
> > > > >> FreeSWITCH-users mailing list
> > > >
> > > > >> FreeSWITCH-users at lists.freeswitch.org
> > > >
> > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >
> > > > >>
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > > >
> > > > >> http://www.freeswitch.org
> > > >
> > > > >
> > > >
> > > > >
> > > >
> > > > > _______________________________________________
> > > >
> > > > > FreeSWITCH-users mailing list
> > > >
> > > > > FreeSWITCH-users at lists.freeswitch.org
> > > >
> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >
> > > > >
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > > >
> > > > > http://www.freeswitch.org
> > > >
> > > > >
> > > >
> > > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > --
> > > >
> > > > Sincerely,
> > > >
> > > >
> > > >
> > > > Giovanni Maruzzelli
> > > >
> > > > Cell : +39-347-2665618
> > > >
> > > >
> > > >
> > > > _______________________________________________
> > > >
> > > > FreeSWITCH-users mailing list
> > > >
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > >
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >
> > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > > >
> > > > http://www.freeswitch.org
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > > http://www.freeswitch.org
> > --
> > http://www.google.com/search?q=vonage+sucks
> > 
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
> > 
> > --
> > This message has been scanned for viruses and
> > dangerous content by N2Net Mailshield, and is
> > believed to be clean.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 
http://www.google.com/search?q=vonage+sucks




From cliff at develix.com  Wed Oct 28 14:55:17 2009
From: cliff at develix.com (Cliff Wells)
Date: Wed, 28 Oct 2009 14:55:17 -0700
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <000f01ca5740$fd9d7a40$f8d86ec0$@net>
References: <bf6923ed0910261446x6a184ecdnee2e73e904406246@mail.gmail.com>
	<bf6923ed0910261501k1ff8c892rfc198c31e30ab286@mail.gmail.com>
	<1256673110.9746.29.camel@portable-evil>
	<000f01ca5740$fd9d7a40$f8d86ec0$@net>
Message-ID: <1256766917.28102.29.camel@portable-evil>

That's really nice looking software.   Thanks for the pointer.

Cliff


On Tue, 2009-10-27 at 16:06 -0400, Gregory Boehnlein wrote:
> I'm fond of Vqmanager from ManagEngine. It is a passive SIP monitor. I.E.
> you mirror the ports that your FS or Asterisk boxes, and VQmanager sniffs
> the mirrors, tracking all sorts of good data.
> 
> You can install it on a Centos box, and get a free trial.
> 
> http://www.manageengine.com/products/vqmanager/index.html
> 
> What is really cool is that it actually monitors the RTP/RTCP as well as all
> of the SIP headers and archives the calls, so you can look at calls from
> several days ago and see EXACTLY what happened on them. I have used this
> extensively to pinpoint bad Level 3 and X/O media gateways.. Much better
> than trying to sniff packets in real-time and MAYBE catch a problem..
> 
> I've also used it to find/fix several SIP issues w/ odd endpoints.. Very
> easy to see..
> 
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> > users-bounces at lists.freeswitch.org] On Behalf Of Cliff Wells
> > Sent: Tuesday, October 27, 2009 3:52 PM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] Estimating Call Capacity
> > 
> > A little off-topic, but since call-capacity is the subject, what are
> > people using to analyze their CDR's to discover this?   I'm handling
> > about 30k calls per day but have only a bandwidth-based guesstimate of
> > the peak number of concurrent calls I'm handling.
> > 
> > If there's an open source solution, I'd appreciate a pointer.
> > 
> > Regards,
> > Cliff
> > 
> > On Mon, 2009-10-26 at 18:01 -0400, Eliot Gable wrote:
> > > Although, FYI, I just benchmarked mod_xml_curl on a separate web app
> > > server from FS with FS on a Dell R710 with their current best
> > > processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and
> > 32
> > > GB memory. The web app server is less than half the power of the
> > R710.
> > > I maxed the web app server at 300 calls per second (both setting up
> > > and tearing down) and the R710 running FS was 65% idle. No audio was
> > > being proxied through FS, though. If I were running the web app
> > server
> > > on an equivalent R710, they probably would have been on-par with each
> > > other in performance. Extrapolating, I expect that in such a case I
> > > should be able to get at least 650 CPS out of FS, though for
> > > production I would probably limit it to 400 CPS or less so I leave
> > > room for miscellaneous tasks. I maxed out the R710 at over 16,000
> > > simultaneous calls (again, no audio proxying) but the only reason I
> > > couldn't do more was because I hit some sort of thread creation limit
> > > in Linux. There was about 17 GB of memory used for this many calls.
> > > This should give you some ballpark idea of what you can accomplish
> > > with FS.
> > >
> > > At some point, I will track down and resolve the thread creation
> > > issue, at which time I believe call limits will be limited either by
> > a
> > > complex combination of available memory, the speed of the processor,
> > > the cost of thread context switching, calls per second setup rate,
> > and
> > > call duration.
> > >
> > > --
> > > Eliot Gable
> > >
> > > > -----Original Message-----
> > > >
> > > > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> > Giovanni Maruzzelli
> > > >
> > > > Sent: Monday, October 26, 2009 4:56 PM
> > > >
> > > > To: freeswitch-users at lists.freeswitch.org
> > > >
> > > > Subject: Re: [Freeswitch-users] Estimating Call Capacity
> > > >
> > > >
> > > >
> > > > On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur
> > > >
> > > > <vinuth.madinur at gmail.com> wrote:
> > > >
> > > > > Here are a few benchmarks that I had stumbled upon.
> > > >
> > > > >
> > http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+
> > x2200+M2
> > > >
> > > >
> > > >
> > > > Please remember NO benchmarks are endorsed by the FS community or
> > > >
> > > > developers, because there are just too many variables, and a simple
> > > >
> > > > figure is just useful for marketing hype, not for real
> > dimensioning.
> > > >
> > > >
> > > >
> > > > You MUST do your own benchmarking, so you get an idea about how to
> > > >
> > > > dimension for your own use case and hardware.
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > > Thanks,
> > > >
> > > > > Vinuth.
> > > >
> > > > >
> > > >
> > > > > On Tue, Oct 27, 2009 at 1:43 AM, Brian West
> > <brian at freeswitch.org> wrote:
> > > >
> > > > >>
> > > >
> > > > >> I highly doubt it... You can wait for someone to post their
> > results
> > > >
> > > > >> but in the end you'll have to do your own load testing because
> > not
> > > >
> > > > >> everyone's numbers will jive with your use case.  Which is the
> > reason
> > > >
> > > > >> the project never posts or endorses a set call count.
> > > >
> > > > >>
> > > >
> > > > >> /b
> > > >
> > > > >>
> > > >
> > > > >> On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote:
> > > >
> > > > >>
> > > >
> > > > >> > Are there any benchmarking test results available publicly?
> > > >
> > > > >> > ________________________________________
> > > >
> > > > >>
> > > >
> > > > >>
> > > >
> > > > >> _______________________________________________
> > > >
> > > > >> FreeSWITCH-users mailing list
> > > >
> > > > >> FreeSWITCH-users at lists.freeswitch.org
> > > >
> > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >
> > > > >>
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > > >
> > > > >> http://www.freeswitch.org
> > > >
> > > > >
> > > >
> > > > >
> > > >
> > > > > _______________________________________________
> > > >
> > > > > FreeSWITCH-users mailing list
> > > >
> > > > > FreeSWITCH-users at lists.freeswitch.org
> > > >
> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >
> > > > >
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > > >
> > > > > http://www.freeswitch.org
> > > >
> > > > >
> > > >
> > > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > --
> > > >
> > > > Sincerely,
> > > >
> > > >
> > > >
> > > > Giovanni Maruzzelli
> > > >
> > > > Cell : +39-347-2665618
> > > >
> > > >
> > > >
> > > > _______________________________________________
> > > >
> > > > FreeSWITCH-users mailing list
> > > >
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > >
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >
> > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > > >
> > > > http://www.freeswitch.org
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > > http://www.freeswitch.org
> > --
> > http://www.google.com/search?q=vonage+sucks
> > 
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
> > 
> > --
> > This message has been scanned for viruses and
> > dangerous content by N2Net Mailshield, and is
> > believed to be clean.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 
http://www.google.com/search?q=vonage+sucks




From cliff at develix.com  Wed Oct 28 14:58:09 2009
From: cliff at develix.com (Cliff Wells)
Date: Wed, 28 Oct 2009 14:58:09 -0700
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <4AE755C3.6060307@gmail.com>
References: <bf6923ed0910261446x6a184ecdnee2e73e904406246@mail.gmail.com>
	<bf6923ed0910261501k1ff8c892rfc198c31e30ab286@mail.gmail.com>
	<1256673110.9746.29.camel@portable-evil> <4AE755C3.6060307@gmail.com>
Message-ID: <1256767089.17442.2.camel@portable-evil>

Hi Shelby,

Thanks!  That's pretty useful.   I also note that this same info is
available from the CLI, although I was curious as to what some of the
numbers indicated:

UP 0 years, 13 days, 19 hours, 25 minutes, 48 seconds, 887 milliseconds, 975 microseconds
529509 session(s) since startup
26 session(s) 0/30
1000 session(s) max

Specifically, 26 sessions 0/30... I take it this means there are 26
current sessions, but I'm unsure of what the "0/30" means.

Cliff


On Tue, 2009-10-27 at 15:19 -0500, Shelby Ramsey wrote:
> Cliff,
> 
> Try using xml_rpc ... status or show channels will give you what you need.
> 
> SDR
> 
> Cliff Wells wrote:
> > A little off-topic, but since call-capacity is the subject, what are
> > people using to analyze their CDR's to discover this?   I'm handling
> > about 30k calls per day but have only a bandwidth-based guesstimate of
> > the peak number of concurrent calls I'm handling.
> >
> > If there's an open source solution, I'd appreciate a pointer.
> >
> > Regards,
> > Cliff
> >
> > On Mon, 2009-10-26 at 18:01 -0400, Eliot Gable wrote:
> >   
> >> Although, FYI, I just benchmarked mod_xml_curl on a separate web app
> >> server from FS with FS on a Dell R710 with their current best
> >> processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and 32
> >> GB memory. The web app server is less than half the power of the R710.
> >> I maxed the web app server at 300 calls per second (both setting up
> >> and tearing down) and the R710 running FS was 65% idle. No audio was
> >> being proxied through FS, though. If I were running the web app server
> >> on an equivalent R710, they probably would have been on-par with each
> >> other in performance. Extrapolating, I expect that in such a case I
> >> should be able to get at least 650 CPS out of FS, though for
> >> production I would probably limit it to 400 CPS or less so I leave
> >> room for miscellaneous tasks. I maxed out the R710 at over 16,000
> >> simultaneous calls (again, no audio proxying) but the only reason I
> >> couldn't do more was because I hit some sort of thread creation limit
> >> in Linux. There was about 17 GB of memory used for this many calls.
> >> This should give you some ballpark idea of what you can accomplish
> >> with FS.
> >>
> >> At some point, I will track down and resolve the thread creation
> >> issue, at which time I believe call limits will be limited either by a
> >> complex combination of available memory, the speed of the processor,
> >> the cost of thread context switching, calls per second setup rate, and
> >> call duration.
> >>
> >> --
> >> Eliot Gable
> >>
> >>     
> >>> -----Original Message-----
> >>>
> >>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli
> >>>
> >>> Sent: Monday, October 26, 2009 4:56 PM
> >>>
> >>> To: freeswitch-users at lists.freeswitch.org
> >>>
> >>> Subject: Re: [Freeswitch-users] Estimating Call Capacity
> >>>
> >>>
> >>>
> >>> On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur
> >>>
> >>> <vinuth.madinur at gmail.com> wrote:
> >>>
> >>>       
> >>>> Here are a few benchmarks that I had stumbled upon.
> >>>>         
> >>>> http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2
> >>>>         
> >>>
> >>> Please remember NO benchmarks are endorsed by the FS community or
> >>>
> >>> developers, because there are just too many variables, and a simple
> >>>
> >>> figure is just useful for marketing hype, not for real dimensioning.
> >>>
> >>>
> >>>
> >>> You MUST do your own benchmarking, so you get an idea about how to
> >>>
> >>> dimension for your own use case and hardware.
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>       
> >>>> Thanks,
> >>>>         
> >>>> Vinuth.
> >>>>         
> >>>> On Tue, Oct 27, 2009 at 1:43 AM, Brian West <brian at freeswitch.org> wrote:
> >>>>         
> >>>>> I highly doubt it... You can wait for someone to post their results
> >>>>>           
> >>>>> but in the end you'll have to do your own load testing because not
> >>>>>           
> >>>>> everyone's numbers will jive with your use case.  Which is the reason
> >>>>>           
> >>>>> the project never posts or endorses a set call count.
> >>>>>           
> >>>>> /b
> >>>>>           
> >>>>> On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote:
> >>>>>           
> >>>>>> Are there any benchmarking test results available publicly?
> >>>>>>             
> >>>>>> ________________________________________
> >>>>>>             
> >>>>> _______________________________________________
> >>>>>           
> >>>>> FreeSWITCH-users mailing list
> >>>>>           
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>           
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>           
> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>           
> >>>>> http://www.freeswitch.org
> >>>>>           
> >>>> _______________________________________________
> >>>>         
> >>>> FreeSWITCH-users mailing list
> >>>>         
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>         
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>         
> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>         
> >>>> http://www.freeswitch.org
> >>>>         
> >>>
> >>>
> >>>
> >>>
> >>>
> >>> --
> >>>
> >>> Sincerely,
> >>>
> >>>
> >>>
> >>> Giovanni Maruzzelli
> >>>
> >>> Cell : +39-347-2665618
> >>>
> >>>
> >>>
> >>> _______________________________________________
> >>>
> >>> FreeSWITCH-users mailing list
> >>>
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>>
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>
> >>> http://www.freeswitch.org
> >>>       
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>     
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 
http://www.google.com/search?q=vonage+sucks




From brian at freeswitch.org  Wed Oct 28 15:02:16 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 28 Oct 2009 17:02:16 -0500
Subject: [Freeswitch-users] Estimating Call Capacity
In-Reply-To: <1256767089.17442.2.camel@portable-evil>
References: <bf6923ed0910261446x6a184ecdnee2e73e904406246@mail.gmail.com>
	<bf6923ed0910261501k1ff8c892rfc198c31e30ab286@mail.gmail.com>
	<1256673110.9746.29.camel@portable-evil>
	<4AE755C3.6060307@gmail.com>
	<1256767089.17442.2.camel@portable-evil>
Message-ID: <499FBF53-6D73-4C3D-8059-EBDD5C4E3C25@freeswitch.org>

sessions per second now/max

/b

On Oct 28, 2009, at 4:58 PM, Cliff Wells wrote:

> current sessions, but I'm unsure of what the "0/30" means.




From msc at freeswitch.org  Wed Oct 28 15:28:24 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 28 Oct 2009 15:28:24 -0700
Subject: [Freeswitch-users] Problem with hangin bri
In-Reply-To: <4AE8B3B6.9000106@wp.pl>
References: <69a9ce230910280946h5eae8c58m72f9b9492f08a329@mail.gmail.com>
	<87f2f3b90910281117i38eabfaalbf412adcc7d608fe@mail.gmail.com>
	<4AE8B3B6.9000106@wp.pl>
Message-ID: <87f2f3b90910281528t633b765x7c2bb858f41ce154@mail.gmail.com>

Thanks. Can you collect debug logs of this happening? See
http://wiki.freeswitch.org/wiki/Reporting_Bugs for helpful tips on
collecting debug information. Use pastebin to dump all the log info and
reply here with the link. We don't have too many BRI users but I believe
there are a few so hopefully we can help you get up and running.
-MC

2009/10/28 Mariusz Ko?odziejczyk <mariusz_kolo at wp.pl>

> Hi
>
> I'm also working on this project, so i can answer your questions
>
> Which version of FreeSWITCH are you running?
>
> FreeSWITCH Version 1.0.trunk (15246)
>
> Which PRI library are you using?
> openzap Native stack
>
> openzap.conf
>
> [span zt BRI1]
> trunk_type => bri
> b-channel => 1-2
> d-channel=> 3
>
> openzap.conf
>
> <configuration name="openzap.conf" description="OpenZAP Configuration">
>  <settings>
>    <param name="debug" value="0"/>
>  </settings>
>
>   <pri_spans>
>    <span id="1">
>     <param name="mode" value="user"/>
>     <param name="dialect" value="q931"/>
>     <param name="dialplan" value="XML"/>
>     <param name="context" value="default"/>
>    </span>
>  </pri_spans>
> </configuration>
>
> Which BRI card are you using?
>
> Producer: http://www.phoniceq.com/
> card model: http://quadbri.phoniceq.com/
>
> Card instalation process (instruction from producer)
>
> 1) download bristuff staff from
>
> http://junghanns.net/downloads/bristuff-0.4.0-RC3h.tar.gz
> or
> http://junghanns.net/downloads/bristuff-0.3.0-PRE-1y-z.tar.gz
>
> unpack it and go to bristuff-*
>
> 2) download patcher from
>
> http://quadbri.phoniceq.com/driver/bristuff/qozap-bristuff-0.3.0-PRE-1y-j-enableLEDS.patch
>
> patch it using
>
> patch -p0 < qozap-bristuff-0.3.0-PRE-1y-j-enableLEDS.patch
>
> 3) you can check card using zttest (result should be 99.x)
>
> Producer has said, that we are first client, it wants to use this card in
> freeswitch
>
> we are using 1 port (S/T interface). Our NT is "NT1 plus 2b1q"
>
>
> Thanks
>
> Michael Collins pisze:
> > Okay, obligatory questions:
> > Which version of FreeSWITCH are you running?
> > Which PRI library are you using?
> > Which BRI card are you using?
> >
> > -MC
> >
> > On Wed, Oct 28, 2009 at 9:46 AM, Jakub Pawli?ski <jakub at pawlinski.pl
> > <mailto:jakub at pawlinski.pl>> wrote:
> >
> >     Hi,
> >     I have some problems with bri status. I have 3 chanel isdn modem,
> >     and zaptel compatible quad bri card.  I can invoke calls from my
> >     voip phone to cell phone, and vice versa, but when i make inbound
> >     and outbound connection in nearly same time something goes wrong
> >     with chanells and after few calls all of them has hangup status.
> >
> >     There is log about that in attachement, see "is already in use
> >     waiting for it to become available." phrase. Time of this event is
> >     about 14:43:35. Unload and Load open_zap module helped, but its
> >     not an solution because of lost connections.
> >
> >
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> --
> Mariusz Ko?odziejczyk
>
> Advanced Developing Architecture S.C.
>
> tel.   : +48 609 381 316
> e-mail : mariusz_kolo at wp.pl
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Wed Oct 28 15:33:28 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 28 Oct 2009 15:33:28 -0700
Subject: [Freeswitch-users] Multi-tenancy context manipulation
In-Reply-To: <1256753834.7175.21.camel@kombu>
References: <1256753834.7175.21.camel@kombu>
Message-ID: <87f2f3b90910281533l36647887u85375f005824228b@mail.gmail.com>

On Wed, Oct 28, 2009 at 11:17 AM, Martin Hickman <marty at makolink.com> wrote:

> Hello, it's my first post, and I'm quite new to Freeswitch...
>
> I'm looking for a bit of direction / advice on Multi-tenant set-ups. I
> have configured Freeswitch as per the fine Multi-tenant guide on the
> Wiki, but this leaves some aspects such as transfers and IVR still in
> the default context.
>
> For example, transfering a call that is passed to features.xml, or
> connecting to an extension thorough demo_ivr will fail if the call is
> left in the default context.
>
> Would you recommend a separate features.xml for each tenants context? If
> the features provided by features.xml and the ivr are the same for all
> tenants, could you advise on the best way to implement one features.xml
> for all tenants?
>
> Kind regards,
> Marty Hickman
>
> How many contexts and tenants are you looking at? Also, will that change in
the future, that is, will you be adding tenants? Keep that in mind because
you might be looking at a good case for using mod_xml_curl...
-MC
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From bottleman at icf.org.ru  Wed Oct 28 17:26:34 2009
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Thu, 29 Oct 2009 03:26:34 +0300 (MSK)
Subject: [Freeswitch-users] mod_opal - call charged before H.225 connect
In-Reply-To: <65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
References: <65d96fc80910061027x2ab58c13s9e31c2f58e5fe01e@mail.gmail.com>
	<65d96fc80910061037w3c2d8feasc86ab5990ada0562@mail.gmail.com>
	<191c3a030910061225m3167daam55b163103fc42641@mail.gmail.com>
	<65d96fc80910061347k3b16fcf3pe527459c43470bad@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.0910290314590.32176@srv.icf.org.ru>

On 2009-10-06 22:47 +0200, Tihomir Culjaga wrote freeswitch-users at lists.fre...:


Hi, now i have another questions about fs and not sip protocols,
call scheme like this - user1-sip->fs-sip->user2, 
as i see if in sip profile enable inbound-proxy-media SDP is goes
end-to-end if sip is on both call legs, if call goes throuch another 
protocol module, for example my mod_h323 (call scheme like 
user1-sip->fs-h323->someH323Ep) there is no way to negotiate 
codec because i have no SDP from called party for example and have no 
way to insert it on sip leg, or i don't see this way?

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From craig at overthewire.com.au  Wed Oct 28 17:40:05 2009
From: craig at overthewire.com.au (Craig Askings)
Date: Thu, 29 Oct 2009 10:40:05 +1000
Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.5pre3 is
	available for testing!
In-Reply-To: <87f2f3b90910281000r4c2021b8u7232320614737e30@mail.gmail.com>
References: <87f2f3b90910281000r4c2021b8u7232320614737e30@mail.gmail.com>
Message-ID: <8cc991dd0910281740w2c7a9d10q93dd5b5f0b4dbd6@mail.gmail.com>

Is the plan to run 1.0.5 as a stable branch with bug fix updates or will it
be the case of just follow the trunk like it is with any problems you
encounter in 1.0.4?

2009/10/29 Michael Collins <msc at freeswitch.org>

> Hello FreeSWITCHers,
>
> The latest FreeSWITCH version is now available for download on the files<http://files.freeswitch.org/freeswitch-1.0.5pre3.tar.gz>site. The announcement story is in the main
> FreeSWITCH <http://www.freeswitch.org/node/212> site. Please download and
> test, and then test some more. We need your feedback. The sooner we get your
> feedback, the more quickly we can roll the official 1.0.5.
>
> --
Craig Askings

Network Engineer | Over the Wire Pty Ltd
craig at overthewire.com.au | www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365
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From jason at jasonjgw.net  Wed Oct 28 18:26:23 2009
From: jason at jasonjgw.net (Jason White)
Date: Thu, 29 Oct 2009 12:26:23 +1100
Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.5pre3 is
 available for testing!
In-Reply-To: <8cc991dd0910281740w2c7a9d10q93dd5b5f0b4dbd6@mail.gmail.com>
References: <87f2f3b90910281000r4c2021b8u7232320614737e30@mail.gmail.com>
	<8cc991dd0910281740w2c7a9d10q93dd5b5f0b4dbd6@mail.gmail.com>
Message-ID: <20091029012622.GA3778@jdc.jasonjgw.net>

Craig Askings <craig at overthewire.com.au> wrote:
> Is the plan to run 1.0.5 as a stable branch with bug fix updates or will it
> be the case of just follow the trunk like it is with any problems you
> encounter in 1.0.4?

I can't speak for the developers, but I would expect the current policy to
continue: bug fixes are placed in trunk, and bug reports should be submitted
against whatever the current trunk revision is at the time of lodging the bug
reports.

The maintainers of FreeSWITCH are very effective, in general, at keeping SVN
trunk relatively stable.




From info at daccii.it  Wed Oct 28 14:05:17 2009
From: info at daccii.it (Albano Daniele Salvatore - Lavoro)
Date: Wed, 28 Oct 2009 22:05:17 +0100
Subject: [Freeswitch-users] Freeswitch seems to doesn't reknow dial tone
 after the first call using OpenZAP (analog spans)
Message-ID: <4AE8B20D.4040608@daccii.it>

Hi to all,

this is my first message here!

I'm trying to setup a freeswitch box using 1.0.4 version compiled from 
sources, ubuntu 8.04.03 lts with lastest updates and lastest zaptel 
modules/tools compiled from sources. As hardware i'm using an OpenVox 
A800P.

My problem is the following: if i do a call trought openzap it works, 
but when i hang down and try a second call it doesn't work!

Looking with ztmonitor and hearing recorded audio file it seems that 
freeswitch do the job on the first outgoing call but, on the second, it 
does nothing because it doesn't reknow dial tone (but it is there). If i 
call the freeswitch mananged number (i can answer or not, nothing 
change) i can do another call, but, in any case, the successive call 
doesn't start.

Looking with ztmonitor 1 -v i see that before the first call i get the 
special dial tone (a continue 425 frequency to state that the line is 
free to dial sended by the provider) but after the first call is done, 
before the second, i didn't see any special dial tone ... that appears 
for less than a sec when freeswitch starts to compose the number.

Here logs for first call
------
2009-10-28 21:38:39.616112 [INFO] ozmod_zt.c:636 Setting echo cancel to 
64 taps for 1:1
2009-10-28 21:38:39.616112 [DEBUG] ozmod_analog.c:450 Executing state 
handler on 1:1 for DIALING
2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:398 
(OpenZAP/1:1/xyz) Running State Change CS_INIT
2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:481 
(OpenZAP/1:1/xyz) State INIT
2009-10-28 21:38:39.616112 [DEBUG] mod_openzap.c:368 (OpenZAP/1:1/xyz) 
State Change CS_INIT -> CS_ROUTING
2009-10-28 21:38:39.616112 [DEBUG] switch_core_session.c:932 Send signal 
OpenZAP/1:1/xyz [BREAK]
2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:481 
(OpenZAP/1:1/xyz) State INIT going to sleep
2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:398 
(OpenZAP/1:1/xyz) Running State Change CS_ROUTING
2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:484 
(OpenZAP/1:1/xyz) State ROUTING
2009-10-28 21:38:39.616112 [DEBUG] mod_openzap.c:391 OpenZAP/1:1/xyz 
CHANNEL ROUTING
2009-10-28 21:38:39.616112 [DEBUG] switch_ivr_originate.c:63 
(OpenZAP/1:1/xyz) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2009-10-28 21:38:39.616112 [DEBUG] switch_core_session.c:932 Send signal 
OpenZAP/1:1/xyz [BREAK]
2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:484 
(OpenZAP/1:1/xyz) State ROUTING going to sleep
2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:398 
(OpenZAP/1:1/xyz) Running State Change CS_CONSUME_MEDIA
2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:503 
(OpenZAP/1:1/xyz) State CONSUME_MEDIA
2009-10-28 21:38:40.475897 [DEBUG] ozmod_analog.c:655 Detected tone DIAL 
on 1:1
2009-10-28 21:38:40.475897 [DEBUG] mod_openzap.c:1216 got FXO sig 1:1 
[TONE_DETECTED]
------

Here logs for second call
------
2009-10-28 21:39:07.105916 [INFO] ozmod_zt.c:636 Setting echo cancel to 
64 taps for 1:1
2009-10-28 21:39:07.105916 [DEBUG] ozmod_analog.c:450 Executing state 
handler on 1:1 for DIALING
2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:398 
(OpenZAP/1:1/xyz) Running State Change CS_INIT
2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:481 
(OpenZAP/1:1/xyz) State INIT
2009-10-28 21:39:07.118644 [DEBUG] mod_openzap.c:368 (OpenZAP/1:1/xyz) 
State Change CS_INIT -> CS_ROUTING
2009-10-28 21:39:07.118644 [DEBUG] switch_core_session.c:932 Send signal 
OpenZAP/1:1/xyz [BREAK]
2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:481 
(OpenZAP/1:1/xyz) State INIT going to sleep
2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:398 
(OpenZAP/1:1/xyz) Running State Change CS_ROUTING
2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:484 
(OpenZAP/1:1/xyz) State ROUTING
2009-10-28 21:39:07.118644 [DEBUG] mod_openzap.c:391 OpenZAP/1:1/xyz 
CHANNEL ROUTING
2009-10-28 21:39:07.118644 [DEBUG] switch_ivr_originate.c:63 
(OpenZAP/1:1/xyz) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2009-10-28 21:39:07.118644 [DEBUG] switch_core_session.c:932 Send signal 
OpenZAP/1:1/xyz [BREAK]
2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:484 
(OpenZAP/1:1/xyz) State ROUTING going to sleep
2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:398 
(OpenZAP/1:1/xyz) Running State Change CS_CONSUME_MEDIA
2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:503 
(OpenZAP/1:1/xyz) State CONSUME_MEDIA
------

Following my configs!

Really thanks for the help,


Best Regards,
Daniele

---
ZAPTEL.CONF
---
# Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#

# It must be in the module loading order


# Span 1: OPVXA1200/0 "OpenVox A1200P/A800P Board 1" (MASTER)
fxsks=1
fxsks=2
fxsks=3
fxsks=4
fxoks=5

# Global data

loadzone        = it
defaultzone     = it
---

ZT.CONF
---
[defaults]
codec_ms => 20
wink_ms => 150
flash_ms => 750
echo_cancel_level => 64
rxgain => 0.0
txgain => 0.0
---

OPENZAP.CONF
---
[span zt]
number => xyz
fxo-channel => 1
number => xyz
fxs-channel => 2
number => xyz
fxs-channel => 3
number => xyz
fxs-channel => 4

[span zt]
fxs-channel => 5
---

TONES.CONF
---
[it]
generate-dial => %(200, 200, 425);%(600, 1000, 425)
detect-dial => 425

generate-ring => %(1000, 4000, 425)
detect-ring => 425

generate-busy => %(500, 500, 425)
detect-busy => 425

generate-callwaiting-sas => %(400,100,425);%(250,100,425);%(150,0,425);
detect-callwaiting-sas => 425
---

OPENZAP.CONF.XML
---
<configuration name="openzap.conf" description="OpenZAP Configuration">
   <settings>
       <param name="debug" value="0"/>
       <param name="hold-music" value="$${moh_uri}"/>
   </settings>

   <analog_spans>

       <span id="1">
           <param name="tonegroup" value="it"/>
           <param name="digit-timeout" value="2000"/>
           <param name="max-digits" value="11"/>
           <param name="dialplan" value="XML"/>
           <param name="context" value="default"/>
           <param name="enable-callerid" value="true"/>
       </span>

       <span id="2">
           <param name="tonegroup" value="it"/>
           <param name="digit-timeout" value="2000"/>
           <param name="max-digits" value="11"/>
           <param name="dialplan" value="XML"/>
           <param name="context" value="default"/>
           <param name="enable-callerid" value="true"/>
       </span>

   </analog_spans>
</configuration>
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From davis.erwin at gmail.com  Wed Oct 28 19:00:14 2009
From: davis.erwin at gmail.com (Erwin Davis)
Date: Wed, 28 Oct 2009 19:00:14 -0700
Subject: [Freeswitch-users] error in loading spidermonkey
In-Reply-To: <8F404160-9658-4D67-A074-53022C2F5560@jerris.com>
References: <a38a91d50910281039pdf28a4fh536f44d238b9a954@mail.gmail.com>
	<8F404160-9658-4D67-A074-53022C2F5560@jerris.com>
Message-ID: <a38a91d50910281900s79dd1675se5f8bac4bc722ec6@mail.gmail.com>

fixed. Thanks,

On Wed, Oct 28, 2009 at 11:51 AM, Michael Jerris <mike at jerris.com> wrote:

> The libcurl is broken on your distro.  You can fix this by configuring
> with --without-libcurl which will use our working in tree copy instead
> of the broken one from your distro.
>
> Mike
>
> On Oct 28, 2009, at 1:39 PM, Erwin Davis wrote:
>
> > Hi, I got an error in loading mod_spidermonkey. my fs in ver 1.04
> > runs on fedora 8 VM. Any clue? thanks,
> >
> > [CRIT] switch_loadable_module.c:871 Error Loading module /usr/local/
> > freeswitch/mod/mod_spidermonkey.so
> > **/usr/lib/libnss3.so: undefined symbol: PR_UnloadLibrary**
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From egable+freeswitch at gmail.com  Wed Oct 28 19:21:05 2009
From: egable+freeswitch at gmail.com (Eliot Gable)
Date: Wed, 28 Oct 2009 22:21:05 -0400
Subject: [Freeswitch-users] SIP provider with extern rtp server
In-Reply-To: <8AC09649-2585-4BE7-A959-A7AC41650789@myrvold.org>
References: <A2D2E918-B494-4179-96BA-CA3A7717846C@myrvold.org>
	<bf6923ed0910270658t61bb6687ne0b7d3ec5090e2ff@mail.gmail.com>
	<CB8008BD-4066-4B5B-9043-214CFE938CD8@myrvold.org>
	<bf6923ed0910271034w6c7e0cc5t685a5377764755e6@mail.gmail.com>
	<FA2C693B-FCE1-4BDD-BCE5-302AC699CEDB@myrvold.org>
	<87f2f3b90910281112g6e72d22elcfd653991ecd50cc@mail.gmail.com>
	<8AC09649-2585-4BE7-A959-A7AC41650789@myrvold.org>
Message-ID: <bf6923ed0910281921uad8c525i850c08361b21538d@mail.gmail.com>

See that 200 OK that keeps coming in over and over and over and over
again? That's because they never received your ACK. If you can turn on
sofia loglevel to 9 and then watch where you send the ACK, you will
probably have your answer to why the other system did not receive it.
If you're still not sure what's going on, post another pastebin with
sofia loglevel set to 9.


On Wed, Oct 28, 2009 at 4:51 PM, Ivan C Myrvold <ivan at myrvold.org> wrote:
> Oh, what happened to it?
> Anyway, here is a new pb:
> http://pastebin.freeswitch.org/10867
> Ivan
> Den 28. okt. 2009 kl. 19:12 skrev Michael Collins:
>
>
> On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold <ivan at myrvold.org> wrote:
>>
>> Here is a debug log from a call from an internal phone out to an
>> external (my iPhone with nbr 91316356):
>> http://pastebin.freeswitch.org/108578
>>
>> Ivan
>>
> Uh... you wanna try that PB number again?
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero



From davis.erwin at gmail.com  Wed Oct 28 19:22:33 2009
From: davis.erwin at gmail.com (Erwin Davis)
Date: Wed, 28 Oct 2009 19:22:33 -0700
Subject: [Freeswitch-users] pass arguments into javascript
Message-ID: <a38a91d50910281922q59aad78s223851b06a6617ec@mail.gmail.com>

Hi, new to javascript. I tried to pass two arguments into javascript,

<action application="javascript" data="test.js $1 $2"/>


In test.js, I tried to use argv[1]  to retrieve $1 and argv[2] to retrieve
$2, however, the javascript test.js complained about argv[] as undefined
variables. How to retrieve the passing arguments in a javascript same as the
case above. Thanks,
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From egable+freeswitch at gmail.com  Wed Oct 28 20:23:09 2009
From: egable+freeswitch at gmail.com (Eliot Gable)
Date: Wed, 28 Oct 2009 23:23:09 -0400
Subject: [Freeswitch-users] pass arguments into javascript
In-Reply-To: <a38a91d50910281922q59aad78s223851b06a6617ec@mail.gmail.com>
References: <a38a91d50910281922q59aad78s223851b06a6617ec@mail.gmail.com>
Message-ID: <bf6923ed0910282023w5fbf8197uf1fd7fef7475f8ce@mail.gmail.com>

Should work fine. I use this:

var calling_num = argv[0];
var called_num = argv[1];

Are you sure you actually had valid data in $1 and $2? Try to call it
from the CLI:

jsrun test.js testvar1 testvar2



On Wed, Oct 28, 2009 at 10:22 PM, Erwin Davis <davis.erwin at gmail.com> wrote:
> Hi, new to javascript. I tried to pass two arguments into javascript,
>
> <action application="javascript" data="test.js $1 $2"/>
>
>
> In test.js, I tried to use argv[1]? to retrieve $1 and argv[2] to retrieve
> $2, however, the javascript test.js complained about argv[] as undefined
> variables. How to retrieve the passing arguments in a javascript same as the
> case above. Thanks,
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero



From davis.erwin at gmail.com  Thu Oct 29 07:48:32 2009
From: davis.erwin at gmail.com (Erwin Davis)
Date: Thu, 29 Oct 2009 10:48:32 -0400
Subject: [Freeswitch-users] pass arguments into javascript
In-Reply-To: <bf6923ed0910282023w5fbf8197uf1fd7fef7475f8ce@mail.gmail.com>
References: <a38a91d50910281922q59aad78s223851b06a6617ec@mail.gmail.com>
	<bf6923ed0910282023w5fbf8197uf1fd7fef7475f8ce@mail.gmail.com>
Message-ID: <a38a91d50910290748l2edb6df7xb1971a833bc7fabf@mail.gmail.com>

yes, I forgot the single quota in javascript statement. Now I saw the
aruments. Thanks,

On 10/28/09, Eliot Gable
<egable+freeswitch at gmail.com<egable%2Bfreeswitch at gmail.com>>
wrote:
>
> Should work fine. I use this:
>
> var calling_num = argv[0];
> var called_num = argv[1];
>
> Are you sure you actually had valid data in $1 and $2? Try to call it
> from the CLI:
>
> jsrun test.js testvar1 testvar2
>
>
>
>
> On Wed, Oct 28, 2009 at 10:22 PM, Erwin Davis <davis.erwin at gmail.com>
> wrote:
> > Hi, new to javascript. I tried to pass two arguments into javascript,
> >
> > <action application="javascript" data="test.js $1 $2"/>
> >
> >
> > In test.js, I tried to use argv[1]  to retrieve $1 and argv[2] to
> retrieve
> > $2, however, the javascript test.js complained about argv[] as undefined
> > variables. How to retrieve the passing arguments in a javascript same as
> the
> > case above. Thanks,
> >
>
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Eliot Gable
>
> "We do not inherit the Earth from our ancestors: we borrow it from our
> children." ~David Brower
>
> "I decided the words were too conservative for me. We're not borrowing
> from our children, we're stealing from them--and it's not even
> considered to be a crime." ~David Brower
>
> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
> live; not live to eat.) ~Marcus Tullius Cicero
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mi+thun at aldan.algebra.com  Thu Oct 29 07:56:33 2009
From: mi+thun at aldan.algebra.com (Mikhail T.)
Date: Thu, 29 Oct 2009 10:56:33 -0400
Subject: [Freeswitch-users] FreeSWITCH on FreeBSD
In-Reply-To: <4AE9BF36.3090201@gmail.com>
References: <4AE87EDB.90306@aldan.algebra.com> <4AE9BF36.3090201@gmail.com>
Message-ID: <4AE9AD21.9000300@aldan.algebra.com>

Richard E Neese ???????(??):
>>      * What hardware is known to work well on FreeBSD?
>>      * What else -- other than a computer with a phone card -- will I
>> need?
>>      * Do end-user phones have to be regular analog phones (all
>>        plugged-in to the computer), or IP-phones, such as made by Avaya
>>        (connecting via an Ethernet switch)?
>>      * How is the computer connecting to the world? Do I plug-in the
>>        incoming analog phone line(s) somewhere, or is it going to go
>>        through the Internet?
>>      * If the connection to the world is over the Internet, don't we
>> need
>>        an account with someone else, or is Verizon known to work with
>>        FreeSWITCH-based PBX-systems directly?
>>      * Should I not simply buy a used Avaya-system like this:
>>       
>> http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=110450339950&_trksid=p2759.l1259
>>
>>        What will I gain and what will I lose (other than personal
>>        satisfaction, of course) by going that route instead of building
>>        my own box?
Thank you very much, Richard, for you answering some of my questions. A
few remain, however:
> using freeswitch on bsd is free. nd all you kneed is phones like
> polycom soundpoint or aastra  or snome or cisco . Sip phones.
So, the phones connect -- via CAT6 cables -- to my network switch, and
talk to my FreeBSD server running FreeSWITCH. Ok. But how does the
server connect to the world? Somewhere between me talking into a Polycom
handset in New Jersey and my friend in hearing me in Ukraine, the audio
has to hit the telephone network... Where does this happen?

Do I plug-in a regular phone line into the same server? If so, what
hardware do I need inside it?
Or do I use a VOIP-provider -- connecting my server to them via the
Internet? And if so, which providers should I consider -- would Verizon
themselves qualify?

I ask, because the phone-service provided with Verizon's FIOS is already
a VOIP of some kind -- there is no copper line any more, it is all done
through optic cable... Verizon-provided device connects to that fiber on
one end, and gives you a phone-line on the other...

So it would be rather silly for my SIP-phones to convert the analog
speech into packets, then for the FreeSWITCH server to convert them back
into analog audio, and then for Verizon's equipment right in my house to
digitize back...

Thanks! Yours,

    -mi




From chris at cloudtel.com  Thu Oct 29 08:25:37 2009
From: chris at cloudtel.com (Chris Burns)
Date: Thu, 29 Oct 2009 11:25:37 -0400
Subject: [Freeswitch-users] Determining Frame Size
In-Reply-To: <4256bf830910280937n1f946518w423597fa6584c26f@mail.gmail.com>
References: <4256bf830910280937n1f946518w423597fa6584c26f@mail.gmail.com>
Message-ID: <200910291125.38079.chris@cloudtel.com>

Depends on your codec. Frame size = ptime. So it can vary from call to call, 
but most you will find it to be 20ms.

So if we are using speex-wb at 20ms, each frame is 0.02 seconds long. 50 
frames = 1 second of sound.

On October 28, 2009 12:37:56 pm Matthew Fong wrote:
> I noticed that the wait_for_silence's <silence_hits> <listen_hits> use
> number of frames as base values (rather than seconds or milliseconds). I'd
> like to wait for 5 seconds of consistent silence, but I'm wondering how I
> can determine the sample rate (and thus frames per second) for an existing
> call. Is there a variable I can read? Thanks.
>
> --matt





From anthony.minessale at gmail.com  Thu Oct 29 09:12:03 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 29 Oct 2009 11:12:03 -0500
Subject: [Freeswitch-users] Freeswitch seems to doesn't reknow dial tone
	after the first call using OpenZAP (analog spans)
In-Reply-To: <4AE8B20D.4040608@daccii.it>
References: <4AE8B20D.4040608@daccii.it>
Message-ID: <191c3a030910290912x9489784s4cc8e224363f3796@mail.gmail.com>

You may want to work with FreeSWITCH svn trunk for the time being.
There have been some changes and if you have a problem to fix it's easier to
debug with that version.


2009/10/28 Albano Daniele Salvatore - Lavoro <info at daccii.it>

> Hi to all,
>
> this is my first message here!
>
> I'm trying to setup a freeswitch box using 1.0.4 version compiled from
> sources, ubuntu 8.04.03 lts with lastest updates and lastest zaptel
> modules/tools compiled from sources. As hardware i'm using an OpenVox A800P.
>
> My problem is the following: if i do a call trought openzap it works, but
> when i hang down and try a second call it doesn't work!
>
> Looking with ztmonitor and hearing recorded audio file it seems that
> freeswitch do the job on the first outgoing call but, on the second, it does
> nothing because it doesn't reknow dial tone (but it is there). If i call the
> freeswitch mananged number (i can answer or not, nothing change) i can do
> another call, but, in any case, the successive call doesn't start.
>
> Looking with ztmonitor 1 -v i see that before the first call i get the
> special dial tone (a continue 425 frequency to state that the line is free
> to dial sended by the provider) but after the first call is done, before the
> second, i didn't see any special dial tone ... that appears for less than a
> sec when freeswitch starts to compose the number.
>
> Here logs for first call
> ------
> 2009-10-28 21:38:39.616112 [INFO] ozmod_zt.c:636 Setting echo cancel to 64
> taps for 1:1
> 2009-10-28 21:38:39.616112 [DEBUG] ozmod_analog.c:450 Executing state
> handler on 1:1 for DIALING
> 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:398
> (OpenZAP/1:1/xyz) Running State Change CS_INIT
> 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:481
> (OpenZAP/1:1/xyz) State INIT
> 2009-10-28 21:38:39.616112 [DEBUG] mod_openzap.c:368 (OpenZAP/1:1/xyz)
> State Change CS_INIT -> CS_ROUTING
> 2009-10-28 21:38:39.616112 [DEBUG] switch_core_session.c:932 Send signal
> OpenZAP/1:1/xyz [BREAK]
> 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:481
> (OpenZAP/1:1/xyz) State INIT going to sleep
> 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:398
> (OpenZAP/1:1/xyz) Running State Change CS_ROUTING
> 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:484
> (OpenZAP/1:1/xyz) State ROUTING
> 2009-10-28 21:38:39.616112 [DEBUG] mod_openzap.c:391 OpenZAP/1:1/xyz
> CHANNEL ROUTING
> 2009-10-28 21:38:39.616112 [DEBUG] switch_ivr_originate.c:63
> (OpenZAP/1:1/xyz) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2009-10-28 21:38:39.616112 [DEBUG] switch_core_session.c:932 Send signal
> OpenZAP/1:1/xyz [BREAK]
> 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:484
> (OpenZAP/1:1/xyz) State ROUTING going to sleep
> 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:398
> (OpenZAP/1:1/xyz) Running State Change CS_CONSUME_MEDIA
> 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:503
> (OpenZAP/1:1/xyz) State CONSUME_MEDIA
> 2009-10-28 21:38:40.475897 [DEBUG] ozmod_analog.c:655 Detected tone DIAL on
> 1:1
> 2009-10-28 21:38:40.475897 [DEBUG] mod_openzap.c:1216 got FXO sig 1:1
> [TONE_DETECTED]
> ------
>
> Here logs for second call
> ------
> 2009-10-28 21:39:07.105916 [INFO] ozmod_zt.c:636 Setting echo cancel to 64
> taps for 1:1
> 2009-10-28 21:39:07.105916 [DEBUG] ozmod_analog.c:450 Executing state
> handler on 1:1 for DIALING
> 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:398
> (OpenZAP/1:1/xyz) Running State Change CS_INIT
> 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:481
> (OpenZAP/1:1/xyz) State INIT
> 2009-10-28 21:39:07.118644 [DEBUG] mod_openzap.c:368 (OpenZAP/1:1/xyz)
> State Change CS_INIT -> CS_ROUTING
> 2009-10-28 21:39:07.118644 [DEBUG] switch_core_session.c:932 Send signal
> OpenZAP/1:1/xyz [BREAK]
> 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:481
> (OpenZAP/1:1/xyz) State INIT going to sleep
> 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:398
> (OpenZAP/1:1/xyz) Running State Change CS_ROUTING
> 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:484
> (OpenZAP/1:1/xyz) State ROUTING
> 2009-10-28 21:39:07.118644 [DEBUG] mod_openzap.c:391 OpenZAP/1:1/xyz
> CHANNEL ROUTING
> 2009-10-28 21:39:07.118644 [DEBUG] switch_ivr_originate.c:63
> (OpenZAP/1:1/xyz) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2009-10-28 21:39:07.118644 [DEBUG] switch_core_session.c:932 Send signal
> OpenZAP/1:1/xyz [BREAK]
> 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:484
> (OpenZAP/1:1/xyz) State ROUTING going to sleep
> 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:398
> (OpenZAP/1:1/xyz) Running State Change CS_CONSUME_MEDIA
> 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:503
> (OpenZAP/1:1/xyz) State CONSUME_MEDIA
> ------
>
> Following my configs!
>
> Really thanks for the help,
>
>
> Best Regards,
> Daniele
>
> ---
> ZAPTEL.CONF
> ---
> # Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
> # Zaptel Configuration File
> #
> # This file is parsed by the Zaptel Configurator, ztcfg
> #
>
> # It must be in the module loading order
>
>
> # Span 1: OPVXA1200/0 "OpenVox A1200P/A800P Board 1" (MASTER)
> fxsks=1
> fxsks=2
> fxsks=3
> fxsks=4
> fxoks=5
>
> # Global data
>
> loadzone        = it
> defaultzone     = it
> ---
>
> ZT.CONF
> ---
> [defaults]
> codec_ms => 20
> wink_ms => 150
> flash_ms => 750
> echo_cancel_level => 64
> rxgain => 0.0
> txgain => 0.0
> ---
>
> OPENZAP.CONF
> ---
> [span zt]
> number => xyz
> fxo-channel => 1
> number => xyz
> fxs-channel => 2
> number => xyz
> fxs-channel => 3
> number => xyz
> fxs-channel => 4
>
> [span zt]
> fxs-channel => 5
> ---
>
> TONES.CONF
> ---
> [it]
> generate-dial => %(200, 200, 425);%(600, 1000, 425)
> detect-dial => 425
>
> generate-ring => %(1000, 4000, 425)
> detect-ring => 425
>
> generate-busy => %(500, 500, 425)
> detect-busy => 425
>
> generate-callwaiting-sas => %(400,100,425);%(250,100,425);%(150,0,425);
> detect-callwaiting-sas => 425
> ---
>
> OPENZAP.CONF.XML
> ---
> <configuration name="openzap.conf" description="OpenZAP Configuration">
>  <settings>
>      <param name="debug" value="0"/>
>      <param name="hold-music" value="$${moh_uri}"/>
>  </settings>
>
>  <analog_spans>
>
>      <span id="1">
>          <param name="tonegroup" value="it"/>
>          <param name="digit-timeout" value="2000"/>
>          <param name="max-digits" value="11"/>
>          <param name="dialplan" value="XML"/>
>          <param name="context" value="default"/>
>          <param name="enable-callerid" value="true"/>
>      </span>
>
>      <span id="2">
>          <param name="tonegroup" value="it"/>
>          <param name="digit-timeout" value="2000"/>
>          <param name="max-digits" value="11"/>
>          <param name="dialplan" value="XML"/>
>          <param name="context" value="default"/>
>          <param name="enable-callerid" value="true"/>
>      </span>
>
>  </analog_spans>
> </configuration>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Thu Oct 29 10:14:05 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 29 Oct 2009 12:14:05 -0500
Subject: [Freeswitch-users] Employment Opportunities
Message-ID: <191c3a030910291014i741ec928p6ee001ac584c495d@mail.gmail.com>

We're looking for a few individuals to work with us on the CudaTel Product
powered by FreeSWITCH.

AJAX Developer:
Must have strong skill in HTTP/CGI/Javascript preferably some exp in jquery.
We would prefer if are not overly obsessed with any standards but respect
them enough to try to follow them in a well-balanced way.
Best if you live near Silicon Valley, Ann Arbor Michigan, or Milwaukee WI
but not mandatory.
Must be self-motivated. OCD a plus.  Willing to work late if necessary to
get things done right.
Working knowledge of Perl, DBI-X Catalyst and TT2 would make your life
easier.
Knowing a little about FS and Telephony would be nice.
Must be able to work diligently from a ssh terminal.

Back-end Developer:
Good knowledge of 2 or more Perl, DBI-X Catalyst and TT2.
Attention to detail, good debugging skills.
Best if you live near Silicon Valley, Ann Arbor Michigan, or Milwaukee WI
but not mandatory.
Must be self-motivated. OCD a plus.  Willing to work late if necessary to
get things done right.
Ability to work well with the guy in the other job description above.
Knowing a little about FS and Telephony would be nice.
Must be able to work diligently from a ssh terminal.


Both positions involve a hands-on "extreme programming" type environment
with shared screens and a permanent line of communication to your peers.
Send CR to jobs at freeswitch.org

-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Thu Oct 29 10:26:52 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 29 Oct 2009 10:26:52 -0700
Subject: [Freeswitch-users] Freeswitch seems to doesn't reknow dial tone
	after the first call using OpenZAP (analog spans)
In-Reply-To: <191c3a030910290912x9489784s4cc8e224363f3796@mail.gmail.com>
References: <4AE8B20D.4040608@daccii.it>
	<191c3a030910290912x9489784s4cc8e224363f3796@mail.gmail.com>
Message-ID: <87f2f3b90910291026u2c62ded6v5b9458a9c2e85877@mail.gmail.com>

On Thu, Oct 29, 2009 at 9:12 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> You may want to work with FreeSWITCH svn trunk for the time being.
> There have been some changes and if you have a problem to fix it's easier
> to debug with that version.
>
>
> 2009/10/28 Albano Daniele Salvatore - Lavoro <info at daccii.it>
>
>> Hi to all,
>>
>> this is my first message here!
>>
>> Welcome!

Like Tony said, the latest SVN trunk is probably what you want. We are
already doing pre-releases of 1.0.5. SVN trunk is extremely stable. Also,
you can get realtime help via IRC at irc.freenode.net, channels #freeswitch
and #openzap.

-MC
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From r.wilczynski at gmail.com  Thu Oct 29 02:46:06 2009
From: r.wilczynski at gmail.com (=?ISO-8859-2?Q?Robert_Wilczy=F1ski?=)
Date: Thu, 29 Oct 2009 10:46:06 +0100
Subject: [Freeswitch-users] Gateway registration fails: IP used for realm in
	place of configured realm value
Message-ID: <7e07d6d80910290246o2daea1e7o2b8be91b137ad2c8@mail.gmail.com>

Hi,

We have two FS instances production (prod) and gateway (gw)

when prod starts it tries to register a gateway:

<include>
??? <gateway name="fs_gateway">
??? ??? <param name="realm" value="24ccdomain"/>
??? ??? <param name="proxy" value="$${local_ip_v4}:15060" />
??? ??? <param name="username" value="24cc"/>
??? ??? <param name="password" value="24ccpass"/>
??? ??? <param name="expire-seconds" value="60"/>
??? ??? <param name="register" value="true"/>
??? </gateway>
</include>

and fails:

2009-10-29 10:19:45.449220 [NOTICE] sofia_glue.c:3534 Reload XML [Success]
2009-10-29 10:19:45.450221 [INFO] mod_enum.c:808 ENUM Reloaded
2009-10-29 10:19:45.450221 [NOTICE] sofia_reg.c:2117 Added gateway
'fs_gateway' to profile 'external'
2009-10-29 10:19:45.450221 [NOTICE] sofia.c:3093 Adding Alias
[outbound] for profile [external]
2009-10-29 10:19:45.450221 [NOTICE] sofia.c:3093 Adding Alias [nat]
for profile [external]
2009-10-29 10:19:45.451221 [NOTICE] sofia.c:3108 Started Profile
external [sofia_reg_external]
2009-10-29 10:19:45.452221 [INFO] switch_time.c:661 Timezone reloaded
530 definitions
2009-10-29 10:19:45.628743 [NOTICE] sofia_reg.c:333 Registering fs_gateway
2009-10-29 10:19:45.996790 [NOTICE] sofia.c:1202 Waiting for worker thread
2009-10-29 10:19:45.998790 [NOTICE] sofia.c:1729 Adding Alias
[192.168.1.20] for profile [internal]
2009-10-29 10:19:45.998790 [NOTICE] sofia.c:3093 Adding Alias
[default] for profile [internal]
2009-10-29 10:19:45.998790 [NOTICE] sofia.c:3108 Started Profile
internal [sofia_reg_internal]
2009-10-29 10:19:53.354724 [ERR] sofia_reg.c:1425 fs_gateway
Registration Failed with status Forbidden [403]. failure #1
2009-10-29 10:19:53.440735 [WARNING] sofia_reg.c:364 fs_gateway Failed
Registration, setting retry to 60 seconds.
2009-10-29 10:20:53.108812 [NOTICE] sofia_reg.c:333 Registering fs_gateway
2009-10-29 10:20:53.116813 [ERR] sofia_reg.c:1425 fs_gateway
Registration Failed with status Forbidden [403]. failure #2
2009-10-29 10:20:53.208825 [WARNING] sofia_reg.c:364 fs_gateway Failed
Registration, setting retry to 90 seconds.

gw is configured like that:

<include>
??? <domain name="24ccdomain">
??? ??? <params>
??? ??? ??? <param name="dial-string"
value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
??? ??? </params>

??? ??? <groups>
??? ??? ??? <group name="default">
??? ??? ??? ??? <users>
??? ??? ??? ??? ??? <user id="24cc">
??? ??? ??? ??? ??? ??? <params>
??? ??? ??? ??? ??? ??? ??? <param name="password" value="24ccpass"/>
??? ??? ??? ??? ??? ??? </params>
??? ??? ??? ??? ??? ??? <variables>
??? ??? ??? ??? ??? ??? ??? <variable name="user_context" value="default"/>
??? ??? ??? ??? ??? ??? ??? <variable name="effective_caller_id_name"
value="24 CallCenter"/>
??? ??? ??? ??? ??? ??? </variables>
??? ??? ??? ??? ??? </user>
??? ??? ??? ??? </users>
??? ??? ??? </group>
??? ??? </groups>
??? </domain>
</include>

and outputs:

2009-10-29 10:11:29.165700 [WARNING] sofia_reg.c:1737 Can't find user
[24cc at 192.168.1.20]
You must define a domain called '192.168.1.20' in your directory and
add a user with the id="24cc" attribute
and you must configure your device to use the proper domain in it's
authentication credentials.

on each registration attempt from prod.

This worked perfectly fine with one of previous revisions but started
failing on 15257. Using siptrace we were able to narrow down the issue
to prod sending an IP address as realm instead of gateway configured
realm (as it did in previous revision):

Authorization: Digest username="24cc", realm="192.168.1.20",
nonce="4d82d624-c884-6340-8f7f-a4575f2583bd",
cnonce="Es23bj8MEi29T7UQ80i7sw", algorithm=MD5,
uri="sip:192.168.1.20:15060",
response="afea161876aad137657bf1f923d18659", qop=auth, nc=00000004

Is there any configuration setting we are missing that would bring
back previous behavior?

--
Best regards,
Robert Wilczynski



From brian at freeswitch.org  Thu Oct 29 11:31:26 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 29 Oct 2009 13:31:26 -0500
Subject: [Freeswitch-users] Gateway registration fails: IP used for
	realm in place of configured realm value
In-Reply-To: <7e07d6d80910290246o2daea1e7o2b8be91b137ad2c8@mail.gmail.com>
References: <7e07d6d80910290246o2daea1e7o2b8be91b137ad2c8@mail.gmail.com>
Message-ID: <6E1B3448-2BD7-492C-B129-59B2D813B8EE@freeswitch.org>

It's because you need to set the from-user and from-domain because  
FreeSWITCH don't care about the realm because WE provide you the realm  
in our challenge... you only provide realm for broken providers such  
as BroadVoice.

remove realm from your setting.  Add from-user and from-domain

/b

On Oct 29, 2009, at 4:46 AM, Robert Wilczy?ski wrote:

> Hi,
>
> We have two FS instances production (prod) and gateway (gw)
>
> when prod starts it tries to register a gateway:
>
> <include>
>     <gateway name="fs_gateway">
>         <param name="realm" value="24ccdomain"/>
>         <param name="proxy" value="$${local_ip_v4}:15060" />
>         <param name="username" value="24cc"/>
>         <param name="password" value="24ccpass"/>
>         <param name="expire-seconds" value="60"/>
>         <param name="register" value="true"/>
>     </gateway>
> </include>
>
> and fails:
>
> 2009-10-29 10:19:45.449220 [NOTICE] sofia_glue.c:3534 Reload XML  
> [Success]
> 2009-10-29 10:19:45.450221 [INFO] mod_enum.c:808 ENUM Reloaded
> 2009-10-29 10:19:45.450221 [NOTICE] sofia_reg.c:2117 Added gateway
> 'fs_gateway' to profile 'external'
> 2009-10-29 10:19:45.450221 [NOTICE] sofia.c:3093 Adding Alias
> [outbound] for profile [external]
> 2009-10-29 10:19:45.450221 [NOTICE] sofia.c:3093 Adding Alias [nat]
> for profile [external]
> 2009-10-29 10:19:45.451221 [NOTICE] sofia.c:3108 Started Profile
> external [sofia_reg_external]
> 2009-10-29 10:19:45.452221 [INFO] switch_time.c:661 Timezone reloaded
> 530 definitions
> 2009-10-29 10:19:45.628743 [NOTICE] sofia_reg.c:333 Registering  
> fs_gateway
> 2009-10-29 10:19:45.996790 [NOTICE] sofia.c:1202 Waiting for worker  
> thread
> 2009-10-29 10:19:45.998790 [NOTICE] sofia.c:1729 Adding Alias
> [192.168.1.20] for profile [internal]
> 2009-10-29 10:19:45.998790 [NOTICE] sofia.c:3093 Adding Alias
> [default] for profile [internal]
> 2009-10-29 10:19:45.998790 [NOTICE] sofia.c:3108 Started Profile
> internal [sofia_reg_internal]
> 2009-10-29 10:19:53.354724 [ERR] sofia_reg.c:1425 fs_gateway
> Registration Failed with status Forbidden [403]. failure #1
> 2009-10-29 10:19:53.440735 [WARNING] sofia_reg.c:364 fs_gateway Failed
> Registration, setting retry to 60 seconds.
> 2009-10-29 10:20:53.108812 [NOTICE] sofia_reg.c:333 Registering  
> fs_gateway
> 2009-10-29 10:20:53.116813 [ERR] sofia_reg.c:1425 fs_gateway
> Registration Failed with status Forbidden [403]. failure #2
> 2009-10-29 10:20:53.208825 [WARNING] sofia_reg.c:364 fs_gateway Failed
> Registration, setting retry to 90 seconds.
>
> gw is configured like that:
>
> <include>
>     <domain name="24ccdomain">
>         <params>
>             <param name="dial-string"
> value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact($ 
> {dialed_user}@${dialed_domain})}"/>
>         </params>
>
>         <groups>
>             <group name="default">
>                 <users>
>                     <user id="24cc">
>                         <params>
>                             <param name="password" value="24ccpass"/>
>                         </params>
>                         <variables>
>                             <variable name="user_context"  
> value="default"/>
>                             <variable name="effective_caller_id_name"
> value="24 CallCenter"/>
>                         </variables>
>                     </user>
>                 </users>
>             </group>
>         </groups>
>     </domain>
> </include>
>
> and outputs:
>
> 2009-10-29 10:11:29.165700 [WARNING] sofia_reg.c:1737 Can't find user
> [24cc at 192.168.1.20]
> You must define a domain called '192.168.1.20' in your directory and
> add a user with the id="24cc" attribute
> and you must configure your device to use the proper domain in it's
> authentication credentials.
>
> on each registration attempt from prod.
>
> This worked perfectly fine with one of previous revisions but started
> failing on 15257. Using siptrace we were able to narrow down the issue
> to prod sending an IP address as realm instead of gateway configured
> realm (as it did in previous revision):
>
> Authorization: Digest username="24cc", realm="192.168.1.20",
> nonce="4d82d624-c884-6340-8f7f-a4575f2583bd",
> cnonce="Es23bj8MEi29T7UQ80i7sw", algorithm=MD5,
> uri="sip:192.168.1.20:15060",
> response="afea161876aad137657bf1f923d18659", qop=auth, nc=00000004
>
> Is there any configuration setting we are missing that would bring
> back previous behavior?
>
> --
> Best regards,
> Robert Wilczynski
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From larclap at yahoo.com  Thu Oct 29 16:20:19 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Thu, 29 Oct 2009 16:20:19 -0700
Subject: [Freeswitch-users] Error 488- Incompatible Destination
Message-ID: <010a01ca58ee$622ae700$2680b500$@com>

I must have been playing with the settings on a SNOM 320 because suddenly it
cannot connect to FreeSwitch. 

 

I have looked at similar posts and it seems like it must be a codec problem
on the SNOM, but I cannot figure it out. 

 

Can someone point me in the right direction?

http://pastebin.freeswitch.org/10887

 

Thanks, Lars

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From brian at freeswitch.org  Thu Oct 29 16:37:29 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 29 Oct 2009 18:37:29 -0500
Subject: [Freeswitch-users] Error 488- Incompatible Destination
In-Reply-To: <010a01ca58ee$622ae700$2680b500$@com>
References: <010a01ca58ee$622ae700$2680b500$@com>
Message-ID: <849DE6E6-1191-47C2-9151-7A97C8B35A57@freeswitch.org>

No its not ALL of the sudden... Please check the SRTP settings on the  
identity.  Thats why you're getting that... the phone defaults to  
sending crypto in AVP which is invalid... we reject it.

This has been this way for some time I suspect you have updated your  
snom recently or aren't on 7.1.35 or higher which has the options to  
fix this on the phone.  set it to SAVP

RTP/SAVP: to off will fix it. Unless you're doing SRTP then set it to  
mandatory.

/b

On Oct 29, 2009, at 6:20 PM, Lars Zeb wrote:

> I must have been playing with the settings on a SNOM 320 because  
> suddenly it cannot connect to FreeSwitch.
>
> I have looked at similar posts and it seems like it must be a codec  
> problem on the SNOM, but I cannot figure it out.
>
> Can someone point me in the right direction?
>
> http://pastebin.freeswitch.org/10887
>
> Thanks, Lars

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From anthony.minessale at gmail.com  Thu Oct 29 16:55:42 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 29 Oct 2009 18:55:42 -0500
Subject: [Freeswitch-users] Error 488- Incompatible Destination
In-Reply-To: <849DE6E6-1191-47C2-9151-7A97C8B35A57@freeswitch.org>
References: <010a01ca58ee$622ae700$2680b500$@com>
	<849DE6E6-1191-47C2-9151-7A97C8B35A57@freeswitch.org>
Message-ID: <191c3a030910291655i2ee1472r642352bdeea5dfbf@mail.gmail.com>

ironically, snom reported this as a bug to us when we used to tolerate it so
they made us fix it thus breaking it on their own phones.


On Thu, Oct 29, 2009 at 6:37 PM, Brian West <brian at freeswitch.org> wrote:

> No its not ALL of the sudden... Please check the SRTP settings on the
> identity.  Thats why you're getting that... the phone defaults to sending
> crypto in AVP which is invalid... we reject it.
>
> This has been this way for some time I suspect you have updated your snom
> recently or aren't on 7.1.35 or higher which has the options to fix this on
> the phone.  set it to SAVP
>
> RTP/SAVP: to off will fix it. Unless you're doing SRTP then set it to
> mandatory.
>
> /b
>
> On Oct 29, 2009, at 6:20 PM, Lars Zeb wrote:
>
> I must have been playing with the settings on a SNOM 320 because suddenly
> it cannot connect to FreeSwitch.
>
> I have looked at similar posts and it seems like it must be a codec problem
> on the SNOM, but I cannot figure it out.
>
> Can someone point me in the right direction?
>
> http://pastebin.freeswitch.org/10887
>
> Thanks, Lars
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From larclap at yahoo.com  Thu Oct 29 17:30:26 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Thu, 29 Oct 2009 17:30:26 -0700
Subject: [Freeswitch-users] Problem with inbound call answered but
	no	sound
In-Reply-To: <8D1A994D-2DCD-4016-9BEF-D6FBD3FBAA1E@freeswitch.org>
References: <016e01ca56ad$56362380$02a26a80$@com>	<168575F8-6507-4AD2-B448-2D6FDA0152B6@freeswitch.org>	<018801ca56b2$1e63e0f0$5b2ba2d0$@com>	<BACDC899-CE4F-42E1-B2A0-C0C0E31F8787@freeswitch.org>	<00d601ca571e$519f7300$f4de5900$@com>
	<8D1A994D-2DCD-4016-9BEF-D6FBD3FBAA1E@freeswitch.org>
Message-ID: <012101ca58f8$3d1b3f50$b751bdf0$@com>

Updated to v15279 and all is OK now.

 

Thanks, Brian

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Wednesday, October 28, 2009 7:13 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Problem with inbound call answered but no
sound

 

What kind of router are you behind?

 

/b

 

On Oct 27, 2009, at 10:58 AM, Lars Zeb wrote:





Thanks for the reply, Brian.

 

Did something in FS change between v15183 and v15225 to make this occur? I
ask because this same configuration worked OK in the earlier version.

 

Lars

 

 

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From larclap at yahoo.com  Thu Oct 29 17:40:57 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Thu, 29 Oct 2009 17:40:57 -0700
Subject: [Freeswitch-users] Error 488- Incompatible Destination
In-Reply-To: <849DE6E6-1191-47C2-9151-7A97C8B35A57@freeswitch.org>
References: <010a01ca58ee$622ae700$2680b500$@com>
	<849DE6E6-1191-47C2-9151-7A97C8B35A57@freeswitch.org>
Message-ID: <012601ca58f9$a5b1e2c0$f115a840$@com>

I meant "suddenly" after I had been messing with the SNOM's settings. The
firmware is 7.3.14. The SRTP stuff is:

 

Symmetrical RTP:       off 

RTP Encryption:        on 

Dynamic G.726 payload: on 

SRTP Auth-tag:         AES-32 

RTP/SAVP:             off

 

Lars

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, October 29, 2009 4:37 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Error 488- Incompatible Destination

 

No its not ALL of the sudden... Please check the SRTP settings on the
identity.  Thats why you're getting that... the phone defaults to sending
crypto in AVP which is invalid... we reject it.

 

This has been this way for some time I suspect you have updated your snom
recently or aren't on 7.1.35 or higher which has the options to fix this on
the phone.  set it to SAVP

 

RTP/SAVP: to off will fix it. Unless you're doing SRTP then set it to
mandatory.

 

/b

 

On Oct 29, 2009, at 6:20 PM, Lars Zeb wrote:





I must have been playing with the settings on a SNOM 320 because suddenly it
cannot connect to FreeSwitch.

 

I have looked at similar posts and it seems like it must be a codec problem
on the SNOM, but I cannot figure it out.

 

Can someone point me in the right direction?

http://pastebin.freeswitch.org/10887

 

Thanks, Lars

 

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From brian at freeswitch.org  Thu Oct 29 18:06:07 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 29 Oct 2009 20:06:07 -0500
Subject: [Freeswitch-users] Error 488- Incompatible Destination
In-Reply-To: <012601ca58f9$a5b1e2c0$f115a840$@com>
References: <010a01ca58ee$622ae700$2680b500$@com>
	<849DE6E6-1191-47C2-9151-7A97C8B35A57@freeswitch.org>
	<012601ca58f9$a5b1e2c0$f115a840$@com>
Message-ID: <CB02F8C3-FC81-41C5-9BD7-DEE5C126F1F0@freeswitch.org>

SEE THIS TAG?  If you have SRTP on you MUST put this to Optional if  
you care for it to work... read the debug a little better it'll tell  
you to reference an RFC on this also.

/b

On Oct 29, 2009, at 7:40 PM, Lars Zeb wrote:

> RTP/SAVP:             off

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From JCasale at activenetwerx.com  Thu Oct 29 19:22:48 2009
From: JCasale at activenetwerx.com (Joseph L. Casale)
Date: Fri, 30 Oct 2009 02:22:48 +0000
Subject: [Freeswitch-users] Error building rpm's from trunk
Message-ID: <ABF9510930E1374BA4B4C61A01104FBDC89637@Monterossa.activenetwerx.local>

I fetched a copy of trunk, tar and gzipped it up and placed it in my
SOURCES dir, I edited the bzip extension in the spec file and tried to
build rpms but got this error:


Checking for unpackaged file(s): /usr/lib/rpm/check-files /var/tmp/freeswitch-1.0.4-1-root-rpm-builder
error: Installed (but unpackaged) file(s) found:
   /opt/freeswitch/mod/ozmod_sangoma_boost.so

RPM build errors:
    Installed (but unpackaged) file(s) found:
   /opt/freeswitch/mod/ozmod_sangoma_boost.so


Anyone know if trunk successfully builds into rpms on CentOSx86 or is
this an issue on my end?

Thanks!
jlc



From brian at freeswitch.org  Thu Oct 29 19:39:57 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 29 Oct 2009 21:39:57 -0500
Subject: [Freeswitch-users] Error building rpm's from trunk
In-Reply-To: <ABF9510930E1374BA4B4C61A01104FBDC89637@Monterossa.activenetwerx.local>
References: <ABF9510930E1374BA4B4C61A01104FBDC89637@Monterossa.activenetwerx.local>
Message-ID: <67478BE8-E29B-4A35-B0E4-2746DF09BB29@freeswitch.org>

perl -i -pe 's/ss7_boost/sangoma_boost/g' freeswitch.spec

or svn up.

/b

On Oct 29, 2009, at 9:22 PM, Joseph L. Casale wrote:

> I fetched a copy of trunk, tar and gzipped it up and placed it in my
> SOURCES dir, I edited the bzip extension in the spec file and tried to
> build rpms but got this error:
>
>
> Checking for unpackaged file(s): /usr/lib/rpm/check-files /var/tmp/ 
> freeswitch-1.0.4-1-root-rpm-builder
> error: Installed (but unpackaged) file(s) found:
>   /opt/freeswitch/mod/ozmod_sangoma_boost.so
>
> RPM build errors:
>    Installed (but unpackaged) file(s) found:
>   /opt/freeswitch/mod/ozmod_sangoma_boost.so
>
>
> Anyone know if trunk successfully builds into rpms on CentOSx86 or is
> this an issue on my end?
>
> Thanks!
> jlc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From brian at freeswitch.org  Thu Oct 29 19:41:24 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 29 Oct 2009 21:41:24 -0500
Subject: [Freeswitch-users] CELT Codec on Friday's meeting
Message-ID: <26F3FA5E-ED72-4EAD-ACA4-C3D037F1D655@freeswitch.org>

Please update to 1.0.5pre4 if you plan on using the CELT codec on the  
conference tomorrow.  I'm rolling the pre4 tarball right now.

Thanks,
Brian




From ujjval at simplesignal.com  Thu Oct 29 21:34:30 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Thu, 29 Oct 2009 21:34:30 -0700
Subject: [Freeswitch-users] Setting up Conference with Moderator
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local>
	<3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local>
	<E97CD3D0-52FF-44D1-BB89-23816045CE21@gmail.com>
	<71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>,
	<114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local>
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local>

Rob:

 For some reason, I don't see the DTMF appear on the fs_CLI when using the below configuration....so it basically timesout.

UK



-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ujjval Karihaloo
Sent: Monday, October 26, 2009 9:21 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator

Thx a lot Rob, reading the wiki your way or using IVR seems correct..
===============
The wiki also says that the wait-mod might be  "used in conjunction 
with an IVR where the moderators are authenticated with an extra pass-
code", which is what I did.  I guess that's why I didn't understand 
the point of the +pin.
======================

I will try it out.

Again thx a lot for your help. Will keep everyone posted.

________________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Forman [rob4manhere at gmail.com]
Sent: Friday, October 23, 2009 12:22 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator

I just re-tested with the pin in my dial plan:

   <action application="conference" data="conference 123456 at default
+flags{}+1234" />

And it doesn't challenge me for the pin.  I just drop right in.  I
figured this is how it was intended, since the wiki says the pin is
set initially and only challenged in later attempts [by future callers]:

"The first time a conference name (confname) is used, it will be
created on demand, and the pin will be set to what ever is specified
at that time: the pin in the data string if specified, or if not, the
"pin" setting in the conference profile, and if that is also
unspecified, then there is no pin protection. Any later attempt to
join the conference must specify the same pin number, if one existed
when it was created. "


The wiki also says that the wait-mod might be  "used in conjunction
with an IVR where the moderators are authenticated with an extra pass-
code", which is what I did.  I guess that's why I didn't understand
the point of the +pin.

I'm sure there's a scenario where its used and useful, the wiki just
doesn't explain it.

Rob

On Oct 23, 2009, at 12:43 PM, Brian West wrote:

> Well first off you're not defining a pine here...
>
> confname at profilename+flags{mute|deaf|waste|moderator}+[conference pin
> number]
>
> That might be why its not asking for a pin.
>
> /b
>
> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:
>
>>      <entry action="menu-exec-app" digits="1" param="conference
>> 123456 at default+flags{}" />
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From velu.technical at gmail.com  Thu Oct 29 23:57:41 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Fri, 30 Oct 2009 12:27:41 +0530
Subject: [Freeswitch-users] Flushing the Event buffer in Perl Event Socket
Message-ID: <1452e2980910292357i38379319ib4283f7189d05abe@mail.gmail.com>

Dear All,
      I receiving the events in while loop by using recvEventTimed method in
ESL.pm. I have to flush that Event buffer after some particular time. How
can I do it?

Thanks,
Velusamy
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From codecomplete at free.fr  Fri Oct 30 05:06:17 2009
From: codecomplete at free.fr (Fred-145)
Date: Fri, 30 Oct 2009 05:06:17 -0700 (PDT)
Subject: [Freeswitch-users] PBX vs. SIP proxy?
In-Reply-To: <4AD914FE.4020709@gmail.com>
References: <25920788.post@talk.nabble.com> <4AD914FE.4020709@gmail.com>
Message-ID: <26128938.post@talk.nabble.com>


Thanks Meftah for the help. So it looks like an SIP proxy is just to handle
the registration process, while a PBX offers services like ringing phones,
setting up calls, handling voice mail, etc.
-- 
View this message in context: http://old.nabble.com/PBX-vs.-SIP-proxy--tp25920788p26128938.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From rob4manhere at gmail.com  Fri Oct 30 06:22:55 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Fri, 30 Oct 2009 08:22:55 -0500
Subject: [Freeswitch-users] Setting up Conference with Moderator
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local>
	<3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local>
	<E97CD3D0-52FF-44D1-BB89-23816045CE21@gmail.com>
	<71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>,
	<114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local>
	<3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local>
Message-ID: <F88E7B78-4CB8-45BD-98CE-81195862A891@gmail.com>

I've never had any problem with that.  Is your logging at debug level  
so you can see the RECV DTFM in the log/fs_cli?  Are you calling from  
a SIP phone on the pbx, or via a PSTN provider?  Maybe your provider  
isn't passing them through.

Make sure your logging is turned up then try something simpler, like  
calling the echo application, and see if DTFM comes through.

Rob

On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote:

> Rob:
>
> For some reason, I don't see the DTMF appear on the fs_CLI when  
> using the below configuration....so it basically timesout.
>
> UK
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Ujjval Karihaloo
> Sent: Monday, October 26, 2009 9:21 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
>
> Thx a lot Rob, reading the wiki your way or using IVR seems correct..
> ===============
> The wiki also says that the wait-mod might be  "used in conjunction
> with an IVR where the moderators are authenticated with an extra pass-
> code", which is what I did.  I guess that's why I didn't understand
> the point of the +pin.
> ======================
>
> I will try it out.
>
> Again thx a lot for your help. Will keep everyone posted.
>
> ________________________________________
> From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Rob Forman [rob4manhere at gmail.com]
> Sent: Friday, October 23, 2009 12:22 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
>
> I just re-tested with the pin in my dial plan:
>
>   <action application="conference" data="conference 123456 at default
> +flags{}+1234" />
>
> And it doesn't challenge me for the pin.  I just drop right in.  I
> figured this is how it was intended, since the wiki says the pin is
> set initially and only challenged in later attempts [by future  
> callers]:
>
> "The first time a conference name (confname) is used, it will be
> created on demand, and the pin will be set to what ever is specified
> at that time: the pin in the data string if specified, or if not, the
> "pin" setting in the conference profile, and if that is also
> unspecified, then there is no pin protection. Any later attempt to
> join the conference must specify the same pin number, if one existed
> when it was created. "
>
>
> The wiki also says that the wait-mod might be  "used in conjunction
> with an IVR where the moderators are authenticated with an extra pass-
> code", which is what I did.  I guess that's why I didn't understand
> the point of the +pin.
>
> I'm sure there's a scenario where its used and useful, the wiki just
> doesn't explain it.
>
> Rob
>
> On Oct 23, 2009, at 12:43 PM, Brian West wrote:
>
>> Well first off you're not defining a pine here...
>>
>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference pin
>> number]
>>
>> That might be why its not asking for a pin.
>>
>> /b
>>
>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:
>>
>>>     <entry action="menu-exec-app" digits="1" param="conference
>>> 123456 at default+flags{}" />
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From anthony.minessale at gmail.com  Fri Oct 30 06:50:02 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 30 Oct 2009 08:50:02 -0500
Subject: [Freeswitch-users] Flushing the Event buffer in Perl Event
	Socket
In-Reply-To: <1452e2980910292357i38379319ib4283f7189d05abe@mail.gmail.com>
References: <1452e2980910292357i38379319ib4283f7189d05abe@mail.gmail.com>
Message-ID: <191c3a030910300650w7b80568eu4c41c805b9372acc@mail.gmail.com>

read them in a timed loop of some small number of MS until you get a timeout
meaning you have flushed them all.


On Fri, Oct 30, 2009 at 1:57 AM, velusamy velu <velu.technical at gmail.com>wrote:

> Dear All,
>       I receiving the events in while loop by using recvEventTimed method
> in ESL.pm. I have to flush that Event buffer after some particular time. How
> can I do it?
>
> Thanks,
> Velusamy
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From ujjval at simplesignal.com  Fri Oct 30 07:43:07 2009
From: ujjval at simplesignal.com (Ujjval Karihaloo)
Date: Fri, 30 Oct 2009 07:43:07 -0700
Subject: [Freeswitch-users] Setting up Conference with Moderator
In-Reply-To: <F88E7B78-4CB8-45BD-98CE-81195862A891@gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local>
	<3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local>
	<E97CD3D0-52FF-44D1-BB89-23816045CE21@gmail.com>
	<71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>,
	<114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local>
	<3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local>
	<F88E7B78-4CB8-45BD-98CE-81195862A891@gmail.com>
Message-ID: <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local>

It's strange... a tcpdump tells me that there is no DTMF from my provider when using IVR, but when I call into a TN that goes directly into the Conference App, I see DTMF from the provider.



-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Forman
Sent: Friday, October 30, 2009 7:23 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator

I've never had any problem with that.  Is your logging at debug level  
so you can see the RECV DTFM in the log/fs_cli?  Are you calling from  
a SIP phone on the pbx, or via a PSTN provider?  Maybe your provider  
isn't passing them through.

Make sure your logging is turned up then try something simpler, like  
calling the echo application, and see if DTFM comes through.

Rob

On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote:

> Rob:
>
> For some reason, I don't see the DTMF appear on the fs_CLI when  
> using the below configuration....so it basically timesout.
>
> UK
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Ujjval Karihaloo
> Sent: Monday, October 26, 2009 9:21 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
>
> Thx a lot Rob, reading the wiki your way or using IVR seems correct..
> ===============
> The wiki also says that the wait-mod might be  "used in conjunction
> with an IVR where the moderators are authenticated with an extra pass-
> code", which is what I did.  I guess that's why I didn't understand
> the point of the +pin.
> ======================
>
> I will try it out.
>
> Again thx a lot for your help. Will keep everyone posted.
>
> ________________________________________
> From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Rob Forman [rob4manhere at gmail.com]
> Sent: Friday, October 23, 2009 12:22 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
>
> I just re-tested with the pin in my dial plan:
>
>   <action application="conference" data="conference 123456 at default
> +flags{}+1234" />
>
> And it doesn't challenge me for the pin.  I just drop right in.  I
> figured this is how it was intended, since the wiki says the pin is
> set initially and only challenged in later attempts [by future  
> callers]:
>
> "The first time a conference name (confname) is used, it will be
> created on demand, and the pin will be set to what ever is specified
> at that time: the pin in the data string if specified, or if not, the
> "pin" setting in the conference profile, and if that is also
> unspecified, then there is no pin protection. Any later attempt to
> join the conference must specify the same pin number, if one existed
> when it was created. "
>
>
> The wiki also says that the wait-mod might be  "used in conjunction
> with an IVR where the moderators are authenticated with an extra pass-
> code", which is what I did.  I guess that's why I didn't understand
> the point of the +pin.
>
> I'm sure there's a scenario where its used and useful, the wiki just
> doesn't explain it.
>
> Rob
>
> On Oct 23, 2009, at 12:43 PM, Brian West wrote:
>
>> Well first off you're not defining a pine here...
>>
>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference pin
>> number]
>>
>> That might be why its not asking for a pin.
>>
>> /b
>>
>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:
>>
>>>     <entry action="menu-exec-app" digits="1" param="conference
>>> 123456 at default+flags{}" />
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From rob4manhere at gmail.com  Fri Oct 30 08:34:28 2009
From: rob4manhere at gmail.com (Rob Forman)
Date: Fri, 30 Oct 2009 10:34:28 -0500
Subject: [Freeswitch-users] Setting up Conference with Moderator
In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local>
	<3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local>
	<E97CD3D0-52FF-44D1-BB89-23816045CE21@gmail.com>
	<71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>,
	<114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local>
	<3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local>
	<F88E7B78-4CB8-45BD-98CE-81195862A891@gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local>
Message-ID: <D7D48F51-0917-4D9A-B8B4-DE8F552BB254@gmail.com>

Hm, strange.  I haven't seen that before.  Can you pastebin your logs  
at debug level?

On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote:

> It's strange... a tcpdump tells me that there is no DTMF from my  
> provider when using IVR, but when I call into a TN that goes  
> directly into the Conference App, I see DTMF from the provider.
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Rob Forman
> Sent: Friday, October 30, 2009 7:23 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
>
> I've never had any problem with that.  Is your logging at debug level
> so you can see the RECV DTFM in the log/fs_cli?  Are you calling from
> a SIP phone on the pbx, or via a PSTN provider?  Maybe your provider
> isn't passing them through.
>
> Make sure your logging is turned up then try something simpler, like
> calling the echo application, and see if DTFM comes through.
>
> Rob
>
> On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote:
>
>> Rob:
>>
>> For some reason, I don't see the DTMF appear on the fs_CLI when
>> using the below configuration....so it basically timesout.
>>
>> UK
>>
>>
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org
>> ] On Behalf Of Ujjval Karihaloo
>> Sent: Monday, October 26, 2009 9:21 AM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
>>
>> Thx a lot Rob, reading the wiki your way or using IVR seems correct..
>> ===============
>> The wiki also says that the wait-mod might be  "used in conjunction
>> with an IVR where the moderators are authenticated with an extra  
>> pass-
>> code", which is what I did.  I guess that's why I didn't understand
>> the point of the +pin.
>> ======================
>>
>> I will try it out.
>>
>> Again thx a lot for your help. Will keep everyone posted.
>>
>> ________________________________________
>> From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org
>> ] On Behalf Of Rob Forman [rob4manhere at gmail.com]
>> Sent: Friday, October 23, 2009 12:22 PM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
>>
>> I just re-tested with the pin in my dial plan:
>>
>>  <action application="conference" data="conference 123456 at default
>> +flags{}+1234" />
>>
>> And it doesn't challenge me for the pin.  I just drop right in.  I
>> figured this is how it was intended, since the wiki says the pin is
>> set initially and only challenged in later attempts [by future
>> callers]:
>>
>> "The first time a conference name (confname) is used, it will be
>> created on demand, and the pin will be set to what ever is specified
>> at that time: the pin in the data string if specified, or if not, the
>> "pin" setting in the conference profile, and if that is also
>> unspecified, then there is no pin protection. Any later attempt to
>> join the conference must specify the same pin number, if one existed
>> when it was created. "
>>
>>
>> The wiki also says that the wait-mod might be  "used in conjunction
>> with an IVR where the moderators are authenticated with an extra  
>> pass-
>> code", which is what I did.  I guess that's why I didn't understand
>> the point of the +pin.
>>
>> I'm sure there's a scenario where its used and useful, the wiki just
>> doesn't explain it.
>>
>> Rob
>>
>> On Oct 23, 2009, at 12:43 PM, Brian West wrote:
>>
>>> Well first off you're not defining a pine here...
>>>
>>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference  
>>> pin
>>> number]
>>>
>>> That might be why its not asking for a pin.
>>>
>>> /b
>>>
>>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:
>>>
>>>>    <entry action="menu-exec-app" digits="1" param="conference
>>>> 123456 at default+flags{}" />
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian.stafford at lattice-voice.com  Fri Oct 30 08:47:40 2009
From: brian.stafford at lattice-voice.com (Brian Stafford)
Date: Fri, 30 Oct 2009 15:47:40 +0000
Subject: [Freeswitch-users] mod_valet_parking: auto reports on wrong leg of
	call
Message-ID: <4AEB0A9C.7010907@lattice-voice.com>

Hi all

I did a 'make update' to 15289 and I found the auto feature added to 
mod_valet_parking so I figured I'd try it out.  Everything works fine 
except that the extension number for the parked call is reported on the 
wrong leg of the call.  It'd be good to sort this out as this is a 
useful feature for us.

Regards
Brian Stafford



From msc at freeswitch.org  Fri Oct 30 08:53:44 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 30 Oct 2009 08:53:44 -0700
Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Starting
	Shortly, Please Start 	Calling In
Message-ID: <87f2f3b90910300853s7ff6c38bt67f660e9d8581bdf@mail.gmail.com>

Hey everyone! The weekly meeting will start in a few minutes. Here's the
agenda:

http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_30

Please feel free to add other subjects. If you have something to share with
the community that would be great. Also, if you have questions for the dev
team by all means bring them too.

-Michael
-------------- next part --------------
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From brian at freeswitch.org  Fri Oct 30 09:00:29 2009
From: brian at freeswitch.org (Brian West)
Date: Fri, 30 Oct 2009 11:00:29 -0500
Subject: [Freeswitch-users] mod_valet_parking: auto reports on wrong leg
	of call
In-Reply-To: <4AEB0A9C.7010907@lattice-voice.com>
References: <4AEB0A9C.7010907@lattice-voice.com>
Message-ID: <BC287CE9-67CA-4E6A-A6D1-38ABBC2C51A7@freeswitch.org>

You have to be doing it wrong then.

Can you show us your dialplan you should have two extensions one for  
the lot range and one to attended transfer someone into the lot.

/b




On Oct 30, 2009, at 10:47 AM, Brian Stafford wrote:

> Hi all
>
> I did a 'make update' to 15289 and I found the auto feature added to
> mod_valet_parking so I figured I'd try it out.  Everything works fine
> except that the extension number for the parked call is reported on  
> the
> wrong leg of the call.  It'd be good to sort this out as this is a
> useful feature for us.
>
> Regards
> Brian Stafford




From djbinter at yahoo.com  Fri Oct 30 09:03:29 2009
From: djbinter at yahoo.com (DJB)
Date: Fri, 30 Oct 2009 09:03:29 -0700 (PDT)
Subject: [Freeswitch-users] Freeswitch in signaling path only
Message-ID: <429925.60305.qm@web37508.mail.mud.yahoo.com>

I am wondering why I cannot do as condition#2.

For Lua in dialplan, when I have the followings:


--WORKING--
(Condition#1)
.
.
session:execute("set","bypass_media=true")
session:execute("set","hangup_after_bridge=true")
session:execute("set","continue_on_fail=true")
.
.
 session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.1|sofia/external/" .. called_num .. "@1.1.1.2")
.
.

--NOT WORKING--
(Condition#2)
Note:  FS tries to be in media path and send re-invite.
.
.
session:execute("set","bypass_media=true")
session:execute("set","hangup_after_bridge=true")
session:execute("set","continue_on_fail=true")
.
.
 session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.1")
 session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.2")
.
.

Thank you,
Dorn B.


      



From djbinter at yahoo.com  Fri Oct 30 09:04:05 2009
From: djbinter at yahoo.com (DJB)
Date: Fri, 30 Oct 2009 09:04:05 -0700 (PDT)
Subject: [Freeswitch-users] CDR CSV variables
Message-ID: <609781.88322.qm@web37502.mail.mud.yahoo.com>

I wonder if I don't want to have b-leg in cdr csv, is there any variables that can give me the actual gateway ip address that is actually went out.

For instance, if I have this in my dialplan:
<action application="bridge" data="sofia/default/$1 at 192.168.1.4|sofia/default/$1 at 192.168.1.5"/>
the only value that I can think of from cdr csv is to get remote_ip_last_arg, but it would contains the whole line of both ip addresses.

Thank you,
Dorn B.



      



From brian.stafford at lattice-voice.com  Fri Oct 30 09:16:38 2009
From: brian.stafford at lattice-voice.com (Brian Stafford)
Date: Fri, 30 Oct 2009 16:16:38 +0000
Subject: [Freeswitch-users] mod_valet_parking: auto reports on wrong leg
 of call
In-Reply-To: <BC287CE9-67CA-4E6A-A6D1-38ABBC2C51A7@freeswitch.org>
References: <4AEB0A9C.7010907@lattice-voice.com>
	<BC287CE9-67CA-4E6A-A6D1-38ABBC2C51A7@freeswitch.org>
Message-ID: <4AEB1166.80002@lattice-voice.com>

Brian West wrote:
> You have to be doing it wrong then.
>
> Can you show us your dialplan you should have two extensions one for  
> the lot range and one to attended transfer someone into the lot.
>
> /b
>   
The relevant excerpt from the dialplan is

<extension name="valet_unpark">
<condition field="destination_number" expression="^(41[0-9])$">
<action application="answer"/>
<action application="valet_park" data="valet_lot $1"/>
</condition>
</extension>

<extension name="valet_park">
<condition field="destination_number" expression="^(420)$">
<action application="answer"/>
<action application="valet_park" data="valet_lot auto in 410 419"/>
</condition>
</extension>

x410-419 are the slots and 420 parks a call. Parking by picking one of 
410-419 works fine and subsequently dialling them from another works 
fine, I added x420 for the auto feature.

Regards
Brian



From ivan at myrvold.org  Fri Oct 30 09:16:37 2009
From: ivan at myrvold.org (Ivan C Myrvold)
Date: Fri, 30 Oct 2009 17:16:37 +0100
Subject: [Freeswitch-users] SIP provider with extern rtp server
In-Reply-To: <bf6923ed0910281921uad8c525i850c08361b21538d@mail.gmail.com>
References: <A2D2E918-B494-4179-96BA-CA3A7717846C@myrvold.org>
	<bf6923ed0910270658t61bb6687ne0b7d3ec5090e2ff@mail.gmail.com>
	<CB8008BD-4066-4B5B-9043-214CFE938CD8@myrvold.org>
	<bf6923ed0910271034w6c7e0cc5t685a5377764755e6@mail.gmail.com>
	<FA2C693B-FCE1-4BDD-BCE5-302AC699CEDB@myrvold.org>
	<87f2f3b90910281112g6e72d22elcfd653991ecd50cc@mail.gmail.com>
	<8AC09649-2585-4BE7-A959-A7AC41650789@myrvold.org>
	<bf6923ed0910281921uad8c525i850c08361b21538d@mail.gmail.com>
Message-ID: <544D39F2-40AB-41B4-BF18-89D7492B17EE@myrvold.org>

I have already set debug to 9, on both profiles.

Ivan


Den 29. okt. 2009 kl. 03:21 skrev Eliot Gable:

> See that 200 OK that keeps coming in over and over and over and over
> again? That's because they never received your ACK. If you can turn on
> sofia loglevel to 9 and then watch where you send the ACK, you will
> probably have your answer to why the other system did not receive it.
> If you're still not sure what's going on, post another pastebin with
> sofia loglevel set to 9.
>
>
> On Wed, Oct 28, 2009 at 4:51 PM, Ivan C Myrvold <ivan at myrvold.org>  
> wrote:
>> Oh, what happened to it?
>> Anyway, here is a new pb:
>> http://pastebin.freeswitch.org/10867
>> Ivan
>> Den 28. okt. 2009 kl. 19:12 skrev Michael Collins:
>>
>>
>> On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold <ivan at myrvold.org>  
>> wrote:
>>>
>>> Here is a debug log from a call from an internal phone out to an
>>> external (my iPhone with nbr 91316356):
>>> http://pastebin.freeswitch.org/108578
>>>
>>> Ivan
>>>
>> Uh... you wanna try that PB number again?
>> -MC
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>>
>
>
>
> -- 
> Eliot Gable
>
> "We do not inherit the Earth from our ancestors: we borrow it from our
> children." ~David Brower
>
> "I decided the words were too conservative for me. We're not borrowing
> from our children, we're stealing from them--and it's not even
> considered to be a crime." ~David Brower
>
> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
> live; not live to eat.) ~Marcus Tullius Cicero
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>




From chris at cloudtel.com  Fri Oct 30 10:00:01 2009
From: chris at cloudtel.com (Chris Burns)
Date: Fri, 30 Oct 2009 13:00:01 -0400
Subject: [Freeswitch-users] Freeswitch in signaling path only
In-Reply-To: <429925.60305.qm@web37508.mail.mud.yahoo.com>
References: <429925.60305.qm@web37508.mail.mud.yahoo.com>
Message-ID: <200910301300.01766.chris@cloudtel.com>

Do you have a debug level message from switch_ivr_originate.c in your log?
"Channel is already up, delaying proxy mode 'till both legs are answered."

Set bypass_media b4 each bridge. It is unsetting on you and setting 
bypass_media_after_bridge because you already answered the channel running 
the lua script.

On October 30, 2009 12:03:29 pm DJB wrote:
> I am wondering why I cannot do as condition#2.
>
> For Lua in dialplan, when I have the followings:
>
>
> --WORKING--
> (Condition#1)
> .
> .
> session:execute("set","bypass_media=true")
> session:execute("set","hangup_after_bridge=true")
> session:execute("set","continue_on_fail=true")
> .
> .
>  session:execute("bridge","sofia/external/" .. called_num ..
> "@1.1.1.1|sofia/external/" .. called_num .. "@1.1.1.2") .
> .
>
> --NOT WORKING--
> (Condition#2)
> Note:  FS tries to be in media path and send re-invite.
> .
> .
> session:execute("set","bypass_media=true")
> session:execute("set","hangup_after_bridge=true")
> session:execute("set","continue_on_fail=true")
> .
> .
>  session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.1")
>  session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.2")
> .
> .
>
> Thank you,
> Dorn B.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From egable+freeswitch at gmail.com  Fri Oct 30 10:30:54 2009
From: egable+freeswitch at gmail.com (Eliot Gable)
Date: Fri, 30 Oct 2009 13:30:54 -0400
Subject: [Freeswitch-users] SIP provider with extern rtp server
In-Reply-To: <544D39F2-40AB-41B4-BF18-89D7492B17EE@myrvold.org>
References: <A2D2E918-B494-4179-96BA-CA3A7717846C@myrvold.org>
	<bf6923ed0910270658t61bb6687ne0b7d3ec5090e2ff@mail.gmail.com>
	<CB8008BD-4066-4B5B-9043-214CFE938CD8@myrvold.org>
	<bf6923ed0910271034w6c7e0cc5t685a5377764755e6@mail.gmail.com>
	<FA2C693B-FCE1-4BDD-BCE5-302AC699CEDB@myrvold.org>
	<87f2f3b90910281112g6e72d22elcfd653991ecd50cc@mail.gmail.com>
	<8AC09649-2585-4BE7-A959-A7AC41650789@myrvold.org>
	<bf6923ed0910281921uad8c525i850c08361b21538d@mail.gmail.com>
	<544D39F2-40AB-41B4-BF18-89D7492B17EE@myrvold.org>
Message-ID: <bf6923ed0910301030x375fd6b9m75328c0e3a58b70f@mail.gmail.com>

fsctl loglevel debug
console loglevel debug
sofia profile internal siptrace on
sofia profile external siptrace on
sofia loglevel all 9
^^^^^^^^^^^^^^^^^^^^^

Then run your call, then do this:

sofia loglevel all 0
sofia profile external siptrace off
sofia profile internal siptrace off
fsctl loglevel warning
console loglevel warning

On Fri, Oct 30, 2009 at 12:16 PM, Ivan C Myrvold <ivan at myrvold.org> wrote:
> I have already set debug to 9, on both profiles.
>
> Ivan
>
>
> Den 29. okt. 2009 kl. 03:21 skrev Eliot Gable:
>
>> See that 200 OK that keeps coming in over and over and over and over
>> again? That's because they never received your ACK. If you can turn on
>> sofia loglevel to 9 and then watch where you send the ACK, you will
>> probably have your answer to why the other system did not receive it.
>> If you're still not sure what's going on, post another pastebin with
>> sofia loglevel set to 9.
>>
>>
>> On Wed, Oct 28, 2009 at 4:51 PM, Ivan C Myrvold <ivan at myrvold.org>
>> wrote:
>>> Oh, what happened to it?
>>> Anyway, here is a new pb:
>>> http://pastebin.freeswitch.org/10867
>>> Ivan
>>> Den 28. okt. 2009 kl. 19:12 skrev Michael Collins:
>>>
>>>
>>> On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold <ivan at myrvold.org>
>>> wrote:
>>>>
>>>> Here is a debug log from a call from an internal phone out to an
>>>> external (my iPhone with nbr 91316356):
>>>> http://pastebin.freeswitch.org/108578
>>>>
>>>> Ivan
>>>>
>>> Uh... you wanna try that PB number again?
>>> -MC
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>> --
>> Eliot Gable
>>
>> "We do not inherit the Earth from our ancestors: we borrow it from our
>> children." ~David Brower
>>
>> "I decided the words were too conservative for me. We're not borrowing
>> from our children, we're stealing from them--and it's not even
>> considered to be a crime." ~David Brower
>>
>> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
>> live; not live to eat.) ~Marcus Tullius Cicero
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero



From r.wilczynski at gmail.com  Fri Oct 30 11:28:06 2009
From: r.wilczynski at gmail.com (=?ISO-8859-2?Q?Robert_Wilczy=F1ski?=)
Date: Fri, 30 Oct 2009 19:28:06 +0100
Subject: [Freeswitch-users] Gateway registration fails: IP used for
	realm in place of configured realm value
Message-ID: <56bfa1be0910301128n6169ee34h540f959042b9ad26@mail.gmail.com>

Hi Brian,

Thanks for the explanation. This seems to have solved the issue.

Robert.

> ---------- Forwarded message ----------
> From:?Brian West <brian at freeswitch.org>
> To:?freeswitch-users at lists.freeswitch.org
> Date:?Thu, 29 Oct 2009 13:31:26 -0500
> Subject:?Re: [Freeswitch-users] Gateway registration fails: IP used for realm in place of configured realm value
> It's because you need to set the from-user and from-domain because FreeSWITCH don't care about the realm because WE provide you the realm in our challenge... you only provide realm for broken providers such as BroadVoice.
>
> remove realm from your setting. ?Add from-user and from-domain
>
> /b
>
> On Oct 29, 2009, at 4:46 AM, Robert Wilczy?ski wrote:
>
>> Hi,
>>
>> We have two FS instances production (prod) and gateway (gw)
>>
>> when prod starts it tries to register a gateway:
>>
>> <include>
>> ? ?<gateway name="fs_gateway">
>> ? ? ? ?<param name="realm" value="24ccdomain"/>
>> ? ? ? ?<param name="proxy" value="$${local_ip_v4}:15060" />
>> ? ? ? ?<param name="username" value="24cc"/>
>> ? ? ? ?<param name="password" value="24ccpass"/>
>> ? ? ? ?<param name="expire-seconds" value="60"/>
>> ? ? ? ?<param name="register" value="true"/>
>> ? ?</gateway>
>> </include>
>>
>> and fails:
>>
>> 2009-10-29 10:19:45.449220 [NOTICE] sofia_glue.c:3534 Reload XML [Success]
>> 2009-10-29 10:19:45.450221 [INFO] mod_enum.c:808 ENUM Reloaded
>> 2009-10-29 10:19:45.450221 [NOTICE] sofia_reg.c:2117 Added gateway
>> 'fs_gateway' to profile 'external'
>> 2009-10-29 10:19:45.450221 [NOTICE] sofia.c:3093 Adding Alias
>> [outbound] for profile [external]
>> 2009-10-29 10:19:45.450221 [NOTICE] sofia.c:3093 Adding Alias [nat]
>> for profile [external]
>> 2009-10-29 10:19:45.451221 [NOTICE] sofia.c:3108 Started Profile
>> external [sofia_reg_external]
>> 2009-10-29 10:19:45.452221 [INFO] switch_time.c:661 Timezone reloaded
>> 530 definitions
>> 2009-10-29 10:19:45.628743 [NOTICE] sofia_reg.c:333 Registering fs_gateway
>> 2009-10-29 10:19:45.996790 [NOTICE] sofia.c:1202 Waiting for worker thread
>> 2009-10-29 10:19:45.998790 [NOTICE] sofia.c:1729 Adding Alias
>> [192.168.1.20] for profile [internal]
>> 2009-10-29 10:19:45.998790 [NOTICE] sofia.c:3093 Adding Alias
>> [default] for profile [internal]
>> 2009-10-29 10:19:45.998790 [NOTICE] sofia.c:3108 Started Profile
>> internal [sofia_reg_internal]
>> 2009-10-29 10:19:53.354724 [ERR] sofia_reg.c:1425 fs_gateway
>> Registration Failed with status Forbidden [403]. failure #1
>> 2009-10-29 10:19:53.440735 [WARNING] sofia_reg.c:364 fs_gateway Failed
>> Registration, setting retry to 60 seconds.
>> 2009-10-29 10:20:53.108812 [NOTICE] sofia_reg.c:333 Registering fs_gateway
>> 2009-10-29 10:20:53.116813 [ERR] sofia_reg.c:1425 fs_gateway
>> Registration Failed with status Forbidden [403]. failure #2
>> 2009-10-29 10:20:53.208825 [WARNING] sofia_reg.c:364 fs_gateway Failed
>> Registration, setting retry to 90 seconds.
>>
>> gw is configured like that:
>>
>> <include>
>> ? ?<domain name="24ccdomain">
>> ? ? ? ?<params>
>> ? ? ? ? ? ?<param name="dial-string"
>> value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>> ? ? ? ?</params>
>>
>> ? ? ? ?<groups>
>> ? ? ? ? ? ?<group name="default">
>> ? ? ? ? ? ? ? ?<users>
>> ? ? ? ? ? ? ? ? ? ?<user id="24cc">
>> ? ? ? ? ? ? ? ? ? ? ? ?<params>
>> ? ? ? ? ? ? ? ? ? ? ? ? ? ?<param name="password" value="24ccpass"/>
>> ? ? ? ? ? ? ? ? ? ? ? ?</params>
>> ? ? ? ? ? ? ? ? ? ? ? ?<variables>
>> ? ? ? ? ? ? ? ? ? ? ? ? ? ?<variable name="user_context" value="default"/>
>> ? ? ? ? ? ? ? ? ? ? ? ? ? ?<variable name="effective_caller_id_name"
>> value="24 CallCenter"/>
>> ? ? ? ? ? ? ? ? ? ? ? ?</variables>
>> ? ? ? ? ? ? ? ? ? ?</user>
>> ? ? ? ? ? ? ? ?</users>
>> ? ? ? ? ? ?</group>
>> ? ? ? ?</groups>
>> ? ?</domain>
>> </include>
>>
>> and outputs:
>>
>> 2009-10-29 10:11:29.165700 [WARNING] sofia_reg.c:1737 Can't find user
>> [24cc at 192.168.1.20]
>> You must define a domain called '192.168.1.20' in your directory and
>> add a user with the id="24cc" attribute
>> and you must configure your device to use the proper domain in it's
>> authentication credentials.
>>
>> on each registration attempt from prod.
>>
>> This worked perfectly fine with one of previous revisions but started
>> failing on 15257. Using siptrace we were able to narrow down the issue
>> to prod sending an IP address as realm instead of gateway configured
>> realm (as it did in previous revision):
>>
>> Authorization: Digest username="24cc", realm="192.168.1.20",
>> nonce="4d82d624-c884-6340-8f7f-a4575f2583bd",
>> cnonce="Es23bj8MEi29T7UQ80i7sw", algorithm=MD5,
>> uri="sip:192.168.1.20:15060",
>> response="afea161876aad137657bf1f923d18659", qop=auth, nc=00000004
>>
>> Is there any configuration setting we are missing that would bring
>> back previous behavior?
>>
>> --
>> Best regards,
>> Robert Wilczynski
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org



From djbinter at yahoo.com  Fri Oct 30 11:31:12 2009
From: djbinter at yahoo.com (DJB)
Date: Fri, 30 Oct 2009 11:31:12 -0700 (PDT)
Subject: [Freeswitch-users] Freeswitch in signaling path only
In-Reply-To: <200910301300.01766.chris@cloudtel.com>
References: <429925.60305.qm@web37508.mail.mud.yahoo.com>
	<200910301300.01766.chris@cloudtel.com>
Message-ID: <312533.76921.qm@web37505.mail.mud.yahoo.com>

Now i have as follows, but it's still the same result.  By the way, I am running:  FreeSWITCH Version 1.0.4 (exported)
.
.
.
session:execute("set","hangup_after_bridge=true")
session:execute("set","continue_on_fail=true")
session:execute("set","originate_timeout=2")
session:execute("set","originate_retries=3")
session:execute("set","progress_timeout=15")
.
.
.
while row do
 local gw_ip_address = row.gw_ip_address
 local cust_name = row.cust_name
 session:execute("set", "accountcode=" ..cust_name .. "")
 session:execute("set","bypass_media=true")
 session:execute("bridge","sofia/external/" .. called_num .. "@XX.XX.XX.XX.146")
 session:execute("set","bypass_media=true")
 session:execute("bridge","sofia/external/" .. called_num .. "@XX.XX.XX.XX.105")
 -- Block for testing -- session:execute("bridge","sofia/external/" .. called_num .. "@" .. gw_ip_address .."")
 row = cur:fetch (row, "a")
end

Here is the debug for switch_ivr_originate.c:

2009-10-30 11:09:52.877832 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C
S_CONSUME_MEDIA
2009-10-30 11:09:53.17811 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX.1
46]
2009-10-30 11:09:54.285453 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.105) State Change CS_ROUTING -> C
S_CONSUME_MEDIA
2009-10-30 11:09:54.422426 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX.
105]
2009-10-30 11:09:55.694761 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C
S_CONSUME_MEDIA
2009-10-30 11:09:55.836036 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX.
146]
2009-10-30 11:09:57.107697 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.105) State Change CS_ROUTING -> C
S_CONSUME_MEDIA
2009-10-30 11:09:57.254664 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX.
105]
2009-10-30 11:12:03.129097 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C
S_CONSUME_MEDIA
2009-10-30 11:12:03.273055 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX.
146]
2009-10-30 11:12:04.546410 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.105) State Change CS_ROUTING -> C
S_CONSUME_MEDIA
2009-10-30 11:12:04.682661 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX.
105]
2009-10-30 11:12:15.781701 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 16 [NORMAL_CLEARING]
2009-10-30 11:12:33.349162 [DEBUG] switch_ivr_originate.c:63 (sofia/external/6463924215 at XX.XX.XX.146) State Change CS_ROUTING -> C
S_CONSUME_MEDIA
2009-10-30 11:12:33.470989 [DEBUG] switch_ivr_originate.c:2061 Originate Resulted in Success: [sofia/external/6463924215 at XX.XX.XX.
146]
2009-10-30 11:12:34.724641 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL]
2009-10-30 11:12:34.730634 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL]
2009-10-30 11:12:34.750637 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL]


FIRST ROUTE: XX.XX.XX.146 
and I tried to failed the first route and it gave 500 back, then it goes to the next one.
SECOND ROUTE:  XX.XX.XX.105


Thank you,
Dorn B.



----- Original Message ----
From: Chris Burns <chris at cloudtel.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Fri, October 30, 2009 10:00:01 AM
Subject: Re: [Freeswitch-users] Freeswitch in signaling path only

Do you have a debug level message from switch_ivr_originate.c in your log?
"Channel is already up, delaying proxy mode 'till both legs are answered."

Set bypass_media b4 each bridge. It is unsetting on you and setting 
bypass_media_after_bridge because you already answered the channel running 
the lua script.

On October 30, 2009 12:03:29 pm DJB wrote:
> I am wondering why I cannot do as condition#2.
>
> For Lua in dialplan, when I have the followings:
>
>
> --WORKING--
> (Condition#1)
> .
> .
> session:execute("set","bypass_media=true")
> session:execute("set","hangup_after_bridge=true")
> session:execute("set","continue_on_fail=true")
> .
> .
>  session:execute("bridge","sofia/external/" .. called_num ..
> "@1.1.1.1|sofia/external/" .. called_num .. "@1.1.1.2") .
> .
>
> --NOT WORKING--
> (Condition#2)
> Note:  FS tries to be in media path and send re-invite.
> .
> .
> session:execute("set","bypass_media=true")
> session:execute("set","hangup_after_bridge=true")
> session:execute("set","continue_on_fail=true")
> .
> .
>  session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.1")
>  session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.2")
> .
> .
>
> Thank you,
> Dorn B.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



      



From pedroprado at msn.com  Fri Oct 30 11:53:00 2009
From: pedroprado at msn.com (Pedro Prado)
Date: Fri, 30 Oct 2009 16:53:00 -0200
Subject: [Freeswitch-users] Control Panel
Message-ID: <SNT102-W5248A31FD39933F270318BC1B60@phx.gbl>


Hi,

Which control panel are you using with FreeSwitch? WikiPbx, FreePbx or other?

Thanks,
Pedro Prado
 		 	   		  
_________________________________________________________________
Voc? sabia que com o Hotmail voc? tem espa?o ilimitado para guardar seus e-mails? Come?e a usar j?!
http://www.microsoft.com/brasil/windows/windowslive/products/hotmail.aspx
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From tleyden at branchcut.com  Fri Oct 30 12:16:34 2009
From: tleyden at branchcut.com (Traun Leyden)
Date: Fri, 30 Oct 2009 23:46:34 +0430
Subject: [Freeswitch-users] Control Panel
In-Reply-To: <SNT102-W5248A31FD39933F270318BC1B60@phx.gbl>
References: <SNT102-W5248A31FD39933F270318BC1B60@phx.gbl>
Message-ID: <f9f80d540910301216u669f19c2tb3845b93d261247c@mail.gmail.com>

>From watching the mailing list, I've seen:

95% are using straight XML combined with their home-cooked solutions based
on mod_xml_curl, which makes it pretty easy to write your own custom GUI for
freeswitch.

Out of the rest, the most are probably using pfsense, freepbx, and a small
handful of people using wikipbx (based on forum activity, guess would be
50ish).  There's also TCAPI (php).

Domain-based multi-tenancy is on the horizon for wikipbx, which has been
chugging along slowly, but has not died.  BroadTone (broadtone.net) has
kindly donated some test servers and a web developer, so kudos goes out to
them.

2009/10/30 Pedro Prado <pedroprado at msn.com>

>  Hi,
>
> Which control panel are you using with FreeSwitch? WikiPbx, FreePbx or
> other?
>
> Thanks,
> Pedro Prado
>
> ------------------------------
> Navegue com seguran?a com o Novo Internet Explorer 8. Baixe agora, ?
> gratis!<http://brasil.microsoft.com.br/IE8/mergulhe/?utm_source=MSN%3BHotmail&utm_medium=Tagline&utm_campaign=IE8>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Fri Oct 30 12:19:39 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 30 Oct 2009 12:19:39 -0700
Subject: [Freeswitch-users] Control Panel
In-Reply-To: <SNT102-W5248A31FD39933F270318BC1B60@phx.gbl>
References: <SNT102-W5248A31FD39933F270318BC1B60@phx.gbl>
Message-ID: <87f2f3b90910301219j45531d15y70482aaa4fb923a2@mail.gmail.com>

2009/10/30 Pedro Prado <pedroprado at msn.com>

>  Hi,
>
> Which control panel are you using with FreeSwitch? WikiPbx, FreePbx or
> other?
>
These are all at various stages of production/usability:
Most of the people here just use emacs/vim/etc. and then use the built-in
webserver for simple stuff. (see http://wiki.freeswitch.org/wiki/Webapi)

For the others:
WikiPBX is an enigma. I haven't heard of anything new in quite some time. I
don't recommend it.
FreePBX V3 - very promising, in early development, don't expect anything
usable right away
FusionPBX - Farther along, simpler than FreePBX, closest to being usable in
production
FSGUI - Donated to contrib by jmesquita; simple, work in progress,
interesting to play with

CudaTel - It's the bomb! ;)

-MC
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From msc at freeswitch.org  Fri Oct 30 12:32:45 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 30 Oct 2009 12:32:45 -0700
Subject: [Freeswitch-users] PBX vs. SIP proxy?
In-Reply-To: <26128938.post@talk.nabble.com>
References: <25920788.post@talk.nabble.com> <4AD914FE.4020709@gmail.com>
	<26128938.post@talk.nabble.com>
Message-ID: <87f2f3b90910301232v16191998occa33e0485598da1@mail.gmail.com>

On Fri, Oct 30, 2009 at 5:06 AM, Fred-145 <codecomplete at free.fr> wrote:

>
> Thanks Meftah for the help. So it looks like an SIP proxy is just to handle
> the registration process, while a PBX offers services like ringing phones,
> setting up calls, handling voice mail, etc.
>

Of course, as usual, the lines are getting blurred. Look at opensips.org and
see all the crazy things that a good SIP proxy can actually do.
-MC
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From codecomplete at free.fr  Fri Oct 30 13:22:45 2009
From: codecomplete at free.fr (Fred-145)
Date: Fri, 30 Oct 2009 13:22:45 -0700 (PDT)
Subject: [Freeswitch-users] PBX vs. SIP proxy?
In-Reply-To: <87f2f3b90910301232v16191998occa33e0485598da1@mail.gmail.com>
References: <25920788.post@talk.nabble.com> <4AD914FE.4020709@gmail.com>
	<26128938.post@talk.nabble.com>
	<87f2f3b90910301232v16191998occa33e0485598da1@mail.gmail.com>
Message-ID: <26136363.post@talk.nabble.com>


Thanks for the link. But then, since OpenSIPs doesn't run on Windows, I'll
stick to FreeSwitch :)
-- 
View this message in context: http://old.nabble.com/PBX-vs.-SIP-proxy--tp25920788p26136363.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From ivan at myrvold.org  Fri Oct 30 13:26:23 2009
From: ivan at myrvold.org (Ivan C Myrvold)
Date: Fri, 30 Oct 2009 21:26:23 +0100
Subject: [Freeswitch-users] SIP provider with extern rtp server
In-Reply-To: <bf6923ed0910301030x375fd6b9m75328c0e3a58b70f@mail.gmail.com>
References: <A2D2E918-B494-4179-96BA-CA3A7717846C@myrvold.org>
	<bf6923ed0910270658t61bb6687ne0b7d3ec5090e2ff@mail.gmail.com>
	<CB8008BD-4066-4B5B-9043-214CFE938CD8@myrvold.org>
	<bf6923ed0910271034w6c7e0cc5t685a5377764755e6@mail.gmail.com>
	<FA2C693B-FCE1-4BDD-BCE5-302AC699CEDB@myrvold.org>
	<87f2f3b90910281112g6e72d22elcfd653991ecd50cc@mail.gmail.com>
	<8AC09649-2585-4BE7-A959-A7AC41650789@myrvold.org>
	<bf6923ed0910281921uad8c525i850c08361b21538d@mail.gmail.com>
	<544D39F2-40AB-41B4-BF18-89D7492B17EE@myrvold.org>
	<bf6923ed0910301030x375fd6b9m75328c0e3a58b70f@mail.gmail.com>
Message-ID: <8BB98561-BAA3-46B4-939F-FBA5EF79BD06@myrvold.org>

Yes, now I got a more detailed trace. Thank you for helping me with  
this.

A new pastebin at http://pastebin.freeswitch.org/10905

Ivan

Den 30. okt. 2009 kl. 18:30 skrev Eliot Gable:

> fsctl loglevel debug
> console loglevel debug
> sofia profile internal siptrace on
> sofia profile external siptrace on
> sofia loglevel all 9
> ^^^^^^^^^^^^^^^^^^^^^
>
> Then run your call, then do this:
>
> sofia loglevel all 0
> sofia profile external siptrace off
> sofia profile internal siptrace off
> fsctl loglevel warning
> console loglevel warning
>
> On Fri, Oct 30, 2009 at 12:16 PM, Ivan C Myrvold <ivan at myrvold.org>  
> wrote:
>> I have already set debug to 9, on both profiles.
>>
>> Ivan
>>
>>
>> Den 29. okt. 2009 kl. 03:21 skrev Eliot Gable:
>>
>>> See that 200 OK that keeps coming in over and over and over and over
>>> again? That's because they never received your ACK. If you can  
>>> turn on
>>> sofia loglevel to 9 and then watch where you send the ACK, you will
>>> probably have your answer to why the other system did not receive  
>>> it.
>>> If you're still not sure what's going on, post another pastebin with
>>> sofia loglevel set to 9.
>>>
>>>
>>> On Wed, Oct 28, 2009 at 4:51 PM, Ivan C Myrvold <ivan at myrvold.org>
>>> wrote:
>>>> Oh, what happened to it?
>>>> Anyway, here is a new pb:
>>>> http://pastebin.freeswitch.org/10867
>>>> Ivan
>>>> Den 28. okt. 2009 kl. 19:12 skrev Michael Collins:
>>>>
>>>>
>>>> On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold <ivan at myrvold.org>
>>>> wrote:
>>>>>
>>>>> Here is a debug log from a call from an internal phone out to an
>>>>> external (my iPhone with nbr 91316356):
>>>>> http://pastebin.freeswitch.org/108578
>>>>>
>>>>> Ivan
>>>>>
>>>> Uh... you wanna try that PB number again?
>>>> -MC
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>> users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>> users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>>
>>> --
>>> Eliot Gable
>>>
>>> "We do not inherit the Earth from our ancestors: we borrow it from  
>>> our
>>> children." ~David Brower
>>>
>>> "I decided the words were too conservative for me. We're not  
>>> borrowing
>>> from our children, we're stealing from them--and it's not even
>>> considered to be a crime." ~David Brower
>>>
>>> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
>>> live; not live to eat.) ~Marcus Tullius Cicero
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>
>
>
> -- 
> Eliot Gable
>
> "We do not inherit the Earth from our ancestors: we borrow it from our
> children." ~David Brower
>
> "I decided the words were too conservative for me. We're not borrowing
> from our children, we're stealing from them--and it's not even
> considered to be a crime." ~David Brower
>
> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
> live; not live to eat.) ~Marcus Tullius Cicero
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>




From diego.viola at gmail.com  Fri Oct 30 13:28:48 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 30 Oct 2009 20:28:48 +0000
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <86a32abc0910162312le6c08b8jb53ae83bcafadab1@mail.gmail.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
	<6b65470d0910131009y1cf136c3q8c0932375c19f272@mail.gmail.com>
	<86a32abc0910161510ma323718g410bcf1029156077@mail.gmail.com>
	<a87d43fd0910161729u42db5ea7qafb5604c0680954@mail.gmail.com>
	<86a32abc0910161811h15eac05dwc3fe0c2533b7397a@mail.gmail.com>
	<86a32abc0910161847k415b9722ubc5439ea1b9f6e78@mail.gmail.com>
	<86a32abc0910161854k7fd3fca6pc0686e928c772d2c@mail.gmail.com>
	<86a32abc0910162050m6bad9a9do171b268e083b3d0d@mail.gmail.com>
	<87f2f3b90910162254q23893e4cx160fe4a5ae64cc91@mail.gmail.com>
	<86a32abc0910162312le6c08b8jb53ae83bcafadab1@mail.gmail.com>
Message-ID: <86a32abc0910301328h642726et2bae30997f8f37ae@mail.gmail.com>

This needs to be converted to documentation form also.

http://wiki.freeswitch.org/wiki/Proxy_Media#How_to_detect_when_it_should_be_used

On Sat, Oct 17, 2009 at 6:12 AM, Diego Viola <diego.viola at gmail.com> wrote:

> Thanks, our wiki is getting a lot better :).
>
> Diego
>
> On Sat, Oct 17, 2009 at 5:54 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Diego, when I get some time I will help convert these from IRC to wiki.
>> Thanks,
>> MC
>>
>>
>> On Fri, Oct 16, 2009 at 8:50 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>
>>> This needs to be converted to documentation form too.
>>>
>>> http://wiki.freeswitch.org/wiki/Mod_vmd#Related_conversations
>>>
>>> Diego
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From msc at freeswitch.org  Fri Oct 30 14:05:03 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 30 Oct 2009 14:05:03 -0700
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <86a32abc0910301328h642726et2bae30997f8f37ae@mail.gmail.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
	<86a32abc0910161510ma323718g410bcf1029156077@mail.gmail.com>
	<a87d43fd0910161729u42db5ea7qafb5604c0680954@mail.gmail.com>
	<86a32abc0910161811h15eac05dwc3fe0c2533b7397a@mail.gmail.com>
	<86a32abc0910161847k415b9722ubc5439ea1b9f6e78@mail.gmail.com>
	<86a32abc0910161854k7fd3fca6pc0686e928c772d2c@mail.gmail.com>
	<86a32abc0910162050m6bad9a9do171b268e083b3d0d@mail.gmail.com>
	<87f2f3b90910162254q23893e4cx160fe4a5ae64cc91@mail.gmail.com>
	<86a32abc0910162312le6c08b8jb53ae83bcafadab1@mail.gmail.com>
	<86a32abc0910301328h642726et2bae30997f8f37ae@mail.gmail.com>
Message-ID: <87f2f3b90910301405j4fdb3f48we864e386fdc878ad@mail.gmail.com>

thanks for the heads up! I'll ask someone in the know to fill that out.
-MC

On Fri, Oct 30, 2009 at 1:28 PM, Diego Viola <diego.viola at gmail.com> wrote:

> This needs to be converted to documentation form also.
>
>
> http://wiki.freeswitch.org/wiki/Proxy_Media#How_to_detect_when_it_should_be_used
>
>
> On Sat, Oct 17, 2009 at 6:12 AM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> Thanks, our wiki is getting a lot better :).
>>
>> Diego
>>
>> On Sat, Oct 17, 2009 at 5:54 AM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Diego, when I get some time I will help convert these from IRC to wiki.
>>> Thanks,
>>> MC
>>>
>>>
>>> On Fri, Oct 16, 2009 at 8:50 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>
>>>> This needs to be converted to documentation form too.
>>>>
>>>> http://wiki.freeswitch.org/wiki/Mod_vmd#Related_conversations
>>>>
>>>> Diego
>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From diego.viola at gmail.com  Fri Oct 30 14:36:27 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 30 Oct 2009 21:36:27 +0000
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <87f2f3b90910301405j4fdb3f48we864e386fdc878ad@mail.gmail.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
	<a87d43fd0910161729u42db5ea7qafb5604c0680954@mail.gmail.com>
	<86a32abc0910161811h15eac05dwc3fe0c2533b7397a@mail.gmail.com>
	<86a32abc0910161847k415b9722ubc5439ea1b9f6e78@mail.gmail.com>
	<86a32abc0910161854k7fd3fca6pc0686e928c772d2c@mail.gmail.com>
	<86a32abc0910162050m6bad9a9do171b268e083b3d0d@mail.gmail.com>
	<87f2f3b90910162254q23893e4cx160fe4a5ae64cc91@mail.gmail.com>
	<86a32abc0910162312le6c08b8jb53ae83bcafadab1@mail.gmail.com>
	<86a32abc0910301328h642726et2bae30997f8f37ae@mail.gmail.com>
	<87f2f3b90910301405j4fdb3f48we864e386fdc878ad@mail.gmail.com>
Message-ID: <86a32abc0910301436q25cbbdc8h182c586819ccad34@mail.gmail.com>

No problem, when I have some time I need to merge these two pages also.

http://wiki.freeswitch.org/wiki/Multi-tenant
http://wiki.freeswitch.org/wiki/Multiple_Companies

=D

Diego


On Fri, Oct 30, 2009 at 9:05 PM, Michael Collins <msc at freeswitch.org> wrote:

> thanks for the heads up! I'll ask someone in the know to fill that out.
> -MC
>
>
> On Fri, Oct 30, 2009 at 1:28 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> This needs to be converted to documentation form also.
>>
>>
>> http://wiki.freeswitch.org/wiki/Proxy_Media#How_to_detect_when_it_should_be_used
>>
>>
>> On Sat, Oct 17, 2009 at 6:12 AM, Diego Viola <diego.viola at gmail.com>wrote:
>>
>>> Thanks, our wiki is getting a lot better :).
>>>
>>> Diego
>>>
>>> On Sat, Oct 17, 2009 at 5:54 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> Diego, when I get some time I will help convert these from IRC to wiki.
>>>> Thanks,
>>>> MC
>>>>
>>>>
>>>> On Fri, Oct 16, 2009 at 8:50 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>>
>>>>> This needs to be converted to documentation form too.
>>>>>
>>>>> http://wiki.freeswitch.org/wiki/Mod_vmd#Related_conversations
>>>>>
>>>>> Diego
>>>>>
>>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Fri Oct 30 14:52:51 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 30 Oct 2009 22:52:51 +0100
Subject: [Freeswitch-users] mod_t38gateway
Message-ID: <65d96fc80910301452x2a831733h92b05861d6c94123@mail.gmail.com>

does anybody know how does it work and how to use it in a dialplan?


freeswitch at nemesis>
freeswitch at nemesis>
freeswitch at nemesis> load mod_t38gateway
API CALL [load(mod_t38gateway)] output:
+OK

2009-10-30 22:44:38.204268 [NOTICE] mod_t38gateway.c:147 T.38 gateway
enabled
2009-10-30 22:44:38.204268 [CONSOLE] switch_loadable_module.c:889
Successfully Loaded [mod_t38gateway]
2009-10-30 22:44:38.204268 [NOTICE] switch_loadable_module.c:248 Adding
Application 't38gateway'
2009-10-30 22:44:38.205374 [NOTICE] switch_loadable_module.c:270 Adding API
Function 't38gateway'
freeswitch at nemesis>
freeswitch at nemesis>
freeswitch at nemesis>
freeswitch at nemesis> t38gateway
API CALL [t38gateway()] output:
-USAGE: <uuid> <command>

freeswitch at nemesis>

T.
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From hjqlopez at hotmail.com  Fri Oct 30 15:07:04 2009
From: hjqlopez at hotmail.com (Humberto Quintana)
Date: Fri, 30 Oct 2009 18:07:04 -0400
Subject: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media
Message-ID: <COL104-W179212DC517C7BD2DB189D0B60@phx.gbl>


Hi everybody,

I'd like some help with this situation that is 'haunting' me :-) 

My scenario is as follows:

inbound-bypass-media is set in the profile because we dont want FS handling the media.

1. A calls B
2. FS sends to B the A's SDP
3. B answers
4. FS sends to A the B's SDP
5. Media going directly between A and B
6. B REFERs the call to C (blind transfer with no reINVITE for Hold)
7. FS accepts(202) the REFER and sends the NOTIFY
7a. B and FS send the BYE
8. FS sends an INVITE  to C with A's SDP
9. C answers
10. FS doesn't send a reINVITE to A to let it know about C's SDP


Is that the expected FS behavior or is this a bug?


I'd appreciate any hints. Thanks,


Humberto


 		 	   		  
_________________________________________________________________
Lots of fantastic Windows 7 offers, in one convenient place. Get the perfect deal for you now.
http://go.microsoft.com/?linkid=9691633
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From diego.viola at gmail.com  Fri Oct 30 15:09:34 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 30 Oct 2009 22:09:34 +0000
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <86a32abc0910301436q25cbbdc8h182c586819ccad34@mail.gmail.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
	<86a32abc0910161811h15eac05dwc3fe0c2533b7397a@mail.gmail.com>
	<86a32abc0910161847k415b9722ubc5439ea1b9f6e78@mail.gmail.com>
	<86a32abc0910161854k7fd3fca6pc0686e928c772d2c@mail.gmail.com>
	<86a32abc0910162050m6bad9a9do171b268e083b3d0d@mail.gmail.com>
	<87f2f3b90910162254q23893e4cx160fe4a5ae64cc91@mail.gmail.com>
	<86a32abc0910162312le6c08b8jb53ae83bcafadab1@mail.gmail.com>
	<86a32abc0910301328h642726et2bae30997f8f37ae@mail.gmail.com>
	<87f2f3b90910301405j4fdb3f48we864e386fdc878ad@mail.gmail.com>
	<86a32abc0910301436q25cbbdc8h182c586819ccad34@mail.gmail.com>
Message-ID: <86a32abc0910301509i10ac4f21rc90d4f6130e0759e@mail.gmail.com>

Another one to convert:

http://wiki.freeswitch.org/wiki/User:Agx#CAMP_ON

On Fri, Oct 30, 2009 at 9:36 PM, Diego Viola <diego.viola at gmail.com> wrote:

> No problem, when I have some time I need to merge these two pages also.
>
> http://wiki.freeswitch.org/wiki/Multi-tenant
> http://wiki.freeswitch.org/wiki/Multiple_Companies
>
> =D
>
> Diego
>
>
>
> On Fri, Oct 30, 2009 at 9:05 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> thanks for the heads up! I'll ask someone in the know to fill that out.
>> -MC
>>
>>
>> On Fri, Oct 30, 2009 at 1:28 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>
>>> This needs to be converted to documentation form also.
>>>
>>>
>>> http://wiki.freeswitch.org/wiki/Proxy_Media#How_to_detect_when_it_should_be_used
>>>
>>>
>>> On Sat, Oct 17, 2009 at 6:12 AM, Diego Viola <diego.viola at gmail.com>wrote:
>>>
>>>> Thanks, our wiki is getting a lot better :).
>>>>
>>>> Diego
>>>>
>>>> On Sat, Oct 17, 2009 at 5:54 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>> Diego, when I get some time I will help convert these from IRC to wiki.
>>>>> Thanks,
>>>>> MC
>>>>>
>>>>>
>>>>> On Fri, Oct 16, 2009 at 8:50 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>>>
>>>>>> This needs to be converted to documentation form too.
>>>>>>
>>>>>> http://wiki.freeswitch.org/wiki/Mod_vmd#Related_conversations
>>>>>>
>>>>>> Diego
>>>>>>
>>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From msc at freeswitch.org  Fri Oct 30 15:19:32 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 30 Oct 2009 15:19:32 -0700
Subject: [Freeswitch-users] Some documentation thoughts
In-Reply-To: <86a32abc0910301509i10ac4f21rc90d4f6130e0759e@mail.gmail.com>
References: <87f2f3b90910121538q5cb6d58fq5d12906c8e5d354e@mail.gmail.com>
	<86a32abc0910161847k415b9722ubc5439ea1b9f6e78@mail.gmail.com>
	<86a32abc0910161854k7fd3fca6pc0686e928c772d2c@mail.gmail.com>
	<86a32abc0910162050m6bad9a9do171b268e083b3d0d@mail.gmail.com>
	<87f2f3b90910162254q23893e4cx160fe4a5ae64cc91@mail.gmail.com>
	<86a32abc0910162312le6c08b8jb53ae83bcafadab1@mail.gmail.com>
	<86a32abc0910301328h642726et2bae30997f8f37ae@mail.gmail.com>
	<87f2f3b90910301405j4fdb3f48we864e386fdc878ad@mail.gmail.com>
	<86a32abc0910301436q25cbbdc8h182c586819ccad34@mail.gmail.com>
	<86a32abc0910301509i10ac4f21rc90d4f6130e0759e@mail.gmail.com>
Message-ID: <87f2f3b90910301519x54ac6c6fr13d546b19e493090@mail.gmail.com>

actually that's a user's personal page so you don't have to worry about that
one. :)
-MC

On Fri, Oct 30, 2009 at 3:09 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Another one to convert:
>
> http://wiki.freeswitch.org/wiki/User:Agx#CAMP_ON
>
>
> On Fri, Oct 30, 2009 at 9:36 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> No problem, when I have some time I need to merge these two pages also.
>>
>> http://wiki.freeswitch.org/wiki/Multi-tenant
>> http://wiki.freeswitch.org/wiki/Multiple_Companies
>>
>> =D
>>
>> Diego
>>
>>
>>
>> On Fri, Oct 30, 2009 at 9:05 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> thanks for the heads up! I'll ask someone in the know to fill that out.
>>> -MC
>>>
>>>
>>> On Fri, Oct 30, 2009 at 1:28 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>
>>>> This needs to be converted to documentation form also.
>>>>
>>>>
>>>> http://wiki.freeswitch.org/wiki/Proxy_Media#How_to_detect_when_it_should_be_used
>>>>
>>>>
>>>> On Sat, Oct 17, 2009 at 6:12 AM, Diego Viola <diego.viola at gmail.com>wrote:
>>>>
>>>>> Thanks, our wiki is getting a lot better :).
>>>>>
>>>>> Diego
>>>>>
>>>>> On Sat, Oct 17, 2009 at 5:54 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>
>>>>>> Diego, when I get some time I will help convert these from IRC to
>>>>>> wiki.
>>>>>> Thanks,
>>>>>> MC
>>>>>>
>>>>>>
>>>>>> On Fri, Oct 16, 2009 at 8:50 PM, Diego Viola <diego.viola at gmail.com>wrote:
>>>>>>
>>>>>>> This needs to be converted to documentation form too.
>>>>>>>
>>>>>>> http://wiki.freeswitch.org/wiki/Mod_vmd#Related_conversations
>>>>>>>
>>>>>>> Diego
>>>>>>>
>>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Fri Oct 30 15:25:50 2009
From: brian at freeswitch.org (Brian West)
Date: Fri, 30 Oct 2009 17:25:50 -0500
Subject: [Freeswitch-users] no REINVITE on Blind Transfer with
	bypass_media
In-Reply-To: <COL104-W179212DC517C7BD2DB189D0B60@phx.gbl>
References: <COL104-W179212DC517C7BD2DB189D0B60@phx.gbl>
Message-ID: <41A5BD86-AC6C-4E0F-BD93-229DD9AF9622@freeswitch.org>

First you forgot to mention what SVN rev you're on...

/b

On Oct 30, 2009, at 5:07 PM, Humberto Quintana wrote:

> Hi everybody,
>
> I'd like some help with this situation that is 'haunting' me :-)
>
> My scenario is as follows:
>
> inbound-bypass-media is set in the profile because we dont want FS  
> handling the media.
>
> 1. A calls B
> 2. FS sends to B the A's SDP
> 3. B answers
> 4. FS sends to A the B's SDP
> 5. Media going directly between A and B
> 6. B REFERs the call to C (blind transfer with no reINVITE for Hold)
> 7. FS accepts(202) the REFER and sends the NOTIFY
> 7a. B and FS send the BYE
> 8. FS sends an INVITE  to C with A's SDP
> 9. C answers
> 10. FS doesn't send a reINVITE to A to let it know about C's SDP
>
>
> Is that the expected FS behavior or is this a bug?
>
>
> I'd appreciate any hints. Thanks,
>
>
> Humberto
>
>
>
> Lots of fantastic offers on Windows 7, in one convenient place. Get  
> a deal on Windows 7 now  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Fri Oct 30 15:58:40 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 30 Oct 2009 17:58:40 -0500
Subject: [Freeswitch-users] no REINVITE on Blind Transfer with
	bypass_media
In-Reply-To: <COL104-W179212DC517C7BD2DB189D0B60@phx.gbl>
References: <COL104-W179212DC517C7BD2DB189D0B60@phx.gbl>
Message-ID: <191c3a030910301558t505ce395u9ccbd36ce3df3033@mail.gmail.com>

profile options

<param name="media-option" value="resume-media-on-hold"/>
<param name="media-option" value="bypass-media-after-att-xfer"/>


On Fri, Oct 30, 2009 at 5:07 PM, Humberto Quintana <hjqlopez at hotmail.com>wrote:

>  Hi everybody,
>
> I'd like some help with this situation that is 'haunting' me :-)
>
> My scenario is as follows:
>
> inbound-bypass-media is set in the profile because we dont want FS handling
> the media.
>
> 1. A calls B
> 2. FS sends to B the A's SDP
> 3. B answers
> 4. FS sends to A the B's SDP
> 5. Media going directly between A and B
> 6. B REFERs the call to C (blind transfer with no reINVITE for Hold)
> 7. FS accepts(202) the REFER and sends the NOTIFY
> 7a. B and FS send the BYE
> 8. FS sends an INVITE  to C with A's SDP
> 9. C answers
> 10. FS doesn't send a reINVITE to A to let it know about C's SDP
>
>
> Is that the expected FS behavior or is this a bug?
>
>
> I'd appreciate any hints. Thanks,
>
>
> Humberto
>
>
>
> ------------------------------
> Lots of fantastic offers on Windows 7, in one convenient place. Get a deal
> on Windows 7 now <http://go.microsoft.com/?linkid=9691628>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From dome at tel.co.th  Sat Oct 31 04:08:24 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Sat, 31 Oct 2009 18:08:24 +0700
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
Message-ID: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>

Dear All,
          I'm disable all CDR and use nibble_bill (modify) and
odbc_query for create CDR and dedcut balance. but i found many zombie
channel and can't kill
Like a
51555b02-c5ba-11de-af77-774e15238dbe,outbound,2009-10-30
18:11:38,1256951498,sofia/external/66848087977 at xxx.xxx.xx.191:7050,CS_REPORTING,FreeSWITCH,66850304266,xxx.xxx.xxx.191,7050,,,XML,public,,,,,

I'm using FreeSWITCH Version 1.0.5pre3 (hacked)

How to fix ?

Dome C.



From info at daccii.it  Sat Oct 31 05:40:31 2009
From: info at daccii.it (Albano Daniele Salvatore - Lavoro)
Date: Sat, 31 Oct 2009 13:40:31 +0100
Subject: [Freeswitch-users] Freeswitch seems to doesn't reknow dial tone
 after the first call using OpenZAP (analog spans)
In-Reply-To: <87f2f3b90910291026u2c62ded6v5b9458a9c2e85877@mail.gmail.com>
References: <4AE8B20D.4040608@daccii.it>	<191c3a030910290912x9489784s4cc8e224363f3796@mail.gmail.com>
	<87f2f3b90910291026u2c62ded6v5b9458a9c2e85877@mail.gmail.com>
Message-ID: <4AEC303F.4020207@daccii.it>

Hi,

i've done more tests, with svn too, but nothing to do. The strange thing 
is when freeswitch shutdown and close zaptel channel i get back the dial 
tone from my provider (i'm checking trought ztmonitor 1 -vv). Can be 
something is missing when span get closed?

PS1: with svn i can do only one call, but i can receive calls without 
problems
PS2: i joined yesterday on irc but i had few time ... hope to join monday


Best Regards,
Daniele

Michael Collins ha scritto:
> 
> 
> On Thu, Oct 29, 2009 at 9:12 AM, Anthony Minessale 
> <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>> wrote:
> 
>     You may want to work with FreeSWITCH svn trunk for the time being.
>     There have been some changes and if you have a problem to fix it's
>     easier to debug with that version.
> 
> 
>     2009/10/28 Albano Daniele Salvatore - Lavoro <info at daccii.it
>     <mailto:info at daccii.it>>
> 
>         Hi to all,
> 
>         this is my first message here!
> 
> Welcome!
> 
> Like Tony said, the latest SVN trunk is probably what you want. We are 
> already doing pre-releases of 1.0.5. SVN trunk is extremely stable. 
> Also, you can get realtime help via IRC at irc.freenode.net 
> <http://irc.freenode.net>, channels #freeswitch and #openzap.
> 
> -MC
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From mike at jerris.com  Sat Oct 31 08:25:23 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 31 Oct 2009 11:25:23 -0400
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <5e414ed0910240513q316905ai5cf8c2ef63b52f60@mail.gmail.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>
	<191c3a030910131101p418e0961t373db68285248ff6@mail.gmail.com>
	<538F4CF43B294D0EACABA281E1913B4C@chaos>
	<65d96fc80910131224m73f35b87g355be818ad3c0caf@mail.gmail.com>
	<48B637E0506F4A809E4C0CAF39BD08BB@chaos>
	<65d96fc80910132348t202905fbub57cc4c814eb4e21@mail.gmail.com>
	<5e414ed0910140731w1c7ebedr150e69cda8073155@mail.gmail.com>
	<191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com>
	<5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com>
	<191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com>
	<5e414ed0910240513q316905ai5cf8c2ef63b52f60@mail.gmail.com>
Message-ID: <ACB65510-C38F-4345-B555-BD2DCF61B444@jerris.com>

This may be possible with tcp, how could this work on udp?  Can you  
provide an rfc reference on this?

Mike

On Oct 24, 2009, at 8:13 AM, Dennis wrote:

> ok, as written, i come back after some tests with fs and a thomson  
> cirpack.
>
> it did not work - at least in our tests.
>
> we are using socket outbound and when a call comes in, it starts the
> socket of fs. the number may be 123456. fs sends the respond 484 and
> our carrier receives this information. but fs ends the call with
> hangup_cause = invalid_number_format.
> the carrier has one more digit for the phone number and sends 1234567
> and the above mentioned behavior repeats.
>
> the behavior we want and expected is, that the call stays in the
> socket after response 484, so that the carrier can send the 1234567
> into the same socket.
> the management, when fs should send response 484 and when fs should be
> answered would be programmed by us.
> it also important, that fs keeps the call in the socket, so we can
> tell fs, to answer the call after x seconds anyway.
>
> any ideas, what we could do?




From freeswitch-list at puzzled.xs4all.nl  Sat Oct 31 08:48:53 2009
From: freeswitch-list at puzzled.xs4all.nl (Patrick List)
Date: Sat, 31 Oct 2009 16:48:53 +0100
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <ACB65510-C38F-4345-B555-BD2DCF61B444@jerris.com>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>	<191c3a030910131101p418e0961t373db68285248ff6@mail.gmail.com>	<538F4CF43B294D0EACABA281E1913B4C@chaos>	<65d96fc80910131224m73f35b87g355be818ad3c0caf@mail.gmail.com>	<48B637E0506F4A809E4C0CAF39BD08BB@chaos>	<65d96fc80910132348t202905fbub57cc4c814eb4e21@mail.gmail.com>	<5e414ed0910140731w1c7ebedr150e69cda8073155@mail.gmail.com>	<191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com>	<5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com>	<191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com>	<5e414ed0910240513q316905ai5cf8c2ef63b52f60@mail.gmail.com>
	<ACB65510-C38F-4345-B555-BD2DCF61B444@jerris.com>
Message-ID: <4AEC5C65.6050800@puzzled.xs4all.nl>

> On Oct 24, 2009, at 8:13 AM, Dennis wrote:
>
>> ok, as written, i come back after some tests with fs and a thomson
>> cirpack.

No idea if this is useful as I'm a noob with fs. If not please excuse 
the noise. In the past Asterisk to work properly with Cirpack needed the 
following patch:

diff -uNr asterisk-1.4.19.org/main/rtp.c asterisk-1.4.19/main/rtp.c
--- asterisk-1.4.19.org/main/rtp.c	2007-10-08 22:06:33.000000000 +0200
+++ asterisk-1.4.19/main/rtp.c	2007-11-11 13:12:28.000000000 +0100
@@ -1383,6 +1383,7 @@
  	[34] = {1, AST_FORMAT_H263},
  	[103] = {1, AST_FORMAT_H263_PLUS},
  	[97] = {1, AST_FORMAT_ILBC},
+	[96] = {0, AST_RTP_DTMF},
  	[99] = {1, AST_FORMAT_H264},
  	[101] = {0, AST_RTP_DTMF},
  	[110] = {1, AST_FORMAT_SPEEX},

Maybe this helps.

Regards,
Patrick



From brian at freeswitch.org  Sat Oct 31 08:58:37 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 31 Oct 2009 10:58:37 -0500
Subject: [Freeswitch-users] SIP Overlap support?
In-Reply-To: <4AEC5C65.6050800@puzzled.xs4all.nl>
References: <5e414ed0910130651s69a55d75sc189c999800ea28c@mail.gmail.com>	<191c3a030910131101p418e0961t373db68285248ff6@mail.gmail.com>	<538F4CF43B294D0EACABA281E1913B4C@chaos>	<65d96fc80910131224m73f35b87g355be818ad3c0caf@mail.gmail.com>	<48B637E0506F4A809E4C0CAF39BD08BB@chaos>	<65d96fc80910132348t202905fbub57cc4c814eb4e21@mail.gmail.com>	<5e414ed0910140731w1c7ebedr150e69cda8073155@mail.gmail.com>	<191c3a030910140747s629ecf34h7c3beb34ed6e521@mail.gmail.com>	<5e414ed0910150047h100fe0cex71981629e29eaed5@mail.gmail.com>	<191c3a030910150653w170ef943w4822549b076c8ab2@mail.gmail.com>	<5e414ed0910240513q316905ai5cf8c2ef63b52f60@mail.gmail.com>
	<ACB65510-C38F-4345-B555-BD2DCF61B444@jerris.com>
	<4AEC5C65.6050800@puzzled.xs4all.nl>
Message-ID: <188D171E-C1E9-439B-BCCB-EE5E80BD21B7@freeswitch.org>

We have <param name="rfc2833-pt" value="101"/> on the profile and you  
to set this.

So you can set it to 96 if needed.  But you shouldn't have to do that  
if they say 96 and we say 101 they should be listening on 101 from us  
and we should be listening on 96 from them... thats why its called an  
RTP map.

/b

On Oct 31, 2009, at 10:48 AM, Patrick List wrote:

>> On Oct 24, 2009, at 8:13 AM, Dennis wrote:
>>
>>> ok, as written, i come back after some tests with fs and a thomson
>>> cirpack.
>
> No idea if this is useful as I'm a noob with fs. If not please excuse
> the noise. In the past Asterisk to work properly with Cirpack needed  
> the
> following patch:
>
> diff -uNr asterisk-1.4.19.org/main/rtp.c asterisk-1.4.19/main/rtp.c
> --- asterisk-1.4.19.org/main/rtp.c	2007-10-08 22:06:33.000000000 +0200
> +++ asterisk-1.4.19/main/rtp.c	2007-11-11 13:12:28.000000000 +0100
> @@ -1383,6 +1383,7 @@
>  	[34] = {1, AST_FORMAT_H263},
>  	[103] = {1, AST_FORMAT_H263_PLUS},
>  	[97] = {1, AST_FORMAT_ILBC},
> +	[96] = {0, AST_RTP_DTMF},
>  	[99] = {1, AST_FORMAT_H264},
>  	[101] = {0, AST_RTP_DTMF},
>  	[110] = {1, AST_FORMAT_SPEEX},
>
> Maybe this helps.
>
> Regards,
> Patrick
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From rupa at rupa.com  Sat Oct 31 08:59:18 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 31 Oct 2009 10:59:18 -0500
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
Message-ID: <b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>

I believe this means you are hung up in CDR reporting.  You can get a
core of the running system and look at the stack traces to see where
exactly it is hung up.

Try using:

support-d/fscore_pb gcore yourname

This will put a copy of the stacktrace on the pastebin for poeple (or
you) to look at.

On Sat, Oct 31, 2009 at 6:08 AM, Dome Charoenyost <dome at tel.co.th> wrote:
> Dear All,
> ? ? ? ? ?I'm disable all CDR and use nibble_bill (modify) and
> odbc_query for create CDR and dedcut balance. but i found many zombie
> channel and can't kill
> Like a
> 51555b02-c5ba-11de-af77-774e15238dbe,outbound,2009-10-30
> 18:11:38,1256951498,sofia/external/66848087977 at xxx.xxx.xx.191:7050,CS_REPORTING,FreeSWITCH,66850304266,xxx.xxx.xxx.191,7050,,,XML,public,,,,,
>
> I'm using FreeSWITCH Version 1.0.5pre3 (hacked)
>
> How to fix ?
>
> Dome C.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa



From brian at freeswitch.org  Sat Oct 31 09:11:49 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 31 Oct 2009 11:11:49 -0500
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
Message-ID: <140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>

So you're doing your billing INLINE with the session using the OSBC  
Query app?  Not the most optimal way!

/b

On Oct 31, 2009, at 10:59 AM, Rupa Schomaker wrote:

> I believe this means you are hung up in CDR reporting.  You can get a
> core of the running system and look at the stack traces to see where
> exactly it is hung up.
>
> Try using:
>
> support-d/fscore_pb gcore yourname
>
> This will put a copy of the stacktrace on the pastebin for poeple (or
> you) to look at.
>
> On Sat, Oct 31, 2009 at 6:08 AM, Dome Charoenyost <dome at tel.co.th>  
> wrote:
>> Dear All,
>>          I'm disable all CDR and use nibble_bill (modify) and
>> odbc_query for create CDR and dedcut balance. but i found many zombie
>> channel and can't kill
>> Like a
>> 51555b02-c5ba-11de-af77-774e15238dbe,outbound,2009-10-30
>> 18:11:38,1256951498,sofia/external/66848087977 at xxx.xxx.xx. 
>> 191:7050,CS_REPORTING,FreeSWITCH,66850304266,xxx.xxx.xxx. 
>> 191,7050,,,XML,public,,,,,
>>
>> I'm using FreeSWITCH Version 1.0.5pre3 (hacked)
>>
>> How to fix ?
>>
>> Dome C.

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From dome at tel.co.th  Sat Oct 31 09:32:35 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Sat, 31 Oct 2009 23:32:35 +0700
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
	<140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
Message-ID: <8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>

I use odbc_query for retrive balance and get LCR from my billing DB.
and use nibble_bill

Dome C.


2009/10/31 Brian West <brian at freeswitch.org>:
> So you're doing your billing INLINE with the session using the OSBC Query
> app? ?Not the most optimal way!
> /b
> On Oct 31, 2009, at 10:59 AM, Rupa Schomaker wrote:
>
> I believe this means you are hung up in CDR reporting. ?You can get a
> core of the running system and look at the stack traces to see where
> exactly it is hung up.
>
> Try using:
>
> support-d/fscore_pb gcore yourname
>
> This will put a copy of the stacktrace on the pastebin for poeple (or
> you) to look at.
>
> On Sat, Oct 31, 2009 at 6:08 AM, Dome Charoenyost <dome at tel.co.th> wrote:
>
> Dear All,
>
> ? ? ? ? ?I'm disable all CDR and use nibble_bill (modify) and
>
> odbc_query for create CDR and dedcut balance. but i found many zombie
>
> channel and can't kill
>
> Like a
>
> 51555b02-c5ba-11de-af77-774e15238dbe,outbound,2009-10-30
>
> 18:11:38,1256951498,sofia/external/66848087977 at xxx.xxx.xx.191:7050,CS_REPORTING,FreeSWITCH,66850304266,xxx.xxx.xxx.191,7050,,,XML,public,,,,,
>
> I'm using FreeSWITCH Version 1.0.5pre3 (hacked)
>
> How to fix ?
>
> Dome C.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From brian at freeswitch.org  Sat Oct 31 09:44:39 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 31 Oct 2009 11:44:39 -0500
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
	<140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
Message-ID: <AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>

You should never do billing inline with the session thread is all I'm  
saying.

/b

On Oct 31, 2009, at 11:32 AM, Dome Charoenyost wrote:

> I use odbc_query for retrive balance and get LCR from my billing DB.
> and use nibble_bill
>
> Dome C.




From jmesquita at freeswitch.org  Sat Oct 31 09:56:17 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 31 Oct 2009 14:56:17 -0200
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
	<140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
	<AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
Message-ID: <ebbc2d550910310956x3ce32c53n44efed28a54ef36c@mail.gmail.com>

Dome, just to explain what Brian is saying:

Doing billing inline in this case means that the session thread (what you
see in show channels). If, for whatever reason, something goes wrong on the
DB connection or task for the billing, this session thread will be stuck
with it leaving it dangling around the system. This is what most likely is
happening to you. The right way to do it is to let the session thread go
(which mod_cdr_csv does) and then process the billing.

None of the methods you have described do that. You might consider
post-processing your CDR/billing information to avoid coming up with this
kind of problem that are very hard to predict.

Hope that helps,

JM

PS: Nonetheless, I still think it is valid to get a core dump like Rupa
metioned.

On Sat, Oct 31, 2009 at 2:44 PM, Brian West <brian at freeswitch.org> wrote:

> You should never do billing inline with the session thread is all I'm
> saying.
>
> /b
>
> On Oct 31, 2009, at 11:32 AM, Dome Charoenyost wrote:
>
> > I use odbc_query for retrive balance and get LCR from my billing DB.
> > and use nibble_bill
> >
> > Dome C.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dome at tel.co.th  Sat Oct 31 10:05:27 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Sun, 1 Nov 2009 00:05:27 +0700
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
	<140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
	<AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
Message-ID: <8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com>

2009/10/31 Brian West <brian at freeswitch.org>:
> You should never do billing inline with the session thread is all I'm
> saying.
Sorry I don't understand.  This is my example dialplan
       <action application="set"
data="dialoption=nibble_account=${caller_id_number},nibble_rate=0.001,ignore_early_media=false,originate_timeout=20/>
        <action application="odbc_query" data="select carrier_name As
provider_id,balance::text As lcr_balance,(balance/rate)::int as
lcr_timeout,(balance/rate*60)::int as lcr_timeout_sec  from
lcr('${caller_id_number}','$1','external');"/>
        <action application="log" data="ALERT Balance ${lcr_balance}"/>
        <action application="log" data="ALERT timeout ${lcr_timeout}"/>
        <action application="execute_extension"
data="ck_zerobalance_${lcr_balance} XML public"/>
        <action application="execute_extension"
data="tz_say_min_${lcr_timeout} XML public"/>
        <action application="sched_hangup" data="+${lcr_timeout_sec}
alloted_timeout"/>
        <action application="log" data="ALERT timeout_sec ${lcr_timeout_sec}"/>
        <action application="odbc_query" data="select
'route_'||route_id::text As name,gw_prefix||prefix||gw_suffix As value
from lcr('${caller_id_number}','$1','external') ;"/>
        <action application="bridge" data="{${dialoption}},${route_1}"/>
        <action application="bridge" data="{${dialoption}},${route_2}"/>
        <action application="bridge" data="{${dialoption}},${route_3}"/>


Dome C.

>
> /b
>
> On Oct 31, 2009, at 11:32 AM, Dome Charoenyost wrote:
>
>> I use odbc_query for retrive balance and get LCR from my billing DB.
>> and use nibble_bill
>>
>> Dome C.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From dome at tel.co.th  Sat Oct 31 10:08:26 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Sun, 1 Nov 2009 00:08:26 +0700
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <ebbc2d550910310956x3ce32c53n44efed28a54ef36c@mail.gmail.com>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
	<140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
	<AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
	<ebbc2d550910310956x3ce32c53n44efed28a54ef36c@mail.gmail.com>
Message-ID: <8ccbff060910311008k51b89440i8c210a09ba82b9ab@mail.gmail.com>

Thanks JM.
   So mod_nibblebill not good for running with high traffic. so we
can't do realtime billing ?

Dome C.


2009/10/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
> Dome, just to explain what Brian is saying:
>
> Doing billing inline in this case means that the session thread (what you
> see in show channels). If, for whatever reason, something goes wrong on the
> DB connection or task for the billing, this session thread will be stuck
> with it leaving it dangling around the system. This is what most likely is
> happening to you. The right way to do it is to let the session thread go
> (which mod_cdr_csv does) and then process the billing.
>
> None of the methods you have described do that. You might consider
> post-processing your CDR/billing information to avoid coming up with this
> kind of problem that are very hard to predict.
>
> Hope that helps,
>
> JM
>
> PS: Nonetheless, I still think it is valid to get a core dump like Rupa
> metioned.
>
> On Sat, Oct 31, 2009 at 2:44 PM, Brian West <brian at freeswitch.org> wrote:
>>
>> You should never do billing inline with the session thread is all I'm
>> saying.
>>
>> /b
>>
>> On Oct 31, 2009, at 11:32 AM, Dome Charoenyost wrote:
>>
>> > I use odbc_query for retrive balance and get LCR from my billing DB.
>> > and use nibble_bill
>> >
>> > Dome C.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From brian at freeswitch.org  Sat Oct 31 10:11:08 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 31 Oct 2009 12:11:08 -0500
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
	<140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
	<AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
	<8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com>
Message-ID: <EEDE7B6D-F722-4557-BBFE-AC0031C0BA86@freeswitch.org>

I think once you get the backtrace like rupa said we can see that  
maybe odbc_query is really hanging or something similar.

/b

On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:

> 2009/10/31 Brian West <brian at freeswitch.org>:
>> You should never do billing inline with the session thread is all I'm
>> saying.

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From jmesquita at freeswitch.org  Sat Oct 31 10:18:13 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 31 Oct 2009 15:18:13 -0200
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <EEDE7B6D-F722-4557-BBFE-AC0031C0BA86@freeswitch.org>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
	<140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
	<AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
	<8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com>
	<EEDE7B6D-F722-4557-BBFE-AC0031C0BA86@freeswitch.org>
Message-ID: <ebbc2d550910311018m7bc9b86bi120649c36dbf3413@mail.gmail.com>

No, mod_nibblebill definetely needs to be enhanced but it is not the problem
and it can be used with high load traffic.

The one I am not sure about is odbc_query since it was not developed for
that.

Do what Rupa said, please.

Regards,

JM

On Sat, Oct 31, 2009 at 3:11 PM, Brian West <brian at freeswitch.org> wrote:

> I think once you get the backtrace like rupa said we can see that maybe
> odbc_query is really hanging or something similar.
>
> /b
>
> On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:
>
> 2009/10/31 Brian West <brian at freeswitch.org>:
>
> You should never do billing inline with the session thread is all I'm
>
> saying.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dome at tel.co.th  Sat Oct 31 10:21:26 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Sun, 1 Nov 2009 00:21:26 +0700
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <EEDE7B6D-F722-4557-BBFE-AC0031C0BA86@freeswitch.org>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
	<140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
	<AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
	<8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com>
	<EEDE7B6D-F722-4557-BBFE-AC0031C0BA86@freeswitch.org>
Message-ID: <8ccbff060910311021k45ceeemf2a56c2dd65d2a6c@mail.gmail.com>

i try
support-d/fscore_pb gcore root
but only get

[New Thread 0xb7656b90 (LWP 28445)]
[New Thread 0xb77bbb90 (LWP 28444)]
0xb7a5e007 in select () from /lib/libc.so.6
Saved corefile core.28443

i got core file but no more file

Dome C.

2009/11/1 Brian West <brian at freeswitch.org>:
> I think once you get the backtrace like rupa said we can see that maybe
> odbc_query is really hanging or something similar.
> /b
> On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:
>
> 2009/10/31 Brian West <brian at freeswitch.org>:
>
> You should never do billing inline with the session thread is all I'm
>
> saying.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From dome at tel.co.th  Sat Oct 31 10:34:07 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Sun, 1 Nov 2009 00:34:07 +0700
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <ebbc2d550910311018m7bc9b86bi120649c36dbf3413@mail.gmail.com>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
	<140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
	<AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
	<8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com>
	<EEDE7B6D-F722-4557-BBFE-AC0031C0BA86@freeswitch.org>
	<ebbc2d550910311018m7bc9b86bi120649c36dbf3413@mail.gmail.com>
Message-ID: <8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com>

I found bug in fscore_pb   $pwd should be $mypwd

Now i post already. please help me check.

Dome C.


2009/11/1 Jo?o Mesquita <jmesquita at freeswitch.org>:
> No, mod_nibblebill definetely needs to be enhanced but it is not the problem
> and it can be used with high load traffic.
>
> The one I am not sure about is odbc_query since it was not developed for
> that.
>
> Do what Rupa said, please.
>
> Regards,
>
> JM
>
> On Sat, Oct 31, 2009 at 3:11 PM, Brian West <brian at freeswitch.org> wrote:
>>
>> I think once you get the backtrace like rupa said we can see that maybe
>> odbc_query is really hanging or something similar.
>> /b
>> On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:
>>
>> 2009/10/31 Brian West <brian at freeswitch.org>:
>>
>> You should never do billing inline with the session thread is all I'm
>>
>> saying.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From mike at jerris.com  Sat Oct 31 10:42:52 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 31 Oct 2009 13:42:52 -0400
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179
In-Reply-To: <87f2f3b90910221054l8abc3far9a18a66f0959f950@mail.gmail.com>
References: <mailman.1516.1256123625.2495.freeswitch-users@lists.freeswitch.org>
	<a60fed5e0910212108v4755203y693b8bdec5501f5f@mail.gmail.com>
	<87f2f3b90910212141w72b8f5a2oce032adb974d71e2@mail.gmail.com>
	<b39273530910220551q14d22150ldf8d4bff0b6fd011@mail.gmail.com>
	<87f2f3b90910221054l8abc3far9a18a66f0959f950@mail.gmail.com>
Message-ID: <509453DC-57BF-43D0-B704-FF0F2BB58EC0@jerris.com>

You actually can use these in conditions.  Just need to be careful  
that the var you are conditioning on is already set.

Mike

On Oct 22, 2009, at 1:54 PM, Michael Collins wrote:

>
>
> On Thu, Oct 22, 2009 at 5:51 AM, Rupa Schomaker <rupa at rupa.com> wrote:
> cond would be helpful here?  I updated the wiki on this one just now
> with a bit more detail.  It is a api call. so, you'd use it like:
>
> ${cond(eval ? trueval : falseval)}
>
> so to get a value of ERR if the var my myvar is > 15 you could:
>
> ${cond(${myvar} > 15 ? ERR : OK)}
>
> If both sides of the comparison operator are numeric then it does
> numeric comparison otherwise it does lexical string comparison.
>
> Rupa,
>
> Yes, you can do the set/cond API trick but you can only do it in the  
> action or anti-action tags, not in the condition tags. I'm sure you  
> know that but I want all those reading this thread to make the  
> connection.
> -MC
>

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From stevendt at primrosebank.net  Sat Oct 31 10:37:19 2009
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Sat, 31 Oct 2009 17:37:19 -0000
Subject: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question
Message-ID: <C4ADBF041AB44AE399847AF51ED740A1@bp1.ad.bp.com>

Hi,

I am new to FreeSwitch  - and VOIP too (as the following questions will indicate). 

I am trying to setup a VOIP system at home and am just getting started, I have been trawling through the Docs/Wiki and realise that I have a lot to learn !

Anyway, here's the first (of what I'm sure will be many !) questions, hopefully someone will take the type to help me out......

OK, I have already worked out that FreeSwitch seems to be much better supported under Linux than Windows (is that a fair comment ?), but while I have a long term intention to give Linux a go, it's something that is on the back burner (and has been for some time). At the moment, the current plan is to use FreeSwitch under WindowsXP and I have loaded the latest Windows Install. Everything seems to be working fine so far - well, in test mode anyway, my VOIP phones and teleco interface won't be here until next week, so at the moment, I'm just trying out the system with X-Lite.

The first question that I have is about WAPSERVER. This was installed as part of the FreeSwitch install. At the moment, I am manually starting FreeSwitch and WAPSERVER does not seem to start automatically. When I manually start it, it says that it is OFFLINE until I manually put it ONLINE. So :-

1. What does WAPSERVER actually do ?
2. How should it be started ?
3. My next step is to configure the Server PC to auto-logon to my Windows 2003 Server domain and put FreeSwitch in the Startup Group - do I need to add WAPSERVER there too ?

The next question is about the ServerIP address. Not thinking far enough ahead, the machine that I installed FreeSwitch on uses DHCP to get its IP Address. I guess that the FreeSwitch server should really have a static IP Address. If I change the PC to have a static IP Address, then 

1. Do I need to re-install FreeSwitch
2. If not, is there a or is there command to get FreeSwitch to updated itself with a new IP Address ?
3. Maybe the Server does not need to know the change - maybe if only affect the clients (Phones/SoftPhones ?)

Thanks & Best Regards
Dave
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From sicfslist at gmail.com  Sat Oct 31 11:13:13 2009
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Sat, 31 Oct 2009 13:13:13 -0500
Subject: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question
In-Reply-To: <C4ADBF041AB44AE399847AF51ED740A1@bp1.ad.bp.com>
References: <C4ADBF041AB44AE399847AF51ED740A1@bp1.ad.bp.com>
Message-ID: <4AEC7E39.5030206@gmail.com>

Dave,

You should check out the settings in $FS_DIR/conf/sofia_profiles/ and 
$FS_DIR/conf/vars.xml.  To make the changes take effect just restart FS 
or use the API commands for reloading the sofia profiles.

You'll definitely want to set a static IP address on the FS server (or 
make sure the DHCP server always assigns the same IP address to the FS 
server).  Otherwise you'll constantly be changing the clients that have 
to connect to FS.

SDR


Dave Stevenson wrote:
> Hi,
>  
> I am new to FreeSwitch  - and VOIP too (as the following questions 
> will indicate).
>  
> I am trying to setup a VOIP system at home and am just getting 
> started, I have been trawling through the Docs/Wiki and realise that I 
> have a lot to learn !
>  
> Anyway, here's the first (of what I'm sure will be many !) questions, 
> hopefully someone will take the type to help me out......
>  
> OK, I have already worked out that FreeSwitch seems to be much better 
> supported under Linux than Windows (is that a fair comment ?), but 
> while I have a long term intention to give Linux a go, it's something 
> that is on the back burner (and has been for some time). At the 
> moment, the current plan is to use FreeSwitch under WindowsXP and I 
> have loaded the latest Windows Install. Everything seems to be working 
> fine so far - well, in test mode anyway, my VOIP phones and teleco 
> interface won't be here until next week, so at the moment, I'm just 
> trying out the system with X-Lite.
>  
> The first question that I have is about WAPSERVER. This was installed 
> as part of the FreeSwitch install. At the moment, I am manually 
> starting FreeSwitch and WAPSERVER does not seem to start 
> automatically. When I manually start it, it says that it is OFFLINE 
> until I manually put it ONLINE. So :-
>  
> 1. What does WAPSERVER actually do ?
> 2. How should it be started ?
> 3. My next step is to configure the Server PC to auto-logon to my 
> Windows 2003 Server domain and put FreeSwitch in the Startup Group - 
> do I need to add WAPSERVER there too ?
>  
> The next question is about the ServerIP address. Not thinking far 
> enough ahead, the machine that I installed FreeSwitch on uses DHCP to 
> get its IP Address. I guess that the FreeSwitch server should really 
> have a static IP Address. If I change the PC to have a static IP 
> Address, then
>  
> 1. Do I need to re-install FreeSwitch
> 2. If not, is there a or is there command to get FreeSwitch to updated 
> itself with a new IP Address ?
> 3. Maybe the Server does not need to know the change - maybe if only 
> affect the clients (Phones/SoftPhones ?)
>  
> Thanks & Best Regards
> Dave
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   




From rupa at rupa.com  Sat Oct 31 11:17:52 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 31 Oct 2009 13:17:52 -0500
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
	<140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
	<AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
	<8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com>
	<EEDE7B6D-F722-4557-BBFE-AC0031C0BA86@freeswitch.org>
	<ebbc2d550910311018m7bc9b86bi120649c36dbf3413@mail.gmail.com>
	<8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com>
Message-ID: <b39273530910311117k3b4bcfa5s98a5b720b1775e20@mail.gmail.com>

fscore_pb has been updated.

Next time put in the pastebin url.  This one was 10911.

Ok, a suspicious thread looks below.  If I read it right, you are
doing an originate from within the hangup handler.  That doesn't look
normal or sane to me.

Thread 41 (process 30732):
#0  0xb7a5e007 in select () from /lib/libc.so.6
#1  0xb7dde39f in apr_sleep (t=1000) at time/unix/time.c:246
#2  0xb7db12c0 in do_sleep (t=1000) at src/switch_time.c:109
#3  0xb7db15ac in switch_cond_next () at src/switch_time.c:196
#4  0xb7d862b8 in switch_ivr_originate (session=0x0, bleg=0xb48c2f00,
cause=0xb48c2ed4, bridgeto=0xb1133340
"{origination_caller_id_number=0857151465,callback_uuid=c32f54cc-c6b8-11de-af77-774e15238dbe,ignore_early_media=true}sofia/external/66857151465 at 115.178.62.191:7050|sofia/external/66857151465 at 115.178.62"...,
timelimit_sec=60, table=0x0, cid_name_override=0xb7e8c976
"FreeSWITCH", cid_num_override=0x90ec0800 "0857151465",
caller_profile_override=0x0, ovars=0x0, flags=0) at
src/switch_ivr_originate.c:2132
#5  0xb6f0def1 in originate_function (cmd=0x90164ce2
"{origination_caller_id_number=0857151465,callback_uuid=c32f54cc-c6b8-11de-af77-774e15238dbe,ignore_early_media=true}sofia/external/66857151465 at 115.178.62.191:7050|sofia/external/66857151465 at 115.178.62"...,
session=0x0, stream=0xb48c3004) at mod_commands.c:2413
#6  0xb7d66fb1 in switch_api_execute (cmd=0x90164cd8 "originate",
arg=0x90164ce2 "{origination_caller_id_number=0857151465,callback_uuid=c32f54cc-c6b8-11de-af77-774e15238dbe,ignore_early_media=true}sofia/external/66857151465 at 115.178.62.191:7050|sofia/external/66857151465 at 115.178.62"...,
session=0x0, stream=0xb48c3004) at src/switch_loadable_module.c:1567
#7  0xb7d53c4a in switch_core_session_hangup_state
(session=0x958ab678) at src/switch_core_state_machine.c:506
#8  0xb7d385e6 in switch_channel_perform_hangup (channel=0x958afd70,
file=0xb6af8788 "mod_dptools.c", func=0xb6af8ba8 "hangup_function",
line=637, hangup_cause=SWITCH_CAUSE_NORMAL_CLEARING) at
src/switch_channel.c:1898
#9  0xb6aee906 in hangup_function (session=0x958ab678, data=0xb7e82ea5
"") at mod_dptools.c:637
#10 0xb7d4fe1f in switch_core_session_exec (session=0x958ab678,
application_interface=0xb6e10ec0, arg=0xb7e82ea5 "") at
src/switch_core_session.c:1552
#11 0xb7d4f73e in switch_core_session_execute_application
(session=0x958ab678, app=0x90163a80 "hangup", arg=0xb7e82ea5 "") at
src/switch_core_session.c:1474
#12 0xb7d51131 in switch_core_standard_on_execute (session=0x958ab678)
at src/switch_core_state_machine.c:166
#13 0xb7d5215a in switch_core_session_run (session=0x958ab678) at
src/switch_core_state_machine.c:340
#14 0xb7d4e199 in switch_core_session_thread (thread=0x9020bd98,
obj=0x958ab678) at src/switch_core_session.c:1135
#15 0xb7ddc2fc in dummy_worker (opaque=0x9020bd98) at
threadproc/unix/thread.c:138
#16 0xb7caaf3b in start_thread () from /lib/libpthread.so.0
#17 0xb7a64bee in clone () from /lib/libc.so.6


On Sat, Oct 31, 2009 at 12:34 PM, Dome Charoenyost <dome at tel.co.th> wrote:
> I found bug in fscore_pb ? $pwd should be $mypwd
>
> Now i post already. please help me check.
>
> Dome C.
>
>
> 2009/11/1 Jo?o Mesquita <jmesquita at freeswitch.org>:
>> No, mod_nibblebill definetely needs to be enhanced but it is not the problem
>> and it can be used with high load traffic.
>>
>> The one I am not sure about is odbc_query since it was not developed for
>> that.
>>
>> Do what Rupa said, please.
>>
>> Regards,
>>
>> JM
>>
>> On Sat, Oct 31, 2009 at 3:11 PM, Brian West <brian at freeswitch.org> wrote:
>>>
>>> I think once you get the backtrace like rupa said we can see that maybe
>>> odbc_query is really hanging or something similar.
>>> /b
>>> On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:
>>>
>>> 2009/10/31 Brian West <brian at freeswitch.org>:
>>>
>>> You should never do billing inline with the session thread is all I'm
>>>
>>> saying.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa



From dujinfang at gmail.com  Sat Oct 31 11:24:31 2009
From: dujinfang at gmail.com (Seven Du)
Date: Sun, 1 Nov 2009 02:24:31 +0800
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
	<140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
	<AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
	<8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com>
	<EEDE7B6D-F722-4557-BBFE-AC0031C0BA86@freeswitch.org>
	<ebbc2d550910311018m7bc9b86bi120649c36dbf3413@mail.gmail.com>
	<8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com>
Message-ID: <23f91030910311124s1f8844ddw63da23c2ca7ab8a9@mail.gmail.com>

I also got some zombie channels, if someone can help me take a look that's
really nice.

http://pastebin.freeswitch.org/10912

I only loaded mod_cdr_csv
Is it ok to use mod_xml_cdr?

Thanks.


2009/11/1 Dome Charoenyost <dome at tel.co.th>

> I found bug in fscore_pb   $pwd should be $mypwd
>
> Now i post already. please help me check.
>
> Dome C.
>
>
> 2009/11/1 Jo?o Mesquita <jmesquita at freeswitch.org>:
> > No, mod_nibblebill definetely needs to be enhanced but it is not the
> problem
> > and it can be used with high load traffic.
> >
> > The one I am not sure about is odbc_query since it was not developed for
> > that.
> >
> > Do what Rupa said, please.
> >
> > Regards,
> >
> > JM
> >
> > On Sat, Oct 31, 2009 at 3:11 PM, Brian West <brian at freeswitch.org>
> wrote:
> >>
> >> I think once you get the backtrace like rupa said we can see that maybe
> >> odbc_query is really hanging or something similar.
> >> /b
> >> On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:
> >>
> >> 2009/10/31 Brian West <brian at freeswitch.org>:
> >>
> >> You should never do billing inline with the session thread is all I'm
> >>
> >> saying.
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From stevendt at primrosebank.net  Sat Oct 31 11:37:28 2009
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Sat, 31 Oct 2009 18:37:28 -0000
Subject: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question
References: <C4ADBF041AB44AE399847AF51ED740A1@bp1.ad.bp.com>
	<4AEC7E39.5030206@gmail.com>
Message-ID: <FEF9804494E84F9CB430906C465A67F7@bp1.ad.bp.com>

Hi Shelby,

thanks for the reply on the IP Address Question.

I do not have a $FS_DIR/conf/sofia_profiles/ directory ?

$FS_DIR/conf/vars.xml has a line :-
<X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>

I'd guess that this means that the Server uses the machines IP address 
automatically, so, I can change the IP address to fixed and don't need to 
make any changes at the Server end - just need to update the Clients.

regards
Dave


----- Original Message ----- 
From: "Shelby Ramsey" <sicfslist at gmail.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, October 31, 2009 6:13 PM
Subject: Re: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question


> Dave,
>
> You should check out the settings in $FS_DIR/conf/sofia_profiles/ and
> $FS_DIR/conf/vars.xml.  To make the changes take effect just restart FS
> or use the API commands for reloading the sofia profiles.
>
> You'll definitely want to set a static IP address on the FS server (or
> make sure the DHCP server always assigns the same IP address to the FS
> server).  Otherwise you'll constantly be changing the clients that have
> to connect to FS.
>
> SDR
>
>
> Dave Stevenson wrote:
>> Hi,
>>
>> I am new to FreeSwitch  - and VOIP too (as the following questions
>> will indicate).
>>
>> I am trying to setup a VOIP system at home and am just getting
>> started, I have been trawling through the Docs/Wiki and realise that I
>> have a lot to learn !
>>
>> Anyway, here's the first (of what I'm sure will be many !) questions,
>> hopefully someone will take the type to help me out......
>>
>> OK, I have already worked out that FreeSwitch seems to be much better
>> supported under Linux than Windows (is that a fair comment ?), but
>> while I have a long term intention to give Linux a go, it's something
>> that is on the back burner (and has been for some time). At the
>> moment, the current plan is to use FreeSwitch under WindowsXP and I
>> have loaded the latest Windows Install. Everything seems to be working
>> fine so far - well, in test mode anyway, my VOIP phones and teleco
>> interface won't be here until next week, so at the moment, I'm just
>> trying out the system with X-Lite.
>>
>> The first question that I have is about WAPSERVER. This was installed
>> as part of the FreeSwitch install. At the moment, I am manually
>> starting FreeSwitch and WAPSERVER does not seem to start
>> automatically. When I manually start it, it says that it is OFFLINE
>> until I manually put it ONLINE. So :-
>>
>> 1. What does WAPSERVER actually do ?
>> 2. How should it be started ?
>> 3. My next step is to configure the Server PC to auto-logon to my
>> Windows 2003 Server domain and put FreeSwitch in the Startup Group -
>> do I need to add WAPSERVER there too ?
>>
>> The next question is about the ServerIP address. Not thinking far
>> enough ahead, the machine that I installed FreeSwitch on uses DHCP to
>> get its IP Address. I guess that the FreeSwitch server should really
>> have a static IP Address. If I change the PC to have a static IP
>> Address, then
>>
>> 1. Do I need to re-install FreeSwitch
>> 2. If not, is there a or is there command to get FreeSwitch to updated
>> itself with a new IP Address ?
>> 3. Maybe the Server does not need to know the change - maybe if only
>> affect the clients (Phones/SoftPhones ?)
>>
>> Thanks & Best Regards
>> Dave
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 




From sicfslist at gmail.com  Sat Oct 31 12:07:39 2009
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Sat, 31 Oct 2009 14:07:39 -0500
Subject: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question
In-Reply-To: <FEF9804494E84F9CB430906C465A67F7@bp1.ad.bp.com>
References: <C4ADBF041AB44AE399847AF51ED740A1@bp1.ad.bp.com>	<4AEC7E39.5030206@gmail.com>
	<FEF9804494E84F9CB430906C465A67F7@bp1.ad.bp.com>
Message-ID: <4AEC8AFB.9040306@gmail.com>

Dave,

I'm not sure how the file structure works on Windows.  But sofia is the 
SIP endpoint (and what you're wanting to confgure).

The bind_server setting only relates to mod_dingaling (I think).  Yes 
you can change the IP address ... but you'll need to restart FS or 
reload the sofia profiles.  The conf settings for external_sip_ip, etc. 
are the ones that matter. 

Good luck!  You can also hop on freenode @ #freeswitch and everyone 
there is quite helpful.

SDR




Dave Stevenson wrote:
> Hi Shelby,
>
> thanks for the reply on the IP Address Question.
>
> I do not have a $FS_DIR/conf/sofia_profiles/ directory ?
>
> $FS_DIR/conf/vars.xml has a line :-
> <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>
>
> I'd guess that this means that the Server uses the machines IP address 
> automatically, so, I can change the IP address to fixed and don't need to 
> make any changes at the Server end - just need to update the Clients.
>
> regards
> Dave
>
>
> ----- Original Message ----- 
> From: "Shelby Ramsey" <sicfslist at gmail.com>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Saturday, October 31, 2009 6:13 PM
> Subject: Re: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question
>
>
>   
>> Dave,
>>
>> You should check out the settings in $FS_DIR/conf/sofia_profiles/ and
>> $FS_DIR/conf/vars.xml.  To make the changes take effect just restart FS
>> or use the API commands for reloading the sofia profiles.
>>
>> You'll definitely want to set a static IP address on the FS server (or
>> make sure the DHCP server always assigns the same IP address to the FS
>> server).  Otherwise you'll constantly be changing the clients that have
>> to connect to FS.
>>
>> SDR
>>
>>
>> Dave Stevenson wrote:
>>     
>>> Hi,
>>>
>>> I am new to FreeSwitch  - and VOIP too (as the following questions
>>> will indicate).
>>>
>>> I am trying to setup a VOIP system at home and am just getting
>>> started, I have been trawling through the Docs/Wiki and realise that I
>>> have a lot to learn !
>>>
>>> Anyway, here's the first (of what I'm sure will be many !) questions,
>>> hopefully someone will take the type to help me out......
>>>
>>> OK, I have already worked out that FreeSwitch seems to be much better
>>> supported under Linux than Windows (is that a fair comment ?), but
>>> while I have a long term intention to give Linux a go, it's something
>>> that is on the back burner (and has been for some time). At the
>>> moment, the current plan is to use FreeSwitch under WindowsXP and I
>>> have loaded the latest Windows Install. Everything seems to be working
>>> fine so far - well, in test mode anyway, my VOIP phones and teleco
>>> interface won't be here until next week, so at the moment, I'm just
>>> trying out the system with X-Lite.
>>>
>>> The first question that I have is about WAPSERVER. This was installed
>>> as part of the FreeSwitch install. At the moment, I am manually
>>> starting FreeSwitch and WAPSERVER does not seem to start
>>> automatically. When I manually start it, it says that it is OFFLINE
>>> until I manually put it ONLINE. So :-
>>>
>>> 1. What does WAPSERVER actually do ?
>>> 2. How should it be started ?
>>> 3. My next step is to configure the Server PC to auto-logon to my
>>> Windows 2003 Server domain and put FreeSwitch in the Startup Group -
>>> do I need to add WAPSERVER there too ?
>>>
>>> The next question is about the ServerIP address. Not thinking far
>>> enough ahead, the machine that I installed FreeSwitch on uses DHCP to
>>> get its IP Address. I guess that the FreeSwitch server should really
>>> have a static IP Address. If I change the PC to have a static IP
>>> Address, then
>>>
>>> 1. Do I need to re-install FreeSwitch
>>> 2. If not, is there a or is there command to get FreeSwitch to updated
>>> itself with a new IP Address ?
>>> 3. Maybe the Server does not need to know the change - maybe if only
>>> affect the clients (Phones/SoftPhones ?)
>>>
>>> Thanks & Best Regards
>>> Dave
>>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   




From tayeb.meftah at gmail.com  Sat Oct 31 13:21:20 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sat, 31 Oct 2009 20:21:20 +0000
Subject: [Freeswitch-users] Mod_pjsip
Message-ID: <4AEC9C40.502@gmail.com>

		Hi my FreeSWITCH friends
Today we talked about FreeSWITCH libraries and we found pjsip, which is 
a lot better than sofia, it can do everything sofia can do and more, 
it's thread-safe and performs well, I would love to see a mod_pjsip.
The library is GPL but they also offer alternative licensing, don't get 
me wrong, Sofia is great but I think FreeSWITCH will benefit from a 
thread-safe SIP library.
don't say that i need some stuf for free, but we need to make fs a very
also we love freeswitch to run in a Embedded PC like the blackfin 
CPU/boards, Pjsip is very good for Embedded SIP Devicesawesome Open 
Source project
With much love,
DelphiWorld____ Information from ESET NOD32 Antivirus, version of virus 
signature database 4539 (20091024) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com




__________ Information from ESET NOD32 Antivirus, version of virus 
signature database 4539 (20091024) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com




__________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com





From rupa at rupa.com  Sat Oct 31 12:28:29 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 31 Oct 2009 14:28:29 -0500
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <23f91030910311124s1f8844ddw63da23c2ca7ab8a9@mail.gmail.com>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
	<140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
	<AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
	<8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com>
	<EEDE7B6D-F722-4557-BBFE-AC0031C0BA86@freeswitch.org>
	<ebbc2d550910311018m7bc9b86bi120649c36dbf3413@mail.gmail.com>
	<8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com>
	<23f91030910311124s1f8844ddw63da23c2ca7ab8a9@mail.gmail.com>
Message-ID: <b39273530910311228w2cc72ec1le65d6ffc79ae5b95@mail.gmail.com>

Nothing jumps out at me on this one.

mod_cdr_csv and/or mod_xml_cdr should be fine.

On Sat, Oct 31, 2009 at 1:24 PM, Seven Du <dujinfang at gmail.com> wrote:
> I also got some zombie channels, if someone can help me take a look that's
> really nice.
>
> http://pastebin.freeswitch.org/10912
>
> I only loaded mod_cdr_csv
> Is it ok to use mod_xml_cdr?
>
> Thanks.
>
>
> 2009/11/1 Dome Charoenyost <dome at tel.co.th>
>>
>> I found bug in fscore_pb ? $pwd should be $mypwd
>>
>> Now i post already. please help me check.
>>
>> Dome C.
>>
>>
>> 2009/11/1 Jo?o Mesquita <jmesquita at freeswitch.org>:
>> > No, mod_nibblebill definetely needs to be enhanced but it is not the
>> > problem
>> > and it can be used with high load traffic.
>> >
>> > The one I am not sure about is odbc_query since it was not developed for
>> > that.
>> >
>> > Do what Rupa said, please.
>> >
>> > Regards,
>> >
>> > JM
>> >
>> > On Sat, Oct 31, 2009 at 3:11 PM, Brian West <brian at freeswitch.org>
>> > wrote:
>> >>
>> >> I think once you get the backtrace like rupa said we can see that maybe
>> >> odbc_query is really hanging or something similar.
>> >> /b
>> >> On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:
>> >>
>> >> 2009/10/31 Brian West <brian at freeswitch.org>:
>> >>
>> >> You should never do billing inline with the session thread is all I'm
>> >>
>> >> saying.
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa



From stevendt at primrosebank.net  Sat Oct 31 12:32:53 2009
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Sat, 31 Oct 2009 19:32:53 -0000
Subject: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question
References: <C4ADBF041AB44AE399847AF51ED740A1@bp1.ad.bp.com>	<4AEC7E39.5030206@gmail.com><FEF9804494E84F9CB430906C465A67F7@bp1.ad.bp.com>
	<4AEC8AFB.9040306@gmail.com>
Message-ID: <7DA26A496A784BE398B9356B8D5A27B6@bp1.ad.bp.com>

Thanks Shelby,

the vars.xml file tells me that if external_sip_ip is unspecified (which I 
think it is), then the bind_server_ip is used, i.e., "auto"

Getting off topic, I stumbled on freenode today, I wasn't sure what I needed 
to do to get IRC running on Windows - yet, another "to do" !

(The link on http://wiki.freeswitch.org/wiki/Main_Page to IRC Chat 
(irc://irc.freenode.net/freeswitch) appears to be an invalid link)


regards
Dave


----- Original Message ----- 
From: "Shelby Ramsey" <sicfslist at gmail.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, October 31, 2009 7:07 PM
Subject: Re: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question


> Dave,
>
> I'm not sure how the file structure works on Windows.  But sofia is the
> SIP endpoint (and what you're wanting to confgure).
>
> The bind_server setting only relates to mod_dingaling (I think).  Yes
> you can change the IP address ... but you'll need to restart FS or
> reload the sofia profiles.  The conf settings for external_sip_ip, etc.
> are the ones that matter.
>
> Good luck!  You can also hop on freenode @ #freeswitch and everyone
> there is quite helpful.
>
> SDR
>
>
>
>
> Dave Stevenson wrote:
>> Hi Shelby,
>>
>> thanks for the reply on the IP Address Question.
>>
>> I do not have a $FS_DIR/conf/sofia_profiles/ directory ?
>>
>> $FS_DIR/conf/vars.xml has a line :-
>> <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>
>>
>> I'd guess that this means that the Server uses the machines IP address
>> automatically, so, I can change the IP address to fixed and don't need to
>> make any changes at the Server end - just need to update the Clients.
>>
>> regards
>> Dave
>>
>>
>> ----- Original Message ----- 
>> From: "Shelby Ramsey" <sicfslist at gmail.com>
>> To: <freeswitch-users at lists.freeswitch.org>
>> Sent: Saturday, October 31, 2009 6:13 PM
>> Subject: Re: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question
>>
>>
>>
>>> Dave,
>>>
>>> You should check out the settings in $FS_DIR/conf/sofia_profiles/ and
>>> $FS_DIR/conf/vars.xml.  To make the changes take effect just restart FS
>>> or use the API commands for reloading the sofia profiles.
>>>
>>> You'll definitely want to set a static IP address on the FS server (or
>>> make sure the DHCP server always assigns the same IP address to the FS
>>> server).  Otherwise you'll constantly be changing the clients that have
>>> to connect to FS.
>>>
>>> SDR
>>>
>>>
>>> Dave Stevenson wrote:
>>>
>>>> Hi,
>>>>
>>>> I am new to FreeSwitch  - and VOIP too (as the following questions
>>>> will indicate).
>>>>
>>>> I am trying to setup a VOIP system at home and am just getting
>>>> started, I have been trawling through the Docs/Wiki and realise that I
>>>> have a lot to learn !
>>>>
>>>> Anyway, here's the first (of what I'm sure will be many !) questions,
>>>> hopefully someone will take the type to help me out......
>>>>
>>>> OK, I have already worked out that FreeSwitch seems to be much better
>>>> supported under Linux than Windows (is that a fair comment ?), but
>>>> while I have a long term intention to give Linux a go, it's something
>>>> that is on the back burner (and has been for some time). At the
>>>> moment, the current plan is to use FreeSwitch under WindowsXP and I
>>>> have loaded the latest Windows Install. Everything seems to be working
>>>> fine so far - well, in test mode anyway, my VOIP phones and teleco
>>>> interface won't be here until next week, so at the moment, I'm just
>>>> trying out the system with X-Lite.
>>>>
>>>> The first question that I have is about WAPSERVER. This was installed
>>>> as part of the FreeSwitch install. At the moment, I am manually
>>>> starting FreeSwitch and WAPSERVER does not seem to start
>>>> automatically. When I manually start it, it says that it is OFFLINE
>>>> until I manually put it ONLINE. So :-
>>>>
>>>> 1. What does WAPSERVER actually do ?
>>>> 2. How should it be started ?
>>>> 3. My next step is to configure the Server PC to auto-logon to my
>>>> Windows 2003 Server domain and put FreeSwitch in the Startup Group -
>>>> do I need to add WAPSERVER there too ?
>>>>
>>>> The next question is about the ServerIP address. Not thinking far
>>>> enough ahead, the machine that I installed FreeSwitch on uses DHCP to
>>>> get its IP Address. I guess that the FreeSwitch server should really
>>>> have a static IP Address. If I change the PC to have a static IP
>>>> Address, then
>>>>
>>>> 1. Do I need to re-install FreeSwitch
>>>> 2. If not, is there a or is there command to get FreeSwitch to updated
>>>> itself with a new IP Address ?
>>>> 3. Maybe the Server does not need to know the change - maybe if only
>>>> affect the clients (Phones/SoftPhones ?)
>>>>
>>>> Thanks & Best Regards
>>>> Dave
>>>> ------------------------------------------------------------------------
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 




From mgg at giagnocavo.net  Sat Oct 31 13:00:52 2009
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Sat, 31 Oct 2009 16:00:52 -0400
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <b39273530910311228w2cc72ec1le65d6ffc79ae5b95@mail.gmail.com>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
	<140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
	<AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
	<8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com>
	<EEDE7B6D-F722-4557-BBFE-AC0031C0BA86@freeswitch.org>
	<ebbc2d550910311018m7bc9b86bi120649c36dbf3413@mail.gmail.com>
	<8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com>
	<23f91030910311124s1f8844ddw63da23c2ca7ab8a9@mail.gmail.com>
	<b39273530910311228w2cc72ec1le65d6ffc79ae5b95@mail.gmail.com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C6703243E2960@mse17be1.mse17.exchange.ms>

Which revision? Wasn't there some read-write lock issues on sessions checked in during the last week?

FWIW, we do inline billing, but the message queue timeouts are very low. Upgrading to current SVN kills us (~15025 works fine) -- some channels stay in CS_EXECUTE, some in CS_REPORTING. We know it's not our code blocking, because there'd be transaction timeouts within 30 seconds, and these are hanging around for a long time.

I suspect it's something with how we do our originates and state handlers. If we _overwrite_ the default originate handlers AND patch originate to blank out dialplan/context after duplicating the caller profile, it works fine. If we "do the right thing" and let the default handler set CS_CONSUME_MEDIA, then we get the calls hanging around forever. As I said, I suspect this is mismatch in what FS does and what our app assumes it does.

-Michael

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker
Sent: Saturday, October 31, 2009 1:28 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Many CS_REPORTING state Zombie session

Nothing jumps out at me on this one.

mod_cdr_csv and/or mod_xml_cdr should be fine.

On Sat, Oct 31, 2009 at 1:24 PM, Seven Du <dujinfang at gmail.com> wrote:
> I also got some zombie channels, if someone can help me take a look that's
> really nice.
>
> http://pastebin.freeswitch.org/10912
>
> I only loaded mod_cdr_csv
> Is it ok to use mod_xml_cdr?
>
> Thanks.
>
>
> 2009/11/1 Dome Charoenyost <dome at tel.co.th>
>>
>> I found bug in fscore_pb ? $pwd should be $mypwd
>>
>> Now i post already. please help me check.
>>
>> Dome C.
>>
>>
>> 2009/11/1 Jo?o Mesquita <jmesquita at freeswitch.org>:
>> > No, mod_nibblebill definetely needs to be enhanced but it is not the
>> > problem
>> > and it can be used with high load traffic.
>> >
>> > The one I am not sure about is odbc_query since it was not developed for
>> > that.
>> >
>> > Do what Rupa said, please.
>> >
>> > Regards,
>> >
>> > JM
>> >
>> > On Sat, Oct 31, 2009 at 3:11 PM, Brian West <brian at freeswitch.org>
>> > wrote:
>> >>
>> >> I think once you get the backtrace like rupa said we can see that maybe
>> >> odbc_query is really hanging or something similar.
>> >> /b
>> >> On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:
>> >>
>> >> 2009/10/31 Brian West <brian at freeswitch.org>:
>> >>
>> >> You should never do billing inline with the session thread is all I'm
>> >>
>> >> saying.
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From sicfslist at gmail.com  Sat Oct 31 13:09:04 2009
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Sat, 31 Oct 2009 15:09:04 -0500
Subject: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question
In-Reply-To: <7DA26A496A784BE398B9356B8D5A27B6@bp1.ad.bp.com>
References: <C4ADBF041AB44AE399847AF51ED740A1@bp1.ad.bp.com>	<4AEC7E39.5030206@gmail.com><FEF9804494E84F9CB430906C465A67F7@bp1.ad.bp.com>	<4AEC8AFB.9040306@gmail.com>
	<7DA26A496A784BE398B9356B8D5A27B6@bp1.ad.bp.com>
Message-ID: <4AEC9960.8010204@gmail.com>

Dave,

As I mentioned before ... the bind_server_ip directive is only for 
mod_dingaling.  I don't think sofia will use it.

As for IRC ... I have no knowledge of Windows machines or apps ... but I 
know there is an IRC client for FireFox called chatzilla.

SDR



Dave Stevenson wrote:
> Thanks Shelby,
>
> the vars.xml file tells me that if external_sip_ip is unspecified (which I 
> think it is), then the bind_server_ip is used, i.e., "auto"
>
> Getting off topic, I stumbled on freenode today, I wasn't sure what I needed 
> to do to get IRC running on Windows - yet, another "to do" !
>
> (The link on http://wiki.freeswitch.org/wiki/Main_Page to IRC Chat 
> (irc://irc.freenode.net/freeswitch) appears to be an invalid link)
>
>
> regards
> Dave
>
>
> ----- Original Message ----- 
> From: "Shelby Ramsey" <sicfslist at gmail.com>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Saturday, October 31, 2009 7:07 PM
> Subject: Re: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question
>
>
>   
>> Dave,
>>
>> I'm not sure how the file structure works on Windows.  But sofia is the
>> SIP endpoint (and what you're wanting to confgure).
>>
>> The bind_server setting only relates to mod_dingaling (I think).  Yes
>> you can change the IP address ... but you'll need to restart FS or
>> reload the sofia profiles.  The conf settings for external_sip_ip, etc.
>> are the ones that matter.
>>
>> Good luck!  You can also hop on freenode @ #freeswitch and everyone
>> there is quite helpful.
>>
>> SDR
>>
>>
>>
>>
>> Dave Stevenson wrote:
>>     
>>> Hi Shelby,
>>>
>>> thanks for the reply on the IP Address Question.
>>>
>>> I do not have a $FS_DIR/conf/sofia_profiles/ directory ?
>>>
>>> $FS_DIR/conf/vars.xml has a line :-
>>> <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>
>>>
>>> I'd guess that this means that the Server uses the machines IP address
>>> automatically, so, I can change the IP address to fixed and don't need to
>>> make any changes at the Server end - just need to update the Clients.
>>>
>>> regards
>>> Dave
>>>
>>>
>>> ----- Original Message ----- 
>>> From: "Shelby Ramsey" <sicfslist at gmail.com>
>>> To: <freeswitch-users at lists.freeswitch.org>
>>> Sent: Saturday, October 31, 2009 6:13 PM
>>> Subject: Re: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question
>>>
>>>
>>>
>>>       
>>>> Dave,
>>>>
>>>> You should check out the settings in $FS_DIR/conf/sofia_profiles/ and
>>>> $FS_DIR/conf/vars.xml.  To make the changes take effect just restart FS
>>>> or use the API commands for reloading the sofia profiles.
>>>>
>>>> You'll definitely want to set a static IP address on the FS server (or
>>>> make sure the DHCP server always assigns the same IP address to the FS
>>>> server).  Otherwise you'll constantly be changing the clients that have
>>>> to connect to FS.
>>>>
>>>> SDR
>>>>
>>>>
>>>> Dave Stevenson wrote:
>>>>
>>>>         
>>>>> Hi,
>>>>>
>>>>> I am new to FreeSwitch  - and VOIP too (as the following questions
>>>>> will indicate).
>>>>>
>>>>> I am trying to setup a VOIP system at home and am just getting
>>>>> started, I have been trawling through the Docs/Wiki and realise that I
>>>>> have a lot to learn !
>>>>>
>>>>> Anyway, here's the first (of what I'm sure will be many !) questions,
>>>>> hopefully someone will take the type to help me out......
>>>>>
>>>>> OK, I have already worked out that FreeSwitch seems to be much better
>>>>> supported under Linux than Windows (is that a fair comment ?), but
>>>>> while I have a long term intention to give Linux a go, it's something
>>>>> that is on the back burner (and has been for some time). At the
>>>>> moment, the current plan is to use FreeSwitch under WindowsXP and I
>>>>> have loaded the latest Windows Install. Everything seems to be working
>>>>> fine so far - well, in test mode anyway, my VOIP phones and teleco
>>>>> interface won't be here until next week, so at the moment, I'm just
>>>>> trying out the system with X-Lite.
>>>>>
>>>>> The first question that I have is about WAPSERVER. This was installed
>>>>> as part of the FreeSwitch install. At the moment, I am manually
>>>>> starting FreeSwitch and WAPSERVER does not seem to start
>>>>> automatically. When I manually start it, it says that it is OFFLINE
>>>>> until I manually put it ONLINE. So :-
>>>>>
>>>>> 1. What does WAPSERVER actually do ?
>>>>> 2. How should it be started ?
>>>>> 3. My next step is to configure the Server PC to auto-logon to my
>>>>> Windows 2003 Server domain and put FreeSwitch in the Startup Group -
>>>>> do I need to add WAPSERVER there too ?
>>>>>
>>>>> The next question is about the ServerIP address. Not thinking far
>>>>> enough ahead, the machine that I installed FreeSwitch on uses DHCP to
>>>>> get its IP Address. I guess that the FreeSwitch server should really
>>>>> have a static IP Address. If I change the PC to have a static IP
>>>>> Address, then
>>>>>
>>>>> 1. Do I need to re-install FreeSwitch
>>>>> 2. If not, is there a or is there command to get FreeSwitch to updated
>>>>> itself with a new IP Address ?
>>>>> 3. Maybe the Server does not need to know the change - maybe if only
>>>>> affect the clients (Phones/SoftPhones ?)
>>>>>
>>>>> Thanks & Best Regards
>>>>> Dave
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>           
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>         
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   




From diego.viola at gmail.com  Sat Oct 31 13:14:10 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 31 Oct 2009 20:14:10 +0000
Subject: [Freeswitch-users] Mod_pjsip
In-Reply-To: <4AEC9C40.502@gmail.com>
References: <4AEC9C40.502@gmail.com>
Message-ID: <86a32abc0910311314m21345e7fh1f3c673bee0e5667@mail.gmail.com>

This is pretty interesting, is sofia thread safe as well?

Diego

On Sat, Oct 31, 2009 at 8:21 PM, Meftah Tayeb <tayeb.meftah at gmail.com> wrote:
> ? ? ? ? ? ? ? ?Hi my FreeSWITCH friends
> Today we talked about FreeSWITCH libraries and we found pjsip, which is
> a lot better than sofia, it can do everything sofia can do and more,
> it's thread-safe and performs well, I would love to see a mod_pjsip.
> The library is GPL but they also offer alternative licensing, don't get
> me wrong, Sofia is great but I think FreeSWITCH will benefit from a
> thread-safe SIP library.
> don't say that i need some stuf for free, but we need to make fs a very
> also we love freeswitch to run in a Embedded PC like the blackfin
> CPU/boards, Pjsip is very good for Embedded SIP Devicesawesome Open
> Source project
> With much love,
> DelphiWorld____ Information from ESET NOD32 Antivirus, version of virus
> signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus
> signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From msc at freeswitch.org  Sat Oct 31 13:16:07 2009
From: msc at freeswitch.org (Michael S Collins)
Date: Sat, 31 Oct 2009 13:16:07 -0700
Subject: [Freeswitch-users] Mod_pjsip
In-Reply-To: <4AEC9C40.502@gmail.com>
References: <4AEC9C40.502@gmail.com>
Message-ID: <14A50F45-F0E7-4FEF-8C1C-66EA24CA64A7@freeswitch.org>

I can guarantee that the FS devs are well aware of pj-sip. If it was/ 
is a viable alternative then it would be considered. The fact that it  
isn't being used is a pretty good indication that it isn't suitable  
for FS at this time.

-MV

Sent from my iPhone

On Oct 31, 2009, at 1:21 PM, Meftah Tayeb <tayeb.meftah at gmail.com>  
wrote:

>        Hi my FreeSWITCH friends
> Today we talked about FreeSWITCH libraries and we found pjsip, which  
> is
> a lot better than sofia, it can do everything sofia can do and more,
> it's thread-safe and performs well, I would love to see a mod_pjsip.
> The library is GPL but they also offer alternative licensing, don't  
> get
> me wrong, Sofia is great but I think FreeSWITCH will benefit from a
> thread-safe SIP library.
> don't say that i need some stuf for free, but we need to make fs a  
> very
> also we love freeswitch to run in a Embedded PC like the blackfin
> CPU/boards, Pjsip is very good for Embedded SIP Devicesawesome Open
> Source project
> With much love,
> DelphiWorld____ Information from ESET NOD32 Antivirus, version of  
> virus
> signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus
> signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus  
> signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From stkn at freeswitch.org  Sat Oct 31 14:04:58 2009
From: stkn at freeswitch.org (Stefan Knoblich)
Date: Sat, 31 Oct 2009 22:04:58 +0100
Subject: [Freeswitch-users] Mod_pjsip
In-Reply-To: <14A50F45-F0E7-4FEF-8C1C-66EA24CA64A7@freeswitch.org>
References: <4AEC9C40.502@gmail.com>
	<14A50F45-F0E7-4FEF-8C1C-66EA24CA64A7@freeswitch.org>
Message-ID: <4AECA67A.7030708@freeswitch.org>

Michael S Collins wrote:
> I can guarantee that the FS devs are well aware of pj-sip. If it was/ 
> is a viable alternative then it would be considered. The fact that it  
> isn't being used is a pretty good indication that it isn't suitable  
> for FS at this time.
> 
> -MV
> 
> Sent from my iPhone

We already mentioned some of the reasons why it did get dropped 3 years ago
(first two points from memory, last two from old IRC logs):


[License incompatible (GPL), but i think tony tried to negotiate on alternate license terms]

Not possible to have multiple SIP profiles (due to global variables being used in the lib).

A race-condition under high load, that couldn't be resolved back then (with the help of the pjsip developers).

And the sofia module "just working" and surviving the scalability tests, so all efforts were focussed on
mod_sofia and pjsip got dropped.



stkn



From shiyanov at gmail.com  Sat Oct 31 14:09:56 2009
From: shiyanov at gmail.com (Artem Shiyanov)
Date: Sun, 1 Nov 2009 00:09:56 +0300
Subject: [Freeswitch-users] [mod_socket] Can't set channel variable SOMETIMES
Message-ID: <d211f54a0910311409n530c4049re16243aa524334a4@mail.gmail.com>

Hello!

I've got strange problem:
In my app which talks to FreeSwitch via mod_socket there is such logic:
<pre conditions: channel1 - newly created, parked channell - with help of
'uuid_create' and 'originate ... &park'; channel2 is talking with channel3>
for (channel in {array of channel1, channel2, channel3}) {
 set value for the channel variable X
}

uuid_transfer channel1 extension Y
uuid_transfer -both channel2 extension Y


In extension Y there is a logic based on the channel variable X.
The problem itself: every second launching falls, the reason - channel
variable X hasn't been set for one of channel1..3.
Sniffer shows that commands are sent (and confirmated by FreeSwitch)
consiquently, FreeSwitch sends OK for each command in time.
Temporary sollution that I found is to wait 0.5 sec after 'set' commands
have been sent and before transfer any channel to extension Y.

Maybe some already faced with similar problem?


Thanks,
Artem
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From anthony.minessale at gmail.com  Sat Oct 31 14:26:09 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 31 Oct 2009 16:26:09 -0500
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6703243E2960@mse17be1.mse17.exchange.ms>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
	<AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
	<8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com>
	<EEDE7B6D-F722-4557-BBFE-AC0031C0BA86@freeswitch.org>
	<ebbc2d550910311018m7bc9b86bi120649c36dbf3413@mail.gmail.com>
	<8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com>
	<23f91030910311124s1f8844ddw63da23c2ca7ab8a9@mail.gmail.com>
	<b39273530910311228w2cc72ec1le65d6ffc79ae5b95@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C6703243E2960@mse17be1.mse17.exchange.ms>
Message-ID: <191c3a030910311426of22a4fdyb6be7cabfb992bd6@mail.gmail.com>

it something doing a read/write lock then not clearing it or its bad build
and you have to rebuild clean.
the trace had no session threads present in it.
look in the code of any non-standard mods you are using for something that
did either switch_ivr_originate or switch_core_session_locate and never
called switch_core_session_rwunlock

#define SWITCH_DEBUG_RWLOCKS at the top of switch.h
and make clean ; make install to get rwlock debugging.


On Sat, Oct 31, 2009 at 3:00 PM, Michael Giagnocavo <mgg at giagnocavo.net>wrote:

> Which revision? Wasn't there some read-write lock issues on sessions
> checked in during the last week?
>
> FWIW, we do inline billing, but the message queue timeouts are very low.
> Upgrading to current SVN kills us (~15025 works fine) -- some channels stay
> in CS_EXECUTE, some in CS_REPORTING. We know it's not our code blocking,
> because there'd be transaction timeouts within 30 seconds, and these are
> hanging around for a long time.
>
> I suspect it's something with how we do our originates and state handlers.
> If we _overwrite_ the default originate handlers AND patch originate to
> blank out dialplan/context after duplicating the caller profile, it works
> fine. If we "do the right thing" and let the default handler set
> CS_CONSUME_MEDIA, then we get the calls hanging around forever. As I said, I
> suspect this is mismatch in what FS does and what our app assumes it does.
>
> -Michael
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker
> Sent: Saturday, October 31, 2009 1:28 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Many CS_REPORTING state Zombie session
>
> Nothing jumps out at me on this one.
>
> mod_cdr_csv and/or mod_xml_cdr should be fine.
>
> On Sat, Oct 31, 2009 at 1:24 PM, Seven Du <dujinfang at gmail.com> wrote:
> > I also got some zombie channels, if someone can help me take a look
> that's
> > really nice.
> >
> > http://pastebin.freeswitch.org/10912
> >
> > I only loaded mod_cdr_csv
> > Is it ok to use mod_xml_cdr?
> >
> > Thanks.
> >
> >
> > 2009/11/1 Dome Charoenyost <dome at tel.co.th>
> >>
> >> I found bug in fscore_pb   $pwd should be $mypwd
> >>
> >> Now i post already. please help me check.
> >>
> >> Dome C.
> >>
> >>
> >> 2009/11/1 Jo?o Mesquita <jmesquita at freeswitch.org>:
> >> > No, mod_nibblebill definetely needs to be enhanced but it is not the
> >> > problem
> >> > and it can be used with high load traffic.
> >> >
> >> > The one I am not sure about is odbc_query since it was not developed
> for
> >> > that.
> >> >
> >> > Do what Rupa said, please.
> >> >
> >> > Regards,
> >> >
> >> > JM
> >> >
> >> > On Sat, Oct 31, 2009 at 3:11 PM, Brian West <brian at freeswitch.org>
> >> > wrote:
> >> >>
> >> >> I think once you get the backtrace like rupa said we can see that
> maybe
> >> >> odbc_query is really hanging or something similar.
> >> >> /b
> >> >> On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:
> >> >>
> >> >> 2009/10/31 Brian West <brian at freeswitch.org>:
> >> >>
> >> >> You should never do billing inline with the session thread is all I'm
> >> >>
> >> >> saying.
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >>
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Sat Oct 31 14:31:55 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 31 Oct 2009 16:31:55 -0500
Subject: [Freeswitch-users] [mod_socket] Can't set channel variable
	SOMETIMES
In-Reply-To: <d211f54a0910311409n530c4049re16243aa524334a4@mail.gmail.com>
References: <d211f54a0910311409n530c4049re16243aa524334a4@mail.gmail.com>
Message-ID: <191c3a030910311431r7f73c725g3f0834243e6c4a0f@mail.gmail.com>

wait for execute_complete events to be sure you application execute request
was received and processed or use uuid_setvar instead which is instant.



On Sat, Oct 31, 2009 at 4:09 PM, Artem Shiyanov <shiyanov at gmail.com> wrote:

> Hello!
>
> I've got strange problem:
> In my app which talks to FreeSwitch via mod_socket there is such logic:
> <pre conditions: channel1 - newly created, parked channell - with help of
> 'uuid_create' and 'originate ... &park'; channel2 is talking with channel3>
> for (channel in {array of channel1, channel2, channel3}) {
>  set value for the channel variable X
> }
>
> uuid_transfer channel1 extension Y
> uuid_transfer -both channel2 extension Y
>
>
> In extension Y there is a logic based on the channel variable X.
> The problem itself: every second launching falls, the reason - channel
> variable X hasn't been set for one of channel1..3.
> Sniffer shows that commands are sent (and confirmated by FreeSwitch)
> consiquently, FreeSwitch sends OK for each command in time.
> Temporary sollution that I found is to wait 0.5 sec after 'set' commands
> have been sent and before transfer any channel to extension Y.
>
> Maybe some already faced with similar problem?
>
>
> Thanks,
> Artem
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From tayeb.meftah at gmail.com  Sat Oct 31 15:44:28 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sat, 31 Oct 2009 22:44:28 +0000
Subject: [Freeswitch-users] Mod_pjsip
In-Reply-To: <4AECA67A.7030708@freeswitch.org>
References: <4AEC9C40.502@gmail.com>	<14A50F45-F0E7-4FEF-8C1C-66EA24CA64A7@freeswitch.org>
	<4AECA67A.7030708@freeswitch.org>
Message-ID: <4AECBDCC.9000105@gmail.com>

hi,
Pjsip support ICE, STUN and TURN!
to STKN:
if you don't pjsip, please stop talking or exit the discution
we want to kype Freeswitch Clean and universal
Stefan Knoblich a ?crit :
> Michael S Collins wrote:
>   
>> I can guarantee that the FS devs are well aware of pj-sip. If it was/ 
>> is a viable alternative then it would be considered. The fact that it  
>> isn't being used is a pretty good indication that it isn't suitable  
>> for FS at this time.
>>
>> -MV
>>
>> Sent from my iPhone
>>     
>
> We already mentioned some of the reasons why it did get dropped 3 years ago
> (first two points from memory, last two from old IRC logs):
>
>
> [License incompatible (GPL), but i think tony tried to negotiate on alternate license terms]
>
> Not possible to have multiple SIP profiles (due to global variables being used in the lib).
>
> A race-condition under high load, that couldn't be resolved back then (with the help of the pjsip developers).
>
> And the sofia module "just working" and surviving the scalability tests, so all efforts were focussed on
> mod_sofia and pjsip got dropped.
>
>
>
> stkn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
>   





__________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From stkn at freeswitch.org  Sat Oct 31 14:53:48 2009
From: stkn at freeswitch.org (Stefan Knoblich)
Date: Sat, 31 Oct 2009 22:53:48 +0100
Subject: [Freeswitch-users] Mod_pjsip
In-Reply-To: <4AECBDCC.9000105@gmail.com>
References: <4AEC9C40.502@gmail.com>	<14A50F45-F0E7-4FEF-8C1C-66EA24CA64A7@freeswitch.org>	<4AECA67A.7030708@freeswitch.org>
	<4AECBDCC.9000105@gmail.com>
Message-ID: <4AECB1EC.6050301@freeswitch.org>

Meftah Tayeb wrote:
> hi,
> Pjsip support ICE, STUN and TURN!
> to STKN:
> if you don't pjsip, please stop talking or exit the discution
> we want to kype Freeswitch Clean and universal

Huh?

In which part did i tell people to stop doing whatever they want with pjsip?

In my last email or on IRC?

I'm merely pointing out the reasons why mod_pjsip was abandoned by the core team 3 years ago
(so some of the points may have been fixed in the meantime).

If somebody feels the need to resurrect it, well, please do.
But don't expect Tony or anyone else from the core team to do that for you.

This is OpenSource after all...


stkn



From anthony.minessale at gmail.com  Sat Oct 31 14:55:04 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 31 Oct 2009 16:55:04 -0500
Subject: [Freeswitch-users] Mod_pjsip
In-Reply-To: <4AECBDCC.9000105@gmail.com>
References: <4AEC9C40.502@gmail.com>
	<14A50F45-F0E7-4FEF-8C1C-66EA24CA64A7@freeswitch.org>
	<4AECA67A.7030708@freeswitch.org> <4AECBDCC.9000105@gmail.com>
Message-ID: <191c3a030910311455j2408a0a2r6caa49375ed10b78@mail.gmail.com>

Meftah,

He is 100% correct.  Please do not insult my volunteer developers. Without
help from him you would not have any FreeSWITCH right now so please drop
this subject we are not using pjsip.



On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb <tayeb.meftah at gmail.com>wrote:

>  hi,
> Pjsip support ICE, STUN and TURN!
> to STKN:
> if you don't pjsip, please stop talking or exit the discution
> we want to kype Freeswitch Clean and universal
> Stefan Knoblich a ?crit :
>
> Michael S Collins wrote:
>
>
>  I can guarantee that the FS devs are well aware of pj-sip. If it was/
> is a viable alternative then it would be considered. The fact that it
> isn't being used is a pretty good indication that it isn't suitable
> for FS at this time.
>
> -MV
>
> Sent from my iPhone
>
>
>
> We already mentioned some of the reasons why it did get dropped 3 years ago
> (first two points from memory, last two from old IRC logs):
>
>
> [License incompatible (GPL), but i think tony tried to negotiate on alternate license terms]
>
> Not possible to have multiple SIP profiles (due to global variables being used in the lib).
>
> A race-condition under high load, that couldn't be resolved back then (with the help of the pjsip developers).
>
> And the sofia module "just working" and surviving the scalability tests, so all efforts were focussed on
> mod_sofia and pjsip got dropped.
>
>
>
> stkn
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
> http://www.eset.com
>
>
>
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus
> signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From tayeb.meftah at gmail.com  Sat Oct 31 16:06:45 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sat, 31 Oct 2009 23:06:45 +0000
Subject: [Freeswitch-users] Mod_pjsip
In-Reply-To: <191c3a030910311455j2408a0a2r6caa49375ed10b78@mail.gmail.com>
References: <4AEC9C40.502@gmail.com>	<14A50F45-F0E7-4FEF-8C1C-66EA24CA64A7@freeswitch.org>	<4AECA67A.7030708@freeswitch.org>
	<4AECBDCC.9000105@gmail.com>
	<191c3a030910311455j2408a0a2r6caa49375ed10b78@mail.gmail.com>
Message-ID: <4AECC305.5000805@gmail.com>

hi Anthony
i agry
i say that because STKN hate all my suggestions.
about pjsip, i will contribute aditional module in the contrib.
thanks Anthony
Anthony Minessale a ?crit :
> Meftah,
>
> He is 100% correct.  Please do not insult my volunteer developers. 
> Without help from him you would not have any FreeSWITCH right now so 
> please drop this subject we are not using pjsip.
>
>
>
> On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb <tayeb.meftah at gmail.com 
> <mailto:tayeb.meftah at gmail.com>> wrote:
>
>     hi,
>     Pjsip support ICE, STUN and TURN!
>     to STKN:
>     if you don't pjsip, please stop talking or exit the discution
>     we want to kype Freeswitch Clean and universal
>     Stefan Knoblich a ?crit :
>>     Michael S Collins wrote:
>>       
>>>     I can guarantee that the FS devs are well aware of pj-sip. If it was/ 
>>>     is a viable alternative then it would be considered. The fact that it  
>>>     isn't being used is a pretty good indication that it isn't suitable  
>>>     for FS at this time.
>>>
>>>     -MV
>>>
>>>     Sent from my iPhone
>>>         
>>     We already mentioned some of the reasons why it did get dropped 3
>>     years ago (first two points from memory, last two from old IRC
>>     logs): [License incompatible (GPL), but i think tony tried to
>>     negotiate on alternate license terms] Not possible to have
>>     multiple SIP profiles (due to global variables being used in the
>>     lib). A race-condition under high load, that couldn't be resolved
>>     back then (with the help of the pjsip developers). And the sofia
>>     module "just working" and surviving the scalability tests, so all
>>     efforts were focussed on mod_sofia and pjsip got dropped. stkn
>>     _______________________________________________ FreeSWITCH-users
>>     mailing list FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>
>>
>>
>>     __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________
>>
>>     The message was checked by ESET NOD32 Antivirus.
>>
>>     http://www.eset.com
>>
>>
>>       
>
>
>
>
>     __________ Information from ESET NOD32 Antivirus, version of virus
>     signature database 4539 (20091024) __________
>
>     The message was checked by ESET NOD32 Antivirus.
>
>     http://www.eset.com
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>   




__________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From anthony.minessale at gmail.com  Sat Oct 31 15:17:19 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 31 Oct 2009 17:17:19 -0500
Subject: [Freeswitch-users] Mod_pjsip
In-Reply-To: <4AECC305.5000805@gmail.com>
References: <4AEC9C40.502@gmail.com>
	<14A50F45-F0E7-4FEF-8C1C-66EA24CA64A7@freeswitch.org>
	<4AECA67A.7030708@freeswitch.org> <4AECBDCC.9000105@gmail.com>
	<191c3a030910311455j2408a0a2r6caa49375ed10b78@mail.gmail.com>
	<4AECC305.5000805@gmail.com>
Message-ID: <191c3a030910311517v46a9a830xa59350f6f8e2f025@mail.gmail.com>

Meftah,
Feel free.

P.S.

STKN was the guy who made the first mod_pjsip for FS that we abandoned years
ago. So you should believe him.
Both him and I agreed it was not working out.  So if you don't believe me,
find out for yourself.




On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb <tayeb.meftah at gmail.com>wrote:

>  hi Anthony
> i agry
> i say that because STKN hate all my suggestions.
> about pjsip, i will contribute aditional module in the contrib.
> thanks Anthony
> Anthony Minessale a ?crit :
>
> Meftah,
>
> He is 100% correct.  Please do not insult my volunteer developers. Without
> help from him you would not have any FreeSWITCH right now so please drop
> this subject we are not using pjsip.
>
>
>
> On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb <tayeb.meftah at gmail.com>wrote:
>
> hi,
> Pjsip support ICE, STUN and TURN!
> to STKN:
> if you don't pjsip, please stop talking or exit the discution
> we want to kype Freeswitch Clean and universal
> Stefan Knoblich a ?crit :
>
>  Michael S Collins wrote:
>
>
>  I can guarantee that the FS devs are well aware of pj-sip. If it was/
> is a viable alternative then it would be considered. The fact that it
> isn't being used is a pretty good indication that it isn't suitable
> for FS at this time.
>
> -MV
>
> Sent from my iPhone
>
>
>
> We already mentioned some of the reasons why it did get dropped 3 years ago
> (first two points from memory, last two from old IRC logs):
>
>
> [License incompatible (GPL), but i think tony tried to negotiate on alternate license terms]
>
> Not possible to have multiple SIP profiles (due to global variables being used in the lib).
>
> A race-condition under high load, that couldn't be resolved back then (with the help of the pjsip developers).
>
> And the sofia module "just working" and surviving the scalability tests, so all efforts were focussed on
> mod_sofia and pjsip got dropped.
>
>
>
> stkn
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
> http://www.eset.com
>
>
>
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus
> signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
> http://www.eset.com
>
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus
> signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From tayeb.meftah at gmail.com  Sat Oct 31 16:27:27 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sat, 31 Oct 2009 23:27:27 +0000
Subject: [Freeswitch-users] Mod_pjsip
In-Reply-To: <191c3a030910311455j2408a0a2r6caa49375ed10b78@mail.gmail.com>
References: <4AEC9C40.502@gmail.com>	<14A50F45-F0E7-4FEF-8C1C-66EA24CA64A7@freeswitch.org>	<4AECA67A.7030708@freeswitch.org>
	<4AECBDCC.9000105@gmail.com>
	<191c3a030910311455j2408a0a2r6caa49375ed10b78@mail.gmail.com>
Message-ID: <4AECC7DF.9060209@gmail.com>

hello anthony
we are realy sory about this subject
"Ok, I apologize, I really love you and respect you, and I believe you"
i will drop it from now.
Anthony Minessale a ?crit :
> Meftah,
>
> He is 100% correct.  Please do not insult my volunteer developers. 
> Without help from him you would not have any FreeSWITCH right now so 
> please drop this subject we are not using pjsip.
>
>
>
> On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb <tayeb.meftah at gmail.com 
> <mailto:tayeb.meftah at gmail.com>> wrote:
>
>     hi,
>     Pjsip support ICE, STUN and TURN!
>     to STKN:
>     if you don't pjsip, please stop talking or exit the discution
>     we want to kype Freeswitch Clean and universal
>     Stefan Knoblich a ?crit :
>>     Michael S Collins wrote:
>>       
>>>     I can guarantee that the FS devs are well aware of pj-sip. If it was/ 
>>>     is a viable alternative then it would be considered. The fact that it  
>>>     isn't being used is a pretty good indication that it isn't suitable  
>>>     for FS at this time.
>>>
>>>     -MV
>>>
>>>     Sent from my iPhone
>>>         
>>     We already mentioned some of the reasons why it did get dropped 3
>>     years ago (first two points from memory, last two from old IRC
>>     logs): [License incompatible (GPL), but i think tony tried to
>>     negotiate on alternate license terms] Not possible to have
>>     multiple SIP profiles (due to global variables being used in the
>>     lib). A race-condition under high load, that couldn't be resolved
>>     back then (with the help of the pjsip developers). And the sofia
>>     module "just working" and surviving the scalability tests, so all
>>     efforts were focussed on mod_sofia and pjsip got dropped. stkn
>>     _______________________________________________ FreeSWITCH-users
>>     mailing list FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>
>>
>>
>>     __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________
>>
>>     The message was checked by ESET NOD32 Antivirus.
>>
>>     http://www.eset.com
>>
>>
>>       
>
>
>
>
>     __________ Information from ESET NOD32 Antivirus, version of virus
>     signature database 4539 (20091024) __________
>
>     The message was checked by ESET NOD32 Antivirus.
>
>     http://www.eset.com
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>   




__________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From tayeb.meftah at gmail.com  Sat Oct 31 16:32:03 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sat, 31 Oct 2009 23:32:03 +0000
Subject: [Freeswitch-users] Mod_pjsip
In-Reply-To: <191c3a030910311517v46a9a830xa59350f6f8e2f025@mail.gmail.com>
References: <4AEC9C40.502@gmail.com>	<14A50F45-F0E7-4FEF-8C1C-66EA24CA64A7@freeswitch.org>	<4AECA67A.7030708@freeswitch.org>
	<4AECBDCC.9000105@gmail.com>	<191c3a030910311455j2408a0a2r6caa49375ed10b78@mail.gmail.com>	<4AECC305.5000805@gmail.com>
	<191c3a030910311517v46a9a830xa59350f6f8e2f025@mail.gmail.com>
Message-ID: <4AECC8F3.7090208@gmail.com>

Anthony Minessale a ?crit :
> Meftah,
> Feel free.

> thanks

> P.S.
>
> STKN was the guy who made the first mod_pjsip for FS that we abandoned 
> years ago. So you should believe him.
> Both him and I agreed it was not working out.  So if you don't believe 
> me, find out for yourself.
> anthony, why i don't believe  you?
never say that.
i believe you and all Freeswitch Staf and thank you and to all 
Freeswitch Staf.
>
>
>
> On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb <tayeb.meftah at gmail.com 
> <mailto:tayeb.meftah at gmail.com>> wrote:
>
>     hi Anthony
>     i agry
>     i say that because STKN hate all my suggestions.
>     about pjsip, i will contribute aditional module in the contrib.
>     thanks Anthony
>     Anthony Minessale a ?crit :
>>     Meftah,
>>
>>     He is 100% correct.  Please do not insult my volunteer
>>     developers. Without help from him you would not have any
>>     FreeSWITCH right now so please drop this subject we are not using
>>     pjsip.
>>
>>
>>
>>     On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb
>>     <tayeb.meftah at gmail.com <mailto:tayeb.meftah at gmail.com>> wrote:
>>
>>         hi,
>>         Pjsip support ICE, STUN and TURN!
>>         to STKN:
>>         if you don't pjsip, please stop talking or exit the discution
>>         we want to kype Freeswitch Clean and universal
>>         Stefan Knoblich a ?crit :
>>>         Michael S Collins wrote:
>>>           
>>>>         I can guarantee that the FS devs are well aware of pj-sip. If it was/ 
>>>>         is a viable alternative then it would be considered. The fact that it  
>>>>         isn't being used is a pretty good indication that it isn't suitable  
>>>>         for FS at this time.
>>>>
>>>>         -MV
>>>>
>>>>         Sent from my iPhone
>>>>             
>>>         We already mentioned some of the reasons why it did get
>>>         dropped 3 years ago (first two points from memory, last two
>>>         from old IRC logs): [License incompatible (GPL), but i think
>>>         tony tried to negotiate on alternate license terms] Not
>>>         possible to have multiple SIP profiles (due to global
>>>         variables being used in the lib). A race-condition under
>>>         high load, that couldn't be resolved back then (with the
>>>         help of the pjsip developers). And the sofia module "just
>>>         working" and surviving the scalability tests, so all efforts
>>>         were focussed on mod_sofia and pjsip got dropped. stkn
>>>         _______________________________________________
>>>         FreeSWITCH-users mailing list
>>>         FreeSWITCH-users at lists.freeswitch.org
>>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>         http://www.freeswitch.org
>>>
>>>
>>>
>>>         __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________
>>>
>>>         The message was checked by ESET NOD32 Antivirus.
>>>
>>>         http://www.eset.com
>>>
>>>
>>>           
>>
>>
>>
>>
>>         __________ Information from ESET NOD32 Antivirus, version of
>>         virus signature database 4539 (20091024) __________
>>
>>         The message was checked by ESET NOD32 Antivirus.
>>
>>         http://www.eset.com
>>
>>         _______________________________________________
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>
>>     -- 
>>     Anthony Minessale II
>>
>>     FreeSWITCH http://www.freeswitch.org/
>>     ClueCon http://www.cluecon.com/
>>     Twitter: http://twitter.com/FreeSWITCH_wire
>>
>>     AIM: anthm
>>     MSN:anthony_minessale at hotmail.com
>>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>
>>     FreeSWITCH Developer Conference
>>     sip:888 at conference.freeswitch.org
>>     <mailto:sip%3A888 at conference.freeswitch.org>
>>     iax:guest at conference.freeswitch.org/888
>>     <http://iax:guest at conference.freeswitch.org/888>
>>     googletalk:conf+888 at conference.freeswitch.org
>>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>     pstn:213-799-1400
>>      
>>     _______________________________________________ FreeSWITCH-users
>>     mailing list FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org __________ Information from ESET NOD32
>>     Antivirus, version of virus signature database 4539 (20091024)
>>     __________ The message was checked by ESET NOD32 Antivirus.
>>     http://www.eset.com
>
>
>
>     __________ Information from ESET NOD32 Antivirus, version of virus
>     signature database 4539 (20091024) __________
>
>     The message was checked by ESET NOD32 Antivirus.
>
>     http://www.eset.com
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>   




__________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From dome at tel.co.th  Sat Oct 31 19:02:07 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Sun, 1 Nov 2009 09:02:07 +0700
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <b39273530910311117k3b4bcfa5s98a5b720b1775e20@mail.gmail.com>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<b39273530910310859l148b92f3g9e561418c8e1d0a8@mail.gmail.com>
	<140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
	<AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
	<8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com>
	<EEDE7B6D-F722-4557-BBFE-AC0031C0BA86@freeswitch.org>
	<ebbc2d550910311018m7bc9b86bi120649c36dbf3413@mail.gmail.com>
	<8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com>
	<b39273530910311117k3b4bcfa5s98a5b720b1775e20@mail.gmail.com>
Message-ID: <8ccbff060910311902u7434bc6m76caba40178ecc1e@mail.gmail.com>

Yes. i user api_hangup_hook for do callback.
<action application="set" data="api_hangup_hook=originate
{origination_caller_id_number=${caller_id_number},callback_uuid=${uuid},ignore_early_media=true}${route_1}|${route_2}|${route_3}
call_ivr${caller_id_number} XML public "/>

may be need originate_timeout

Dome C.


2009/11/1 Rupa Schomaker <rupa at rupa.com>:
> fscore_pb has been updated.
>
> Next time put in the pastebin url. ?This one was 10911.
>
> Ok, a suspicious thread looks below. ?If I read it right, you are
> doing an originate from within the hangup handler. ?That doesn't look
> normal or sane to me.
>
> Thread 41 (process 30732):
> #0 ?0xb7a5e007 in select () from /lib/libc.so.6
> #1 ?0xb7dde39f in apr_sleep (t=1000) at time/unix/time.c:246
> #2 ?0xb7db12c0 in do_sleep (t=1000) at src/switch_time.c:109
> #3 ?0xb7db15ac in switch_cond_next () at src/switch_time.c:196
> #4 ?0xb7d862b8 in switch_ivr_originate (session=0x0, bleg=0xb48c2f00,
> cause=0xb48c2ed4, bridgeto=0xb1133340
> "{origination_caller_id_number=0857151465,callback_uuid=c32f54cc-c6b8-11de-af77-774e15238dbe,ignore_early_media=true}sofia/external/66857151465 at 115.178.62.191:7050|sofia/external/66857151465 at 115.178.62"...,
> timelimit_sec=60, table=0x0, cid_name_override=0xb7e8c976
> "FreeSWITCH", cid_num_override=0x90ec0800 "0857151465",
> caller_profile_override=0x0, ovars=0x0, flags=0) at
> src/switch_ivr_originate.c:2132
> #5 ?0xb6f0def1 in originate_function (cmd=0x90164ce2
> "{origination_caller_id_number=0857151465,callback_uuid=c32f54cc-c6b8-11de-af77-774e15238dbe,ignore_early_media=true}sofia/external/66857151465 at 115.178.62.191:7050|sofia/external/66857151465 at 115.178.62"...,
> session=0x0, stream=0xb48c3004) at mod_commands.c:2413
> #6 ?0xb7d66fb1 in switch_api_execute (cmd=0x90164cd8 "originate",
> arg=0x90164ce2 "{origination_caller_id_number=0857151465,callback_uuid=c32f54cc-c6b8-11de-af77-774e15238dbe,ignore_early_media=true}sofia/external/66857151465 at 115.178.62.191:7050|sofia/external/66857151465 at 115.178.62"...,
> session=0x0, stream=0xb48c3004) at src/switch_loadable_module.c:1567
> #7 ?0xb7d53c4a in switch_core_session_hangup_state
> (session=0x958ab678) at src/switch_core_state_machine.c:506
> #8 ?0xb7d385e6 in switch_channel_perform_hangup (channel=0x958afd70,
> file=0xb6af8788 "mod_dptools.c", func=0xb6af8ba8 "hangup_function",
> line=637, hangup_cause=SWITCH_CAUSE_NORMAL_CLEARING) at
> src/switch_channel.c:1898
> #9 ?0xb6aee906 in hangup_function (session=0x958ab678, data=0xb7e82ea5
> "") at mod_dptools.c:637
> #10 0xb7d4fe1f in switch_core_session_exec (session=0x958ab678,
> application_interface=0xb6e10ec0, arg=0xb7e82ea5 "") at
> src/switch_core_session.c:1552
> #11 0xb7d4f73e in switch_core_session_execute_application
> (session=0x958ab678, app=0x90163a80 "hangup", arg=0xb7e82ea5 "") at
> src/switch_core_session.c:1474
> #12 0xb7d51131 in switch_core_standard_on_execute (session=0x958ab678)
> at src/switch_core_state_machine.c:166
> #13 0xb7d5215a in switch_core_session_run (session=0x958ab678) at
> src/switch_core_state_machine.c:340
> #14 0xb7d4e199 in switch_core_session_thread (thread=0x9020bd98,
> obj=0x958ab678) at src/switch_core_session.c:1135
> #15 0xb7ddc2fc in dummy_worker (opaque=0x9020bd98) at
> threadproc/unix/thread.c:138
> #16 0xb7caaf3b in start_thread () from /lib/libpthread.so.0
> #17 0xb7a64bee in clone () from /lib/libc.so.6
>
>
> On Sat, Oct 31, 2009 at 12:34 PM, Dome Charoenyost <dome at tel.co.th> wrote:
>> I found bug in fscore_pb ? $pwd should be $mypwd
>>
>> Now i post already. please help me check.
>>
>> Dome C.
>>
>>
>> 2009/11/1 Jo?o Mesquita <jmesquita at freeswitch.org>:
>>> No, mod_nibblebill definetely needs to be enhanced but it is not the problem
>>> and it can be used with high load traffic.
>>>
>>> The one I am not sure about is odbc_query since it was not developed for
>>> that.
>>>
>>> Do what Rupa said, please.
>>>
>>> Regards,
>>>
>>> JM
>>>
>>> On Sat, Oct 31, 2009 at 3:11 PM, Brian West <brian at freeswitch.org> wrote:
>>>>
>>>> I think once you get the backtrace like rupa said we can see that maybe
>>>> odbc_query is really hanging or something similar.
>>>> /b
>>>> On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:
>>>>
>>>> 2009/10/31 Brian West <brian at freeswitch.org>:
>>>>
>>>> You should never do billing inline with the session thread is all I'm
>>>>
>>>> saying.
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From anthony.minessale at gmail.com  Sat Oct 31 19:23:23 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 31 Oct 2009 21:23:23 -0500
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <8ccbff060910311902u7434bc6m76caba40178ecc1e@mail.gmail.com>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<140B81A7-8C10-41AA-85FE-95D862557B8F@freeswitch.org>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
	<AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
	<8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com>
	<EEDE7B6D-F722-4557-BBFE-AC0031C0BA86@freeswitch.org>
	<ebbc2d550910311018m7bc9b86bi120649c36dbf3413@mail.gmail.com>
	<8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com>
	<b39273530910311117k3b4bcfa5s98a5b720b1775e20@mail.gmail.com>
	<8ccbff060910311902u7434bc6m76caba40178ecc1e@mail.gmail.com>
Message-ID: <191c3a030910311923t5f81019fyc4e7487cd61502b8@mail.gmail.com>

Use bgapi originate ...

On Oct 31, 2009 9:09 PM, "Dome Charoenyost" <dome at tel.co.th> wrote:

Yes. i user api_hangup_hook for do callback.
<action application="set" data="api_hangup_hook=originate
{origination_caller_id_number=${caller_id_number},callback_uuid=${uuid},ignore_early_media=true}${route_1}|${route_2}|${route_3}
call_ivr${caller_id_number} XML public "/>

may be need originate_timeout

Dome C.


2009/11/1 Rupa Schomaker <rupa at rupa.com>:

> fscore_pb has been updated. > > Next time put in the pastebin url.  This
one was 10911. > > Ok, a ...
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From chris at cloudtel.com  Sat Oct 31 19:24:47 2009
From: chris at cloudtel.com (Chris Burns)
Date: Sat, 31 Oct 2009 22:24:47 -0400
Subject: [Freeswitch-users] Mod_pjsip
In-Reply-To: <4AECC8F3.7090208@gmail.com>
References: <4AEC9C40.502@gmail.com>
	<191c3a030910311517v46a9a830xa59350f6f8e2f025@mail.gmail.com>
	<4AECC8F3.7090208@gmail.com>
Message-ID: <200910312224.47236.chris@cloudtel.com>

My favorite part of this 'civilized' discussion on IRC was when DelphiWord and 
diegoviola sat around tryin to take the piss outta stkn on this issue for 
seemingly no reason. Thanks for making the channel a cool place, guys ;)

On October 31, 2009 07:32:03 pm Meftah Tayeb wrote:
> Anthony Minessale a ?crit :
> > Meftah,
> > Feel free.
> >
> > thanks
> >
> > P.S.
> >
> > STKN was the guy who made the first mod_pjsip for FS that we abandoned
> > years ago. So you should believe him.
> > Both him and I agreed it was not working out.  So if you don't believe
> > me, find out for yourself.
> > anthony, why i don't believe  you?
>
> never say that.
> i believe you and all Freeswitch Staf and thank you and to all
> Freeswitch Staf.
>
> > On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb <tayeb.meftah at gmail.com
> > <mailto:tayeb.meftah at gmail.com>> wrote:
> >
> >     hi Anthony
> >     i agry
> >     i say that because STKN hate all my suggestions.
> >     about pjsip, i will contribute aditional module in the contrib.
> >     thanks Anthony
> >
> >     Anthony Minessale a ?crit :
> >>     Meftah,
> >>
> >>     He is 100% correct.  Please do not insult my volunteer
> >>     developers. Without help from him you would not have any
> >>     FreeSWITCH right now so please drop this subject we are not using
> >>     pjsip.
> >>
> >>
> >>
> >>     On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb
> >>     <tayeb.meftah at gmail.com <mailto:tayeb.meftah at gmail.com>> wrote:
> >>
> >>         hi,
> >>         Pjsip support ICE, STUN and TURN!
> >>         to STKN:
> >>         if you don't pjsip, please stop talking or exit the discution
> >>         we want to kype Freeswitch Clean and universal
> >>
> >>         Stefan Knoblich a ?crit :
> >>>         Michael S Collins wrote:
> >>>>         I can guarantee that the FS devs are well aware of pj-sip. If
> >>>> it was/ is a viable alternative then it would be considered. The fact
> >>>> that it isn't being used is a pretty good indication that it isn't
> >>>> suitable for FS at this time.
> >>>>
> >>>>         -MV
> >>>>
> >>>>         Sent from my iPhone
> >>>
> >>>         We already mentioned some of the reasons why it did get
> >>>         dropped 3 years ago (first two points from memory, last two
> >>>         from old IRC logs): [License incompatible (GPL), but i think
> >>>         tony tried to negotiate on alternate license terms] Not
> >>>         possible to have multiple SIP profiles (due to global
> >>>         variables being used in the lib). A race-condition under
> >>>         high load, that couldn't be resolved back then (with the
> >>>         help of the pjsip developers). And the sofia module "just
> >>>         working" and surviving the scalability tests, so all efforts
> >>>         were focussed on mod_sofia and pjsip got dropped. stkn
> >>>         _______________________________________________
> >>>         FreeSWITCH-users mailing list
> >>>         FreeSWITCH-users at lists.freeswitch.org
> >>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>        
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> >>>s http://www.freeswitch.org
> >>>
> >>>
> >>>
> >>>         __________ Information from ESET NOD32 Antivirus, version of
> >>> virus signature database 4539 (20091024) __________
> >>>
> >>>         The message was checked by ESET NOD32 Antivirus.
> >>>
> >>>         http://www.eset.com
> >>
> >>         __________ Information from ESET NOD32 Antivirus, version of
> >>         virus signature database 4539 (20091024) __________
> >>
> >>         The message was checked by ESET NOD32 Antivirus.
> >>
> >>         http://www.eset.com
> >>
> >>         _______________________________________________
> >>         FreeSWITCH-users mailing list
> >>         FreeSWITCH-users at lists.freeswitch.org
> >>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>        
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> >>     --
> >>     Anthony Minessale II
> >>
> >>     FreeSWITCH http://www.freeswitch.org/
> >>     ClueCon http://www.cluecon.com/
> >>     Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >>     AIM: anthm
> >>     MSN:anthony_minessale at hotmail.com
> >>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
> >>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> >>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >>
> >>     FreeSWITCH Developer Conference
> >>     sip:888 at conference.freeswitch.org
> >>     <mailto:sip%3A888 at conference.freeswitch.org>
> >>     iax:guest at conference.freeswitch.org/888
> >>     <http://iax:guest at conference.freeswitch.org/888>
> >>     googletalk:conf+888 at conference.freeswitch.org
> >>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>     pstn:213-799-1400
> >>
> >>     _______________________________________________ FreeSWITCH-users
> >>     mailing list FreeSWITCH-users at lists.freeswitch.org
> >>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>    
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org __________ Information from ESET NOD32
> >> Antivirus, version of virus signature database 4539 (20091024)
> >> __________ The message was checked by ESET NOD32 Antivirus.
> >> http://www.eset.com
> >
> >     __________ Information from ESET NOD32 Antivirus, version of virus
> >     signature database 4539 (20091024) __________
> >
> >     The message was checked by ESET NOD32 Antivirus.
> >
> >     http://www.eset.com
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >    
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com
> > <mailto:MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> > <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org
> > <mailto:sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > <http://iax:guest at conference.freeswitch.org/888>
> > googletalk:conf+888 at conference.freeswitch.org
> > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:213-799-1400
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > __________ Information from ESET NOD32 Antivirus, version of virus
> > signature database 4539 (20091024) __________
> >
> > The message was checked by ESET NOD32 Antivirus.
> >
> > http://www.eset.com
>
> __________ Information from ESET NOD32 Antivirus, version of virus
> signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com





From dome at tel.co.th  Sat Oct 31 19:45:25 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Sun, 1 Nov 2009 09:45:25 +0700
Subject: [Freeswitch-users] Many CS_REPORTING state Zombie session
In-Reply-To: <191c3a030910311923t5f81019fyc4e7487cd61502b8@mail.gmail.com>
References: <8ccbff060910310408s37e5a45s45790e70235613f8@mail.gmail.com>
	<8ccbff060910310932n1710f601rcf063dac56f807a3@mail.gmail.com>
	<AD6346AF-A59E-4285-8954-0F4D94855B82@freeswitch.org>
	<8ccbff060910311005o35b011c8x75b0353f0e2ae0f7@mail.gmail.com>
	<EEDE7B6D-F722-4557-BBFE-AC0031C0BA86@freeswitch.org>
	<ebbc2d550910311018m7bc9b86bi120649c36dbf3413@mail.gmail.com>
	<8ccbff060910311034x53349625u711e3ab286b167f8@mail.gmail.com>
	<b39273530910311117k3b4bcfa5s98a5b720b1775e20@mail.gmail.com>
	<8ccbff060910311902u7434bc6m76caba40178ecc1e@mail.gmail.com>
	<191c3a030910311923t5f81019fyc4e7487cd61502b8@mail.gmail.com>
Message-ID: <8ccbff060910311945j5be2fa85j95e7e1b422dcc3a9@mail.gmail.com>

How to use bgapi in my flow.
1. user call did
2. FS send ringing and check balance, LCR from DB (by mod_odbc_quey)
3. Hangup (by use or timeout)
4. FS callback to user and bridge to IVR

I'm not sure  bgapi can do after channel hangup.

Dome C.


2009/11/1 Anthony Minessale <anthony.minessale at gmail.com>:
> Use bgapi originate ...
>
> On Oct 31, 2009 9:09 PM, "Dome Charoenyost" <dome at tel.co.th> wrote:
>
> Yes. i user api_hangup_hook for do callback.
> <action application="set" data="api_hangup_hook=originate
> {origination_caller_id_number=${caller_id_number},callback_uuid=${uuid},ignore_early_media=true}${route_1}|${route_2}|${route_3}
> call_ivr${caller_id_number} XML public "/>
>
> may be need originate_timeout
>
> Dome C.
>
>
> 2009/11/1 Rupa Schomaker <rupa at rupa.com>:
>
>> fscore_pb has been updated. > > Next time put in the pastebin url. ?This
>> one was 10911. > > Ok, a ...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From mike at jerris.com  Sat Oct 31 19:57:32 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 31 Oct 2009 22:57:32 -0400
Subject: [Freeswitch-users] Mod_pjsip
In-Reply-To: <4AEC9C40.502@gmail.com>
References: <4AEC9C40.502@gmail.com>
Message-ID: <BC20832A-CC71-471E-9E2A-62742334F3F2@jerris.com>

As the other reasons have already been covered about the history of  
mod_pjsip, I'll just make a few comments here:

If someone is really interested in resurecting mod_pjsip, the code is  
around here somewhere (although I can't find it on my quick look).   
Just let us know.

The licensing is not really an issue, that was resolved long ago.

There is no major thread safety issue in sofia.  There are some minor  
issues which we easily address in the module and an interest in thread  
pooling, which would increase performance.

pjsip and sofia always performed quite similar to each other, they are  
both quite good sip libraries

We had extensive thread safety and race condition issues in pjsip that  
never got resolved while we were using it.  It is quite possible they  
have been fixed sense, but many months were spent trying to fix them  
to no success.

We already support ice in the freeswitch rtp stack.  A sip stack  
supporting ice and turn does little for us and would be the worst way  
to implement it in freeswitch.  Sofia already supports sdp parsing for  
all of this, if you are interested in ICE/TURN support in freeswitch,  
it needs to be done in the RTP stack mostly, regardless of which sip  
stack.

Sofia has been shown to run well already in embedded environments.

Mike

On Oct 31, 2009, at 4:21 PM, Meftah Tayeb wrote:

> 		Hi my FreeSWITCH friends
> Today we talked about FreeSWITCH libraries and we found pjsip, which  
> is
> a lot better than sofia, it can do everything sofia can do and more,
> it's thread-safe and performs well, I would love to see a mod_pjsip.
> The library is GPL but they also offer alternative licensing, don't  
> get
> me wrong, Sofia is great but I think FreeSWITCH will benefit from a
> thread-safe SIP library.
> don't say that i need some stuf for free, but we need to make fs a  
> very
> also we love freeswitch to run in a Embedded PC like the blackfin
> CPU/boards, Pjsip is very good for Embedded SIP Devicesawesome Open
> Source project
> With much love,




From gromovd at gmail.com  Sat Oct 31 22:02:18 2009
From: gromovd at gmail.com (Dmitry Gromov)
Date: Sun, 1 Nov 2009 01:02:18 -0400
Subject: [Freeswitch-users] SIP Proxy with direct media path
Message-ID: <e21d97a50910312202s5082b8cch5f809da20fddb0ab@mail.gmail.com>

Hi!

I was a long time user of MySipSwitch service (www.mysipswitch.com) but it
looks like they are planning to close their service. I can't use the
replacement they have and would actually prefer some locally installed
solution.

Question - is it possible to use FreeSwitch exactly like MySipSwitch - just
SIP Proxy with very flexible dial plan?
With MySipSwitch it is possible to:
- register service with multiple SIP providers as client
- register your ATA with MySipSwitch
- create smart dial plan (in Ruby) to route calls based on prefixes and/or
call source and some other logic
- accept and route calls from all registered providers

MySipSwitch does not have any media capabilities - RTP is always established
directly between call parties (no re-Invites either).

If this is possible - is there any beginner guide for this?

Also - I know that FreeSwitch could run in Windows as well as in Linux -
would it be possible to use the above functionality in both OSes?

Thanks,
Dmitry

-- 
DG
NJ
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From mike at jerris.com  Sat Oct 31 22:15:26 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 1 Nov 2009 01:15:26 -0400
Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.5pre3 is
	available for testing!
In-Reply-To: <8cc991dd0910281740w2c7a9d10q93dd5b5f0b4dbd6@mail.gmail.com>
References: <87f2f3b90910281000r4c2021b8u7232320614737e30@mail.gmail.com>
	<8cc991dd0910281740w2c7a9d10q93dd5b5f0b4dbd6@mail.gmail.com>
Message-ID: <91BCC471-AA4C-4E5A-973C-8380E54DAB51@jerris.com>

We still do plan on branching 1.0 into bugfix only.  This has not yet  
happened but may happen at some point after 1.0.5.  In the mean time,  
the vast majority of the work lately has been fixes with small feature  
improvements, most all of this would stay in a 1.0 branch even if we  
were already branched.  Trunk remains for the most part very stable  
with new features mostly coming in new modules.

Mike

On Oct 28, 2009, at 8:40 PM, Craig Askings wrote:

> Is the plan to run 1.0.5 as a stable branch with bug fix updates or  
> will it be the case of just follow the trunk like it is with any  
> problems you encounter in 1.0.4?
>
> 2009/10/29 Michael Collins <msc at freeswitch.org>
> Hello FreeSWITCHers,
>
> The latest FreeSWITCH version is now available for download on the  
> files site. The announcement story is in the main FreeSWITCH site.  
> Please download and test, and then test some more. We need your  
> feedback. The sooner we get your feedback, the more quickly we can  
> roll the official 1.0.5.
>
> -- 
> Craig Askings
>
> Network Engineer | Over the Wire Pty Ltd
> craig at overthewire.com.au | www.overthewire.com.au
> Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From jason at jasonjgw.net  Sat Oct 31 22:21:19 2009
From: jason at jasonjgw.net (Jason White)
Date: Sun, 1 Nov 2009 16:21:19 +1100
Subject: [Freeswitch-users] SIP Proxy with direct media path
In-Reply-To: <e21d97a50910312202s5082b8cch5f809da20fddb0ab@mail.gmail.com>
References: <e21d97a50910312202s5082b8cch5f809da20fddb0ab@mail.gmail.com>
Message-ID: <20091101052119.GA28137@jdc.jasonjgw.net>

> With MySipSwitch it is possible to:
> - register service with multiple SIP providers as client

This is easy once you understand the FreeSWITCH XML configuration.
> - register your ATA with MySipSwitch

That's really a special case of the above.
> - create smart dial plan (in Ruby) to route calls based on prefixes and/or
> call source and some other logic

FreeSWITCH can do better than that: there are lots of variables which can be
matched by Perl-compatible regular expressions, including the destination
number.
> - accept and route calls from all registered providers

That's handled in the same way as the previous item.
> 
> MySipSwitch does not have any media capabilities - RTP is always established
> directly between call parties (no re-Invites either).

FreeSWITCH gives you a choice: you can use proxy media or bypass media modes
if you don't want it to handle the media.
> 
> If this is possible - is there any beginner guide for this?

http://wiki.freeswitch.org/ is the best there is.

It is being improved by the community over time.

You can also take advantage of the IRC channel, the FreeSWITCH conference and
of course the mailing list.




From msc at freeswitch.org  Sat Oct 31 22:24:59 2009
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 31 Oct 2009 22:24:59 -0700
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179
In-Reply-To: <509453DC-57BF-43D0-B704-FF0F2BB58EC0@jerris.com>
References: <mailman.1516.1256123625.2495.freeswitch-users@lists.freeswitch.org>
	<a60fed5e0910212108v4755203y693b8bdec5501f5f@mail.gmail.com>
	<87f2f3b90910212141w72b8f5a2oce032adb974d71e2@mail.gmail.com>
	<b39273530910220551q14d22150ldf8d4bff0b6fd011@mail.gmail.com>
	<87f2f3b90910221054l8abc3far9a18a66f0959f950@mail.gmail.com>
	<509453DC-57BF-43D0-B704-FF0F2BB58EC0@jerris.com>
Message-ID: <87f2f3b90910312224s6968f278i86e35469c781d60c@mail.gmail.com>

How would you do an expression like: if $x < 24 in a condition tag? Just
curious. I would like to make sure that is properly documented.
-MC

On Sat, Oct 31, 2009 at 10:42 AM, Michael Jerris <mike at jerris.com> wrote:

> You actually can use these in conditions.  Just need to be careful that the
> var you are conditioning on is already set.
>
> Mike
>
> On Oct 22, 2009, at 1:54 PM, Michael Collins wrote:
>
>
>
> On Thu, Oct 22, 2009 at 5:51 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>
>> cond would be helpful here?  I updated the wiki on this one just now
>> with a bit more detail.  It is a api call. so, you'd use it like:
>>
>> ${cond(eval ? trueval : falseval)}
>>
>> so to get a value of ERR if the var my myvar is > 15 you could:
>>
>> ${cond(${myvar} > 15 ? ERR : OK)}
>>
>> If both sides of the comparison operator are numeric then it does
>> numeric comparison otherwise it does lexical string comparison.
>>
>
> Rupa,
>
> Yes, you can do the set/cond API trick but you can only do it in the action
> or anti-action tags, not in the condition tags. I'm sure you know that but I
> want all those reading this thread to make the connection.
> -MC
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Sat Oct 31 22:33:18 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 1 Nov 2009 01:33:18 -0400
Subject: [Freeswitch-users] Setting up Conference with Moderator
In-Reply-To: <D7D48F51-0917-4D9A-B8B4-DE8F552BB254@gmail.com>
References: <3C04B27FC880044F8FCD735D0D952FF71701E84202@EXMBXCLUS01.citservers.local>
	<3C04B27FC880044F8FCD735D0D952FF71701E84338@EXMBXCLUS01.citservers.local>
	<E97CD3D0-52FF-44D1-BB89-23816045CE21@gmail.com>
	<71BBDC06-B669-4473-92DB-8B52713ACB23@freeswitch.org>,
	<114C4FF2-CA52-4C8A-81D2-16B4977E7B63@gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71701B6DCE6@EXMBXCLUS01.citservers.local>
	<3C04B27FC880044F8FCD735D0D952FF71702E7C7E5@EXMBXCLUS01.citservers.local>
	<F88E7B78-4CB8-45BD-98CE-81195862A891@gmail.com>
	<3C04B27FC880044F8FCD735D0D952FF71702E7C85F@EXMBXCLUS01.citservers.local>
	<D7D48F51-0917-4D9A-B8B4-DE8F552BB254@gmail.com>
Message-ID: <89D54263-7234-4F9A-8E22-40139F103DD3@jerris.com>

Have you answered the call?

On Oct 30, 2009, at 11:34 AM, Rob Forman wrote:

> Hm, strange.  I haven't seen that before.  Can you pastebin your logs
> at debug level?
>
> On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote:
>
>> It's strange... a tcpdump tells me that there is no DTMF from my
>> provider when using IVR, but when I call into a TN that goes
>> directly into the Conference App, I see DTMF from the provider.
>>




From mike at jerris.com  Sat Oct 31 22:34:38 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 1 Nov 2009 01:34:38 -0400
Subject: [Freeswitch-users] mod_t38gateway
In-Reply-To: <65d96fc80910301452x2a831733h92b05861d6c94123@mail.gmail.com>
References: <65d96fc80910301452x2a831733h92b05861d6c94123@mail.gmail.com>
Message-ID: <ECDA870C-7549-450A-A338-3BFFF600E592@jerris.com>

This is a non working module, just a shell for development.

Mike

On Oct 30, 2009, at 5:52 PM, Tihomir Culjaga wrote:

> does anybody know how does it work and how to use it in a dialplan?
>




From mike at jerris.com  Sat Oct 31 22:38:10 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 1 Nov 2009 01:38:10 -0400
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179
In-Reply-To: <87f2f3b90910312224s6968f278i86e35469c781d60c@mail.gmail.com>
References: <mailman.1516.1256123625.2495.freeswitch-users@lists.freeswitch.org>
	<a60fed5e0910212108v4755203y693b8bdec5501f5f@mail.gmail.com>
	<87f2f3b90910212141w72b8f5a2oce032adb974d71e2@mail.gmail.com>
	<b39273530910220551q14d22150ldf8d4bff0b6fd011@mail.gmail.com>
	<87f2f3b90910221054l8abc3far9a18a66f0959f950@mail.gmail.com>
	<509453DC-57BF-43D0-B704-FF0F2BB58EC0@jerris.com>
	<87f2f3b90910312224s6968f278i86e35469c781d60c@mail.gmail.com>
Message-ID: <AE75D709-C68C-458D-9A18-C6AB0E14C5F0@jerris.com>

see rupa's explanation below.


On Nov 1, 2009, at 1:24 AM, Michael Collins wrote:

>> How would you do an expression like: if $x < 24 in a condition tag?  
>> Just curious. I would like to make sure that is properly documented.
>> -MC
>>
>>
>>>
>>>
>>> On Thu, Oct 22, 2009 at 5:51 AM, Rupa Schomaker <rupa at rupa.com>  
>>> wrote:
>>> ${cond(${myvar} > 15 ? ERR : OK)}
>>>
>>> If both sides of the comparison operator are numeric then it does
>>> numeric comparison otherwise it does lexical string comparison.
>>

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